From 4dc69be22163bab880384858f30cb8cc76ad47f9 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 4 Jan 2011 20:16:07 +0530 Subject: ASoC: sst v2: Add sn95031 codec driver This patch adds the sn95031 asoc codec driver. This driver currently supports only playback. Capture and jack detection to be added later Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sn95031.c | 495 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sn95031.h | 99 +++++++++ 4 files changed, 600 insertions(+) create mode 100644 sound/soc/codecs/sn95031.c create mode 100644 sound/soc/codecs/sn95031.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 054191ed08e..fa42be529a7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -32,6 +32,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 + select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS @@ -173,6 +174,9 @@ config SND_SOC_MAX98088 config SND_SOC_PCM3008 tristate +config SND_SOC_SN95031 + tristate + config SND_SOC_SPDIF tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6a1e17bbbf8..76304d47891 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -18,6 +18,7 @@ snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-alc5623-objs := alc5623.o +snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-stac9766-objs := stac9766.o @@ -97,6 +98,7 @@ obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o +obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c new file mode 100644 index 00000000000..146b74467ae --- /dev/null +++ b/sound/soc/codecs/sn95031.c @@ -0,0 +1,495 @@ +/* + * sn95031.c - TI sn95031 Codec driver + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul + * Author: Harsha Priya + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * + */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include +#include +#include +#include +#include +#include +#include +#include +#include "sn95031.h" + +#define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100) +#define SN95031_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) + +/* + * todo: + * capture paths + * jack detection + * PM functions + */ + +static inline unsigned int sn95031_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 value = 0; + int ret; + + ret = intel_scu_ipc_ioread8(reg, &value); + if (ret) + pr_err("read of %x failed, err %d\n", reg, ret); + return value; + +} + +static inline int sn95031_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + int ret; + + ret = intel_scu_ipc_iowrite8(reg, value); + if (ret) + pr_err("write of %x failed, err %d\n", reg, ret); + return ret; +} + +static int sn95031_set_vaud_bias(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { + pr_debug("vaud_bias powering up pll\n"); + /* power up the pll */ + snd_soc_write(codec, SN95031_AUDPLLCTRL, BIT(5)); + /* enable pcm 2 */ + snd_soc_update_bits(codec, SN95031_PCM2C2, + BIT(0), BIT(0)); + } + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + pr_debug("vaud_bias power up rail\n"); + /* power up the rail */ + snd_soc_write(codec, SN95031_VAUD, + BIT(2)|BIT(1)|BIT(0)); + msleep(1); + } else if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) { + /* turn off pcm */ + pr_debug("vaud_bias power dn pcm\n"); + snd_soc_update_bits(codec, SN95031_PCM2C2, BIT(0), 0); + snd_soc_write(codec, SN95031_AUDPLLCTRL, 0); + } + break; + + + case SND_SOC_BIAS_OFF: + pr_debug("vaud_bias _OFF doing rail shutdown\n"); + snd_soc_write(codec, SN95031_VAUD, BIT(3)); + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +static int sn95031_vhs_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) { + pr_debug("VHS SND_SOC_DAPM_EVENT_ON doing rail startup now\n"); + /* power up the rail */ + snd_soc_write(w->codec, SN95031_VHSP, 0x3D); + snd_soc_write(w->codec, SN95031_VHSN, 0x3F); + msleep(1); + } else if (SND_SOC_DAPM_EVENT_OFF(event)) { + pr_debug("VHS SND_SOC_DAPM_EVENT_OFF doing rail shutdown\n"); + snd_soc_write(w->codec, SN95031_VHSP, 0xC4); + snd_soc_write(w->codec, SN95031_VHSN, 0x04); + } + return 0; +} + +static int sn95031_vihf_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) { + pr_debug("VIHF SND_SOC_DAPM_EVENT_ON doing rail startup now\n"); + /* power up the rail */ + snd_soc_write(w->codec, SN95031_VIHF, 0x27); + msleep(1); + } else if (SND_SOC_DAPM_EVENT_OFF(event)) { + pr_debug("VIHF SND_SOC_DAPM_EVENT_OFF doing rail shutdown\n"); + snd_soc_write(w->codec, SN95031_VIHF, 0x24); + } + return 0; +} + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { + + /* all end points mic, hs etc */ + SND_SOC_DAPM_OUTPUT("HPOUTL"), + SND_SOC_DAPM_OUTPUT("HPOUTR"), + SND_SOC_DAPM_OUTPUT("EPOUT"), + SND_SOC_DAPM_OUTPUT("IHFOUTL"), + SND_SOC_DAPM_OUTPUT("IHFOUTR"), + SND_SOC_DAPM_OUTPUT("LINEOUTL"), + SND_SOC_DAPM_OUTPUT("LINEOUTR"), + SND_SOC_DAPM_OUTPUT("VIB1OUT"), + SND_SOC_DAPM_OUTPUT("VIB2OUT"), + + SND_SOC_DAPM_SUPPLY("Headset Rail", SND_SOC_NOPM, 0, 0, + sn95031_vhs_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("Speaker Rail", SND_SOC_NOPM, 0, 0, + sn95031_vihf_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* playback path driver enables */ + SND_SOC_DAPM_PGA("Headset Left Playback", + SN95031_DRIVEREN, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headset Right Playback", + SN95031_DRIVEREN, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("Speaker Left Playback", + SN95031_DRIVEREN, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("Speaker Right Playback", + SN95031_DRIVEREN, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Vibra1 Playback", + SN95031_DRIVEREN, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Vibra2 Playback", + SN95031_DRIVEREN, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("Earpiece Playback", + SN95031_DRIVEREN, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Lineout Left Playback", + SN95031_LOCTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Lineout Right Playback", + SN95031_LOCTL, 4, 0, NULL, 0), + + /* playback path filter enable */ + SND_SOC_DAPM_PGA("Headset Left Filter", + SN95031_HSEPRXCTRL, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headset Right Filter", + SN95031_HSEPRXCTRL, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("Speaker Left Filter", + SN95031_IHFRXCTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Speaker Right Filter", + SN95031_IHFRXCTRL, 1, 0, NULL, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("HSDAC Left", "Headset", + SN95031_DACCONFIG, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset", + SN95031_DACCONFIG, 1, 0), + SND_SOC_DAPM_DAC("IHFDAC Left", "Speaker", + SN95031_DACCONFIG, 2, 0), + SND_SOC_DAPM_DAC("IHFDAC Right", "Speaker", + SN95031_DACCONFIG, 3, 0), + SND_SOC_DAPM_DAC("Vibra1 DAC", "Vibra1", + SN95031_VIB1C5, 1, 0), + SND_SOC_DAPM_DAC("Vibra2 DAC", "Vibra2", + SN95031_VIB2C5, 1, 0), +}; + +static const struct snd_soc_dapm_route sn95031_audio_map[] = { + /* headset and earpiece map */ + { "HPOUTL", NULL, "Headset Left Playback" }, + { "HPOUTR", NULL, "Headset Right Playback" }, + { "EPOUT", NULL, "Earpiece Playback" }, + { "Headset Left Playback", NULL, "Headset Left Filter"}, + { "Headset Right Playback", NULL, "Headset Right Filter"}, + { "Earpiece Playback", NULL, "Headset Left Filter"}, + { "Headset Left Filter", NULL, "HSDAC Left"}, + { "Headset Right Filter", NULL, "HSDAC Right"}, + { "HSDAC Left", NULL, "Headset Rail"}, + { "HSDAC Right", NULL, "Headset Rail"}, + + /* speaker map */ + { "IHFOUTL", "NULL", "Speaker Left Playback"}, + { "IHFOUTR", "NULL", "Speaker Right Playback"}, + { "Speaker Left Playback", NULL, "Speaker Left Filter"}, + { "Speaker Right Playback", NULL, "Speaker Right Filter"}, + { "Speaker Left Filter", NULL, "IHFDAC Left"}, + { "Speaker Right Filter", NULL, "IHFDAC Right"}, + { "IHFDAC Left", NULL, "Speaker Rail"}, + { "IHFDAC Right", NULL, "Speaker Rail"}, + + /* vibra map */ + {"VIB1OUT", NULL, "Vibra1 Playback"}, + {"Vibra1 Playback", NULL, "Vibra1 DAC"}, + + {"VIB2OUT", NULL, "Vibra2 Playback"}, + {"Vibra2 Playback", NULL, "Vibra2 DAC"}, + + /* lineout */ + {"LINEOUTL", NULL, "Lineout Left Playback"}, + {"LINEOUTR", NULL, "Lineout Right Playback"}, + {"Lineout Left Playback", NULL, "Headset Left Filter"}, + {"Lineout Left Playback", NULL, "Speaker Left Filter"}, + {"Lineout Left Playback", NULL, "Vibra1 DAC"}, + {"Lineout Right Playback", NULL, "Headset Right Filter"}, + {"Lineout Right Playback", NULL, "Speaker Right Filter"}, + {"Lineout Right Playback", NULL, "Vibra2 DAC"}, +}; + +/* speaker and headset mutes, for audio pops and clicks */ +static int sn95031_pcm_hs_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, + SN95031_HSLVOLCTRL, BIT(7), (!mute << 7)); + snd_soc_update_bits(dai->codec, + SN95031_HSRVOLCTRL, BIT(7), (!mute << 7)); + return 0; +} + +static int sn95031_pcm_spkr_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, + SN95031_IHFLVOLCTRL, BIT(7), (!mute << 7)); + snd_soc_update_bits(dai->codec, + SN95031_IHFRVOLCTRL, BIT(7), (!mute << 7)); + return 0; +} + +int sn95031_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + unsigned int format, rate; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + format = BIT(4)|BIT(5); + break; + + case SNDRV_PCM_FORMAT_S24_LE: + format = 0; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(dai->codec, SN95031_PCM2C2, + BIT(4)|BIT(5), format); + + switch (params_rate(params)) { + case 48000: + pr_debug("RATE_48000\n"); + rate = 0; + break; + + case 44100: + pr_debug("RATE_44100\n"); + rate = BIT(7); + break; + + default: + pr_err("ERR rate %d\n", params_rate(params)); + return -EINVAL; + } + snd_soc_update_bits(dai->codec, SN95031_PCM1C1, BIT(7), rate); + + return 0; +} + +/* Codec DAI section */ +static struct snd_soc_dai_ops sn95031_headset_dai_ops = { + .digital_mute = sn95031_pcm_hs_mute, + .hw_params = sn95031_pcm_hw_params, +}; + +static struct snd_soc_dai_ops sn95031_speaker_dai_ops = { + .digital_mute = sn95031_pcm_spkr_mute, + .hw_params = sn95031_pcm_hw_params, +}; + +static struct snd_soc_dai_ops sn95031_vib1_dai_ops = { + .hw_params = sn95031_pcm_hw_params, +}; + +static struct snd_soc_dai_ops sn95031_vib2_dai_ops = { + .hw_params = sn95031_pcm_hw_params, +}; + +struct snd_soc_dai_driver sn95031_dais[] = { +{ + .name = "SN95031 Headset", + .playback = { + .stream_name = "Headset", + .channels_min = 2, + .channels_max = 2, + .rates = SN95031_RATES, + .formats = SN95031_FORMATS, + }, + .ops = &sn95031_headset_dai_ops, +}, +{ .name = "SN95031 Speaker", + .playback = { + .stream_name = "Speaker", + .channels_min = 2, + .channels_max = 2, + .rates = SN95031_RATES, + .formats = SN95031_FORMATS, + }, + .ops = &sn95031_speaker_dai_ops, +}, +{ .name = "SN95031 Vibra1", + .playback = { + .stream_name = "Vibra1", + .channels_min = 1, + .channels_max = 1, + .rates = SN95031_RATES, + .formats = SN95031_FORMATS, + }, + .ops = &sn95031_vib1_dai_ops, +}, +{ .name = "SN95031 Vibra2", + .playback = { + .stream_name = "Vibra2", + .channels_min = 1, + .channels_max = 1, + .rates = SN95031_RATES, + .formats = SN95031_FORMATS, + }, + .ops = &sn95031_vib2_dai_ops, +}, +}; + +/* codec registration */ +static int sn95031_codec_probe(struct snd_soc_codec *codec) +{ + int ret; + + pr_debug("codec_probe called\n"); + + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + codec->dapm.idle_bias_off = 1; + + /* PCM interface config + * This sets the pcm rx slot conguration to max 6 slots + * for max 4 dais (2 stereo and 2 mono) + */ + snd_soc_write(codec, SN95031_PCM2RXSLOT01, 0x10); + snd_soc_write(codec, SN95031_PCM2RXSLOT23, 0x32); + snd_soc_write(codec, SN95031_PCM2RXSLOT45, 0x54); + /* pcm port setting + * This sets the pcm port to slave and clock at 19.2Mhz which + * can support 6slots, sampling rate set per stream in hw-params + */ + snd_soc_write(codec, SN95031_PCM1C1, 0x00); + snd_soc_write(codec, SN95031_PCM2C1, 0x01); + snd_soc_write(codec, SN95031_PCM2C2, 0x0A); + snd_soc_write(codec, SN95031_HSMIXER, BIT(0)|BIT(4)); + /* vendor vibra workround, the vibras are muted by + * custom register so unmute them + */ + snd_soc_write(codec, SN95031_SSR5, 0x80); + snd_soc_write(codec, SN95031_SSR6, 0x80); + snd_soc_write(codec, SN95031_VIB1C5, 0x00); + snd_soc_write(codec, SN95031_VIB2C5, 0x00); + /* configure vibras for pcm port */ + snd_soc_write(codec, SN95031_VIB1C3, 0x00); + snd_soc_write(codec, SN95031_VIB2C3, 0x00); + + /* soft mute ramp time */ + snd_soc_write(codec, SN95031_SOFTMUTE, 0x3); + /* fix the initial volume at 1dB, + * default in +9dB, + * 1dB give optimal swing on DAC, amps + */ + snd_soc_write(codec, SN95031_HSLVOLCTRL, 0x08); + snd_soc_write(codec, SN95031_HSRVOLCTRL, 0x08); + snd_soc_write(codec, SN95031_IHFLVOLCTRL, 0x08); + snd_soc_write(codec, SN95031_IHFRVOLCTRL, 0x08); + /* dac mode and lineout workaround */ + snd_soc_write(codec, SN95031_SSR2, 0x10); + snd_soc_write(codec, SN95031_SSR3, 0x40); + + ret = snd_soc_dapm_new_controls(&codec->dapm, sn95031_dapm_widgets, + ARRAY_SIZE(sn95031_dapm_widgets)); + if (ret) + pr_err("soc_dapm_new_control failed %d", ret); + ret = snd_soc_dapm_add_routes(&codec->dapm, sn95031_audio_map, + ARRAY_SIZE(sn95031_audio_map)); + if (ret) + pr_err("soc_dapm_add_routes failed %d", ret); + + return ret; +} + +static int sn95031_codec_remove(struct snd_soc_codec *codec) +{ + pr_debug("codec_remove called\n"); + sn95031_set_vaud_bias(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +struct snd_soc_codec_driver sn95031_codec = { + .probe = sn95031_codec_probe, + .remove = sn95031_codec_remove, + .read = sn95031_read, + .write = sn95031_write, + .set_bias_level = sn95031_set_vaud_bias, +}; + +static int __devinit sn95031_device_probe(struct platform_device *pdev) +{ + pr_debug("codec device probe called for %s\n", dev_name(&pdev->dev)); + return snd_soc_register_codec(&pdev->dev, &sn95031_codec, + sn95031_dais, ARRAY_SIZE(sn95031_dais)); +} + +static int __devexit sn95031_device_remove(struct platform_device *pdev) +{ + pr_debug("codec device remove called\n"); + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver sn95031_codec_driver = { + .driver = { + .name = "sn95031", + .owner = THIS_MODULE, + }, + .probe = sn95031_device_probe, + .remove = sn95031_device_remove, +}; + +static int __init sn95031_init(void) +{ + pr_debug("driver init called\n"); + return platform_driver_register(&sn95031_codec_driver); +} +module_init(sn95031_init); + +static void __exit sn95031_exit(void) +{ + pr_debug("driver exit called\n"); + platform_driver_unregister(&sn95031_codec_driver); +} +module_exit(sn95031_exit); + +MODULE_DESCRIPTION("ASoC Intel(R) SN95031 codec driver"); +MODULE_AUTHOR("Vinod Koul "); +MODULE_AUTHOR("Harsha Priya "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:sn95031"); diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h new file mode 100644 index 00000000000..b17a39b2aef --- /dev/null +++ b/sound/soc/codecs/sn95031.h @@ -0,0 +1,99 @@ +/* + * sn95031.h - TI sn95031 Codec driver + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul + * Author: Harsha Priya + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * + */ +#ifndef _SN95031_H +#define _SN95031_H + +/*register map*/ +#define SN95031_VAUD 0xDB +#define SN95031_VHSP 0xDC +#define SN95031_VHSN 0xDD +#define SN95031_VIHF 0xC9 + +#define SN95031_AUDPLLCTRL 0x240 +#define SN95031_DMICBUF0123 0x241 +#define SN95031_DMICBUF45 0x242 +#define SN95031_DMICGPO 0x244 +#define SN95031_DMICMUX 0x245 +#define SN95031_DMICLK 0x246 +#define SN95031_MICBIAS 0x247 +#define SN95031_ADCCONFIG 0x248 +#define SN95031_MICAMP1 0x249 +#define SN95031_MICAMP2 0x24A +#define SN95031_NOISEMUX 0x24B +#define SN95031_AUDIOMUX12 0x24C +#define SN95031_AUDIOMUX34 0x24D +#define SN95031_AUDIOSINC 0x24E +#define SN95031_AUDIOTXEN 0x24F +#define SN95031_HSEPRXCTRL 0x250 +#define SN95031_IHFRXCTRL 0x251 +#define SN95031_HSMIXER 0x256 +#define SN95031_DACCONFIG 0x257 +#define SN95031_SOFTMUTE 0x258 +#define SN95031_HSLVOLCTRL 0x259 +#define SN95031_HSRVOLCTRL 0x25A +#define SN95031_IHFLVOLCTRL 0x25B +#define SN95031_IHFRVOLCTRL 0x25C +#define SN95031_DRIVEREN 0x25D +#define SN95031_LOCTL 0x25E +#define SN95031_VIB1C1 0x25F +#define SN95031_VIB1C2 0x260 +#define SN95031_VIB1C3 0x261 +#define SN95031_VIB1SPIPCM1 0x262 +#define SN95031_VIB1SPIPCM2 0x263 +#define SN95031_VIB1C5 0x264 +#define SN95031_VIB2C1 0x265 +#define SN95031_VIB2C2 0x266 +#define SN95031_VIB2C3 0x267 +#define SN95031_VIB2SPIPCM1 0x268 +#define SN95031_VIB2SPIPCM2 0x269 +#define SN95031_VIB2C5 0x26A +#define SN95031_BTNCTRL1 0x26B +#define SN95031_BTNCTRL2 0x26C +#define SN95031_PCM1TXSLOT01 0x26D +#define SN95031_PCM1TXSLOT23 0x26E +#define SN95031_PCM1TXSLOT45 0x26F +#define SN95031_PCM1RXSLOT0_3 0x270 +#define SN95031_PCM1RXSLOT45 0x271 +#define SN95031_PCM2TXSLOT01 0x272 +#define SN95031_PCM2TXSLOT23 0x273 +#define SN95031_PCM2TXSLOT45 0x274 +#define SN95031_PCM2RXSLOT01 0x275 +#define SN95031_PCM2RXSLOT23 0x276 +#define SN95031_PCM2RXSLOT45 0x277 +#define SN95031_PCM1C1 0x278 +#define SN95031_PCM1C2 0x279 +#define SN95031_PCM1C3 0x27A +#define SN95031_PCM2C1 0x27B +#define SN95031_PCM2C2 0x27C +/*end codec register defn*/ + +/*vendor defn these are not part of avp*/ +#define SN95031_SSR2 0x381 +#define SN95031_SSR3 0x382 +#define SN95031_SSR5 0x384 +#define SN95031_SSR6 0x385 + +#endif -- cgit v1.2.3 From d927fdae5cb2ca36f0c5b61e528078e8c1261607 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 4 Jan 2011 20:16:32 +0530 Subject: ASoC: sst v2: Add mid platform driver This patch adds the platform driver for mid asoc drivers. This platfrom driver sends commands to sst dsp engine driver for the dai operations. For this purpose it depends on intel_sst driver which is currently in staging tree Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/mid-x86/sst_platform.c | 484 +++++++++++++++++++++++++++++++++++++++ sound/soc/mid-x86/sst_platform.h | 63 +++++ 2 files changed, 547 insertions(+) create mode 100644 sound/soc/mid-x86/sst_platform.c create mode 100644 sound/soc/mid-x86/sst_platform.h (limited to 'sound') diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c new file mode 100644 index 00000000000..189d546fbf9 --- /dev/null +++ b/sound/soc/mid-x86/sst_platform.c @@ -0,0 +1,484 @@ +/* + * sst_platform.c - Intel MID Platform driver + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul + * Author: Harsha Priya + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * + */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include +#include +#include +#include +#include +#include +#include "../../../drivers/staging/intel_sst/intel_sst_ioctl.h" +#include "../../../drivers/staging/intel_sst/intel_sst.h" +#include "sst_platform.h" + +static struct snd_pcm_hardware sst_platform_pcm_hw = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_DOUBLE | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_MMAP| + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_U16 | + SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_U24 | + SNDRV_PCM_FMTBIT_S32 | SNDRV_PCM_FMTBIT_U32), + .rates = (SNDRV_PCM_RATE_8000| + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000), + .rate_min = SST_MIN_RATE, + .rate_max = SST_MAX_RATE, + .channels_min = SST_MIN_CHANNEL, + .channels_max = SST_MAX_CHANNEL, + .buffer_bytes_max = SST_MAX_BUFFER, + .period_bytes_min = SST_MIN_PERIOD_BYTES, + .period_bytes_max = SST_MAX_PERIOD_BYTES, + .periods_min = SST_MIN_PERIODS, + .periods_max = SST_MAX_PERIODS, + .fifo_size = SST_FIFO_SIZE, +}; + +/* MFLD - MSIC */ +struct snd_soc_dai_driver sst_platform_dai[] = { +{ + .name = "Headset-cpu-dai", + .id = 0, + .playback = { + .channels_min = SST_STEREO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "Speaker-cpu-dai", + .id = 1, + .playback = { + .channels_min = SST_MONO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "Vibra1-cpu-dai", + .id = 2, + .playback = { + .channels_min = SST_MONO, + .channels_max = SST_MONO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +{ + .name = "Vibra2-cpu-dai", + .id = 3, + .playback = { + .channels_min = SST_MONO, + .channels_max = SST_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, +}, +}; +/* helper functions */ +static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) +{ + struct sst_runtime_stream *stream = + substream->runtime->private_data; + struct snd_sst_stream_params param = {{{0,},},}; + struct snd_sst_params str_params = {0}; + int ret_val; + + /* set codec params and inform SST driver the same */ + + param.uc.pcm_params.codec = SST_CODEC_TYPE_PCM; + param.uc.pcm_params.num_chan = (u8) substream->runtime->channels; + param.uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits; + param.uc.pcm_params.reserved = 0; + param.uc.pcm_params.sfreq = substream->runtime->rate; + param.uc.pcm_params.ring_buffer_size = + snd_pcm_lib_buffer_bytes(substream); + param.uc.pcm_params.period_count = substream->runtime->period_size; + param.uc.pcm_params.ring_buffer_addr = + virt_to_phys(substream->dma_buffer.area); + substream->runtime->dma_area = substream->dma_buffer.area; + + pr_debug("period_cnt = %d\n", param.uc.pcm_params.period_count); + pr_debug("sfreq= %d, wd_sz = %d\n", + param.uc.pcm_params.sfreq, param.uc.pcm_params.pcm_wd_sz); + + str_params.sparams = param; + str_params.codec = SST_CODEC_TYPE_PCM; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + str_params.ops = STREAM_OPS_PLAYBACK; + str_params.device_type = substream->pcm->device + 1; + pr_debug("Playbck stream,Device %d\n", + substream->pcm->device); + } else { + str_params.ops = STREAM_OPS_CAPTURE; + str_params.device_type = SND_SST_DEVICE_CAPTURE; + pr_debug("Capture stream,Device %d\n", + substream->pcm->device); + } + ret_val = stream->sstdrv_ops->control_set(SST_SND_ALLOC, &str_params); + pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val); + if (ret_val < 0) + return ret_val; + + stream->stream_info.str_id = ret_val; + pr_debug("str id : %d\n", stream->stream_info.str_id); + + return ret_val; +} + + +static void sst_period_elapsed(void *mad_substream) +{ + struct snd_pcm_substream *substream = mad_substream; + struct sst_runtime_stream *stream; + + if (!substream || !substream->runtime) + return; + stream = substream->runtime->private_data; + if (!stream) + return; + + spin_lock(&stream->status_lock); + if (stream->stream_status != SST_PLATFORM_RUNNING) { + spin_unlock(&stream->status_lock); + return; + } + spin_unlock(&stream->status_lock); + snd_pcm_period_elapsed(substream); + return; +} + +static int sst_platform_init_stream(struct snd_pcm_substream *substream) +{ + struct sst_runtime_stream *stream = + substream->runtime->private_data; + int ret_val; + + pr_debug("setting buffer ptr param\n"); + spin_lock(&stream->status_lock); + stream->stream_status = SST_PLATFORM_INIT; + spin_unlock(&stream->status_lock); + stream->stream_info.period_elapsed = sst_period_elapsed; + stream->stream_info.mad_substream = substream; + stream->stream_info.buffer_ptr = 0; + stream->stream_info.sfreq = substream->runtime->rate; + ret_val = stream->sstdrv_ops->control_set(SST_SND_STREAM_INIT, + &stream->stream_info); + if (ret_val) + pr_err("control_set ret error %d\n", ret_val); + return ret_val; + +} +/* end -- helper functions */ + +static int sst_platform_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime; + struct sst_runtime_stream *stream; + int ret_val = 0; + + pr_debug("sst_platform_open called\n"); + + runtime = substream->runtime; + runtime->hw = sst_platform_pcm_hw; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + spin_lock_init(&stream->status_lock); + stream->stream_info.str_id = 0; + + spin_lock(&stream->status_lock); + stream->stream_status = SST_PLATFORM_INIT; + spin_unlock(&stream->status_lock); + + stream->stream_info.mad_substream = substream; + /* allocate memory for SST API set */ + stream->sstdrv_ops = kzalloc(sizeof(*stream->sstdrv_ops), + GFP_KERNEL); + if (!stream->sstdrv_ops) { + pr_err("sst: mem allocation for ops fail\n"); + kfree(stream); + return -ENOMEM; + } + stream->sstdrv_ops->vendor_id = MSIC_VENDOR_ID; + + /* registering with SST driver to get access to SST APIs to use */ + ret_val = register_sst_card(stream->sstdrv_ops); + if (ret_val) { + pr_err("sst: sst card registration failed\n"); + return ret_val; + } + runtime->private_data = stream; + return snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); +} + +static int sst_platform_close(struct snd_pcm_substream *substream) +{ + struct sst_runtime_stream *stream; + int ret_val = 0, str_id; + + pr_debug("sst_platform_close called\n"); + + stream = substream->runtime->private_data; + str_id = stream->stream_info.str_id; + + if (str_id) + ret_val = stream->sstdrv_ops->control_set( + SST_SND_FREE, &str_id); + + kfree(stream->sstdrv_ops); + kfree(stream); + return ret_val; +} + +static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct sst_runtime_stream *stream; + int ret_val = 0, str_id; + + pr_debug("sst_platform_pcm_prepare called\n"); + + stream = substream->runtime->private_data; + str_id = stream->stream_info.str_id; + if (stream->stream_info.str_id) { + ret_val = stream->sstdrv_ops->control_set( + SST_SND_DROP, &str_id); + return ret_val; + } + + ret_val = sst_platform_alloc_stream(substream); + if (ret_val < 0) + return ret_val; + snprintf(substream->pcm->id, sizeof(substream->pcm->id), + "%d", stream->stream_info.str_id); + + ret_val = sst_platform_init_stream(substream); + if (ret_val) + return ret_val; + substream->runtime->hw.info = SNDRV_PCM_INFO_BLOCK_TRANSFER; + return ret_val; +} + +static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + int ret_val = 0, str_id; + struct sst_runtime_stream *stream; + + pr_debug("sst_platform_pcm_trigger called\n"); + + stream = substream->runtime->private_data; + + str_id = stream->stream_info.str_id; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + pr_debug("sst: Trigger Start\n"); + ret_val = stream->sstdrv_ops->control_set( + SST_SND_START, &str_id); + if (ret_val) + break; + spin_lock(&stream->status_lock); + stream->stream_status = SST_PLATFORM_RUNNING; + spin_unlock(&stream->status_lock); + stream->stream_info.mad_substream = substream; + break; + case SNDRV_PCM_TRIGGER_STOP: + pr_debug("sst: in stop\n"); + ret_val = stream->sstdrv_ops->control_set( + SST_SND_DROP, &str_id); + if (ret_val) + break; + spin_lock(&stream->status_lock); + stream->stream_status = SST_PLATFORM_DROPPED; + spin_unlock(&stream->status_lock); + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + pr_debug("sst: in pause\n"); + ret_val = stream->sstdrv_ops->control_set( + SST_SND_PAUSE, &str_id); + if (ret_val) + break; + spin_lock(&stream->status_lock); + stream->stream_status = SST_PLATFORM_PAUSED; + spin_unlock(&stream->status_lock); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + pr_debug("sst: in pause release\n"); + ret_val = stream->sstdrv_ops->control_set( + SST_SND_RESUME, &str_id); + if (ret_val) + break; + spin_lock(&stream->status_lock); + stream->stream_status = SST_PLATFORM_RUNNING; + spin_unlock(&stream->status_lock); + break; + default: + ret_val = -EINVAL; + } + return ret_val; +} + + +static snd_pcm_uframes_t sst_platform_pcm_pointer + (struct snd_pcm_substream *substream) +{ + struct sst_runtime_stream *stream; + int ret_val; + struct pcm_stream_info *str_info; + + + stream = substream->runtime->private_data; + spin_lock(&stream->status_lock); + if (stream->stream_status == SST_PLATFORM_INIT) { + spin_unlock(&stream->status_lock); + return 0; + } + spin_unlock(&stream->status_lock); + + str_info = &stream->stream_info; + ret_val = stream->sstdrv_ops->control_set( + SST_SND_BUFFER_POINTER, str_info); + if (ret_val) { + pr_err("sst: error code = %d\n", ret_val); + return ret_val; + } + + return stream->stream_info.buffer_ptr; +} + + +static struct snd_pcm_ops sst_platform_ops = { + .open = sst_platform_open, + .close = sst_platform_close, + .ioctl = snd_pcm_lib_ioctl, + .prepare = sst_platform_pcm_prepare, + .trigger = sst_platform_pcm_trigger, + .pointer = sst_platform_pcm_pointer, +}; + +static void sst_pcm_free(struct snd_pcm *pcm) +{ + pr_debug("sst_pcm_free called\n"); + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +int sst_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + int retval = 0; + + pr_debug("sst_pcm_new called\n"); + + if (dai->driver->playback.channels_min || + dai->driver->capture.channels_min) { + retval = snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + SST_MIN_BUFFER, SST_MAX_BUFFER); + if (retval) { + pr_err("dma buffer allocationf fail\n"); + return retval; + } + } + + return retval; +} +struct snd_soc_platform_driver sst_soc_platform_drv = { + .ops = &sst_platform_ops, + .pcm_new = sst_pcm_new, + .pcm_free = sst_pcm_free, +}; + +static int sst_platform_probe(struct platform_device *pdev) +{ + int ret; + + pr_debug("sst_platform_probe called\n"); + ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv); + if (ret) { + pr_err("registering soc platform failed\n"); + return ret; + } + ret = snd_soc_register_dais(&pdev->dev, + sst_platform_dai, ARRAY_SIZE(sst_platform_dai)); + if (ret) { + pr_err("registering cpu dais failed\n"); + snd_soc_unregister_platform(&pdev->dev); + } + return ret; +} + +static int sst_platform_remove(struct platform_device *pdev) +{ + + snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sst_platform_dai)); + + snd_soc_unregister_platform(&pdev->dev); + pr_debug("sst_platform_remove sucess\n"); + + return 0; +} + +static struct platform_driver sst_platform_driver = { + .driver = { + .name = "sst-platform", + .owner = THIS_MODULE, + }, + .probe = sst_platform_probe, + .remove = sst_platform_remove, +}; + +static int __init sst_soc_platform_init(void) +{ + pr_debug("sst_soc_platform_init called\n"); + return platform_driver_register(&sst_platform_driver); +} +module_init(sst_soc_platform_init); + +static void __exit sst_soc_platform_exit(void) +{ + platform_driver_unregister(&sst_platform_driver); + pr_debug("sst_soc_platform_exit sucess\n"); +} +module_exit(sst_soc_platform_exit); + +MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver"); +MODULE_AUTHOR("Vinod Koul "); +MODULE_AUTHOR("Harsha Priya "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platfrom: sst-platform"); diff --git a/sound/soc/mid-x86/sst_platform.h b/sound/soc/mid-x86/sst_platform.h new file mode 100644 index 00000000000..df370286694 --- /dev/null +++ b/sound/soc/mid-x86/sst_platform.h @@ -0,0 +1,63 @@ +/* + * sst_platform.h - Intel MID Platform driver header file + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul + * Author: Harsha Priya + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * + */ + +#ifndef __SST_PLATFORMDRV_H__ +#define __SST_PLATFORMDRV_H__ + +#define SST_MONO 1 +#define SST_STEREO 2 +#define SST_MAX_CAP 5 + +#define SST_MIN_RATE 8000 +#define SST_MAX_RATE 48000 +#define SST_MIN_CHANNEL 1 +#define SST_MAX_CHANNEL 5 +#define SST_MAX_BUFFER (800*1024) +#define SST_MIN_BUFFER (800*1024) +#define SST_MIN_PERIOD_BYTES 32 +#define SST_MAX_PERIOD_BYTES SST_MAX_BUFFER +#define SST_MIN_PERIODS 2 +#define SST_MAX_PERIODS (1024*2) +#define SST_FIFO_SIZE 0 +#define SST_CARD_NAMES "intel_mid_card" +#define MSIC_VENDOR_ID 3 + +struct sst_runtime_stream { + int stream_status; + struct pcm_stream_info stream_info; + struct intel_sst_card_ops *sstdrv_ops; + spinlock_t status_lock; +}; + +enum sst_drv_status { + SST_PLATFORM_INIT = 1, + SST_PLATFORM_STARTED, + SST_PLATFORM_RUNNING, + SST_PLATFORM_PAUSED, + SST_PLATFORM_DROPPED, +}; + +#endif -- cgit v1.2.3 From 55c720369dbe1dd558b87478e3448df837fbe7a3 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 4 Jan 2011 20:16:50 +0530 Subject: ASoC sst v2: Add medfield machine driver This patch adds the medfield machine driver Machine driver glues the sn95031 codec driver and sst_platform driver to form the asoc sound driver Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/mid-x86/mfld_machine.c | 296 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 296 insertions(+) create mode 100644 sound/soc/mid-x86/mfld_machine.c (limited to 'sound') diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c new file mode 100644 index 00000000000..1a330be1a01 --- /dev/null +++ b/sound/soc/mid-x86/mfld_machine.c @@ -0,0 +1,296 @@ +/* + * mfld_machine.c - ASoc Machine driver for Intel Medfield MID platform + * + * Copyright (C) 2010 Intel Corp + * Author: Vinod Koul + * Author: Harsha Priya + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include +#include +#include +#include +#include +#include +#include "../codecs/sn95031.h" + +#define MID_MONO 1 +#define MID_STEREO 2 +#define MID_MAX_CAP 5 + +static unsigned int hs_switch; +static unsigned int lo_dac; + +/* sound card controls */ +static const char *headset_switch_text[] = {"Earpiece", "Headset"}; + +static const char *lo_text[] = {"Vibra", "Headset", "IHF", "None"}; + +static const struct soc_enum headset_enum = + SOC_ENUM_SINGLE_EXT(2, headset_switch_text); + +static const struct soc_enum lo_enum = + SOC_ENUM_SINGLE_EXT(4, lo_text); + +static int headset_get_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = hs_switch; + return 0; +} + +static int headset_set_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0] == hs_switch) + return 0; + + if (ucontrol->value.integer.value[0]) { + pr_debug("hs_set HS path\n"); + snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTL"); + snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); + } else { + pr_debug("hs_set EP path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); + snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); + } + snd_soc_dapm_sync(&codec->dapm); + hs_switch = ucontrol->value.integer.value[0]; + + return 0; +} + +static void lo_enable_out_pins(struct snd_soc_codec *codec) +{ + snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTL"); + snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTL"); + snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT"); + snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT"); + if (hs_switch) { + snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTL"); + snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); + } else { + snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); + snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); + } +} + +static int lo_get_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = lo_dac; + return 0; +} + +static int lo_set_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0] == lo_dac) + return 0; + + /* we dont want to work with last state of lineout so just enable all + * pins and then disable pins not required + */ + lo_enable_out_pins(codec); + switch (ucontrol->value.integer.value[0]) { + case 0: + pr_debug("set vibra path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "VIB1OUT"); + snd_soc_dapm_disable_pin(&codec->dapm, "VIB2OUT"); + snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0); + break; + + case 1: + pr_debug("set hs path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); + snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); + snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22); + break; + + case 2: + pr_debug("set spkr path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTL"); + snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTR"); + snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x44); + break; + + case 3: + pr_debug("set null path\n"); + snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTL"); + snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTR"); + snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x66); + break; + } + snd_soc_dapm_sync(&codec->dapm); + lo_dac = ucontrol->value.integer.value[0]; + return 0; +} + +static const struct snd_kcontrol_new mfld_snd_controls[] = { + SOC_ENUM_EXT("Playback Switch", headset_enum, + headset_get_switch, headset_set_switch), + SOC_ENUM_EXT("Lineout Mux", lo_enum, + lo_get_switch, lo_set_switch), +}; + +static int mfld_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret_val; + + ret_val = snd_soc_add_controls(codec, mfld_snd_controls, + ARRAY_SIZE(mfld_snd_controls)); + if (ret_val) { + pr_err("soc_add_controls failed %d", ret_val); + return ret_val; + } + /* default is earpiece pin, userspace sets it explcitly */ + snd_soc_dapm_disable_pin(dapm, "HPOUTL"); + snd_soc_dapm_disable_pin(dapm, "HPOUTR"); + /* default is lineout NC, userspace sets it explcitly */ + snd_soc_dapm_disable_pin(dapm, "LINEOUTL"); + snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); + lo_dac = 3; + hs_switch = 0; + return snd_soc_dapm_sync(dapm); +} + +struct snd_soc_dai_link mfld_msic_dailink[] = { + { + .name = "Medfield Headset", + .stream_name = "Headset", + .cpu_dai_name = "Headset-cpu-dai", + .codec_dai_name = "SN95031 Headset", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = mfld_init, + }, + { + .name = "Medfield Speaker", + .stream_name = "Speaker", + .cpu_dai_name = "Speaker-cpu-dai", + .codec_dai_name = "SN95031 Speaker", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = NULL, + }, + { + .name = "Medfield Vibra", + .stream_name = "Vibra1", + .cpu_dai_name = "Vibra1-cpu-dai", + .codec_dai_name = "SN95031 Vibra1", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = NULL, + }, + { + .name = "Medfield Haptics", + .stream_name = "Vibra2", + .cpu_dai_name = "Vibra2-cpu-dai", + .codec_dai_name = "SN95031 Vibra2", + .codec_name = "sn95031", + .platform_name = "sst-platform", + .init = NULL, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_mfld = { + .name = "medfield_audio", + .dai_link = mfld_msic_dailink, + .num_links = ARRAY_SIZE(mfld_msic_dailink), +}; + +static int __devinit snd_mfld_mc_probe(struct platform_device *pdev) +{ + struct platform_device *socdev; + int ret_val = 0; + + pr_debug("snd_mfld_mc_probe called\n"); + + socdev = platform_device_alloc("soc-audio", -1); + if (!socdev) { + pr_err("soc-audio device allocation failed\n"); + return -ENOMEM; + } + platform_set_drvdata(socdev, &snd_soc_card_mfld); + ret_val = platform_device_add(socdev); + if (ret_val) { + pr_err("Unable to add soc-audio device, err %d\n", ret_val); + platform_device_put(socdev); + } + + platform_set_drvdata(pdev, socdev); + + pr_debug("successfully exited probe\n"); + return ret_val; +} + +static int __devexit snd_mfld_mc_remove(struct platform_device *pdev) +{ + struct platform_device *socdev = platform_get_drvdata(pdev); + pr_debug("snd_mfld_mc_remove called\n"); + + platform_device_unregister(socdev); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver snd_mfld_mc_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "msic_audio", + }, + .probe = snd_mfld_mc_probe, + .remove = __devexit_p(snd_mfld_mc_remove), +}; + +static int __init snd_mfld_driver_init(void) +{ + pr_debug("snd_mfld_driver_init called\n"); + return platform_driver_register(&snd_mfld_mc_driver); +} +module_init(snd_mfld_driver_init); + +static void __exit snd_mfld_driver_exit(void) +{ + pr_debug("snd_mfld_driver_exit called\n"); + platform_driver_unregister(&snd_mfld_mc_driver); +} +module_exit(snd_mfld_driver_exit); + +MODULE_DESCRIPTION("ASoC Intel(R) MID Machine driver"); +MODULE_AUTHOR("Vinod Koul "); +MODULE_AUTHOR("Harsha Priya "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:msic-audio"); -- cgit v1.2.3 From e62255f2adf2e6407d9ac72f5d5ec17712a4fc1d Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 4 Jan 2011 20:17:04 +0530 Subject: ASoC: sst v2: Add makefiles and kconfigs changes This patch adds the makefile and kconfig changes for mid asoc drivers: platform and machine driver which are introduced in 2 preceeding patches Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/mid-x86/Kconfig | 13 +++++++++++++ sound/soc/mid-x86/Makefile | 7 +++++++ 4 files changed, 22 insertions(+) create mode 100644 sound/soc/mid-x86/Kconfig create mode 100644 sound/soc/mid-x86/Makefile (limited to 'sound') diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index a3efc52a34d..821ab0b759b 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -50,6 +50,7 @@ source "sound/soc/jz4740/Kconfig" source "sound/soc/nuc900/Kconfig" source "sound/soc/omap/Kconfig" source "sound/soc/kirkwood/Kconfig" +source "sound/soc/mid-x86/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/samsung/Kconfig" source "sound/soc/s6000/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index ce913bf5213..df83d2143af 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -10,6 +10,7 @@ obj-$(CONFIG_SND_SOC) += ep93xx/ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += imx/ obj-$(CONFIG_SND_SOC) += jz4740/ +obj-$(CONFIG_SND_SOC) += mid-x86/ obj-$(CONFIG_SND_SOC) += nuc900/ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += kirkwood/ diff --git a/sound/soc/mid-x86/Kconfig b/sound/soc/mid-x86/Kconfig new file mode 100644 index 00000000000..e0123a40822 --- /dev/null +++ b/sound/soc/mid-x86/Kconfig @@ -0,0 +1,13 @@ +config SND_MFLD_MACHINE + tristate "SOC Machine Audio driver for Intel Medfield MID platform" + select SND_SOC_SN95031 + select SND_SST_PLATFORM + help + This adds support for ASoC machine driver for Intel(R) MID Medfield platform + used as alsa device in audio substem in Intel(R) MID devices + Say Y if you have such a device + If unsure select "N". + +config SND_SST_PLATFORM + tristate + depends on SND_INTEL_SST diff --git a/sound/soc/mid-x86/Makefile b/sound/soc/mid-x86/Makefile new file mode 100644 index 00000000000..a1a93314b84 --- /dev/null +++ b/sound/soc/mid-x86/Makefile @@ -0,0 +1,7 @@ +EXTRA_CFLAGS += -DDEBUG + +snd-soc-sst-platform-objs := sst_platform.o +snd-soc-mfld-machine-objs := mfld_machine.o + +obj-$(CONFIG_SND_SST_PLATFORM) += snd-soc-sst-platform.o +obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o -- cgit v1.2.3 From b22dab8883250d5fd91c8c4a7bd00c5985ac82ad Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 7 Jan 2011 16:20:28 +0530 Subject: ASoC: sn95031 fix the code style and format inconsistencies this patch fixes inconsistencies commented by Mark. This also fixes few other style things in audio_map & header file Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 30 +++++++++++++++--------------- sound/soc/codecs/sn95031.h | 12 ++++++------ 2 files changed, 21 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 146b74467ae..593632cf791 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -235,21 +235,21 @@ static const struct snd_soc_dapm_route sn95031_audio_map[] = { { "IHFDAC Right", NULL, "Speaker Rail"}, /* vibra map */ - {"VIB1OUT", NULL, "Vibra1 Playback"}, - {"Vibra1 Playback", NULL, "Vibra1 DAC"}, + { "VIB1OUT", NULL, "Vibra1 Playback"}, + { "Vibra1 Playback", NULL, "Vibra1 DAC"}, - {"VIB2OUT", NULL, "Vibra2 Playback"}, - {"Vibra2 Playback", NULL, "Vibra2 DAC"}, + { "VIB2OUT", NULL, "Vibra2 Playback"}, + { "Vibra2 Playback", NULL, "Vibra2 DAC"}, /* lineout */ - {"LINEOUTL", NULL, "Lineout Left Playback"}, - {"LINEOUTR", NULL, "Lineout Right Playback"}, - {"Lineout Left Playback", NULL, "Headset Left Filter"}, - {"Lineout Left Playback", NULL, "Speaker Left Filter"}, - {"Lineout Left Playback", NULL, "Vibra1 DAC"}, - {"Lineout Right Playback", NULL, "Headset Right Filter"}, - {"Lineout Right Playback", NULL, "Speaker Right Filter"}, - {"Lineout Right Playback", NULL, "Vibra2 DAC"}, + { "LINEOUTL", NULL, "Lineout Left Playback"}, + { "LINEOUTR", NULL, "Lineout Right Playback"}, + { "Lineout Left Playback", NULL, "Headset Left Filter"}, + { "Lineout Left Playback", NULL, "Speaker Left Filter"}, + { "Lineout Left Playback", NULL, "Vibra1 DAC"}, + { "Lineout Right Playback", NULL, "Headset Right Filter"}, + { "Lineout Right Playback", NULL, "Speaker Right Filter"}, + { "Lineout Right Playback", NULL, "Vibra2 DAC"}, }; /* speaker and headset mutes, for audio pops and clicks */ @@ -444,8 +444,8 @@ static int sn95031_codec_remove(struct snd_soc_codec *codec) } struct snd_soc_codec_driver sn95031_codec = { - .probe = sn95031_codec_probe, - .remove = sn95031_codec_remove, + .probe = sn95031_codec_probe, + .remove = sn95031_codec_remove, .read = sn95031_read, .write = sn95031_write, .set_bias_level = sn95031_set_vaud_bias, @@ -488,7 +488,7 @@ static void __exit sn95031_exit(void) } module_exit(sn95031_exit); -MODULE_DESCRIPTION("ASoC Intel(R) SN95031 codec driver"); +MODULE_DESCRIPTION("ASoC TI SN95031 codec driver"); MODULE_AUTHOR("Vinod Koul "); MODULE_AUTHOR("Harsha Priya "); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h index b17a39b2aef..e2b17d908ae 100644 --- a/sound/soc/codecs/sn95031.h +++ b/sound/soc/codecs/sn95031.h @@ -35,13 +35,13 @@ #define SN95031_AUDPLLCTRL 0x240 #define SN95031_DMICBUF0123 0x241 #define SN95031_DMICBUF45 0x242 -#define SN95031_DMICGPO 0x244 -#define SN95031_DMICMUX 0x245 +#define SN95031_DMICGPO 0x244 +#define SN95031_DMICMUX 0x245 #define SN95031_DMICLK 0x246 -#define SN95031_MICBIAS 0x247 +#define SN95031_MICBIAS 0x247 #define SN95031_ADCCONFIG 0x248 -#define SN95031_MICAMP1 0x249 -#define SN95031_MICAMP2 0x24A +#define SN95031_MICAMP1 0x249 +#define SN95031_MICAMP2 0x24A #define SN95031_NOISEMUX 0x24B #define SN95031_AUDIOMUX12 0x24C #define SN95031_AUDIOMUX34 0x24D @@ -49,7 +49,7 @@ #define SN95031_AUDIOTXEN 0x24F #define SN95031_HSEPRXCTRL 0x250 #define SN95031_IHFRXCTRL 0x251 -#define SN95031_HSMIXER 0x256 +#define SN95031_HSMIXER 0x256 #define SN95031_DACCONFIG 0x257 #define SN95031_SOFTMUTE 0x258 #define SN95031_HSLVOLCTRL 0x259 -- cgit v1.2.3 From f5c6b2aa33cdacb4902c7b946395f7e73b00d4ce Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 7 Jan 2011 16:20:56 +0530 Subject: ASoC: sst platform - fix the style inconsistency this patch fixes the style inconsistency by removing empty space in MODULE_ALIAS also removes a redundant return statement Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/mid-x86/sst_platform.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 189d546fbf9..0e994c16e7a 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -176,7 +176,6 @@ static void sst_period_elapsed(void *mad_substream) } spin_unlock(&stream->status_lock); snd_pcm_period_elapsed(substream); - return; } static int sst_platform_init_stream(struct snd_pcm_substream *substream) @@ -481,4 +480,4 @@ MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver"); MODULE_AUTHOR("Vinod Koul "); MODULE_AUTHOR("Harsha Priya "); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platfrom: sst-platform"); +MODULE_ALIAS("platform:sst-platform"); -- cgit v1.2.3 From fd0d08d9edc8ffc900896e1d1f7f287e6fde1ab3 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 7 Jan 2011 16:21:12 +0530 Subject: ASoC: mid-x86 - remove the flag in makefile previous commit e62255f2adf2 introduced DDEBUG flag in Makefile This causes all debug statemenets to be ON. This patch removes this flag Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/mid-x86/Makefile | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/mid-x86/Makefile b/sound/soc/mid-x86/Makefile index a1a93314b84..63988333946 100644 --- a/sound/soc/mid-x86/Makefile +++ b/sound/soc/mid-x86/Makefile @@ -1,5 +1,3 @@ -EXTRA_CFLAGS += -DDEBUG - snd-soc-sst-platform-objs := sst_platform.o snd-soc-mfld-machine-objs := mfld_machine.o -- cgit v1.2.3 From 4f644ea70d54bf851fc0ebc77fa0f66fce7cb7d2 Mon Sep 17 00:00:00 2001 From: Seungwhan Youn Date: Fri, 7 Jan 2011 13:46:52 +0900 Subject: ASoC: SAMSUNG: Remove AC97 header file This patch moves AC97 specific definitions, they doesn't need to be shared, into 'ac97.c' from 'ac97.h'. This patch also remove header file, 'ac97.h', and remove includes on machine drivers. Signed-off-by: Seungwhan Youn Acked-by: Jassi Brar Signed-off-by: Mark Brown --- sound/soc/samsung/ac97.c | 4 +++- sound/soc/samsung/ac97.h | 21 --------------------- sound/soc/samsung/ln2440sbc_alc650.c | 1 - sound/soc/samsung/smdk2443_wm9710.c | 1 - sound/soc/samsung/smdk_wm9713.c | 1 - 5 files changed, 3 insertions(+), 25 deletions(-) delete mode 100644 sound/soc/samsung/ac97.h (limited to 'sound') diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 4770a955034..1e84ee01eac 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -25,11 +25,13 @@ #include #include "dma.h" -#include "ac97.h" #define AC_CMD_ADDR(x) (x << 16) #define AC_CMD_DATA(x) (x & 0xffff) +#define S3C_AC97_DAI_PCM 0 +#define S3C_AC97_DAI_MIC 1 + struct s3c_ac97_info { struct clk *ac97_clk; void __iomem *regs; diff --git a/sound/soc/samsung/ac97.h b/sound/soc/samsung/ac97.h deleted file mode 100644 index 0d0e1b51145..00000000000 --- a/sound/soc/samsung/ac97.h +++ /dev/null @@ -1,21 +0,0 @@ -/* sound/soc/samsung/ac97.h - * - * ALSA SoC Audio Layer - S3C AC97 Controller driver - * Evolved from s3c2443-ac97.h - * - * Copyright (c) 2010 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh - * Credits: Graeme Gregory, Sean Choi - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __S3C_AC97_H_ -#define __S3C_AC97_H_ - -#define S3C_AC97_DAI_PCM 0 -#define S3C_AC97_DAI_MIC 1 - -#endif /* __S3C_AC97_H_ */ diff --git a/sound/soc/samsung/ln2440sbc_alc650.c b/sound/soc/samsung/ln2440sbc_alc650.c index a2bb34def74..02414290169 100644 --- a/sound/soc/samsung/ln2440sbc_alc650.c +++ b/sound/soc/samsung/ln2440sbc_alc650.c @@ -23,7 +23,6 @@ #include #include "dma.h" -#include "ac97.h" static struct snd_soc_card ln2440sbc; diff --git a/sound/soc/samsung/smdk2443_wm9710.c b/sound/soc/samsung/smdk2443_wm9710.c index 3be7e7e92d6..4bd8c0b6396 100644 --- a/sound/soc/samsung/smdk2443_wm9710.c +++ b/sound/soc/samsung/smdk2443_wm9710.c @@ -19,7 +19,6 @@ #include #include "dma.h" -#include "ac97.h" static struct snd_soc_card smdk2443; diff --git a/sound/soc/samsung/smdk_wm9713.c b/sound/soc/samsung/smdk_wm9713.c index ae5fed6f772..d58d86e7e19 100644 --- a/sound/soc/samsung/smdk_wm9713.c +++ b/sound/soc/samsung/smdk_wm9713.c @@ -16,7 +16,6 @@ #include #include "dma.h" -#include "ac97.h" static struct snd_soc_card smdk; -- cgit v1.2.3 From f7f741810f14f971337b15892007622c3af1f109 Mon Sep 17 00:00:00 2001 From: Seungwhan Youn Date: Fri, 7 Jan 2011 13:51:26 +0900 Subject: ASoC: SAMSUNG: Clean-up DMA header file This patch moves DMA specific definitions, they doesn't need to be shared, into 'dma.c' from 'dma.h'. And remove unnecessery definitions on 'dma.h'. Signed-off-by: Seungwhan Youn Acked-by: Jassi Brar Signed-off-by: Mark Brown --- sound/soc/samsung/dma.c | 3 +++ sound/soc/samsung/dma.h | 8 -------- 2 files changed, 3 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 21240198c5d..c2cdb4597e5 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -32,6 +32,9 @@ #include "dma.h" +#define ST_RUNNING (1<<0) +#define ST_OPENED (1<<1) + static const struct snd_pcm_hardware dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index f8cd2b4223a..c50659269a4 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -12,9 +12,6 @@ #ifndef _S3C_AUDIO_H #define _S3C_AUDIO_H -#define ST_RUNNING (1<<0) -#define ST_OPENED (1<<1) - struct s3c_dma_params { struct s3c2410_dma_client *client; /* stream identifier */ int channel; /* Channel ID */ @@ -22,9 +19,4 @@ struct s3c_dma_params { int dma_size; /* Size of the DMA transfer */ }; -#define S3C24XX_DAI_I2S 0 - -/* platform data */ -extern struct snd_ac97_bus_ops s3c24xx_ac97_ops; - #endif -- cgit v1.2.3 From 9c6df19e55799d23863eb414b0ee850c4fd3a174 Mon Sep 17 00:00:00 2001 From: Seungwhan Youn Date: Fri, 7 Jan 2011 13:57:23 +0900 Subject: ASoC: SAMSUNG: Move PCM specific definitions into pcm.c This patch moves PCM specific definitions, they doesn't need to be shared, into 'pcm.c' from 'pcm.h'. Signed-off-by: Seungwhan Youn Acked-by: Jassi Brar Signed-off-by: Mark Brown --- sound/soc/samsung/pcm.c | 107 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/samsung/pcm.h | 107 ------------------------------------------------ 2 files changed, 107 insertions(+), 107 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 48d0b750406..091db5dce07 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -32,6 +32,113 @@ #include "dma.h" #include "pcm.h" +/*Register Offsets */ +#define S3C_PCM_CTL 0x00 +#define S3C_PCM_CLKCTL 0x04 +#define S3C_PCM_TXFIFO 0x08 +#define S3C_PCM_RXFIFO 0x0C +#define S3C_PCM_IRQCTL 0x10 +#define S3C_PCM_IRQSTAT 0x14 +#define S3C_PCM_FIFOSTAT 0x18 +#define S3C_PCM_CLRINT 0x20 + +/* PCM_CTL Bit-Fields */ +#define S3C_PCM_CTL_TXDIPSTICK_MASK 0x3f +#define S3C_PCM_CTL_TXDIPSTICK_SHIFT 13 +#define S3C_PCM_CTL_RXDIPSTICK_MASK 0x3f +#define S3C_PCM_CTL_RXDIPSTICK_SHIFT 7 +#define S3C_PCM_CTL_TXDMA_EN (0x1 << 6) +#define S3C_PCM_CTL_RXDMA_EN (0x1 << 5) +#define S3C_PCM_CTL_TXMSB_AFTER_FSYNC (0x1 << 4) +#define S3C_PCM_CTL_RXMSB_AFTER_FSYNC (0x1 << 3) +#define S3C_PCM_CTL_TXFIFO_EN (0x1 << 2) +#define S3C_PCM_CTL_RXFIFO_EN (0x1 << 1) +#define S3C_PCM_CTL_ENABLE (0x1 << 0) + +/* PCM_CLKCTL Bit-Fields */ +#define S3C_PCM_CLKCTL_SERCLK_EN (0x1 << 19) +#define S3C_PCM_CLKCTL_SERCLKSEL_PCLK (0x1 << 18) +#define S3C_PCM_CLKCTL_SCLKDIV_MASK 0x1ff +#define S3C_PCM_CLKCTL_SYNCDIV_MASK 0x1ff +#define S3C_PCM_CLKCTL_SCLKDIV_SHIFT 9 +#define S3C_PCM_CLKCTL_SYNCDIV_SHIFT 0 + +/* PCM_TXFIFO Bit-Fields */ +#define S3C_PCM_TXFIFO_DVALID (0x1 << 16) +#define S3C_PCM_TXFIFO_DATA_MSK (0xffff << 0) + +/* PCM_RXFIFO Bit-Fields */ +#define S3C_PCM_RXFIFO_DVALID (0x1 << 16) +#define S3C_PCM_RXFIFO_DATA_MSK (0xffff << 0) + +/* PCM_IRQCTL Bit-Fields */ +#define S3C_PCM_IRQCTL_IRQEN (0x1 << 14) +#define S3C_PCM_IRQCTL_WRDEN (0x1 << 12) +#define S3C_PCM_IRQCTL_TXEMPTYEN (0x1 << 11) +#define S3C_PCM_IRQCTL_TXALMSTEMPTYEN (0x1 << 10) +#define S3C_PCM_IRQCTL_TXFULLEN (0x1 << 9) +#define S3C_PCM_IRQCTL_TXALMSTFULLEN (0x1 << 8) +#define S3C_PCM_IRQCTL_TXSTARVEN (0x1 << 7) +#define S3C_PCM_IRQCTL_TXERROVRFLEN (0x1 << 6) +#define S3C_PCM_IRQCTL_RXEMPTEN (0x1 << 5) +#define S3C_PCM_IRQCTL_RXALMSTEMPTEN (0x1 << 4) +#define S3C_PCM_IRQCTL_RXFULLEN (0x1 << 3) +#define S3C_PCM_IRQCTL_RXALMSTFULLEN (0x1 << 2) +#define S3C_PCM_IRQCTL_RXSTARVEN (0x1 << 1) +#define S3C_PCM_IRQCTL_RXERROVRFLEN (0x1 << 0) + +/* PCM_IRQSTAT Bit-Fields */ +#define S3C_PCM_IRQSTAT_IRQPND (0x1 << 13) +#define S3C_PCM_IRQSTAT_WRD_XFER (0x1 << 12) +#define S3C_PCM_IRQSTAT_TXEMPTY (0x1 << 11) +#define S3C_PCM_IRQSTAT_TXALMSTEMPTY (0x1 << 10) +#define S3C_PCM_IRQSTAT_TXFULL (0x1 << 9) +#define S3C_PCM_IRQSTAT_TXALMSTFULL (0x1 << 8) +#define S3C_PCM_IRQSTAT_TXSTARV (0x1 << 7) +#define S3C_PCM_IRQSTAT_TXERROVRFL (0x1 << 6) +#define S3C_PCM_IRQSTAT_RXEMPT (0x1 << 5) +#define S3C_PCM_IRQSTAT_RXALMSTEMPT (0x1 << 4) +#define S3C_PCM_IRQSTAT_RXFULL (0x1 << 3) +#define S3C_PCM_IRQSTAT_RXALMSTFULL (0x1 << 2) +#define S3C_PCM_IRQSTAT_RXSTARV (0x1 << 1) +#define S3C_PCM_IRQSTAT_RXERROVRFL (0x1 << 0) + +/* PCM_FIFOSTAT Bit-Fields */ +#define S3C_PCM_FIFOSTAT_TXCNT_MSK (0x3f << 14) +#define S3C_PCM_FIFOSTAT_TXFIFOEMPTY (0x1 << 13) +#define S3C_PCM_FIFOSTAT_TXFIFOALMSTEMPTY (0x1 << 12) +#define S3C_PCM_FIFOSTAT_TXFIFOFULL (0x1 << 11) +#define S3C_PCM_FIFOSTAT_TXFIFOALMSTFULL (0x1 << 10) +#define S3C_PCM_FIFOSTAT_RXCNT_MSK (0x3f << 4) +#define S3C_PCM_FIFOSTAT_RXFIFOEMPTY (0x1 << 3) +#define S3C_PCM_FIFOSTAT_RXFIFOALMSTEMPTY (0x1 << 2) +#define S3C_PCM_FIFOSTAT_RXFIFOFULL (0x1 << 1) +#define S3C_PCM_FIFOSTAT_RXFIFOALMSTFULL (0x1 << 0) + +/** + * struct s3c_pcm_info - S3C PCM Controller information + * @dev: The parent device passed to use from the probe. + * @regs: The pointer to the device register block. + * @dma_playback: DMA information for playback channel. + * @dma_capture: DMA information for capture channel. + */ +struct s3c_pcm_info { + spinlock_t lock; + struct device *dev; + void __iomem *regs; + + unsigned int sclk_per_fs; + + /* Whether to keep PCMSCLK enabled even when idle(no active xfer) */ + unsigned int idleclk; + + struct clk *pclk; + struct clk *cclk; + + struct s3c_dma_params *dma_playback; + struct s3c_dma_params *dma_capture; +}; + static struct s3c2410_dma_client s3c_pcm_dma_client_out = { .name = "PCM Stereo out" }; diff --git a/sound/soc/samsung/pcm.h b/sound/soc/samsung/pcm.h index 03393dcf852..726baf81461 100644 --- a/sound/soc/samsung/pcm.h +++ b/sound/soc/samsung/pcm.h @@ -9,116 +9,9 @@ #ifndef __S3C_PCM_H #define __S3C_PCM_H __FILE__ -/*Register Offsets */ -#define S3C_PCM_CTL (0x00) -#define S3C_PCM_CLKCTL (0x04) -#define S3C_PCM_TXFIFO (0x08) -#define S3C_PCM_RXFIFO (0x0C) -#define S3C_PCM_IRQCTL (0x10) -#define S3C_PCM_IRQSTAT (0x14) -#define S3C_PCM_FIFOSTAT (0x18) -#define S3C_PCM_CLRINT (0x20) - -/* PCM_CTL Bit-Fields */ -#define S3C_PCM_CTL_TXDIPSTICK_MASK (0x3f) -#define S3C_PCM_CTL_TXDIPSTICK_SHIFT (13) -#define S3C_PCM_CTL_RXDIPSTICK_MASK (0x3f) -#define S3C_PCM_CTL_RXDIPSTICK_SHIFT (7) -#define S3C_PCM_CTL_TXDMA_EN (0x1<<6) -#define S3C_PCM_CTL_RXDMA_EN (0x1<<5) -#define S3C_PCM_CTL_TXMSB_AFTER_FSYNC (0x1<<4) -#define S3C_PCM_CTL_RXMSB_AFTER_FSYNC (0x1<<3) -#define S3C_PCM_CTL_TXFIFO_EN (0x1<<2) -#define S3C_PCM_CTL_RXFIFO_EN (0x1<<1) -#define S3C_PCM_CTL_ENABLE (0x1<<0) - -/* PCM_CLKCTL Bit-Fields */ -#define S3C_PCM_CLKCTL_SERCLK_EN (0x1<<19) -#define S3C_PCM_CLKCTL_SERCLKSEL_PCLK (0x1<<18) -#define S3C_PCM_CLKCTL_SCLKDIV_MASK (0x1ff) -#define S3C_PCM_CLKCTL_SYNCDIV_MASK (0x1ff) -#define S3C_PCM_CLKCTL_SCLKDIV_SHIFT (9) -#define S3C_PCM_CLKCTL_SYNCDIV_SHIFT (0) - -/* PCM_TXFIFO Bit-Fields */ -#define S3C_PCM_TXFIFO_DVALID (0x1<<16) -#define S3C_PCM_TXFIFO_DATA_MSK (0xffff<<0) - -/* PCM_RXFIFO Bit-Fields */ -#define S3C_PCM_RXFIFO_DVALID (0x1<<16) -#define S3C_PCM_RXFIFO_DATA_MSK (0xffff<<0) - -/* PCM_IRQCTL Bit-Fields */ -#define S3C_PCM_IRQCTL_IRQEN (0x1<<14) -#define S3C_PCM_IRQCTL_WRDEN (0x1<<12) -#define S3C_PCM_IRQCTL_TXEMPTYEN (0x1<<11) -#define S3C_PCM_IRQCTL_TXALMSTEMPTYEN (0x1<<10) -#define S3C_PCM_IRQCTL_TXFULLEN (0x1<<9) -#define S3C_PCM_IRQCTL_TXALMSTFULLEN (0x1<<8) -#define S3C_PCM_IRQCTL_TXSTARVEN (0x1<<7) -#define S3C_PCM_IRQCTL_TXERROVRFLEN (0x1<<6) -#define S3C_PCM_IRQCTL_RXEMPTEN (0x1<<5) -#define S3C_PCM_IRQCTL_RXALMSTEMPTEN (0x1<<4) -#define S3C_PCM_IRQCTL_RXFULLEN (0x1<<3) -#define S3C_PCM_IRQCTL_RXALMSTFULLEN (0x1<<2) -#define S3C_PCM_IRQCTL_RXSTARVEN (0x1<<1) -#define S3C_PCM_IRQCTL_RXERROVRFLEN (0x1<<0) - -/* PCM_IRQSTAT Bit-Fields */ -#define S3C_PCM_IRQSTAT_IRQPND (0x1<<13) -#define S3C_PCM_IRQSTAT_WRD_XFER (0x1<<12) -#define S3C_PCM_IRQSTAT_TXEMPTY (0x1<<11) -#define S3C_PCM_IRQSTAT_TXALMSTEMPTY (0x1<<10) -#define S3C_PCM_IRQSTAT_TXFULL (0x1<<9) -#define S3C_PCM_IRQSTAT_TXALMSTFULL (0x1<<8) -#define S3C_PCM_IRQSTAT_TXSTARV (0x1<<7) -#define S3C_PCM_IRQSTAT_TXERROVRFL (0x1<<6) -#define S3C_PCM_IRQSTAT_RXEMPT (0x1<<5) -#define S3C_PCM_IRQSTAT_RXALMSTEMPT (0x1<<4) -#define S3C_PCM_IRQSTAT_RXFULL (0x1<<3) -#define S3C_PCM_IRQSTAT_RXALMSTFULL (0x1<<2) -#define S3C_PCM_IRQSTAT_RXSTARV (0x1<<1) -#define S3C_PCM_IRQSTAT_RXERROVRFL (0x1<<0) - -/* PCM_FIFOSTAT Bit-Fields */ -#define S3C_PCM_FIFOSTAT_TXCNT_MSK (0x3f<<14) -#define S3C_PCM_FIFOSTAT_TXFIFOEMPTY (0x1<<13) -#define S3C_PCM_FIFOSTAT_TXFIFOALMSTEMPTY (0x1<<12) -#define S3C_PCM_FIFOSTAT_TXFIFOFULL (0x1<<11) -#define S3C_PCM_FIFOSTAT_TXFIFOALMSTFULL (0x1<<10) -#define S3C_PCM_FIFOSTAT_RXCNT_MSK (0x3f<<4) -#define S3C_PCM_FIFOSTAT_RXFIFOEMPTY (0x1<<3) -#define S3C_PCM_FIFOSTAT_RXFIFOALMSTEMPTY (0x1<<2) -#define S3C_PCM_FIFOSTAT_RXFIFOFULL (0x1<<1) -#define S3C_PCM_FIFOSTAT_RXFIFOALMSTFULL (0x1<<0) - #define S3C_PCM_CLKSRC_PCLK 0 #define S3C_PCM_CLKSRC_MUX 1 #define S3C_PCM_SCLK_PER_FS 0 -/** - * struct s3c_pcm_info - S3C PCM Controller information - * @dev: The parent device passed to use from the probe. - * @regs: The pointer to the device register block. - * @dma_playback: DMA information for playback channel. - * @dma_capture: DMA information for capture channel. - */ -struct s3c_pcm_info { - spinlock_t lock; - struct device *dev; - void __iomem *regs; - - unsigned int sclk_per_fs; - - /* Whether to keep PCMSCLK enabled even when idle(no active xfer) */ - unsigned int idleclk; - - struct clk *pclk; - struct clk *cclk; - - struct s3c_dma_params *dma_playback; - struct s3c_dma_params *dma_capture; -}; - #endif /* __S3C_PCM_H */ -- cgit v1.2.3 From 0d51a9cbb6c1275cc0f3840c36f8b7840a438c59 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 6 Jan 2011 16:04:57 +0000 Subject: ASoC: Taint the kernel if debugfs is used to write directly to CODECs Since direct register writes may confuse the drivers and are supposed to be used only in diagnostic situations discourage their use in production by tainting the kernel when we do a write. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bac7291b6ff..96bf381683f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -209,6 +209,10 @@ static ssize_t codec_reg_write_file(struct file *file, start++; if (strict_strtoul(start, 16, &value)) return -EINVAL; + + /* Userspace has been fiddling around behind the kernel's back */ + add_taint(TAINT_USER); + snd_soc_write(codec, reg, value); return buf_size; } -- cgit v1.2.3 From 2aa86323d815bab62a7e1d3ef8ed6c81a6dfeffa Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Mon, 10 Jan 2011 10:10:56 +0000 Subject: ASoC: soc-core: Add support for NULL default register caches The infrastructure for handling NULL default register maps is already included in the soc-cache code, just ensure that we don't try to dereference a NULL pointer while accessing a NULL register cache. Signed-off-by: Dimitris Papastamos Acked-by: Timur Tabi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 96bf381683f..4274435853d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3507,11 +3507,13 @@ int snd_soc_register_codec(struct device *dev, * kernel might have freed the array by the time we initialize * the cache. */ - codec->reg_def_copy = kmemdup(codec_drv->reg_cache_default, - reg_size, GFP_KERNEL); - if (!codec->reg_def_copy) { - ret = -ENOMEM; - goto fail; + if (codec_drv->reg_cache_default) { + codec->reg_def_copy = kmemdup(codec_drv->reg_cache_default, + reg_size, GFP_KERNEL); + if (!codec->reg_def_copy) { + ret = -ENOMEM; + goto fail; + } } } -- cgit v1.2.3 From faff4bb067d15a3bc0dde8c50cbc1a7075e314de Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 7 Jan 2011 22:36:11 -0700 Subject: ASoC: Export debugfs root dentry A couple Tegra ASoC drivers will create debugfs entries. Mark requested these by under debugfs/asoc/ not just debugfs/. To enable this, export the dentry representing debugfs/asoc/. Also, rename debugfs_root -> asoc_debugfs_root now it's exported to prevent potential symbol name clashes. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4274435853d..0484e504b58 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -48,7 +48,8 @@ static DEFINE_MUTEX(pcm_mutex); static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); #ifdef CONFIG_DEBUG_FS -static struct dentry *debugfs_root; +struct dentry *asoc_debugfs_root; +EXPORT_SYMBOL_GPL(asoc_debugfs_root); #endif static DEFINE_MUTEX(client_mutex); @@ -360,7 +361,7 @@ static const struct file_operations platform_list_fops = { static void soc_init_card_debugfs(struct snd_soc_card *card) { card->debugfs_card_root = debugfs_create_dir(card->name, - debugfs_root); + asoc_debugfs_root); if (!card->debugfs_card_root) { dev_warn(card->dev, "ASoC: Failed to create codec debugfs directory\n"); @@ -3583,22 +3584,22 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); static int __init snd_soc_init(void) { #ifdef CONFIG_DEBUG_FS - debugfs_root = debugfs_create_dir("asoc", NULL); - if (IS_ERR(debugfs_root) || !debugfs_root) { + asoc_debugfs_root = debugfs_create_dir("asoc", NULL); + if (IS_ERR(asoc_debugfs_root) || !asoc_debugfs_root) { printk(KERN_WARNING "ASoC: Failed to create debugfs directory\n"); - debugfs_root = NULL; + asoc_debugfs_root = NULL; } - if (!debugfs_create_file("codecs", 0444, debugfs_root, NULL, + if (!debugfs_create_file("codecs", 0444, asoc_debugfs_root, NULL, &codec_list_fops)) pr_warn("ASoC: Failed to create CODEC list debugfs file\n"); - if (!debugfs_create_file("dais", 0444, debugfs_root, NULL, + if (!debugfs_create_file("dais", 0444, asoc_debugfs_root, NULL, &dai_list_fops)) pr_warn("ASoC: Failed to create DAI list debugfs file\n"); - if (!debugfs_create_file("platforms", 0444, debugfs_root, NULL, + if (!debugfs_create_file("platforms", 0444, asoc_debugfs_root, NULL, &platform_list_fops)) pr_warn("ASoC: Failed to create platform list debugfs file\n"); #endif @@ -3610,7 +3611,7 @@ module_init(snd_soc_init); static void __exit snd_soc_exit(void) { #ifdef CONFIG_DEBUG_FS - debugfs_remove_recursive(debugfs_root); + debugfs_remove_recursive(asoc_debugfs_root); #endif platform_driver_unregister(&soc_driver); } -- cgit v1.2.3 From f0d8af4f528ee6dd63670521d429edac67934c06 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 7 Jan 2011 22:36:12 -0700 Subject: ASoC: tegra: Add tegra-das driver The DAS (Digital Audio Switch) is a mux/crossbar which sits between the DACs (Digital Audio Controllers) and the DAPs (Digital Audio Ports). Audio data may be routed between DACs and DAPs in various combinations as required by board design/application. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_das.c | 263 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra_das.h | 135 +++++++++++++++++++++++ 2 files changed, 398 insertions(+) create mode 100644 sound/soc/tegra/tegra_das.c create mode 100644 sound/soc/tegra/tegra_das.h (limited to 'sound') diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c new file mode 100644 index 00000000000..796d36d5188 --- /dev/null +++ b/sound/soc/tegra/tegra_das.c @@ -0,0 +1,263 @@ +/* + * tegra_das.c - Tegra DAS driver + * + * Author: Stephen Warren + * Copyright (C) 2010 - NVIDIA, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "tegra_das.h" + +#define DRV_NAME "tegra-das" + +static struct tegra_das *das; + +static inline void tegra_das_write(u32 reg, u32 val) +{ + __raw_writel(val, das->regs + reg); +} + +static inline u32 tegra_das_read(u32 reg) +{ + return __raw_readl(das->regs + reg); +} + +int tegra_das_connect_dap_to_dac(int dap, int dac) +{ + u32 addr; + u32 reg; + + if (!das) + return -ENODEV; + + addr = TEGRA_DAS_DAP_CTRL_SEL + + (dap * TEGRA_DAS_DAP_CTRL_SEL_STRIDE); + reg = dac << TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P; + + tegra_das_write(addr, reg); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra_das_connect_dap_to_dac); + +int tegra_das_connect_dap_to_dap(int dap, int otherdap, int master, + int sdata1rx, int sdata2rx) +{ + u32 addr; + u32 reg; + + if (!das) + return -ENODEV; + + addr = TEGRA_DAS_DAP_CTRL_SEL + + (dap * TEGRA_DAS_DAP_CTRL_SEL_STRIDE); + reg = otherdap << TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P | + !!sdata2rx << TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P | + !!sdata1rx << TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P | + !!master << TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P; + + tegra_das_write(addr, reg); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra_das_connect_dap_to_dap); + +int tegra_das_connect_dac_to_dap(int dac, int dap) +{ + u32 addr; + u32 reg; + + if (!das) + return -ENODEV; + + addr = TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL + + (dac * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE); + reg = dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P | + dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P | + dap << TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P; + + tegra_das_write(addr, reg); + + return 0; +} +EXPORT_SYMBOL_GPL(tegra_das_connect_dac_to_dap); + +#ifdef CONFIG_DEBUG_FS +static int tegra_das_show(struct seq_file *s, void *unused) +{ + int i; + u32 addr; + u32 reg; + + for (i = 0; i < TEGRA_DAS_DAP_CTRL_SEL_COUNT; i++) { + addr = TEGRA_DAS_DAP_CTRL_SEL + + (i * TEGRA_DAS_DAP_CTRL_SEL_STRIDE); + reg = tegra_das_read(addr); + seq_printf(s, "TEGRA_DAS_DAP_CTRL_SEL[%d] = %08x\n", i, reg); + } + + for (i = 0; i < TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT; i++) { + addr = TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL + + (i * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE); + reg = tegra_das_read(addr); + seq_printf(s, "TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL[%d] = %08x\n", + i, reg); + } + + return 0; +} + +static int tegra_das_debug_open(struct inode *inode, struct file *file) +{ + return single_open(file, tegra_das_show, inode->i_private); +} + +static const struct file_operations tegra_das_debug_fops = { + .open = tegra_das_debug_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static void tegra_das_debug_add(struct tegra_das *das) +{ + das->debug = debugfs_create_file(DRV_NAME, S_IRUGO, asoc_debugfs_root, + das, &tegra_das_debug_fops); +} + +static void tegra_das_debug_remove(struct tegra_das *das) +{ + if (das->debug) + debugfs_remove(das->debug); +} +#else +static inline void tegra_das_debug_add(struct tegra_das *das) +{ +} + +static inline void tegra_das_debug_remove(struct tegra_das *das) +{ +} +#endif + +static int __devinit tegra_das_probe(struct platform_device *pdev) +{ + struct resource *res, *region; + int ret = 0; + + if (das) + return -ENODEV; + + das = kzalloc(sizeof(struct tegra_das), GFP_KERNEL); + if (!das) { + dev_err(&pdev->dev, "Can't allocate tegra_das\n"); + ret = -ENOMEM; + goto exit; + } + das->dev = &pdev->dev; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "No memory resource\n"); + ret = -ENODEV; + goto err_free; + } + + region = request_mem_region(res->start, resource_size(res), + pdev->name); + if (!region) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + ret = -EBUSY; + goto err_free; + } + + das->regs = ioremap(res->start, resource_size(res)); + if (!das->regs) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err_release; + } + + tegra_das_debug_add(das); + + platform_set_drvdata(pdev, das); + + return 0; + +err_release: + release_mem_region(res->start, resource_size(res)); +err_free: + kfree(das); + das = 0; +exit: + return ret; +} + +static int __devexit tegra_das_remove(struct platform_device *pdev) +{ + struct resource *res; + + if (!das) + return -ENODEV; + + platform_set_drvdata(pdev, NULL); + + tegra_das_debug_remove(das); + + iounmap(das->regs); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(res->start, resource_size(res)); + + kfree(das); + das = 0; + + return 0; +} + +static struct platform_driver tegra_das_driver = { + .probe = tegra_das_probe, + .remove = __devexit_p(tegra_das_remove), + .driver = { + .name = DRV_NAME, + }, +}; + +static int __init tegra_das_modinit(void) +{ + return platform_driver_register(&tegra_das_driver); +} +module_init(tegra_das_modinit); + +static void __exit tegra_das_modexit(void) +{ + platform_driver_unregister(&tegra_das_driver); +} +module_exit(tegra_das_modexit); + +MODULE_AUTHOR("Stephen Warren "); +MODULE_DESCRIPTION("Tegra DAS driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/tegra/tegra_das.h b/sound/soc/tegra/tegra_das.h new file mode 100644 index 00000000000..2c96c7b3c45 --- /dev/null +++ b/sound/soc/tegra/tegra_das.h @@ -0,0 +1,135 @@ +/* + * tegra_das.h - Definitions for Tegra DAS driver + * + * Author: Stephen Warren + * Copyright (C) 2010 - NVIDIA, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TEGRA_DAS_H__ +#define __TEGRA_DAS_H__ + +/* Register TEGRA_DAS_DAP_CTRL_SEL */ +#define TEGRA_DAS_DAP_CTRL_SEL 0x00 +#define TEGRA_DAS_DAP_CTRL_SEL_COUNT 5 +#define TEGRA_DAS_DAP_CTRL_SEL_STRIDE 4 +#define TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_P 31 +#define TEGRA_DAS_DAP_CTRL_SEL_DAP_MS_SEL_S 1 +#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_P 30 +#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA1_TX_RX_S 1 +#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_P 29 +#define TEGRA_DAS_DAP_CTRL_SEL_DAP_SDATA2_TX_RX_S 1 +#define TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_P 0 +#define TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL_S 5 + +/* Values for field TEGRA_DAS_DAP_CTRL_SEL_DAP_CTRL_SEL */ +#define TEGRA_DAS_DAP_SEL_DAC1 0 +#define TEGRA_DAS_DAP_SEL_DAC2 1 +#define TEGRA_DAS_DAP_SEL_DAC3 2 +#define TEGRA_DAS_DAP_SEL_DAP1 16 +#define TEGRA_DAS_DAP_SEL_DAP2 17 +#define TEGRA_DAS_DAP_SEL_DAP3 18 +#define TEGRA_DAS_DAP_SEL_DAP4 19 +#define TEGRA_DAS_DAP_SEL_DAP5 20 + +/* Register TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL */ +#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL 0x40 +#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_COUNT 3 +#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_STRIDE 4 +#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_P 28 +#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL_S 4 +#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_P 24 +#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL_S 4 +#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_P 0 +#define TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL_S 4 + +/* + * Values for: + * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA2_SEL + * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_SDATA1_SEL + * TEGRA_DAS_DAC_INPUT_DATA_CLK_SEL_DAC_CLK_SEL + */ +#define TEGRA_DAS_DAC_SEL_DAP1 0 +#define TEGRA_DAS_DAC_SEL_DAP2 1 +#define TEGRA_DAS_DAC_SEL_DAP3 2 +#define TEGRA_DAS_DAC_SEL_DAP4 3 +#define TEGRA_DAS_DAC_SEL_DAP5 4 + +/* + * Names/IDs of the DACs/DAPs. + */ + +#define TEGRA_DAS_DAP_ID_1 0 +#define TEGRA_DAS_DAP_ID_2 1 +#define TEGRA_DAS_DAP_ID_3 2 +#define TEGRA_DAS_DAP_ID_4 3 +#define TEGRA_DAS_DAP_ID_5 4 + +#define TEGRA_DAS_DAC_ID_1 0 +#define TEGRA_DAS_DAC_ID_2 1 +#define TEGRA_DAS_DAC_ID_3 2 + +struct tegra_das { + struct device *dev; + void __iomem *regs; + struct dentry *debug; +}; + +/* + * Terminology: + * DAS: Digital audio switch (HW module controlled by this driver) + * DAP: Digital audio port (port/pins on Tegra device) + * DAC: Digital audio controller (e.g. I2S or AC97 controller elsewhere) + * + * The Tegra DAS is a mux/cross-bar which can connect each DAP to a specific + * DAC, or another DAP. When DAPs are connected, one must be the master and + * one the slave. Each DAC allows selection of a specific DAP for input, to + * cater for the case where N DAPs are connected to 1 DAC for broadcast + * output. + * + * This driver is dumb; no attempt is made to ensure that a valid routing + * configuration is programmed. + */ + +/* + * Connect a DAP to to a DAC + * dap_id: DAP to connect: TEGRA_DAS_DAP_ID_* + * dac_sel: DAC to connect to: TEGRA_DAS_DAP_SEL_DAC* + */ +extern int tegra_das_connect_dap_to_dac(int dap_id, int dac_sel); + +/* + * Connect a DAP to to another DAP + * dap_id: DAP to connect: TEGRA_DAS_DAP_ID_* + * other_dap_sel: DAP to connect to: TEGRA_DAS_DAP_SEL_DAP* + * master: Is this DAP the master (1) or slave (0) + * sdata1rx: Is this DAP's SDATA1 pin RX (1) or TX (0) + * sdata2rx: Is this DAP's SDATA2 pin RX (1) or TX (0) + */ +extern int tegra_das_connect_dap_to_dap(int dap_id, int other_dap_sel, + int master, int sdata1rx, + int sdata2rx); + +/* + * Connect a DAC's input to a DAP + * (DAC outputs are selected by the DAP) + * dac_id: DAC ID to connect: TEGRA_DAS_DAC_ID_* + * dap_sel: DAP to receive input from: TEGRA_DAS_DAC_SEL_DAP* + */ +extern int tegra_das_connect_dac_to_dap(int dac_id, int dap_sel); + +#endif -- cgit v1.2.3 From 7605eb5bc327497aed1816d6238d3b64e032b491 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 7 Jan 2011 22:36:13 -0700 Subject: ASoC: tegra: Add tegra-pcm driver This provides an ASoC platform driver that manages Tegra's APB DMA controller. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_pcm.c | 401 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra_pcm.h | 55 ++++++ 2 files changed, 456 insertions(+) create mode 100644 sound/soc/tegra/tegra_pcm.c create mode 100644 sound/soc/tegra/tegra_pcm.h (limited to 'sound') diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c new file mode 100644 index 00000000000..663ea9fa0ca --- /dev/null +++ b/sound/soc/tegra/tegra_pcm.c @@ -0,0 +1,401 @@ +/* + * tegra_pcm.c - Tegra PCM driver + * + * Author: Stephen Warren + * Copyright (C) 2010 - NVIDIA, Inc. + * + * Based on code copyright/by: + * + * Copyright (c) 2009-2010, NVIDIA Corporation. + * Scott Peterson + * Vijay Mali + * + * Copyright (C) 2010 Google, Inc. + * Iliyan Malchev + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include + +#include "tegra_pcm.h" + +static const struct snd_pcm_hardware tegra_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_INTERLEAVED, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 2, + .channels_max = 2, + .period_bytes_min = 1024, + .period_bytes_max = PAGE_SIZE, + .periods_min = 2, + .periods_max = 8, + .buffer_bytes_max = PAGE_SIZE * 8, + .fifo_size = 4, +}; + +static void tegra_pcm_queue_dma(struct tegra_runtime_data *prtd) +{ + struct snd_pcm_substream *substream = prtd->substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + struct tegra_dma_req *dma_req; + unsigned long addr; + + dma_req = &prtd->dma_req[prtd->dma_req_idx]; + prtd->dma_req_idx = 1 - prtd->dma_req_idx; + + addr = buf->addr + prtd->dma_pos; + prtd->dma_pos += dma_req->size; + if (prtd->dma_pos >= prtd->dma_pos_end) + prtd->dma_pos = 0; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_req->source_addr = addr; + else + dma_req->dest_addr = addr; + + tegra_dma_enqueue_req(prtd->dma_chan, dma_req); +} + +static void dma_complete_callback(struct tegra_dma_req *req) +{ + struct tegra_runtime_data *prtd = (struct tegra_runtime_data *)req->dev; + struct snd_pcm_substream *substream = prtd->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + + spin_lock(&prtd->lock); + + if (!prtd->running) { + spin_unlock(&prtd->lock); + return; + } + + if (++prtd->period_index >= runtime->periods) + prtd->period_index = 0; + + tegra_pcm_queue_dma(prtd); + + spin_unlock(&prtd->lock); + + snd_pcm_period_elapsed(substream); +} + +static void setup_dma_tx_request(struct tegra_dma_req *req, + struct tegra_pcm_dma_params * dmap) +{ + req->complete = dma_complete_callback; + req->to_memory = false; + req->dest_addr = dmap->addr; + req->dest_wrap = dmap->wrap; + req->source_bus_width = 32; + req->source_wrap = 0; + req->dest_bus_width = dmap->width; + req->req_sel = dmap->req_sel; +} + +static void setup_dma_rx_request(struct tegra_dma_req *req, + struct tegra_pcm_dma_params * dmap) +{ + req->complete = dma_complete_callback; + req->to_memory = true; + req->source_addr = dmap->addr; + req->dest_wrap = 0; + req->source_bus_width = dmap->width; + req->source_wrap = dmap->wrap; + req->dest_bus_width = 32; + req->req_sel = dmap->req_sel; +} + +static int tegra_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct tegra_runtime_data *prtd; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct tegra_pcm_dma_params * dmap; + int ret = 0; + + prtd = kzalloc(sizeof(struct tegra_runtime_data), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + runtime->private_data = prtd; + prtd->substream = substream; + + spin_lock_init(&prtd->lock); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dmap = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + setup_dma_tx_request(&prtd->dma_req[0], dmap); + setup_dma_tx_request(&prtd->dma_req[1], dmap); + } else { + dmap = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + setup_dma_rx_request(&prtd->dma_req[0], dmap); + setup_dma_rx_request(&prtd->dma_req[1], dmap); + } + + prtd->dma_req[0].dev = prtd; + prtd->dma_req[1].dev = prtd; + + prtd->dma_chan = tegra_dma_allocate_channel(TEGRA_DMA_MODE_ONESHOT); + if (IS_ERR(prtd->dma_chan)) { + ret = PTR_ERR(prtd->dma_chan); + goto err; + } + + /* Set HW params now that initialization is complete */ + snd_soc_set_runtime_hwparams(substream, &tegra_pcm_hardware); + + /* Ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto err; + + return 0; + +err: + if (prtd->dma_chan) { + tegra_dma_free_channel(prtd->dma_chan); + } + + kfree(prtd); + + return ret; +} + +static int tegra_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct tegra_runtime_data *prtd = runtime->private_data; + + tegra_dma_free_channel(prtd->dma_chan); + + kfree(prtd); + + return 0; +} + +static int tegra_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct tegra_runtime_data *prtd = runtime->private_data; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + prtd->dma_req[0].size = params_period_bytes(params); + prtd->dma_req[1].size = prtd->dma_req[0].size; + + return 0; +} + +static int tegra_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + + return 0; +} + +static int tegra_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct tegra_runtime_data *prtd = runtime->private_data; + unsigned long flags; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->dma_pos = 0; + prtd->dma_pos_end = frames_to_bytes(runtime, runtime->periods * runtime->period_size); + prtd->period_index = 0; + prtd->dma_req_idx = 0; + /* Fall-through */ + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&prtd->lock, flags); + prtd->running = 1; + spin_unlock_irqrestore(&prtd->lock, flags); + tegra_pcm_queue_dma(prtd); + tegra_pcm_queue_dma(prtd); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&prtd->lock, flags); + prtd->running = 0; + spin_unlock_irqrestore(&prtd->lock, flags); + tegra_dma_dequeue_req(prtd->dma_chan, &prtd->dma_req[0]); + tegra_dma_dequeue_req(prtd->dma_chan, &prtd->dma_req[1]); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t tegra_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct tegra_runtime_data *prtd = runtime->private_data; + + return prtd->period_index * runtime->period_size; +} + + +static int tegra_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops tegra_pcm_ops = { + .open = tegra_pcm_open, + .close = tegra_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = tegra_pcm_hw_params, + .hw_free = tegra_pcm_hw_free, + .trigger = tegra_pcm_trigger, + .pointer = tegra_pcm_pointer, + .mmap = tegra_pcm_mmap, +}; + +static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = tegra_pcm_hardware.buffer_bytes_max; + + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->bytes = size; + + return 0; +} + +static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + if (!buf->area) + return; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; +} + +static u64 tegra_dma_mask = DMA_BIT_MASK(32); + +static int tegra_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &tegra_dma_mask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->driver->playback.channels_min) { + ret = tegra_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto err; + } + + if (dai->driver->capture.channels_min) { + ret = tegra_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto err_free_play; + } + + return 0; + +err_free_play: + tegra_pcm_deallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); +err: + return ret; +} + +static void tegra_pcm_free(struct snd_pcm *pcm) +{ + tegra_pcm_deallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); + tegra_pcm_deallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); +} + +struct snd_soc_platform_driver tegra_pcm_platform = { + .ops = &tegra_pcm_ops, + .pcm_new = tegra_pcm_new, + .pcm_free = tegra_pcm_free, +}; + +static int __devinit tegra_pcm_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &tegra_pcm_platform); +} + +static int __devexit tegra_pcm_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver tegra_pcm_driver = { + .driver = { + .name = "tegra-pcm-audio", + .owner = THIS_MODULE, + }, + .probe = tegra_pcm_platform_probe, + .remove = __devexit_p(tegra_pcm_platform_remove), +}; + +static int __init snd_tegra_pcm_init(void) +{ + return platform_driver_register(&tegra_pcm_driver); +} +module_init(snd_tegra_pcm_init); + +static void __exit snd_tegra_pcm_exit(void) +{ + platform_driver_unregister(&tegra_pcm_driver); +} +module_exit(snd_tegra_pcm_exit); + +MODULE_AUTHOR("Stephen Warren "); +MODULE_DESCRIPTION("Tegra PCM ASoC driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/tegra/tegra_pcm.h b/sound/soc/tegra/tegra_pcm.h new file mode 100644 index 00000000000..dbb90339fe0 --- /dev/null +++ b/sound/soc/tegra/tegra_pcm.h @@ -0,0 +1,55 @@ +/* + * tegra_pcm.h - Definitions for Tegra PCM driver + * + * Author: Stephen Warren + * Copyright (C) 2010 - NVIDIA, Inc. + * + * Based on code copyright/by: + * + * Copyright (c) 2009-2010, NVIDIA Corporation. + * Scott Peterson + * + * Copyright (C) 2010 Google, Inc. + * Iliyan Malchev + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TEGRA_PCM_H__ +#define __TEGRA_PCM_H__ + +#include + +struct tegra_pcm_dma_params { + unsigned long addr; + unsigned long wrap; + unsigned long width; + unsigned long req_sel; +}; + +struct tegra_runtime_data { + struct snd_pcm_substream *substream; + spinlock_t lock; + int running; + int dma_pos; + int dma_pos_end; + int period_index; + int dma_req_idx; + struct tegra_dma_req dma_req[2]; + struct tegra_dma_channel *dma_chan; +}; + +#endif -- cgit v1.2.3 From 71f78e22146c522b26fc2074fcd9cb81806895b1 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 7 Jan 2011 22:36:14 -0700 Subject: ASoC: tegra: Add tegra-i2s driver This provides an ASoC DAI interface for Tegra's I2S controller. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_i2s.c | 502 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra_i2s.h | 165 +++++++++++++++ 2 files changed, 667 insertions(+) create mode 100644 sound/soc/tegra/tegra_i2s.c create mode 100644 sound/soc/tegra/tegra_i2s.h (limited to 'sound') diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c new file mode 100644 index 00000000000..9b7a22af52a --- /dev/null +++ b/sound/soc/tegra/tegra_i2s.c @@ -0,0 +1,502 @@ +/* + * tegra_i2s.c - Tegra I2S driver + * + * Author: Stephen Warren + * Copyright (C) 2010 - NVIDIA, Inc. + * + * Based on code copyright/by: + * + * Copyright (c) 2009-2010, NVIDIA Corporation. + * Scott Peterson + * + * Copyright (C) 2010 Google, Inc. + * Iliyan Malchev + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "tegra_das.h" +#include "tegra_i2s.h" + +#define DRV_NAME "tegra-i2s" + +static inline void tegra_i2s_write(struct tegra_i2s *i2s, u32 reg, u32 val) +{ + __raw_writel(val, i2s->regs + reg); +} + +static inline u32 tegra_i2s_read(struct tegra_i2s *i2s, u32 reg) +{ + return __raw_readl(i2s->regs + reg); +} + +#ifdef CONFIG_DEBUG_FS +static int tegra_i2s_show(struct seq_file *s, void *unused) +{ +#define REG(r) { r, #r } + static const struct { + int offset; + const char *name; + } regs[] = { + REG(TEGRA_I2S_CTRL), + REG(TEGRA_I2S_STATUS), + REG(TEGRA_I2S_TIMING), + REG(TEGRA_I2S_FIFO_SCR), + REG(TEGRA_I2S_PCM_CTRL), + REG(TEGRA_I2S_NW_CTRL), + REG(TEGRA_I2S_TDM_CTRL), + REG(TEGRA_I2S_TDM_TX_RX_CTRL), + }; +#undef REG + + struct tegra_i2s *i2s = s->private; + int i; + + for (i = 0; i < ARRAY_SIZE(regs); i++) { + u32 val = tegra_i2s_read(i2s, regs[i].offset); + seq_printf(s, "%s = %08x\n", regs[i].name, val); + } + + return 0; +} + +static int tegra_i2s_debug_open(struct inode *inode, struct file *file) +{ + return single_open(file, tegra_i2s_show, inode->i_private); +} + +static const struct file_operations tegra_i2s_debug_fops = { + .open = tegra_i2s_debug_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id) +{ + char name[] = DRV_NAME ".0"; + + snprintf(name, sizeof(name), DRV_NAME".%1d", id); + i2s->debug = debugfs_create_file(name, S_IRUGO, asoc_debugfs_root, + i2s, &tegra_i2s_debug_fops); +} + +static void tegra_i2s_debug_remove(struct tegra_i2s *i2s) +{ + if (i2s->debug) + debugfs_remove(i2s->debug); +} +#else +static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s) +{ +} + +static inline void tegra_i2s_debug_remove(struct tegra_i2s *i2s) +{ +} +#endif + +static int tegra_i2s_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_MASTER_ENABLE; + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + i2s->reg_ctrl |= TEGRA_I2S_CTRL_MASTER_ENABLE; + break; + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + i2s->reg_ctrl &= ~(TEGRA_I2S_CTRL_BIT_FORMAT_MASK | + TEGRA_I2S_CTRL_LRCK_MASK); + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_DSP; + i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_DSP_B: + i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_DSP; + i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_R_LOW; + break; + case SND_SOC_DAIFMT_I2S: + i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_I2S; + i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_RIGHT_J: + i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_RJM; + i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; + break; + case SND_SOC_DAIFMT_LEFT_J: + i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_FORMAT_LJM; + i2s->reg_ctrl |= TEGRA_I2S_CTRL_LRCK_L_LOW; + break; + default: + return -EINVAL; + } + + return 0; +} + +static int tegra_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct device *dev = substream->pcm->card->dev; + struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); + u32 reg; + int ret, sample_size, srate, i2sclock, bitcnt; + + i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_BIT_SIZE_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_16; + sample_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_24; + sample_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + i2s->reg_ctrl |= TEGRA_I2S_CTRL_BIT_SIZE_32; + sample_size = 32; + break; + default: + return -EINVAL; + } + + srate = params_rate(params); + + /* Final "* 2" required by Tegra hardware */ + i2sclock = srate * params_channels(params) * sample_size * 2; + + ret = clk_set_rate(i2s->clk_i2s, i2sclock); + if (ret) { + dev_err(dev, "Can't set I2S clock rate: %d\n", ret); + return ret; + } + + bitcnt = (i2sclock / (2 * srate)) - 1; + if (bitcnt < 0 || bitcnt > TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US) + return -EINVAL; + reg = bitcnt << TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT; + + if (i2sclock % (2 * srate)) + reg |= TEGRA_I2S_TIMING_NON_SYM_ENABLE; + + tegra_i2s_write(i2s, TEGRA_I2S_TIMING, reg); + + tegra_i2s_write(i2s, TEGRA_I2S_FIFO_SCR, + TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS | + TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS); + + return 0; +} + +static void tegra_i2s_start_playback(struct tegra_i2s *i2s) +{ + i2s->reg_ctrl |= TEGRA_I2S_CTRL_FIFO1_ENABLE; + tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); +} + +static void tegra_i2s_stop_playback(struct tegra_i2s *i2s) +{ + i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_FIFO1_ENABLE; + tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); +} + +static void tegra_i2s_start_capture(struct tegra_i2s *i2s) +{ + i2s->reg_ctrl |= TEGRA_I2S_CTRL_FIFO2_ENABLE; + tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); +} + +static void tegra_i2s_stop_capture(struct tegra_i2s *i2s) +{ + i2s->reg_ctrl &= ~TEGRA_I2S_CTRL_FIFO2_ENABLE; + tegra_i2s_write(i2s, TEGRA_I2S_CTRL, i2s->reg_ctrl); +} + +static int tegra_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct tegra_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + if (!i2s->clk_refs) + clk_enable(i2s->clk_i2s); + i2s->clk_refs++; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tegra_i2s_start_playback(i2s); + else + tegra_i2s_start_capture(i2s); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tegra_i2s_stop_playback(i2s); + else + tegra_i2s_stop_capture(i2s); + i2s->clk_refs--; + if (!i2s->clk_refs) + clk_disable(i2s->clk_i2s); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int tegra_i2s_probe(struct snd_soc_dai *dai) +{ + struct tegra_i2s * i2s = snd_soc_dai_get_drvdata(dai); + + dai->capture_dma_data = &i2s->capture_dma_data; + dai->playback_dma_data = &i2s->playback_dma_data; + + return 0; +} + +static struct snd_soc_dai_ops tegra_i2s_dai_ops = { + .set_fmt = tegra_i2s_set_fmt, + .hw_params = tegra_i2s_hw_params, + .trigger = tegra_i2s_trigger, +}; + +struct snd_soc_dai_driver tegra_i2s_dai[] = { + { + .name = DRV_NAME ".0", + .probe = tegra_i2s_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &tegra_i2s_dai_ops, + .symmetric_rates = 1, + }, + { + .name = DRV_NAME ".1", + .probe = tegra_i2s_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &tegra_i2s_dai_ops, + .symmetric_rates = 1, + }, +}; + +static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) +{ + struct tegra_i2s * i2s; + char clk_name[12]; /* tegra-i2s.0 */ + struct resource *mem, *memregion, *dmareq; + int ret; + + if ((pdev->id < 0) || + (pdev->id >= ARRAY_SIZE(tegra_i2s_dai))) { + dev_err(&pdev->dev, "ID %d out of range\n", pdev->id); + return -EINVAL; + } + + /* + * FIXME: Until a codec driver exists for the tegra DAS, hard-code a + * 1:1 mapping between audio controllers and audio ports. + */ + ret = tegra_das_connect_dap_to_dac(TEGRA_DAS_DAP_ID_1 + pdev->id, + TEGRA_DAS_DAP_SEL_DAC1 + pdev->id); + if (ret) { + dev_err(&pdev->dev, "Can't set up DAP connection\n"); + return ret; + } + ret = tegra_das_connect_dac_to_dap(TEGRA_DAS_DAC_ID_1 + pdev->id, + TEGRA_DAS_DAC_SEL_DAP1 + pdev->id); + if (ret) { + dev_err(&pdev->dev, "Can't set up DAC connection\n"); + return ret; + } + + i2s = kzalloc(sizeof(struct tegra_i2s), GFP_KERNEL); + if (!i2s) { + dev_err(&pdev->dev, "Can't allocate tegra_i2s\n"); + ret = -ENOMEM; + goto exit; + } + dev_set_drvdata(&pdev->dev, i2s); + + snprintf(clk_name, sizeof(clk_name), DRV_NAME ".%d", pdev->id); + i2s->clk_i2s = clk_get_sys(clk_name, NULL); + if (IS_ERR_OR_NULL(i2s->clk_i2s)) { + pr_err("Can't retrieve i2s clock\n"); + ret = PTR_ERR(i2s->clk_i2s); + goto err_free; + } + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "No memory resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmareq) { + dev_err(&pdev->dev, "No DMA resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + memregion = request_mem_region(mem->start, resource_size(mem), + DRV_NAME); + if (!memregion) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + ret = -EBUSY; + goto err_clk_put; + } + + i2s->regs = ioremap(mem->start, resource_size(mem)); + if (!i2s->regs) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err_release; + } + + i2s->capture_dma_data.addr = mem->start + TEGRA_I2S_FIFO2; + i2s->capture_dma_data.wrap = 4; + i2s->capture_dma_data.width = 32; + i2s->capture_dma_data.req_sel = dmareq->start; + + i2s->playback_dma_data.addr = mem->start + TEGRA_I2S_FIFO1; + i2s->playback_dma_data.wrap = 4; + i2s->playback_dma_data.width = 32; + i2s->playback_dma_data.req_sel = dmareq->start; + + i2s->reg_ctrl = TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED; + + ret = snd_soc_register_dai(&pdev->dev, &tegra_i2s_dai[pdev->id]); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + ret = -ENOMEM; + goto err_unmap; + } + + tegra_i2s_debug_add(i2s, pdev->id); + + return 0; + +err_unmap: + iounmap(i2s->regs); +err_release: + release_mem_region(mem->start, resource_size(mem)); +err_clk_put: + clk_put(i2s->clk_i2s); +err_free: + kfree(i2s); +exit: + return ret; +} + +static int __devexit tegra_i2s_platform_remove(struct platform_device *pdev) +{ + struct tegra_i2s *i2s = dev_get_drvdata(&pdev->dev); + struct resource *res; + + snd_soc_unregister_dai(&pdev->dev); + + tegra_i2s_debug_remove(i2s); + + iounmap(i2s->regs); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(res->start, resource_size(res)); + + clk_put(i2s->clk_i2s); + + kfree(i2s); + + return 0; +} + +static struct platform_driver tegra_i2s_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + .probe = tegra_i2s_platform_probe, + .remove = __devexit_p(tegra_i2s_platform_remove), +}; + +static int __init snd_tegra_i2s_init(void) +{ + return platform_driver_register(&tegra_i2s_driver); +} +module_init(snd_tegra_i2s_init); + +static void __exit snd_tegra_i2s_exit(void) +{ + platform_driver_unregister(&tegra_i2s_driver); +} +module_exit(snd_tegra_i2s_exit); + +MODULE_AUTHOR("Stephen Warren "); +MODULE_DESCRIPTION("Tegra I2S ASoC driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/tegra/tegra_i2s.h b/sound/soc/tegra/tegra_i2s.h new file mode 100644 index 00000000000..2b38a096f46 --- /dev/null +++ b/sound/soc/tegra/tegra_i2s.h @@ -0,0 +1,165 @@ +/* + * tegra_i2s.h - Definitions for Tegra I2S driver + * + * Author: Stephen Warren + * Copyright (C) 2010 - NVIDIA, Inc. + * + * Based on code copyright/by: + * + * Copyright (c) 2009-2010, NVIDIA Corporation. + * Scott Peterson + * + * Copyright (C) 2010 Google, Inc. + * Iliyan Malchev + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TEGRA_I2S_H__ +#define __TEGRA_I2S_H__ + +#include "tegra_pcm.h" + +/* Register offsets from TEGRA_I2S1_BASE and TEGRA_I2S2_BASE */ + +#define TEGRA_I2S_CTRL 0x00 +#define TEGRA_I2S_STATUS 0x04 +#define TEGRA_I2S_TIMING 0x08 +#define TEGRA_I2S_FIFO_SCR 0x0c +#define TEGRA_I2S_PCM_CTRL 0x10 +#define TEGRA_I2S_NW_CTRL 0x14 +#define TEGRA_I2S_TDM_CTRL 0x20 +#define TEGRA_I2S_TDM_TX_RX_CTRL 0x24 +#define TEGRA_I2S_FIFO1 0x40 +#define TEGRA_I2S_FIFO2 0x80 + +/* Fields in TEGRA_I2S_CTRL */ + +#define TEGRA_I2S_CTRL_FIFO2_TX_ENABLE (1 << 30) +#define TEGRA_I2S_CTRL_FIFO1_ENABLE (1 << 29) +#define TEGRA_I2S_CTRL_FIFO2_ENABLE (1 << 28) +#define TEGRA_I2S_CTRL_FIFO1_RX_ENABLE (1 << 27) +#define TEGRA_I2S_CTRL_FIFO_LPBK_ENABLE (1 << 26) +#define TEGRA_I2S_CTRL_MASTER_ENABLE (1 << 25) + +#define TEGRA_I2S_LRCK_LEFT_LOW 0 +#define TEGRA_I2S_LRCK_RIGHT_LOW 1 + +#define TEGRA_I2S_CTRL_LRCK_SHIFT 24 +#define TEGRA_I2S_CTRL_LRCK_MASK (1 << TEGRA_I2S_CTRL_LRCK_SHIFT) +#define TEGRA_I2S_CTRL_LRCK_L_LOW (TEGRA_I2S_LRCK_LEFT_LOW << TEGRA_I2S_CTRL_LRCK_SHIFT) +#define TEGRA_I2S_CTRL_LRCK_R_LOW (TEGRA_I2S_LRCK_RIGHT_LOW << TEGRA_I2S_CTRL_LRCK_SHIFT) + +#define TEGRA_I2S_BIT_FORMAT_I2S 0 +#define TEGRA_I2S_BIT_FORMAT_RJM 1 +#define TEGRA_I2S_BIT_FORMAT_LJM 2 +#define TEGRA_I2S_BIT_FORMAT_DSP 3 + +#define TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT 10 +#define TEGRA_I2S_CTRL_BIT_FORMAT_MASK (3 << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA_I2S_CTRL_BIT_FORMAT_I2S (TEGRA_I2S_BIT_FORMAT_I2S << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA_I2S_CTRL_BIT_FORMAT_RJM (TEGRA_I2S_BIT_FORMAT_RJM << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA_I2S_CTRL_BIT_FORMAT_LJM (TEGRA_I2S_BIT_FORMAT_LJM << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) +#define TEGRA_I2S_CTRL_BIT_FORMAT_DSP (TEGRA_I2S_BIT_FORMAT_DSP << TEGRA_I2S_CTRL_BIT_FORMAT_SHIFT) + +#define TEGRA_I2S_BIT_SIZE_16 0 +#define TEGRA_I2S_BIT_SIZE_20 1 +#define TEGRA_I2S_BIT_SIZE_24 2 +#define TEGRA_I2S_BIT_SIZE_32 3 + +#define TEGRA_I2S_CTRL_BIT_SIZE_SHIFT 8 +#define TEGRA_I2S_CTRL_BIT_SIZE_MASK (3 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA_I2S_CTRL_BIT_SIZE_16 (TEGRA_I2S_BIT_SIZE_16 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA_I2S_CTRL_BIT_SIZE_20 (TEGRA_I2S_BIT_SIZE_20 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA_I2S_CTRL_BIT_SIZE_24 (TEGRA_I2S_BIT_SIZE_24 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) +#define TEGRA_I2S_CTRL_BIT_SIZE_32 (TEGRA_I2S_BIT_SIZE_32 << TEGRA_I2S_CTRL_BIT_SIZE_SHIFT) + +#define TEGRA_I2S_FIFO_16_LSB 0 +#define TEGRA_I2S_FIFO_20_LSB 1 +#define TEGRA_I2S_FIFO_24_LSB 2 +#define TEGRA_I2S_FIFO_32 3 +#define TEGRA_I2S_FIFO_PACKED 7 + +#define TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT 4 +#define TEGRA_I2S_CTRL_FIFO_FORMAT_MASK (7 << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA_I2S_CTRL_FIFO_FORMAT_16_LSB (TEGRA_I2S_FIFO_16_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA_I2S_CTRL_FIFO_FORMAT_20_LSB (TEGRA_I2S_FIFO_20_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA_I2S_CTRL_FIFO_FORMAT_24_LSB (TEGRA_I2S_FIFO_24_LSB << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA_I2S_CTRL_FIFO_FORMAT_32 (TEGRA_I2S_FIFO_32 << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) +#define TEGRA_I2S_CTRL_FIFO_FORMAT_PACKED (TEGRA_I2S_FIFO_PACKED << TEGRA_I2S_CTRL_FIFO_FORMAT_SHIFT) + +#define TEGRA_I2S_CTRL_IE_FIFO1_ERR (1 << 3) +#define TEGRA_I2S_CTRL_IE_FIFO2_ERR (1 << 2) +#define TEGRA_I2S_CTRL_QE_FIFO1 (1 << 1) +#define TEGRA_I2S_CTRL_QE_FIFO2 (1 << 0) + +/* Fields in TEGRA_I2S_STATUS */ + +#define TEGRA_I2S_STATUS_FIFO1_RDY (1 << 31) +#define TEGRA_I2S_STATUS_FIFO2_RDY (1 << 30) +#define TEGRA_I2S_STATUS_FIFO1_BSY (1 << 29) +#define TEGRA_I2S_STATUS_FIFO2_BSY (1 << 28) +#define TEGRA_I2S_STATUS_FIFO1_ERR (1 << 3) +#define TEGRA_I2S_STATUS_FIFO2_ERR (1 << 2) +#define TEGRA_I2S_STATUS_QS_FIFO1 (1 << 1) +#define TEGRA_I2S_STATUS_QS_FIFO2 (1 << 0) + +/* Fields in TEGRA_I2S_TIMING */ + +#define TEGRA_I2S_TIMING_NON_SYM_ENABLE (1 << 12) +#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT 0 +#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US 0x7fff +#define TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK (TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US << TEGRA_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT) + +/* Fields in TEGRA_I2S_FIFO_SCR */ + +#define TEGRA_I2S_FIFO_SCR_FIFO2_FULL_EMPTY_COUNT_SHIFT 24 +#define TEGRA_I2S_FIFO_SCR_FIFO1_FULL_EMPTY_COUNT_SHIFT 16 +#define TEGRA_I2S_FIFO_SCR_FIFO_FULL_EMPTY_COUNT_MASK 0x3f + +#define TEGRA_I2S_FIFO_SCR_FIFO2_CLR (1 << 12) +#define TEGRA_I2S_FIFO_SCR_FIFO1_CLR (1 << 8) + +#define TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT 0 +#define TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS 1 +#define TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS 2 +#define TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS 3 + +#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT 4 +#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_MASK (3 << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_ONE_SLOT (TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_EIGHT_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) +#define TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_TWELVE_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_SHIFT) + +#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT 0 +#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_MASK (3 << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_ONE_SLOT (TEGRA_I2S_FIFO_ATN_LVL_ONE_SLOT << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_FOUR_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_EIGHT_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_EIGHT_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) +#define TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_TWELVE_SLOTS (TEGRA_I2S_FIFO_ATN_LVL_TWELVE_SLOTS << TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_SHIFT) + +struct tegra_i2s { + struct clk *clk_i2s; + int clk_refs; + struct tegra_pcm_dma_params capture_dma_data; + struct tegra_pcm_dma_params playback_dma_data; + void __iomem *regs; + struct dentry *debug; + u32 reg_ctrl; +}; + +#endif -- cgit v1.2.3 From a50a399b8ba169816d8afae66bfd42fbb65b1973 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 7 Jan 2011 22:36:15 -0700 Subject: ASoC: tegra: Machine utility code Many portions of Tegra ASoC machine drivers will be similar or identical. To avoid cut/paste, this file will act as a repository for all that common code. For now, it solely includes code to reprogram the audio PLL for 44.1KHz- vs. 48KHz-based sample rates. Signed-Off-By: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_asoc_utils.c | 154 +++++++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra_asoc_utils.h | 31 ++++++++ 2 files changed, 185 insertions(+) create mode 100644 sound/soc/tegra/tegra_asoc_utils.c create mode 100644 sound/soc/tegra/tegra_asoc_utils.h (limited to 'sound') diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c new file mode 100644 index 00000000000..711ab7ff5ce --- /dev/null +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -0,0 +1,154 @@ +/* + * tegra_asoc_utils.c - Harmony machine ASoC driver + * + * Author: Stephen Warren + * Copyright (C) 2010 - NVIDIA, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include + +#include "tegra_asoc_utils.h" + +#define PREFIX "ASoC Tegra: " + +static struct clk *clk_pll_a; +static struct clk *clk_pll_a_out0; +static struct clk *clk_cdev1; + +static int set_baseclock, set_mclk; + +int tegra_asoc_utils_set_rate(int srate, int mclk, int *mclk_change) +{ + int new_baseclock; + int err; + + switch (srate) { + case 11025: + case 22050: + case 44100: + case 88200: + new_baseclock = 56448000; + break; + case 8000: + case 16000: + case 32000: + case 48000: + case 64000: + case 96000: + new_baseclock = 73728000; + break; + default: + return -EINVAL; + } + + *mclk_change = ((new_baseclock != set_baseclock) || + (mclk != set_mclk)); + if (!*mclk_change) + return 0; + + set_baseclock = 0; + set_mclk = 0; + + clk_disable(clk_cdev1); + clk_disable(clk_pll_a_out0); + clk_disable(clk_pll_a); + + err = clk_set_rate(clk_pll_a, new_baseclock); + if (err) { + pr_err(PREFIX "Can't set pll_a rate: %d\n", err); + return err; + } + + err = clk_set_rate(clk_pll_a_out0, mclk); + if (err) { + pr_err(PREFIX "Can't set pll_a_out0 rate: %d\n", err); + return err; + } + + /* Don't set cdev1 rate; its locked to pll_a_out0 */ + + err = clk_enable(clk_pll_a); + if (err) { + pr_err(PREFIX "Can't enable pll_a: %d\n", err); + return err; + } + + err = clk_enable(clk_pll_a_out0); + if (err) { + pr_err(PREFIX "Can't enable pll_a_out0: %d\n", err); + return err; + } + + err = clk_enable(clk_cdev1); + if (err) { + pr_err(PREFIX "Can't enable cdev1: %d\n", err); + return err; + } + + set_baseclock = new_baseclock; + set_mclk = mclk; + + return 0; +} + +int tegra_asoc_utils_init(void) +{ + int ret; + + clk_pll_a = clk_get_sys(NULL, "pll_a"); + if (IS_ERR_OR_NULL(clk_pll_a)) { + pr_err(PREFIX "Can't retrieve clk pll_a\n"); + ret = PTR_ERR(clk_pll_a); + goto err; + } + + clk_pll_a_out0 = clk_get_sys(NULL, "pll_a_out0"); + if (IS_ERR_OR_NULL(clk_pll_a_out0)) { + pr_err(PREFIX "Can't retrieve clk pll_a_out0\n"); + ret = PTR_ERR(clk_pll_a_out0); + goto err; + } + + clk_cdev1 = clk_get_sys(NULL, "cdev1"); + if (IS_ERR_OR_NULL(clk_cdev1)) { + pr_err(PREFIX "Can't retrieve clk cdev1\n"); + ret = PTR_ERR(clk_cdev1); + goto err; + } + + return 0; + +err: + if (!IS_ERR_OR_NULL(clk_cdev1)) + clk_put(clk_cdev1); + if (!IS_ERR_OR_NULL(clk_pll_a_out0)) + clk_put(clk_pll_a_out0); + if (!IS_ERR_OR_NULL(clk_pll_a)) + clk_put(clk_pll_a); + return ret; +} + +void tegra_asoc_utils_fini(void) +{ + clk_put(clk_cdev1); + clk_put(clk_pll_a_out0); + clk_put(clk_pll_a); +} + diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h new file mode 100644 index 00000000000..855f8f6e44c --- /dev/null +++ b/sound/soc/tegra/tegra_asoc_utils.h @@ -0,0 +1,31 @@ +/* + * tegra_asoc_utils.h - Definitions for Tegra DAS driver + * + * Author: Stephen Warren + * Copyright (C) 2010 - NVIDIA, Inc. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TEGRA_ASOC_UTILS_H__ +#define __TEGRA_ASOC_UTILS_H_ + +int tegra_asoc_utils_set_rate(int srate, int mclk_rate, int *mclk_change); +int tegra_asoc_utils_init(void); +void tegra_asoc_utils_fini(void); + +#endif + -- cgit v1.2.3 From a8bf1ba1c803d63311a2fb3c38c172df6a606faa Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 7 Jan 2011 22:36:16 -0700 Subject: ASoC: tegra: Harmony machine support Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/harmony.c | 179 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 179 insertions(+) create mode 100644 sound/soc/tegra/harmony.c (limited to 'sound') diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c new file mode 100644 index 00000000000..bf0dbafbf6d --- /dev/null +++ b/sound/soc/tegra/harmony.c @@ -0,0 +1,179 @@ +/* + * harmony.c - Harmony machine ASoC driver + * + * Author: Stephen Warren + * Copyright (C) 2010 - NVIDIA, Inc. + * + * Based on code copyright/by: + * + * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd. + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include +#include + +#include "tegra_das.h" +#include "tegra_i2s.h" +#include "tegra_pcm.h" +#include "tegra_asoc_utils.h" + +#define PREFIX "ASoC Harmony: " + +static struct platform_device *harmony_snd_device; + +static int harmony_asoc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int srate, mclk, mclk_change; + int err; + + srate = params_rate(params); + switch (srate) { + case 64000: + case 88200: + case 96000: + mclk = 128 * srate; + break; + default: + mclk = 256 * srate; + break; + } + /* FIXME: Codec only requires >= 3MHz if OSR==0 */ + while (mclk < 6000000) + mclk *= 2; + + err = tegra_asoc_utils_set_rate(srate, mclk, &mclk_change); + if (err < 0) { + pr_err(PREFIX "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (err < 0) { + pr_err(PREFIX "codec_dai fmt not set\n"); + return err; + } + + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (err < 0) { + pr_err(PREFIX "cpu_dai fmt not set\n"); + return err; + } + + if (mclk_change) { + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, SND_SOC_CLOCK_IN); + if (err < 0) { + pr_err(PREFIX "codec_dai clock not set\n"); + return err; + } + } + + return 0; +} + +static struct snd_soc_ops harmony_asoc_ops = { + .hw_params = harmony_asoc_hw_params, +}; + +static struct snd_soc_dai_link harmony_wm8903_dai = { + .name = "WM8903", + .stream_name = "WM8903 PCM", + .codec_name = "wm8903-codec.0-001a", + .platform_name = "tegra-pcm-audio", + .cpu_dai_name = "tegra-i2s.0", + .codec_dai_name = "wm8903-hifi", + .ops = &harmony_asoc_ops, +}; + +static struct snd_soc_card snd_soc_harmony = { + .name = "tegra-harmony", + .dai_link = &harmony_wm8903_dai, + .num_links = 1, +}; + +static int __init harmony_soc_modinit(void) +{ + int ret; + + if (!machine_is_harmony()) { + pr_err(PREFIX "Not running on Tegra Harmony!\n"); + return -ENODEV; + } + + ret = tegra_asoc_utils_init(); + if (ret) { + return ret; + } + + /* + * Create and register platform device + */ + harmony_snd_device = platform_device_alloc("soc-audio", -1); + if (harmony_snd_device == NULL) { + pr_err(PREFIX "platform_device_alloc failed\n"); + ret = -ENOMEM; + goto err_clock_utils; + } + + platform_set_drvdata(harmony_snd_device, &snd_soc_harmony); + + ret = platform_device_add(harmony_snd_device); + if (ret) { + pr_err(PREFIX "platform_device_add failed (%d)\n", + ret); + goto err_device_put; + } + + return 0; + +err_device_put: + platform_device_put(harmony_snd_device); +err_clock_utils: + tegra_asoc_utils_fini(); + return ret; +} +module_init(harmony_soc_modinit); + +static void __exit harmony_soc_modexit(void) +{ + platform_device_unregister(harmony_snd_device); + + tegra_asoc_utils_fini(); +} +module_exit(harmony_soc_modexit); + +MODULE_AUTHOR("Stephen Warren "); +MODULE_DESCRIPTION("Harmony machine ASoC driver"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 8b75d714a6ae6445841006ae2074dae095608c36 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 7 Jan 2011 22:36:17 -0700 Subject: ASoC: tegra: Kconfig and Makefile Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/tegra/Kconfig | 27 +++++++++++++++++++++++++++ sound/soc/tegra/Makefile | 14 ++++++++++++++ 4 files changed, 43 insertions(+) create mode 100644 sound/soc/tegra/Kconfig create mode 100644 sound/soc/tegra/Makefile (limited to 'sound') diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 821ab0b759b..8224db5f043 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -55,6 +55,7 @@ source "sound/soc/pxa/Kconfig" source "sound/soc/samsung/Kconfig" source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" +source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" # Supported codecs diff --git a/sound/soc/Makefile b/sound/soc/Makefile index df83d2143af..1ed61c5df2c 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -18,4 +18,5 @@ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += samsung/ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ +obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig new file mode 100644 index 00000000000..5514f1c7ca2 --- /dev/null +++ b/sound/soc/tegra/Kconfig @@ -0,0 +1,27 @@ +config SND_TEGRA_SOC + tristate "SoC Audio for the Tegra System-on-Chip" + depends on ARCH_TEGRA && TEGRA_SYSTEM_DMA + default m + help + Say Y or M here if you want support for SoC audio on Tegra. + +config SND_TEGRA_SOC_I2S + tristate + depends on SND_TEGRA_SOC + depends on !TEGRA_I2S_AUDIO + default m + help + Say Y or M if you want to add support for codecs attached to the + Tegra I2S interface. You will also need to select the individual + machine drivers to support below. + +config SND_TEGRA_SOC_HARMONY + tristate "SoC Audio support for Tegra Harmony reference board" + depends on SND_TEGRA_SOC && MACH_HARMONY && I2C + default m + select SND_TEGRA_SOC_I2S + select SND_SOC_WM8903 + help + Say Y or M here if you want to add support for SoC audio on the + Tegra Harmony reference board. + diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile new file mode 100644 index 00000000000..dfd2ab9d975 --- /dev/null +++ b/sound/soc/tegra/Makefile @@ -0,0 +1,14 @@ +# Tegra platform Support +snd-soc-tegra-das-objs := tegra_das.o +snd-soc-tegra-pcm-objs := tegra_pcm.o +snd-soc-tegra-i2s-objs := tegra_i2s.o + +obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-das.o +obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-pcm.o +obj-$(CONFIG_SND_TEGRA_SOC_I2S) += snd-soc-tegra-i2s.o + +# Tegra machine Support +snd-soc-tegra-harmony-objs := harmony.o +snd-soc-tegra-harmony-objs += tegra_asoc_utils.o + +obj-$(CONFIG_SND_TEGRA_SOC_HARMONY) += snd-soc-tegra-harmony.o -- cgit v1.2.3 From 8a9dab1a555e3f2088c68cae792dfd7e854e65e4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Jan 2011 22:25:21 +0000 Subject: ASoC: Update name of debugfs root symbol to snd_soc_ Everything else is using snd_soc_ so we should use it here too. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 20 ++++++++++---------- sound/soc/tegra/tegra_das.c | 5 +++-- sound/soc/tegra/tegra_i2s.c | 2 +- 3 files changed, 14 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0484e504b58..2ff708a4111 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -48,8 +48,8 @@ static DEFINE_MUTEX(pcm_mutex); static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); #ifdef CONFIG_DEBUG_FS -struct dentry *asoc_debugfs_root; -EXPORT_SYMBOL_GPL(asoc_debugfs_root); +struct dentry *snd_soc_debugfs_root; +EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); #endif static DEFINE_MUTEX(client_mutex); @@ -361,7 +361,7 @@ static const struct file_operations platform_list_fops = { static void soc_init_card_debugfs(struct snd_soc_card *card) { card->debugfs_card_root = debugfs_create_dir(card->name, - asoc_debugfs_root); + snd_soc_debugfs_root); if (!card->debugfs_card_root) { dev_warn(card->dev, "ASoC: Failed to create codec debugfs directory\n"); @@ -3584,22 +3584,22 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); static int __init snd_soc_init(void) { #ifdef CONFIG_DEBUG_FS - asoc_debugfs_root = debugfs_create_dir("asoc", NULL); - if (IS_ERR(asoc_debugfs_root) || !asoc_debugfs_root) { + snd_soc_debugfs_root = debugfs_create_dir("asoc", NULL); + if (IS_ERR(snd_soc_debugfs_root) || !snd_soc_debugfs_root) { printk(KERN_WARNING "ASoC: Failed to create debugfs directory\n"); - asoc_debugfs_root = NULL; + snd_soc_debugfs_root = NULL; } - if (!debugfs_create_file("codecs", 0444, asoc_debugfs_root, NULL, + if (!debugfs_create_file("codecs", 0444, snd_soc_debugfs_root, NULL, &codec_list_fops)) pr_warn("ASoC: Failed to create CODEC list debugfs file\n"); - if (!debugfs_create_file("dais", 0444, asoc_debugfs_root, NULL, + if (!debugfs_create_file("dais", 0444, snd_soc_debugfs_root, NULL, &dai_list_fops)) pr_warn("ASoC: Failed to create DAI list debugfs file\n"); - if (!debugfs_create_file("platforms", 0444, asoc_debugfs_root, NULL, + if (!debugfs_create_file("platforms", 0444, snd_soc_debugfs_root, NULL, &platform_list_fops)) pr_warn("ASoC: Failed to create platform list debugfs file\n"); #endif @@ -3611,7 +3611,7 @@ module_init(snd_soc_init); static void __exit snd_soc_exit(void) { #ifdef CONFIG_DEBUG_FS - debugfs_remove_recursive(asoc_debugfs_root); + debugfs_remove_recursive(snd_soc_debugfs_root); #endif platform_driver_unregister(&soc_driver); } diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c index 796d36d5188..01eb9c9301d 100644 --- a/sound/soc/tegra/tegra_das.c +++ b/sound/soc/tegra/tegra_das.c @@ -144,8 +144,9 @@ static const struct file_operations tegra_das_debug_fops = { static void tegra_das_debug_add(struct tegra_das *das) { - das->debug = debugfs_create_file(DRV_NAME, S_IRUGO, asoc_debugfs_root, - das, &tegra_das_debug_fops); + das->debug = debugfs_create_file(DRV_NAME, S_IRUGO, + snd_soc_debugfs_root, das, + &tegra_das_debug_fops); } static void tegra_das_debug_remove(struct tegra_das *das) diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 9b7a22af52a..1730509c8ac 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -104,7 +104,7 @@ static void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id) char name[] = DRV_NAME ".0"; snprintf(name, sizeof(name), DRV_NAME".%1d", id); - i2s->debug = debugfs_create_file(name, S_IRUGO, asoc_debugfs_root, + i2s->debug = debugfs_create_file(name, S_IRUGO, snd_soc_debugfs_root, i2s, &tegra_i2s_debug_fops); } -- cgit v1.2.3 From 5c68536f32a46c8ccc98df3d9b154e1f33f72447 Mon Sep 17 00:00:00 2001 From: Harsha Priya Date: Tue, 11 Jan 2011 14:50:35 +0530 Subject: ASoC: sst_platform created helper functions Few funtions can be modularized in this driver to make them look cleaner like managing the stream status with locks and filling pcm parameters. This patch adds helper functions to do the same. Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/mid-x86/sst_platform.c | 123 +++++++++++++++++---------------------- 1 file changed, 53 insertions(+), 70 deletions(-) (limited to 'sound') diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 0e994c16e7a..a4e3fa3761f 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -105,7 +105,45 @@ struct snd_soc_dai_driver sst_platform_dai[] = { }, }, }; + /* helper functions */ +static inline void sst_set_stream_status(struct sst_runtime_stream *stream, + int state) +{ + spin_lock(&stream->status_lock); + stream->stream_status = state; + spin_unlock(&stream->status_lock); +} + +static inline int sst_get_stream_status(struct sst_runtime_stream *stream) +{ + int state; + + spin_lock(&stream->status_lock); + state = stream->stream_status; + spin_unlock(&stream->status_lock); + return state; +} + +static void sst_fill_pcm_params(struct snd_pcm_substream *substream, + struct snd_sst_stream_params *param) +{ + + param->uc.pcm_params.codec = SST_CODEC_TYPE_PCM; + param->uc.pcm_params.num_chan = (u8) substream->runtime->channels; + param->uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits; + param->uc.pcm_params.reserved = 0; + param->uc.pcm_params.sfreq = substream->runtime->rate; + param->uc.pcm_params.ring_buffer_size = + snd_pcm_lib_buffer_bytes(substream); + param->uc.pcm_params.period_count = substream->runtime->period_size; + param->uc.pcm_params.ring_buffer_addr = + virt_to_phys(substream->dma_buffer.area); + pr_debug("period_cnt = %d\n", param->uc.pcm_params.period_count); + pr_debug("sfreq= %d, wd_sz = %d\n", + param->uc.pcm_params.sfreq, param->uc.pcm_params.pcm_wd_sz); +} + static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) { struct sst_runtime_stream *stream = @@ -115,26 +153,10 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) int ret_val; /* set codec params and inform SST driver the same */ - - param.uc.pcm_params.codec = SST_CODEC_TYPE_PCM; - param.uc.pcm_params.num_chan = (u8) substream->runtime->channels; - param.uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits; - param.uc.pcm_params.reserved = 0; - param.uc.pcm_params.sfreq = substream->runtime->rate; - param.uc.pcm_params.ring_buffer_size = - snd_pcm_lib_buffer_bytes(substream); - param.uc.pcm_params.period_count = substream->runtime->period_size; - param.uc.pcm_params.ring_buffer_addr = - virt_to_phys(substream->dma_buffer.area); + sst_fill_pcm_params(substream, ¶m); substream->runtime->dma_area = substream->dma_buffer.area; - - pr_debug("period_cnt = %d\n", param.uc.pcm_params.period_count); - pr_debug("sfreq= %d, wd_sz = %d\n", - param.uc.pcm_params.sfreq, param.uc.pcm_params.pcm_wd_sz); - str_params.sparams = param; - str_params.codec = SST_CODEC_TYPE_PCM; - + str_params.codec = param.uc.pcm_params.codec; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { str_params.ops = STREAM_OPS_PLAYBACK; str_params.device_type = substream->pcm->device + 1; @@ -153,28 +175,23 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) stream->stream_info.str_id = ret_val; pr_debug("str id : %d\n", stream->stream_info.str_id); - return ret_val; } - static void sst_period_elapsed(void *mad_substream) { struct snd_pcm_substream *substream = mad_substream; struct sst_runtime_stream *stream; + int status; if (!substream || !substream->runtime) return; stream = substream->runtime->private_data; if (!stream) return; - - spin_lock(&stream->status_lock); - if (stream->stream_status != SST_PLATFORM_RUNNING) { - spin_unlock(&stream->status_lock); + status = sst_get_stream_status(stream); + if (status != SST_PLATFORM_RUNNING) return; - } - spin_unlock(&stream->status_lock); snd_pcm_period_elapsed(substream); } @@ -185,9 +202,7 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) int ret_val; pr_debug("setting buffer ptr param\n"); - spin_lock(&stream->status_lock); - stream->stream_status = SST_PLATFORM_INIT; - spin_unlock(&stream->status_lock); + sst_set_stream_status(stream, SST_PLATFORM_INIT); stream->stream_info.period_elapsed = sst_period_elapsed; stream->stream_info.mad_substream = substream; stream->stream_info.buffer_ptr = 0; @@ -208,21 +223,14 @@ static int sst_platform_open(struct snd_pcm_substream *substream) int ret_val = 0; pr_debug("sst_platform_open called\n"); - runtime = substream->runtime; runtime->hw = sst_platform_pcm_hw; - stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) return -ENOMEM; - spin_lock_init(&stream->status_lock); stream->stream_info.str_id = 0; - - spin_lock(&stream->status_lock); - stream->stream_status = SST_PLATFORM_INIT; - spin_unlock(&stream->status_lock); - + sst_set_stream_status(stream, SST_PLATFORM_INIT); stream->stream_info.mad_substream = substream; /* allocate memory for SST API set */ stream->sstdrv_ops = kzalloc(sizeof(*stream->sstdrv_ops), @@ -233,7 +241,6 @@ static int sst_platform_open(struct snd_pcm_substream *substream) return -ENOMEM; } stream->sstdrv_ops->vendor_id = MSIC_VENDOR_ID; - /* registering with SST driver to get access to SST APIs to use */ ret_val = register_sst_card(stream->sstdrv_ops); if (ret_val) { @@ -251,14 +258,11 @@ static int sst_platform_close(struct snd_pcm_substream *substream) int ret_val = 0, str_id; pr_debug("sst_platform_close called\n"); - stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; - if (str_id) ret_val = stream->sstdrv_ops->control_set( SST_SND_FREE, &str_id); - kfree(stream->sstdrv_ops); kfree(stream); return ret_val; @@ -270,7 +274,6 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) int ret_val = 0, str_id; pr_debug("sst_platform_pcm_prepare called\n"); - stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (stream->stream_info.str_id) { @@ -299,11 +302,8 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, struct sst_runtime_stream *stream; pr_debug("sst_platform_pcm_trigger called\n"); - stream = substream->runtime->private_data; - str_id = stream->stream_info.str_id; - switch (cmd) { case SNDRV_PCM_TRIGGER_START: pr_debug("sst: Trigger Start\n"); @@ -311,9 +311,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, SST_SND_START, &str_id); if (ret_val) break; - spin_lock(&stream->status_lock); - stream->stream_status = SST_PLATFORM_RUNNING; - spin_unlock(&stream->status_lock); + sst_set_stream_status(stream, SST_PLATFORM_RUNNING); stream->stream_info.mad_substream = substream; break; case SNDRV_PCM_TRIGGER_STOP: @@ -322,9 +320,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, SST_SND_DROP, &str_id); if (ret_val) break; - spin_lock(&stream->status_lock); - stream->stream_status = SST_PLATFORM_DROPPED; - spin_unlock(&stream->status_lock); + sst_set_stream_status(stream, SST_PLATFORM_DROPPED); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: pr_debug("sst: in pause\n"); @@ -332,9 +328,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, SST_SND_PAUSE, &str_id); if (ret_val) break; - spin_lock(&stream->status_lock); - stream->stream_status = SST_PLATFORM_PAUSED; - spin_unlock(&stream->status_lock); + sst_set_stream_status(stream, SST_PLATFORM_PAUSED); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: pr_debug("sst: in pause release\n"); @@ -342,9 +336,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, SST_SND_RESUME, &str_id); if (ret_val) break; - spin_lock(&stream->status_lock); - stream->stream_status = SST_PLATFORM_RUNNING; - spin_unlock(&stream->status_lock); + sst_set_stream_status(stream, SST_PLATFORM_RUNNING); break; default: ret_val = -EINVAL; @@ -357,18 +349,13 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer (struct snd_pcm_substream *substream) { struct sst_runtime_stream *stream; - int ret_val; + int ret_val, status; struct pcm_stream_info *str_info; - stream = substream->runtime->private_data; - spin_lock(&stream->status_lock); - if (stream->stream_status == SST_PLATFORM_INIT) { - spin_unlock(&stream->status_lock); + status = sst_get_stream_status(stream); + if (status == SST_PLATFORM_INIT) return 0; - } - spin_unlock(&stream->status_lock); - str_info = &stream->stream_info; ret_val = stream->sstdrv_ops->control_set( SST_SND_BUFFER_POINTER, str_info); @@ -376,7 +363,6 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer pr_err("sst: error code = %d\n", ret_val); return ret_val; } - return stream->stream_info.buffer_ptr; } @@ -402,7 +388,6 @@ int sst_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, int retval = 0; pr_debug("sst_pcm_new called\n"); - if (dai->driver->playback.channels_min || dai->driver->capture.channels_min) { retval = snd_pcm_lib_preallocate_pages_for_all(pcm, @@ -414,7 +399,6 @@ int sst_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, return retval; } } - return retval; } struct snd_soc_platform_driver sst_soc_platform_drv = { @@ -433,6 +417,7 @@ static int sst_platform_probe(struct platform_device *pdev) pr_err("registering soc platform failed\n"); return ret; } + ret = snd_soc_register_dais(&pdev->dev, sst_platform_dai, ARRAY_SIZE(sst_platform_dai)); if (ret) { @@ -446,10 +431,8 @@ static int sst_platform_remove(struct platform_device *pdev) { snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sst_platform_dai)); - snd_soc_unregister_platform(&pdev->dev); pr_debug("sst_platform_remove sucess\n"); - return 0; } -- cgit v1.2.3 From 4fde768ecf39360eaed44152c2556dd10c3e0ce7 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Tue, 11 Jan 2011 09:28:32 +0000 Subject: ASoC: WM8995: Remember to flush the cache on resume Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 3d2110c1d81..96e0dc09f20 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1498,6 +1498,7 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: snd_soc_update_bits(codec, WM8995_POWER_MANAGEMENT_1, WM8995_BG_ENA_MASK, 0); + codec->cache_sync = 1; break; } -- cgit v1.2.3 From 180c329d6b0db49687c88303e7f3754e7afdc3bf Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Mon, 10 Jan 2011 15:58:13 -0600 Subject: ASoC: let snd_soc_update_bits() return an error code Update snd_soc_update_bits() so that it returns a negative error code if the the read or write operation fails. Note that currently, a lot of the lower-level read functions have an unsigned integer return type (and some of them even try to return a negative number), but this code still appears to work in those cases. An examination of the code shows that all current callers are compatible with this change. Signed-off-by: Timur Tabi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2ff708a4111..6cc68140fa6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2137,19 +2137,27 @@ EXPORT_SYMBOL_GPL(snd_soc_write); * * Writes new register value. * - * Returns 1 for change else 0. + * Returns 1 for change, 0 for no change, or negative error code. */ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, unsigned int mask, unsigned int value) { int change; unsigned int old, new; + int ret; - old = snd_soc_read(codec, reg); + ret = snd_soc_read(codec, reg); + if (ret < 0) + return ret; + + old = ret; new = (old & ~mask) | value; change = old != new; - if (change) - snd_soc_write(codec, reg, new); + if (change) { + ret = snd_soc_write(codec, reg, new); + if (ret < 0) + return ret; + } return change; } -- cgit v1.2.3 From 0378b6acc8ee20720dce23d5157185e09189d41c Mon Sep 17 00:00:00 2001 From: Seungwhan Youn Date: Tue, 11 Jan 2011 07:26:06 +0900 Subject: ASoC: SAMSUNG: Clean-up header includes This patch remove including unnecessary/duplicated headers which relative with Samsung SoCs. Signed-off-by: Seungwhan Youn Acked-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/samsung/ac97.c | 4 +--- sound/soc/samsung/dma.c | 8 +------- sound/soc/samsung/goni_wm8994.c | 10 +--------- sound/soc/samsung/h1940_uda1380.c | 9 --------- sound/soc/samsung/i2s.c | 3 +-- sound/soc/samsung/jive_wm8750.c | 11 ----------- sound/soc/samsung/ln2440sbc_alc650.c | 6 ------ sound/soc/samsung/neo1973_gta02_wm8753.c | 14 ++------------ sound/soc/samsung/pcm.c | 11 +---------- sound/soc/samsung/rx1950_uda1380.c | 11 ----------- sound/soc/samsung/s3c-i2s-v2.c | 3 +-- sound/soc/samsung/s3c2412-i2s.c | 12 +----------- sound/soc/samsung/s3c24xx-i2s.c | 14 +------------- sound/soc/samsung/s3c24xx_simtec.c | 7 ------- sound/soc/samsung/s3c24xx_simtec_hermes.c | 10 ---------- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 12 ------------ sound/soc/samsung/s3c24xx_uda134x.c | 9 +-------- sound/soc/samsung/smartq_wm8987.c | 6 ------ sound/soc/samsung/smdk2443_wm9710.c | 6 ------ sound/soc/samsung/smdk_spdif.c | 5 ----- sound/soc/samsung/smdk_wm8580.c | 7 +------ sound/soc/samsung/smdk_wm9713.c | 4 ---- sound/soc/samsung/spdif.c | 3 +-- 23 files changed, 13 insertions(+), 172 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 1e84ee01eac..f97110e72e8 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -12,16 +12,14 @@ * published by the Free Software Foundation. */ -#include -#include #include #include #include #include -#include #include +#include #include #include "dma.h" diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index c2cdb4597e5..9bce1df1f0d 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -14,17 +14,11 @@ * option) any later version. */ -#include -#include -#include -#include #include #include -#include -#include -#include #include +#include #include #include diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c index 34dd9ef1b9c..f6b3a3ce591 100644 --- a/sound/soc/samsung/goni_wm8994.c +++ b/sound/soc/samsung/goni_wm8994.c @@ -11,21 +11,13 @@ * */ -#include -#include -#include -#include #include #include + #include #include -#include -#include -#include #include "../codecs/wm8994.h" -#include "dma.h" -#include "i2s.h" #define MACHINE_NAME 0 #define CPU_VOICE_DAI 1 diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index c45f7ce14d6..241f55d0066 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -13,25 +13,16 @@ * */ -#include -#include -#include -#include #include #include -#include #include #include - #include - #include -#include "dma.h" #include "s3c24xx-i2s.h" -#include "../codecs/uda1380.h" static unsigned int rates[] = { 11025, diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index d00ac3a7102..ffa09b3b2ca 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -15,9 +15,8 @@ #include #include -#include -#include #include +#include #include diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 08802520e01..3b53ad54bc3 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -11,22 +11,11 @@ * published by the Free Software Foundation. */ -#include -#include -#include -#include -#include -#include - -#include -#include #include #include -#include "dma.h" #include "s3c2412-i2s.h" - #include "../codecs/wm8750.h" static const struct snd_soc_dapm_route audio_map[] = { diff --git a/sound/soc/samsung/ln2440sbc_alc650.c b/sound/soc/samsung/ln2440sbc_alc650.c index 02414290169..bd91c19a6c0 100644 --- a/sound/soc/samsung/ln2440sbc_alc650.c +++ b/sound/soc/samsung/ln2440sbc_alc650.c @@ -16,14 +16,8 @@ * */ -#include -#include -#include -#include #include -#include "dma.h" - static struct snd_soc_card ln2440sbc; static struct snd_soc_dai_link ln2440sbc_dai[] = { diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c index 3eec610c10f..69e08fd1ff0 100644 --- a/sound/soc/samsung/neo1973_gta02_wm8753.c +++ b/sound/soc/samsung/neo1973_gta02_wm8753.c @@ -13,25 +13,15 @@ * option) any later version. */ -#include -#include -#include -#include -#include #include -#include -#include + #include #include - #include - -#include -#include #include + #include "../codecs/wm8753.h" -#include "dma.h" #include "s3c24xx-i2s.h" static struct snd_soc_card neo1973_gta02; diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 091db5dce07..38aac7d57a5 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -11,20 +11,11 @@ * published by the Free Software Foundation. */ -#include -#include -#include -#include #include -#include -#include #include -#include -#include -#include -#include #include +#include #include #include diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 5a4587e0288..5f2479ce9dd 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -17,26 +17,15 @@ * */ -#include -#include -#include -#include #include -#include #include -#include #include #include - -#include - #include -#include "dma.h" #include "s3c24xx-i2s.h" -#include "../codecs/uda1380.h" static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd); static int rx1950_startup(struct snd_pcm_substream *substream); diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 094f36e41e8..52074a2b069 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -20,9 +20,8 @@ #include #include -#include -#include #include +#include #include diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 7ea83786712..841ab14c110 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -16,21 +16,13 @@ * option) any later version. */ -#include -#include -#include #include #include #include -#include #include -#include -#include -#include -#include #include -#include +#include #include #include @@ -39,8 +31,6 @@ #include "regs-i2s-v2.h" #include "s3c2412-i2s.h" -#define S3C2412_I2S_DEBUG 0 - static struct s3c2410_dma_client s3c2412_dma_client_out = { .name = "I2S PCM Stereo out" }; diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 13e41ed8e22..63d8849d80b 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -14,28 +14,16 @@ * option) any later version. */ -#include -#include -#include #include #include -#include #include #include -#include -#include -#include -#include #include +#include -#include #include -#include - -#include #include - #include #include "dma.h" diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c index a434032d183..349566f0686 100644 --- a/sound/soc/samsung/s3c24xx_simtec.c +++ b/sound/soc/samsung/s3c24xx_simtec.c @@ -7,20 +7,13 @@ * published by the Free Software Foundation. */ -#include -#include -#include #include #include -#include -#include -#include #include #include -#include "dma.h" #include "s3c24xx-i2s.h" #include "s3c24xx_simtec.h" diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c index bb4292e3596..d7b3e6e9d78 100644 --- a/sound/soc/samsung/s3c24xx_simtec_hermes.c +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -7,18 +7,8 @@ * published by the Free Software Foundation. */ -#include -#include -#include - -#include -#include #include -#include - -#include "dma.h" -#include "s3c24xx-i2s.h" #include "s3c24xx_simtec.h" static const struct snd_soc_dapm_widget dapm_widgets[] = { diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index fbba4e36772..ff6168f5f2e 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -7,22 +7,10 @@ * published by the Free Software Foundation. */ -#include -#include -#include - -#include -#include #include -#include - -#include "dma.h" -#include "s3c24xx-i2s.h" #include "s3c24xx_simtec.h" -#include "../codecs/tlv320aic23.h" - /* supported machines: * * Machine Connections AMP diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index cdc8ecbcb8e..ce749a10ec0 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -11,22 +11,15 @@ * published by the Free Software Foundation. */ -#include #include -#include #include -#include -#include + #include #include -#include #include -#include "dma.h" #include "s3c24xx-i2s.h" -#include "../codecs/uda134x.h" - /* #define ENFORCE_RATES 1 */ /* diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 61e2b5219d4..0a2c4f22303 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -13,20 +13,14 @@ * */ -#include -#include #include -#include -#include #include #include #include -#include "dma.h" #include "i2s.h" - #include "../codecs/wm8750.h" /* diff --git a/sound/soc/samsung/smdk2443_wm9710.c b/sound/soc/samsung/smdk2443_wm9710.c index 4bd8c0b6396..3a0dbfc793f 100644 --- a/sound/soc/samsung/smdk2443_wm9710.c +++ b/sound/soc/samsung/smdk2443_wm9710.c @@ -12,14 +12,8 @@ * */ -#include -#include -#include -#include #include -#include "dma.h" - static struct snd_soc_card smdk2443; static struct snd_soc_dai_link smdk2443_dai[] = { diff --git a/sound/soc/samsung/smdk_spdif.c b/sound/soc/samsung/smdk_spdif.c index cb2f4d04ab7..d42fe8df714 100644 --- a/sound/soc/samsung/smdk_spdif.c +++ b/sound/soc/samsung/smdk_spdif.c @@ -10,15 +10,10 @@ * */ -#include -#include #include -#include - #include -#include "dma.h" #include "spdif.h" /* Audio clock settings are belonged to board specific part. Every diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index b2cff1a44ae..8aacf23d6f3 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -10,17 +10,12 @@ * option) any later version. */ -#include -#include -#include -#include -#include #include +#include #include #include "../codecs/wm8580.h" -#include "dma.h" #include "i2s.h" /* diff --git a/sound/soc/samsung/smdk_wm9713.c b/sound/soc/samsung/smdk_wm9713.c index d58d86e7e19..fffe3c1dd1b 100644 --- a/sound/soc/samsung/smdk_wm9713.c +++ b/sound/soc/samsung/smdk_wm9713.c @@ -11,12 +11,8 @@ * */ -#include -#include #include -#include "dma.h" - static struct snd_soc_card smdk; /* diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index f0816404ea3..28c491dacf7 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -13,9 +13,8 @@ #include #include -#include -#include #include +#include #include #include -- cgit v1.2.3 From 1321e8838bf9bbc883450675d5c501cb8510f1a4 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Tue, 11 Jan 2011 11:29:49 +0000 Subject: ASoC: soc-cache: Clean up the cache manipulation code Use Takashi's clean up code to make the cache manipulation code more readable. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 230 ++++++++++++++++++-------------------------------- 1 file changed, 82 insertions(+), 148 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index a9ebc078bea..9e534429ea6 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -761,6 +761,49 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); +static bool snd_soc_set_cache_val(void *base, unsigned int idx, + unsigned int val, unsigned int word_size) +{ + switch (word_size) { + case 1: { + u8 *cache = base; + if (cache[idx] == val) + return true; + cache[idx] = val; + break; + } + case 2: { + u16 *cache = base; + if (cache[idx] == val) + return true; + cache[idx] = val; + break; + } + default: + BUG(); + } + return false; +} + +static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx, + unsigned int word_size) +{ + switch (word_size) { + case 1: { + const u8 *cache = base; + return cache[idx]; + } + case 2: { + const u16 *cache = base; + return cache[idx]; + } + default: + BUG(); + } + /* unreachable */ + return -1; +} + struct snd_soc_rbtree_node { struct rb_node node; unsigned int reg; @@ -924,7 +967,12 @@ static int snd_soc_rbtree_cache_exit(struct snd_soc_codec *codec) static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec) { + struct snd_soc_rbtree_node *rbtree_node; struct snd_soc_rbtree_ctx *rbtree_ctx; + unsigned int val; + unsigned int word_size; + int i; + int ret; codec->reg_cache = kmalloc(sizeof *rbtree_ctx, GFP_KERNEL); if (!codec->reg_cache) @@ -936,53 +984,25 @@ static int snd_soc_rbtree_cache_init(struct snd_soc_codec *codec) if (!codec->reg_def_copy) return 0; -/* - * populate the rbtree with the initialized registers. All other - * registers will be inserted into the tree when they are first written. - * - * The reasoning behind this, is that we need to step through and - * dereference the cache in u8/u16 increments without sacrificing - * portability. This could also be done using memcpy() but that would - * be slightly more cryptic. - */ -#define snd_soc_rbtree_populate(cache) \ -({ \ - int ret, i; \ - struct snd_soc_rbtree_node *rbtree_node; \ - \ - ret = 0; \ - cache = codec->reg_def_copy; \ - for (i = 0; i < codec->driver->reg_cache_size; ++i) { \ - if (!cache[i]) \ - continue; \ - rbtree_node = kzalloc(sizeof *rbtree_node, GFP_KERNEL); \ - if (!rbtree_node) { \ - ret = -ENOMEM; \ - snd_soc_cache_exit(codec); \ - break; \ - } \ - rbtree_node->reg = i; \ - rbtree_node->value = cache[i]; \ - rbtree_node->defval = cache[i]; \ - snd_soc_rbtree_insert(&rbtree_ctx->root, \ - rbtree_node); \ - } \ - ret; \ -}) - - switch (codec->driver->reg_word_size) { - case 1: { - const u8 *cache; - - return snd_soc_rbtree_populate(cache); - } - case 2: { - const u16 *cache; - - return snd_soc_rbtree_populate(cache); - } - default: - BUG(); + /* + * populate the rbtree with the initialized registers. All other + * registers will be inserted when they are first modified. + */ + word_size = codec->driver->reg_word_size; + for (i = 0; i < codec->driver->reg_cache_size; ++i) { + val = snd_soc_get_cache_val(codec->reg_def_copy, i, word_size); + if (!val) + continue; + rbtree_node = kzalloc(sizeof *rbtree_node, GFP_KERNEL); + if (!rbtree_node) { + ret = -ENOMEM; + snd_soc_cache_exit(codec); + break; + } + rbtree_node->reg = i; + rbtree_node->value = val; + rbtree_node->defval = val; + snd_soc_rbtree_insert(&rbtree_ctx->root, rbtree_node); } return 0; @@ -1165,29 +1185,10 @@ static int snd_soc_lzo_cache_write(struct snd_soc_codec *codec, } /* write the new value to the cache */ - switch (codec->driver->reg_word_size) { - case 1: { - u8 *cache; - cache = lzo_block->dst; - if (cache[blkpos] == value) { - kfree(lzo_block->dst); - goto out; - } - cache[blkpos] = value; - } - break; - case 2: { - u16 *cache; - cache = lzo_block->dst; - if (cache[blkpos] == value) { - kfree(lzo_block->dst); - goto out; - } - cache[blkpos] = value; - } - break; - default: - BUG(); + if (snd_soc_set_cache_val(lzo_block->dst, blkpos, value, + codec->driver->reg_word_size)) { + kfree(lzo_block->dst); + goto out; } /* prepare the source to be the decompressed block */ @@ -1241,25 +1242,10 @@ static int snd_soc_lzo_cache_read(struct snd_soc_codec *codec, /* decompress the block */ ret = snd_soc_lzo_decompress_cache_block(codec, lzo_block); - if (ret >= 0) { + if (ret >= 0) /* fetch the value from the cache */ - switch (codec->driver->reg_word_size) { - case 1: { - u8 *cache; - cache = lzo_block->dst; - *value = cache[blkpos]; - } - break; - case 2: { - u16 *cache; - cache = lzo_block->dst; - *value = cache[blkpos]; - } - break; - default: - BUG(); - } - } + *value = snd_soc_get_cache_val(lzo_block->dst, blkpos, + codec->driver->reg_word_size); kfree(lzo_block->dst); /* restore the pointer and length of the compressed block */ @@ -1414,28 +1400,10 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) ret = snd_soc_cache_read(codec, i, &val); if (ret) return ret; - if (codec_drv->reg_cache_default) { - switch (codec_drv->reg_word_size) { - case 1: { - const u8 *cache; - - cache = codec_drv->reg_cache_default; - if (cache[i] == val) - continue; - } - break; - case 2: { - const u16 *cache; - - cache = codec_drv->reg_cache_default; - if (cache[i] == val) - continue; - } - break; - default: - BUG(); - } - } + if (codec_drv->reg_cache_default) + if (snd_soc_get_cache_val(codec_drv->reg_cache_default, + i, codec_drv->reg_word_size) == val) + continue; ret = snd_soc_write(codec, i, val); if (ret) return ret; @@ -1448,50 +1416,16 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) static int snd_soc_flat_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - switch (codec->driver->reg_word_size) { - case 1: { - u8 *cache; - - cache = codec->reg_cache; - cache[reg] = value; - } - break; - case 2: { - u16 *cache; - - cache = codec->reg_cache; - cache[reg] = value; - } - break; - default: - BUG(); - } - + snd_soc_set_cache_val(codec->reg_cache, reg, value, + codec->driver->reg_word_size); return 0; } static int snd_soc_flat_cache_read(struct snd_soc_codec *codec, unsigned int reg, unsigned int *value) { - switch (codec->driver->reg_word_size) { - case 1: { - u8 *cache; - - cache = codec->reg_cache; - *value = cache[reg]; - } - break; - case 2: { - u16 *cache; - - cache = codec->reg_cache; - *value = cache[reg]; - } - break; - default: - BUG(); - } - + *value = snd_soc_get_cache_val(codec->reg_cache, reg, + codec->driver->reg_word_size); return 0; } -- cgit v1.2.3 From 861f2faf4fbd4fb46a436d41dcb5ff75b243ab3a Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Tue, 11 Jan 2011 11:43:51 +0000 Subject: ASoC: soc-core: Simplify compress_type overriding functionality Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6cc68140fa6..fb170a07376 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1748,6 +1748,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) list_for_each_entry(codec, &codec_list, list) { if (codec->cache_init) continue; + /* by default we don't override the compress_type */ + compress_type = 0; /* check to see if we need to override the compress_type */ for (i = 0; i < card->num_configs; ++i) { codec_conf = &card->codec_conf[i]; @@ -1758,18 +1760,6 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) break; } } - if (i == card->num_configs) { - /* no need to override the compress_type so - * go ahead and do the standard thing */ - ret = snd_soc_init_codec_cache(codec, 0); - if (ret < 0) { - mutex_unlock(&card->mutex); - return; - } - continue; - } - /* override the compress_type with the one supplied in - * the machine driver */ ret = snd_soc_init_codec_cache(codec, compress_type); if (ret < 0) { mutex_unlock(&card->mutex); -- cgit v1.2.3 From 9f040c7941df09eca6e6a15cfa766b719d8cafb3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Jan 2011 19:06:08 +0000 Subject: ASoC: Fix indentation in soc_remove() Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fb170a07376..04475c18003 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1896,7 +1896,7 @@ static int soc_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); int i; - if (card->instantiated) { + if (card->instantiated) { /* make sure any delayed work runs */ for (i = 0; i < card->num_rtd; i++) { -- cgit v1.2.3 From d779fce5d79525d66269c8f6e430e1515d697f3d Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Wed, 12 Jan 2011 10:22:28 +0000 Subject: ASoC: soc-cache: Ensure flat compression uses a copy of the defaults cache With the addition of the cache fallback functionality, it is necessary to ensure that if the register defaults cache was marked as __devinitconst and the LZO compression is not compiled in the kernel, the fallback to flat compression will still use a copy of the defaults cache. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 16 ++++------------ 1 file changed, 4 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 9e534429ea6..19b29fb3ca4 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1400,8 +1400,8 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) ret = snd_soc_cache_read(codec, i, &val); if (ret) return ret; - if (codec_drv->reg_cache_default) - if (snd_soc_get_cache_val(codec_drv->reg_cache_default, + if (codec->reg_def_copy) + if (snd_soc_get_cache_val(codec->reg_def_copy, i, codec_drv->reg_word_size) == val) continue; ret = snd_soc_write(codec, i, val); @@ -1446,16 +1446,8 @@ static int snd_soc_flat_cache_init(struct snd_soc_codec *codec) codec_drv = codec->driver; reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; - /* - * for flat compression, we don't need to keep a copy of the - * original defaults register cache as it will definitely not - * be marked as __devinitconst - */ - kfree(codec->reg_def_copy); - codec->reg_def_copy = NULL; - - if (codec_drv->reg_cache_default) - codec->reg_cache = kmemdup(codec_drv->reg_cache_default, + if (codec->reg_def_copy) + codec->reg_cache = kmemdup(codec->reg_def_copy, reg_size, GFP_KERNEL); else codec->reg_cache = kzalloc(reg_size, GFP_KERNEL); -- cgit v1.2.3 From aea170a099793abcd0e6de46b947458073204241 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Wed, 12 Jan 2011 10:38:58 +0000 Subject: ASoC: soc-cache: Add reg_size as a member to snd_soc_codec Simplify the use of reg_size, by calculating it once and storing it in the codec structure for later reference. The value of reg_size is reg_cache_size * reg_word_size. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 26 ++++++++------------------ sound/soc/soc-core.c | 1 + 2 files changed, 9 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 19b29fb3ca4..b2e333f5a38 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1100,34 +1100,28 @@ static inline int snd_soc_lzo_get_blkindex(struct snd_soc_codec *codec, unsigned int reg) { const struct snd_soc_codec_driver *codec_drv; - size_t reg_size; codec_drv = codec->driver; - reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; return (reg * codec_drv->reg_word_size) / - DIV_ROUND_UP(reg_size, snd_soc_lzo_block_count()); + DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()); } static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec, unsigned int reg) { const struct snd_soc_codec_driver *codec_drv; - size_t reg_size; codec_drv = codec->driver; - reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; - return reg % (DIV_ROUND_UP(reg_size, snd_soc_lzo_block_count()) / + return reg % (DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()) / codec_drv->reg_word_size); } static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec) { const struct snd_soc_codec_driver *codec_drv; - size_t reg_size; codec_drv = codec->driver; - reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; - return DIV_ROUND_UP(reg_size, snd_soc_lzo_block_count()); + return DIV_ROUND_UP(codec->reg_size, snd_soc_lzo_block_count()); } static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec) @@ -1287,7 +1281,7 @@ static int snd_soc_lzo_cache_exit(struct snd_soc_codec *codec) static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) { struct snd_soc_lzo_ctx **lzo_blocks; - size_t reg_size, bmp_size; + size_t bmp_size; const struct snd_soc_codec_driver *codec_drv; int ret, tofree, i, blksize, blkcount; const char *p, *end; @@ -1295,7 +1289,6 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) ret = 0; codec_drv = codec->driver; - reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; /* * If we have not been given a default register cache @@ -1307,8 +1300,7 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) tofree = 1; if (!codec->reg_def_copy) { - codec->reg_def_copy = kzalloc(reg_size, - GFP_KERNEL); + codec->reg_def_copy = kzalloc(codec->reg_size, GFP_KERNEL); if (!codec->reg_def_copy) return -ENOMEM; } @@ -1356,7 +1348,7 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) blksize = snd_soc_lzo_get_blksize(codec); p = codec->reg_def_copy; - end = codec->reg_def_copy + reg_size; + end = codec->reg_def_copy + codec->reg_size; /* compress the register map and fill the lzo blocks */ for (i = 0; i < blkcount; ++i, p += blksize) { lzo_blocks[i]->src = p; @@ -1441,16 +1433,14 @@ static int snd_soc_flat_cache_exit(struct snd_soc_codec *codec) static int snd_soc_flat_cache_init(struct snd_soc_codec *codec) { const struct snd_soc_codec_driver *codec_drv; - size_t reg_size; codec_drv = codec->driver; - reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; if (codec->reg_def_copy) codec->reg_cache = kmemdup(codec->reg_def_copy, - reg_size, GFP_KERNEL); + codec->reg_size, GFP_KERNEL); else - codec->reg_cache = kzalloc(reg_size, GFP_KERNEL); + codec->reg_cache = kzalloc(codec->reg_size, GFP_KERNEL); if (!codec->reg_cache) return -ENOMEM; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 04475c18003..cbac50b69c3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3500,6 +3500,7 @@ int snd_soc_register_codec(struct device *dev, /* allocate CODEC register cache */ if (codec_drv->reg_cache_size && codec_drv->reg_word_size) { reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; + codec->reg_size = reg_size; /* it is necessary to make a copy of the default register cache * because in the case of using a compression type that requires * the default register cache to be marked as __devinitconst the -- cgit v1.2.3 From a193436c5d52e8398af22462beaa03407fa89b2e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 12 Jan 2011 15:32:07 +0900 Subject: ASoC: sh: fsi-ak4642: fixup platform device id Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/fsi-ak4642.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index a14820ac966..56cd3422310 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -18,6 +18,7 @@ struct fsi_ak4642_data { const char *cpu_dai; const char *codec; const char *platform; + int id; }; static int fsi_ak4642_dai_init(struct snd_soc_pcm_runtime *rtd) @@ -60,7 +61,7 @@ static int fsi_ak4642_probe(struct platform_device *pdev) pdata = (struct fsi_ak4642_data *)id_entry->driver_data; - fsi_snd_device = platform_device_alloc("soc-audio", FSI_PORT_A); + fsi_snd_device = platform_device_alloc("soc-audio", pdata->id); if (!fsi_snd_device) goto out; @@ -93,6 +94,7 @@ static struct fsi_ak4642_data fsi_a_ak4642 = { .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0012", .platform = "sh_fsi.0", + .id = FSI_PORT_A, }; static struct fsi_ak4642_data fsi_b_ak4642 = { @@ -101,6 +103,7 @@ static struct fsi_ak4642_data fsi_b_ak4642 = { .cpu_dai = "fsib-dai", .codec = "ak4642-codec.0-0012", .platform = "sh_fsi.0", + .id = FSI_PORT_B, }; static struct fsi_ak4642_data fsi_a_ak4643 = { @@ -109,6 +112,7 @@ static struct fsi_ak4642_data fsi_a_ak4643 = { .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0013", .platform = "sh_fsi.0", + .id = FSI_PORT_A, }; static struct fsi_ak4642_data fsi_b_ak4643 = { @@ -117,6 +121,7 @@ static struct fsi_ak4642_data fsi_b_ak4643 = { .cpu_dai = "fsib-dai", .codec = "ak4642-codec.0-0013", .platform = "sh_fsi.0", + .id = FSI_PORT_B, }; static struct fsi_ak4642_data fsi2_a_ak4642 = { @@ -125,6 +130,7 @@ static struct fsi_ak4642_data fsi2_a_ak4642 = { .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0012", .platform = "sh_fsi2", + .id = FSI_PORT_A, }; static struct fsi_ak4642_data fsi2_b_ak4642 = { @@ -133,6 +139,7 @@ static struct fsi_ak4642_data fsi2_b_ak4642 = { .cpu_dai = "fsib-dai", .codec = "ak4642-codec.0-0012", .platform = "sh_fsi2", + .id = FSI_PORT_B, }; static struct fsi_ak4642_data fsi2_a_ak4643 = { @@ -141,6 +148,7 @@ static struct fsi_ak4642_data fsi2_a_ak4643 = { .cpu_dai = "fsia-dai", .codec = "ak4642-codec.0-0013", .platform = "sh_fsi2", + .id = FSI_PORT_A, }; static struct fsi_ak4642_data fsi2_b_ak4643 = { @@ -149,6 +157,7 @@ static struct fsi_ak4642_data fsi2_b_ak4643 = { .cpu_dai = "fsib-dai", .codec = "ak4642-codec.0-0013", .platform = "sh_fsi2", + .id = FSI_PORT_B, }; static struct platform_device_id fsi_id_table[] = { -- cgit v1.2.3 From 066d16c3e8194677a9aaeb06a45e4014387d16f1 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Thu, 13 Jan 2011 12:20:36 +0000 Subject: ASoC: soc-cache: Add support for default readable()/volatile() functions For common scenarios, device drivers can provide a table of all the registers that are at least either readable/writable/volatile. The idea is that if a register lookup fails, all of its read/write/vol members will be zero and will be treated as default. This also reduces the size of the register access array. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 49 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 49 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 1a36b36c5ba..d97a59f6a24 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1603,3 +1603,52 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) return -EINVAL; } EXPORT_SYMBOL_GPL(snd_soc_cache_sync); + +static int snd_soc_get_reg_access_index(struct snd_soc_codec *codec, + unsigned int reg) +{ + const struct snd_soc_codec_driver *codec_drv; + unsigned int min, max, index; + + codec_drv = codec->driver; + min = 0; + max = codec_drv->reg_access_size - 1; + do { + index = (min + max) / 2; + if (codec_drv->reg_access_default[index].reg == reg) + return index; + if (codec_drv->reg_access_default[index].reg < reg) + min = index + 1; + else + max = index; + } while (min <= max); + return -1; +} + +int snd_soc_default_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + int index; + + if (reg >= codec->driver->reg_cache_size) + return 1; + index = snd_soc_get_reg_access_index(codec, reg); + if (index < 0) + return 0; + return codec->driver->reg_access_default[index].vol; +} +EXPORT_SYMBOL_GPL(snd_soc_default_volatile_register); + +int snd_soc_default_readable_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + int index; + + if (reg >= codec->driver->reg_cache_size) + return 1; + index = snd_soc_get_reg_access_index(codec, reg); + if (index < 0) + return 0; + return codec->driver->reg_access_default[index].read; +} +EXPORT_SYMBOL_GPL(snd_soc_default_readable_register); -- cgit v1.2.3 From d4754ec91c7b094298f0b2ba02327e6887671edc Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Thu, 13 Jan 2011 12:20:37 +0000 Subject: ASoC: Update users of readable_register()/volatile_register() Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 4 ++-- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 4 ++-- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 10 +++++----- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 4 ++-- sound/soc/soc-core.c | 4 ++-- 15 files changed, 23 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 8b51245f231..c0fccadaea9 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -193,12 +193,12 @@ static struct cs4270_mode_ratios cs4270_mode_ratios[] = { /* The number of MCLK/LRCK ratios supported by the CS4270 */ #define NUM_MCLK_RATIOS ARRAY_SIZE(cs4270_mode_ratios) -static int cs4270_reg_is_readable(unsigned int reg) +static int cs4270_reg_is_readable(struct snd_soc_codec *codec, unsigned int reg) { return (reg >= CS4270_FIRSTREG) && (reg <= CS4270_LASTREG); } -static int cs4270_reg_is_volatile(unsigned int reg) +static int cs4270_reg_is_volatile(struct snd_soc_codec *codec, unsigned int reg) { /* Unreadable registers are considered volatile */ if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG)) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 37133c40e76..b6ecc7e8967 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -608,7 +608,7 @@ static struct { { 0xFF, 0x00, 1 }, /* FF */ }; -static int max98088_volatile_register(unsigned int reg) +static int max98088_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { return max98088_access[reg].vol; } diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 5eb2f501ce3..83e86f077ee 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -58,7 +58,7 @@ static const u16 wm8523_reg[WM8523_REGISTER_COUNT] = { 0x0000, /* R8 - ZERO_DETECT */ }; -static int wm8523_volatile_register(unsigned int reg) +static int wm8523_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8523_DEVICE_ID: diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 6dae1b40c9f..6785688f880 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -175,7 +175,7 @@ static int txsrc_put(struct snd_kcontrol *kcontrol, return 0; } -static int wm8804_volatile(unsigned int reg) +static int wm8804_volatile(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8804_RST_DEVID1: diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index cd0959926d1..449ea09a193 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -180,7 +180,7 @@ static const u16 wm8900_reg_defaults[WM8900_MAXREG] = { /* Remaining registers all zero */ }; -static int wm8900_volatile_register(unsigned int reg) +static int wm8900_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8900_REG_ID: diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 987476a5895..a2a446cb180 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -232,7 +232,7 @@ struct wm8903_priv { int mic_delay; }; -static int wm8903_volatile_register(unsigned int reg) +static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8903_SW_RESET_AND_ID: diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 9de44a4c05c..17a8fe9b39b 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -596,7 +596,7 @@ static struct { { 0x003F, 0x003F, 0 }, /* R248 - FLL NCO Test 1 */ }; -static int wm8904_volatile_register(unsigned int reg) +static int wm8904_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { return wm8904_access[reg].vol; } diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 55252e7d02c..cdee8103d09 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -291,7 +291,7 @@ struct wm8961_priv { int sysclk; }; -static int wm8961_volatile_register(unsigned int reg) +static int wm8961_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8961_SOFTWARE_RESET: diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b9cb1fcf8c9..7c02924bedd 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1938,7 +1938,7 @@ static const struct wm8962_reg_access { [21139] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21139 - VSS_XTS32_0 */ }; -static int wm8962_volatile_register(unsigned int reg) +static int wm8962_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { if (wm8962_reg_access[reg].vol) return 1; @@ -1946,7 +1946,7 @@ static int wm8962_volatile_register(unsigned int reg) return 0; } -static int wm8962_readable_register(unsigned int reg) +static int wm8962_readable_register(struct snd_soc_codec *codec, unsigned int reg) { if (wm8962_reg_access[reg].read) return 1; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 18c0d9ce7c3..379fa22c5b6 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -242,7 +242,7 @@ struct wm8993_priv { int fll_src; }; -static int wm8993_volatile(unsigned int reg) +static int wm8993_volatile(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8993_SOFTWARE_RESET: diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 247a6a99feb..0bb0bb40b84 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -109,7 +109,7 @@ struct wm8994_priv { struct wm8994_pdata *pdata; }; -static int wm8994_readable(unsigned int reg) +static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM8994_GPIO_1: @@ -136,7 +136,7 @@ static int wm8994_readable(unsigned int reg) return wm8994_access_masks[reg].readable != 0; } -static int wm8994_volatile(unsigned int reg) +static int wm8994_volatile(struct snd_soc_codec *codec, unsigned int reg) { if (reg >= WM8994_CACHE_SIZE) return 1; @@ -164,7 +164,7 @@ static int wm8994_write(struct snd_soc_codec *codec, unsigned int reg, BUG_ON(reg > WM8994_MAX_REGISTER); - if (!wm8994_volatile(reg)) { + if (!wm8994_volatile(codec, reg)) { ret = snd_soc_cache_write(codec, reg, value); if (ret != 0) dev_err(codec->dev, "Cache write to %x failed: %d\n", @@ -182,7 +182,7 @@ static unsigned int wm8994_read(struct snd_soc_codec *codec, BUG_ON(reg > WM8994_MAX_REGISTER); - if (!wm8994_volatile(reg) && wm8994_readable(reg) && + if (!wm8994_volatile(codec, reg) && wm8994_readable(codec, reg) && reg < codec->driver->reg_cache_size) { ret = snd_soc_cache_read(codec, reg, &val); if (ret >= 0) @@ -2943,7 +2943,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) /* Read our current status back from the chip - we don't want to * reset as this may interfere with the GPIO or LDO operation. */ for (i = 0; i < WM8994_CACHE_SIZE; i++) { - if (!wm8994_readable(i) || wm8994_volatile(i)) + if (!wm8994_readable(codec, i) || wm8994_volatile(codec, i)) continue; ret = wm8994_reg_read(codec->control_data, i); diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index ac210ccebd4..f0f678de489 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -909,7 +909,7 @@ static const struct snd_soc_dapm_route wm8995_intercon[] = { { "SPK2R", NULL, "SPK2R Driver" } }; -static int wm8995_volatile(unsigned int reg) +static int wm8995_volatile(struct snd_soc_codec *codec, unsigned int reg) { /* out of bounds registers are generally considered * volatile to support register banks that are partially diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 43825b2102a..5c224dd917d 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -169,7 +169,7 @@ struct wm9081_priv { struct wm9081_retune_mobile_config *retune; }; -static int wm9081_volatile_register(unsigned int reg) +static int wm9081_volatile_register(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM9081_SOFTWARE_RESET: diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index a788c429704..d40bfc9f880 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -144,7 +144,7 @@ struct wm9090_priv { void *control_data; }; -static int wm9090_volatile(unsigned int reg) +static int wm9090_volatile(struct snd_soc_codec *codec, unsigned int reg) { switch (reg) { case WM9090_SOFTWARE_RESET: @@ -518,7 +518,7 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec, for (i = 1; i < codec->driver->reg_cache_size; i++) { if (reg_cache[i] == wm9090_reg_defaults[i]) continue; - if (wm9090_volatile(i)) + if (wm9090_volatile(codec, i)) continue; ret = snd_soc_write(codec, i, reg_cache[i]); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cbac50b69c3..b5e5758456b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -84,7 +84,7 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) count += sprintf(buf, "%s registers\n", codec->name); for (i = 0; i < codec->driver->reg_cache_size; i += step) { - if (codec->driver->readable_register && !codec->driver->readable_register(i)) + if (codec->driver->readable_register && !codec->driver->readable_register(codec, i)) continue; count += sprintf(buf + count, "%2x: ", i); @@ -2030,7 +2030,7 @@ static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg) { if (codec->driver->volatile_register) - return codec->driver->volatile_register(reg); + return codec->driver->volatile_register(codec, reg); else return 0; } -- cgit v1.2.3 From 1500b7b5ffaacb8199e0a61162f5d349fb19acbe Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Thu, 13 Jan 2011 12:20:38 +0000 Subject: ASoC: Automatically assign the default readable()/volatile() functions Ensure that all calls to readable_register()/volatile_register() go via the snd_soc_codec function pointers. If the default register access table has been given but no functions for handling readable()/volatile() registers, use the default ones provided by soc-cache. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b5e5758456b..30d76e8bc9d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -84,7 +84,7 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) count += sprintf(buf, "%s registers\n", codec->name); for (i = 0; i < codec->driver->reg_cache_size; i += step) { - if (codec->driver->readable_register && !codec->driver->readable_register(codec, i)) + if (codec->readable_register && !codec->readable_register(codec, i)) continue; count += sprintf(buf + count, "%2x: ", i); @@ -2029,8 +2029,8 @@ static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) */ int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg) { - if (codec->driver->volatile_register) - return codec->driver->volatile_register(codec, reg); + if (codec->volatile_register) + return codec->volatile_register(codec, reg); else return 0; } @@ -3489,6 +3489,8 @@ int snd_soc_register_codec(struct device *dev, codec->write = codec_drv->write; codec->read = codec_drv->read; + codec->volatile_register = codec_drv->volatile_register; + codec->readable_register = codec_drv->readable_register; codec->dapm.bias_level = SND_SOC_BIAS_OFF; codec->dapm.dev = dev; codec->dapm.codec = codec; @@ -3517,6 +3519,13 @@ int snd_soc_register_codec(struct device *dev, } } + if (codec_drv->reg_access_size && codec_drv->reg_access_default) { + if (!codec->volatile_register) + codec->volatile_register = snd_soc_default_volatile_register; + if (!codec->readable_register) + codec->readable_register = snd_soc_default_readable_register; + } + for (i = 0; i < num_dai; i++) { fixup_codec_formats(&dai_drv[i].playback); fixup_codec_formats(&dai_drv[i].capture); -- cgit v1.2.3 From 422650e65a41a61b2f92396dfa4faa6a4df89913 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 11 Jan 2011 12:48:53 -0700 Subject: ASoC: tegra: s/IS_ERR_OR_NULL/IS_ERR/ for clk_get_sys A recent discussion on linux-arm-kernel noted that the value returned by clk_get_sys is an opaque token, and not strictly a pointer; it is meaningful only to the clock API, clients should not dereference the value, and the clock API must accept any non-IS_ERR value it returned. Hence, only IS_ERR is appropriate to interpret the result, not IS_ERR_OR_NULL. I checked that clk_get_sys in both ASoC's for-next and Tegra's for-next do behave as described; NULL is not returned in the case of error. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_asoc_utils.c | 20 +++++++++----------- sound/soc/tegra/tegra_i2s.c | 2 +- 2 files changed, 10 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index 711ab7ff5ce..cfe2ea890dc 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -113,35 +113,33 @@ int tegra_asoc_utils_init(void) int ret; clk_pll_a = clk_get_sys(NULL, "pll_a"); - if (IS_ERR_OR_NULL(clk_pll_a)) { + if (IS_ERR(clk_pll_a)) { pr_err(PREFIX "Can't retrieve clk pll_a\n"); ret = PTR_ERR(clk_pll_a); goto err; } clk_pll_a_out0 = clk_get_sys(NULL, "pll_a_out0"); - if (IS_ERR_OR_NULL(clk_pll_a_out0)) { + if (IS_ERR(clk_pll_a_out0)) { pr_err(PREFIX "Can't retrieve clk pll_a_out0\n"); ret = PTR_ERR(clk_pll_a_out0); - goto err; + goto err_put_pll_a; } clk_cdev1 = clk_get_sys(NULL, "cdev1"); - if (IS_ERR_OR_NULL(clk_cdev1)) { + if (IS_ERR(clk_cdev1)) { pr_err(PREFIX "Can't retrieve clk cdev1\n"); ret = PTR_ERR(clk_cdev1); - goto err; + goto err_put_pll_a_out0; } return 0; +err_put_pll_a_out0: + clk_put(clk_pll_a_out0); +err_put_pll_a: + clk_put(clk_pll_a); err: - if (!IS_ERR_OR_NULL(clk_cdev1)) - clk_put(clk_cdev1); - if (!IS_ERR_OR_NULL(clk_pll_a_out0)) - clk_put(clk_pll_a_out0); - if (!IS_ERR_OR_NULL(clk_pll_a)) - clk_put(clk_pll_a); return ret; } diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 1730509c8ac..6d668785e9a 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -385,7 +385,7 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) snprintf(clk_name, sizeof(clk_name), DRV_NAME ".%d", pdev->id); i2s->clk_i2s = clk_get_sys(clk_name, NULL); - if (IS_ERR_OR_NULL(i2s->clk_i2s)) { + if (IS_ERR(i2s->clk_i2s)) { pr_err("Can't retrieve i2s clock\n"); ret = PTR_ERR(i2s->clk_i2s); goto err_free; -- cgit v1.2.3 From b0e264855cc42106cc41426e68651b5d67df444d Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 13 Jan 2011 22:48:02 +0530 Subject: ASoC: soc core move card cleanup from soc_remove() In soc_remove() the card resources are cleaned up. This can also be done in card_unregister() This patch move this cleanup into a new function and calls it from card_unregister. This paves way for further work to allow card registartion from machine. Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 51 +++++++++++++++++++++++++++++---------------------- 1 file changed, 29 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 30d76e8bc9d..318c3a72021 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1890,37 +1890,42 @@ static int soc_probe(struct platform_device *pdev) return 0; } -/* removes a socdev */ -static int soc_remove(struct platform_device *pdev) +static int soc_cleanup_card_resources(struct snd_soc_card *card) { - struct snd_soc_card *card = platform_get_drvdata(pdev); + struct platform_device *pdev = to_platform_device(card->dev); int i; - if (card->instantiated) { + /* make sure any delayed work runs */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + flush_delayed_work_sync(&rtd->delayed_work); + } - /* make sure any delayed work runs */ - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; - flush_delayed_work_sync(&rtd->delayed_work); - } + /* remove auxiliary devices */ + for (i = 0; i < card->num_aux_devs; i++) + soc_remove_aux_dev(card, i); - /* remove auxiliary devices */ - for (i = 0; i < card->num_aux_devs; i++) - soc_remove_aux_dev(card, i); + /* remove and free each DAI */ + for (i = 0; i < card->num_rtd; i++) + soc_remove_dai_link(card, i); - /* remove and free each DAI */ - for (i = 0; i < card->num_rtd; i++) - soc_remove_dai_link(card, i); + soc_cleanup_card_debugfs(card); - soc_cleanup_card_debugfs(card); + /* remove the card */ + if (card->remove) + card->remove(pdev); - /* remove the card */ - if (card->remove) - card->remove(pdev); + kfree(card->rtd); + snd_card_free(card->snd_card); + return 0; + +} + +/* removes a socdev */ +static int soc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); - kfree(card->rtd); - snd_card_free(card->snd_card); - } snd_soc_unregister_card(card); return 0; } @@ -3153,6 +3158,8 @@ static int snd_soc_register_card(struct snd_soc_card *card) */ static int snd_soc_unregister_card(struct snd_soc_card *card) { + if (card->instantiated) + soc_cleanup_card_resources(card); mutex_lock(&client_mutex); list_del(&card->list); mutex_unlock(&client_mutex); -- cgit v1.2.3 From 150dd2f8c42bdb3c51533459e3ff5075d8720260 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 13 Jan 2011 22:48:32 +0530 Subject: ASoC: soc core move the card debugfs initialization The card debugfs initialization is done in soc_probe but would be better if it is done when the card in registered This patch moves the debugfs initialization to register_card() Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 318c3a72021..9c5e7cff3f0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1879,8 +1879,6 @@ static int soc_probe(struct platform_device *pdev) INIT_LIST_HEAD(&card->paths); INIT_LIST_HEAD(&card->dapm_list); - soc_init_card_debugfs(card); - ret = snd_soc_register_card(card); if (ret != 0) { dev_err(&pdev->dev, "Failed to register card\n"); @@ -3123,6 +3121,8 @@ static int snd_soc_register_card(struct snd_soc_card *card) if (!card->name || !card->dev) return -EINVAL; + soc_init_card_debugfs(card); + card->rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime) * (card->num_links + card->num_aux_devs), GFP_KERNEL); -- cgit v1.2.3 From 4e10bda05d6c7d4aba509bbbab5ba748d54c702f Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 13 Jan 2011 22:48:52 +0530 Subject: ASoC: soc core add inline to handle card list initialzation Currently the soc_probe initializes the card hence it does the card list initialzation. But if machines directly register the card they would need to do these steps, so putting them as inline would save lot of code This patch adds an inline to do list initialzation Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9c5e7cff3f0..83057127b2f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1872,12 +1872,7 @@ static int soc_probe(struct platform_device *pdev) /* Bodge while we unpick instantiation */ card->dev = &pdev->dev; - INIT_LIST_HEAD(&card->dai_dev_list); - INIT_LIST_HEAD(&card->codec_dev_list); - INIT_LIST_HEAD(&card->platform_dev_list); - INIT_LIST_HEAD(&card->widgets); - INIT_LIST_HEAD(&card->paths); - INIT_LIST_HEAD(&card->dapm_list); + snd_soc_initialize_card_lists(card); ret = snd_soc_register_card(card); if (ret != 0) { -- cgit v1.2.3 From b0ee5fbab742d234f87c0186a5cae2aba85ba7a7 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 13 Jan 2011 15:21:53 -0700 Subject: ASoC: tegra: Remove TEGRA_I2S_AUDIO from Kconfig That config variable doesn't exist in the mainline kernel, and hence the dependency shouldn't either. In linux-tegra-2.6.36, the dependency did exist to avoid a conflict with the old non-ALSA Tegra I2S driver. However, this isn't and won't be upstreamed. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/Kconfig | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 5514f1c7ca2..66b504f06c2 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -8,7 +8,6 @@ config SND_TEGRA_SOC config SND_TEGRA_SOC_I2S tristate depends on SND_TEGRA_SOC - depends on !TEGRA_I2S_AUDIO default m help Say Y or M if you want to add support for codecs attached to the -- cgit v1.2.3 From 62ffac4d70ed996d303da0ac8e32c89b9810115c Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 13 Jan 2011 15:22:08 -0700 Subject: ASoC: tegra: Add DAPM widgets/routes for Harmony With this change, I can capture from a microphone plugged into the mic jack on Harmony (after unmuting Left Input PGA, and maybe turning up the gain there too). Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/harmony.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) (limited to 'sound') diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index bf0dbafbf6d..b160b7113f4 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -107,6 +107,36 @@ static struct snd_soc_ops harmony_asoc_ops = { .hw_params = harmony_asoc_hw_params, }; +static const struct snd_soc_dapm_widget harmony_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route harmony_audio_map[] = { + {"Headphone Jack", NULL, "HPOUTR"}, + {"Headphone Jack", NULL, "HPOUTL"}, + {"Mic Bias", NULL, "Mic Jack"}, + {"IN1L", NULL, "Mic Bias"}, +}; + +static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_new_controls(dapm, harmony_dapm_widgets, + ARRAY_SIZE(harmony_dapm_widgets)); + + snd_soc_dapm_add_routes(dapm, harmony_audio_map, + ARRAY_SIZE(harmony_audio_map)); + + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); + snd_soc_dapm_sync(dapm); + + return 0; +} + static struct snd_soc_dai_link harmony_wm8903_dai = { .name = "WM8903", .stream_name = "WM8903 PCM", @@ -114,6 +144,7 @@ static struct snd_soc_dai_link harmony_wm8903_dai = { .platform_name = "tegra-pcm-audio", .cpu_dai_name = "tegra-i2s.0", .codec_dai_name = "wm8903-hifi", + .init = harmony_asoc_init, .ops = &harmony_asoc_ops, }; -- cgit v1.2.3 From 70a7ca34dbdcc6f0ed332baf2b308bab2871424a Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Fri, 14 Jan 2011 19:22:48 +0530 Subject: ASoC: soc core allow machine driver to register the card The machine driver can't register the card directly and need to do this thru soc-audio device creation This patch allows the register and unregister card to be directly called by machine drivers Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 83057127b2f..69117b686fd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -58,8 +58,6 @@ static LIST_HEAD(dai_list); static LIST_HEAD(platform_list); static LIST_HEAD(codec_list); -static int snd_soc_register_card(struct snd_soc_card *card); -static int snd_soc_unregister_card(struct snd_soc_card *card); static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); /* @@ -1870,6 +1868,13 @@ static int soc_probe(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); int ret = 0; + /* + * no card, so machine driver should be registering card + * we should not be here in that case so ret error + */ + if (!card) + return -EINVAL; + /* Bodge while we unpick instantiation */ card->dev = &pdev->dev; snd_soc_initialize_card_lists(card); @@ -3105,11 +3110,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); * * @card: Card to register * - * Note that currently this is an internal only function: it will be - * exposed to machine drivers after further backporting of ASoC v2 - * registration APIs. */ -static int snd_soc_register_card(struct snd_soc_card *card) +int snd_soc_register_card(struct snd_soc_card *card) { int i; @@ -3141,17 +3143,15 @@ static int snd_soc_register_card(struct snd_soc_card *card) return 0; } +EXPORT_SYMBOL_GPL(snd_soc_register_card); /** * snd_soc_unregister_card - Unregister a card with the ASoC core * * @card: Card to unregister * - * Note that currently this is an internal only function: it will be - * exposed to machine drivers after further backporting of ASoC v2 - * registration APIs. */ -static int snd_soc_unregister_card(struct snd_soc_card *card) +int snd_soc_unregister_card(struct snd_soc_card *card) { if (card->instantiated) soc_cleanup_card_resources(card); @@ -3162,6 +3162,7 @@ static int snd_soc_unregister_card(struct snd_soc_card *card) return 0; } +EXPORT_SYMBOL_GPL(snd_soc_unregister_card); /* * Simplify DAI link configuration by removing ".-1" from device names -- cgit v1.2.3 From 203db220718c735dcb959fddc64e94fff3b52f73 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Fri, 14 Jan 2011 15:49:40 +0000 Subject: ASoC: WM8991: Add initial WM8991 driver The WM8991 is a highly integrated ultra-low power hi-fi CODEC designed for handsets rich in multimedia features such as GPS, mobile TV, digital audio playback and gaming. This driver was originally written by Graeme Gregory and has been maintained out of tree by Mark Brown and Dimitris Papastamos. Signed-off-by: Graeme Gregory Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8991.c | 1427 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8991.h | 833 ++++++++++++++++++++++++++ 4 files changed, 2266 insertions(+) create mode 100644 sound/soc/codecs/wm8991.c create mode 100644 sound/soc/codecs/wm8991.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 61e36efbf27..a18cff4afbc 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -77,6 +77,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8985 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C + select SND_SOC_WM8991 if I2C select SND_SOC_WM8993 if I2C select SND_SOC_WM8994 if MFD_WM8994 select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI @@ -308,6 +309,9 @@ config SND_SOC_WM8988 config SND_SOC_WM8990 tristate +config SND_SOC_WM8991 + tristate + config SND_SOC_WM8993 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 333910a9f8f..68e76af894b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -62,6 +62,7 @@ snd-soc-wm8978-objs := wm8978.o snd-soc-wm8985-objs := wm8985.o snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o +snd-soc-wm8991-objs := wm8991.o snd-soc-wm8993-objs := wm8993.o snd-soc-wm8994-objs := wm8994.o wm8994-tables.o snd-soc-wm8995-objs := wm8995.o @@ -143,6 +144,7 @@ obj-$(CONFIG_SND_SOC_WM8978) += snd-soc-wm8978.o obj-$(CONFIG_SND_SOC_WM8985) += snd-soc-wm8985.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o +obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o obj-$(CONFIG_SND_SOC_WM8994) += snd-soc-wm8994.o obj-$(CONFIG_SND_SOC_WM8995) += snd-soc-wm8995.o diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c new file mode 100644 index 00000000000..28fdfd66661 --- /dev/null +++ b/sound/soc/codecs/wm8991.c @@ -0,0 +1,1427 @@ +/* + * wm8991.c -- WM8991 ALSA Soc Audio driver + * + * Copyright 2007-2010 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * linux@wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8991.h" + +struct wm8991_priv { + enum snd_soc_control_type control_type; + unsigned int pcmclk; +}; + +static const u16 wm8991_reg_defs[] = { + 0x8991, /* R0 - Reset */ + 0x0000, /* R1 - Power Management (1) */ + 0x6000, /* R2 - Power Management (2) */ + 0x0000, /* R3 - Power Management (3) */ + 0x4050, /* R4 - Audio Interface (1) */ + 0x4000, /* R5 - Audio Interface (2) */ + 0x01C8, /* R6 - Clocking (1) */ + 0x0000, /* R7 - Clocking (2) */ + 0x0040, /* R8 - Audio Interface (3) */ + 0x0040, /* R9 - Audio Interface (4) */ + 0x0004, /* R10 - DAC CTRL */ + 0x00C0, /* R11 - Left DAC Digital Volume */ + 0x00C0, /* R12 - Right DAC Digital Volume */ + 0x0000, /* R13 - Digital Side Tone */ + 0x0100, /* R14 - ADC CTRL */ + 0x00C0, /* R15 - Left ADC Digital Volume */ + 0x00C0, /* R16 - Right ADC Digital Volume */ + 0x0000, /* R17 */ + 0x0000, /* R18 - GPIO CTRL 1 */ + 0x1000, /* R19 - GPIO1 & GPIO2 */ + 0x1010, /* R20 - GPIO3 & GPIO4 */ + 0x1010, /* R21 - GPIO5 & GPIO6 */ + 0x8000, /* R22 - GPIOCTRL 2 */ + 0x0800, /* R23 - GPIO_POL */ + 0x008B, /* R24 - Left Line Input 1&2 Volume */ + 0x008B, /* R25 - Left Line Input 3&4 Volume */ + 0x008B, /* R26 - Right Line Input 1&2 Volume */ + 0x008B, /* R27 - Right Line Input 3&4 Volume */ + 0x0000, /* R28 - Left Output Volume */ + 0x0000, /* R29 - Right Output Volume */ + 0x0066, /* R30 - Line Outputs Volume */ + 0x0022, /* R31 - Out3/4 Volume */ + 0x0079, /* R32 - Left OPGA Volume */ + 0x0079, /* R33 - Right OPGA Volume */ + 0x0003, /* R34 - Speaker Volume */ + 0x0003, /* R35 - ClassD1 */ + 0x0000, /* R36 */ + 0x0100, /* R37 - ClassD3 */ + 0x0000, /* R38 */ + 0x0000, /* R39 - Input Mixer1 */ + 0x0000, /* R40 - Input Mixer2 */ + 0x0000, /* R41 - Input Mixer3 */ + 0x0000, /* R42 - Input Mixer4 */ + 0x0000, /* R43 - Input Mixer5 */ + 0x0000, /* R44 - Input Mixer6 */ + 0x0000, /* R45 - Output Mixer1 */ + 0x0000, /* R46 - Output Mixer2 */ + 0x0000, /* R47 - Output Mixer3 */ + 0x0000, /* R48 - Output Mixer4 */ + 0x0000, /* R49 - Output Mixer5 */ + 0x0000, /* R50 - Output Mixer6 */ + 0x0180, /* R51 - Out3/4 Mixer */ + 0x0000, /* R52 - Line Mixer1 */ + 0x0000, /* R53 - Line Mixer2 */ + 0x0000, /* R54 - Speaker Mixer */ + 0x0000, /* R55 - Additional Control */ + 0x0000, /* R56 - AntiPOP1 */ + 0x0000, /* R57 - AntiPOP2 */ + 0x0000, /* R58 - MICBIAS */ + 0x0000, /* R59 */ + 0x0008, /* R60 - PLL1 */ + 0x0031, /* R61 - PLL2 */ + 0x0026, /* R62 - PLL3 */ +}; + +#define wm8991_reset(c) snd_soc_write(c, WM8991_RESET, 0) + +static const unsigned int rec_mix_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 7, TLV_DB_LINEAR_ITEM(-1500, 600), +}; + +static const unsigned int in_pga_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 0x1F, TLV_DB_LINEAR_ITEM(-1650, 3000), +}; + +static const unsigned int out_mix_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 7, TLV_DB_LINEAR_ITEM(0, -2100), +}; + +static const unsigned int out_pga_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 127, TLV_DB_LINEAR_ITEM(-7300, 600), +}; + +static const unsigned int out_omix_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 7, TLV_DB_LINEAR_ITEM(-600, 0), +}; + +static const unsigned int out_dac_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 255, TLV_DB_LINEAR_ITEM(-7163, 0), +}; + +static const unsigned int in_adc_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 255, TLV_DB_LINEAR_ITEM(-7163, 1763), +}; + +static const unsigned int out_sidetone_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 31, TLV_DB_LINEAR_ITEM(-3600, 0), +}; + +static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int ret; + u16 val; + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (ret < 0) + return ret; + + /* now hit the volume update bits (always bit 8) */ + val = snd_soc_read(codec, reg); + return snd_soc_write(codec, reg, val | 0x0100); +} + +static const char *wm8991_digital_sidetone[] = +{"None", "Left ADC", "Right ADC", "Reserved"}; + +static const struct soc_enum wm8991_left_digital_sidetone_enum = + SOC_ENUM_SINGLE(WM8991_DIGITAL_SIDE_TONE, + WM8991_ADC_TO_DACL_SHIFT, + WM8991_ADC_TO_DACL_MASK, + wm8991_digital_sidetone); + +static const struct soc_enum wm8991_right_digital_sidetone_enum = + SOC_ENUM_SINGLE(WM8991_DIGITAL_SIDE_TONE, + WM8991_ADC_TO_DACR_SHIFT, + WM8991_ADC_TO_DACR_MASK, + wm8991_digital_sidetone); + +static const char *wm8991_adcmode[] = +{"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"}; + +static const struct soc_enum wm8991_right_adcmode_enum = + SOC_ENUM_SINGLE(WM8991_ADC_CTRL, + WM8991_ADC_HPF_CUT_SHIFT, + WM8991_ADC_HPF_CUT_MASK, + wm8991_adcmode); + +static const struct snd_kcontrol_new wm8991_snd_controls[] = { + /* INMIXL */ + SOC_SINGLE("LIN12 PGA Boost", WM8991_INPUT_MIXER3, WM8991_L12MNBST_BIT, 1, 0), + SOC_SINGLE("LIN34 PGA Boost", WM8991_INPUT_MIXER3, WM8991_L34MNBST_BIT, 1, 0), + /* INMIXR */ + SOC_SINGLE("RIN12 PGA Boost", WM8991_INPUT_MIXER3, WM8991_R12MNBST_BIT, 1, 0), + SOC_SINGLE("RIN34 PGA Boost", WM8991_INPUT_MIXER3, WM8991_R34MNBST_BIT, 1, 0), + + /* LOMIX */ + SOC_SINGLE_TLV("LOMIX LIN3 Bypass Volume", WM8991_OUTPUT_MIXER3, + WM8991_LLI3LOVOL_SHIFT, WM8991_LLI3LOVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("LOMIX RIN12 PGA Bypass Volume", WM8991_OUTPUT_MIXER3, + WM8991_LR12LOVOL_SHIFT, WM8991_LR12LOVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("LOMIX LIN12 PGA Bypass Volume", WM8991_OUTPUT_MIXER3, + WM8991_LL12LOVOL_SHIFT, WM8991_LL12LOVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("LOMIX RIN3 Bypass Volume", WM8991_OUTPUT_MIXER5, + WM8991_LRI3LOVOL_SHIFT, WM8991_LRI3LOVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("LOMIX AINRMUX Bypass Volume", WM8991_OUTPUT_MIXER5, + WM8991_LRBLOVOL_SHIFT, WM8991_LRBLOVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("LOMIX AINLMUX Bypass Volume", WM8991_OUTPUT_MIXER5, + WM8991_LRBLOVOL_SHIFT, WM8991_LRBLOVOL_MASK, 1, out_mix_tlv), + + /* ROMIX */ + SOC_SINGLE_TLV("ROMIX RIN3 Bypass Volume", WM8991_OUTPUT_MIXER4, + WM8991_RRI3ROVOL_SHIFT, WM8991_RRI3ROVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("ROMIX LIN12 PGA Bypass Volume", WM8991_OUTPUT_MIXER4, + WM8991_RL12ROVOL_SHIFT, WM8991_RL12ROVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("ROMIX RIN12 PGA Bypass Volume", WM8991_OUTPUT_MIXER4, + WM8991_RR12ROVOL_SHIFT, WM8991_RR12ROVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("ROMIX LIN3 Bypass Volume", WM8991_OUTPUT_MIXER6, + WM8991_RLI3ROVOL_SHIFT, WM8991_RLI3ROVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("ROMIX AINLMUX Bypass Volume", WM8991_OUTPUT_MIXER6, + WM8991_RLBROVOL_SHIFT, WM8991_RLBROVOL_MASK, 1, out_mix_tlv), + SOC_SINGLE_TLV("ROMIX AINRMUX Bypass Volume", WM8991_OUTPUT_MIXER6, + WM8991_RRBROVOL_SHIFT, WM8991_RRBROVOL_MASK, 1, out_mix_tlv), + + /* LOUT */ + SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOUT Volume", WM8991_LEFT_OUTPUT_VOLUME, + WM8991_LOUTVOL_SHIFT, WM8991_LOUTVOL_MASK, 0, out_pga_tlv), + SOC_SINGLE("LOUT ZC", WM8991_LEFT_OUTPUT_VOLUME, WM8991_LOZC_BIT, 1, 0), + + /* ROUT */ + SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROUT Volume", WM8991_RIGHT_OUTPUT_VOLUME, + WM8991_ROUTVOL_SHIFT, WM8991_ROUTVOL_MASK, 0, out_pga_tlv), + SOC_SINGLE("ROUT ZC", WM8991_RIGHT_OUTPUT_VOLUME, WM8991_ROZC_BIT, 1, 0), + + /* LOPGA */ + SOC_WM899X_OUTPGA_SINGLE_R_TLV("LOPGA Volume", WM8991_LEFT_OPGA_VOLUME, + WM8991_LOPGAVOL_SHIFT, WM8991_LOPGAVOL_MASK, 0, out_pga_tlv), + SOC_SINGLE("LOPGA ZC Switch", WM8991_LEFT_OPGA_VOLUME, + WM8991_LOPGAZC_BIT, 1, 0), + + /* ROPGA */ + SOC_WM899X_OUTPGA_SINGLE_R_TLV("ROPGA Volume", WM8991_RIGHT_OPGA_VOLUME, + WM8991_ROPGAVOL_SHIFT, WM8991_ROPGAVOL_MASK, 0, out_pga_tlv), + SOC_SINGLE("ROPGA ZC Switch", WM8991_RIGHT_OPGA_VOLUME, + WM8991_ROPGAZC_BIT, 1, 0), + + SOC_SINGLE("LON Mute Switch", WM8991_LINE_OUTPUTS_VOLUME, + WM8991_LONMUTE_BIT, 1, 0), + SOC_SINGLE("LOP Mute Switch", WM8991_LINE_OUTPUTS_VOLUME, + WM8991_LOPMUTE_BIT, 1, 0), + SOC_SINGLE("LOP Attenuation Switch", WM8991_LINE_OUTPUTS_VOLUME, + WM8991_LOATTN_BIT, 1, 0), + SOC_SINGLE("RON Mute Switch", WM8991_LINE_OUTPUTS_VOLUME, + WM8991_RONMUTE_BIT, 1, 0), + SOC_SINGLE("ROP Mute Switch", WM8991_LINE_OUTPUTS_VOLUME, + WM8991_ROPMUTE_BIT, 1, 0), + SOC_SINGLE("ROP Attenuation Switch", WM8991_LINE_OUTPUTS_VOLUME, + WM8991_ROATTN_BIT, 1, 0), + + SOC_SINGLE("OUT3 Mute Switch", WM8991_OUT3_4_VOLUME, + WM8991_OUT3MUTE_BIT, 1, 0), + SOC_SINGLE("OUT3 Attenuation Switch", WM8991_OUT3_4_VOLUME, + WM8991_OUT3ATTN_BIT, 1, 0), + + SOC_SINGLE("OUT4 Mute Switch", WM8991_OUT3_4_VOLUME, + WM8991_OUT4MUTE_BIT, 1, 0), + SOC_SINGLE("OUT4 Attenuation Switch", WM8991_OUT3_4_VOLUME, + WM8991_OUT4ATTN_BIT, 1, 0), + + SOC_SINGLE("Speaker Mode Switch", WM8991_CLASSD1, + WM8991_CDMODE_BIT, 1, 0), + + SOC_SINGLE("Speaker Output Attenuation Volume", WM8991_SPEAKER_VOLUME, + WM8991_SPKVOL_SHIFT, WM8991_SPKVOL_MASK, 0), + SOC_SINGLE("Speaker DC Boost Volume", WM8991_CLASSD3, + WM8991_DCGAIN_SHIFT, WM8991_DCGAIN_MASK, 0), + SOC_SINGLE("Speaker AC Boost Volume", WM8991_CLASSD3, + WM8991_ACGAIN_SHIFT, WM8991_ACGAIN_MASK, 0), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left DAC Digital Volume", + WM8991_LEFT_DAC_DIGITAL_VOLUME, + WM8991_DACL_VOL_SHIFT, + WM8991_DACL_VOL_MASK, + 0, + out_dac_tlv), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right DAC Digital Volume", + WM8991_RIGHT_DAC_DIGITAL_VOLUME, + WM8991_DACR_VOL_SHIFT, + WM8991_DACR_VOL_MASK, + 0, + out_dac_tlv), + + SOC_ENUM("Left Digital Sidetone", wm8991_left_digital_sidetone_enum), + SOC_ENUM("Right Digital Sidetone", wm8991_right_digital_sidetone_enum), + + SOC_SINGLE_TLV("Left Digital Sidetone Volume", WM8991_DIGITAL_SIDE_TONE, + WM8991_ADCL_DAC_SVOL_SHIFT, WM8991_ADCL_DAC_SVOL_MASK, 0, + out_sidetone_tlv), + SOC_SINGLE_TLV("Right Digital Sidetone Volume", WM8991_DIGITAL_SIDE_TONE, + WM8991_ADCR_DAC_SVOL_SHIFT, WM8991_ADCR_DAC_SVOL_MASK, 0, + out_sidetone_tlv), + + SOC_SINGLE("ADC Digital High Pass Filter Switch", WM8991_ADC_CTRL, + WM8991_ADC_HPF_ENA_BIT, 1, 0), + + SOC_ENUM("ADC HPF Mode", wm8991_right_adcmode_enum), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left ADC Digital Volume", + WM8991_LEFT_ADC_DIGITAL_VOLUME, + WM8991_ADCL_VOL_SHIFT, + WM8991_ADCL_VOL_MASK, + 0, + in_adc_tlv), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("Right ADC Digital Volume", + WM8991_RIGHT_ADC_DIGITAL_VOLUME, + WM8991_ADCR_VOL_SHIFT, + WM8991_ADCR_VOL_MASK, + 0, + in_adc_tlv), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN12 Volume", + WM8991_LEFT_LINE_INPUT_1_2_VOLUME, + WM8991_LIN12VOL_SHIFT, + WM8991_LIN12VOL_MASK, + 0, + in_pga_tlv), + + SOC_SINGLE("LIN12 ZC Switch", WM8991_LEFT_LINE_INPUT_1_2_VOLUME, + WM8991_LI12ZC_BIT, 1, 0), + + SOC_SINGLE("LIN12 Mute Switch", WM8991_LEFT_LINE_INPUT_1_2_VOLUME, + WM8991_LI12MUTE_BIT, 1, 0), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("LIN34 Volume", + WM8991_LEFT_LINE_INPUT_3_4_VOLUME, + WM8991_LIN34VOL_SHIFT, + WM8991_LIN34VOL_MASK, + 0, + in_pga_tlv), + + SOC_SINGLE("LIN34 ZC Switch", WM8991_LEFT_LINE_INPUT_3_4_VOLUME, + WM8991_LI34ZC_BIT, 1, 0), + + SOC_SINGLE("LIN34 Mute Switch", WM8991_LEFT_LINE_INPUT_3_4_VOLUME, + WM8991_LI34MUTE_BIT, 1, 0), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN12 Volume", + WM8991_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8991_RIN12VOL_SHIFT, + WM8991_RIN12VOL_MASK, + 0, + in_pga_tlv), + + SOC_SINGLE("RIN12 ZC Switch", WM8991_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8991_RI12ZC_BIT, 1, 0), + + SOC_SINGLE("RIN12 Mute Switch", WM8991_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8991_RI12MUTE_BIT, 1, 0), + + SOC_WM899X_OUTPGA_SINGLE_R_TLV("RIN34 Volume", + WM8991_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8991_RIN34VOL_SHIFT, + WM8991_RIN34VOL_MASK, + 0, + in_pga_tlv), + + SOC_SINGLE("RIN34 ZC Switch", WM8991_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8991_RI34ZC_BIT, 1, 0), + + SOC_SINGLE("RIN34 Mute Switch", WM8991_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8991_RI34MUTE_BIT, 1, 0), +}; + +/* + * _DAPM_ Controls + */ +static int inmixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + u16 reg, fakepower; + + reg = snd_soc_read(w->codec, WM8991_POWER_MANAGEMENT_2); + fakepower = snd_soc_read(w->codec, WM8991_INTDRIVBITS); + + if (fakepower & ((1 << WM8991_INMIXL_PWR_BIT) | + (1 << WM8991_AINLMUX_PWR_BIT))) + reg |= WM8991_AINL_ENA; + else + reg &= ~WM8991_AINL_ENA; + + if (fakepower & ((1 << WM8991_INMIXR_PWR_BIT) | + (1 << WM8991_AINRMUX_PWR_BIT))) + reg |= WM8991_AINR_ENA; + else + reg &= ~WM8991_AINL_ENA; + + snd_soc_write(w->codec, WM8991_POWER_MANAGEMENT_2, reg); + return 0; +} + +static int outmixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + u32 reg_shift = kcontrol->private_value & 0xfff; + int ret = 0; + u16 reg; + + switch (reg_shift) { + case WM8991_SPEAKER_MIXER | (WM8991_LDSPK_BIT << 8): + reg = snd_soc_read(w->codec, WM8991_OUTPUT_MIXER1); + if (reg & WM8991_LDLO) { + printk(KERN_WARNING + "Cannot set as Output Mixer 1 LDLO Set\n"); + ret = -1; + } + break; + + case WM8991_SPEAKER_MIXER | (WM8991_RDSPK_BIT << 8): + reg = snd_soc_read(w->codec, WM8991_OUTPUT_MIXER2); + if (reg & WM8991_RDRO) { + printk(KERN_WARNING + "Cannot set as Output Mixer 2 RDRO Set\n"); + ret = -1; + } + break; + + case WM8991_OUTPUT_MIXER1 | (WM8991_LDLO_BIT << 8): + reg = snd_soc_read(w->codec, WM8991_SPEAKER_MIXER); + if (reg & WM8991_LDSPK) { + printk(KERN_WARNING + "Cannot set as Speaker Mixer LDSPK Set\n"); + ret = -1; + } + break; + + case WM8991_OUTPUT_MIXER2 | (WM8991_RDRO_BIT << 8): + reg = snd_soc_read(w->codec, WM8991_SPEAKER_MIXER); + if (reg & WM8991_RDSPK) { + printk(KERN_WARNING + "Cannot set as Speaker Mixer RDSPK Set\n"); + ret = -1; + } + break; + } + + return ret; +} + +/* INMIX dB values */ +static const unsigned int in_mix_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0, 7, TLV_DB_LINEAR_ITEM(-1200, 600), +}; + +/* Left In PGA Connections */ +static const struct snd_kcontrol_new wm8991_dapm_lin12_pga_controls[] = { + SOC_DAPM_SINGLE("LIN1 Switch", WM8991_INPUT_MIXER2, WM8991_LMN1_BIT, 1, 0), + SOC_DAPM_SINGLE("LIN2 Switch", WM8991_INPUT_MIXER2, WM8991_LMP2_BIT, 1, 0), +}; + +static const struct snd_kcontrol_new wm8991_dapm_lin34_pga_controls[] = { + SOC_DAPM_SINGLE("LIN3 Switch", WM8991_INPUT_MIXER2, WM8991_LMN3_BIT, 1, 0), + SOC_DAPM_SINGLE("LIN4 Switch", WM8991_INPUT_MIXER2, WM8991_LMP4_BIT, 1, 0), +}; + +/* Right In PGA Connections */ +static const struct snd_kcontrol_new wm8991_dapm_rin12_pga_controls[] = { + SOC_DAPM_SINGLE("RIN1 Switch", WM8991_INPUT_MIXER2, WM8991_RMN1_BIT, 1, 0), + SOC_DAPM_SINGLE("RIN2 Switch", WM8991_INPUT_MIXER2, WM8991_RMP2_BIT, 1, 0), +}; + +static const struct snd_kcontrol_new wm8991_dapm_rin34_pga_controls[] = { + SOC_DAPM_SINGLE("RIN3 Switch", WM8991_INPUT_MIXER2, WM8991_RMN3_BIT, 1, 0), + SOC_DAPM_SINGLE("RIN4 Switch", WM8991_INPUT_MIXER2, WM8991_RMP4_BIT, 1, 0), +}; + +/* INMIXL */ +static const struct snd_kcontrol_new wm8991_dapm_inmixl_controls[] = { + SOC_DAPM_SINGLE_TLV("Record Left Volume", WM8991_INPUT_MIXER3, + WM8991_LDBVOL_SHIFT, WM8991_LDBVOL_MASK, 0, in_mix_tlv), + SOC_DAPM_SINGLE_TLV("LIN2 Volume", WM8991_INPUT_MIXER5, WM8991_LI2BVOL_SHIFT, + 7, 0, in_mix_tlv), + SOC_DAPM_SINGLE("LINPGA12 Switch", WM8991_INPUT_MIXER3, WM8991_L12MNB_BIT, + 1, 0), + SOC_DAPM_SINGLE("LINPGA34 Switch", WM8991_INPUT_MIXER3, WM8991_L34MNB_BIT, + 1, 0), +}; + +/* INMIXR */ +static const struct snd_kcontrol_new wm8991_dapm_inmixr_controls[] = { + SOC_DAPM_SINGLE_TLV("Record Right Volume", WM8991_INPUT_MIXER4, + WM8991_RDBVOL_SHIFT, WM8991_RDBVOL_MASK, 0, in_mix_tlv), + SOC_DAPM_SINGLE_TLV("RIN2 Volume", WM8991_INPUT_MIXER6, WM8991_RI2BVOL_SHIFT, + 7, 0, in_mix_tlv), + SOC_DAPM_SINGLE("RINPGA12 Switch", WM8991_INPUT_MIXER3, WM8991_L12MNB_BIT, + 1, 0), + SOC_DAPM_SINGLE("RINPGA34 Switch", WM8991_INPUT_MIXER3, WM8991_L34MNB_BIT, + 1, 0), +}; + +/* AINLMUX */ +static const char *wm8991_ainlmux[] = +{"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"}; + +static const struct soc_enum wm8991_ainlmux_enum = + SOC_ENUM_SINGLE(WM8991_INPUT_MIXER1, WM8991_AINLMODE_SHIFT, + ARRAY_SIZE(wm8991_ainlmux), wm8991_ainlmux); + +static const struct snd_kcontrol_new wm8991_dapm_ainlmux_controls = + SOC_DAPM_ENUM("Route", wm8991_ainlmux_enum); + +/* DIFFINL */ + +/* AINRMUX */ +static const char *wm8991_ainrmux[] = +{"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"}; + +static const struct soc_enum wm8991_ainrmux_enum = + SOC_ENUM_SINGLE(WM8991_INPUT_MIXER1, WM8991_AINRMODE_SHIFT, + ARRAY_SIZE(wm8991_ainrmux), wm8991_ainrmux); + +static const struct snd_kcontrol_new wm8991_dapm_ainrmux_controls = + SOC_DAPM_ENUM("Route", wm8991_ainrmux_enum); + +/* RXVOICE */ +static const struct snd_kcontrol_new wm8991_dapm_rxvoice_controls[] = { + SOC_DAPM_SINGLE_TLV("LIN4RXN", WM8991_INPUT_MIXER5, WM8991_LR4BVOL_SHIFT, + WM8991_LR4BVOL_MASK, 0, in_mix_tlv), + SOC_DAPM_SINGLE_TLV("RIN4RXP", WM8991_INPUT_MIXER6, WM8991_RL4BVOL_SHIFT, + WM8991_RL4BVOL_MASK, 0, in_mix_tlv), +}; + +/* LOMIX */ +static const struct snd_kcontrol_new wm8991_dapm_lomix_controls[] = { + SOC_DAPM_SINGLE("LOMIX Right ADC Bypass Switch", WM8991_OUTPUT_MIXER1, + WM8991_LRBLO_BIT, 1, 0), + SOC_DAPM_SINGLE("LOMIX Left ADC Bypass Switch", WM8991_OUTPUT_MIXER1, + WM8991_LLBLO_BIT, 1, 0), + SOC_DAPM_SINGLE("LOMIX RIN3 Bypass Switch", WM8991_OUTPUT_MIXER1, + WM8991_LRI3LO_BIT, 1, 0), + SOC_DAPM_SINGLE("LOMIX LIN3 Bypass Switch", WM8991_OUTPUT_MIXER1, + WM8991_LLI3LO_BIT, 1, 0), + SOC_DAPM_SINGLE("LOMIX RIN12 PGA Bypass Switch", WM8991_OUTPUT_MIXER1, + WM8991_LR12LO_BIT, 1, 0), + SOC_DAPM_SINGLE("LOMIX LIN12 PGA Bypass Switch", WM8991_OUTPUT_MIXER1, + WM8991_LL12LO_BIT, 1, 0), + SOC_DAPM_SINGLE("LOMIX Left DAC Switch", WM8991_OUTPUT_MIXER1, + WM8991_LDLO_BIT, 1, 0), +}; + +/* ROMIX */ +static const struct snd_kcontrol_new wm8991_dapm_romix_controls[] = { + SOC_DAPM_SINGLE("ROMIX Left ADC Bypass Switch", WM8991_OUTPUT_MIXER2, + WM8991_RLBRO_BIT, 1, 0), + SOC_DAPM_SINGLE("ROMIX Right ADC Bypass Switch", WM8991_OUTPUT_MIXER2, + WM8991_RRBRO_BIT, 1, 0), + SOC_DAPM_SINGLE("ROMIX LIN3 Bypass Switch", WM8991_OUTPUT_MIXER2, + WM8991_RLI3RO_BIT, 1, 0), + SOC_DAPM_SINGLE("ROMIX RIN3 Bypass Switch", WM8991_OUTPUT_MIXER2, + WM8991_RRI3RO_BIT, 1, 0), + SOC_DAPM_SINGLE("ROMIX LIN12 PGA Bypass Switch", WM8991_OUTPUT_MIXER2, + WM8991_RL12RO_BIT, 1, 0), + SOC_DAPM_SINGLE("ROMIX RIN12 PGA Bypass Switch", WM8991_OUTPUT_MIXER2, + WM8991_RR12RO_BIT, 1, 0), + SOC_DAPM_SINGLE("ROMIX Right DAC Switch", WM8991_OUTPUT_MIXER2, + WM8991_RDRO_BIT, 1, 0), +}; + +/* LONMIX */ +static const struct snd_kcontrol_new wm8991_dapm_lonmix_controls[] = { + SOC_DAPM_SINGLE("LONMIX Left Mixer PGA Switch", WM8991_LINE_MIXER1, + WM8991_LLOPGALON_BIT, 1, 0), + SOC_DAPM_SINGLE("LONMIX Right Mixer PGA Switch", WM8991_LINE_MIXER1, + WM8991_LROPGALON_BIT, 1, 0), + SOC_DAPM_SINGLE("LONMIX Inverted LOP Switch", WM8991_LINE_MIXER1, + WM8991_LOPLON_BIT, 1, 0), +}; + +/* LOPMIX */ +static const struct snd_kcontrol_new wm8991_dapm_lopmix_controls[] = { + SOC_DAPM_SINGLE("LOPMIX Right Mic Bypass Switch", WM8991_LINE_MIXER1, + WM8991_LR12LOP_BIT, 1, 0), + SOC_DAPM_SINGLE("LOPMIX Left Mic Bypass Switch", WM8991_LINE_MIXER1, + WM8991_LL12LOP_BIT, 1, 0), + SOC_DAPM_SINGLE("LOPMIX Left Mixer PGA Switch", WM8991_LINE_MIXER1, + WM8991_LLOPGALOP_BIT, 1, 0), +}; + +/* RONMIX */ +static const struct snd_kcontrol_new wm8991_dapm_ronmix_controls[] = { + SOC_DAPM_SINGLE("RONMIX Right Mixer PGA Switch", WM8991_LINE_MIXER2, + WM8991_RROPGARON_BIT, 1, 0), + SOC_DAPM_SINGLE("RONMIX Left Mixer PGA Switch", WM8991_LINE_MIXER2, + WM8991_RLOPGARON_BIT, 1, 0), + SOC_DAPM_SINGLE("RONMIX Inverted ROP Switch", WM8991_LINE_MIXER2, + WM8991_ROPRON_BIT, 1, 0), +}; + +/* ROPMIX */ +static const struct snd_kcontrol_new wm8991_dapm_ropmix_controls[] = { + SOC_DAPM_SINGLE("ROPMIX Left Mic Bypass Switch", WM8991_LINE_MIXER2, + WM8991_RL12ROP_BIT, 1, 0), + SOC_DAPM_SINGLE("ROPMIX Right Mic Bypass Switch", WM8991_LINE_MIXER2, + WM8991_RR12ROP_BIT, 1, 0), + SOC_DAPM_SINGLE("ROPMIX Right Mixer PGA Switch", WM8991_LINE_MIXER2, + WM8991_RROPGAROP_BIT, 1, 0), +}; + +/* OUT3MIX */ +static const struct snd_kcontrol_new wm8991_dapm_out3mix_controls[] = { + SOC_DAPM_SINGLE("OUT3MIX LIN4RXN Bypass Switch", WM8991_OUT3_4_MIXER, + WM8991_LI4O3_BIT, 1, 0), + SOC_DAPM_SINGLE("OUT3MIX Left Out PGA Switch", WM8991_OUT3_4_MIXER, + WM8991_LPGAO3_BIT, 1, 0), +}; + +/* OUT4MIX */ +static const struct snd_kcontrol_new wm8991_dapm_out4mix_controls[] = { + SOC_DAPM_SINGLE("OUT4MIX Right Out PGA Switch", WM8991_OUT3_4_MIXER, + WM8991_RPGAO4_BIT, 1, 0), + SOC_DAPM_SINGLE("OUT4MIX RIN4RXP Bypass Switch", WM8991_OUT3_4_MIXER, + WM8991_RI4O4_BIT, 1, 0), +}; + +/* SPKMIX */ +static const struct snd_kcontrol_new wm8991_dapm_spkmix_controls[] = { + SOC_DAPM_SINGLE("SPKMIX LIN2 Bypass Switch", WM8991_SPEAKER_MIXER, + WM8991_LI2SPK_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX LADC Bypass Switch", WM8991_SPEAKER_MIXER, + WM8991_LB2SPK_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX Left Mixer PGA Switch", WM8991_SPEAKER_MIXER, + WM8991_LOPGASPK_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX Left DAC Switch", WM8991_SPEAKER_MIXER, + WM8991_LDSPK_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX Right DAC Switch", WM8991_SPEAKER_MIXER, + WM8991_RDSPK_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX Right Mixer PGA Switch", WM8991_SPEAKER_MIXER, + WM8991_ROPGASPK_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX RADC Bypass Switch", WM8991_SPEAKER_MIXER, + WM8991_RL12ROP_BIT, 1, 0), + SOC_DAPM_SINGLE("SPKMIX RIN2 Bypass Switch", WM8991_SPEAKER_MIXER, + WM8991_RI2SPK_BIT, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8991_dapm_widgets[] = { + /* Input Side */ + /* Input Lines */ + SND_SOC_DAPM_INPUT("LIN1"), + SND_SOC_DAPM_INPUT("LIN2"), + SND_SOC_DAPM_INPUT("LIN3"), + SND_SOC_DAPM_INPUT("LIN4RXN"), + SND_SOC_DAPM_INPUT("RIN3"), + SND_SOC_DAPM_INPUT("RIN4RXP"), + SND_SOC_DAPM_INPUT("RIN1"), + SND_SOC_DAPM_INPUT("RIN2"), + SND_SOC_DAPM_INPUT("Internal ADC Source"), + + /* DACs */ + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8991_POWER_MANAGEMENT_2, + WM8991_ADCL_ENA_BIT, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8991_POWER_MANAGEMENT_2, + WM8991_ADCR_ENA_BIT, 0), + + /* Input PGAs */ + SND_SOC_DAPM_MIXER("LIN12 PGA", WM8991_POWER_MANAGEMENT_2, WM8991_LIN12_ENA_BIT, + 0, &wm8991_dapm_lin12_pga_controls[0], + ARRAY_SIZE(wm8991_dapm_lin12_pga_controls)), + SND_SOC_DAPM_MIXER("LIN34 PGA", WM8991_POWER_MANAGEMENT_2, WM8991_LIN34_ENA_BIT, + 0, &wm8991_dapm_lin34_pga_controls[0], + ARRAY_SIZE(wm8991_dapm_lin34_pga_controls)), + SND_SOC_DAPM_MIXER("RIN12 PGA", WM8991_POWER_MANAGEMENT_2, WM8991_RIN12_ENA_BIT, + 0, &wm8991_dapm_rin12_pga_controls[0], + ARRAY_SIZE(wm8991_dapm_rin12_pga_controls)), + SND_SOC_DAPM_MIXER("RIN34 PGA", WM8991_POWER_MANAGEMENT_2, WM8991_RIN34_ENA_BIT, + 0, &wm8991_dapm_rin34_pga_controls[0], + ARRAY_SIZE(wm8991_dapm_rin34_pga_controls)), + + /* INMIXL */ + SND_SOC_DAPM_MIXER_E("INMIXL", WM8991_INTDRIVBITS, WM8991_INMIXL_PWR_BIT, 0, + &wm8991_dapm_inmixl_controls[0], + ARRAY_SIZE(wm8991_dapm_inmixl_controls), + inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* AINLMUX */ + SND_SOC_DAPM_MUX_E("AINLMUX", WM8991_INTDRIVBITS, WM8991_AINLMUX_PWR_BIT, 0, + &wm8991_dapm_ainlmux_controls, inmixer_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* INMIXR */ + SND_SOC_DAPM_MIXER_E("INMIXR", WM8991_INTDRIVBITS, WM8991_INMIXR_PWR_BIT, 0, + &wm8991_dapm_inmixr_controls[0], + ARRAY_SIZE(wm8991_dapm_inmixr_controls), + inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* AINRMUX */ + SND_SOC_DAPM_MUX_E("AINRMUX", WM8991_INTDRIVBITS, WM8991_AINRMUX_PWR_BIT, 0, + &wm8991_dapm_ainrmux_controls, inmixer_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8991_POWER_MANAGEMENT_3, + WM8991_DACL_ENA_BIT, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8991_POWER_MANAGEMENT_3, + WM8991_DACR_ENA_BIT, 0), + + /* LOMIX */ + SND_SOC_DAPM_MIXER_E("LOMIX", WM8991_POWER_MANAGEMENT_3, WM8991_LOMIX_ENA_BIT, + 0, &wm8991_dapm_lomix_controls[0], + ARRAY_SIZE(wm8991_dapm_lomix_controls), + outmixer_event, SND_SOC_DAPM_PRE_REG), + + /* LONMIX */ + SND_SOC_DAPM_MIXER("LONMIX", WM8991_POWER_MANAGEMENT_3, WM8991_LON_ENA_BIT, 0, + &wm8991_dapm_lonmix_controls[0], + ARRAY_SIZE(wm8991_dapm_lonmix_controls)), + + /* LOPMIX */ + SND_SOC_DAPM_MIXER("LOPMIX", WM8991_POWER_MANAGEMENT_3, WM8991_LOP_ENA_BIT, 0, + &wm8991_dapm_lopmix_controls[0], + ARRAY_SIZE(wm8991_dapm_lopmix_controls)), + + /* OUT3MIX */ + SND_SOC_DAPM_MIXER("OUT3MIX", WM8991_POWER_MANAGEMENT_1, WM8991_OUT3_ENA_BIT, 0, + &wm8991_dapm_out3mix_controls[0], + ARRAY_SIZE(wm8991_dapm_out3mix_controls)), + + /* SPKMIX */ + SND_SOC_DAPM_MIXER_E("SPKMIX", WM8991_POWER_MANAGEMENT_1, WM8991_SPK_ENA_BIT, 0, + &wm8991_dapm_spkmix_controls[0], + ARRAY_SIZE(wm8991_dapm_spkmix_controls), outmixer_event, + SND_SOC_DAPM_PRE_REG), + + /* OUT4MIX */ + SND_SOC_DAPM_MIXER("OUT4MIX", WM8991_POWER_MANAGEMENT_1, WM8991_OUT4_ENA_BIT, 0, + &wm8991_dapm_out4mix_controls[0], + ARRAY_SIZE(wm8991_dapm_out4mix_controls)), + + /* ROPMIX */ + SND_SOC_DAPM_MIXER("ROPMIX", WM8991_POWER_MANAGEMENT_3, WM8991_ROP_ENA_BIT, 0, + &wm8991_dapm_ropmix_controls[0], + ARRAY_SIZE(wm8991_dapm_ropmix_controls)), + + /* RONMIX */ + SND_SOC_DAPM_MIXER("RONMIX", WM8991_POWER_MANAGEMENT_3, WM8991_RON_ENA_BIT, 0, + &wm8991_dapm_ronmix_controls[0], + ARRAY_SIZE(wm8991_dapm_ronmix_controls)), + + /* ROMIX */ + SND_SOC_DAPM_MIXER_E("ROMIX", WM8991_POWER_MANAGEMENT_3, WM8991_ROMIX_ENA_BIT, + 0, &wm8991_dapm_romix_controls[0], + ARRAY_SIZE(wm8991_dapm_romix_controls), + outmixer_event, SND_SOC_DAPM_PRE_REG), + + /* LOUT PGA */ + SND_SOC_DAPM_PGA("LOUT PGA", WM8991_POWER_MANAGEMENT_1, WM8991_LOUT_ENA_BIT, 0, + NULL, 0), + + /* ROUT PGA */ + SND_SOC_DAPM_PGA("ROUT PGA", WM8991_POWER_MANAGEMENT_1, WM8991_ROUT_ENA_BIT, 0, + NULL, 0), + + /* LOPGA */ + SND_SOC_DAPM_PGA("LOPGA", WM8991_POWER_MANAGEMENT_3, WM8991_LOPGA_ENA_BIT, 0, + NULL, 0), + + /* ROPGA */ + SND_SOC_DAPM_PGA("ROPGA", WM8991_POWER_MANAGEMENT_3, WM8991_ROPGA_ENA_BIT, 0, + NULL, 0), + + /* MICBIAS */ + SND_SOC_DAPM_MICBIAS("MICBIAS", WM8991_POWER_MANAGEMENT_1, + WM8991_MICBIAS_ENA_BIT, 0), + + SND_SOC_DAPM_OUTPUT("LON"), + SND_SOC_DAPM_OUTPUT("LOP"), + SND_SOC_DAPM_OUTPUT("OUT3"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("SPKN"), + SND_SOC_DAPM_OUTPUT("SPKP"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_OUTPUT("OUT4"), + SND_SOC_DAPM_OUTPUT("ROP"), + SND_SOC_DAPM_OUTPUT("RON"), + SND_SOC_DAPM_OUTPUT("OUT"), + + SND_SOC_DAPM_OUTPUT("Internal DAC Sink"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Make DACs turn on when playing even if not mixed into any outputs */ + {"Internal DAC Sink", NULL, "Left DAC"}, + {"Internal DAC Sink", NULL, "Right DAC"}, + + /* Make ADCs turn on when recording even if not mixed from any inputs */ + {"Left ADC", NULL, "Internal ADC Source"}, + {"Right ADC", NULL, "Internal ADC Source"}, + + /* Input Side */ + /* LIN12 PGA */ + {"LIN12 PGA", "LIN1 Switch", "LIN1"}, + {"LIN12 PGA", "LIN2 Switch", "LIN2"}, + /* LIN34 PGA */ + {"LIN34 PGA", "LIN3 Switch", "LIN3"}, + {"LIN34 PGA", "LIN4 Switch", "LIN4RXN"}, + /* INMIXL */ + {"INMIXL", "Record Left Volume", "LOMIX"}, + {"INMIXL", "LIN2 Volume", "LIN2"}, + {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"}, + {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"}, + /* AINLMUX */ + {"AINLMUX", "INMIXL Mix", "INMIXL"}, + {"AINLMUX", "DIFFINL Mix", "LIN12 PGA"}, + {"AINLMUX", "DIFFINL Mix", "LIN34 PGA"}, + {"AINLMUX", "RXVOICE Mix", "LIN4RXN"}, + {"AINLMUX", "RXVOICE Mix", "RIN4RXP"}, + /* ADC */ + {"Left ADC", NULL, "AINLMUX"}, + + /* RIN12 PGA */ + {"RIN12 PGA", "RIN1 Switch", "RIN1"}, + {"RIN12 PGA", "RIN2 Switch", "RIN2"}, + /* RIN34 PGA */ + {"RIN34 PGA", "RIN3 Switch", "RIN3"}, + {"RIN34 PGA", "RIN4 Switch", "RIN4RXP"}, + /* INMIXL */ + {"INMIXR", "Record Right Volume", "ROMIX"}, + {"INMIXR", "RIN2 Volume", "RIN2"}, + {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"}, + {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"}, + /* AINRMUX */ + {"AINRMUX", "INMIXR Mix", "INMIXR"}, + {"AINRMUX", "DIFFINR Mix", "RIN12 PGA"}, + {"AINRMUX", "DIFFINR Mix", "RIN34 PGA"}, + {"AINRMUX", "RXVOICE Mix", "LIN4RXN"}, + {"AINRMUX", "RXVOICE Mix", "RIN4RXP"}, + /* ADC */ + {"Right ADC", NULL, "AINRMUX"}, + + /* LOMIX */ + {"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"}, + {"LOMIX", "LOMIX LIN3 Bypass Switch", "LIN3"}, + {"LOMIX", "LOMIX LIN12 PGA Bypass Switch", "LIN12 PGA"}, + {"LOMIX", "LOMIX RIN12 PGA Bypass Switch", "RIN12 PGA"}, + {"LOMIX", "LOMIX Right ADC Bypass Switch", "AINRMUX"}, + {"LOMIX", "LOMIX Left ADC Bypass Switch", "AINLMUX"}, + {"LOMIX", "LOMIX Left DAC Switch", "Left DAC"}, + + /* ROMIX */ + {"ROMIX", "ROMIX RIN3 Bypass Switch", "RIN3"}, + {"ROMIX", "ROMIX LIN3 Bypass Switch", "LIN3"}, + {"ROMIX", "ROMIX LIN12 PGA Bypass Switch", "LIN12 PGA"}, + {"ROMIX", "ROMIX RIN12 PGA Bypass Switch", "RIN12 PGA"}, + {"ROMIX", "ROMIX Right ADC Bypass Switch", "AINRMUX"}, + {"ROMIX", "ROMIX Left ADC Bypass Switch", "AINLMUX"}, + {"ROMIX", "ROMIX Right DAC Switch", "Right DAC"}, + + /* SPKMIX */ + {"SPKMIX", "SPKMIX LIN2 Bypass Switch", "LIN2"}, + {"SPKMIX", "SPKMIX RIN2 Bypass Switch", "RIN2"}, + {"SPKMIX", "SPKMIX LADC Bypass Switch", "AINLMUX"}, + {"SPKMIX", "SPKMIX RADC Bypass Switch", "AINRMUX"}, + {"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"}, + {"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"}, + {"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"}, + {"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"}, + + /* LONMIX */ + {"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"}, + {"LONMIX", "LONMIX Right Mixer PGA Switch", "ROPGA"}, + {"LONMIX", "LONMIX Inverted LOP Switch", "LOPMIX"}, + + /* LOPMIX */ + {"LOPMIX", "LOPMIX Right Mic Bypass Switch", "RIN12 PGA"}, + {"LOPMIX", "LOPMIX Left Mic Bypass Switch", "LIN12 PGA"}, + {"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"}, + + /* OUT3MIX */ + {"OUT3MIX", "OUT3MIX LIN4RXN Bypass Switch", "LIN4RXN"}, + {"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"}, + + /* OUT4MIX */ + {"OUT4MIX", "OUT4MIX Right Out PGA Switch", "ROPGA"}, + {"OUT4MIX", "OUT4MIX RIN4RXP Bypass Switch", "RIN4RXP"}, + + /* RONMIX */ + {"RONMIX", "RONMIX Right Mixer PGA Switch", "ROPGA"}, + {"RONMIX", "RONMIX Left Mixer PGA Switch", "LOPGA"}, + {"RONMIX", "RONMIX Inverted ROP Switch", "ROPMIX"}, + + /* ROPMIX */ + {"ROPMIX", "ROPMIX Left Mic Bypass Switch", "LIN12 PGA"}, + {"ROPMIX", "ROPMIX Right Mic Bypass Switch", "RIN12 PGA"}, + {"ROPMIX", "ROPMIX Right Mixer PGA Switch", "ROPGA"}, + + /* Out Mixer PGAs */ + {"LOPGA", NULL, "LOMIX"}, + {"ROPGA", NULL, "ROMIX"}, + + {"LOUT PGA", NULL, "LOMIX"}, + {"ROUT PGA", NULL, "ROMIX"}, + + /* Output Pins */ + {"LON", NULL, "LONMIX"}, + {"LOP", NULL, "LOPMIX"}, + {"OUT", NULL, "OUT3MIX"}, + {"LOUT", NULL, "LOUT PGA"}, + {"SPKN", NULL, "SPKMIX"}, + {"ROUT", NULL, "ROUT PGA"}, + {"OUT4", NULL, "OUT4MIX"}, + {"ROP", NULL, "ROPMIX"}, + {"RON", NULL, "RONMIX"}, +}; + +/* PLL divisors */ +struct _pll_div { + u32 div2; + u32 n; + u32 k; +}; + +/* The size in bits of the pll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 16) * 10) + +static void pll_factors(struct _pll_div *pll_div, unsigned int target, + unsigned int source) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod; + + + Ndiv = target / source; + if (Ndiv < 6) { + source >>= 1; + pll_div->div2 = 1; + Ndiv = target / source; + } else + pll_div->div2 = 0; + + if ((Ndiv < 6) || (Ndiv > 12)) + printk(KERN_WARNING + "WM8991 N value outwith recommended range! N = %d\n", Ndiv); + + pll_div->n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div->k = K; +} + +static int wm8991_set_dai_pll(struct snd_soc_dai *codec_dai, + int pll_id, int src, unsigned int freq_in, unsigned int freq_out) +{ + u16 reg; + struct snd_soc_codec *codec = codec_dai->codec; + struct _pll_div pll_div; + + if (freq_in && freq_out) { + pll_factors(&pll_div, freq_out * 4, freq_in); + + /* Turn on PLL */ + reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_2); + reg |= WM8991_PLL_ENA; + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_2, reg); + + /* sysclk comes from PLL */ + reg = snd_soc_read(codec, WM8991_CLOCKING_2); + snd_soc_write(codec, WM8991_CLOCKING_2, reg | WM8991_SYSCLK_SRC); + + /* set up N , fractional mode and pre-divisor if neccessary */ + snd_soc_write(codec, WM8991_PLL1, pll_div.n | WM8991_SDM | + (pll_div.div2 ? WM8991_PRESCALE : 0)); + snd_soc_write(codec, WM8991_PLL2, (u8)(pll_div.k>>8)); + snd_soc_write(codec, WM8991_PLL3, (u8)(pll_div.k & 0xFF)); + } else { + /* Turn on PLL */ + reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_2); + reg &= ~WM8991_PLL_ENA; + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_2, reg); + } + return 0; +} + +/* + * Set's ADC and Voice DAC format. + */ +static int wm8991_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 audio1, audio3; + + audio1 = snd_soc_read(codec, WM8991_AUDIO_INTERFACE_1); + audio3 = snd_soc_read(codec, WM8991_AUDIO_INTERFACE_3); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + audio3 &= ~WM8991_AIF_MSTR1; + break; + case SND_SOC_DAIFMT_CBM_CFM: + audio3 |= WM8991_AIF_MSTR1; + break; + default: + return -EINVAL; + } + + audio1 &= ~WM8991_AIF_FMT_MASK; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + audio1 |= WM8991_AIF_TMF_I2S; + audio1 &= ~WM8991_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_RIGHT_J: + audio1 |= WM8991_AIF_TMF_RIGHTJ; + audio1 &= ~WM8991_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_LEFT_J: + audio1 |= WM8991_AIF_TMF_LEFTJ; + audio1 &= ~WM8991_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_DSP_A: + audio1 |= WM8991_AIF_TMF_DSP; + audio1 &= ~WM8991_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_DSP_B: + audio1 |= WM8991_AIF_TMF_DSP | WM8991_AIF_LRCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, WM8991_AUDIO_INTERFACE_1, audio1); + snd_soc_write(codec, WM8991_AUDIO_INTERFACE_3, audio3); + return 0; +} + +static int wm8991_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM8991_MCLK_DIV: + reg = snd_soc_read(codec, WM8991_CLOCKING_2) & + ~WM8991_MCLK_DIV_MASK; + snd_soc_write(codec, WM8991_CLOCKING_2, reg | div); + break; + case WM8991_DACCLK_DIV: + reg = snd_soc_read(codec, WM8991_CLOCKING_2) & + ~WM8991_DAC_CLKDIV_MASK; + snd_soc_write(codec, WM8991_CLOCKING_2, reg | div); + break; + case WM8991_ADCCLK_DIV: + reg = snd_soc_read(codec, WM8991_CLOCKING_2) & + ~WM8991_ADC_CLKDIV_MASK; + snd_soc_write(codec, WM8991_CLOCKING_2, reg | div); + break; + case WM8991_BCLK_DIV: + reg = snd_soc_read(codec, WM8991_CLOCKING_1) & + ~WM8991_BCLK_DIV_MASK; + snd_soc_write(codec, WM8991_CLOCKING_1, reg | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +/* + * Set PCM DAI bit size and sample rate. + */ +static int wm8991_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + u16 audio1 = snd_soc_read(codec, WM8991_AUDIO_INTERFACE_1); + + audio1 &= ~WM8991_AIF_WL_MASK; + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + audio1 |= WM8991_AIF_WL_20BITS; + break; + case SNDRV_PCM_FORMAT_S24_LE: + audio1 |= WM8991_AIF_WL_24BITS; + break; + case SNDRV_PCM_FORMAT_S32_LE: + audio1 |= WM8991_AIF_WL_32BITS; + break; + } + + snd_soc_write(codec, WM8991_AUDIO_INTERFACE_1, audio1); + return 0; +} + +static int wm8991_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 val; + + val = snd_soc_read(codec, WM8991_DAC_CTRL) & ~WM8991_DAC_MUTE; + if (mute) + snd_soc_write(codec, WM8991_DAC_CTRL, val | WM8991_DAC_MUTE); + else + snd_soc_write(codec, WM8991_DAC_CTRL, val); + return 0; +} + +static int wm8991_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 val; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* VMID=2*50k */ + val = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_1) & + ~WM8991_VMID_MODE_MASK; + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, val | 0x2); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + snd_soc_cache_sync(codec); + /* Enable all output discharge bits */ + snd_soc_write(codec, WM8991_ANTIPOP1, WM8991_DIS_LLINE | + WM8991_DIS_RLINE | WM8991_DIS_OUT3 | + WM8991_DIS_OUT4 | WM8991_DIS_LOUT | + WM8991_DIS_ROUT); + + /* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */ + snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST | + WM8991_BUFDCOPEN | WM8991_POBCTRL | + WM8991_VMIDTOG); + + /* Delay to allow output caps to discharge */ + msleep(300); + + /* Disable VMIDTOG */ + snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST | + WM8991_BUFDCOPEN | WM8991_POBCTRL); + + /* disable all output discharge bits */ + snd_soc_write(codec, WM8991_ANTIPOP1, 0); + + /* Enable outputs */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1b00); + + msleep(50); + + /* Enable VMID at 2x50k */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1f02); + + msleep(100); + + /* Enable VREF */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1f03); + + msleep(600); + + /* Enable BUFIOEN */ + snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST | + WM8991_BUFDCOPEN | WM8991_POBCTRL | + WM8991_BUFIOEN); + + /* Disable outputs */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x3); + + /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ + snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_BUFIOEN); + } + + /* VMID=2*250k */ + val = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_1) & + ~WM8991_VMID_MODE_MASK; + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, val | 0x4); + break; + + case SND_SOC_BIAS_OFF: + /* Enable POBCTRL and SOFT_ST */ + snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST | + WM8991_POBCTRL | WM8991_BUFIOEN); + + /* Enable POBCTRL, SOFT_ST and BUFDCOPEN */ + snd_soc_write(codec, WM8991_ANTIPOP2, WM8991_SOFTST | + WM8991_BUFDCOPEN | WM8991_POBCTRL | + WM8991_BUFIOEN); + + /* mute DAC */ + val = snd_soc_read(codec, WM8991_DAC_CTRL); + snd_soc_write(codec, WM8991_DAC_CTRL, val | WM8991_DAC_MUTE); + + /* Enable any disabled outputs */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1f03); + + /* Disable VMID */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x1f01); + + msleep(300); + + /* Enable all output discharge bits */ + snd_soc_write(codec, WM8991_ANTIPOP1, WM8991_DIS_LLINE | + WM8991_DIS_RLINE | WM8991_DIS_OUT3 | + WM8991_DIS_OUT4 | WM8991_DIS_LOUT | + WM8991_DIS_ROUT); + + /* Disable VREF */ + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, 0x0); + + /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ + snd_soc_write(codec, WM8991_ANTIPOP2, 0x0); + codec->cache_sync = 1; + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +static int wm8991_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + wm8991_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8991_resume(struct snd_soc_codec *codec) +{ + wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +/* power down chip */ +static int wm8991_remove(struct snd_soc_codec *codec) +{ + wm8991_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8991_probe(struct snd_soc_codec *codec) +{ + struct wm8991_priv *wm8991; + int ret; + unsigned int reg; + + wm8991 = snd_soc_codec_get_drvdata(codec); + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm8991->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); + return ret; + } + + ret = wm8991_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + return ret; + } + + wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + reg = snd_soc_read(codec, WM8991_AUDIO_INTERFACE_4); + snd_soc_write(codec, WM8991_AUDIO_INTERFACE_4, reg | WM8991_ALRCGPIO1); + + reg = snd_soc_read(codec, WM8991_GPIO1_GPIO2) & + ~WM8991_GPIO1_SEL_MASK; + snd_soc_write(codec, WM8991_GPIO1_GPIO2, reg | 1); + + reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_1); + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, reg | WM8991_VREF_ENA| + WM8991_VMID_MODE_MASK); + + reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_2); + snd_soc_write(codec, WM8991_POWER_MANAGEMENT_2, reg | WM8991_OPCLK_ENA); + + snd_soc_write(codec, WM8991_DAC_CTRL, 0); + snd_soc_write(codec, WM8991_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); + snd_soc_write(codec, WM8991_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); + + snd_soc_add_controls(codec, wm8991_snd_controls, + ARRAY_SIZE(wm8991_snd_controls)); + + snd_soc_dapm_new_controls(&codec->dapm, wm8991_dapm_widgets, + ARRAY_SIZE(wm8991_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, audio_map, + ARRAY_SIZE(audio_map)); + return 0; +} + +#define WM8991_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops wm8991_ops = { + .hw_params = wm8991_hw_params, + .digital_mute = wm8991_mute, + .set_fmt = wm8991_set_dai_fmt, + .set_clkdiv = wm8991_set_dai_clkdiv, + .set_pll = wm8991_set_dai_pll +}; + +/* + * The WM8991 supports 2 different and mutually exclusive DAI + * configurations. + * + * 1. ADC/DAC on Primary Interface + * 2. ADC on Primary Interface/DAC on secondary + */ +static struct snd_soc_dai_driver wm8991_dai = { + /* ADC/DAC on primary */ + .name = "wm8991", + .id = 1, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = WM8991_FORMATS + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = WM8991_FORMATS + }, + .ops = &wm8991_ops +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm8991 = { + .probe = wm8991_probe, + .remove = wm8991_remove, + .suspend = wm8991_suspend, + .resume = wm8991_resume, + .set_bias_level = wm8991_set_bias_level, + .reg_cache_size = WM8991_MAX_REGISTER + 1, + .reg_word_size = sizeof(u16), + .reg_cache_default = wm8991_reg_defs +}; + +static __devinit int wm8991_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8991_priv *wm8991; + int ret; + + wm8991 = kzalloc(sizeof *wm8991, GFP_KERNEL); + if (!wm8991) + return -ENOMEM; + + wm8991->control_type = SND_SOC_I2C; + i2c_set_clientdata(i2c, wm8991); + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_wm8991, &wm8991_dai, 1); + if (ret < 0) + kfree(wm8991); + return ret; +} + +static __devexit int wm8991_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id wm8991_i2c_id[] = { + { "wm8991", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8991_i2c_id); + +static struct i2c_driver wm8991_i2c_driver = { + .driver = { + .name = "wm8991", + .owner = THIS_MODULE, + }, + .probe = wm8991_i2c_probe, + .remove = __devexit_p(wm8991_i2c_remove), + .id_table = wm8991_i2c_id, +}; + +static int __init wm8991_modinit(void) +{ + int ret; + ret = i2c_add_driver(&wm8991_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8991 I2C driver: %d\n", + ret); + } + return 0; +} +module_init(wm8991_modinit); + +static void __exit wm8991_exit(void) +{ + i2c_del_driver(&wm8991_i2c_driver); +} +module_exit(wm8991_exit); + +MODULE_DESCRIPTION("ASoC WM8991 driver"); +MODULE_AUTHOR("Graeme Gregory"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8991.h b/sound/soc/codecs/wm8991.h new file mode 100644 index 00000000000..8a942efd18a --- /dev/null +++ b/sound/soc/codecs/wm8991.h @@ -0,0 +1,833 @@ +/* + * wm8991.h -- audio driver for WM8991 + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef _WM8991_H +#define _WM8991_H + +/* + * Register values. + */ +#define WM8991_RESET 0x00 +#define WM8991_POWER_MANAGEMENT_1 0x01 +#define WM8991_POWER_MANAGEMENT_2 0x02 +#define WM8991_POWER_MANAGEMENT_3 0x03 +#define WM8991_AUDIO_INTERFACE_1 0x04 +#define WM8991_AUDIO_INTERFACE_2 0x05 +#define WM8991_CLOCKING_1 0x06 +#define WM8991_CLOCKING_2 0x07 +#define WM8991_AUDIO_INTERFACE_3 0x08 +#define WM8991_AUDIO_INTERFACE_4 0x09 +#define WM8991_DAC_CTRL 0x0A +#define WM8991_LEFT_DAC_DIGITAL_VOLUME 0x0B +#define WM8991_RIGHT_DAC_DIGITAL_VOLUME 0x0C +#define WM8991_DIGITAL_SIDE_TONE 0x0D +#define WM8991_ADC_CTRL 0x0E +#define WM8991_LEFT_ADC_DIGITAL_VOLUME 0x0F +#define WM8991_RIGHT_ADC_DIGITAL_VOLUME 0x10 +#define WM8991_GPIO_CTRL_1 0x12 +#define WM8991_GPIO1_GPIO2 0x13 +#define WM8991_GPIO3_GPIO4 0x14 +#define WM8991_GPIO5_GPIO6 0x15 +#define WM8991_GPIOCTRL_2 0x16 +#define WM8991_GPIO_POL 0x17 +#define WM8991_LEFT_LINE_INPUT_1_2_VOLUME 0x18 +#define WM8991_LEFT_LINE_INPUT_3_4_VOLUME 0x19 +#define WM8991_RIGHT_LINE_INPUT_1_2_VOLUME 0x1A +#define WM8991_RIGHT_LINE_INPUT_3_4_VOLUME 0x1B +#define WM8991_LEFT_OUTPUT_VOLUME 0x1C +#define WM8991_RIGHT_OUTPUT_VOLUME 0x1D +#define WM8991_LINE_OUTPUTS_VOLUME 0x1E +#define WM8991_OUT3_4_VOLUME 0x1F +#define WM8991_LEFT_OPGA_VOLUME 0x20 +#define WM8991_RIGHT_OPGA_VOLUME 0x21 +#define WM8991_SPEAKER_VOLUME 0x22 +#define WM8991_CLASSD1 0x23 +#define WM8991_CLASSD3 0x25 +#define WM8991_INPUT_MIXER1 0x27 +#define WM8991_INPUT_MIXER2 0x28 +#define WM8991_INPUT_MIXER3 0x29 +#define WM8991_INPUT_MIXER4 0x2A +#define WM8991_INPUT_MIXER5 0x2B +#define WM8991_INPUT_MIXER6 0x2C +#define WM8991_OUTPUT_MIXER1 0x2D +#define WM8991_OUTPUT_MIXER2 0x2E +#define WM8991_OUTPUT_MIXER3 0x2F +#define WM8991_OUTPUT_MIXER4 0x30 +#define WM8991_OUTPUT_MIXER5 0x31 +#define WM8991_OUTPUT_MIXER6 0x32 +#define WM8991_OUT3_4_MIXER 0x33 +#define WM8991_LINE_MIXER1 0x34 +#define WM8991_LINE_MIXER2 0x35 +#define WM8991_SPEAKER_MIXER 0x36 +#define WM8991_ADDITIONAL_CONTROL 0x37 +#define WM8991_ANTIPOP1 0x38 +#define WM8991_ANTIPOP2 0x39 +#define WM8991_MICBIAS 0x3A +#define WM8991_PLL1 0x3C +#define WM8991_PLL2 0x3D +#define WM8991_PLL3 0x3E +#define WM8991_INTDRIVBITS 0x3F + +#define WM8991_REGISTER_COUNT 60 +#define WM8991_MAX_REGISTER 0x3F + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - Reset + */ +#define WM8991_SW_RESET_CHIP_ID_MASK 0xFFFF /* SW_RESET_CHIP_ID - [15:0] */ + +/* + * R1 (0x01) - Power Management (1) + */ +#define WM8991_SPK_ENA 0x1000 /* SPK_ENA */ +#define WM8991_SPK_ENA_BIT 12 +#define WM8991_OUT3_ENA 0x0800 /* OUT3_ENA */ +#define WM8991_OUT3_ENA_BIT 11 +#define WM8991_OUT4_ENA 0x0400 /* OUT4_ENA */ +#define WM8991_OUT4_ENA_BIT 10 +#define WM8991_LOUT_ENA 0x0200 /* LOUT_ENA */ +#define WM8991_LOUT_ENA_BIT 9 +#define WM8991_ROUT_ENA 0x0100 /* ROUT_ENA */ +#define WM8991_ROUT_ENA_BIT 8 +#define WM8991_MICBIAS_ENA 0x0010 /* MICBIAS_ENA */ +#define WM8991_MICBIAS_ENA_BIT 4 +#define WM8991_VMID_MODE_MASK 0x0006 /* VMID_MODE - [2:1] */ +#define WM8991_VREF_ENA 0x0001 /* VREF_ENA */ +#define WM8991_VREF_ENA_BIT 0 + +/* + * R2 (0x02) - Power Management (2) + */ +#define WM8991_PLL_ENA 0x8000 /* PLL_ENA */ +#define WM8991_PLL_ENA_BIT 15 +#define WM8991_TSHUT_ENA 0x4000 /* TSHUT_ENA */ +#define WM8991_TSHUT_ENA_BIT 14 +#define WM8991_TSHUT_OPDIS 0x2000 /* TSHUT_OPDIS */ +#define WM8991_TSHUT_OPDIS_BIT 13 +#define WM8991_OPCLK_ENA 0x0800 /* OPCLK_ENA */ +#define WM8991_OPCLK_ENA_BIT 11 +#define WM8991_AINL_ENA 0x0200 /* AINL_ENA */ +#define WM8991_AINL_ENA_BIT 9 +#define WM8991_AINR_ENA 0x0100 /* AINR_ENA */ +#define WM8991_AINR_ENA_BIT 8 +#define WM8991_LIN34_ENA 0x0080 /* LIN34_ENA */ +#define WM8991_LIN34_ENA_BIT 7 +#define WM8991_LIN12_ENA 0x0040 /* LIN12_ENA */ +#define WM8991_LIN12_ENA_BIT 6 +#define WM8991_RIN34_ENA 0x0020 /* RIN34_ENA */ +#define WM8991_RIN34_ENA_BIT 5 +#define WM8991_RIN12_ENA 0x0010 /* RIN12_ENA */ +#define WM8991_RIN12_ENA_BIT 4 +#define WM8991_ADCL_ENA 0x0002 /* ADCL_ENA */ +#define WM8991_ADCL_ENA_BIT 1 +#define WM8991_ADCR_ENA 0x0001 /* ADCR_ENA */ +#define WM8991_ADCR_ENA_BIT 0 + +/* + * R3 (0x03) - Power Management (3) + */ +#define WM8991_LON_ENA 0x2000 /* LON_ENA */ +#define WM8991_LON_ENA_BIT 13 +#define WM8991_LOP_ENA 0x1000 /* LOP_ENA */ +#define WM8991_LOP_ENA_BIT 12 +#define WM8991_RON_ENA 0x0800 /* RON_ENA */ +#define WM8991_RON_ENA_BIT 11 +#define WM8991_ROP_ENA 0x0400 /* ROP_ENA */ +#define WM8991_ROP_ENA_BIT 10 +#define WM8991_LOPGA_ENA 0x0080 /* LOPGA_ENA */ +#define WM8991_LOPGA_ENA_BIT 7 +#define WM8991_ROPGA_ENA 0x0040 /* ROPGA_ENA */ +#define WM8991_ROPGA_ENA_BIT 6 +#define WM8991_LOMIX_ENA 0x0020 /* LOMIX_ENA */ +#define WM8991_LOMIX_ENA_BIT 5 +#define WM8991_ROMIX_ENA 0x0010 /* ROMIX_ENA */ +#define WM8991_ROMIX_ENA_BIT 4 +#define WM8991_DACL_ENA 0x0002 /* DACL_ENA */ +#define WM8991_DACL_ENA_BIT 1 +#define WM8991_DACR_ENA 0x0001 /* DACR_ENA */ +#define WM8991_DACR_ENA_BIT 0 + +/* + * R4 (0x04) - Audio Interface (1) + */ +#define WM8991_AIFADCL_SRC 0x8000 /* AIFADCL_SRC */ +#define WM8991_AIFADCR_SRC 0x4000 /* AIFADCR_SRC */ +#define WM8991_AIFADC_TDM 0x2000 /* AIFADC_TDM */ +#define WM8991_AIFADC_TDM_CHAN 0x1000 /* AIFADC_TDM_CHAN */ +#define WM8991_AIF_BCLK_INV 0x0100 /* AIF_BCLK_INV */ +#define WM8991_AIF_LRCLK_INV 0x0080 /* AIF_LRCLK_INV */ +#define WM8991_AIF_WL_MASK 0x0060 /* AIF_WL - [6:5] */ +#define WM8991_AIF_WL_16BITS (0 << 5) +#define WM8991_AIF_WL_20BITS (1 << 5) +#define WM8991_AIF_WL_24BITS (2 << 5) +#define WM8991_AIF_WL_32BITS (3 << 5) +#define WM8991_AIF_FMT_MASK 0x0018 /* AIF_FMT - [4:3] */ +#define WM8991_AIF_TMF_RIGHTJ (0 << 3) +#define WM8991_AIF_TMF_LEFTJ (1 << 3) +#define WM8991_AIF_TMF_I2S (2 << 3) +#define WM8991_AIF_TMF_DSP (3 << 3) + +/* + * R5 (0x05) - Audio Interface (2) + */ +#define WM8991_DACL_SRC 0x8000 /* DACL_SRC */ +#define WM8991_DACR_SRC 0x4000 /* DACR_SRC */ +#define WM8991_AIFDAC_TDM 0x2000 /* AIFDAC_TDM */ +#define WM8991_AIFDAC_TDM_CHAN 0x1000 /* AIFDAC_TDM_CHAN */ +#define WM8991_DAC_BOOST_MASK 0x0C00 /* DAC_BOOST - [11:10] */ +#define WM8991_DAC_COMP 0x0010 /* DAC_COMP */ +#define WM8991_DAC_COMPMODE 0x0008 /* DAC_COMPMODE */ +#define WM8991_ADC_COMP 0x0004 /* ADC_COMP */ +#define WM8991_ADC_COMPMODE 0x0002 /* ADC_COMPMODE */ +#define WM8991_LOOPBACK 0x0001 /* LOOPBACK */ + +/* + * R6 (0x06) - Clocking (1) + */ +#define WM8991_TOCLK_RATE 0x8000 /* TOCLK_RATE */ +#define WM8991_TOCLK_ENA 0x4000 /* TOCLK_ENA */ +#define WM8991_OPCLKDIV_MASK 0x1E00 /* OPCLKDIV - [12:9] */ +#define WM8991_DCLKDIV_MASK 0x01C0 /* DCLKDIV - [8:6] */ +#define WM8991_BCLK_DIV_MASK 0x001E /* BCLK_DIV - [4:1] */ +#define WM8991_BCLK_DIV_1 (0x0 << 1) +#define WM8991_BCLK_DIV_1_5 (0x1 << 1) +#define WM8991_BCLK_DIV_2 (0x2 << 1) +#define WM8991_BCLK_DIV_3 (0x3 << 1) +#define WM8991_BCLK_DIV_4 (0x4 << 1) +#define WM8991_BCLK_DIV_5_5 (0x5 << 1) +#define WM8991_BCLK_DIV_6 (0x6 << 1) +#define WM8991_BCLK_DIV_8 (0x7 << 1) +#define WM8991_BCLK_DIV_11 (0x8 << 1) +#define WM8991_BCLK_DIV_12 (0x9 << 1) +#define WM8991_BCLK_DIV_16 (0xA << 1) +#define WM8991_BCLK_DIV_22 (0xB << 1) +#define WM8991_BCLK_DIV_24 (0xC << 1) +#define WM8991_BCLK_DIV_32 (0xD << 1) +#define WM8991_BCLK_DIV_44 (0xE << 1) +#define WM8991_BCLK_DIV_48 (0xF << 1) + +/* + * R7 (0x07) - Clocking (2) + */ +#define WM8991_MCLK_SRC 0x8000 /* MCLK_SRC */ +#define WM8991_SYSCLK_SRC 0x4000 /* SYSCLK_SRC */ +#define WM8991_CLK_FORCE 0x2000 /* CLK_FORCE */ +#define WM8991_MCLK_DIV_MASK 0x1800 /* MCLK_DIV - [12:11] */ +#define WM8991_MCLK_DIV_1 (0 << 11) +#define WM8991_MCLK_DIV_2 ( 2 << 11) +#define WM8991_MCLK_INV 0x0400 /* MCLK_INV */ +#define WM8991_ADC_CLKDIV_MASK 0x00E0 /* ADC_CLKDIV - [7:5] */ +#define WM8991_ADC_CLKDIV_1 (0 << 5) +#define WM8991_ADC_CLKDIV_1_5 (1 << 5) +#define WM8991_ADC_CLKDIV_2 (2 << 5) +#define WM8991_ADC_CLKDIV_3 (3 << 5) +#define WM8991_ADC_CLKDIV_4 (4 << 5) +#define WM8991_ADC_CLKDIV_5_5 (5 << 5) +#define WM8991_ADC_CLKDIV_6 (6 << 5) +#define WM8991_DAC_CLKDIV_MASK 0x001C /* DAC_CLKDIV - [4:2] */ +#define WM8991_DAC_CLKDIV_1 (0 << 2) +#define WM8991_DAC_CLKDIV_1_5 (1 << 2) +#define WM8991_DAC_CLKDIV_2 (2 << 2) +#define WM8991_DAC_CLKDIV_3 (3 << 2) +#define WM8991_DAC_CLKDIV_4 (4 << 2) +#define WM8991_DAC_CLKDIV_5_5 (5 << 2) +#define WM8991_DAC_CLKDIV_6 (6 << 2) + +/* + * R8 (0x08) - Audio Interface (3) + */ +#define WM8991_AIF_MSTR1 0x8000 /* AIF_MSTR1 */ +#define WM8991_AIF_MSTR2 0x4000 /* AIF_MSTR2 */ +#define WM8991_AIF_SEL 0x2000 /* AIF_SEL */ +#define WM8991_ADCLRC_DIR 0x0800 /* ADCLRC_DIR */ +#define WM8991_ADCLRC_RATE_MASK 0x07FF /* ADCLRC_RATE - [10:0] */ + +/* + * R9 (0x09) - Audio Interface (4) + */ +#define WM8991_ALRCGPIO1 0x8000 /* ALRCGPIO1 */ +#define WM8991_ALRCBGPIO6 0x4000 /* ALRCBGPIO6 */ +#define WM8991_AIF_TRIS 0x2000 /* AIF_TRIS */ +#define WM8991_DACLRC_DIR 0x0800 /* DACLRC_DIR */ +#define WM8991_DACLRC_RATE_MASK 0x07FF /* DACLRC_RATE - [10:0] */ + +/* + * R10 (0x0A) - DAC CTRL + */ +#define WM8991_AIF_LRCLKRATE 0x0400 /* AIF_LRCLKRATE */ +#define WM8991_DAC_MONO 0x0200 /* DAC_MONO */ +#define WM8991_DAC_SB_FILT 0x0100 /* DAC_SB_FILT */ +#define WM8991_DAC_MUTERATE 0x0080 /* DAC_MUTERATE */ +#define WM8991_DAC_MUTEMODE 0x0040 /* DAC_MUTEMODE */ +#define WM8991_DEEMP_MASK 0x0030 /* DEEMP - [5:4] */ +#define WM8991_DAC_MUTE 0x0004 /* DAC_MUTE */ +#define WM8991_DACL_DATINV 0x0002 /* DACL_DATINV */ +#define WM8991_DACR_DATINV 0x0001 /* DACR_DATINV */ + +/* + * R11 (0x0B) - Left DAC Digital Volume + */ +#define WM8991_DAC_VU 0x0100 /* DAC_VU */ +#define WM8991_DACL_VOL_MASK 0x00FF /* DACL_VOL - [7:0] */ +#define WM8991_DACL_VOL_SHIFT 0 +/* + * R12 (0x0C) - Right DAC Digital Volume + */ +#define WM8991_DAC_VU 0x0100 /* DAC_VU */ +#define WM8991_DACR_VOL_MASK 0x00FF /* DACR_VOL - [7:0] */ +#define WM8991_DACR_VOL_SHIFT 0 +/* + * R13 (0x0D) - Digital Side Tone + */ +#define WM8991_ADCL_DAC_SVOL_MASK 0x0F /* ADCL_DAC_SVOL - [12:9] */ +#define WM8991_ADCL_DAC_SVOL_SHIFT 9 +#define WM8991_ADCR_DAC_SVOL_MASK 0x0F /* ADCR_DAC_SVOL - [8:5] */ +#define WM8991_ADCR_DAC_SVOL_SHIFT 5 +#define WM8991_ADC_TO_DACL_MASK 0x03 /* ADC_TO_DACL - [3:2] */ +#define WM8991_ADC_TO_DACL_SHIFT 2 +#define WM8991_ADC_TO_DACR_MASK 0x03 /* ADC_TO_DACR - [1:0] */ +#define WM8991_ADC_TO_DACR_SHIFT 0 + +/* + * R14 (0x0E) - ADC CTRL + */ +#define WM8991_ADC_HPF_ENA 0x0100 /* ADC_HPF_ENA */ +#define WM8991_ADC_HPF_ENA_BIT 8 +#define WM8991_ADC_HPF_CUT_MASK 0x03 /* ADC_HPF_CUT - [6:5] */ +#define WM8991_ADC_HPF_CUT_SHIFT 5 +#define WM8991_ADCL_DATINV 0x0002 /* ADCL_DATINV */ +#define WM8991_ADCL_DATINV_BIT 1 +#define WM8991_ADCR_DATINV 0x0001 /* ADCR_DATINV */ +#define WM8991_ADCR_DATINV_BIT 0 + +/* + * R15 (0x0F) - Left ADC Digital Volume + */ +#define WM8991_ADC_VU 0x0100 /* ADC_VU */ +#define WM8991_ADCL_VOL_MASK 0x00FF /* ADCL_VOL - [7:0] */ +#define WM8991_ADCL_VOL_SHIFT 0 + +/* + * R16 (0x10) - Right ADC Digital Volume + */ +#define WM8991_ADC_VU 0x0100 /* ADC_VU */ +#define WM8991_ADCR_VOL_MASK 0x00FF /* ADCR_VOL - [7:0] */ +#define WM8991_ADCR_VOL_SHIFT 0 + +/* + * R18 (0x12) - GPIO CTRL 1 + */ +#define WM8991_IRQ 0x1000 /* IRQ */ +#define WM8991_TEMPOK 0x0800 /* TEMPOK */ +#define WM8991_MICSHRT 0x0400 /* MICSHRT */ +#define WM8991_MICDET 0x0200 /* MICDET */ +#define WM8991_PLL_LCK 0x0100 /* PLL_LCK */ +#define WM8991_GPI8_STATUS 0x0080 /* GPI8_STATUS */ +#define WM8991_GPI7_STATUS 0x0040 /* GPI7_STATUS */ +#define WM8991_GPIO6_STATUS 0x0020 /* GPIO6_STATUS */ +#define WM8991_GPIO5_STATUS 0x0010 /* GPIO5_STATUS */ +#define WM8991_GPIO4_STATUS 0x0008 /* GPIO4_STATUS */ +#define WM8991_GPIO3_STATUS 0x0004 /* GPIO3_STATUS */ +#define WM8991_GPIO2_STATUS 0x0002 /* GPIO2_STATUS */ +#define WM8991_GPIO1_STATUS 0x0001 /* GPIO1_STATUS */ + +/* + * R19 (0x13) - GPIO1 & GPIO2 + */ +#define WM8991_GPIO2_DEB_ENA 0x8000 /* GPIO2_DEB_ENA */ +#define WM8991_GPIO2_IRQ_ENA 0x4000 /* GPIO2_IRQ_ENA */ +#define WM8991_GPIO2_PU 0x2000 /* GPIO2_PU */ +#define WM8991_GPIO2_PD 0x1000 /* GPIO2_PD */ +#define WM8991_GPIO2_SEL_MASK 0x0F00 /* GPIO2_SEL - [11:8] */ +#define WM8991_GPIO1_DEB_ENA 0x0080 /* GPIO1_DEB_ENA */ +#define WM8991_GPIO1_IRQ_ENA 0x0040 /* GPIO1_IRQ_ENA */ +#define WM8991_GPIO1_PU 0x0020 /* GPIO1_PU */ +#define WM8991_GPIO1_PD 0x0010 /* GPIO1_PD */ +#define WM8991_GPIO1_SEL_MASK 0x000F /* GPIO1_SEL - [3:0] */ + +/* + * R20 (0x14) - GPIO3 & GPIO4 + */ +#define WM8991_GPIO4_DEB_ENA 0x8000 /* GPIO4_DEB_ENA */ +#define WM8991_GPIO4_IRQ_ENA 0x4000 /* GPIO4_IRQ_ENA */ +#define WM8991_GPIO4_PU 0x2000 /* GPIO4_PU */ +#define WM8991_GPIO4_PD 0x1000 /* GPIO4_PD */ +#define WM8991_GPIO4_SEL_MASK 0x0F00 /* GPIO4_SEL - [11:8] */ +#define WM8991_GPIO3_DEB_ENA 0x0080 /* GPIO3_DEB_ENA */ +#define WM8991_GPIO3_IRQ_ENA 0x0040 /* GPIO3_IRQ_ENA */ +#define WM8991_GPIO3_PU 0x0020 /* GPIO3_PU */ +#define WM8991_GPIO3_PD 0x0010 /* GPIO3_PD */ +#define WM8991_GPIO3_SEL_MASK 0x000F /* GPIO3_SEL - [3:0] */ + +/* + * R21 (0x15) - GPIO5 & GPIO6 + */ +#define WM8991_GPIO6_DEB_ENA 0x8000 /* GPIO6_DEB_ENA */ +#define WM8991_GPIO6_IRQ_ENA 0x4000 /* GPIO6_IRQ_ENA */ +#define WM8991_GPIO6_PU 0x2000 /* GPIO6_PU */ +#define WM8991_GPIO6_PD 0x1000 /* GPIO6_PD */ +#define WM8991_GPIO6_SEL_MASK 0x0F00 /* GPIO6_SEL - [11:8] */ +#define WM8991_GPIO5_DEB_ENA 0x0080 /* GPIO5_DEB_ENA */ +#define WM8991_GPIO5_IRQ_ENA 0x0040 /* GPIO5_IRQ_ENA */ +#define WM8991_GPIO5_PU 0x0020 /* GPIO5_PU */ +#define WM8991_GPIO5_PD 0x0010 /* GPIO5_PD */ +#define WM8991_GPIO5_SEL_MASK 0x000F /* GPIO5_SEL - [3:0] */ + +/* + * R22 (0x16) - GPIOCTRL 2 + */ +#define WM8991_RD_3W_ENA 0x8000 /* RD_3W_ENA */ +#define WM8991_MODE_3W4W 0x4000 /* MODE_3W4W */ +#define WM8991_TEMPOK_IRQ_ENA 0x0800 /* TEMPOK_IRQ_ENA */ +#define WM8991_MICSHRT_IRQ_ENA 0x0400 /* MICSHRT_IRQ_ENA */ +#define WM8991_MICDET_IRQ_ENA 0x0200 /* MICDET_IRQ_ENA */ +#define WM8991_PLL_LCK_IRQ_ENA 0x0100 /* PLL_LCK_IRQ_ENA */ +#define WM8991_GPI8_DEB_ENA 0x0080 /* GPI8_DEB_ENA */ +#define WM8991_GPI8_IRQ_ENA 0x0040 /* GPI8_IRQ_ENA */ +#define WM8991_GPI8_ENA 0x0010 /* GPI8_ENA */ +#define WM8991_GPI7_DEB_ENA 0x0008 /* GPI7_DEB_ENA */ +#define WM8991_GPI7_IRQ_ENA 0x0004 /* GPI7_IRQ_ENA */ +#define WM8991_GPI7_ENA 0x0001 /* GPI7_ENA */ + +/* + * R23 (0x17) - GPIO_POL + */ +#define WM8991_IRQ_INV 0x1000 /* IRQ_INV */ +#define WM8991_TEMPOK_POL 0x0800 /* TEMPOK_POL */ +#define WM8991_MICSHRT_POL 0x0400 /* MICSHRT_POL */ +#define WM8991_MICDET_POL 0x0200 /* MICDET_POL */ +#define WM8991_PLL_LCK_POL 0x0100 /* PLL_LCK_POL */ +#define WM8991_GPI8_POL 0x0080 /* GPI8_POL */ +#define WM8991_GPI7_POL 0x0040 /* GPI7_POL */ +#define WM8991_GPIO6_POL 0x0020 /* GPIO6_POL */ +#define WM8991_GPIO5_POL 0x0010 /* GPIO5_POL */ +#define WM8991_GPIO4_POL 0x0008 /* GPIO4_POL */ +#define WM8991_GPIO3_POL 0x0004 /* GPIO3_POL */ +#define WM8991_GPIO2_POL 0x0002 /* GPIO2_POL */ +#define WM8991_GPIO1_POL 0x0001 /* GPIO1_POL */ + +/* + * R24 (0x18) - Left Line Input 1&2 Volume + */ +#define WM8991_IPVU 0x0100 /* IPVU */ +#define WM8991_LI12MUTE 0x0080 /* LI12MUTE */ +#define WM8991_LI12MUTE_BIT 7 +#define WM8991_LI12ZC 0x0040 /* LI12ZC */ +#define WM8991_LI12ZC_BIT 6 +#define WM8991_LIN12VOL_MASK 0x001F /* LIN12VOL - [4:0] */ +#define WM8991_LIN12VOL_SHIFT 0 +/* + * R25 (0x19) - Left Line Input 3&4 Volume + */ +#define WM8991_IPVU 0x0100 /* IPVU */ +#define WM8991_LI34MUTE 0x0080 /* LI34MUTE */ +#define WM8991_LI34MUTE_BIT 7 +#define WM8991_LI34ZC 0x0040 /* LI34ZC */ +#define WM8991_LI34ZC_BIT 6 +#define WM8991_LIN34VOL_MASK 0x001F /* LIN34VOL - [4:0] */ +#define WM8991_LIN34VOL_SHIFT 0 + +/* + * R26 (0x1A) - Right Line Input 1&2 Volume + */ +#define WM8991_IPVU 0x0100 /* IPVU */ +#define WM8991_RI12MUTE 0x0080 /* RI12MUTE */ +#define WM8991_RI12MUTE_BIT 7 +#define WM8991_RI12ZC 0x0040 /* RI12ZC */ +#define WM8991_RI12ZC_BIT 6 +#define WM8991_RIN12VOL_MASK 0x001F /* RIN12VOL - [4:0] */ +#define WM8991_RIN12VOL_SHIFT 0 + +/* + * R27 (0x1B) - Right Line Input 3&4 Volume + */ +#define WM8991_IPVU 0x0100 /* IPVU */ +#define WM8991_RI34MUTE 0x0080 /* RI34MUTE */ +#define WM8991_RI34MUTE_BIT 7 +#define WM8991_RI34ZC 0x0040 /* RI34ZC */ +#define WM8991_RI34ZC_BIT 6 +#define WM8991_RIN34VOL_MASK 0x001F /* RIN34VOL - [4:0] */ +#define WM8991_RIN34VOL_SHIFT 0 + +/* + * R28 (0x1C) - Left Output Volume + */ +#define WM8991_OPVU 0x0100 /* OPVU */ +#define WM8991_LOZC 0x0080 /* LOZC */ +#define WM8991_LOZC_BIT 7 +#define WM8991_LOUTVOL_MASK 0x007F /* LOUTVOL - [6:0] */ +#define WM8991_LOUTVOL_SHIFT 0 +/* + * R29 (0x1D) - Right Output Volume + */ +#define WM8991_OPVU 0x0100 /* OPVU */ +#define WM8991_ROZC 0x0080 /* ROZC */ +#define WM8991_ROZC_BIT 7 +#define WM8991_ROUTVOL_MASK 0x007F /* ROUTVOL - [6:0] */ +#define WM8991_ROUTVOL_SHIFT 0 +/* + * R30 (0x1E) - Line Outputs Volume + */ +#define WM8991_LONMUTE 0x0040 /* LONMUTE */ +#define WM8991_LONMUTE_BIT 6 +#define WM8991_LOPMUTE 0x0020 /* LOPMUTE */ +#define WM8991_LOPMUTE_BIT 5 +#define WM8991_LOATTN 0x0010 /* LOATTN */ +#define WM8991_LOATTN_BIT 4 +#define WM8991_RONMUTE 0x0004 /* RONMUTE */ +#define WM8991_RONMUTE_BIT 2 +#define WM8991_ROPMUTE 0x0002 /* ROPMUTE */ +#define WM8991_ROPMUTE_BIT 1 +#define WM8991_ROATTN 0x0001 /* ROATTN */ +#define WM8991_ROATTN_BIT 0 + +/* + * R31 (0x1F) - Out3/4 Volume + */ +#define WM8991_OUT3MUTE 0x0020 /* OUT3MUTE */ +#define WM8991_OUT3MUTE_BIT 5 +#define WM8991_OUT3ATTN 0x0010 /* OUT3ATTN */ +#define WM8991_OUT3ATTN_BIT 4 +#define WM8991_OUT4MUTE 0x0002 /* OUT4MUTE */ +#define WM8991_OUT4MUTE_BIT 1 +#define WM8991_OUT4ATTN 0x0001 /* OUT4ATTN */ +#define WM8991_OUT4ATTN_BIT 0 + +/* + * R32 (0x20) - Left OPGA Volume + */ +#define WM8991_OPVU 0x0100 /* OPVU */ +#define WM8991_LOPGAZC 0x0080 /* LOPGAZC */ +#define WM8991_LOPGAZC_BIT 7 +#define WM8991_LOPGAVOL_MASK 0x007F /* LOPGAVOL - [6:0] */ +#define WM8991_LOPGAVOL_SHIFT 0 + +/* + * R33 (0x21) - Right OPGA Volume + */ +#define WM8991_OPVU 0x0100 /* OPVU */ +#define WM8991_ROPGAZC 0x0080 /* ROPGAZC */ +#define WM8991_ROPGAZC_BIT 7 +#define WM8991_ROPGAVOL_MASK 0x007F /* ROPGAVOL - [6:0] */ +#define WM8991_ROPGAVOL_SHIFT 0 +/* + * R34 (0x22) - Speaker Volume + */ +#define WM8991_SPKVOL_MASK 0x0003 /* SPKVOL - [1:0] */ +#define WM8991_SPKVOL_SHIFT 0 + +/* + * R35 (0x23) - ClassD1 + */ +#define WM8991_CDMODE 0x0100 /* CDMODE */ +#define WM8991_CDMODE_BIT 8 + +/* + * R37 (0x25) - ClassD3 + */ +#define WM8991_DCGAIN_MASK 0x0007 /* DCGAIN - [5:3] */ +#define WM8991_DCGAIN_SHIFT 3 +#define WM8991_ACGAIN_MASK 0x0007 /* ACGAIN - [2:0] */ +#define WM8991_ACGAIN_SHIFT 0 +/* + * R39 (0x27) - Input Mixer1 + */ +#define WM8991_AINLMODE_MASK 0x000C /* AINLMODE - [3:2] */ +#define WM8991_AINLMODE_SHIFT 2 +#define WM8991_AINRMODE_MASK 0x0003 /* AINRMODE - [1:0] */ +#define WM8991_AINRMODE_SHIFT 0 + +/* + * R40 (0x28) - Input Mixer2 + */ +#define WM8991_LMP4 0x0080 /* LMP4 */ +#define WM8991_LMP4_BIT 7 /* LMP4 */ +#define WM8991_LMN3 0x0040 /* LMN3 */ +#define WM8991_LMN3_BIT 6 /* LMN3 */ +#define WM8991_LMP2 0x0020 /* LMP2 */ +#define WM8991_LMP2_BIT 5 /* LMP2 */ +#define WM8991_LMN1 0x0010 /* LMN1 */ +#define WM8991_LMN1_BIT 4 /* LMN1 */ +#define WM8991_RMP4 0x0008 /* RMP4 */ +#define WM8991_RMP4_BIT 3 /* RMP4 */ +#define WM8991_RMN3 0x0004 /* RMN3 */ +#define WM8991_RMN3_BIT 2 /* RMN3 */ +#define WM8991_RMP2 0x0002 /* RMP2 */ +#define WM8991_RMP2_BIT 1 /* RMP2 */ +#define WM8991_RMN1 0x0001 /* RMN1 */ +#define WM8991_RMN1_BIT 0 /* RMN1 */ + +/* + * R41 (0x29) - Input Mixer3 + */ +#define WM8991_L34MNB 0x0100 /* L34MNB */ +#define WM8991_L34MNB_BIT 8 +#define WM8991_L34MNBST 0x0080 /* L34MNBST */ +#define WM8991_L34MNBST_BIT 7 +#define WM8991_L12MNB 0x0020 /* L12MNB */ +#define WM8991_L12MNB_BIT 5 +#define WM8991_L12MNBST 0x0010 /* L12MNBST */ +#define WM8991_L12MNBST_BIT 4 +#define WM8991_LDBVOL_MASK 0x0007 /* LDBVOL - [2:0] */ +#define WM8991_LDBVOL_SHIFT 0 + +/* + * R42 (0x2A) - Input Mixer4 + */ +#define WM8991_R34MNB 0x0100 /* R34MNB */ +#define WM8991_R34MNB_BIT 8 +#define WM8991_R34MNBST 0x0080 /* R34MNBST */ +#define WM8991_R34MNBST_BIT 7 +#define WM8991_R12MNB 0x0020 /* R12MNB */ +#define WM8991_R12MNB_BIT 5 +#define WM8991_R12MNBST 0x0010 /* R12MNBST */ +#define WM8991_R12MNBST_BIT 4 +#define WM8991_RDBVOL_MASK 0x0007 /* RDBVOL - [2:0] */ +#define WM8991_RDBVOL_SHIFT 0 + +/* + * R43 (0x2B) - Input Mixer5 + */ +#define WM8991_LI2BVOL_MASK 0x07 /* LI2BVOL - [8:6] */ +#define WM8991_LI2BVOL_SHIFT 6 +#define WM8991_LR4BVOL_MASK 0x07 /* LR4BVOL - [5:3] */ +#define WM8991_LR4BVOL_SHIFT 3 +#define WM8991_LL4BVOL_MASK 0x07 /* LL4BVOL - [2:0] */ +#define WM8991_LL4BVOL_SHIFT 0 + +/* + * R44 (0x2C) - Input Mixer6 + */ +#define WM8991_RI2BVOL_MASK 0x07 /* RI2BVOL - [8:6] */ +#define WM8991_RI2BVOL_SHIFT 6 +#define WM8991_RL4BVOL_MASK 0x07 /* RL4BVOL - [5:3] */ +#define WM8991_RL4BVOL_SHIFT 3 +#define WM8991_RR4BVOL_MASK 0x07 /* RR4BVOL - [2:0] */ +#define WM8991_RR4BVOL_SHIFT 0 + +/* + * R45 (0x2D) - Output Mixer1 + */ +#define WM8991_LRBLO 0x0080 /* LRBLO */ +#define WM8991_LRBLO_BIT 7 +#define WM8991_LLBLO 0x0040 /* LLBLO */ +#define WM8991_LLBLO_BIT 6 +#define WM8991_LRI3LO 0x0020 /* LRI3LO */ +#define WM8991_LRI3LO_BIT 5 +#define WM8991_LLI3LO 0x0010 /* LLI3LO */ +#define WM8991_LLI3LO_BIT 4 +#define WM8991_LR12LO 0x0008 /* LR12LO */ +#define WM8991_LR12LO_BIT 3 +#define WM8991_LL12LO 0x0004 /* LL12LO */ +#define WM8991_LL12LO_BIT 2 +#define WM8991_LDLO 0x0001 /* LDLO */ +#define WM8991_LDLO_BIT 0 + +/* + * R46 (0x2E) - Output Mixer2 + */ +#define WM8991_RLBRO 0x0080 /* RLBRO */ +#define WM8991_RLBRO_BIT 7 +#define WM8991_RRBRO 0x0040 /* RRBRO */ +#define WM8991_RRBRO_BIT 6 +#define WM8991_RLI3RO 0x0020 /* RLI3RO */ +#define WM8991_RLI3RO_BIT 5 +#define WM8991_RRI3RO 0x0010 /* RRI3RO */ +#define WM8991_RRI3RO_BIT 4 +#define WM8991_RL12RO 0x0008 /* RL12RO */ +#define WM8991_RL12RO_BIT 3 +#define WM8991_RR12RO 0x0004 /* RR12RO */ +#define WM8991_RR12RO_BIT 2 +#define WM8991_RDRO 0x0001 /* RDRO */ +#define WM8991_RDRO_BIT 0 + +/* + * R47 (0x2F) - Output Mixer3 + */ +#define WM8991_LLI3LOVOL_MASK 0x07 /* LLI3LOVOL - [8:6] */ +#define WM8991_LLI3LOVOL_SHIFT 6 +#define WM8991_LR12LOVOL_MASK 0x07 /* LR12LOVOL - [5:3] */ +#define WM8991_LR12LOVOL_SHIFT 3 +#define WM8991_LL12LOVOL_MASK 0x07 /* LL12LOVOL - [2:0] */ +#define WM8991_LL12LOVOL_SHIFT 0 + +/* + * R48 (0x30) - Output Mixer4 + */ +#define WM8991_RRI3ROVOL_MASK 0x07 /* RRI3ROVOL - [8:6] */ +#define WM8991_RRI3ROVOL_SHIFT 6 +#define WM8991_RL12ROVOL_MASK 0x07 /* RL12ROVOL - [5:3] */ +#define WM8991_RL12ROVOL_SHIFT 3 +#define WM8991_RR12ROVOL_MASK 0x07 /* RR12ROVOL - [2:0] */ +#define WM8991_RR12ROVOL_SHIFT 0 + +/* + * R49 (0x31) - Output Mixer5 + */ +#define WM8991_LRI3LOVOL_MASK 0x07 /* LRI3LOVOL - [8:6] */ +#define WM8991_LRI3LOVOL_SHIFT 6 +#define WM8991_LRBLOVOL_MASK 0x07 /* LRBLOVOL - [5:3] */ +#define WM8991_LRBLOVOL_SHIFT 3 +#define WM8991_LLBLOVOL_MASK 0x07 /* LLBLOVOL - [2:0] */ +#define WM8991_LLBLOVOL_SHIFT 0 + +/* + * R50 (0x32) - Output Mixer6 + */ +#define WM8991_RLI3ROVOL_MASK 0x07 /* RLI3ROVOL - [8:6] */ +#define WM8991_RLI3ROVOL_SHIFT 6 +#define WM8991_RLBROVOL_MASK 0x07 /* RLBROVOL - [5:3] */ +#define WM8991_RLBROVOL_SHIFT 3 +#define WM8991_RRBROVOL_MASK 0x07 /* RRBROVOL - [2:0] */ +#define WM8991_RRBROVOL_SHIFT 0 + +/* + * R51 (0x33) - Out3/4 Mixer + */ +#define WM8991_VSEL_MASK 0x0180 /* VSEL - [8:7] */ +#define WM8991_LI4O3 0x0020 /* LI4O3 */ +#define WM8991_LI4O3_BIT 5 +#define WM8991_LPGAO3 0x0010 /* LPGAO3 */ +#define WM8991_LPGAO3_BIT 4 +#define WM8991_RI4O4 0x0002 /* RI4O4 */ +#define WM8991_RI4O4_BIT 1 +#define WM8991_RPGAO4 0x0001 /* RPGAO4 */ +#define WM8991_RPGAO4_BIT 0 +/* + * R52 (0x34) - Line Mixer1 + */ +#define WM8991_LLOPGALON 0x0040 /* LLOPGALON */ +#define WM8991_LLOPGALON_BIT 6 +#define WM8991_LROPGALON 0x0020 /* LROPGALON */ +#define WM8991_LROPGALON_BIT 5 +#define WM8991_LOPLON 0x0010 /* LOPLON */ +#define WM8991_LOPLON_BIT 4 +#define WM8991_LR12LOP 0x0004 /* LR12LOP */ +#define WM8991_LR12LOP_BIT 2 +#define WM8991_LL12LOP 0x0002 /* LL12LOP */ +#define WM8991_LL12LOP_BIT 1 +#define WM8991_LLOPGALOP 0x0001 /* LLOPGALOP */ +#define WM8991_LLOPGALOP_BIT 0 +/* + * R53 (0x35) - Line Mixer2 + */ +#define WM8991_RROPGARON 0x0040 /* RROPGARON */ +#define WM8991_RROPGARON_BIT 6 +#define WM8991_RLOPGARON 0x0020 /* RLOPGARON */ +#define WM8991_RLOPGARON_BIT 5 +#define WM8991_ROPRON 0x0010 /* ROPRON */ +#define WM8991_ROPRON_BIT 4 +#define WM8991_RL12ROP 0x0004 /* RL12ROP */ +#define WM8991_RL12ROP_BIT 2 +#define WM8991_RR12ROP 0x0002 /* RR12ROP */ +#define WM8991_RR12ROP_BIT 1 +#define WM8991_RROPGAROP 0x0001 /* RROPGAROP */ +#define WM8991_RROPGAROP_BIT 0 + +/* + * R54 (0x36) - Speaker Mixer + */ +#define WM8991_LB2SPK 0x0080 /* LB2SPK */ +#define WM8991_LB2SPK_BIT 7 +#define WM8991_RB2SPK 0x0040 /* RB2SPK */ +#define WM8991_RB2SPK_BIT 6 +#define WM8991_LI2SPK 0x0020 /* LI2SPK */ +#define WM8991_LI2SPK_BIT 5 +#define WM8991_RI2SPK 0x0010 /* RI2SPK */ +#define WM8991_RI2SPK_BIT 4 +#define WM8991_LOPGASPK 0x0008 /* LOPGASPK */ +#define WM8991_LOPGASPK_BIT 3 +#define WM8991_ROPGASPK 0x0004 /* ROPGASPK */ +#define WM8991_ROPGASPK_BIT 2 +#define WM8991_LDSPK 0x0002 /* LDSPK */ +#define WM8991_LDSPK_BIT 1 +#define WM8991_RDSPK 0x0001 /* RDSPK */ +#define WM8991_RDSPK_BIT 0 + +/* + * R55 (0x37) - Additional Control + */ +#define WM8991_VROI 0x0001 /* VROI */ + +/* + * R56 (0x38) - AntiPOP1 + */ +#define WM8991_DIS_LLINE 0x0020 /* DIS_LLINE */ +#define WM8991_DIS_RLINE 0x0010 /* DIS_RLINE */ +#define WM8991_DIS_OUT3 0x0008 /* DIS_OUT3 */ +#define WM8991_DIS_OUT4 0x0004 /* DIS_OUT4 */ +#define WM8991_DIS_LOUT 0x0002 /* DIS_LOUT */ +#define WM8991_DIS_ROUT 0x0001 /* DIS_ROUT */ + +/* + * R57 (0x39) - AntiPOP2 + */ +#define WM8991_SOFTST 0x0040 /* SOFTST */ +#define WM8991_BUFIOEN 0x0008 /* BUFIOEN */ +#define WM8991_BUFDCOPEN 0x0004 /* BUFDCOPEN */ +#define WM8991_POBCTRL 0x0002 /* POBCTRL */ +#define WM8991_VMIDTOG 0x0001 /* VMIDTOG */ + +/* + * R58 (0x3A) - MICBIAS + */ +#define WM8991_MCDSCTH_MASK 0x00C0 /* MCDSCTH - [7:6] */ +#define WM8991_MCDTHR_MASK 0x0038 /* MCDTHR - [5:3] */ +#define WM8991_MCD 0x0004 /* MCD */ +#define WM8991_MBSEL 0x0001 /* MBSEL */ + +/* + * R60 (0x3C) - PLL1 + */ +#define WM8991_SDM 0x0080 /* SDM */ +#define WM8991_PRESCALE 0x0040 /* PRESCALE */ +#define WM8991_PLLN_MASK 0x000F /* PLLN - [3:0] */ + +/* + * R61 (0x3D) - PLL2 + */ +#define WM8991_PLLK1_MASK 0x00FF /* PLLK1 - [7:0] */ + +/* + * R62 (0x3E) - PLL3 + */ +#define WM8991_PLLK2_MASK 0x00FF /* PLLK2 - [7:0] */ + +/* + * R63 (0x3F) - Internal Driver Bits + */ +#define WM8991_INMIXL_PWR_BIT 0 +#define WM8991_AINLMUX_PWR_BIT 1 +#define WM8991_INMIXR_PWR_BIT 2 +#define WM8991_AINRMUX_PWR_BIT 3 + +#define WM8991_MCLK_DIV 0 +#define WM8991_DACCLK_DIV 1 +#define WM8991_ADCCLK_DIV 2 +#define WM8991_BCLK_DIV 3 + +#define SOC_WM899X_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert,\ + tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_get_volsw, .put = wm899x_outpga_put_volsw_vu, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + +#endif /* _WM8991_H */ -- cgit v1.2.3 From a1b3b5eeeebac8acfa7838ef90f5a00a6f9188a0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 24 Dec 2010 16:59:30 +0000 Subject: ASoC: Avoid direct register cache access when setting defaults Directly accessing the register cache means that we can't use anything except a flat register cache so use snd_soc_update_bits(). Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8523.c | 6 +++--- sound/soc/codecs/wm8741.c | 13 ++++++++----- sound/soc/codecs/wm8904.c | 41 ++++++++++++++++++++++++++--------------- sound/soc/codecs/wm8955.c | 27 +++++++++++++++++++-------- sound/soc/codecs/wm8962.c | 30 ++++++++++++++++++++---------- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm9090.c | 41 ++++++++++++++++++++++++----------------- 7 files changed, 101 insertions(+), 59 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 83e86f077ee..4fd4d8dca0f 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -414,7 +414,6 @@ static int wm8523_resume(struct snd_soc_codec *codec) static int wm8523_probe(struct snd_soc_codec *codec) { struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); - u16 *reg_cache = codec->reg_cache; int ret, i; codec->hw_write = (hw_write_t)i2c_master_send; @@ -471,8 +470,9 @@ static int wm8523_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU and enable ZC */ - reg_cache[WM8523_DAC_GAINR] |= WM8523_DACR_VU; - reg_cache[WM8523_DAC_CTRL3] |= WM8523_ZC; + snd_soc_update_bits(codec, WM8523_DAC_GAINR, + WM8523_DACR_VU, WM8523_DACR_VU); + snd_soc_update_bits(codec, WM8523_DAC_CTRL3, WM8523_ZC, WM8523_ZC); wm8523_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 494f2d31d75..25af901fe81 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -421,7 +421,6 @@ static int wm8741_resume(struct snd_soc_codec *codec) static int wm8741_probe(struct snd_soc_codec *codec) { struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); - u16 *reg_cache = codec->reg_cache; int ret = 0; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8741->control_type); @@ -437,10 +436,14 @@ static int wm8741_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU */ - reg_cache[WM8741_DACLLSB_ATTENUATION] |= WM8741_UPDATELL; - reg_cache[WM8741_DACLMSB_ATTENUATION] |= WM8741_UPDATELM; - reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERL; - reg_cache[WM8741_DACRLSB_ATTENUATION] |= WM8741_UPDATERM; + snd_soc_update_bits(codec, WM8741_DACLLSB_ATTENUATION, + WM8741_UPDATELL, WM8741_UPDATELL); + snd_soc_update_bits(codec, WM8741_DACLMSB_ATTENUATION, + WM8741_UPDATELM, WM8741_UPDATELM); + snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, + WM8741_UPDATERL, WM8741_UPDATERL); + snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION, + WM8741_UPDATERM, WM8741_UPDATERM); snd_soc_add_controls(codec, wm8741_snd_controls, ARRAY_SIZE(wm8741_snd_controls)); diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 17a8fe9b39b..443ae580445 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2436,19 +2436,28 @@ static int wm8904_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU and enable ZC */ - reg_cache[WM8904_ADC_DIGITAL_VOLUME_LEFT] |= WM8904_ADC_VU; - reg_cache[WM8904_ADC_DIGITAL_VOLUME_RIGHT] |= WM8904_ADC_VU; - reg_cache[WM8904_DAC_DIGITAL_VOLUME_LEFT] |= WM8904_DAC_VU; - reg_cache[WM8904_DAC_DIGITAL_VOLUME_RIGHT] |= WM8904_DAC_VU; - reg_cache[WM8904_ANALOGUE_OUT1_LEFT] |= WM8904_HPOUT_VU | - WM8904_HPOUTLZC; - reg_cache[WM8904_ANALOGUE_OUT1_RIGHT] |= WM8904_HPOUT_VU | - WM8904_HPOUTRZC; - reg_cache[WM8904_ANALOGUE_OUT2_LEFT] |= WM8904_LINEOUT_VU | - WM8904_LINEOUTLZC; - reg_cache[WM8904_ANALOGUE_OUT2_RIGHT] |= WM8904_LINEOUT_VU | - WM8904_LINEOUTRZC; - reg_cache[WM8904_CLOCK_RATES_0] &= ~WM8904_SR_MODE; + snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT, + WM8904_ADC_VU, WM8904_ADC_VU); + snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_RIGHT, + WM8904_ADC_VU, WM8904_ADC_VU); + snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_LEFT, + WM8904_DAC_VU, WM8904_DAC_VU); + snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_RIGHT, + WM8904_DAC_VU, WM8904_DAC_VU); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_LEFT, + WM8904_HPOUT_VU | WM8904_HPOUTLZC, + WM8904_HPOUT_VU | WM8904_HPOUTLZC); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_RIGHT, + WM8904_HPOUT_VU | WM8904_HPOUTRZC, + WM8904_HPOUT_VU | WM8904_HPOUTRZC); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_LEFT, + WM8904_LINEOUT_VU | WM8904_LINEOUTLZC, + WM8904_LINEOUT_VU | WM8904_LINEOUTLZC); + snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_RIGHT, + WM8904_LINEOUT_VU | WM8904_LINEOUTRZC, + WM8904_LINEOUT_VU | WM8904_LINEOUTRZC); + snd_soc_update_bits(codec, WM8904_CLOCK_RATES_0, + WM8904_SR_MODE, 0); /* Apply configuration from the platform data. */ if (wm8904->pdata) { @@ -2469,10 +2478,12 @@ static int wm8904_probe(struct snd_soc_codec *codec) /* Set Class W by default - this will be managed by the Class * G widget at runtime where bypass paths are available. */ - reg_cache[WM8904_CLASS_W_0] |= WM8904_CP_DYN_PWR; + snd_soc_update_bits(codec, WM8904_CLASS_W_0, + WM8904_CP_DYN_PWR, WM8904_CP_DYN_PWR); /* Use normal bias source */ - reg_cache[WM8904_BIAS_CONTROL_0] &= ~WM8904_POBCTRL; + snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, + WM8904_POBCTRL, 0); wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 7167dfc96aa..5e0214d6293 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -934,16 +934,27 @@ static int wm8955_probe(struct snd_soc_codec *codec) } /* Change some default settings - latch VU and enable ZC */ - reg_cache[WM8955_LEFT_DAC_VOLUME] |= WM8955_LDVU; - reg_cache[WM8955_RIGHT_DAC_VOLUME] |= WM8955_RDVU; - reg_cache[WM8955_LOUT1_VOLUME] |= WM8955_LO1VU | WM8955_LO1ZC; - reg_cache[WM8955_ROUT1_VOLUME] |= WM8955_RO1VU | WM8955_RO1ZC; - reg_cache[WM8955_LOUT2_VOLUME] |= WM8955_LO2VU | WM8955_LO2ZC; - reg_cache[WM8955_ROUT2_VOLUME] |= WM8955_RO2VU | WM8955_RO2ZC; - reg_cache[WM8955_MONOOUT_VOLUME] |= WM8955_MOZC; + snd_soc_update_bits(codec, WM8955_LEFT_DAC_VOLUME, + WM8955_LDVU, WM8955_LDVU); + snd_soc_update_bits(codec, WM8955_RIGHT_DAC_VOLUME, + WM8955_RDVU, WM8955_RDVU); + snd_soc_update_bits(codec, WM8955_LOUT1_VOLUME, + WM8955_LO1VU | WM8955_LO1ZC, + WM8955_LO1VU | WM8955_LO1ZC); + snd_soc_update_bits(codec, WM8955_ROUT1_VOLUME, + WM8955_RO1VU | WM8955_RO1ZC, + WM8955_RO1VU | WM8955_RO1ZC); + snd_soc_update_bits(codec, WM8955_LOUT2_VOLUME, + WM8955_LO2VU | WM8955_LO2ZC, + WM8955_LO2VU | WM8955_LO2ZC); + snd_soc_update_bits(codec, WM8955_ROUT2_VOLUME, + WM8955_RO2VU | WM8955_RO2ZC, + WM8955_RO2VU | WM8955_RO2ZC); + snd_soc_update_bits(codec, WM8955_MONOOUT_VOLUME, + WM8955_MOZC, WM8955_MOZC); /* Also enable adaptive bass boost by default */ - reg_cache[WM8955_BASS_CONTROL] |= WM8955_BB; + snd_soc_update_bits(codec, WM8955_BASS_CONTROL, WM8955_BB, WM8955_BB); /* Set platform data values */ if (pdata) { diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 7c02924bedd..5c7b730a864 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3822,16 +3822,26 @@ static int wm8962_probe(struct snd_soc_codec *codec) } /* Latch volume update bits */ - reg_cache[WM8962_LEFT_INPUT_VOLUME] |= WM8962_IN_VU; - reg_cache[WM8962_RIGHT_INPUT_VOLUME] |= WM8962_IN_VU; - reg_cache[WM8962_LEFT_ADC_VOLUME] |= WM8962_ADC_VU; - reg_cache[WM8962_RIGHT_ADC_VOLUME] |= WM8962_ADC_VU; - reg_cache[WM8962_LEFT_DAC_VOLUME] |= WM8962_DAC_VU; - reg_cache[WM8962_RIGHT_DAC_VOLUME] |= WM8962_DAC_VU; - reg_cache[WM8962_SPKOUTL_VOLUME] |= WM8962_SPKOUT_VU; - reg_cache[WM8962_SPKOUTR_VOLUME] |= WM8962_SPKOUT_VU; - reg_cache[WM8962_HPOUTL_VOLUME] |= WM8962_HPOUT_VU; - reg_cache[WM8962_HPOUTR_VOLUME] |= WM8962_HPOUT_VU; + snd_soc_update_bits(codec, WM8962_LEFT_INPUT_VOLUME, + WM8962_IN_VU, WM8962_IN_VU); + snd_soc_update_bits(codec, WM8962_RIGHT_INPUT_VOLUME, + WM8962_IN_VU, WM8962_IN_VU); + snd_soc_update_bits(codec, WM8962_LEFT_ADC_VOLUME, + WM8962_ADC_VU, WM8962_ADC_VU); + snd_soc_update_bits(codec, WM8962_RIGHT_ADC_VOLUME, + WM8962_ADC_VU, WM8962_ADC_VU); + snd_soc_update_bits(codec, WM8962_LEFT_DAC_VOLUME, + WM8962_DAC_VU, WM8962_DAC_VU); + snd_soc_update_bits(codec, WM8962_RIGHT_DAC_VOLUME, + WM8962_DAC_VU, WM8962_DAC_VU); + snd_soc_update_bits(codec, WM8962_SPKOUTL_VOLUME, + WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); + snd_soc_update_bits(codec, WM8962_SPKOUTR_VOLUME, + WM8962_SPKOUT_VU, WM8962_SPKOUT_VU); + snd_soc_update_bits(codec, WM8962_HPOUTL_VOLUME, + WM8962_HPOUT_VU, WM8962_HPOUT_VU); + snd_soc_update_bits(codec, WM8962_HPOUTR_VOLUME, + WM8962_HPOUT_VU, WM8962_HPOUT_VU); wm8962_add_widgets(codec); diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 4bbc3442703..30fb48ec279 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -965,7 +965,7 @@ static int wm8978_probe(struct snd_soc_codec *codec) * written. */ for (i = 0; i < ARRAY_SIZE(update_reg); i++) - ((u16 *)codec->reg_cache)[update_reg[i]] |= 0x100; + snd_soc_update_bits(codec, update_reg[i], 0x100, 0x100); /* Reset the codec */ ret = snd_soc_write(codec, WM8978_RESET, 0); diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index d40bfc9f880..4de12203e61 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -551,7 +551,6 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec, static int wm9090_probe(struct snd_soc_codec *codec) { struct wm9090_priv *wm9090 = snd_soc_codec_get_drvdata(codec); - u16 *reg_cache = codec->reg_cache; int ret; codec->control_data = wm9090->control_data; @@ -576,22 +575,30 @@ static int wm9090_probe(struct snd_soc_codec *codec) /* Configure some defaults; they will be written out when we * bring the bias up. */ - reg_cache[WM9090_IN1_LINE_INPUT_A_VOLUME] |= WM9090_IN1_VU - | WM9090_IN1A_ZC; - reg_cache[WM9090_IN1_LINE_INPUT_B_VOLUME] |= WM9090_IN1_VU - | WM9090_IN1B_ZC; - reg_cache[WM9090_IN2_LINE_INPUT_A_VOLUME] |= WM9090_IN2_VU - | WM9090_IN2A_ZC; - reg_cache[WM9090_IN2_LINE_INPUT_B_VOLUME] |= WM9090_IN2_VU - | WM9090_IN2B_ZC; - reg_cache[WM9090_SPEAKER_VOLUME_LEFT] |= - WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC; - reg_cache[WM9090_LEFT_OUTPUT_VOLUME] |= - WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC; - reg_cache[WM9090_RIGHT_OUTPUT_VOLUME] |= - WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC; - - reg_cache[WM9090_CLOCKING_1] |= WM9090_TOCLK_ENA; + snd_soc_update_bits(codec, WM9090_IN1_LINE_INPUT_A_VOLUME, + WM9090_IN1_VU | WM9090_IN1A_ZC, + WM9090_IN1_VU | WM9090_IN1A_ZC); + snd_soc_update_bits(codec, WM9090_IN1_LINE_INPUT_B_VOLUME, + WM9090_IN1_VU | WM9090_IN1B_ZC, + WM9090_IN1_VU | WM9090_IN1B_ZC); + snd_soc_update_bits(codec, WM9090_IN2_LINE_INPUT_A_VOLUME, + WM9090_IN2_VU | WM9090_IN2A_ZC, + WM9090_IN2_VU | WM9090_IN2A_ZC); + snd_soc_update_bits(codec, WM9090_IN2_LINE_INPUT_B_VOLUME, + WM9090_IN2_VU | WM9090_IN2B_ZC, + WM9090_IN2_VU | WM9090_IN2B_ZC); + snd_soc_update_bits(codec, WM9090_SPEAKER_VOLUME_LEFT, + WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC, + WM9090_SPKOUT_VU | WM9090_SPKOUTL_ZC); + snd_soc_update_bits(codec, WM9090_LEFT_OUTPUT_VOLUME, + WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC, + WM9090_HPOUT1_VU | WM9090_HPOUT1L_ZC); + snd_soc_update_bits(codec, WM9090_RIGHT_OUTPUT_VOLUME, + WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC, + WM9090_HPOUT1_VU | WM9090_HPOUT1R_ZC); + + snd_soc_update_bits(codec, WM9090_CLOCKING_1, + WM9090_TOCLK_ENA, WM9090_TOCLK_ENA); wm9090_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -- cgit v1.2.3 From 219d8df86805b8bb20b375707e9be734100ce89d Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Mon, 17 Jan 2011 11:00:12 +0000 Subject: ASoC: WM8995: Add regulator handling code Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 102 +++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 97 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index f0f678de489..7d563413df3 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include @@ -30,6 +31,18 @@ #include "wm8995.h" +#define WM8995_NUM_SUPPLIES 8 +static const char *wm8995_supply_names[WM8995_NUM_SUPPLIES] = { + "DCVDD", + "DBVDD1", + "DBVDD2", + "DBVDD3", + "AVDD1", + "AVDD2", + "CPVDD", + "MICVDD" +}; + static const u16 wm8995_reg_defs[WM8995_MAX_REGISTER + 1] = { [0] = 0x8995, [5] = 0x0100, [16] = 0x000b, [17] = 0x000b, [24] = 0x02c0, [25] = 0x02c0, [26] = 0x02c0, [27] = 0x02c0, @@ -126,8 +139,37 @@ struct wm8995_priv { int mclk[2]; int aifclk[2]; struct fll_config fll[2], fll_suspend[2]; + struct regulator_bulk_data supplies[WM8995_NUM_SUPPLIES]; + struct notifier_block disable_nb[WM8995_NUM_SUPPLIES]; + struct snd_soc_codec *codec; }; +/* + * We can't use the same notifier block for more than one supply and + * there's no way I can see to get from a callback to the caller + * except container_of(). + */ +#define WM8995_REGULATOR_EVENT(n) \ +static int wm8995_regulator_event_##n(struct notifier_block *nb, \ + unsigned long event, void *data) \ +{ \ + struct wm8995_priv *wm8995 = container_of(nb, struct wm8995_priv, \ + disable_nb[n]); \ + if (event & REGULATOR_EVENT_DISABLE) { \ + wm8995->codec->cache_sync = 1; \ + } \ + return 0; \ +} + +WM8995_REGULATOR_EVENT(0) +WM8995_REGULATOR_EVENT(1) +WM8995_REGULATOR_EVENT(2) +WM8995_REGULATOR_EVENT(3) +WM8995_REGULATOR_EVENT(4) +WM8995_REGULATOR_EVENT(5) +WM8995_REGULATOR_EVENT(6) +WM8995_REGULATOR_EVENT(7) + static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); static const DECLARE_TLV_DB_SCALE(in1lr_pga_tlv, -1650, 150, 0); static const DECLARE_TLV_DB_SCALE(in1l_boost_tlv, 0, 600, 0); @@ -1483,6 +1525,11 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(wm8995->supplies), + wm8995->supplies); + if (ret) + return ret; + ret = snd_soc_cache_sync(codec); if (ret) { dev_err(codec->dev, @@ -1492,13 +1539,13 @@ static int wm8995_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, WM8995_POWER_MANAGEMENT_1, WM8995_BG_ENA_MASK, WM8995_BG_ENA); - } break; case SND_SOC_BIAS_OFF: snd_soc_update_bits(codec, WM8995_POWER_MANAGEMENT_1, WM8995_BG_ENA_MASK, 0); - codec->cache_sync = 1; + regulator_bulk_disable(ARRAY_SIZE(wm8995->supplies), + wm8995->supplies); break; } @@ -1537,10 +1584,12 @@ static int wm8995_remove(struct snd_soc_codec *codec) static int wm8995_probe(struct snd_soc_codec *codec) { struct wm8995_priv *wm8995; + int i; int ret; codec->dapm.idle_bias_off = 1; wm8995 = snd_soc_codec_get_drvdata(codec); + wm8995->codec = codec; ret = snd_soc_codec_set_cache_io(codec, 16, 16, wm8995->control_type); if (ret < 0) { @@ -1548,21 +1597,58 @@ static int wm8995_probe(struct snd_soc_codec *codec) return ret; } + for (i = 0; i < ARRAY_SIZE(wm8995->supplies); i++) + wm8995->supplies[i].supply = wm8995_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8995->supplies), + wm8995->supplies); + if (ret) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + wm8995->disable_nb[0].notifier_call = wm8995_regulator_event_0; + wm8995->disable_nb[1].notifier_call = wm8995_regulator_event_1; + wm8995->disable_nb[2].notifier_call = wm8995_regulator_event_2; + wm8995->disable_nb[3].notifier_call = wm8995_regulator_event_3; + wm8995->disable_nb[4].notifier_call = wm8995_regulator_event_4; + wm8995->disable_nb[5].notifier_call = wm8995_regulator_event_5; + wm8995->disable_nb[6].notifier_call = wm8995_regulator_event_6; + wm8995->disable_nb[7].notifier_call = wm8995_regulator_event_7; + + /* This should really be moved into the regulator core */ + for (i = 0; i < ARRAY_SIZE(wm8995->supplies); i++) { + ret = regulator_register_notifier(wm8995->supplies[i].consumer, + &wm8995->disable_nb[i]); + if (ret) { + dev_err(codec->dev, + "Failed to register regulator notifier: %d\n", + ret); + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8995->supplies), + wm8995->supplies); + if (ret) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_reg_get; + } + ret = snd_soc_read(codec, WM8995_SOFTWARE_RESET); if (ret < 0) { dev_err(codec->dev, "Failed to read device ID: %d\n", ret); - return ret; + goto err_reg_enable; } if (ret != 0x8995) { dev_err(codec->dev, "Invalid device ID: %#x\n", ret); - return -EINVAL; + goto err_reg_enable; } ret = snd_soc_write(codec, WM8995_SOFTWARE_RESET, 0); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); - return ret; + goto err_reg_enable; } wm8995_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1597,6 +1683,12 @@ static int wm8995_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm8995_intercon)); return 0; + +err_reg_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8995->supplies), wm8995->supplies); +err_reg_get: + regulator_bulk_free(ARRAY_SIZE(wm8995->supplies), wm8995->supplies); + return ret; } #define WM8995_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ -- cgit v1.2.3 From 0d00a85752ab0fb4f0f7758e35d54b931a25d812 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 18 Jan 2011 15:51:45 +0200 Subject: ASoC: core: Remove dapm_sync call from soc_post_component_init snd_soc_dapm_new_widgets will call dapm_power_widgets at the end, so there is no need to call snd_soc_dapm_sync after snd_soc_dapm_new_widgets. Signed-off-by: Peter Ujfalusi Reviewed-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 69117b686fd..9e68984423b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1470,7 +1470,6 @@ static int soc_post_component_init(struct snd_soc_card *card, /* Make sure all DAPM widgets are instantiated */ snd_soc_dapm_new_widgets(&codec->dapm); - snd_soc_dapm_sync(&codec->dapm); /* register the rtd device */ rtd->codec = codec; -- cgit v1.2.3 From 065ae6784f03016ed1c0613863461a162388cf5a Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 18 Jan 2011 19:11:48 +0530 Subject: ASoC: Add dependency on INTEL_SCU_IPC for Intel MID drivers Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/mid-x86/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/mid-x86/Kconfig b/sound/soc/mid-x86/Kconfig index e0123a40822..1ad75383635 100644 --- a/sound/soc/mid-x86/Kconfig +++ b/sound/soc/mid-x86/Kconfig @@ -1,5 +1,6 @@ config SND_MFLD_MACHINE tristate "SOC Machine Audio driver for Intel Medfield MID platform" + depends on INTEL_SCU_IPC select SND_SOC_SN95031 select SND_SST_PLATFORM help -- cgit v1.2.3 From a211701eb1e4a41bbee22d61f35dba8e55925db4 Mon Sep 17 00:00:00 2001 From: Harsha Priya Date: Tue, 11 Jan 2011 14:50:59 +0530 Subject: ASoC: sst_platform porting sst dsp driver interface as per latest in Greg's staging tree The interface between sst platform driver and intel sst dsp driver have been changed in Greg's staging tree - next branch This patch adds the interface changes compatible with the new interface in Greg's staging tree Signed-off-by: Harsha Priya Signed-off-by: Vinod Koul Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/mid-x86/sst_platform.c | 49 +++++++++++++++++----------------------- 1 file changed, 21 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index a4e3fa3761f..1d1f5447b33 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -168,7 +168,7 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) pr_debug("Capture stream,Device %d\n", substream->pcm->device); } - ret_val = stream->sstdrv_ops->control_set(SST_SND_ALLOC, &str_params); + ret_val = stream->sstdrv_ops->pcm_control->open(&str_params); pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val); if (ret_val < 0) return ret_val; @@ -207,8 +207,8 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) stream->stream_info.mad_substream = substream; stream->stream_info.buffer_ptr = 0; stream->stream_info.sfreq = substream->runtime->rate; - ret_val = stream->sstdrv_ops->control_set(SST_SND_STREAM_INIT, - &stream->stream_info); + ret_val = stream->sstdrv_ops->pcm_control->device_control( + SST_SND_STREAM_INIT, &stream->stream_info); if (ret_val) pr_err("control_set ret error %d\n", ret_val); return ret_val; @@ -261,8 +261,7 @@ static int sst_platform_close(struct snd_pcm_substream *substream) stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (str_id) - ret_val = stream->sstdrv_ops->control_set( - SST_SND_FREE, &str_id); + ret_val = stream->sstdrv_ops->pcm_control->close(str_id); kfree(stream->sstdrv_ops); kfree(stream); return ret_val; @@ -277,7 +276,7 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (stream->stream_info.str_id) { - ret_val = stream->sstdrv_ops->control_set( + ret_val = stream->sstdrv_ops->pcm_control->device_control( SST_SND_DROP, &str_id); return ret_val; } @@ -300,6 +299,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, { int ret_val = 0, str_id; struct sst_runtime_stream *stream; + int str_cmd, status; pr_debug("sst_platform_pcm_trigger called\n"); stream = substream->runtime->private_data; @@ -307,40 +307,33 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: pr_debug("sst: Trigger Start\n"); - ret_val = stream->sstdrv_ops->control_set( - SST_SND_START, &str_id); - if (ret_val) - break; - sst_set_stream_status(stream, SST_PLATFORM_RUNNING); + str_cmd = SST_SND_START; + status = SST_PLATFORM_RUNNING; stream->stream_info.mad_substream = substream; break; case SNDRV_PCM_TRIGGER_STOP: pr_debug("sst: in stop\n"); - ret_val = stream->sstdrv_ops->control_set( - SST_SND_DROP, &str_id); - if (ret_val) - break; - sst_set_stream_status(stream, SST_PLATFORM_DROPPED); + str_cmd = SST_SND_DROP; + status = SST_PLATFORM_DROPPED; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: pr_debug("sst: in pause\n"); - ret_val = stream->sstdrv_ops->control_set( - SST_SND_PAUSE, &str_id); - if (ret_val) - break; - sst_set_stream_status(stream, SST_PLATFORM_PAUSED); + str_cmd = SST_SND_PAUSE; + status = SST_PLATFORM_PAUSED; break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: pr_debug("sst: in pause release\n"); - ret_val = stream->sstdrv_ops->control_set( - SST_SND_RESUME, &str_id); - if (ret_val) - break; - sst_set_stream_status(stream, SST_PLATFORM_RUNNING); + str_cmd = SST_SND_RESUME; + status = SST_PLATFORM_RUNNING; break; default: - ret_val = -EINVAL; + return -EINVAL; } + ret_val = stream->sstdrv_ops->pcm_control->device_control(str_cmd, + &str_id); + if (!ret_val) + sst_set_stream_status(stream, status); + return ret_val; } @@ -357,7 +350,7 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer if (status == SST_PLATFORM_INIT) return 0; str_info = &stream->stream_info; - ret_val = stream->sstdrv_ops->control_set( + ret_val = stream->sstdrv_ops->pcm_control->device_control( SST_SND_BUFFER_POINTER, str_info); if (ret_val) { pr_err("sst: error code = %d\n", ret_val); -- cgit v1.2.3 From 828a842f2e02de5d884ee14bd3c21ddbc77ec60e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 15 Jan 2011 13:14:30 +0000 Subject: ASoC: Explicitly say if we're powering up or down Rather than passing the sequence to use for DAPM widgets around by reference explicitly say if we're powering up or down until the point where we need the sequence itself. This should make no practical difference in itself but supports future refactoring. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 37 +++++++++++++++++++++++++------------ 1 file changed, 25 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 499730ab563..57e1c9f7114 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -726,8 +726,15 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) static int dapm_seq_compare(struct snd_soc_dapm_widget *a, struct snd_soc_dapm_widget *b, - int sort[]) + bool power_up) { + int *sort; + + if (power_up) + sort = dapm_up_seq; + else + sort = dapm_down_seq; + if (sort[a->id] != sort[b->id]) return sort[a->id] - sort[b->id]; if (a->reg != b->reg) @@ -741,12 +748,12 @@ static int dapm_seq_compare(struct snd_soc_dapm_widget *a, /* Insert a widget in order into a DAPM power sequence. */ static void dapm_seq_insert(struct snd_soc_dapm_widget *new_widget, struct list_head *list, - int sort[]) + bool power_up) { struct snd_soc_dapm_widget *w; list_for_each_entry(w, list, power_list) - if (dapm_seq_compare(new_widget, w, sort) < 0) { + if (dapm_seq_compare(new_widget, w, power_up) < 0) { list_add_tail(&new_widget->power_list, &w->power_list); return; } @@ -857,7 +864,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, * handled. */ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, - struct list_head *list, int event, int sort[]) + struct list_head *list, int event, bool power_up) { struct snd_soc_dapm_widget *w, *n; LIST_HEAD(pending); @@ -865,6 +872,12 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, int cur_reg = SND_SOC_NOPM; struct snd_soc_dapm_context *cur_dapm = NULL; int ret; + int *sort; + + if (power_up) + sort = dapm_up_seq; + else + sort = dapm_down_seq; list_for_each_entry_safe(w, n, list, power_list) { ret = 0; @@ -1002,10 +1015,10 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) list_for_each_entry(w, &card->widgets, list) { switch (w->id) { case snd_soc_dapm_pre: - dapm_seq_insert(w, &down_list, dapm_down_seq); + dapm_seq_insert(w, &down_list, false); break; case snd_soc_dapm_post: - dapm_seq_insert(w, &up_list, dapm_up_seq); + dapm_seq_insert(w, &up_list, true); break; default: @@ -1025,9 +1038,9 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) trace_snd_soc_dapm_widget_power(w, power); if (power) - dapm_seq_insert(w, &up_list, dapm_up_seq); + dapm_seq_insert(w, &up_list, true); else - dapm_seq_insert(w, &down_list, dapm_down_seq); + dapm_seq_insert(w, &down_list, false); w->power = power; break; @@ -1086,12 +1099,12 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } /* Power down widgets first; try to avoid amplifying pops. */ - dapm_seq_run(dapm, &down_list, event, dapm_down_seq); + dapm_seq_run(dapm, &down_list, event, false); dapm_widget_update(dapm); /* Now power up. */ - dapm_seq_run(dapm, &up_list, event, dapm_up_seq); + dapm_seq_run(dapm, &up_list, event, true); list_for_each_entry(d, &dapm->card->dapm_list, list) { /* If we just powered the last thing off drop to standby bias */ @@ -2372,7 +2385,7 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) if (w->dapm != dapm) continue; if (w->power) { - dapm_seq_insert(w, &down_list, dapm_down_seq); + dapm_seq_insert(w, &down_list, false); w->power = 0; powerdown = 1; } @@ -2383,7 +2396,7 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) */ if (powerdown) { snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_PREPARE); - dapm_seq_run(dapm, &down_list, 0, dapm_down_seq); + dapm_seq_run(dapm, &down_list, 0, false); snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_STANDBY); } } -- cgit v1.2.3 From 20e4859dedfc7e7b620d1756b29f8483c5be5fcc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 15 Jan 2011 13:40:50 +0000 Subject: ASoC: Add support for sequencing within With larger devices there may be many widgets of the same type in series in an audio path. Allow drivers to specify an additional level of ordering within each widget type by adding a subsequence number to widgets and then splitting operations on widgets so that widgets of the same type but different sequence numbers are processed separately. A typical example would be a supply widget which requires that another widget be enabled to provide power or clocking. SND_SOC_DAPM_PGA_S() and SND_SOC_DAPM_SUPPLY_S() macros are provided allowing this to be used with PGAs and supplies as these are the most commonly affected widgets. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 57e1c9f7114..eb7436c7aca 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -737,6 +737,12 @@ static int dapm_seq_compare(struct snd_soc_dapm_widget *a, if (sort[a->id] != sort[b->id]) return sort[a->id] - sort[b->id]; + if (a->subseq != b->subseq) { + if (power_up) + return a->subseq - b->subseq; + else + return b->subseq - a->subseq; + } if (a->reg != b->reg) return a->reg - b->reg; if (a->dapm != b->dapm) @@ -869,6 +875,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_widget *w, *n; LIST_HEAD(pending); int cur_sort = -1; + int cur_subseq = -1; int cur_reg = SND_SOC_NOPM; struct snd_soc_dapm_context *cur_dapm = NULL; int ret; @@ -884,12 +891,13 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, /* Do we need to apply any queued changes? */ if (sort[w->id] != cur_sort || w->reg != cur_reg || - w->dapm != cur_dapm) { + w->dapm != cur_dapm || w->subseq != cur_subseq) { if (!list_empty(&pending)) dapm_seq_run_coalesced(cur_dapm, &pending); INIT_LIST_HEAD(&pending); cur_sort = -1; + cur_subseq = -1; cur_reg = SND_SOC_NOPM; cur_dapm = NULL; } @@ -934,6 +942,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, default: /* Queue it up for application */ cur_sort = sort[w->id]; + cur_subseq = w->subseq; cur_reg = w->reg; cur_dapm = w->dapm; list_move(&w->power_list, &pending); -- cgit v1.2.3 From 474b62d6eee733abdcd36f8e3e5ce504fbb9110b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 18 Jan 2011 16:14:44 +0000 Subject: ASoC: Provide per widget type callback when executing DAPM sequences Many modern devices have features such as DC servos which take time to start. Currently these are handled by per-widget events but this makes it difficult to paralleise operations on multiple widgets, meaning delays can end up being needlessly serialised. By providing a callback to drivers when all widgets of a given type have been handled during a DAPM sequence the core allows drivers to start operations separately and wait for them to complete much more simply. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 16 +++++++++++++++- 2 files changed, 16 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9e68984423b..b0e7689159c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3496,6 +3496,7 @@ int snd_soc_register_codec(struct device *dev, codec->dapm.bias_level = SND_SOC_BIAS_OFF; codec->dapm.dev = dev; codec->dapm.codec = codec; + codec->dapm.seq_notifier = codec_drv->seq_notifier; codec->dev = dev; codec->driver = codec_drv; codec->num_dai = num_dai; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index eb7436c7aca..37b376f4c75 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -878,7 +878,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, int cur_subseq = -1; int cur_reg = SND_SOC_NOPM; struct snd_soc_dapm_context *cur_dapm = NULL; - int ret; + int ret, i; int *sort; if (power_up) @@ -895,6 +895,13 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, if (!list_empty(&pending)) dapm_seq_run_coalesced(cur_dapm, &pending); + if (cur_dapm && cur_dapm->seq_notifier) { + for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) + if (sort[i] == cur_sort) + cur_dapm->seq_notifier(cur_dapm, + i); + } + INIT_LIST_HEAD(&pending); cur_sort = -1; cur_subseq = -1; @@ -956,6 +963,13 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, if (!list_empty(&pending)) dapm_seq_run_coalesced(dapm, &pending); + + if (cur_dapm && cur_dapm->seq_notifier) { + for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) + if (sort[i] == cur_sort) + cur_dapm->seq_notifier(cur_dapm, + i); + } } static void dapm_widget_update(struct snd_soc_dapm_context *dapm) -- cgit v1.2.3 From dad8e7aeeb83a26d267e757e4c1cf69591850477 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Wed, 19 Jan 2011 14:53:36 +0000 Subject: ASoC: soc-cache: Introduce the cache_bypass option This is primarily needed to avoid writing back to the cache whenever we are syncing the cache with the hardware. This gives a performance benefit especially for large register maps. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 36 ++++++++++++++++++++++++------------ 1 file changed, 24 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index d97a59f6a24..1ebff9f12b4 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -25,7 +25,8 @@ static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -49,7 +50,8 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = value & 0x00ff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -106,7 +108,8 @@ static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -130,7 +133,8 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = value & 0x00ff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -191,7 +195,8 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, data[1] = value & 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -216,7 +221,8 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, reg &= 0xff; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -271,7 +277,8 @@ static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, data[2] = value & 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -295,7 +302,8 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -450,7 +458,8 @@ static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, reg &= 0xff; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -476,7 +485,8 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, reg &= 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; @@ -568,7 +578,8 @@ static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec, unsigned int val; if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg)) { + snd_soc_codec_volatile_register(codec, reg) || + codec->cache_bypass) { if (codec->cache_only) return -1; @@ -595,7 +606,8 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, data[3] = value & 0xff; if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size) { + reg < codec->driver->reg_cache_size && + !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) return -1; -- cgit v1.2.3 From 9978007befa5efdc16dc940631c594f420f96b09 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Wed, 19 Jan 2011 14:53:37 +0000 Subject: ASoC: soc-cache: Apply the cache_bypass option Incorporate the use of the cache_bypass functionality in the syncing functions. The snd_soc_flat_cache_sync() need not be hooked as there is no performance benefit from using the cache_bypass option. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 1ebff9f12b4..f8348396379 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -890,7 +890,9 @@ static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec) ret = snd_soc_cache_read(codec, rbnode->reg, &val); if (ret) return ret; + codec->cache_bypass = 1; ret = snd_soc_write(codec, rbnode->reg, val); + codec->cache_bypass = 0; if (ret) return ret; dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", @@ -1148,7 +1150,9 @@ static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec) ret = snd_soc_cache_read(codec, i, &val); if (ret) return ret; + codec->cache_bypass = 1; ret = snd_soc_write(codec, i, val); + codec->cache_bypass = 0; if (ret) return ret; dev_dbg(codec->dev, "Synced register %#x, value = %#x\n", -- cgit v1.2.3 From 64ed98365062b748770b340fd27ae34564a5a322 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 20 Jan 2011 21:43:44 +0000 Subject: ASoC: Staticise twl6040_hs_jack_report() It's an internal function so shouldn't be exported (as sparse points out). Signed-off-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/twl6040.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 4bbf1b15a49..482fcdb59bf 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -724,8 +724,8 @@ static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, return 0; } -void twl6040_hs_jack_report(struct snd_soc_codec *codec, - struct snd_soc_jack *jack, int report) +static void twl6040_hs_jack_report(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, int report) { struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); int status; -- cgit v1.2.3 From 7cfe56172ac14d2031f1896ecb629033f71caafa Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 20 Jan 2011 13:52:08 -0700 Subject: ASoC: wm8903: Expose GPIOs through gpiolib Also, update platform_data GPIO handling to have an explicit "don't touch this pin" option. Add #defines for the GPIO pin functions. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 126 +++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 125 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index a2a446cb180..9c4f2c4febc 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2,6 +2,7 @@ * wm8903.c -- WM8903 ALSA SoC Audio driver * * Copyright 2008 Wolfson Microelectronics + * Copyright 2011 NVIDIA, Inc. * * Author: Mark Brown * @@ -19,6 +20,7 @@ #include #include #include +#include #include #include #include @@ -213,6 +215,7 @@ static u16 wm8903_reg_defaults[] = { }; struct wm8903_priv { + struct snd_soc_codec *codec; int sysclk; int irq; @@ -230,6 +233,10 @@ struct wm8903_priv { int mic_short; int mic_last_report; int mic_delay; + +#ifdef CONFIG_GPIOLIB + struct gpio_chip gpio_chip; +#endif }; static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int reg) @@ -1635,6 +1642,119 @@ static int wm8903_resume(struct snd_soc_codec *codec) return 0; } +#ifdef CONFIG_GPIOLIB +static inline struct wm8903_priv *gpio_to_wm8903(struct gpio_chip *chip) +{ + return container_of(chip, struct wm8903_priv, gpio_chip); +} + +static int wm8903_gpio_request(struct gpio_chip *chip, unsigned offset) +{ + if (offset >= WM8903_NUM_GPIO) + return -EINVAL; + + return 0; +} + +static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) +{ + struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct snd_soc_codec *codec = wm8903->codec; + unsigned int mask, val; + + mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK; + val = (WM8903_GPn_FN_GPIO_INPUT << WM8903_GP1_FN_SHIFT) | + WM8903_GP1_DIR; + + return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, + mask, val); +} + +static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset) +{ + struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct snd_soc_codec *codec = wm8903->codec; + int reg; + + reg = snd_soc_read(codec, WM8903_GPIO_CONTROL_1 + offset); + + return (reg & WM8903_GP1_LVL_MASK) >> WM8903_GP1_LVL_SHIFT; +} + +static int wm8903_gpio_direction_out(struct gpio_chip *chip, + unsigned offset, int value) +{ + struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct snd_soc_codec *codec = wm8903->codec; + unsigned int mask, val; + + mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK | WM8903_GP1_LVL_MASK; + val = (WM8903_GPn_FN_GPIO_OUTPUT << WM8903_GP1_FN_SHIFT) | + (value << WM8903_GP2_LVL_SHIFT); + + return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, + mask, val); +} + +static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value) +{ + struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); + struct snd_soc_codec *codec = wm8903->codec; + + snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, + WM8903_GP1_LVL_MASK, value << WM8903_GP1_LVL_SHIFT); +} + +static struct gpio_chip wm8903_template_chip = { + .label = "wm8903", + .owner = THIS_MODULE, + .request = wm8903_gpio_request, + .direction_input = wm8903_gpio_direction_in, + .get = wm8903_gpio_get, + .direction_output = wm8903_gpio_direction_out, + .set = wm8903_gpio_set, + .can_sleep = 1, +}; + +static void wm8903_init_gpio(struct snd_soc_codec *codec) +{ + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + struct wm8903_platform_data *pdata = dev_get_platdata(codec->dev); + int ret; + + wm8903->gpio_chip = wm8903_template_chip; + wm8903->gpio_chip.ngpio = WM8903_NUM_GPIO; + wm8903->gpio_chip.dev = codec->dev; + + if (pdata && pdata->gpio_base) + wm8903->gpio_chip.base = pdata->gpio_base; + else + wm8903->gpio_chip.base = -1; + + ret = gpiochip_add(&wm8903->gpio_chip); + if (ret != 0) + dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); +} + +static void wm8903_free_gpio(struct snd_soc_codec *codec) +{ + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = gpiochip_remove(&wm8903->gpio_chip); + if (ret != 0) + dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret); +} +#else +static void wm8903_init_gpio(struct snd_soc_codec *codec) +{ +} + +static void wm8903_free_gpio(struct snd_soc_codec *codec) +{ +} +#endif + static int wm8903_probe(struct snd_soc_codec *codec) { struct wm8903_platform_data *pdata = dev_get_platdata(codec->dev); @@ -1643,6 +1763,7 @@ static int wm8903_probe(struct snd_soc_codec *codec) int trigger, irq_pol; u16 val; + wm8903->codec = codec; init_completion(&wm8903->wseq); ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); @@ -1667,7 +1788,7 @@ static int wm8903_probe(struct snd_soc_codec *codec) /* Set up GPIOs and microphone detection */ if (pdata) { for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { - if (!pdata->gpio_cfg[i]) + if (pdata->gpio_cfg[i] == WM8903_GPIO_NO_CONFIG) continue; snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i, @@ -1749,12 +1870,15 @@ static int wm8903_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm8903_snd_controls)); wm8903_add_widgets(codec); + wm8903_init_gpio(codec); + return ret; } /* power down chip */ static int wm8903_remove(struct snd_soc_codec *codec) { + wm8903_free_gpio(codec); wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } -- cgit v1.2.3 From 67b22517d8e48a97e1d2ab10d095c538bbb2374c Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin Date: Wed, 19 Jan 2011 21:22:06 +0300 Subject: ASoC: CS4271 codec support Added support for CS4271 codec to ASoC. Signed-off-by: Alexander Sverdlin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4271.c | 630 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 636 insertions(+) create mode 100644 sound/soc/codecs/cs4271.c (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a9cb2a04ad5..e239345a4d5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C + select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_JZ4740_CODEC if SOC_JZ4740 @@ -157,6 +158,9 @@ config SND_SOC_CS4270_VD33_ERRATA bool depends on SND_SOC_CS4270 +config SND_SOC_CS4271 + tristate + config SND_SOC_CX20442 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 68e76af894b..83b7accd703 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -12,6 +12,7 @@ snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs4270-objs := cs4270.o +snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-dmic-objs := dmic.o @@ -93,6 +94,7 @@ obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o +obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c new file mode 100644 index 00000000000..237ece3f104 --- /dev/null +++ b/sound/soc/codecs/cs4271.c @@ -0,0 +1,630 @@ +/* + * CS4271 ASoC codec driver + * + * Copyright (c) 2010 Alexander Sverdlin + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * This driver support CS4271 codec being master or slave, working + * in control port mode, connected either via SPI or I2C. + * The data format accepted is I2S or left-justified. + * DAPM support not implemented. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* + * CS4271 registers + * High byte represents SPI chip address (0x10) + write command (0) + * Low byte - codec register address + */ +#define CS4271_MODE1 0x2001 /* Mode Control 1 */ +#define CS4271_DACCTL 0x2002 /* DAC Control */ +#define CS4271_DACVOL 0x2003 /* DAC Volume & Mixing Control */ +#define CS4271_VOLA 0x2004 /* DAC Channel A Volume Control */ +#define CS4271_VOLB 0x2005 /* DAC Channel B Volume Control */ +#define CS4271_ADCCTL 0x2006 /* ADC Control */ +#define CS4271_MODE2 0x2007 /* Mode Control 2 */ +#define CS4271_CHIPID 0x2008 /* Chip ID */ + +#define CS4271_FIRSTREG CS4271_MODE1 +#define CS4271_LASTREG CS4271_MODE2 +#define CS4271_NR_REGS ((CS4271_LASTREG & 0xFF) + 1) + +/* Bit masks for the CS4271 registers */ +#define CS4271_MODE1_MODE_MASK 0xC0 +#define CS4271_MODE1_MODE_1X 0x00 +#define CS4271_MODE1_MODE_2X 0x80 +#define CS4271_MODE1_MODE_4X 0xC0 + +#define CS4271_MODE1_DIV_MASK 0x30 +#define CS4271_MODE1_DIV_1 0x00 +#define CS4271_MODE1_DIV_15 0x10 +#define CS4271_MODE1_DIV_2 0x20 +#define CS4271_MODE1_DIV_3 0x30 + +#define CS4271_MODE1_MASTER 0x08 + +#define CS4271_MODE1_DAC_DIF_MASK 0x07 +#define CS4271_MODE1_DAC_DIF_LJ 0x00 +#define CS4271_MODE1_DAC_DIF_I2S 0x01 +#define CS4271_MODE1_DAC_DIF_RJ16 0x02 +#define CS4271_MODE1_DAC_DIF_RJ24 0x03 +#define CS4271_MODE1_DAC_DIF_RJ20 0x04 +#define CS4271_MODE1_DAC_DIF_RJ18 0x05 + +#define CS4271_DACCTL_AMUTE 0x80 +#define CS4271_DACCTL_IF_SLOW 0x40 + +#define CS4271_DACCTL_DEM_MASK 0x30 +#define CS4271_DACCTL_DEM_DIS 0x00 +#define CS4271_DACCTL_DEM_441 0x10 +#define CS4271_DACCTL_DEM_48 0x20 +#define CS4271_DACCTL_DEM_32 0x30 + +#define CS4271_DACCTL_SVRU 0x08 +#define CS4271_DACCTL_SRD 0x04 +#define CS4271_DACCTL_INVA 0x02 +#define CS4271_DACCTL_INVB 0x01 + +#define CS4271_DACVOL_BEQUA 0x40 +#define CS4271_DACVOL_SOFT 0x20 +#define CS4271_DACVOL_ZEROC 0x10 + +#define CS4271_DACVOL_ATAPI_MASK 0x0F +#define CS4271_DACVOL_ATAPI_M_M 0x00 +#define CS4271_DACVOL_ATAPI_M_BR 0x01 +#define CS4271_DACVOL_ATAPI_M_BL 0x02 +#define CS4271_DACVOL_ATAPI_M_BLR2 0x03 +#define CS4271_DACVOL_ATAPI_AR_M 0x04 +#define CS4271_DACVOL_ATAPI_AR_BR 0x05 +#define CS4271_DACVOL_ATAPI_AR_BL 0x06 +#define CS4271_DACVOL_ATAPI_AR_BLR2 0x07 +#define CS4271_DACVOL_ATAPI_AL_M 0x08 +#define CS4271_DACVOL_ATAPI_AL_BR 0x09 +#define CS4271_DACVOL_ATAPI_AL_BL 0x0A +#define CS4271_DACVOL_ATAPI_AL_BLR2 0x0B +#define CS4271_DACVOL_ATAPI_ALR2_M 0x0C +#define CS4271_DACVOL_ATAPI_ALR2_BR 0x0D +#define CS4271_DACVOL_ATAPI_ALR2_BL 0x0E +#define CS4271_DACVOL_ATAPI_ALR2_BLR2 0x0F + +#define CS4271_VOLA_MUTE 0x80 +#define CS4271_VOLA_VOL_MASK 0x7F +#define CS4271_VOLB_MUTE 0x80 +#define CS4271_VOLB_VOL_MASK 0x7F + +#define CS4271_ADCCTL_DITHER16 0x20 + +#define CS4271_ADCCTL_ADC_DIF_MASK 0x10 +#define CS4271_ADCCTL_ADC_DIF_LJ 0x00 +#define CS4271_ADCCTL_ADC_DIF_I2S 0x10 + +#define CS4271_ADCCTL_MUTEA 0x08 +#define CS4271_ADCCTL_MUTEB 0x04 +#define CS4271_ADCCTL_HPFDA 0x02 +#define CS4271_ADCCTL_HPFDB 0x01 + +#define CS4271_MODE2_LOOP 0x10 +#define CS4271_MODE2_MUTECAEQUB 0x08 +#define CS4271_MODE2_FREEZE 0x04 +#define CS4271_MODE2_CPEN 0x02 +#define CS4271_MODE2_PDN 0x01 + +#define CS4271_CHIPID_PART_MASK 0xF0 +#define CS4271_CHIPID_REV_MASK 0x0F + +/* + * Default CS4271 power-up configuration + * Array contains non-existing in hw register at address 0 + * Array do not include Chip ID, as codec driver does not use + * registers read operations at all + */ +static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = { + 0, + 0, + CS4271_DACCTL_AMUTE, + CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, + 0, + 0, + 0, + 0, +}; + +struct cs4271_private { + /* SND_SOC_I2C or SND_SOC_SPI */ + enum snd_soc_control_type bus_type; + void *control_data; + unsigned int mclk; + bool master; + bool deemph; + /* Current sample rate for de-emphasis control */ + int rate; + /* GPIO driving Reset pin, if any */ + int gpio_nreset; + /* GPIO that disable serial bus, if any */ + int gpio_disable; +}; + +struct cs4271_clk_cfg { + unsigned int ratio; /* MCLK / sample rate */ + u8 speed_mode; /* codec speed mode: 1x, 2x, 4x */ + u8 mclk_master; /* ratio bit mask for Master mode */ + u8 mclk_slave; /* ratio bit mask for Slave mode */ +}; + +static struct cs4271_clk_cfg cs4271_clk_tab[] = { + {64, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, + {96, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, + {128, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, + {192, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, + {256, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, + {384, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, + {512, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_2, CS4271_MODE1_DIV_1}, + {768, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3}, + {1024, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3} +}; + +#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab) + +/* + * @freq is the desired MCLK rate + * MCLK rate should (c) be the sample rate, multiplied by one of the + * ratios listed in cs4271_mclk_fs_ratios table + */ +static int cs4271_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + cs4271->mclk = freq; + return 0; +} + +static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs4271->master = 0; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs4271->master = 1; + val |= CS4271_MODE1_MASTER; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + val |= CS4271_MODE1_DAC_DIF_LJ; + snd_soc_update_bits(codec, CS4271_ADCCTL, + CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_LJ); + break; + case SND_SOC_DAIFMT_I2S: + val |= CS4271_MODE1_DAC_DIF_I2S; + snd_soc_update_bits(codec, CS4271_ADCCTL, + CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_I2S); + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + snd_soc_update_bits(codec, CS4271_MODE1, + CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val); + + return 0; +} + +static int cs4271_deemph[] = {0, 44100, 48000, 32000}; + +static int cs4271_set_deemph(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + int i; + int val = CS4271_DACCTL_DEM_DIS; + + if (cs4271->deemph) { + /* Find closest de-emphasis freq */ + val = 1; + for (i = 2; i < ARRAY_SIZE(cs4271_deemph); i++) + if (abs(cs4271_deemph[i] - cs4271->rate) < + abs(cs4271_deemph[val] - cs4271->rate)) + val = i; + val <<= 4; + } + + return snd_soc_update_bits(codec, CS4271_DACCTL, + CS4271_DACCTL_DEM_MASK, val); +} + +static int cs4271_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = cs4271->deemph; + return 0; +} + +static int cs4271_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + cs4271->deemph = ucontrol->value.enumerated.item[0]; + return cs4271_set_deemph(codec); +} + +static int cs4271_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + unsigned int i, ratio, val; + + cs4271->rate = params_rate(params); + ratio = cs4271->mclk / cs4271->rate; + for (i = 0; i < CS4171_NR_RATIOS; i++) + if (cs4271_clk_tab[i].ratio == ratio) + break; + + if ((i == CS4171_NR_RATIOS) || ((ratio == 1024) && cs4271->master)) { + dev_err(codec->dev, "Invalid sample rate\n"); + return -EINVAL; + } + + /* Configure DAC */ + val = cs4271_clk_tab[i].speed_mode; + + if (cs4271->master) + val |= cs4271_clk_tab[i].mclk_master; + else + val |= cs4271_clk_tab[i].mclk_slave; + + snd_soc_update_bits(codec, CS4271_MODE1, + CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val); + + return cs4271_set_deemph(codec); +} + +static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int val_a = 0; + int val_b = 0; + + if (mute) { + val_a = CS4271_VOLA_MUTE; + val_b = CS4271_VOLB_MUTE; + } + + snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); + snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + + return 0; +} + +/* CS4271 controls */ +static DECLARE_TLV_DB_SCALE(cs4271_dac_tlv, -12700, 100, 0); + +static const struct snd_kcontrol_new cs4271_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", CS4271_VOLA, CS4271_VOLB, + 0, 0x7F, 1, cs4271_dac_tlv), + SOC_SINGLE("Digital Loopback Switch", CS4271_MODE2, 4, 1, 0), + SOC_SINGLE("Soft Ramp Switch", CS4271_DACVOL, 5, 1, 0), + SOC_SINGLE("Zero Cross Switch", CS4271_DACVOL, 4, 1, 0), + SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0, + cs4271_get_deemph, cs4271_put_deemph), + SOC_SINGLE("Auto-Mute Switch", CS4271_DACCTL, 7, 1, 0), + SOC_SINGLE("Slow Roll Off Filter Switch", CS4271_DACCTL, 6, 1, 0), + SOC_SINGLE("Soft Volume Ramp-Up Switch", CS4271_DACCTL, 3, 1, 0), + SOC_SINGLE("Soft Ramp-Down Switch", CS4271_DACCTL, 2, 1, 0), + SOC_SINGLE("Left Channel Inversion Switch", CS4271_DACCTL, 1, 1, 0), + SOC_SINGLE("Right Channel Inversion Switch", CS4271_DACCTL, 0, 1, 0), + SOC_DOUBLE("Master Capture Switch", CS4271_ADCCTL, 3, 2, 1, 1), + SOC_SINGLE("Dither 16-Bit Data Switch", CS4271_ADCCTL, 5, 1, 0), + SOC_DOUBLE("High Pass Filter Switch", CS4271_ADCCTL, 1, 0, 1, 1), + SOC_DOUBLE_R("Master Playback Switch", CS4271_VOLA, CS4271_VOLB, + 7, 1, 1), +}; + +static struct snd_soc_dai_ops cs4271_dai_ops = { + .hw_params = cs4271_hw_params, + .set_sysclk = cs4271_set_dai_sysclk, + .set_fmt = cs4271_set_dai_fmt, + .digital_mute = cs4271_digital_mute, +}; + +struct snd_soc_dai_driver cs4271_dai = { + .name = "cs4271-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = CS4271_PCM_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = CS4271_PCM_FORMATS, + }, + .ops = &cs4271_dai_ops, + .symmetric_rates = 1, +}; + +#ifdef CONFIG_PM +static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +{ + /* Set power-down bit */ + snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN); + return 0; +} + +static int cs4271_soc_resume(struct snd_soc_codec *codec) +{ + /* Restore codec state */ + snd_soc_cache_sync(codec); + /* then disable the power-down bit */ + snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + return 0; +} +#else +#define cs4271_soc_suspend NULL +#define cs4271_soc_resume NULL +#endif /* CONFIG_PM */ + +static int cs4271_probe(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; + int ret; + int gpio_nreset = -EINVAL; + int gpio_disable = -EINVAL; + + codec->control_data = cs4271->control_data; + + if (cs4271plat) { + if (gpio_is_valid(cs4271plat->gpio_nreset)) + gpio_nreset = cs4271plat->gpio_nreset; + if (gpio_is_valid(cs4271plat->gpio_disable)) + gpio_disable = cs4271plat->gpio_disable; + } + + if (gpio_disable >= 0) + if (gpio_request(gpio_disable, "CS4271 Disable")) + gpio_disable = -EINVAL; + if (gpio_disable >= 0) + gpio_direction_output(gpio_disable, 0); + + if (gpio_nreset >= 0) + if (gpio_request(gpio_nreset, "CS4271 Reset")) + gpio_nreset = -EINVAL; + if (gpio_nreset >= 0) { + /* Reset codec */ + gpio_direction_output(gpio_nreset, 0); + udelay(1); + gpio_set_value(gpio_nreset, 1); + /* Give the codec time to wake up */ + udelay(1); + } + + cs4271->gpio_nreset = gpio_nreset; + cs4271->gpio_disable = gpio_disable; + + /* + * In case of I2C, chip address specified in board data. + * So cache IO operations use 8 bit codec register address. + * In case of SPI, chip address and register address + * passed together as 16 bit value. + * Anyway, register address is masked with 0xFF inside + * soc-cache code. + */ + if (cs4271->bus_type == SND_SOC_SPI) + ret = snd_soc_codec_set_cache_io(codec, 16, 8, + cs4271->bus_type); + else + ret = snd_soc_codec_set_cache_io(codec, 8, 8, + cs4271->bus_type); + if (ret) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + snd_soc_update_bits(codec, CS4271_MODE2, 0, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN); + snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + /* Power-up sequence requires 85 uS */ + udelay(85); + + return snd_soc_add_controls(codec, cs4271_snd_controls, + ARRAY_SIZE(cs4271_snd_controls)); +} + +static int cs4271_remove(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + int gpio_nreset, gpio_disable; + + gpio_nreset = cs4271->gpio_nreset; + gpio_disable = cs4271->gpio_disable; + + if (gpio_is_valid(gpio_nreset)) { + /* Set codec to the reset state */ + gpio_set_value(gpio_nreset, 0); + gpio_free(gpio_nreset); + } + + if (gpio_is_valid(gpio_disable)) + gpio_free(gpio_disable); + + return 0; +}; + +struct snd_soc_codec_driver soc_codec_dev_cs4271 = { + .probe = cs4271_probe, + .remove = cs4271_remove, + .suspend = cs4271_soc_suspend, + .resume = cs4271_soc_resume, + .reg_cache_default = cs4271_dflt_reg, + .reg_cache_size = ARRAY_SIZE(cs4271_dflt_reg), + .reg_word_size = sizeof(cs4271_dflt_reg[0]), + .compress_type = SND_SOC_FLAT_COMPRESSION, +}; + +#if defined(CONFIG_SPI_MASTER) +static int __devinit cs4271_spi_probe(struct spi_device *spi) +{ + struct cs4271_private *cs4271; + + cs4271 = devm_kzalloc(&spi->dev, sizeof(*cs4271), GFP_KERNEL); + if (!cs4271) + return -ENOMEM; + + spi_set_drvdata(spi, cs4271); + cs4271->control_data = spi; + cs4271->bus_type = SND_SOC_SPI; + + return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271, + &cs4271_dai, 1); +} + +static int __devexit cs4271_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver cs4271_spi_driver = { + .driver = { + .name = "cs4271", + .owner = THIS_MODULE, + }, + .probe = cs4271_spi_probe, + .remove = __devexit_p(cs4271_spi_remove), +}; +#endif /* defined(CONFIG_SPI_MASTER) */ + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static struct i2c_device_id cs4271_i2c_id[] = { + {"cs4271", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id); + +static int __devinit cs4271_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs4271_private *cs4271; + + cs4271 = devm_kzalloc(&client->dev, sizeof(*cs4271), GFP_KERNEL); + if (!cs4271) + return -ENOMEM; + + i2c_set_clientdata(client, cs4271); + cs4271->control_data = client; + cs4271->bus_type = SND_SOC_I2C; + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271, + &cs4271_dai, 1); +} + +static int __devexit cs4271_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver cs4271_i2c_driver = { + .driver = { + .name = "cs4271", + .owner = THIS_MODULE, + }, + .id_table = cs4271_i2c_id, + .probe = cs4271_i2c_probe, + .remove = __devexit_p(cs4271_i2c_remove), +}; +#endif /* defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) */ + +/* + * We only register our serial bus driver here without + * assignment to particular chip. So if any of the below + * fails, there is some problem with I2C or SPI subsystem. + * In most cases this module will be compiled with support + * of only one serial bus. + */ +static int __init cs4271_modinit(void) +{ + int ret; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&cs4271_i2c_driver); + if (ret) { + pr_err("Failed to register CS4271 I2C driver: %d\n", ret); + return ret; + } +#endif + +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&cs4271_spi_driver); + if (ret) { + pr_err("Failed to register CS4271 SPI driver: %d\n", ret); + return ret; + } +#endif + + return 0; +} +module_init(cs4271_modinit); + +static void __exit cs4271_modexit(void) +{ +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&cs4271_spi_driver); +#endif + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&cs4271_i2c_driver); +#endif +} +module_exit(cs4271_modexit); + +MODULE_AUTHOR("Alexander Sverdlin "); +MODULE_DESCRIPTION("Cirrus Logic CS4271 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From c358e640a669b528b32af5442c92b856de623e1c Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Fri, 21 Jan 2011 15:29:02 +0000 Subject: ASoC: soc-cache: Add trace event for snd_soc_cache_sync() This patch makes it easy to see when the syncing process begins and ends. You can also enable the snd_soc_reg_write tracepoint to see which registers are being synced. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index f8348396379..db66dc44add 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -18,6 +18,8 @@ #include #include +#include + static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -1601,18 +1603,26 @@ EXPORT_SYMBOL_GPL(snd_soc_cache_write); int snd_soc_cache_sync(struct snd_soc_codec *codec) { int ret; + const char *name; if (!codec->cache_sync) { return 0; } + if (codec->cache_ops->name) + name = codec->cache_ops->name; + else + name = "unknown"; + if (codec->cache_ops && codec->cache_ops->sync) { if (codec->cache_ops->name) dev_dbg(codec->dev, "Syncing %s cache for %s codec\n", codec->cache_ops->name, codec->name); + trace_snd_soc_cache_sync(codec, name, "start"); ret = codec->cache_ops->sync(codec); if (!ret) codec->cache_sync = 0; + trace_snd_soc_cache_sync(codec, name, "end"); return ret; } -- cgit v1.2.3 From cb9c130aa97bd41887a0a391388ef4070caab4d9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Jan 2011 11:45:34 +0900 Subject: ASoC: ak4642: add SND_SOC_DAIFMT_FORMAT support This patch support LEFT_J / I2S only for now Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 24 ++++++++++++++++++++++++ sound/soc/sh/fsi-ak4642.c | 3 ++- 2 files changed, 26 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index f00eba313df..4be0570e3f1 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -116,6 +116,12 @@ #define BCKO_MASK (1 << 3) #define BCKO_64 BCKO_MASK +#define DIF_MASK (3 << 0) +#define DSP (0 << 0) +#define RIGHT_J (1 << 0) +#define LEFT_J (2 << 0) +#define I2S (3 << 0) + /* MD_CTL2 */ #define FS0 (1 << 0) #define FS1 (1 << 1) @@ -354,6 +360,24 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) snd_soc_update_bits(codec, PW_MGMT2, MS, data); snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); + /* format type */ + data = 0; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + data = LEFT_J; + break; + case SND_SOC_DAIFMT_I2S: + data = I2S; + break; + /* FIXME + * Please add RIGHT_J / DSP support here + */ + default: + return -EINVAL; + break; + } + snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data); + return 0; } diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index 56cd3422310..a722a4c661f 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -26,7 +26,8 @@ static int fsi_ak4642_dai_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *dai = rtd->codec_dai; int ret; - ret = snd_soc_dai_set_fmt(dai, SND_SOC_DAIFMT_CBM_CFM); + ret = snd_soc_dai_set_fmt(dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; -- cgit v1.2.3 From 0d032c19e764ad2f60292f58116f6c2a052a58ab Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Jan 2011 11:45:51 +0900 Subject: ASoC: sh: fsi: Add fsi_get_priv_frm_dai function Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 2b06402801e..1d0a16e8091 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -252,9 +252,8 @@ static struct snd_soc_dai *fsi_get_dai(struct snd_pcm_substream *substream) return rtd->cpu_dai; } -static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream) +static struct fsi_priv *fsi_get_priv_frm_dai(struct snd_soc_dai *dai) { - struct snd_soc_dai *dai = fsi_get_dai(substream); struct fsi_master *master = snd_soc_dai_get_drvdata(dai); if (dai->id == 0) @@ -263,6 +262,11 @@ static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream) return &master->fsib; } +static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream) +{ + return fsi_get_priv_frm_dai(fsi_get_dai(substream)); +} + static u32 fsi_get_info_flags(struct fsi_priv *fsi) { int is_porta = fsi_is_port_a(fsi); -- cgit v1.2.3 From 4d805f7b6607f6e547dc22e5d57c201e43d21c05 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 20 Jan 2011 11:46:02 +0900 Subject: ASoC: sh: fsi: Add snd_soc_dai_set_fmt support This patch add snd_soc_dai_ops :: set_fmt to FSI driver and select master/slave clock mode by snd_soc_dai_set_fmt on fsi-xxx.c instead of platform infomation code. This patch remove fsi_is_master function which is no longer needed. Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Acked-by: Paul Mundt Signed-off-by: Mark Brown --- sound/soc/sh/fsi-ak4642.c | 13 ++++++++---- sound/soc/sh/fsi-da7210.c | 12 +++++++++-- sound/soc/sh/fsi-hdmi.c | 11 ++++++++++ sound/soc/sh/fsi.c | 54 +++++++++++++++++++++++++++-------------------- 4 files changed, 61 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index a722a4c661f..ce058c749e6 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -23,15 +23,20 @@ struct fsi_ak4642_data { static int fsi_ak4642_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *codec = rtd->codec_dai; + struct snd_soc_dai *cpu = rtd->cpu_dai; int ret; - ret = snd_soc_dai_set_fmt(dai, SND_SOC_DAIFMT_LEFT_J | - SND_SOC_DAIFMT_CBM_CFM); + ret = snd_soc_dai_set_fmt(codec, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; - ret = snd_soc_dai_set_sysclk(dai, 0, 11289600, 0); + ret = snd_soc_dai_set_sysclk(codec, 0, 11289600, 0); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_CBS_CFS); return ret; } diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c index e8df9da92f7..9b24ed466ab 100644 --- a/sound/soc/sh/fsi-da7210.c +++ b/sound/soc/sh/fsi-da7210.c @@ -15,11 +15,19 @@ static int fsi_da7210_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *dai = rtd->codec_dai; + struct snd_soc_dai *codec = rtd->codec_dai; + struct snd_soc_dai *cpu = rtd->cpu_dai; + int ret; - return snd_soc_dai_set_fmt(dai, + ret = snd_soc_dai_set_fmt(codec, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_CBS_CFS); + + return ret; } static struct snd_soc_dai_link fsi_da7210_dai = { diff --git a/sound/soc/sh/fsi-hdmi.c b/sound/soc/sh/fsi-hdmi.c index a52dd8ec71d..96d8ce3f321 100644 --- a/sound/soc/sh/fsi-hdmi.c +++ b/sound/soc/sh/fsi-hdmi.c @@ -12,6 +12,16 @@ #include #include +static int fsi_hdmi_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *cpu = rtd->cpu_dai; + int ret; + + ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_CBM_CFM); + + return ret; +} + static struct snd_soc_dai_link fsi_dai_link = { .name = "HDMI", .stream_name = "HDMI", @@ -19,6 +29,7 @@ static struct snd_soc_dai_link fsi_dai_link = { .codec_dai_name = "sh_mobile_hdmi-hifi", .platform_name = "sh_fsi2", .codec_name = "sh-mobile-hdmi", + .init = fsi_hdmi_dai_init, }; static struct snd_soc_card fsi_soc_card = { diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 1d0a16e8091..5f39f364eff 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -78,6 +78,8 @@ /* CKG1 */ #define ACKMD_MASK 0x00007000 #define BPFMD_MASK 0x00000700 +#define DIMD (1 << 4) +#define DOMD (1 << 0) /* A/B MST_CTLR */ #define BP (1 << 4) /* Fix the signal of Biphase output */ @@ -292,21 +294,6 @@ static inline struct fsi_stream *fsi_get_stream(struct fsi_priv *fsi, return is_play ? &fsi->playback : &fsi->capture; } -static int fsi_is_master_mode(struct fsi_priv *fsi, int is_play) -{ - u32 mode; - u32 flags = fsi_get_info_flags(fsi); - - mode = is_play ? SH_FSI_OUT_SLAVE_MODE : SH_FSI_IN_SLAVE_MODE; - - /* return - * 1 : master mode - * 0 : slave mode - */ - - return (mode & flags) != mode; -} - static u32 fsi_get_port_shift(struct fsi_priv *fsi, int is_play) { int is_porta = fsi_is_port_a(fsi); @@ -764,19 +751,11 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, u32 fmt; u32 data; int is_play = fsi_is_play(substream); - int is_master; io = fsi_get_stream(fsi, is_play); pm_runtime_get_sync(dai->dev); - /* CKG1 */ - data = is_play ? (1 << 0) : (1 << 4); - is_master = fsi_is_master_mode(fsi, is_play); - if (is_master) - fsi_reg_mask_set(fsi, CKG1, data, data); - else - fsi_reg_mask_set(fsi, CKG1, data, 0); /* clock inversion (CKG2) */ data = 0; @@ -893,6 +872,34 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } +static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct fsi_priv *fsi = fsi_get_priv_frm_dai(dai); + u32 data = 0; + int ret; + + pm_runtime_get_sync(dai->dev); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + data = DIMD | DOMD; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + ret = -EINVAL; + goto set_fmt_exit; + } + fsi_reg_mask_set(fsi, CKG1, (DIMD | DOMD), data); + ret = 0; + +set_fmt_exit: + pm_runtime_put_sync(dai->dev); + + return ret; +} + static int fsi_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -979,6 +986,7 @@ static struct snd_soc_dai_ops fsi_dai_ops = { .startup = fsi_dai_startup, .shutdown = fsi_dai_shutdown, .trigger = fsi_dai_trigger, + .set_fmt = fsi_dai_set_fmt, .hw_params = fsi_dai_hw_params, }; -- cgit v1.2.3 From 86c3304181a25f127e46c864dc735e21f24484fc Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin Date: Sun, 23 Jan 2011 04:01:15 +0300 Subject: ASoC: EDB93xx machine sound driver with CS4271 Added support for EDB93xx sound with CS4271 CODEC. Features: - Playback, Capture - Sample rates from 8kHz to 96kHz tested Signed-off-by: Alexander Sverdlin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/ep93xx/Kconfig | 9 +++ sound/soc/ep93xx/Makefile | 2 + sound/soc/ep93xx/edb93xx.c | 142 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 153 insertions(+) create mode 100644 sound/soc/ep93xx/edb93xx.c (limited to 'sound') diff --git a/sound/soc/ep93xx/Kconfig b/sound/soc/ep93xx/Kconfig index 57429041189..91a28de9410 100644 --- a/sound/soc/ep93xx/Kconfig +++ b/sound/soc/ep93xx/Kconfig @@ -30,3 +30,12 @@ config SND_EP93XX_SOC_SIMONE help Say Y or M here if you want to add support for AC97 audio on the Simplemachines Sim.One board. + +config SND_EP93XX_SOC_EDB93XX + tristate "SoC Audio support for Cirrus Logic EDB93xx boards" + depends on SND_EP93XX_SOC && (MACH_EDB9301 || MACH_EDB9302 || MACH_EDB9302A || MACH_EDB9307A || MACH_EDB9315A) + select SND_EP93XX_SOC_I2S + select SND_SOC_CS4271 + help + Say Y or M here if you want to add support for I2S audio on the + Cirrus Logic EDB93xx boards. diff --git a/sound/soc/ep93xx/Makefile b/sound/soc/ep93xx/Makefile index 8e7977fb6b7..5514146cbdf 100644 --- a/sound/soc/ep93xx/Makefile +++ b/sound/soc/ep93xx/Makefile @@ -10,6 +10,8 @@ obj-$(CONFIG_SND_EP93XX_SOC_AC97) += snd-soc-ep93xx-ac97.o # EP93XX Machine Support snd-soc-snappercl15-objs := snappercl15.o snd-soc-simone-objs := simone.o +snd-soc-edb93xx-objs := edb93xx.o obj-$(CONFIG_SND_EP93XX_SOC_SNAPPERCL15) += snd-soc-snappercl15.o obj-$(CONFIG_SND_EP93XX_SOC_SIMONE) += snd-soc-simone.o +obj-$(CONFIG_SND_EP93XX_SOC_EDB93XX) += snd-soc-edb93xx.o diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c new file mode 100644 index 00000000000..b270085227f --- /dev/null +++ b/sound/soc/ep93xx/edb93xx.c @@ -0,0 +1,142 @@ +/* + * SoC audio for EDB93xx + * + * Copyright (c) 2010 Alexander Sverdlin + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * This driver support CS4271 codec being master or slave, working + * in control port mode, connected either via SPI or I2C. + * The data format accepted is I2S or left-justified. + * DAPM support not implemented. + */ + +#include +#include +#include +#include +#include +#include +#include +#include "ep93xx-pcm.h" + +#define edb93xx_has_audio() (machine_is_edb9301() || \ + machine_is_edb9302() || \ + machine_is_edb9302a() || \ + machine_is_edb9307a() || \ + machine_is_edb9315a()) + +static int edb93xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int err; + unsigned int rate = params_rate(params); + /* + * We set LRCLK equal to `rate' and SCLK = LRCLK * 64, + * because our sample size is 32 bit * 2 channels. + * I2S standard permits us to transmit more bits than + * the codec uses. + * MCLK = SCLK * 4 is the best recommended value, + * but we have to fall back to ratio 2 for higher + * sample rates. + */ + unsigned int mclk_rate = rate * 64 * ((rate <= 48000) ? 4 : 2); + + err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS); + if (err) + return err; + + err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS); + if (err) + return err; + + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate, + SND_SOC_CLOCK_IN); + if (err) + return err; + + return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_rate, + SND_SOC_CLOCK_OUT); +} + +static struct snd_soc_ops edb93xx_ops = { + .hw_params = edb93xx_hw_params, +}; + +static struct snd_soc_dai_link edb93xx_dai = { + .name = "CS4271", + .stream_name = "CS4271 HiFi", + .platform_name = "ep93xx-pcm-audio", + .cpu_dai_name = "ep93xx-i2s", + .codec_name = "spi0.0", + .codec_dai_name = "cs4271-hifi", + .ops = &edb93xx_ops, +}; + +static struct snd_soc_card snd_soc_edb93xx = { + .name = "EDB93XX", + .dai_link = &edb93xx_dai, + .num_links = 1, +}; + +static struct platform_device *edb93xx_snd_device; + +static int __init edb93xx_init(void) +{ + int ret; + + if (!edb93xx_has_audio()) + return -ENODEV; + + ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97, + EP93XX_SYSCON_I2SCLKDIV_ORIDE | + EP93XX_SYSCON_I2SCLKDIV_SPOL); + if (ret) + return ret; + + edb93xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!edb93xx_snd_device) { + ret = -ENOMEM; + goto free_i2s; + } + + platform_set_drvdata(edb93xx_snd_device, &snd_soc_edb93xx); + ret = platform_device_add(edb93xx_snd_device); + if (ret) + goto device_put; + + return 0; + +device_put: + platform_device_put(edb93xx_snd_device); +free_i2s: + ep93xx_i2s_release(); + return ret; +} +module_init(edb93xx_init); + +static void __exit edb93xx_exit(void) +{ + platform_device_unregister(edb93xx_snd_device); + ep93xx_i2s_release(); +} +module_exit(edb93xx_exit); + +MODULE_AUTHOR("Alexander Sverdlin "); +MODULE_DESCRIPTION("ALSA SoC EDB93xx"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 181e055e6bed80afbf8ba2bb5e3ce84fbd3f633c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 24 Jan 2011 14:05:25 +0000 Subject: ASoC: Fix type for snd_soc_volatile_register() We generally refer to registers as unsigned ints (including in the underlying CODEC driver operation). Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b0e7689159c..14861f95f62 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2029,7 +2029,8 @@ static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) * * Boolean function indiciating if a CODEC register is volatile. */ -int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg) +int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) { if (codec->volatile_register) return codec->volatile_register(codec, reg); -- cgit v1.2.3 From 3f25c9ccb7409e6f61c11e7828c0ddb54ab1cb7b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 24 Jan 2011 10:09:02 +0900 Subject: ASoC: sh: fsi-hdmi: Add FSI port and HDMI selection This patch add platform_device_id which can control PortA/PortB for FSI2-HDMI from platform data. Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/fsi-hdmi.c | 66 +++++++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 61 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi-hdmi.c b/sound/soc/sh/fsi-hdmi.c index 96d8ce3f321..9719985eb82 100644 --- a/sound/soc/sh/fsi-hdmi.c +++ b/sound/soc/sh/fsi-hdmi.c @@ -12,6 +12,12 @@ #include #include +struct fsi_hdmi_data { + const char *cpu_dai; + const char *card; + int id; +}; + static int fsi_hdmi_dai_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *cpu = rtd->cpu_dai; @@ -25,7 +31,6 @@ static int fsi_hdmi_dai_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link fsi_dai_link = { .name = "HDMI", .stream_name = "HDMI", - .cpu_dai_name = "fsib-dai", /* fsi B */ .codec_dai_name = "sh_mobile_hdmi-hifi", .platform_name = "sh_fsi2", .codec_name = "sh-mobile-hdmi", @@ -33,21 +38,33 @@ static struct snd_soc_dai_link fsi_dai_link = { }; static struct snd_soc_card fsi_soc_card = { - .name = "FSI (SH MOBILE HDMI)", .dai_link = &fsi_dai_link, .num_links = 1, }; static struct platform_device *fsi_snd_device; -static int __init fsi_hdmi_init(void) +static int fsi_hdmi_probe(struct platform_device *pdev) { int ret = -ENOMEM; + const struct platform_device_id *id_entry; + struct fsi_hdmi_data *pdata; - fsi_snd_device = platform_device_alloc("soc-audio", FSI_PORT_B); + id_entry = pdev->id_entry; + if (!id_entry) { + dev_err(&pdev->dev, "unknown fsi hdmi\n"); + return -ENODEV; + } + + pdata = (struct fsi_hdmi_data *)id_entry->driver_data; + + fsi_snd_device = platform_device_alloc("soc-audio", pdata->id); if (!fsi_snd_device) goto out; + fsi_dai_link.cpu_dai_name = pdata->cpu_dai; + fsi_soc_card.name = pdata->card; + platform_set_drvdata(fsi_snd_device, &fsi_soc_card); ret = platform_device_add(fsi_snd_device); @@ -58,9 +75,48 @@ out: return ret; } -static void __exit fsi_hdmi_exit(void) +static int fsi_hdmi_remove(struct platform_device *pdev) { platform_device_unregister(fsi_snd_device); + return 0; +} + +static struct fsi_hdmi_data fsi2_a_hdmi = { + .cpu_dai = "fsia-dai", + .card = "FSI2A (SH MOBILE HDMI)", + .id = FSI_PORT_A, +}; + +static struct fsi_hdmi_data fsi2_b_hdmi = { + .cpu_dai = "fsib-dai", + .card = "FSI2B (SH MOBILE HDMI)", + .id = FSI_PORT_B, +}; + +static struct platform_device_id fsi_id_table[] = { + /* FSI 2 */ + { "sh_fsi2_a_hdmi", (kernel_ulong_t)&fsi2_a_hdmi }, + { "sh_fsi2_b_hdmi", (kernel_ulong_t)&fsi2_b_hdmi }, + {}, +}; + +static struct platform_driver fsi_hdmi = { + .driver = { + .name = "fsi-hdmi-audio", + }, + .probe = fsi_hdmi_probe, + .remove = fsi_hdmi_remove, + .id_table = fsi_id_table, +}; + +static int __init fsi_hdmi_init(void) +{ + return platform_driver_register(&fsi_hdmi); +} + +static void __exit fsi_hdmi_exit(void) +{ + platform_driver_unregister(&fsi_hdmi); } module_init(fsi_hdmi_init); -- cgit v1.2.3 From 0d42e6e77f8b872483833a7146286edaaaeb2f39 Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin Date: Fri, 21 Jan 2011 22:22:07 +0300 Subject: ASoC: cs4271.c: improve error handling CS4271 CODEC driver adapted to recently introduced error handling in snd_soc_update_bits(). Added snd_soc_cache_sync() error handling. Signed-off-by: Alexander Sverdlin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 59 +++++++++++++++++++++++++++++++++++------------ 1 file changed, 44 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 237ece3f104..5357ec5f5d7 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -209,6 +209,7 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); unsigned int val = 0; + int ret; switch (format & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -226,22 +227,27 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_LEFT_J: val |= CS4271_MODE1_DAC_DIF_LJ; - snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = snd_soc_update_bits(codec, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_LJ); + if (ret < 0) + return ret; break; case SND_SOC_DAIFMT_I2S: val |= CS4271_MODE1_DAC_DIF_I2S; - snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = snd_soc_update_bits(codec, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_I2S); + if (ret < 0) + return ret; break; default: dev_err(codec->dev, "Invalid DAI format\n"); return -EINVAL; } - snd_soc_update_bits(codec, CS4271_MODE1, + ret = snd_soc_update_bits(codec, CS4271_MODE1, CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val); - + if (ret < 0) + return ret; return 0; } @@ -250,7 +256,7 @@ static int cs4271_deemph[] = {0, 44100, 48000, 32000}; static int cs4271_set_deemph(struct snd_soc_codec *codec) { struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - int i; + int i, ret; int val = CS4271_DACCTL_DEM_DIS; if (cs4271->deemph) { @@ -263,8 +269,11 @@ static int cs4271_set_deemph(struct snd_soc_codec *codec) val <<= 4; } - return snd_soc_update_bits(codec, CS4271_DACCTL, + ret = snd_soc_update_bits(codec, CS4271_DACCTL, CS4271_DACCTL_DEM_MASK, val); + if (ret < 0) + return ret; + return 0; } static int cs4271_get_deemph(struct snd_kcontrol *kcontrol, @@ -294,7 +303,8 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - unsigned int i, ratio, val; + int i, ret; + unsigned int ratio, val; cs4271->rate = params_rate(params); ratio = cs4271->mclk / cs4271->rate; @@ -315,8 +325,10 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, else val |= cs4271_clk_tab[i].mclk_slave; - snd_soc_update_bits(codec, CS4271_MODE1, + ret = snd_soc_update_bits(codec, CS4271_MODE1, CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val); + if (ret < 0) + return ret; return cs4271_set_deemph(codec); } @@ -324,6 +336,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; + int ret; int val_a = 0; int val_b = 0; @@ -332,8 +345,12 @@ static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) val_b = CS4271_VOLB_MUTE; } - snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); - snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + ret = snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); + if (ret < 0) + return ret; + ret = snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + if (ret < 0) + return ret; return 0; } @@ -392,17 +409,25 @@ struct snd_soc_dai_driver cs4271_dai = { #ifdef CONFIG_PM static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) { + int ret; /* Set power-down bit */ - snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN); + ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN); + if (ret < 0) + return ret; return 0; } static int cs4271_soc_resume(struct snd_soc_codec *codec) { + int ret; /* Restore codec state */ - snd_soc_cache_sync(codec); + ret = snd_soc_cache_sync(codec); + if (ret < 0) + return ret; /* then disable the power-down bit */ - snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + if (ret < 0) + return ret; return 0; } #else @@ -467,9 +492,13 @@ static int cs4271_probe(struct snd_soc_codec *codec) return ret; } - snd_soc_update_bits(codec, CS4271_MODE2, 0, + ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN | CS4271_MODE2_CPEN); - snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + if (ret < 0) + return ret; + ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + if (ret < 0) + return ret; /* Power-up sequence requires 85 uS */ udelay(85); -- cgit v1.2.3 From 160afa7f05ef4922b136335cd1793b607ffb0d7c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 24 Jan 2011 10:42:08 +0900 Subject: ASoC: sh: fsi: move chan_num from fsi_stream to fsi_priv Signed-off-by: Kuninori Morimoto Acked-by: Paul Mundt Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 34 +++++++++++++++------------------- 1 file changed, 15 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 5f39f364eff..605ea7f17cf 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -130,7 +130,6 @@ struct fsi_stream { struct snd_pcm_substream *substream; int fifo_max_num; - int chan_num; int buff_offset; int buff_len; @@ -145,6 +144,7 @@ struct fsi_priv { void __iomem *base; struct fsi_master *master; + int chan_num; struct fsi_stream playback; struct fsi_stream capture; @@ -348,7 +348,6 @@ static void fsi_stream_pop(struct fsi_priv *fsi, int is_play) static int fsi_get_fifo_data_num(struct fsi_priv *fsi, int is_play) { u32 status; - struct fsi_stream *io = fsi_get_stream(fsi, is_play); int data_num; status = is_play ? @@ -356,7 +355,7 @@ static int fsi_get_fifo_data_num(struct fsi_priv *fsi, int is_play) fsi_reg_read(fsi, DIFF_ST); data_num = 0x1ff & (status >> 8); - data_num *= io->chan_num; + data_num *= fsi->chan_num; return data_num; } @@ -378,7 +377,7 @@ static int fsi_get_frame_width(struct fsi_priv *fsi, int is_play) struct snd_pcm_substream *substream = io->substream; struct snd_pcm_runtime *runtime = substream->runtime; - return frames_to_bytes(runtime, 1) / io->chan_num; + return frames_to_bytes(runtime, 1) / fsi->chan_num; } static void fsi_count_fifo_err(struct fsi_priv *fsi) @@ -571,10 +570,10 @@ static void fsi_fifo_init(struct fsi_priv *fsi, * 7 channels: 32 ( 32 x 7 = 224) * 8 channels: 32 ( 32 x 8 = 256) */ - for (i = 1; i < io->chan_num; i <<= 1) + for (i = 1; i < fsi->chan_num; i <<= 1) io->fifo_max_num >>= 1; dev_dbg(dai->dev, "%d channel %d store\n", - io->chan_num, io->fifo_max_num); + fsi->chan_num, io->fifo_max_num); /* * set interrupt generation factor @@ -650,7 +649,7 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int stream) * data_num_max : number of FSI fifo free space * data_num : number of ALSA residue data */ - data_num_max = io->fifo_max_num * io->chan_num; + data_num_max = io->fifo_max_num * fsi->chan_num; data_num_max -= fsi_get_fifo_data_num(fsi, is_play); data_num = data_residue_num; @@ -746,14 +745,11 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, { struct fsi_priv *fsi = fsi_get_priv(substream); struct fsi_master *master = fsi_get_master(fsi); - struct fsi_stream *io; u32 flags = fsi_get_info_flags(fsi); u32 fmt; u32 data; int is_play = fsi_is_play(substream); - io = fsi_get_stream(fsi, is_play); - pm_runtime_get_sync(dai->dev); @@ -776,29 +772,29 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, switch (fmt) { case SH_FSI_FMT_MONO: data = CR_MONO; - io->chan_num = 1; + fsi->chan_num = 1; break; case SH_FSI_FMT_MONO_DELAY: data = CR_MONO_D; - io->chan_num = 1; + fsi->chan_num = 1; break; case SH_FSI_FMT_PCM: data = CR_PCM; - io->chan_num = 2; + fsi->chan_num = 2; break; case SH_FSI_FMT_I2S: data = CR_I2S; - io->chan_num = 2; + fsi->chan_num = 2; break; case SH_FSI_FMT_TDM: - io->chan_num = is_play ? + fsi->chan_num = is_play ? SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags); - data = CR_TDM | (io->chan_num - 1); + data = CR_TDM | (fsi->chan_num - 1); break; case SH_FSI_FMT_TDM_DELAY: - io->chan_num = is_play ? + fsi->chan_num = is_play ? SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags); - data = CR_TDM_D | (io->chan_num - 1); + data = CR_TDM_D | (fsi->chan_num - 1); break; case SH_FSI_FMT_SPDIF: if (master->core->ver < 2) { @@ -806,7 +802,7 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, return -EINVAL; } data = CR_BWS_16 | CR_DTMD_SPDIF_PCM | CR_PCM; - io->chan_num = 2; + fsi->chan_num = 2; fsi_spdif_clk_ctrl(fsi, 1); fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD); break; -- cgit v1.2.3 From d7c5762bc72ea4184c413166c063899dffae7385 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 24 Jan 2011 10:42:33 +0900 Subject: ASoC: sh: fsi: free from NULL pointer of struct sh_fsi_platform_info Current FSI driver assumed master->info is not NULL. This patch allow NULL in master->info Signed-off-by: Kuninori Morimoto Acked-by: Paul Mundt Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 21 +++++++++++++++++---- 1 file changed, 17 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 605ea7f17cf..3c53693d726 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -113,6 +113,8 @@ #define FSI_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) +typedef int (*set_rate_func)(struct device *dev, int is_porta, int rate, int enable); + /* * FSI driver use below type name for variable * @@ -269,11 +271,22 @@ static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream) return fsi_get_priv_frm_dai(fsi_get_dai(substream)); } +static set_rate_func fsi_get_info_set_rate(struct fsi_master *master) +{ + if (!master->info) + return NULL; + + return master->info->set_rate; +} + static u32 fsi_get_info_flags(struct fsi_priv *fsi) { int is_porta = fsi_is_port_a(fsi); struct fsi_master *master = fsi_get_master(fsi); + if (!master->info) + return 0; + return is_porta ? master->info->porta_flags : master->info->portb_flags; } @@ -830,12 +843,12 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream, struct fsi_priv *fsi = fsi_get_priv(substream); int is_play = fsi_is_play(substream); struct fsi_master *master = fsi_get_master(fsi); - int (*set_rate)(struct device *dev, int is_porta, int rate, int enable); + set_rate_func set_rate; fsi_irq_disable(fsi, is_play); fsi_clk_ctrl(fsi, 0); - set_rate = master->info->set_rate; + set_rate = fsi_get_info_set_rate(master); if (set_rate && fsi->rate) set_rate(dai->dev, fsi_is_port_a(fsi), fsi->rate, 0); fsi->rate = 0; @@ -902,12 +915,12 @@ static int fsi_dai_hw_params(struct snd_pcm_substream *substream, { struct fsi_priv *fsi = fsi_get_priv(substream); struct fsi_master *master = fsi_get_master(fsi); - int (*set_rate)(struct device *dev, int is_porta, int rate, int enable); + set_rate_func set_rate; int fsi_ver = master->core->ver; long rate = params_rate(params); int ret; - set_rate = master->info->set_rate; + set_rate = fsi_get_info_set_rate(master); if (!set_rate) return 0; -- cgit v1.2.3 From f17c13ca52d5c5a6a164536244a6debb8cd17983 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 24 Jan 2011 10:43:19 +0900 Subject: ASoC: sh: fsi: modify selection method of I2S/PCM/SPDIF format Current format selection of FSI-codecs depended on platform information for FSI, and chip default settings for codecs. It is not understandable/formal method. This patch modify FSI and FSI-codecs to use snd_soc_dai_set_fmt. But FSI can use I2S/PCM and SPDIF format today. It can be selected to I2S/PCM by snd_soc_dai_set_fmt, but can not select SPDIF. So, this patch change FSI platform information to have DAI/SPDIF mode. If platform selects DAI mode (default), FSI-codecs can select I2S/PCM by snd_soc_dai_set_fmt, and if it is SPDIF mode, FSI become SPDIF format. Signed-off-by: Kuninori Morimoto Acked-by: Paul Mundt Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/fsi-ak4642.c | 3 +- sound/soc/sh/fsi-da7210.c | 3 +- sound/soc/sh/fsi.c | 106 ++++++++++++++++++++++++---------------------- 3 files changed, 59 insertions(+), 53 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index ce058c749e6..d6f4703b3c0 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -36,7 +36,8 @@ static int fsi_ak4642_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; - ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_CBS_CFS); + ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_CBS_CFS); return ret; } diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c index 9b24ed466ab..dbafd7ac559 100644 --- a/sound/soc/sh/fsi-da7210.c +++ b/sound/soc/sh/fsi-da7210.c @@ -25,7 +25,8 @@ static int fsi_da7210_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; - ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_CBS_CFS); + ret = snd_soc_dai_set_fmt(cpu, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_CBS_CFS); return ret; } diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 3c53693d726..0c9997e2d8c 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -757,9 +757,7 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct fsi_priv *fsi = fsi_get_priv(substream); - struct fsi_master *master = fsi_get_master(fsi); u32 flags = fsi_get_info_flags(fsi); - u32 fmt; u32 data; int is_play = fsi_is_play(substream); @@ -779,54 +777,6 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, fsi_reg_write(fsi, CKG2, data); - /* do fmt, di fmt */ - data = 0; - fmt = is_play ? SH_FSI_GET_OFMT(flags) : SH_FSI_GET_IFMT(flags); - switch (fmt) { - case SH_FSI_FMT_MONO: - data = CR_MONO; - fsi->chan_num = 1; - break; - case SH_FSI_FMT_MONO_DELAY: - data = CR_MONO_D; - fsi->chan_num = 1; - break; - case SH_FSI_FMT_PCM: - data = CR_PCM; - fsi->chan_num = 2; - break; - case SH_FSI_FMT_I2S: - data = CR_I2S; - fsi->chan_num = 2; - break; - case SH_FSI_FMT_TDM: - fsi->chan_num = is_play ? - SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags); - data = CR_TDM | (fsi->chan_num - 1); - break; - case SH_FSI_FMT_TDM_DELAY: - fsi->chan_num = is_play ? - SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags); - data = CR_TDM_D | (fsi->chan_num - 1); - break; - case SH_FSI_FMT_SPDIF: - if (master->core->ver < 2) { - dev_err(dai->dev, "This FSI can not use SPDIF\n"); - return -EINVAL; - } - data = CR_BWS_16 | CR_DTMD_SPDIF_PCM | CR_PCM; - fsi->chan_num = 2; - fsi_spdif_clk_ctrl(fsi, 1); - fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD); - break; - default: - dev_err(dai->dev, "unknown format.\n"); - return -EINVAL; - } - is_play ? - fsi_reg_write(fsi, DO_FMT, data) : - fsi_reg_write(fsi, DI_FMT, data); - /* irq clear */ fsi_irq_disable(fsi, is_play); fsi_irq_clear_status(fsi); @@ -881,9 +831,52 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } +static int fsi_set_fmt_dai(struct fsi_priv *fsi, unsigned int fmt) +{ + u32 data = 0; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + data = CR_I2S; + fsi->chan_num = 2; + break; + case SND_SOC_DAIFMT_LEFT_J: + data = CR_PCM; + fsi->chan_num = 2; + break; + default: + return -EINVAL; + } + + fsi_reg_write(fsi, DO_FMT, data); + fsi_reg_write(fsi, DI_FMT, data); + + return 0; +} + +static int fsi_set_fmt_spdif(struct fsi_priv *fsi) +{ + struct fsi_master *master = fsi_get_master(fsi); + u32 data = 0; + + if (master->core->ver < 2) + return -EINVAL; + + data = CR_BWS_16 | CR_DTMD_SPDIF_PCM | CR_PCM; + fsi->chan_num = 2; + fsi_spdif_clk_ctrl(fsi, 1); + fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD); + + fsi_reg_write(fsi, DO_FMT, data); + fsi_reg_write(fsi, DI_FMT, data); + + return 0; +} + static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct fsi_priv *fsi = fsi_get_priv_frm_dai(dai); + u32 flags = fsi_get_info_flags(fsi); u32 data = 0; int ret; @@ -901,7 +894,18 @@ static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) goto set_fmt_exit; } fsi_reg_mask_set(fsi, CKG1, (DIMD | DOMD), data); - ret = 0; + + /* set format */ + switch (flags & SH_FSI_FMT_MASK) { + case SH_FSI_FMT_DAI: + ret = fsi_set_fmt_dai(fsi, fmt & SND_SOC_DAIFMT_FORMAT_MASK); + break; + case SH_FSI_FMT_SPDIF: + ret = fsi_set_fmt_spdif(fsi); + break; + default: + ret = -EINVAL; + } set_fmt_exit: pm_runtime_put_sync(dai->dev); -- cgit v1.2.3 From 16af7d60aa27d3fc39e46fd456b8e33d34d60437 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jan 2011 11:35:28 +0000 Subject: ASoC: Staticise non-exported symbols in cs4271 Signed-off-by: Mark Brown Acked-by: Alexander Sverdlin Acked-by: Liam Girdwood --- sound/soc/codecs/cs4271.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 5357ec5f5d7..9c5b7db0ce6 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -386,7 +386,7 @@ static struct snd_soc_dai_ops cs4271_dai_ops = { .digital_mute = cs4271_digital_mute, }; -struct snd_soc_dai_driver cs4271_dai = { +static struct snd_soc_dai_driver cs4271_dai = { .name = "cs4271-hifi", .playback = { .stream_name = "Playback", @@ -526,7 +526,7 @@ static int cs4271_remove(struct snd_soc_codec *codec) return 0; }; -struct snd_soc_codec_driver soc_codec_dev_cs4271 = { +static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .probe = cs4271_probe, .remove = cs4271_remove, .suspend = cs4271_soc_suspend, -- cgit v1.2.3 From 70b2ac126a60c87145ae8a8eb1b4dccaa0bf5468 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jan 2011 14:05:25 +0000 Subject: ASoC: Use card rather than soc-audio device to card PM functions The platform device for the card is tied closely to the soc-audio implementation which we're currently trying to remove in favour of allowing cards to have their own devices. Begin removing it by replacing it with the card in the suspend and resume callbacks we give to cards, also taking the opportunity to remove the legacy suspend types which are currently hard coded anyway. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/pxa/raumfeld.c | 4 ++-- sound/soc/pxa/zylonite.c | 5 ++--- sound/soc/soc-core.c | 9 ++++----- 3 files changed, 8 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 0fd60f42303..db1dd560a58 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -151,13 +151,13 @@ static struct snd_soc_ops raumfeld_cs4270_ops = { .hw_params = raumfeld_cs4270_hw_params, }; -static int raumfeld_line_suspend(struct platform_device *pdev, pm_message_t state) +static int raumfeld_line_suspend(struct snd_soc_card *card) { raumfeld_enable_audio(false); return 0; } -static int raumfeld_line_resume(struct platform_device *pdev) +static int raumfeld_line_resume(struct snd_soc_card *card) { raumfeld_enable_audio(true); return 0; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index b222a7d7202..7b729013a72 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -226,8 +226,7 @@ static int zylonite_remove(struct platform_device *pdev) return 0; } -static int zylonite_suspend_post(struct platform_device *pdev, - pm_message_t state) +static int zylonite_suspend_post(struct snd_soc_card *card) { if (clk_pout) clk_disable(pout); @@ -235,7 +234,7 @@ static int zylonite_suspend_post(struct platform_device *pdev, return 0; } -static int zylonite_resume_pre(struct platform_device *pdev) +static int zylonite_resume_pre(struct snd_soc_card *card) { int ret = 0; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 14861f95f62..446838e7d3e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1011,7 +1011,7 @@ static int soc_suspend(struct device *dev) } if (card->suspend_pre) - card->suspend_pre(pdev, PMSG_SUSPEND); + card->suspend_pre(card); for (i = 0; i < card->num_rtd; i++) { struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; @@ -1078,7 +1078,7 @@ static int soc_suspend(struct device *dev) } if (card->suspend_post) - card->suspend_post(pdev, PMSG_SUSPEND); + card->suspend_post(card); return 0; } @@ -1090,7 +1090,6 @@ static void soc_resume_deferred(struct work_struct *work) { struct snd_soc_card *card = container_of(work, struct snd_soc_card, deferred_resume_work); - struct platform_device *pdev = to_platform_device(card->dev); struct snd_soc_codec *codec; int i; @@ -1104,7 +1103,7 @@ static void soc_resume_deferred(struct work_struct *work) snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D2); if (card->resume_pre) - card->resume_pre(pdev); + card->resume_pre(card); /* resume AC97 DAIs */ for (i = 0; i < card->num_rtd; i++) { @@ -1179,7 +1178,7 @@ static void soc_resume_deferred(struct work_struct *work) } if (card->resume_post) - card->resume_post(pdev); + card->resume_post(card); dev_dbg(card->dev, "resume work completed\n"); -- cgit v1.2.3 From e7361ec4996c170c63c4ac379085896db85ff34d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jan 2011 14:17:20 +0000 Subject: ASoC: Replace pdev with card in machine driver probe and remove In order to support cards instantiated without using soc-audio remove the use of the platform device in the card probe() and remove() ops. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/fsl/mpc8610_hpcd.c | 6 ++---- sound/soc/fsl/p1022_ds.c | 6 ++---- sound/soc/pxa/tosa.c | 4 ++-- sound/soc/pxa/zylonite.c | 4 ++-- sound/soc/soc-core.c | 8 +++----- 5 files changed, 11 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 7d7847a1e66..c16c6b2eff9 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -53,9 +53,8 @@ struct mpc8610_hpcd_data { * * Here we program the DMACR and PMUXCR registers. */ -static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) +static int mpc8610_hpcd_machine_probe(struct snd_soc_card *card) { - struct snd_soc_card *card = platform_get_drvdata(sound_device); struct mpc8610_hpcd_data *machine_data = container_of(card, struct mpc8610_hpcd_data, card); struct ccsr_guts_86xx __iomem *guts; @@ -138,9 +137,8 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) * This function is called to remove the sound device for one SSI. We * de-program the DMACR and PMUXCR register. */ -static int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) +static int mpc8610_hpcd_machine_remove(struct snd_soc_card *card) { - struct snd_soc_card *card = platform_get_drvdata(sound_device); struct mpc8610_hpcd_data *machine_data = container_of(card, struct mpc8610_hpcd_data, card); struct ccsr_guts_86xx __iomem *guts; diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 026b756961e..66e0b68af14 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -85,9 +85,8 @@ struct machine_data { * * Here we program the DMACR and PMUXCR registers. */ -static int p1022_ds_machine_probe(struct platform_device *sound_device) +static int p1022_ds_machine_probe(struct snd_soc_card *card) { - struct snd_soc_card *card = platform_get_drvdata(sound_device); struct machine_data *mdata = container_of(card, struct machine_data, card); struct ccsr_guts_85xx __iomem *guts; @@ -160,9 +159,8 @@ static int p1022_ds_startup(struct snd_pcm_substream *substream) * This function is called to remove the sound device for one SSI. We * de-program the DMACR and PMUXCR register. */ -static int p1022_ds_machine_remove(struct platform_device *sound_device) +static int p1022_ds_machine_remove(struct snd_soc_card *card) { - struct snd_soc_card *card = platform_get_drvdata(sound_device); struct machine_data *mdata = container_of(card, struct machine_data, card); struct ccsr_guts_85xx __iomem *guts; diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index f75804ef089..489139a31cf 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -237,7 +237,7 @@ static struct snd_soc_dai_link tosa_dai[] = { }, }; -static int tosa_probe(struct platform_device *dev) +static int tosa_probe(struct snd_soc_card *card) { int ret; @@ -251,7 +251,7 @@ static int tosa_probe(struct platform_device *dev) return ret; } -static int tosa_remove(struct platform_device *dev) +static int tosa_remove(struct snd_soc_card *card) { gpio_free(TOSA_GPIO_L_MUTE); return 0; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 7b729013a72..c5858296b48 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -189,7 +189,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { }, }; -static int zylonite_probe(struct platform_device *pdev) +static int zylonite_probe(struct snd_soc_card *card) { int ret; @@ -216,7 +216,7 @@ static int zylonite_probe(struct platform_device *pdev) return 0; } -static int zylonite_remove(struct platform_device *pdev) +static int zylonite_remove(struct snd_soc_card *card) { if (clk_pout) { clk_disable(pout); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 446838e7d3e..4bc2365bf1d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1717,7 +1717,6 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec, static void snd_soc_instantiate_card(struct snd_soc_card *card) { - struct platform_device *pdev = to_platform_device(card->dev); struct snd_soc_codec *codec; struct snd_soc_codec_conf *codec_conf; enum snd_soc_compress_type compress_type; @@ -1781,7 +1780,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) /* initialise the sound card only once */ if (card->probe) { - ret = card->probe(pdev); + ret = card->probe(card); if (ret < 0) goto card_probe_error; } @@ -1842,7 +1841,7 @@ probe_dai_err: card_probe_error: if (card->remove) - card->remove(pdev); + card->remove(card); snd_card_free(card->snd_card); @@ -1888,7 +1887,6 @@ static int soc_probe(struct platform_device *pdev) static int soc_cleanup_card_resources(struct snd_soc_card *card) { - struct platform_device *pdev = to_platform_device(card->dev); int i; /* make sure any delayed work runs */ @@ -1909,7 +1907,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) /* remove the card */ if (card->remove) - card->remove(pdev); + card->remove(card); kfree(card->rtd); snd_card_free(card->snd_card); -- cgit v1.2.3 From 6f8ab4ac292f81b9246ddf363bf1c6a2fc7a0629 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jan 2011 14:59:27 +0000 Subject: ASoC: Export card PM callbacks for use in direct registered cards Allow hookup of cards registered directly with the core to the PM operations by exporting the device power management operations to modules, also exporting the default PM operations since it is expected that most cards will end up using exactly the same setup. Note that the callbacks require that the driver data for the card be the snd_soc_card. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4bc2365bf1d..5dffc7a469c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -965,12 +965,11 @@ static struct snd_pcm_ops soc_pcm_ops = { .pointer = soc_pcm_pointer, }; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP /* powers down audio subsystem for suspend */ -static int soc_suspend(struct device *dev) +int snd_soc_suspend(struct device *dev) { - struct platform_device *pdev = to_platform_device(dev); - struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_card *card = dev_get_drvdata(dev); struct snd_soc_codec *codec; int i; @@ -1082,6 +1081,7 @@ static int soc_suspend(struct device *dev) return 0; } +EXPORT_SYMBOL_GPL(snd_soc_suspend); /* deferred resume work, so resume can complete before we finished * setting our codec back up, which can be very slow on I2C @@ -1187,10 +1187,9 @@ static void soc_resume_deferred(struct work_struct *work) } /* powers up audio subsystem after a suspend */ -static int soc_resume(struct device *dev) +int snd_soc_resume(struct device *dev) { - struct platform_device *pdev = to_platform_device(dev); - struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_card *card = dev_get_drvdata(dev); int i; /* AC97 devices might have other drivers hanging off them so @@ -1212,9 +1211,10 @@ static int soc_resume(struct device *dev) return 0; } +EXPORT_SYMBOL_GPL(snd_soc_resume); #else -#define soc_suspend NULL -#define soc_resume NULL +#define snd_soc_suspend NULL +#define snd_soc_resume NULL #endif static struct snd_soc_dai_ops null_dai_ops = { @@ -1924,10 +1924,9 @@ static int soc_remove(struct platform_device *pdev) return 0; } -static int soc_poweroff(struct device *dev) +int snd_soc_poweroff(struct device *dev) { - struct platform_device *pdev = to_platform_device(dev); - struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_card *card = dev_get_drvdata(dev); int i; if (!card->instantiated) @@ -1944,11 +1943,12 @@ static int soc_poweroff(struct device *dev) return 0; } +EXPORT_SYMBOL_GPL(snd_soc_poweroff); -static const struct dev_pm_ops soc_pm_ops = { - .suspend = soc_suspend, - .resume = soc_resume, - .poweroff = soc_poweroff, +const struct dev_pm_ops snd_soc_pm_ops = { + .suspend = snd_soc_suspend, + .resume = snd_soc_resume, + .poweroff = snd_soc_poweroff, }; /* ASoC platform driver */ @@ -1956,7 +1956,7 @@ static struct platform_driver soc_driver = { .driver = { .name = "soc-audio", .owner = THIS_MODULE, - .pm = &soc_pm_ops, + .pm = &snd_soc_pm_ops, }, .probe = soc_probe, .remove = soc_remove, -- cgit v1.2.3 From aaee8ef146111566e1c607bdf368d73fb966be2e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jan 2011 20:53:50 +0000 Subject: ASoC: Make cache status available via debugfs Could just as well live in sysfs but sysfs doesn't have the simple value export helpers debugfs does. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5dffc7a469c..8af47f7a8fc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -235,6 +235,11 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) return; } + debugfs_create_bool("cache_sync", 0444, codec->debugfs_codec_root, + &codec->cache_sync); + debugfs_create_bool("cache_only", 0444, codec->debugfs_codec_root, + &codec->cache_only); + codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, codec->debugfs_codec_root, codec, &codec_reg_fops); -- cgit v1.2.3 From f85a9e0d260905f98d4ca6b66f0e64f63a729dba Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 26 Jan 2011 21:41:28 +0000 Subject: ASoC: Add subsequence information to seq_notify callbacks Allows drivers to distinguish which subsequence is being notified when they get called back. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 37b376f4c75..0f94fd057f2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -899,7 +899,8 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) if (sort[i] == cur_sort) cur_dapm->seq_notifier(cur_dapm, - i); + i, + cur_subseq); } INIT_LIST_HEAD(&pending); @@ -968,7 +969,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) if (sort[i] == cur_sort) cur_dapm->seq_notifier(cur_dapm, - i); + i, cur_subseq); } } -- cgit v1.2.3 From 48529b3b7ca76e688146e13d1a1666fe69703f0c Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 27 Jan 2011 16:47:11 +0200 Subject: ASoC: omap: rx51: Add stereo output support to audio jack Audio jack in Nokia RX-51/N900 is driven by TPA6130 headphone amplifier. This patch adds support for it and stereo output can be active when "Jack Function" == "TV-OUT" || "Headphone". As the TPA6130 can output very high volume levels the output is limited with snd_soc_limit_volume. Limiting value is found from Maemo kernel sources. Signed-off-by: Jarkko Nikula Cc: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/Kconfig | 1 + sound/soc/omap/rx51.c | 43 +++++++++++++++++++++++++++++++++++++++---- 2 files changed, 40 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index a088db6d509..b5922984eac 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -24,6 +24,7 @@ config SND_OMAP_SOC_RX51 select OMAP_MCBSP select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X + select SND_SOC_TPA6130A2 help Say Y if you want to add support for SoC audio on Nokia RX-51 hardware. This is also known as Nokia N900 product. diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 09fb0df8d41..251afbe303c 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -31,6 +31,7 @@ #include #include #include +#include "../codecs/tpa6130a2.h" #include @@ -47,7 +48,8 @@ enum { RX51_JACK_DISABLED, - RX51_JACK_TVOUT, /* tv-out */ + RX51_JACK_TVOUT, /* tv-out with stereo output */ + RX51_JACK_HP, /* headphone: stereo output, no mic */ }; static int rx51_spk_func; @@ -57,6 +59,15 @@ static int rx51_jack_func; static void rx51_ext_control(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = &codec->dapm; + int hp = 0, tvout = 0; + + switch (rx51_jack_func) { + case RX51_JACK_TVOUT: + tvout = 1; + case RX51_JACK_HP: + hp = 1; + break; + } if (rx51_spk_func) snd_soc_dapm_enable_pin(dapm, "Ext Spk"); @@ -66,9 +77,12 @@ static void rx51_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_enable_pin(dapm, "DMic"); else snd_soc_dapm_disable_pin(dapm, "DMic"); + if (hp) + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + else + snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); - gpio_set_value(RX51_TVOUT_SEL_GPIO, - rx51_jack_func == RX51_JACK_TVOUT); + gpio_set_value(RX51_TVOUT_SEL_GPIO, tvout); snd_soc_dapm_sync(dapm); } @@ -153,6 +167,19 @@ static int rx51_spk_event(struct snd_soc_dapm_widget *w, return 0; } +static int rx51_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_codec *codec = w->dapm->codec; + + if (SND_SOC_DAPM_EVENT_ON(event)) + tpa6130a2_stereo_enable(codec, 1); + else + tpa6130a2_stereo_enable(codec, 0); + + return 0; +} + static int rx51_get_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -212,11 +239,14 @@ static struct snd_soc_jack_gpio rx51_av_jack_gpios[] = { static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event), SND_SOC_DAPM_MIC("DMic", NULL), + SND_SOC_DAPM_HP("Headphone Jack", rx51_hp_event), }; static const struct snd_soc_dapm_route audio_map[] = { {"Ext Spk", NULL, "HPLOUT"}, {"Ext Spk", NULL, "HPROUT"}, + {"Headphone Jack", NULL, "LLOUT"}, + {"Headphone Jack", NULL, "RLOUT"}, {"DMic Rate 64", NULL, "Mic Bias 2V"}, {"Mic Bias 2V", NULL, "DMic"}, @@ -224,7 +254,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static const char *spk_function[] = {"Off", "On"}; static const char *input_function[] = {"ADC", "Digital Mic"}; -static const char *jack_function[] = {"Off", "TV-OUT"}; +static const char *jack_function[] = {"Off", "TV-OUT", "Headphone"}; static const struct soc_enum rx51_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), @@ -265,6 +295,11 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) /* Set up RX-51 specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + err = tpa6130a2_add_controls(codec); + if (err < 0) + return err; + snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42); + snd_soc_dapm_sync(dapm); /* AV jack detection */ -- cgit v1.2.3 From 70d29331ac551c11e6e3934f43a548ae8154351e Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 27 Jan 2011 16:24:22 +0200 Subject: ASoC: soc-core: Increment codec and platform driver refcounts before probing Commit f6c2ed5 "ASoC: Fix the device references to codec and platform drivers" moved codec and platform driver refcount increments from soc_bind_dai_link to more appropriate places. Adjust a little them so that refcounts are incremented before executing the driver probe functions. Signed-off-by: Jarkko Nikula Cc: Vinod Koul Cc: Harsha Priya Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e9f81c551e5..9dfbb8fcb76 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1412,26 +1412,31 @@ static int soc_probe_codec(struct snd_soc_card *card, codec->dapm.card = card; soc_set_name_prefix(card, codec); + if (!try_module_get(codec->dev->driver->owner)) + return -ENODEV; + if (codec->driver->probe) { ret = codec->driver->probe(codec); if (ret < 0) { dev_err(codec->dev, "asoc: failed to probe CODEC %s: %d\n", codec->name, ret); - return ret; + goto err_probe; } } soc_init_codec_debugfs(codec); /* mark codec as probed and add to card codec list */ - if (!try_module_get(codec->dev->driver->owner)) - return -ENODEV; - codec->probed = 1; list_add(&codec->card_list, &card->codec_dev_list); list_add(&codec->dapm.list, &card->dapm_list); + return 0; + +err_probe: + module_put(codec->dev->driver->owner); + return ret; } @@ -1549,19 +1554,19 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) /* probe the platform */ if (!platform->probed) { + if (!try_module_get(platform->dev->driver->owner)) + return -ENODEV; + if (platform->driver->probe) { ret = platform->driver->probe(platform); if (ret < 0) { printk(KERN_ERR "asoc: failed to probe platform %s\n", platform->name); + module_put(platform->dev->driver->owner); return ret; } } /* mark platform as probed and add to card platform list */ - - if (!try_module_get(platform->dev->driver->owner)) - return -ENODEV; - platform->probed = 1; list_add(&platform->card_list, &card->platform_dev_list); } -- cgit v1.2.3 From fd94eeef06ed4abc08f58e42f46341d0bc4f7793 Mon Sep 17 00:00:00 2001 From: Harsha Priya Date: Fri, 28 Jan 2011 22:26:53 +0530 Subject: ASoC: sn95031: add capture support This patch adds the support for capture path in sn95031 codec. This codec supports upto 6DMICs, 2 AMICs and Linein. The linein and AMICs are connected through a MUX to ADC. The TX paths can be assigned to any of the ADCs or DMICs. Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 281 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 281 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 593632cf791..40e285df9ae 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -33,6 +33,7 @@ #include #include #include +#include #include "sn95031.h" #define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100) @@ -145,6 +146,129 @@ static int sn95031_vihf_event(struct snd_soc_dapm_widget *w, return 0; } +static int sn95031_dmic12_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + unsigned int ldo = 0, clk_dir = 0, data_dir = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ldo = BIT(5)|BIT(4); + clk_dir = BIT(0); + data_dir = BIT(7); + } + /* program DMIC LDO, clock and set clock */ + snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo); + snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(0), clk_dir); + snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(7), data_dir); + return 0; +} + +static int sn95031_dmic34_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + unsigned int ldo = 0, clk_dir = 0, data_dir = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ldo = BIT(5)|BIT(4); + clk_dir = BIT(2); + data_dir = BIT(1); + } + /* program DMIC LDO, clock and set clock */ + snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(5)|BIT(4), ldo); + snd_soc_update_bits(w->codec, SN95031_DMICBUF0123, BIT(2), clk_dir); + snd_soc_update_bits(w->codec, SN95031_DMICBUF45, BIT(1), data_dir); + return 0; +} + +static int sn95031_dmic56_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + unsigned int ldo = 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) + ldo = BIT(7)|BIT(6); + + /* program DMIC LDO */ + snd_soc_update_bits(w->codec, SN95031_MICBIAS, BIT(7)|BIT(6), ldo); + return 0; +} + +/* mux controls */ +static const char *sn95031_mic_texts[] = { "AMIC", "LineIn" }; + +static const struct soc_enum sn95031_micl_enum = + SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 1, 2, sn95031_mic_texts); + +static const struct snd_kcontrol_new sn95031_micl_mux_control = + SOC_DAPM_ENUM("Route", sn95031_micl_enum); + +static const struct soc_enum sn95031_micr_enum = + SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 3, 2, sn95031_mic_texts); + +static const struct snd_kcontrol_new sn95031_micr_mux_control = + SOC_DAPM_ENUM("Route", sn95031_micr_enum); + +static const char *sn95031_input_texts[] = { "DMIC1", "DMIC2", "DMIC3", + "DMIC4", "DMIC5", "DMIC6", + "ADC Left", "ADC Right" }; + +static const struct soc_enum sn95031_input1_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 0, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input1_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input1_enum); + +static const struct soc_enum sn95031_input2_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 4, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input2_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input2_enum); + +static const struct soc_enum sn95031_input3_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 0, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input3_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input3_enum); + +static const struct soc_enum sn95031_input4_enum = + SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 4, 8, sn95031_input_texts); + +static const struct snd_kcontrol_new sn95031_input4_mux_control = + SOC_DAPM_ENUM("Route", sn95031_input4_enum); + +/* capture path controls */ + +static const char *sn95031_micmode_text[] = {"Single Ended", "Differential"}; + +/* 0dB to 30dB in 10dB steps */ +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 30); + +static const struct soc_enum sn95031_micmode1_enum = + SOC_ENUM_SINGLE(SN95031_MICAMP1, 1, 2, sn95031_micmode_text); +static const struct soc_enum sn95031_micmode2_enum = + SOC_ENUM_SINGLE(SN95031_MICAMP2, 1, 2, sn95031_micmode_text); + +static const char *sn95031_dmic_cfg_text[] = {"GPO", "DMIC"}; + +static const struct soc_enum sn95031_dmic12_cfg_enum = + SOC_ENUM_SINGLE(SN95031_DMICMUX, 0, 2, sn95031_dmic_cfg_text); +static const struct soc_enum sn95031_dmic34_cfg_enum = + SOC_ENUM_SINGLE(SN95031_DMICMUX, 1, 2, sn95031_dmic_cfg_text); +static const struct soc_enum sn95031_dmic56_cfg_enum = + SOC_ENUM_SINGLE(SN95031_DMICMUX, 2, 2, sn95031_dmic_cfg_text); + +static const struct snd_kcontrol_new sn95031_snd_controls[] = { + SOC_ENUM("Mic1Mode Capture Route", sn95031_micmode1_enum), + SOC_ENUM("Mic2Mode Capture Route", sn95031_micmode2_enum), + SOC_ENUM("DMIC12 Capture Route", sn95031_dmic12_cfg_enum), + SOC_ENUM("DMIC34 Capture Route", sn95031_dmic34_cfg_enum), + SOC_ENUM("DMIC56 Capture Route", sn95031_dmic56_cfg_enum), + SOC_SINGLE_TLV("Mic1 Capture Volume", SN95031_MICAMP1, + 2, 4, 0, mic_tlv), + SOC_SINGLE_TLV("Mic2 Capture Volume", SN95031_MICAMP2, + 2, 4, 0, mic_tlv), +}; + /* DAPM widgets */ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { @@ -159,6 +283,36 @@ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("VIB1OUT"), SND_SOC_DAPM_OUTPUT("VIB2OUT"), + SND_SOC_DAPM_INPUT("AMIC1"), /* headset mic */ + SND_SOC_DAPM_INPUT("AMIC2"), + SND_SOC_DAPM_INPUT("DMIC1"), + SND_SOC_DAPM_INPUT("DMIC2"), + SND_SOC_DAPM_INPUT("DMIC3"), + SND_SOC_DAPM_INPUT("DMIC4"), + SND_SOC_DAPM_INPUT("DMIC5"), + SND_SOC_DAPM_INPUT("DMIC6"), + SND_SOC_DAPM_INPUT("LINEINL"), + SND_SOC_DAPM_INPUT("LINEINR"), + + SND_SOC_DAPM_MICBIAS("AMIC1Bias", SN95031_MICBIAS, 2, 0), + SND_SOC_DAPM_MICBIAS("AMIC2Bias", SN95031_MICBIAS, 3, 0), + SND_SOC_DAPM_MICBIAS("DMIC12Bias", SN95031_DMICMUX, 3, 0), + SND_SOC_DAPM_MICBIAS("DMIC34Bias", SN95031_DMICMUX, 4, 0), + SND_SOC_DAPM_MICBIAS("DMIC56Bias", SN95031_DMICMUX, 5, 0), + + SND_SOC_DAPM_SUPPLY("DMIC12supply", SN95031_DMICLK, 0, 0, + sn95031_dmic12_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("DMIC34supply", SN95031_DMICLK, 1, 0, + sn95031_dmic34_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("DMIC56supply", SN95031_DMICLK, 2, 0, + sn95031_dmic56_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_AIF_OUT("PCM_Out", "Capture", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("Headset Rail", SND_SOC_NOPM, 0, 0, sn95031_vhs_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), @@ -209,6 +363,40 @@ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { SN95031_VIB1C5, 1, 0), SND_SOC_DAPM_DAC("Vibra2 DAC", "Vibra2", SN95031_VIB2C5, 1, 0), + + /* capture widgets */ + SND_SOC_DAPM_PGA("LineIn Enable Left", SN95031_MICAMP1, + 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("LineIn Enable Right", SN95031_MICAMP2, + 7, 0, NULL, 0), + + SND_SOC_DAPM_PGA("MIC1 Enable", SN95031_MICAMP1, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIC2 Enable", SN95031_MICAMP2, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("TX1 Enable", SN95031_AUDIOTXEN, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("TX2 Enable", SN95031_AUDIOTXEN, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("TX3 Enable", SN95031_AUDIOTXEN, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("TX4 Enable", SN95031_AUDIOTXEN, 5, 0, NULL, 0), + + /* ADC have null stream as they will be turned ON by TX path */ + SND_SOC_DAPM_ADC("ADC Left", NULL, + SN95031_ADCCONFIG, 0, 0), + SND_SOC_DAPM_ADC("ADC Right", NULL, + SN95031_ADCCONFIG, 2, 0), + + SND_SOC_DAPM_MUX("Mic_InputL Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_micl_mux_control), + SND_SOC_DAPM_MUX("Mic_InputR Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_micr_mux_control), + + SND_SOC_DAPM_MUX("Txpath1 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input1_mux_control), + SND_SOC_DAPM_MUX("Txpath2 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input2_mux_control), + SND_SOC_DAPM_MUX("Txpath3 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input3_mux_control), + SND_SOC_DAPM_MUX("Txpath4 Capture Route", + SND_SOC_NOPM, 0, 0, &sn95031_input4_mux_control), + }; static const struct snd_soc_dapm_route sn95031_audio_map[] = { @@ -250,6 +438,87 @@ static const struct snd_soc_dapm_route sn95031_audio_map[] = { { "Lineout Right Playback", NULL, "Headset Right Filter"}, { "Lineout Right Playback", NULL, "Speaker Right Filter"}, { "Lineout Right Playback", NULL, "Vibra2 DAC"}, + + /* Headset (AMIC1) mic */ + { "AMIC1Bias", NULL, "AMIC1"}, + { "MIC1 Enable", NULL, "AMIC1Bias"}, + { "Mic_InputL Capture Route", "AMIC", "MIC1 Enable"}, + + /* AMIC2 */ + { "AMIC2Bias", NULL, "AMIC2"}, + { "MIC2 Enable", NULL, "AMIC2Bias"}, + { "Mic_InputR Capture Route", "AMIC", "MIC2 Enable"}, + + + /* Linein */ + { "LineIn Enable Left", NULL, "LINEINL"}, + { "LineIn Enable Right", NULL, "LINEINR"}, + { "Mic_InputL Capture Route", "LineIn", "LineIn Enable Left"}, + { "Mic_InputR Capture Route", "LineIn", "LineIn Enable Right"}, + + /* ADC connection */ + { "ADC Left", NULL, "Mic_InputL Capture Route"}, + { "ADC Right", NULL, "Mic_InputR Capture Route"}, + + /*DMIC connections */ + { "DMIC1", NULL, "DMIC12supply"}, + { "DMIC2", NULL, "DMIC12supply"}, + { "DMIC3", NULL, "DMIC34supply"}, + { "DMIC4", NULL, "DMIC34supply"}, + { "DMIC5", NULL, "DMIC56supply"}, + { "DMIC6", NULL, "DMIC56supply"}, + + { "DMIC12Bias", NULL, "DMIC1"}, + { "DMIC12Bias", NULL, "DMIC2"}, + { "DMIC34Bias", NULL, "DMIC3"}, + { "DMIC34Bias", NULL, "DMIC4"}, + { "DMIC56Bias", NULL, "DMIC5"}, + { "DMIC56Bias", NULL, "DMIC6"}, + + /*TX path inputs*/ + { "Txpath1 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath2 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath3 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath4 Capture Route", "ADC Left", "ADC Left"}, + { "Txpath1 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath2 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath3 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath4 Capture Route", "ADC Right", "ADC Right"}, + { "Txpath1 Capture Route", NULL, "DMIC1"}, + { "Txpath2 Capture Route", NULL, "DMIC1"}, + { "Txpath3 Capture Route", NULL, "DMIC1"}, + { "Txpath4 Capture Route", NULL, "DMIC1"}, + { "Txpath1 Capture Route", NULL, "DMIC2"}, + { "Txpath2 Capture Route", NULL, "DMIC2"}, + { "Txpath3 Capture Route", NULL, "DMIC2"}, + { "Txpath4 Capture Route", NULL, "DMIC2"}, + { "Txpath1 Capture Route", NULL, "DMIC3"}, + { "Txpath2 Capture Route", NULL, "DMIC3"}, + { "Txpath3 Capture Route", NULL, "DMIC3"}, + { "Txpath4 Capture Route", NULL, "DMIC3"}, + { "Txpath1 Capture Route", NULL, "DMIC4"}, + { "Txpath2 Capture Route", NULL, "DMIC4"}, + { "Txpath3 Capture Route", NULL, "DMIC4"}, + { "Txpath4 Capture Route", NULL, "DMIC4"}, + { "Txpath1 Capture Route", NULL, "DMIC5"}, + { "Txpath2 Capture Route", NULL, "DMIC5"}, + { "Txpath3 Capture Route", NULL, "DMIC5"}, + { "Txpath4 Capture Route", NULL, "DMIC5"}, + { "Txpath1 Capture Route", NULL, "DMIC6"}, + { "Txpath2 Capture Route", NULL, "DMIC6"}, + { "Txpath3 Capture Route", NULL, "DMIC6"}, + { "Txpath4 Capture Route", NULL, "DMIC6"}, + + /* tx path */ + { "TX1 Enable", NULL, "Txpath1 Capture Route"}, + { "TX2 Enable", NULL, "Txpath2 Capture Route"}, + { "TX3 Enable", NULL, "Txpath3 Capture Route"}, + { "TX4 Enable", NULL, "Txpath4 Capture Route"}, + { "PCM_Out", NULL, "TX1 Enable"}, + { "PCM_Out", NULL, "TX2 Enable"}, + { "PCM_Out", NULL, "TX3 Enable"}, + { "PCM_Out", NULL, "TX4 Enable"}, + }; /* speaker and headset mutes, for audio pops and clicks */ @@ -339,6 +608,13 @@ struct snd_soc_dai_driver sn95031_dais[] = { .rates = SN95031_RATES, .formats = SN95031_FORMATS, }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 5, + .rates = SN95031_RATES, + .formats = SN95031_FORMATS, + }, .ops = &sn95031_headset_dai_ops, }, { .name = "SN95031 Speaker", @@ -390,6 +666,8 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_PCM2RXSLOT01, 0x10); snd_soc_write(codec, SN95031_PCM2RXSLOT23, 0x32); snd_soc_write(codec, SN95031_PCM2RXSLOT45, 0x54); + snd_soc_write(codec, SN95031_PCM2TXSLOT01, 0x10); + snd_soc_write(codec, SN95031_PCM2TXSLOT23, 0x32); /* pcm port setting * This sets the pcm port to slave and clock at 19.2Mhz which * can support 6slots, sampling rate set per stream in hw-params @@ -423,6 +701,9 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_SSR2, 0x10); snd_soc_write(codec, SN95031_SSR3, 0x40); + snd_soc_add_controls(codec, sn95031_snd_controls, + ARRAY_SIZE(sn95031_snd_controls)); + ret = snd_soc_dapm_new_controls(&codec->dapm, sn95031_dapm_widgets, ARRAY_SIZE(sn95031_dapm_widgets)); if (ret) -- cgit v1.2.3 From a7bffdf7d885197857bb04919344dc17ffaf79b9 Mon Sep 17 00:00:00 2001 From: Harsha Priya Date: Fri, 28 Jan 2011 22:27:26 +0530 Subject: ASoC: sst_platform: add support for capture stream on headset dai Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/mid-x86/sst_platform.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 1d1f5447b33..96e6e9c9c5f 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -73,6 +73,12 @@ struct snd_soc_dai_driver sst_platform_dai[] = { .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S24_LE, }, + .capture = { + .channels_min = 1, + .channels_max = 5, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE, + }, }, { .name = "Speaker-cpu-dai", -- cgit v1.2.3 From d316553a0cb4569a9b6260e870cab8b9c102eace Mon Sep 17 00:00:00 2001 From: Harsha Priya Date: Fri, 28 Jan 2011 22:28:32 +0530 Subject: ASoC: mid-x86: Add support for capture in machine driver This configures the capture unused pins Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/mid-x86/mfld_machine.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 1a330be1a01..7925851a5de 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -182,6 +182,9 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); lo_dac = 3; hs_switch = 0; + /* we dont use linein in this so set to NC */ + snd_soc_dapm_disable_pin(dapm, "LINEINL"); + snd_soc_dapm_disable_pin(dapm, "LINEINR"); return snd_soc_dapm_sync(dapm); } -- cgit v1.2.3 From 111c6419ffeef0ef225dfdd2c8b7b03cc6bb228e Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 28 Jan 2011 14:26:35 -0700 Subject: ASoC: Move card list initialization to snd_soc_register_card All ASoC cards need snd_soc_initialize_card_lists called. Previously, it was only called for cards backed by a "soc-audio" platform device, via soc_probe(). However, it's also needed for cards backed by other platform devices, and registered directly via snd_soc_register_card(). Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9dfbb8fcb76..3feddd91b97 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1881,7 +1881,6 @@ static int soc_probe(struct platform_device *pdev) /* Bodge while we unpick instantiation */ card->dev = &pdev->dev; - snd_soc_initialize_card_lists(card); ret = snd_soc_register_card(card); if (ret != 0) { @@ -3122,6 +3121,8 @@ int snd_soc_register_card(struct snd_soc_card *card) if (!card->name || !card->dev) return -EINVAL; + snd_soc_initialize_card_lists(card); + soc_init_card_debugfs(card); card->rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime) * -- cgit v1.2.3 From 72de2b1a9a5b869c1eaea098dd2a75e6bbbf0488 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 28 Jan 2011 14:26:36 -0700 Subject: ASoC: Tegra: Harmony: Don't use soc-audio platform device Previously, snd-soc-tegra-harmony internally instantiated a platform device object whenever the module was loaded. Instead, switch to a more typical model where arch/arm/mach-tegra defines a platform device, and snd-soc-tegra-harmony acts as a driver for such a platform device. Define a new struct tegra_harmony to store driver data in the future. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/harmony.c | 94 +++++++++++++++++++++++++++++++++-------------- 1 file changed, 66 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index b160b7113f4..2d6a15c6bdf 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -2,7 +2,7 @@ * harmony.c - Harmony machine ASoC driver * * Author: Stephen Warren - * Copyright (C) 2010 - NVIDIA, Inc. + * Copyright (C) 2010-2011 - NVIDIA, Inc. * * Based on code copyright/by: * @@ -29,7 +29,11 @@ */ #include + #include +#include +#include + #include #include #include @@ -40,9 +44,11 @@ #include "tegra_pcm.h" #include "tegra_asoc_utils.h" -#define PREFIX "ASoC Harmony: " +#define DRV_NAME "tegra-snd-harmony" +#define PREFIX DRV_NAME ": " -static struct platform_device *harmony_snd_device; +struct tegra_harmony { +}; static int harmony_asoc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -154,56 +160,88 @@ static struct snd_soc_card snd_soc_harmony = { .num_links = 1, }; -static int __init harmony_soc_modinit(void) +static __devinit int tegra_snd_harmony_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_harmony; + struct tegra_harmony *harmony; int ret; if (!machine_is_harmony()) { - pr_err(PREFIX "Not running on Tegra Harmony!\n"); + dev_err(&pdev->dev, "Not running on Tegra Harmony!\n"); return -ENODEV; } - ret = tegra_asoc_utils_init(); - if (ret) { - return ret; + harmony = kzalloc(sizeof(struct tegra_harmony), GFP_KERNEL); + if (!harmony) { + dev_err(&pdev->dev, "Can't allocate tegra_harmony\n"); + return -ENOMEM; } - /* - * Create and register platform device - */ - harmony_snd_device = platform_device_alloc("soc-audio", -1); - if (harmony_snd_device == NULL) { - pr_err(PREFIX "platform_device_alloc failed\n"); - ret = -ENOMEM; - goto err_clock_utils; - } + ret = tegra_asoc_utils_init(); + if (ret) + goto err_free_harmony; - platform_set_drvdata(harmony_snd_device, &snd_soc_harmony); + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, harmony); - ret = platform_device_add(harmony_snd_device); + ret = snd_soc_register_card(card); if (ret) { - pr_err(PREFIX "platform_device_add failed (%d)\n", + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto err_device_put; + goto err_clear_drvdata; } return 0; -err_device_put: - platform_device_put(harmony_snd_device); -err_clock_utils: +err_clear_drvdata: + snd_soc_card_set_drvdata(card, NULL); + platform_set_drvdata(pdev, NULL); + card->dev = NULL; tegra_asoc_utils_fini(); +err_free_harmony: + kfree(harmony); return ret; } -module_init(harmony_soc_modinit); -static void __exit harmony_soc_modexit(void) +static int __devexit tegra_snd_harmony_remove(struct platform_device *pdev) { - platform_device_unregister(harmony_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + snd_soc_card_set_drvdata(card, NULL); + platform_set_drvdata(pdev, NULL); + card->dev = NULL; tegra_asoc_utils_fini(); + + kfree(harmony); + + return 0; +} + +static struct platform_driver tegra_snd_harmony_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + .probe = tegra_snd_harmony_probe, + .remove = __devexit_p(tegra_snd_harmony_remove), +}; + +static int __init snd_tegra_harmony_init(void) +{ + return platform_driver_register(&tegra_snd_harmony_driver); +} +module_init(snd_tegra_harmony_init); + +static void __exit snd_tegra_harmony_exit(void) +{ + platform_driver_unregister(&tegra_snd_harmony_driver); } -module_exit(harmony_soc_modexit); +module_exit(snd_tegra_harmony_exit); MODULE_AUTHOR("Stephen Warren "); MODULE_DESCRIPTION("Harmony machine ASoC driver"); -- cgit v1.2.3 From 6e267645046ecc161791b5a0d2eef6e494681416 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 28 Jan 2011 14:26:37 -0700 Subject: ASoC: Tegra: Harmony: Support the internal speaker Add DAPM widget definitions for the internal speaker paths. Currently, this path is always enabled while playback is active. Add code to control the speaker amplifier GPIO. The GPIO is requested during _init, since that's the first time it is guaranteed that the WM8903 module is loaded, probed, and hence has exported its GPIO chip. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/harmony.c | 51 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 51 insertions(+) (limited to 'sound') diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index 2d6a15c6bdf..0f5a4815093 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -33,6 +33,9 @@ #include #include #include +#include + +#include #include #include @@ -48,6 +51,8 @@ #define PREFIX DRV_NAME ": " struct tegra_harmony { + struct harmony_audio_platform_data *pdata; + int gpio_spkr_en_requested; }; static int harmony_asoc_hw_params(struct snd_pcm_substream *substream, @@ -113,7 +118,22 @@ static struct snd_soc_ops harmony_asoc_ops = { .hw_params = harmony_asoc_hw_params, }; +static int harmony_event_int_spk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct snd_soc_card *card = codec->card; + struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card); + struct harmony_audio_platform_data *pdata = harmony->pdata; + + gpio_set_value_cansleep(pdata->gpio_spkr_en, + !!SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + static const struct snd_soc_dapm_widget harmony_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Int Spk", harmony_event_int_spk), SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), }; @@ -121,6 +141,10 @@ static const struct snd_soc_dapm_widget harmony_dapm_widgets[] = { static const struct snd_soc_dapm_route harmony_audio_map[] = { {"Headphone Jack", NULL, "HPOUTR"}, {"Headphone Jack", NULL, "HPOUTL"}, + {"Int Spk", NULL, "ROP"}, + {"Int Spk", NULL, "RON"}, + {"Int Spk", NULL, "LOP"}, + {"Int Spk", NULL, "LON"}, {"Mic Bias", NULL, "Mic Jack"}, {"IN1L", NULL, "Mic Bias"}, }; @@ -129,6 +153,19 @@ static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_card *card = codec->card; + struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card); + struct harmony_audio_platform_data *pdata = harmony->pdata; + int ret; + + ret = gpio_request(pdata->gpio_spkr_en, "spkr_en"); + if (ret) { + dev_err(card->dev, "cannot get spkr_en gpio\n"); + return ret; + } + harmony->gpio_spkr_en_requested = 1; + + gpio_direction_output(pdata->gpio_spkr_en, 0); snd_soc_dapm_new_controls(dapm, harmony_dapm_widgets, ARRAY_SIZE(harmony_dapm_widgets)); @@ -137,6 +174,7 @@ static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(harmony_audio_map)); snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(dapm, "Int Spk"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); snd_soc_dapm_sync(dapm); @@ -164,6 +202,7 @@ static __devinit int tegra_snd_harmony_probe(struct platform_device *pdev) { struct snd_soc_card *card = &snd_soc_harmony; struct tegra_harmony *harmony; + struct harmony_audio_platform_data *pdata; int ret; if (!machine_is_harmony()) { @@ -171,12 +210,20 @@ static __devinit int tegra_snd_harmony_probe(struct platform_device *pdev) return -ENODEV; } + pdata = pdev->dev.platform_data; + if (!pdata) { + dev_err(&pdev->dev, "no platform data supplied\n"); + return -EINVAL; + } + harmony = kzalloc(sizeof(struct tegra_harmony), GFP_KERNEL); if (!harmony) { dev_err(&pdev->dev, "Can't allocate tegra_harmony\n"); return -ENOMEM; } + harmony->pdata = pdata; + ret = tegra_asoc_utils_init(); if (ret) goto err_free_harmony; @@ -208,6 +255,7 @@ static int __devexit tegra_snd_harmony_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card); + struct harmony_audio_platform_data *pdata = harmony->pdata; snd_soc_unregister_card(card); @@ -217,6 +265,9 @@ static int __devexit tegra_snd_harmony_remove(struct platform_device *pdev) tegra_asoc_utils_fini(); + if (harmony->gpio_spkr_en_requested) + gpio_free(pdata->gpio_spkr_en); + kfree(harmony); return 0; -- cgit v1.2.3 From bc72fe0c0e90b1ff9c12037321626aa5a4b061c3 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 28 Jan 2011 14:26:38 -0700 Subject: ASoC: Tegra: Harmony: Fix indentation issue. Indent with TABs not spaces. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/harmony.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index 0f5a4815093..b5311a3ee95 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -104,11 +104,12 @@ static int harmony_asoc_hw_params(struct snd_pcm_substream *substream, } if (mclk_change) { - err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, SND_SOC_CLOCK_IN); - if (err < 0) { - pr_err(PREFIX "codec_dai clock not set\n"); - return err; - } + err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + pr_err(PREFIX "codec_dai clock not set\n"); + return err; + } } return 0; -- cgit v1.2.3 From c244d477b7a5e0060b05d6ef90b7b29ef72a2188 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 28 Jan 2011 14:26:39 -0700 Subject: ASoC: Tegra: Harmony: Use dev_err not pr_err Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/harmony.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index b5311a3ee95..76793a93c13 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -48,7 +48,6 @@ #include "tegra_asoc_utils.h" #define DRV_NAME "tegra-snd-harmony" -#define PREFIX DRV_NAME ": " struct tegra_harmony { struct harmony_audio_platform_data *pdata; @@ -61,6 +60,8 @@ static int harmony_asoc_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; int srate, mclk, mclk_change; int err; @@ -81,7 +82,7 @@ static int harmony_asoc_hw_params(struct snd_pcm_substream *substream, err = tegra_asoc_utils_set_rate(srate, mclk, &mclk_change); if (err < 0) { - pr_err(PREFIX "Can't configure clocks\n"); + dev_err(card->dev, "Can't configure clocks\n"); return err; } @@ -90,7 +91,7 @@ static int harmony_asoc_hw_params(struct snd_pcm_substream *substream, SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (err < 0) { - pr_err(PREFIX "codec_dai fmt not set\n"); + dev_err(card->dev, "codec_dai fmt not set\n"); return err; } @@ -99,7 +100,7 @@ static int harmony_asoc_hw_params(struct snd_pcm_substream *substream, SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (err < 0) { - pr_err(PREFIX "cpu_dai fmt not set\n"); + dev_err(card->dev, "cpu_dai fmt not set\n"); return err; } @@ -107,7 +108,7 @@ static int harmony_asoc_hw_params(struct snd_pcm_substream *substream, err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, SND_SOC_CLOCK_IN); if (err < 0) { - pr_err(PREFIX "codec_dai clock not set\n"); + dev_err(card->dev, "codec_dai clock not set\n"); return err; } } -- cgit v1.2.3 From d64e57cef0436833cfc9620e350ec6f5b041c9a9 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 28 Jan 2011 14:26:40 -0700 Subject: ASoC: Tegra: utils: Don't use global variables Instead, have the machine driver provide storage for the utility data somehow. For Harmony in particular, store this within struct tegra_harmony, itself referenced by snd_soc_card's drvdata. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/harmony.c | 11 +++-- sound/soc/tegra/tegra_asoc_utils.c | 91 ++++++++++++++++++-------------------- sound/soc/tegra/tegra_asoc_utils.h | 20 +++++++-- 3 files changed, 68 insertions(+), 54 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index 76793a93c13..d1faa63af03 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -50,6 +50,7 @@ #define DRV_NAME "tegra-snd-harmony" struct tegra_harmony { + struct tegra_asoc_utils_data util_data; struct harmony_audio_platform_data *pdata; int gpio_spkr_en_requested; }; @@ -62,6 +63,7 @@ static int harmony_asoc_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = codec->card; + struct tegra_harmony *harmony = snd_soc_card_get_drvdata(card); int srate, mclk, mclk_change; int err; @@ -80,7 +82,8 @@ static int harmony_asoc_hw_params(struct snd_pcm_substream *substream, while (mclk < 6000000) mclk *= 2; - err = tegra_asoc_utils_set_rate(srate, mclk, &mclk_change); + err = tegra_asoc_utils_set_rate(&harmony->util_data, srate, mclk, + &mclk_change); if (err < 0) { dev_err(card->dev, "Can't configure clocks\n"); return err; @@ -226,7 +229,7 @@ static __devinit int tegra_snd_harmony_probe(struct platform_device *pdev) harmony->pdata = pdata; - ret = tegra_asoc_utils_init(); + ret = tegra_asoc_utils_init(&harmony->util_data, &pdev->dev); if (ret) goto err_free_harmony; @@ -247,7 +250,7 @@ err_clear_drvdata: snd_soc_card_set_drvdata(card, NULL); platform_set_drvdata(pdev, NULL); card->dev = NULL; - tegra_asoc_utils_fini(); + tegra_asoc_utils_fini(&harmony->util_data); err_free_harmony: kfree(harmony); return ret; @@ -265,7 +268,7 @@ static int __devexit tegra_snd_harmony_remove(struct platform_device *pdev) platform_set_drvdata(pdev, NULL); card->dev = NULL; - tegra_asoc_utils_fini(); + tegra_asoc_utils_fini(&harmony->util_data); if (harmony->gpio_spkr_en_requested) gpio_free(pdata->gpio_spkr_en); diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index cfe2ea890dc..cb4fc13c7d2 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -21,20 +21,14 @@ */ #include +#include #include #include #include "tegra_asoc_utils.h" -#define PREFIX "ASoC Tegra: " - -static struct clk *clk_pll_a; -static struct clk *clk_pll_a_out0; -static struct clk *clk_cdev1; - -static int set_baseclock, set_mclk; - -int tegra_asoc_utils_set_rate(int srate, int mclk, int *mclk_change) +int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, + int mclk, int *mclk_change) { int new_baseclock; int err; @@ -58,95 +52,98 @@ int tegra_asoc_utils_set_rate(int srate, int mclk, int *mclk_change) return -EINVAL; } - *mclk_change = ((new_baseclock != set_baseclock) || - (mclk != set_mclk)); + *mclk_change = ((new_baseclock != data->set_baseclock) || + (mclk != data->set_mclk)); if (!*mclk_change) return 0; - set_baseclock = 0; - set_mclk = 0; + data->set_baseclock = 0; + data->set_mclk = 0; - clk_disable(clk_cdev1); - clk_disable(clk_pll_a_out0); - clk_disable(clk_pll_a); + clk_disable(data->clk_cdev1); + clk_disable(data->clk_pll_a_out0); + clk_disable(data->clk_pll_a); - err = clk_set_rate(clk_pll_a, new_baseclock); + err = clk_set_rate(data->clk_pll_a, new_baseclock); if (err) { - pr_err(PREFIX "Can't set pll_a rate: %d\n", err); + dev_err(data->dev, "Can't set pll_a rate: %d\n", err); return err; } - err = clk_set_rate(clk_pll_a_out0, mclk); + err = clk_set_rate(data->clk_pll_a_out0, mclk); if (err) { - pr_err(PREFIX "Can't set pll_a_out0 rate: %d\n", err); + dev_err(data->dev, "Can't set pll_a_out0 rate: %d\n", err); return err; } /* Don't set cdev1 rate; its locked to pll_a_out0 */ - err = clk_enable(clk_pll_a); + err = clk_enable(data->clk_pll_a); if (err) { - pr_err(PREFIX "Can't enable pll_a: %d\n", err); + dev_err(data->dev, "Can't enable pll_a: %d\n", err); return err; } - err = clk_enable(clk_pll_a_out0); + err = clk_enable(data->clk_pll_a_out0); if (err) { - pr_err(PREFIX "Can't enable pll_a_out0: %d\n", err); + dev_err(data->dev, "Can't enable pll_a_out0: %d\n", err); return err; } - err = clk_enable(clk_cdev1); + err = clk_enable(data->clk_cdev1); if (err) { - pr_err(PREFIX "Can't enable cdev1: %d\n", err); + dev_err(data->dev, "Can't enable cdev1: %d\n", err); return err; } - set_baseclock = new_baseclock; - set_mclk = mclk; + data->set_baseclock = new_baseclock; + data->set_mclk = mclk; return 0; } -int tegra_asoc_utils_init(void) +int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, + struct device *dev) { int ret; - clk_pll_a = clk_get_sys(NULL, "pll_a"); - if (IS_ERR(clk_pll_a)) { - pr_err(PREFIX "Can't retrieve clk pll_a\n"); - ret = PTR_ERR(clk_pll_a); + data->dev = dev; + + data->clk_pll_a = clk_get_sys(NULL, "pll_a"); + if (IS_ERR(data->clk_pll_a)) { + dev_err(data->dev, "Can't retrieve clk pll_a\n"); + ret = PTR_ERR(data->clk_pll_a); goto err; } - clk_pll_a_out0 = clk_get_sys(NULL, "pll_a_out0"); - if (IS_ERR(clk_pll_a_out0)) { - pr_err(PREFIX "Can't retrieve clk pll_a_out0\n"); - ret = PTR_ERR(clk_pll_a_out0); + data->clk_pll_a_out0 = clk_get_sys(NULL, "pll_a_out0"); + if (IS_ERR(data->clk_pll_a_out0)) { + dev_err(data->dev, "Can't retrieve clk pll_a_out0\n"); + ret = PTR_ERR(data->clk_pll_a_out0); goto err_put_pll_a; } - clk_cdev1 = clk_get_sys(NULL, "cdev1"); - if (IS_ERR(clk_cdev1)) { - pr_err(PREFIX "Can't retrieve clk cdev1\n"); - ret = PTR_ERR(clk_cdev1); + data->clk_cdev1 = clk_get_sys(NULL, "cdev1"); + if (IS_ERR(data->clk_cdev1)) { + dev_err(data->dev, "Can't retrieve clk cdev1\n"); + ret = PTR_ERR(data->clk_cdev1); goto err_put_pll_a_out0; } return 0; err_put_pll_a_out0: - clk_put(clk_pll_a_out0); + clk_put(data->clk_pll_a_out0); err_put_pll_a: - clk_put(clk_pll_a); + clk_put(data->clk_pll_a); err: return ret; } -void tegra_asoc_utils_fini(void) +void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data) { - clk_put(clk_cdev1); - clk_put(clk_pll_a_out0); - clk_put(clk_pll_a); + clk_put(data->clk_cdev1); + clk_put(data->clk_pll_a_out0); + clk_put(data->clk_pll_a); } diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h index 855f8f6e44c..bbba7afdfc2 100644 --- a/sound/soc/tegra/tegra_asoc_utils.h +++ b/sound/soc/tegra/tegra_asoc_utils.h @@ -23,9 +23,23 @@ #ifndef __TEGRA_ASOC_UTILS_H__ #define __TEGRA_ASOC_UTILS_H_ -int tegra_asoc_utils_set_rate(int srate, int mclk_rate, int *mclk_change); -int tegra_asoc_utils_init(void); -void tegra_asoc_utils_fini(void); +struct clk; +struct device; + +struct tegra_asoc_utils_data { + struct device *dev; + struct clk *clk_pll_a; + struct clk *clk_pll_a_out0; + struct clk *clk_cdev1; + int set_baseclock; + int set_mclk; +}; + +int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, + int mclk, int *mclk_change); +int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, + struct device *dev); +void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data); #endif -- cgit v1.2.3 From 713dce4e0b3dfd9d2d159d2fc6ec191916279bc5 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 28 Jan 2011 14:26:41 -0700 Subject: ASoC: Tegra: I2S: Use dev_err not pr_err Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 6d668785e9a..870ee361f75 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -386,7 +386,7 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev) snprintf(clk_name, sizeof(clk_name), DRV_NAME ".%d", pdev->id); i2s->clk_i2s = clk_get_sys(clk_name, NULL); if (IS_ERR(i2s->clk_i2s)) { - pr_err("Can't retrieve i2s clock\n"); + dev_err(&pdev->dev, "Can't retrieve i2s clock\n"); ret = PTR_ERR(i2s->clk_i2s); goto err_free; } -- cgit v1.2.3 From c800587f658d787a1e92385fdde0e3b5e0856f37 Mon Sep 17 00:00:00 2001 From: H Hartley Sweeten Date: Fri, 28 Jan 2011 12:00:18 -0700 Subject: ASoC: neo1973_wm8753 audio support does not require scoop This driver does not use any of the functionality provided by the scoop hardware. Remove the unneeded header. Signed-off-by: H Hartley Sweeten Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/samsung/neo1973_wm8753.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index d20815d5ab2..d3cd6888a81 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -24,7 +24,6 @@ #include #include -#include #include #include #include -- cgit v1.2.3 From d71bb810be602b7194d0dc336854e2f4a3d44492 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 31 Jan 2011 13:41:03 +0000 Subject: ASoC: Accept any logical value for WM8962 GPIO set() Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 5c7b730a864..3b71dd65c96 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3635,7 +3635,7 @@ static void wm8962_gpio_set(struct gpio_chip *chip, unsigned offset, int value) struct snd_soc_codec *codec = wm8962->codec; snd_soc_update_bits(codec, WM8962_GPIO_BASE + offset, - WM8962_GP2_LVL, value << WM8962_GP2_LVL_SHIFT); + WM8962_GP2_LVL, !!value << WM8962_GP2_LVL_SHIFT); } static int wm8962_gpio_direction_out(struct gpio_chip *chip, -- cgit v1.2.3 From c8059930f0aafb77d94d268dfb4fceaa15cd2ad9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 31 Jan 2011 13:41:17 +0000 Subject: ASoC: Accept any logical value WM8903 GPIO set() Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 9c4f2c4febc..3d4c55f3c7b 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1702,7 +1702,8 @@ static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value) struct snd_soc_codec *codec = wm8903->codec; snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - WM8903_GP1_LVL_MASK, value << WM8903_GP1_LVL_SHIFT); + WM8903_GP1_LVL_MASK, + !!value << WM8903_GP1_LVL_SHIFT); } static struct gpio_chip wm8903_template_chip = { -- cgit v1.2.3 From 8cda975174945e3b7119d48341052d0dc0c87cf2 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 31 Jan 2011 14:43:48 +0200 Subject: ASoC: omap: rx51: Add earphone support Earphone in Nokia RX-51/N900 is connected to left HP output of B part of the TLV320AIC34 dual codec. In RX-51 the codec A is used as a traditional codec and the codec B as an auxiliary device. Audio from codec A goes via the codec B to earphone: MONO_LOUT of A -> LINE2R of B (B interconnects) -> HPLOUT of B -> Earphone. Take earphone into use by utilizing the recent ASoC auxiliary and cross-device support. Signed-off-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/rx51.c | 50 ++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 50 insertions(+) (limited to 'sound') diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 251afbe303c..2d7d4115cb7 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -242,6 +242,10 @@ static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", rx51_hp_event), }; +static const struct snd_soc_dapm_widget aic34_dapm_widgetsb[] = { + SND_SOC_DAPM_SPK("Earphone", NULL), +}; + static const struct snd_soc_dapm_route audio_map[] = { {"Ext Spk", NULL, "HPLOUT"}, {"Ext Spk", NULL, "HPROUT"}, @@ -252,6 +256,11 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Mic Bias 2V", NULL, "DMic"}, }; +static const struct snd_soc_dapm_route audio_mapb[] = { + {"b LINE2R", NULL, "MONO_LOUT"}, + {"Earphone", NULL, "b HPLOUT"}, +}; + static const char *spk_function[] = {"Off", "On"}; static const char *input_function[] = {"ADC", "Digital Mic"}; static const char *jack_function[] = {"Off", "TV-OUT", "Headphone"}; @@ -271,6 +280,10 @@ static const struct snd_kcontrol_new aic34_rx51_controls[] = { rx51_get_jack, rx51_set_jack), }; +static const struct snd_kcontrol_new aic34_rx51_controlsb[] = { + SOC_DAPM_PIN_SWITCH("Earphone"), +}; + static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; @@ -314,6 +327,24 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) return err; } +static int rx51_aic34b_init(struct snd_soc_dapm_context *dapm) +{ + int err; + + err = snd_soc_add_controls(dapm->codec, aic34_rx51_controlsb, + ARRAY_SIZE(aic34_rx51_controlsb)); + if (err < 0) + return err; + + err = snd_soc_dapm_new_controls(dapm, aic34_dapm_widgetsb, + ARRAY_SIZE(aic34_dapm_widgetsb)); + if (err < 0) + return 0; + + return snd_soc_dapm_add_routes(dapm, audio_mapb, + ARRAY_SIZE(audio_mapb)); +} + /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link rx51_dai[] = { { @@ -328,11 +359,30 @@ static struct snd_soc_dai_link rx51_dai[] = { }, }; +struct snd_soc_aux_dev rx51_aux_dev[] = { + { + .name = "TLV320AIC34b", + .codec_name = "tlv320aic3x-codec.2-0019", + .init = rx51_aic34b_init, + }, +}; + +static struct snd_soc_codec_conf rx51_codec_conf[] = { + { + .dev_name = "tlv320aic3x-codec.2-0019", + .name_prefix = "b", + }, +}; + /* Audio card */ static struct snd_soc_card rx51_sound_card = { .name = "RX-51", .dai_link = rx51_dai, .num_links = ARRAY_SIZE(rx51_dai), + .aux_dev = rx51_aux_dev, + .num_aux_devs = ARRAY_SIZE(rx51_aux_dev), + .codec_conf = rx51_codec_conf, + .num_configs = ARRAY_SIZE(rx51_codec_conf), }; static struct platform_device *rx51_snd_device; -- cgit v1.2.3 From f9eabc3deedccf545bd73fd72b3af2759d44f6f3 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Mon, 31 Jan 2011 11:00:17 -0700 Subject: ASoC: Tegra: Harmony: Remove redundant !! gpio_set_value* should accept logic values not just 0 or 1. The WM8903 GPIO driver has been fixed to work this way, so remove the redundant !! previously required when it didn't accept values >1. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/harmony.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index d1faa63af03..11e2cb82566 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -132,7 +132,7 @@ static int harmony_event_int_spk(struct snd_soc_dapm_widget *w, struct harmony_audio_platform_data *pdata = harmony->pdata; gpio_set_value_cansleep(pdata->gpio_spkr_en, - !!SND_SOC_DAPM_EVENT_ON(event)); + SND_SOC_DAPM_EVENT_ON(event)); return 0; } -- cgit v1.2.3 From 2bc9a81e2aa3c95b1e709f3109c3bbc6d969cc06 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Tue, 1 Feb 2011 12:24:08 +0000 Subject: ASoC: soc-core: Ensure codec_reg has fixed length fields Make the format of the codec_reg file more easily parsable. Remove the header field which gives the codec name. These changes are important when it comes to extend the debugfs codec_reg file to dump more than PAGE_SIZE bytes to make it easier to calculate offsets within the file. We still need to handle the case when the snd_soc_read() call fails and is outputted. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 27 ++++++++++++++++++++++++--- 1 file changed, 24 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3feddd91b97..205cbd7b149 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -69,10 +69,32 @@ static int pmdown_time = 5000; module_param(pmdown_time, int, 0); MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); +/* returns the minimum number of bytes needed to represent + * a particular given value */ +static int min_bytes_needed(unsigned long val) +{ + int c = 0; + int i; + + for (i = (sizeof val * 8) - 1; i >= 0; --i, ++c) + if (val & (1UL << i)) + break; + c = (sizeof val * 8) - c; + if (!c || (c % 8)) + c = (c + 8) / 8; + else + c /= 8; + return c; +} + /* codec register dump */ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) { int ret, i, step = 1, count = 0; + int wordsize, regsize; + + wordsize = codec->driver->reg_word_size * 2; + regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; if (!codec->driver->reg_cache_size) return 0; @@ -80,12 +102,11 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) if (codec->driver->reg_cache_step) step = codec->driver->reg_cache_step; - count += sprintf(buf, "%s registers\n", codec->name); for (i = 0; i < codec->driver->reg_cache_size; i += step) { if (codec->readable_register && !codec->readable_register(codec, i)) continue; - count += sprintf(buf + count, "%2x: ", i); + count += sprintf(buf + count, "%.*x: ", regsize, i); if (count >= PAGE_SIZE - 1) break; @@ -101,7 +122,7 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) if (ret >= 0) count += snprintf(buf + count, PAGE_SIZE - count, - "%4x", ret); + "%.*x", wordsize, ret); else count += snprintf(buf + count, PAGE_SIZE - count, -- cgit v1.2.3 From 3d8b2ce01b2de7c3c5cbf18b9b97d26cfaf11c14 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 31 Jan 2011 20:14:38 +0000 Subject: ASoC: Use snd_pcm_format_width() in snd_soc_params_to_frame_size() Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-utils.c | 23 +++-------------------- 1 file changed, 3 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 1d07b931f3d..3f45e6a439b 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -28,26 +28,9 @@ int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params) { int sample_size; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - case SNDRV_PCM_FORMAT_S16_BE: - sample_size = 16; - break; - case SNDRV_PCM_FORMAT_S20_3LE: - case SNDRV_PCM_FORMAT_S20_3BE: - sample_size = 20; - break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S24_BE: - sample_size = 24; - break; - case SNDRV_PCM_FORMAT_S32_LE: - case SNDRV_PCM_FORMAT_S32_BE: - sample_size = 32; - break; - default: - return -ENOTSUPP; - } + sample_size = snd_pcm_format_width(params_format(params)); + if (sample_size < 0) + return sample_size; return snd_soc_calc_frame_size(sample_size, params_channels(params), 1); -- cgit v1.2.3 From 88ee1c611d0ac0f771212f907bdfb33905a5dc89 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Feb 2011 10:43:26 +0000 Subject: ASoC: Update PM ifdefs for exported suspend/resume Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 205cbd7b149..10bc3c23b52 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1801,7 +1801,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } card->snd_card->dev = card->dev; -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP /* deferred resume work */ INIT_WORK(&card->deferred_resume_work, soc_resume_deferred); #endif -- cgit v1.2.3 From a591e969fae8b79484cb94a12aceb9799e06ac22 Mon Sep 17 00:00:00 2001 From: Sven Neumann Date: Mon, 31 Jan 2011 16:10:50 +0100 Subject: ASoC: PXA: formatting Signed-off-by: Sven Neumann Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/pxa/raumfeld.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index db1dd560a58..2afabaf5949 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -229,19 +229,19 @@ static struct snd_soc_dai_link raumfeld_dai[] = { { .name = "ak4104", .stream_name = "Playback", - .cpu_dai_name = "pxa-ssp-dai.1", - .codec_dai_name = "ak4104-hifi", - .platform_name = "pxa-pcm-audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "ak4104-hifi", + .platform_name = "pxa-pcm-audio", .ops = &raumfeld_ak4104_ops, - .codec_name = "ak4104-codec.0", + .codec_name = "ak4104-codec.0", }, { .name = "CS4270", .stream_name = "CS4270", - .cpu_dai_name = "pxa-ssp-dai.0", - .platform_name = "pxa-pcm-audio", - .codec_dai_name = "cs4270-hifi", - .codec_name = "cs4270-codec.0-0048", + .cpu_dai_name = "pxa-ssp-dai.0", + .platform_name = "pxa-pcm-audio", + .codec_dai_name = "cs4270-hifi", + .codec_name = "cs4270-codec.0-0048", .ops = &raumfeld_cs4270_ops, },}; -- cgit v1.2.3 From 13fd179f14712cc89b5b3812673891b96e0ab3a2 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Wed, 2 Feb 2011 13:58:58 +0000 Subject: ASoC: soc-core: Support debugfs entries larger than PAGE_SIZE bytes For some codecs with large register maps, it was not possible to dump all registers via the codec_reg file but only up to PAGE_SIZE bytes. This patch fixes this problem. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 110 ++++++++++++++++++++++++++++++++++----------------- 1 file changed, 73 insertions(+), 37 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 10bc3c23b52..0c789f9150c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -87,15 +87,56 @@ static int min_bytes_needed(unsigned long val) return c; } +/* fill buf which is 'len' bytes with a formatted + * string of the form 'reg: value\n' */ +static int format_register_str(struct snd_soc_codec *codec, + unsigned int reg, char *buf, size_t len) +{ + int wordsize = codec->driver->reg_word_size * 2; + int regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; + int ret; + char tmpbuf[len + 1]; + char regbuf[regsize + 1]; + + /* since tmpbuf is allocated on the stack, warn the callers if they + * try to abuse this function */ + WARN_ON(len > 63); + + /* +2 for ': ' and + 1 for '\n' */ + if (wordsize + regsize + 2 + 1 != len) + return -EINVAL; + + ret = snd_soc_read(codec , reg); + if (ret < 0) { + memset(regbuf, 'X', regsize); + regbuf[regsize] = '\0'; + } else { + snprintf(regbuf, regsize + 1, "%.*x", regsize, ret); + } + + /* prepare the buffer */ + snprintf(tmpbuf, len + 1, "%.*x: %s\n", wordsize, reg, regbuf); + /* copy it back to the caller without the '\0' */ + memcpy(buf, tmpbuf, len); + + return 0; +} + /* codec register dump */ -static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) +static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf, + size_t count, loff_t pos) { - int ret, i, step = 1, count = 0; + int i, step = 1; int wordsize, regsize; + int len; + size_t total = 0; + loff_t p = 0; wordsize = codec->driver->reg_word_size * 2; regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; + len = wordsize + regsize + 2 + 1; + if (!codec->driver->reg_cache_size) return 0; @@ -105,51 +146,34 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) for (i = 0; i < codec->driver->reg_cache_size; i += step) { if (codec->readable_register && !codec->readable_register(codec, i)) continue; - - count += sprintf(buf + count, "%.*x: ", regsize, i); - if (count >= PAGE_SIZE - 1) - break; - if (codec->driver->display_register) { count += codec->driver->display_register(codec, buf + count, PAGE_SIZE - count, i); } else { - /* If the read fails it's almost certainly due to - * the register being volatile and the device being - * powered off. - */ - ret = snd_soc_read(codec, i); - if (ret >= 0) - count += snprintf(buf + count, - PAGE_SIZE - count, - "%.*x", wordsize, ret); - else - count += snprintf(buf + count, - PAGE_SIZE - count, - "", ret); + /* only support larger than PAGE_SIZE bytes debugfs + * entries for the default case */ + if (p >= pos) { + if (total + len >= count - 1) + break; + format_register_str(codec, i, buf + total, len); + total += len; + } + p += len; } - - if (count >= PAGE_SIZE - 1) - break; - - count += snprintf(buf + count, PAGE_SIZE - count, "\n"); - if (count >= PAGE_SIZE - 1) - break; } - /* Truncate count; min() would cause a warning */ - if (count >= PAGE_SIZE) - count = PAGE_SIZE - 1; + total = min(total, count - 1); - return count; + return total; } + static ssize_t codec_reg_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_soc_pcm_runtime *rtd = container_of(dev, struct snd_soc_pcm_runtime, dev); - return soc_codec_reg_show(rtd->codec, buf); + return soc_codec_reg_show(rtd->codec, buf, PAGE_SIZE, 0); } static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); @@ -188,16 +212,28 @@ static int codec_reg_open_file(struct inode *inode, struct file *file) } static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, - size_t count, loff_t *ppos) + size_t count, loff_t *ppos) { ssize_t ret; struct snd_soc_codec *codec = file->private_data; - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + char *buf; + + if (*ppos < 0 || !count) + return -EINVAL; + + buf = kmalloc(count, GFP_KERNEL); if (!buf) return -ENOMEM; - ret = soc_codec_reg_show(codec, buf); - if (ret >= 0) - ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); + + ret = soc_codec_reg_show(codec, buf, count, *ppos); + if (ret >= 0) { + if (copy_to_user(user_buf, buf, ret)) { + kfree(buf); + return -EFAULT; + } + *ppos += ret; + } + kfree(buf); return ret; } -- cgit v1.2.3 From c45bfccfa299a9b3a20373963c286b4e9bd3c40f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 3 Feb 2011 14:07:15 +0000 Subject: ASoC: Sort ALC5623 in Kconfig and Makefile Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 83b7accd703..ae10507dd2e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -91,6 +91,7 @@ obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o +obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o @@ -102,7 +103,6 @@ obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o -obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o -- cgit v1.2.3 From 10b6089a69ed7d0fa4b8f30160c3efbf3342cc9b Mon Sep 17 00:00:00 2001 From: Mika Westerberg Date: Fri, 4 Feb 2011 12:22:55 +0200 Subject: ASoC: ep93xx-ac97: remove extra empty line Signed-off-by: Mika Westerberg Signed-off-by: Mark Brown --- sound/soc/ep93xx/ep93xx-ac97.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c index 68a0bae1208..104e95cda0a 100644 --- a/sound/soc/ep93xx/ep93xx-ac97.c +++ b/sound/soc/ep93xx/ep93xx-ac97.c @@ -253,7 +253,6 @@ static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream, struct ep93xx_ac97_info *info = snd_soc_dai_get_drvdata(dai); unsigned v = 0; - switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: -- cgit v1.2.3 From 480b08d0bb1aab29de2545625767dac55c7dcb59 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Sun, 6 Feb 2011 15:35:48 +0530 Subject: ASoC: mid-x86: Fix dependency on intel_sst driver Enabling medfield asoc driver causes compliation error when intel_sst is not selected ERROR: "register_sst_card" [sound/soc/mid-x86/snd-soc-sst-platform.ko] undefined! This patch puts proper dependency to elimate build error Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Reported-by: Andrew Morton Signed-off-by: Mark Brown --- sound/soc/mid-x86/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mid-x86/Kconfig b/sound/soc/mid-x86/Kconfig index 1ad75383635..29350428f1c 100644 --- a/sound/soc/mid-x86/Kconfig +++ b/sound/soc/mid-x86/Kconfig @@ -1,6 +1,7 @@ config SND_MFLD_MACHINE tristate "SOC Machine Audio driver for Intel Medfield MID platform" depends on INTEL_SCU_IPC + depends on SND_INTEL_SST select SND_SOC_SN95031 select SND_SST_PLATFORM help @@ -11,4 +12,3 @@ config SND_MFLD_MACHINE config SND_SST_PLATFORM tristate - depends on SND_INTEL_SST -- cgit v1.2.3 From 338ee25393a5627e8ded5819147f98b919656ce9 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 6 Feb 2011 10:04:11 +0100 Subject: ASoC: codecs: wm8753: Fix DAI mode switching The wm8753 codec supports switching between different DAI modes. The current drivers tries to implement this by changing the DAI driver at runtime. But to properly work this would require support from the ASoC core. So this patch takes a different approch on how the DAI mode switching is implemented. The only difference, from a driver point of view, between the different DAI modes is how to program the DAI format to the hardware. So what this patch is, it stores the current format for each DAI in the drivers private struct and when the DAI mode is changed the format gets simply reprogrammed according to the new DAI mode. Futhermore this patch restricts the changing of the DAI format to when the codec is inactive. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 296 +++++++++++++++++++--------------------------- 1 file changed, 121 insertions(+), 175 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 79b02ae125c..3f09deea8d9 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -55,8 +55,10 @@ static int caps_charge = 2000; module_param(caps_charge, int, 0); MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)"); -static void wm8753_set_dai_mode(struct snd_soc_codec *codec, - struct snd_soc_dai *dai, unsigned int hifi); +static int wm8753_hifi_write_dai_fmt(struct snd_soc_codec *codec, + unsigned int fmt); +static int wm8753_voice_write_dai_fmt(struct snd_soc_codec *codec, + unsigned int fmt); /* * wm8753 register cache @@ -87,6 +89,10 @@ struct wm8753_priv { enum snd_soc_control_type control_type; unsigned int sysclk; unsigned int pcmclk; + + unsigned int voice_fmt; + unsigned int hifi_fmt; + int dai_func; }; @@ -170,9 +176,9 @@ static int wm8753_get_dai(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int mode = snd_soc_read(codec, WM8753_IOCTL); + struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); - ucontrol->value.integer.value[0] = (mode & 0xc) >> 2; + ucontrol->value.integer.value[0] = wm8753->dai_func; return 0; } @@ -180,16 +186,26 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int mode = snd_soc_read(codec, WM8753_IOCTL); struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); + u16 ioctl; + + if (codec->active) + return -EBUSY; + + ioctl = snd_soc_read(codec, WM8753_IOCTL); + + wm8753->dai_func = ucontrol->value.integer.value[0]; + + if (((ioctl >> 2) & 0x3) == wm8753->dai_func) + return 1; + + ioctl = (ioctl & 0x1f3) | (wm8753->dai_func << 2); + snd_soc_write(codec, WM8753_IOCTL, ioctl); - if (((mode & 0xc) >> 2) == ucontrol->value.integer.value[0]) - return 0; - mode &= 0xfff3; - mode |= (ucontrol->value.integer.value[0] << 2); + wm8753_hifi_write_dai_fmt(codec, wm8753->hifi_fmt); + wm8753_voice_write_dai_fmt(codec, wm8753->voice_fmt); - wm8753->dai_func = ucontrol->value.integer.value[0]; return 1; } @@ -828,10 +844,9 @@ static int wm8753_set_dai_sysclk(struct snd_soc_dai *codec_dai, /* * Set's ADC and Voice DAC format. */ -static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 voice = snd_soc_read(codec, WM8753_PCM) & 0x01ec; /* interface format */ @@ -858,13 +873,6 @@ static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } -static int wm8753_pcm_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - wm8753_set_dai_mode(dai->codec, dai, 0); - return 0; -} - /* * Set PCM DAI bit size and sample rate. */ @@ -905,10 +913,9 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, /* * Set's PCM dai fmt and BCLK. */ -static int wm8753_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 voice, ioctl; voice = snd_soc_read(codec, WM8753_PCM) & 0x011f; @@ -999,10 +1006,9 @@ static int wm8753_set_dai_clkdiv(struct snd_soc_dai *codec_dai, /* * Set's HiFi DAC format. */ -static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 hifi = snd_soc_read(codec, WM8753_HIFI) & 0x01e0; /* interface format */ @@ -1032,10 +1038,9 @@ static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai, /* * Set's I2S DAI format. */ -static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_i2s_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 ioctl, hifi; hifi = snd_soc_read(codec, WM8753_HIFI) & 0x011f; @@ -1098,13 +1103,6 @@ static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } -static int wm8753_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - wm8753_set_dai_mode(dai->codec, dai, 1); - return 0; -} - /* * Set PCM DAI bit size and sample rate. */ @@ -1147,61 +1145,117 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8753_mode1v_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 clock; /* set clk source as pcmclk */ clock = snd_soc_read(codec, WM8753_CLOCK) & 0xfffb; snd_soc_write(codec, WM8753_CLOCK, clock); - if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0) - return -EINVAL; - return wm8753_pcm_set_dai_fmt(codec_dai, fmt); + return wm8753_vdac_adc_set_dai_fmt(codec, fmt); } -static int wm8753_mode1h_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0) - return -EINVAL; - return wm8753_i2s_set_dai_fmt(codec_dai, fmt); + return wm8753_hdac_set_dai_fmt(codec, fmt); } -static int wm8753_mode2_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 clock; /* set clk source as pcmclk */ clock = snd_soc_read(codec, WM8753_CLOCK) & 0xfffb; snd_soc_write(codec, WM8753_CLOCK, clock); - if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0) - return -EINVAL; - return wm8753_i2s_set_dai_fmt(codec_dai, fmt); + return wm8753_vdac_adc_set_dai_fmt(codec, fmt); } -static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec *codec, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; u16 clock; /* set clk source as mclk */ clock = snd_soc_read(codec, WM8753_CLOCK) & 0xfffb; snd_soc_write(codec, WM8753_CLOCK, clock | 0x4); - if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0) + if (wm8753_hdac_set_dai_fmt(codec, fmt) < 0) return -EINVAL; - if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0) - return -EINVAL; - return wm8753_i2s_set_dai_fmt(codec_dai, fmt); + return wm8753_vdac_adc_set_dai_fmt(codec, fmt); } +static int wm8753_hifi_write_dai_fmt(struct snd_soc_codec *codec, + unsigned int fmt) +{ + struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (wm8753->dai_func) { + case 0: + ret = wm8753_mode1h_set_dai_fmt(codec, fmt); + break; + case 1: + ret = wm8753_mode2_set_dai_fmt(codec, fmt); + break; + case 2: + case 3: + ret = wm8753_mode3_4_set_dai_fmt(codec, fmt); + break; + default: + break; + } + if (ret) + return ret; + + return wm8753_i2s_set_dai_fmt(codec, fmt); +} + +static int wm8753_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); + + wm8753->hifi_fmt = fmt; + + return wm8753_hifi_write_dai_fmt(codec, fmt); +}; + +static int wm8753_voice_write_dai_fmt(struct snd_soc_codec *codec, + unsigned int fmt) +{ + struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if (wm8753->dai_func != 0) + return 0; + + ret = wm8753_mode1v_set_dai_fmt(codec, fmt); + if (ret) + return ret; + ret = wm8753_pcm_set_dai_fmt(codec, fmt); + if (ret) + return ret; + + return 0; +}; + +static int wm8753_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); + + wm8753->voice_fmt = fmt; + + return wm8753_voice_write_dai_fmt(codec, fmt); +}; + static int wm8753_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; @@ -1268,57 +1322,25 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec, * 3. Voice disabled - HIFI over HIFI * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture */ -static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode1 = { - .startup = wm8753_i2s_startup, +static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode = { .hw_params = wm8753_i2s_hw_params, .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode1h_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, -}; - -static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode1 = { - .startup = wm8753_pcm_startup, - .hw_params = wm8753_pcm_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode1v_set_dai_fmt, + .set_fmt = wm8753_hifi_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, .set_pll = wm8753_set_dai_pll, .set_sysclk = wm8753_set_dai_sysclk, }; -static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode2 = { - .startup = wm8753_pcm_startup, +static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode = { .hw_params = wm8753_pcm_hw_params, .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode2_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, -}; - -static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode3 = { - .startup = wm8753_i2s_startup, - .hw_params = wm8753_i2s_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode3_4_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, -}; - -static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode4 = { - .startup = wm8753_i2s_startup, - .hw_params = wm8753_i2s_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode3_4_set_dai_fmt, + .set_fmt = wm8753_voice_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, .set_pll = wm8753_set_dai_pll, .set_sysclk = wm8753_set_dai_sysclk, }; -static struct snd_soc_dai_driver wm8753_all_dai[] = { +static struct snd_soc_dai_driver wm8753_dai[] = { /* DAI HiFi mode 1 */ { .name = "wm8753-hifi", .playback = { @@ -1326,14 +1348,16 @@ static struct snd_soc_dai_driver wm8753_all_dai[] = { .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, - .formats = WM8753_FORMATS}, + .formats = WM8753_FORMATS + }, .capture = { /* dummy for fast DAI switching */ .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, - .formats = WM8753_FORMATS}, - .ops = &wm8753_dai_ops_hifi_mode1, + .formats = WM8753_FORMATS + }, + .ops = &wm8753_dai_ops_hifi_mode, }, /* DAI Voice mode 1 */ { .name = "wm8753-voice", @@ -1342,97 +1366,19 @@ static struct snd_soc_dai_driver wm8753_all_dai[] = { .channels_min = 1, .channels_max = 1, .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 2, - .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .ops = &wm8753_dai_ops_voice_mode1, -}, -/* DAI HiFi mode 2 - dummy */ -{ .name = "wm8753-hifi", -}, -/* DAI Voice mode 2 */ -{ .name = "wm8753-voice", - .playback = { - .stream_name = "Voice Playback", - .channels_min = 1, - .channels_max = 1, - .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 2, - .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .ops = &wm8753_dai_ops_voice_mode2, -}, -/* DAI HiFi mode 3 */ -{ .name = "wm8753-hifi", - .playback = { - .stream_name = "HiFi Playback", - .channels_min = 1, - .channels_max = 2, - .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 2, - .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .ops = &wm8753_dai_ops_hifi_mode3, -}, -/* DAI Voice mode 3 - dummy */ -{ .name = "wm8753-voice", -}, -/* DAI HiFi mode 4 */ -{ .name = "wm8753-hifi", - .playback = { - .stream_name = "HiFi Playback", - .channels_min = 1, - .channels_max = 2, - .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, + .formats = WM8753_FORMATS, + }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, - .ops = &wm8753_dai_ops_hifi_mode4, -}, -/* DAI Voice mode 4 - dummy */ -{ .name = "wm8753-voice", -}, -}; - -static struct snd_soc_dai_driver wm8753_dai[] = { - { - .name = "wm8753-aif0", - }, - { - .name = "wm8753-aif1", + .formats = WM8753_FORMATS, }, + .ops = &wm8753_dai_ops_voice_mode, +}, }; -static void wm8753_set_dai_mode(struct snd_soc_codec *codec, - struct snd_soc_dai *dai, unsigned int hifi) -{ - struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); - - if (wm8753->dai_func < 4) { - if (hifi) - dai->driver = &wm8753_all_dai[wm8753->dai_func << 1]; - else - dai->driver = &wm8753_all_dai[(wm8753->dai_func << 1) + 1]; - } - snd_soc_write(codec, WM8753_IOCTL, wm8753->dai_func); -} - static void wm8753_work(struct work_struct *work) { struct snd_soc_dapm_context *dapm = -- cgit v1.2.3 From a98a0bc6c92eacd181417a9c0ccd2e8028066622 Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin Date: Thu, 3 Feb 2011 03:11:45 +0300 Subject: ASoC: CS4271: Move Chip Select control out of the CODEC code. Move Chip Select control out of the CODEC code for CS4271. Signed-off-by: Alexander Sverdlin Reviewed-by: H Hartley Sweeten Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 22 +++------------------- 1 file changed, 3 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 9c5b7db0ce6..1791796216c 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -441,22 +441,11 @@ static int cs4271_probe(struct snd_soc_codec *codec) struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; int ret; int gpio_nreset = -EINVAL; - int gpio_disable = -EINVAL; codec->control_data = cs4271->control_data; - if (cs4271plat) { - if (gpio_is_valid(cs4271plat->gpio_nreset)) - gpio_nreset = cs4271plat->gpio_nreset; - if (gpio_is_valid(cs4271plat->gpio_disable)) - gpio_disable = cs4271plat->gpio_disable; - } - - if (gpio_disable >= 0) - if (gpio_request(gpio_disable, "CS4271 Disable")) - gpio_disable = -EINVAL; - if (gpio_disable >= 0) - gpio_direction_output(gpio_disable, 0); + if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset)) + gpio_nreset = cs4271plat->gpio_nreset; if (gpio_nreset >= 0) if (gpio_request(gpio_nreset, "CS4271 Reset")) @@ -471,7 +460,6 @@ static int cs4271_probe(struct snd_soc_codec *codec) } cs4271->gpio_nreset = gpio_nreset; - cs4271->gpio_disable = gpio_disable; /* * In case of I2C, chip address specified in board data. @@ -509,10 +497,9 @@ static int cs4271_probe(struct snd_soc_codec *codec) static int cs4271_remove(struct snd_soc_codec *codec) { struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); - int gpio_nreset, gpio_disable; + int gpio_nreset; gpio_nreset = cs4271->gpio_nreset; - gpio_disable = cs4271->gpio_disable; if (gpio_is_valid(gpio_nreset)) { /* Set codec to the reset state */ @@ -520,9 +507,6 @@ static int cs4271_remove(struct snd_soc_codec *codec) gpio_free(gpio_nreset); } - if (gpio_is_valid(gpio_disable)) - gpio_free(gpio_disable); - return 0; }; -- cgit v1.2.3 From 46fdaa3bec1bcaa36759b1f5e5e27245ae465b8f Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Mon, 7 Feb 2011 22:01:41 +0300 Subject: ASoC: soc-cache: dereferencing before checking The patch c358e640a66 "ASoC: soc-cache: Add trace event for snd_soc_cache_sync()" introduced a dereference of "codec->cache_ops" before we had checked it for NULL. I pulled the check forward, and then pulled everything in an indent level. Signed-off-by: Dan Carpenter Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-cache.c | 25 ++++++++++++------------- 1 file changed, 12 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index db66dc44add..5d76da43b14 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1609,24 +1609,23 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) return 0; } + if (!codec->cache_ops || !codec->cache_ops->sync) + return -EINVAL; + if (codec->cache_ops->name) name = codec->cache_ops->name; else name = "unknown"; - if (codec->cache_ops && codec->cache_ops->sync) { - if (codec->cache_ops->name) - dev_dbg(codec->dev, "Syncing %s cache for %s codec\n", - codec->cache_ops->name, codec->name); - trace_snd_soc_cache_sync(codec, name, "start"); - ret = codec->cache_ops->sync(codec); - if (!ret) - codec->cache_sync = 0; - trace_snd_soc_cache_sync(codec, name, "end"); - return ret; - } - - return -EINVAL; + if (codec->cache_ops->name) + dev_dbg(codec->dev, "Syncing %s cache for %s codec\n", + codec->cache_ops->name, codec->name); + trace_snd_soc_cache_sync(codec, name, "start"); + ret = codec->cache_ops->sync(codec); + if (!ret) + codec->cache_sync = 0; + trace_snd_soc_cache_sync(codec, name, "end"); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_cache_sync); -- cgit v1.2.3 From f7d3e403d7de06f8a42a7f34950ae972547c6d59 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 3 Feb 2011 13:56:13 -0700 Subject: ASoC: Tegra: Harmony: Add headphone jack detection Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/harmony.c | 30 +++++++++++++++++++++++++++++- 1 file changed, 29 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index 11e2cb82566..fcd316e1c94 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -38,6 +38,7 @@ #include #include +#include #include #include #include @@ -123,6 +124,24 @@ static struct snd_soc_ops harmony_asoc_ops = { .hw_params = harmony_asoc_hw_params, }; +static struct snd_soc_jack harmony_hp_jack; + +static struct snd_soc_jack_pin harmony_hp_jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static struct snd_soc_jack_gpio harmony_hp_jack_gpios[] = { + { + .name = "headphone detect", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, + .invert = 1, + } +}; + static int harmony_event_int_spk(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { @@ -178,11 +197,20 @@ static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_add_routes(dapm, harmony_audio_map, ARRAY_SIZE(harmony_audio_map)); - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Int Spk"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); snd_soc_dapm_sync(dapm); + harmony_hp_jack_gpios[0].gpio = pdata->gpio_hp_det; + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, + &harmony_hp_jack); + snd_soc_jack_add_pins(&harmony_hp_jack, + ARRAY_SIZE(harmony_hp_jack_pins), + harmony_hp_jack_pins); + snd_soc_jack_add_gpios(&harmony_hp_jack, + ARRAY_SIZE(harmony_hp_jack_gpios), + harmony_hp_jack_gpios); + return 0; } -- cgit v1.2.3 From 3d8bc3901067700e790e8cc796aa438cffb5ac80 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 3 Feb 2011 13:56:14 -0700 Subject: ASoC: Tegra: Harmony: Add switch control for speaker Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/harmony.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index fcd316e1c94..be95405df3f 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -173,6 +173,10 @@ static const struct snd_soc_dapm_route harmony_audio_map[] = { {"IN1L", NULL, "Mic Bias"}, }; +static const struct snd_kcontrol_new harmony_controls[] = { + SOC_DAPM_PIN_SWITCH("Int Spk"), +}; + static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; @@ -191,13 +195,17 @@ static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd) gpio_direction_output(pdata->gpio_spkr_en, 0); + ret = snd_soc_add_controls(codec, harmony_controls, + ARRAY_SIZE(harmony_controls)); + if (ret < 0) + return ret; + snd_soc_dapm_new_controls(dapm, harmony_dapm_widgets, ARRAY_SIZE(harmony_dapm_widgets)); snd_soc_dapm_add_routes(dapm, harmony_audio_map, ARRAY_SIZE(harmony_audio_map)); - snd_soc_dapm_enable_pin(dapm, "Int Spk"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); snd_soc_dapm_sync(dapm); -- cgit v1.2.3 From 1e2f5932e472f11b987e8339ffc855aa00ecebf5 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 9 Feb 2011 21:44:32 +0530 Subject: ASoC: sn95031: Add jack support in the codec This patch adds support for jack detection and reporting in the codec It however is not fully functional as it doesn't measure adc to figure out what got inserted which will be added later Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 56 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sn95031.h | 9 ++++++++ 2 files changed, 65 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 40e285df9ae..b49d79017d3 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -34,6 +34,7 @@ #include #include #include +#include #include "sn95031.h" #define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100) @@ -649,6 +650,61 @@ struct snd_soc_dai_driver sn95031_dais[] = { }, }; +static inline void sn95031_disable_jack_btn(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_BTNCTRL2, 0x00); +} + +static inline void sn95031_enable_jack_btn(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_BTNCTRL1, 0x77); + snd_soc_write(codec, SN95031_BTNCTRL2, 0x01); +} + +static int sn95031_get_headset_state(struct snd_soc_jack *mfld_jack) +{ + /* Defaulting to HEADSET for now. + * will change after adding soc-jack detection apis */ + int jack_type = SND_JACK_HEADSET; + + pr_debug("jack type detected = %d\n", jack_type); + if (jack_type == SND_JACK_HEADSET) + sn95031_enable_jack_btn(mfld_jack->codec); + return jack_type; +} + +void sn95031_jack_detection(struct mfld_jack_data *jack_data) +{ + unsigned int status; + unsigned int mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_HEADSET; + + pr_debug("interrupt id read in sram = 0x%x\n", jack_data->intr_id); + if (jack_data->intr_id & 0x1) { + pr_debug("short_push detected\n"); + status = SND_JACK_HEADSET | SND_JACK_BTN_0; + } else if (jack_data->intr_id & 0x2) { + pr_debug("long_push detected\n"); + status = SND_JACK_HEADSET | SND_JACK_BTN_1; + } else if (jack_data->intr_id & 0x4) { + pr_debug("headset or headphones inserted\n"); + status = sn95031_get_headset_state(jack_data->mfld_jack); + } else if (jack_data->intr_id & 0x8) { + pr_debug("headset or headphones removed\n"); + status = 0; + sn95031_disable_jack_btn(jack_data->mfld_jack->codec); + } else { + pr_err("unidentified interrupt\n"); + return; + } + + snd_soc_jack_report(jack_data->mfld_jack, status, mask); + /*button pressed and released so we send explicit button release */ + if ((status & SND_JACK_BTN_0) | (status & SND_JACK_BTN_1)) + snd_soc_jack_report(jack_data->mfld_jack, + SND_JACK_HEADSET, mask); +} +EXPORT_SYMBOL_GPL(sn95031_jack_detection); + /* codec registration */ static int sn95031_codec_probe(struct snd_soc_codec *codec) { diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h index e2b17d908ae..2dbae614bac 100644 --- a/sound/soc/codecs/sn95031.h +++ b/sound/soc/codecs/sn95031.h @@ -96,4 +96,13 @@ #define SN95031_SSR5 0x384 #define SN95031_SSR6 0x385 +#define SN95031_AUDIO_GPIO_CTRL 0x070 +struct mfld_jack_data { + int intr_id; + int micbias_vol; + struct snd_soc_jack *mfld_jack; +}; + +extern void sn95031_jack_detection(struct mfld_jack_data *jack_data); + #endif -- cgit v1.2.3 From 42aee9b43e76645cda59bce30a101e7574ce913e Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 9 Feb 2011 21:44:33 +0530 Subject: ASoC: mfld_machine: Add support for jack detection This patch adds support for registering jack interupt and registering jack with core Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/mid-x86/mfld_machine.c | 203 +++++++++++++++++++++++++++++++++------ 1 file changed, 176 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 7925851a5de..45a00670802 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -27,18 +27,53 @@ #include #include #include +#include #include #include #include +#include #include "../codecs/sn95031.h" #define MID_MONO 1 #define MID_STEREO 2 #define MID_MAX_CAP 5 +#define MFLD_JACK_INSERT 0x04 + +enum soc_mic_bias_zones { + MFLD_MV_START = 0, + /* mic bias volutage range for Headphones*/ + MFLD_MV_HP = 400, + /* mic bias volutage range for American Headset*/ + MFLD_MV_AM_HS = 650, + /* mic bias volutage range for Headset*/ + MFLD_MV_HS = 2000, + MFLD_MV_UNDEFINED, +}; static unsigned int hs_switch; static unsigned int lo_dac; +struct mfld_mc_private { + struct platform_device *socdev; + void __iomem *int_base; + struct snd_soc_codec *codec; + u8 interrupt_status; +}; + +struct snd_soc_jack mfld_jack; + +/*Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin mfld_jack_pins[] = { + { + .pin = "Headphones", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "AMIC1", + .mask = SND_JACK_MICROPHONE, + }, +}; + /* sound card controls */ static const char *headset_switch_text[] = {"Earpiece", "Headset"}; @@ -67,13 +102,11 @@ static int headset_set_switch(struct snd_kcontrol *kcontrol, if (ucontrol->value.integer.value[0]) { pr_debug("hs_set HS path\n"); - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "Headphones"); snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); } else { pr_debug("hs_set EP path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); } snd_soc_dapm_sync(&codec->dapm); @@ -91,12 +124,10 @@ static void lo_enable_out_pins(struct snd_soc_codec *codec) snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT"); snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT"); if (hs_switch) { - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_enable_pin(&codec->dapm, "Headphones"); snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); } else { - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); } } @@ -130,8 +161,7 @@ static int lo_set_switch(struct snd_kcontrol *kcontrol, case 1: pr_debug("set hs path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22); break; @@ -162,12 +192,45 @@ static const struct snd_kcontrol_new mfld_snd_controls[] = { lo_get_switch, lo_set_switch), }; +static const struct snd_soc_dapm_widget mfld_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static const struct snd_soc_dapm_route mfld_map[] = { + {"Headphones", NULL, "HPOUTR"}, + {"Headphones", NULL, "HPOUTL"}, + {"Mic", NULL, "AMIC1"}, +}; + +static void mfld_jack_check(unsigned int intr_status) +{ + struct mfld_jack_data jack_data; + + jack_data.mfld_jack = &mfld_jack; + jack_data.intr_id = intr_status; + + sn95031_jack_detection(&jack_data); + /* TODO: add american headset detection post gpiolib support */ +} + static int mfld_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_codec *codec = runtime->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; int ret_val; + /* Add jack sense widgets */ + snd_soc_dapm_new_controls(dapm, mfld_widgets, ARRAY_SIZE(mfld_widgets)); + + /* Set up the map */ + snd_soc_dapm_add_routes(dapm, mfld_map, ARRAY_SIZE(mfld_map)); + + /* always connected */ + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Mic"); + snd_soc_dapm_sync(dapm); + ret_val = snd_soc_add_controls(codec, mfld_snd_controls, ARRAY_SIZE(mfld_snd_controls)); if (ret_val) { @@ -175,8 +238,7 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) return ret_val; } /* default is earpiece pin, userspace sets it explcitly */ - snd_soc_dapm_disable_pin(dapm, "HPOUTL"); - snd_soc_dapm_disable_pin(dapm, "HPOUTR"); + snd_soc_dapm_disable_pin(dapm, "Headphones"); /* default is lineout NC, userspace sets it explcitly */ snd_soc_dapm_disable_pin(dapm, "LINEOUTL"); snd_soc_dapm_disable_pin(dapm, "LINEOUTR"); @@ -185,7 +247,29 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) /* we dont use linein in this so set to NC */ snd_soc_dapm_disable_pin(dapm, "LINEINL"); snd_soc_dapm_disable_pin(dapm, "LINEINR"); - return snd_soc_dapm_sync(dapm); + snd_soc_dapm_sync(dapm); + + /* Headset and button jack detection */ + ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1, &mfld_jack); + if (ret_val) { + pr_err("jack creation failed\n"); + return ret_val; + } + + ret_val = snd_soc_jack_add_pins(&mfld_jack, + ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins); + if (ret_val) { + pr_err("adding jack pins failed\n"); + return ret_val; + } + + /* we want to check if anything is inserted at boot, + * so send a fake event to codec and it will read adc + * to find if anything is there or not */ + mfld_jack_check(MFLD_JACK_INSERT); + return ret_val; } struct snd_soc_dai_link mfld_msic_dailink[] = { @@ -234,37 +318,102 @@ static struct snd_soc_card snd_soc_card_mfld = { .num_links = ARRAY_SIZE(mfld_msic_dailink), }; +static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev) +{ + struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev; + + memcpy_fromio(&mc_private->interrupt_status, + ((void *)(mc_private->int_base)), + sizeof(u8)); + return IRQ_WAKE_THREAD; +} + +static irqreturn_t snd_mfld_jack_detection(int irq, void *data) +{ + struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data; + + if (mfld_jack.codec == NULL) + return IRQ_HANDLED; + mfld_jack_check(mc_drv_ctx->interrupt_status); + + return IRQ_HANDLED; +} + static int __devinit snd_mfld_mc_probe(struct platform_device *pdev) { - struct platform_device *socdev; - int ret_val = 0; + int ret_val = 0, irq; + struct mfld_mc_private *mc_drv_ctx; + struct resource *irq_mem; pr_debug("snd_mfld_mc_probe called\n"); - socdev = platform_device_alloc("soc-audio", -1); - if (!socdev) { - pr_err("soc-audio device allocation failed\n"); + /* retrive the irq number */ + irq = platform_get_irq(pdev, 0); + + /* audio interrupt base of SRAM location where + * interrupts are stored by System FW */ + mc_drv_ctx = kzalloc(sizeof(*mc_drv_ctx), GFP_ATOMIC); + if (!mc_drv_ctx) { + pr_err("allocation failed\n"); return -ENOMEM; } - platform_set_drvdata(socdev, &snd_soc_card_mfld); - ret_val = platform_device_add(socdev); + + irq_mem = platform_get_resource_byname( + pdev, IORESOURCE_MEM, "IRQ_BASE"); + if (!irq_mem) { + pr_err("no mem resource given\n"); + ret_val = -ENODEV; + goto unalloc; + } + mc_drv_ctx->int_base = ioremap_nocache(irq_mem->start, + resource_size(irq_mem)); + if (!mc_drv_ctx->int_base) { + pr_err("Mapping of cache failed\n"); + ret_val = -ENOMEM; + goto unalloc; + } + /* register for interrupt */ + ret_val = request_threaded_irq(irq, snd_mfld_jack_intr_handler, + snd_mfld_jack_detection, + IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx); + if (ret_val) { + pr_err("cannot register IRQ\n"); + goto unalloc; + } + /* create soc device */ + mc_drv_ctx->socdev = platform_device_alloc("soc-audio", -1); + if (!mc_drv_ctx->socdev) { + pr_err("soc-audio device allocation failed\n"); + ret_val = -ENOMEM; + goto freeirq; + } + platform_set_drvdata(mc_drv_ctx->socdev, &snd_soc_card_mfld); + ret_val = platform_device_add(mc_drv_ctx->socdev); if (ret_val) { pr_err("Unable to add soc-audio device, err %d\n", ret_val); - platform_device_put(socdev); + goto unregister; } - - platform_set_drvdata(pdev, socdev); - + platform_set_drvdata(pdev, mc_drv_ctx); pr_debug("successfully exited probe\n"); return ret_val; + +unregister: + platform_device_put(mc_drv_ctx->socdev); +freeirq: + free_irq(irq, mc_drv_ctx); +unalloc: + kfree(mc_drv_ctx); + return ret_val; } static int __devexit snd_mfld_mc_remove(struct platform_device *pdev) { - struct platform_device *socdev = platform_get_drvdata(pdev); - pr_debug("snd_mfld_mc_remove called\n"); + struct mfld_mc_private *mc_drv_ctx = platform_get_drvdata(pdev); - platform_device_unregister(socdev); + pr_debug("snd_mfld_mc_remove called\n"); + free_irq(platform_get_irq(pdev, 0), mc_drv_ctx); + platform_device_unregister(mc_drv_ctx->socdev); + kfree(mc_drv_ctx); platform_set_drvdata(pdev, NULL); return 0; } -- cgit v1.2.3 From 36633237be60c0ec88b11e00d5fa22a305563d03 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 9 Feb 2011 21:44:34 +0530 Subject: ASoC: sn95031: Add support for reading mic bias This patch adds support to read the mic bias voltage when a jack is inserted. It uses ADC to measure. Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 127 +++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/sn95031.h | 24 +++++++++ 2 files changed, 147 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index b49d79017d3..4cc00177ee3 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -40,12 +40,129 @@ #define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100) #define SN95031_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) +/* adc helper functions */ + +/* enables mic bias voltage */ +static void sn95031_enable_mic_bias(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, SN95031_VAUD, BIT(2)|BIT(1)|BIT(0)); + snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(2), BIT(2)); +} + +/* Enable/Disable the ADC depending on the argument */ +static void configure_adc(struct snd_soc_codec *sn95031_codec, int val) +{ + int value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); + + if (val) { + /* Enable and start the ADC */ + value |= (SN95031_ADC_ENBL | SN95031_ADC_START); + value &= (~SN95031_ADC_NO_LOOP); + } else { + /* Just stop the ADC */ + value &= (~SN95031_ADC_START); + } + snd_soc_write(sn95031_codec, SN95031_ADC1CNTL1, value); +} + /* - * todo: - * capture paths - * jack detection - * PM functions + * finds an empty channel for conversion + * If the ADC is not enabled then start using 0th channel + * itself. Otherwise find an empty channel by looking for a + * channel in which the stopbit is set to 1. returns the index + * of the first free channel if succeeds or an error code. + * + * Context: can sleep + * */ +static int find_free_channel(struct snd_soc_codec *sn95031_codec) +{ + int ret = 0, i, value; + + /* check whether ADC is enabled */ + value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1); + + if ((value & SN95031_ADC_ENBL) == 0) + return 0; + + /* ADC is already enabled; Looking for an empty channel */ + for (i = 0; i < SN95031_ADC_CHANLS_MAX; i++) { + value = snd_soc_read(sn95031_codec, + SN95031_ADC_CHNL_START_ADDR + i); + if (value & SN95031_STOPBIT_MASK) { + ret = i; + break; + } + } + return (ret > SN95031_ADC_LOOP_MAX) ? (-EINVAL) : ret; +} + +/* Initialize the ADC for reading micbias values. Can sleep. */ +static int sn95031_initialize_adc(struct snd_soc_codec *sn95031_codec) +{ + int base_addr, chnl_addr; + int value; + static int channel_index; + + /* Index of the first channel in which the stop bit is set */ + channel_index = find_free_channel(sn95031_codec); + if (channel_index < 0) { + pr_err("No free ADC channels"); + return channel_index; + } + + base_addr = SN95031_ADC_CHNL_START_ADDR + channel_index; + + if (!(channel_index == 0 || channel_index == SN95031_ADC_LOOP_MAX)) { + /* Reset stop bit for channels other than 0 and 12 */ + value = snd_soc_read(sn95031_codec, base_addr); + /* Set the stop bit to zero */ + snd_soc_write(sn95031_codec, base_addr, value & 0xEF); + /* Index of the first free channel */ + base_addr++; + channel_index++; + } + + /* Since this is the last channel, set the stop bit + to 1 by ORing the DIE_SENSOR_CODE with 0x10 */ + snd_soc_write(sn95031_codec, base_addr, + SN95031_AUDIO_DETECT_CODE | 0x10); + + chnl_addr = SN95031_ADC_DATA_START_ADDR + 2 * channel_index; + pr_debug("mid_initialize : %x", chnl_addr); + configure_adc(sn95031_codec, 1); + return chnl_addr; +} + + +/* reads the ADC registers and gets the mic bias value in mV. */ +static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec) +{ + u16 adc_adr = sn95031_initialize_adc(codec); + u16 adc_val1, adc_val2; + unsigned int mic_bias; + + sn95031_enable_mic_bias(codec); + + /* Enable the sound card for conversion before reading */ + snd_soc_write(codec, SN95031_ADC1CNTL3, 0x05); + /* Re-toggle the RRDATARD bit */ + snd_soc_write(codec, SN95031_ADC1CNTL3, 0x04); + + /* Read the higher bits of data */ + msleep(1000); + adc_val1 = snd_soc_read(codec, adc_adr); + adc_adr++; + adc_val2 = snd_soc_read(codec, adc_adr); + + /* Adding lower two bits to the higher bits */ + mic_bias = (adc_val1 << 2) + (adc_val2 & 3); + mic_bias = (mic_bias * SN95031_ADC_ONE_LSB_MULTIPLIER) / 1000; + pr_debug("mic bias = %dmV\n", mic_bias); + return mic_bias; +} +EXPORT_SYMBOL_GPL(sn95031_get_mic_bias); +/*end - adc helper functions */ static inline unsigned int sn95031_read(struct snd_soc_codec *codec, unsigned int reg) @@ -663,6 +780,8 @@ static inline void sn95031_enable_jack_btn(struct snd_soc_codec *codec) static int sn95031_get_headset_state(struct snd_soc_jack *mfld_jack) { + int micbias = sn95031_get_mic_bias(mfld_jack->codec); + /* Defaulting to HEADSET for now. * will change after adding soc-jack detection apis */ int jack_type = SND_JACK_HEADSET; diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h index 2dbae614bac..20376d234fb 100644 --- a/sound/soc/codecs/sn95031.h +++ b/sound/soc/codecs/sn95031.h @@ -96,7 +96,31 @@ #define SN95031_SSR5 0x384 #define SN95031_SSR6 0x385 +/* ADC registers */ + +#define SN95031_ADC1CNTL1 0x1C0 +#define SN95031_ADC_ENBL 0x10 +#define SN95031_ADC_START 0x08 +#define SN95031_ADC1CNTL3 0x1C2 +#define SN95031_ADCTHERM_ENBL 0x04 +#define SN95031_ADCRRDATA_ENBL 0x05 +#define SN95031_STOPBIT_MASK 16 +#define SN95031_ADCTHERM_MASK 4 +#define SN95031_ADC_CHANLS_MAX 15 /* Number of ADC channels */ +#define SN95031_ADC_LOOP_MAX (SN95031_ADC_CHANLS_MAX - 1) +#define SN95031_ADC_NO_LOOP 0x07 #define SN95031_AUDIO_GPIO_CTRL 0x070 + +/* ADC channel code values */ +#define SN95031_AUDIO_DETECT_CODE 0x06 + +/* ADC base addresses */ +#define SN95031_ADC_CHNL_START_ADDR 0x1C5 /* increments by 1 */ +#define SN95031_ADC_DATA_START_ADDR 0x1D4 /* increments by 2 */ +/* multipier to convert to mV */ +#define SN95031_ADC_ONE_LSB_MULTIPLIER 2346 + + struct mfld_jack_data { int intr_id; int micbias_vol; -- cgit v1.2.3 From d852f446b71b80c59821a08cdb595aa266210daf Mon Sep 17 00:00:00 2001 From: Vaibhav Bedia Date: Wed, 9 Feb 2011 18:39:52 +0530 Subject: ASoC: Davinci: Use resource_size() helper function This patch modifies the Davinci i2s and mcasp drivers to make use of the resource_size() helper function for readability. Signed-off-by: Vaibhav Bedia Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-i2s.c | 6 +++--- sound/soc/davinci/davinci-mcasp.c | 6 +++--- 2 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 9e0e565e6ed..f8c8487aba5 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -658,7 +658,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) return -ENODEV; } - ioarea = request_mem_region(mem->start, (mem->end - mem->start) + 1, + ioarea = request_mem_region(mem->start, resource_size(mem), pdev->name); if (!ioarea) { dev_err(&pdev->dev, "McBSP region already claimed\n"); @@ -731,7 +731,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) err_free_mem: kfree(dev); err_release_region: - release_mem_region(mem->start, (mem->end - mem->start) + 1); + release_mem_region(mem->start, resource_size(mem)); return ret; } @@ -747,7 +747,7 @@ static int davinci_i2s_remove(struct platform_device *pdev) dev->clk = NULL; kfree(dev); mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(mem->start, (mem->end - mem->start) + 1); + release_mem_region(mem->start, resource_size(mem)); return 0; } diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index fb55d2c5d70..9b5d5ca37a9 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -868,7 +868,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } ioarea = request_mem_region(mem->start, - (mem->end - mem->start) + 1, pdev->name); + resource_size(mem), pdev->name); if (!ioarea) { dev_err(&pdev->dev, "Audio region already claimed\n"); ret = -EBUSY; @@ -933,7 +933,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) return 0; err_release_region: - release_mem_region(mem->start, (mem->end - mem->start) + 1); + release_mem_region(mem->start, resource_size(mem)); err_release_data: kfree(dev); @@ -951,7 +951,7 @@ static int davinci_mcasp_remove(struct platform_device *pdev) dev->clk = NULL; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(mem->start, (mem->end - mem->start) + 1); + release_mem_region(mem->start, resource_size(mem)); kfree(dev); -- cgit v1.2.3 From eef6d7b8c24e8dd5af6094ef453a05c80b68d36e Mon Sep 17 00:00:00 2001 From: Vaibhav Bedia Date: Wed, 9 Feb 2011 18:39:53 +0530 Subject: ASoC: Davinci: Call clk_disable() and clk_put() in case of error In case of any error in probe() function, clk_disable() and clk_put() should be called if clk_enable() and clk_get() went through. Signed-off-by: Vaibhav Bedia Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-i2s.c | 9 ++++++--- sound/soc/davinci/davinci-mcasp.c | 9 ++++++--- 2 files changed, 12 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index f8c8487aba5..e763d58e51f 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -707,7 +707,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); ret = -ENXIO; - goto err_free_mem; + goto err_release_clk; } dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start; @@ -715,7 +715,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); ret = -ENXIO; - goto err_free_mem; + goto err_release_clk; } dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; dev->dev = &pdev->dev; @@ -724,10 +724,13 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &davinci_i2s_dai); if (ret != 0) - goto err_free_mem; + goto err_release_clk; return 0; +err_release_clk: + clk_disable(dev->clk); + clk_put(dev->clk); err_free_mem: kfree(dev); err_release_region: diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9b5d5ca37a9..e7ae621a3af 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -906,7 +906,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); ret = -ENODEV; - goto err_release_region; + goto err_release_clk; } dma_data->channel = res->start; @@ -921,7 +921,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); ret = -ENODEV; - goto err_release_region; + goto err_release_clk; } dma_data->channel = res->start; @@ -929,9 +929,12 @@ static int davinci_mcasp_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &davinci_mcasp_dai[pdata->op_mode]); if (ret != 0) - goto err_release_region; + goto err_release_clk; return 0; +err_release_clk: + clk_disable(dev->clk); + clk_put(dev->clk); err_release_region: release_mem_region(mem->start, resource_size(mem)); err_release_data: -- cgit v1.2.3 From 4f82f0285297b89bb1d612b52dfd38fd572a8282 Mon Sep 17 00:00:00 2001 From: Vaibhav Bedia Date: Wed, 9 Feb 2011 18:39:54 +0530 Subject: ASoC: Davinci: Replace usage of IO_ADDRESS with ioremap() This patch modifies the Davinci i2s and mcasp drivers to make use of ioremap() instead of IO_ADDRESS() Signed-off-by: Vaibhav Bedia Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-i2s.c | 19 +++++++++++++------ sound/soc/davinci/davinci-mcasp.c | 20 ++++++++++++++------ 2 files changed, 27 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index e763d58e51f..d0d60b8a54d 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -694,20 +694,25 @@ static int davinci_i2s_probe(struct platform_device *pdev) } clk_enable(dev->clk); - dev->base = (void __iomem *)IO_ADDRESS(mem->start); + dev->base = ioremap(mem->start, resource_size(mem)); + if (!dev->base) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err_release_clk; + } dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = - (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG); + (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG); dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = - (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG); + (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG); /* first TX, then RX */ res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); ret = -ENXIO; - goto err_release_clk; + goto err_iounmap; } dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start; @@ -715,7 +720,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); ret = -ENXIO; - goto err_release_clk; + goto err_iounmap; } dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; dev->dev = &pdev->dev; @@ -724,10 +729,12 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &davinci_i2s_dai); if (ret != 0) - goto err_release_clk; + goto err_iounmap; return 0; +err_iounmap: + iounmap(dev->base); err_release_clk: clk_disable(dev->clk); clk_put(dev->clk); diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e7ae621a3af..a5af834c8ef 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -885,7 +885,13 @@ static int davinci_mcasp_probe(struct platform_device *pdev) clk_enable(dev->clk); dev->clk_active = 1; - dev->base = (void __iomem *)IO_ADDRESS(mem->start); + dev->base = ioremap(mem->start, resource_size(mem)); + if (!dev->base) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err_release_clk; + } + dev->op_mode = pdata->op_mode; dev->tdm_slots = pdata->tdm_slots; dev->num_serializer = pdata->num_serializer; @@ -899,14 +905,14 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset + - io_v2p(dev->base)); + mem->start); /* first TX, then RX */ res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); ret = -ENODEV; - goto err_release_clk; + goto err_iounmap; } dma_data->channel = res->start; @@ -915,13 +921,13 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset + - io_v2p(dev->base)); + mem->start); res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!res) { dev_err(&pdev->dev, "no DMA resource\n"); ret = -ENODEV; - goto err_release_clk; + goto err_iounmap; } dma_data->channel = res->start; @@ -929,9 +935,11 @@ static int davinci_mcasp_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &davinci_mcasp_dai[pdata->op_mode]); if (ret != 0) - goto err_release_clk; + goto err_iounmap; return 0; +err_iounmap: + iounmap(dev->base); err_release_clk: clk_disable(dev->clk); clk_put(dev->clk); -- cgit v1.2.3 From 4b592c919c694de79c31d5fde59c169fc79595a9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Feb 2011 13:47:06 +0000 Subject: ASoC: Remove redundant -codec from WM8903 driver name It causes noisy -codecs to appear in things like .codec_name. Signed-off-by: Mark Brown Acked-by: Stephen Warren Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 2 +- sound/soc/tegra/harmony.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 3d4c55f3c7b..7c84ebf3420 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1932,7 +1932,7 @@ MODULE_DEVICE_TABLE(i2c, wm8903_i2c_id); static struct i2c_driver wm8903_i2c_driver = { .driver = { - .name = "wm8903-codec", + .name = "wm8903", .owner = THIS_MODULE, }, .probe = wm8903_i2c_probe, diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index be95405df3f..61befcc281a 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -225,7 +225,7 @@ static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link harmony_wm8903_dai = { .name = "WM8903", .stream_name = "WM8903 PCM", - .codec_name = "wm8903-codec.0-001a", + .codec_name = "wm8903.0-001a", .platform_name = "tegra-pcm-audio", .cpu_dai_name = "tegra-i2s.0", .codec_dai_name = "wm8903-hifi", -- cgit v1.2.3 From 1d8d62d637577afc568da32e3ecc7bddc11db56d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Feb 2011 13:47:07 +0000 Subject: ASoC: Display WM8903 chip revision alphabetically Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 7c84ebf3420..e6ba4d05bf2 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1781,8 +1781,8 @@ static int wm8903_probe(struct snd_soc_codec *codec) } val = snd_soc_read(codec, WM8903_REVISION_NUMBER); - dev_info(codec->dev, "WM8903 revision %d\n", - val & WM8903_CHIP_REV_MASK); + dev_info(codec->dev, "WM8903 revision %c\n", + (val & WM8903_CHIP_REV_MASK) + 'A'); wm8903_reset(codec); -- cgit v1.2.3 From 1e113bf9e088f1a6f4a1cdadce598ccc340f8fc1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Feb 2011 13:47:08 +0000 Subject: ASoC: Add support for AIF channel muxing on WM8903 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 66 ++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 62 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index e6ba4d05bf2..0190c5aa44f 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -674,6 +674,22 @@ static const struct soc_enum lsidetone_enum = static const struct soc_enum rsidetone_enum = SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text); +static const char *aif_text[] = { + "Left", "Right" +}; + +static const struct soc_enum lcapture_enum = + SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 7, 2, aif_text); + +static const struct soc_enum rcapture_enum = + SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 6, 2, aif_text); + +static const struct soc_enum lplay_enum = + SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 5, 2, aif_text); + +static const struct soc_enum rplay_enum = + SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 4, 2, aif_text); + static const struct snd_kcontrol_new wm8903_snd_controls[] = { /* Input PGAs - No TLV since the scale depends on PGA mode */ @@ -791,6 +807,18 @@ static const struct snd_kcontrol_new lsidetone_mux = static const struct snd_kcontrol_new rsidetone_mux = SOC_DAPM_ENUM("DACR Sidetone Mux", rsidetone_enum); +static const struct snd_kcontrol_new lcapture_mux = + SOC_DAPM_ENUM("Left Capture Mux", lcapture_enum); + +static const struct snd_kcontrol_new rcapture_mux = + SOC_DAPM_ENUM("Right Capture Mux", rcapture_enum); + +static const struct snd_kcontrol_new lplay_mux = + SOC_DAPM_ENUM("Left Playback Mux", lplay_enum); + +static const struct snd_kcontrol_new rplay_mux = + SOC_DAPM_ENUM("Right Playback Mux", rplay_enum); + static const struct snd_kcontrol_new left_output_mixer[] = { SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_LEFT_MIX_0, 3, 1, 0), SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_LEFT_MIX_0, 2, 1, 0), @@ -854,14 +882,26 @@ SND_SOC_DAPM_MUX("Right Input Mode Mux", SND_SOC_NOPM, 0, 0, &rinput_mode_mux), SND_SOC_DAPM_PGA("Left Input PGA", WM8903_POWER_MANAGEMENT_0, 1, 0, NULL, 0), SND_SOC_DAPM_PGA("Right Input PGA", WM8903_POWER_MANAGEMENT_0, 0, 0, NULL, 0), -SND_SOC_DAPM_ADC("ADCL", "Left HiFi Capture", WM8903_POWER_MANAGEMENT_6, 1, 0), -SND_SOC_DAPM_ADC("ADCR", "Right HiFi Capture", WM8903_POWER_MANAGEMENT_6, 0, 0), +SND_SOC_DAPM_ADC("ADCL", NULL, WM8903_POWER_MANAGEMENT_6, 1, 0), +SND_SOC_DAPM_ADC("ADCR", NULL, WM8903_POWER_MANAGEMENT_6, 0, 0), + +SND_SOC_DAPM_MUX("Left Capture Mux", SND_SOC_NOPM, 0, 0, &lcapture_mux), +SND_SOC_DAPM_MUX("Right Capture Mux", SND_SOC_NOPM, 0, 0, &rcapture_mux), + +SND_SOC_DAPM_AIF_OUT("AIFTXL", "Left HiFi Capture", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIFTXR", "Right HiFi Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &lsidetone_mux), SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &rsidetone_mux), -SND_SOC_DAPM_DAC("DACL", "Left Playback", WM8903_POWER_MANAGEMENT_6, 3, 0), -SND_SOC_DAPM_DAC("DACR", "Right Playback", WM8903_POWER_MANAGEMENT_6, 2, 0), +SND_SOC_DAPM_AIF_IN("AIFRXL", "Left Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIFRXR", "Right Playback", 0, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_MUX("Left Playback Mux", SND_SOC_NOPM, 0, 0, &lplay_mux), +SND_SOC_DAPM_MUX("Right Playback Mux", SND_SOC_NOPM, 0, 0, &rplay_mux), + +SND_SOC_DAPM_DAC("DACL", NULL, WM8903_POWER_MANAGEMENT_6, 3, 0), +SND_SOC_DAPM_DAC("DACR", NULL, WM8903_POWER_MANAGEMENT_6, 2, 0), SND_SOC_DAPM_MIXER("Left Output Mixer", WM8903_POWER_MANAGEMENT_1, 1, 0, left_output_mixer, ARRAY_SIZE(left_output_mixer)), @@ -943,18 +983,36 @@ static const struct snd_soc_dapm_route intercon[] = { { "Left Input PGA", NULL, "Left Input Mode Mux" }, { "Right Input PGA", NULL, "Right Input Mode Mux" }, + { "Left Capture Mux", "Left", "ADCL" }, + { "Left Capture Mux", "Right", "ADCR" }, + + { "Right Capture Mux", "Left", "ADCL" }, + { "Right Capture Mux", "Right", "ADCR" }, + + { "AIFTXL", NULL, "Left Capture Mux" }, + { "AIFTXR", NULL, "Right Capture Mux" }, + { "ADCL", NULL, "Left Input PGA" }, { "ADCL", NULL, "CLK_DSP" }, { "ADCR", NULL, "Right Input PGA" }, { "ADCR", NULL, "CLK_DSP" }, + { "Left Playback Mux", "Left", "AIFRXL" }, + { "Left Playback Mux", "Right", "AIFRXR" }, + + { "Right Playback Mux", "Left", "AIFRXL" }, + { "Right Playback Mux", "Right", "AIFRXR" }, + { "DACL Sidetone", "Left", "ADCL" }, { "DACL Sidetone", "Right", "ADCR" }, { "DACR Sidetone", "Left", "ADCL" }, { "DACR Sidetone", "Right", "ADCR" }, + { "DACL", NULL, "Left Playback Mux" }, { "DACL", NULL, "DACL Sidetone" }, { "DACL", NULL, "CLK_DSP" }, + + { "DACR", NULL, "Right Playback Mux" }, { "DACR", NULL, "DACR Sidetone" }, { "DACR", NULL, "CLK_DSP" }, -- cgit v1.2.3 From 13a9983eb197254dffd9ea63a2d5f12c54eb651c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Feb 2011 17:42:55 +0000 Subject: ASoC: Convert WM8903 to use PGA_S for output stage enables This simplfies the code and slightly reduces the startup time. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 180 ++++++++++++++++------------------------------ 1 file changed, 60 insertions(+), 120 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 0190c5aa44f..9793775d579 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -246,6 +246,8 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re case WM8903_REVISION_NUMBER: case WM8903_INTERRUPT_STATUS_1: case WM8903_WRITE_SEQUENCER_4: + case WM8903_POWER_MANAGEMENT_3: + case WM8903_POWER_MANAGEMENT_2: return 1; default: @@ -304,11 +306,6 @@ static void wm8903_reset(struct snd_soc_codec *codec) sizeof(wm8903_reg_defaults)); } -#define WM8903_OUTPUT_SHORT 0x8 -#define WM8903_OUTPUT_OUT 0x4 -#define WM8903_OUTPUT_INT 0x2 -#define WM8903_OUTPUT_IN 0x1 - static int wm8903_cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -318,99 +315,6 @@ static int wm8903_cp_event(struct snd_soc_dapm_widget *w, return 0; } -/* - * Event for headphone and line out amplifier power changes. Special - * power up/down sequences are required in order to maximise pop/click - * performance. - */ -static int wm8903_output_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - u16 val; - u16 reg; - u16 dcs_reg; - u16 dcs_bit; - int shift; - - switch (w->reg) { - case WM8903_POWER_MANAGEMENT_2: - reg = WM8903_ANALOGUE_HP_0; - dcs_bit = 0 + w->shift; - break; - case WM8903_POWER_MANAGEMENT_3: - reg = WM8903_ANALOGUE_LINEOUT_0; - dcs_bit = 2 + w->shift; - break; - default: - BUG(); - return -EINVAL; /* Spurious warning from some compilers */ - } - - switch (w->shift) { - case 0: - shift = 0; - break; - case 1: - shift = 4; - break; - default: - BUG(); - return -EINVAL; /* Spurious warning from some compilers */ - } - - if (event & SND_SOC_DAPM_PRE_PMU) { - val = snd_soc_read(codec, reg); - - /* Short the output */ - val &= ~(WM8903_OUTPUT_SHORT << shift); - snd_soc_write(codec, reg, val); - } - - if (event & SND_SOC_DAPM_POST_PMU) { - val = snd_soc_read(codec, reg); - - val |= (WM8903_OUTPUT_IN << shift); - snd_soc_write(codec, reg, val); - - val |= (WM8903_OUTPUT_INT << shift); - snd_soc_write(codec, reg, val); - - /* Turn on the output ENA_OUTP */ - val |= (WM8903_OUTPUT_OUT << shift); - snd_soc_write(codec, reg, val); - - /* Enable the DC servo */ - dcs_reg = snd_soc_read(codec, WM8903_DC_SERVO_0); - dcs_reg |= dcs_bit; - snd_soc_write(codec, WM8903_DC_SERVO_0, dcs_reg); - - /* Remove the short */ - val |= (WM8903_OUTPUT_SHORT << shift); - snd_soc_write(codec, reg, val); - } - - if (event & SND_SOC_DAPM_PRE_PMD) { - val = snd_soc_read(codec, reg); - - /* Short the output */ - val &= ~(WM8903_OUTPUT_SHORT << shift); - snd_soc_write(codec, reg, val); - - /* Disable the DC servo */ - dcs_reg = snd_soc_read(codec, WM8903_DC_SERVO_0); - dcs_reg &= ~dcs_bit; - snd_soc_write(codec, WM8903_DC_SERVO_0, dcs_reg); - - /* Then disable the intermediate and output stages */ - val &= ~((WM8903_OUTPUT_OUT | WM8903_OUTPUT_INT | - WM8903_OUTPUT_IN) << shift); - snd_soc_write(codec, reg, val); - } - - return 0; -} - /* * When used with DAC outputs only the WM8903 charge pump supports * operation in class W mode, providing very low power consumption @@ -913,23 +817,40 @@ SND_SOC_DAPM_MIXER("Left Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 1, 0, SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0, right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), -SND_SOC_DAPM_PGA_E("Left Headphone Output PGA", WM8903_POWER_MANAGEMENT_2, - 1, 0, NULL, 0, wm8903_output_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD), -SND_SOC_DAPM_PGA_E("Right Headphone Output PGA", WM8903_POWER_MANAGEMENT_2, - 0, 0, NULL, 0, wm8903_output_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD), - -SND_SOC_DAPM_PGA_E("Left Line Output PGA", WM8903_POWER_MANAGEMENT_3, 1, 0, - NULL, 0, wm8903_output_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD), -SND_SOC_DAPM_PGA_E("Right Line Output PGA", WM8903_POWER_MANAGEMENT_3, 0, 0, - NULL, 0, wm8903_output_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0, + 4, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0, + 0, 0, NULL, 0), + +SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 4, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 0, 0, + NULL, 0), + +SND_SOC_DAPM_PGA_S("HPL_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 7, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPL_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 3, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0), + +SND_SOC_DAPM_PGA_S("LINEOUTL_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 7, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 6, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 5, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTR_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 3, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0, + NULL, 0), + +SND_SOC_DAPM_PGA_S("HPL_DCS", 3, WM8903_DC_SERVO_0, 3, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_DCS", 3, WM8903_DC_SERVO_0, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTL_DCS", 3, WM8903_DC_SERVO_0, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTR_DCS", 3, WM8903_DC_SERVO_0, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("Left Speaker PGA", WM8903_POWER_MANAGEMENT_5, 1, 0, NULL, 0), @@ -1045,11 +966,30 @@ static const struct snd_soc_dapm_route intercon[] = { { "Left Speaker PGA", NULL, "Left Speaker Mixer" }, { "Right Speaker PGA", NULL, "Right Speaker Mixer" }, - { "HPOUTL", NULL, "Left Headphone Output PGA" }, - { "HPOUTR", NULL, "Right Headphone Output PGA" }, - - { "LINEOUTL", NULL, "Left Line Output PGA" }, - { "LINEOUTR", NULL, "Right Line Output PGA" }, + { "HPL_ENA_DLY", NULL, "Left Headphone Output PGA" }, + { "HPR_ENA_DLY", NULL, "Right Headphone Output PGA" }, + { "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" }, + { "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" }, + + { "HPL_DCS", NULL, "HPL_ENA_DLY" }, + { "HPR_DCS", NULL, "HPR_ENA_DLY" }, + { "LINEOUTL_DCS", NULL, "LINEOUTL_ENA_DLY" }, + { "LINEOUTR_DCS", NULL, "LINEOUTR_ENA_DLY" }, + + { "HPL_ENA_OUTP", NULL, "HPL_DCS" }, + { "HPR_ENA_OUTP", NULL, "HPR_DCS" }, + { "LINEOUTL_ENA_OUTP", NULL, "LINEOUTL_DCS" }, + { "LINEOUTR_ENA_OUTP", NULL, "LINEOUTR_DCS" }, + + { "HPL_RMV_SHORT", NULL, "HPL_ENA_OUTP" }, + { "HPR_RMV_SHORT", NULL, "HPR_ENA_OUTP" }, + { "LINEOUTL_RMV_SHORT", NULL, "LINEOUTL_ENA_OUTP" }, + { "LINEOUTR_RMV_SHORT", NULL, "LINEOUTR_ENA_OUTP" }, + + { "HPOUTL", NULL, "HPL_RMV_SHORT" }, + { "HPOUTR", NULL, "HPR_RMV_SHORT" }, + { "LINEOUTL", NULL, "LINEOUTL_RMV_SHORT" }, + { "LINEOUTR", NULL, "LINEOUTR_RMV_SHORT" }, { "LOP", NULL, "Left Speaker PGA" }, { "LON", NULL, "Left Speaker PGA" }, -- cgit v1.2.3 From 2c8be5a26e42cfc4906c4daa8a5a5c82610ddb3d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Feb 2011 17:42:56 +0000 Subject: ASoC: Dynamically manage CLK_SYS in WM8903 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 9793775d579..389d2e8088f 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -860,10 +860,18 @@ SND_SOC_DAPM_PGA("Right Speaker PGA", WM8903_POWER_MANAGEMENT_5, 0, 0, SND_SOC_DAPM_SUPPLY("Charge Pump", WM8903_CHARGE_PUMP_0, 0, 0, wm8903_cp_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8903_CLOCK_RATES_2, 1, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("CLK_SYS", WM8903_CLOCK_RATES_2, 2, 0, NULL, 0), }; static const struct snd_soc_dapm_route intercon[] = { + { "CLK_DSP", NULL, "CLK_SYS" }, + { "Mic Bias", NULL, "CLK_SYS" }, + { "HPL_DCS", NULL, "CLK_SYS" }, + { "HPR_DCS", NULL, "CLK_SYS" }, + { "LINEOUTL_DCS", NULL, "CLK_SYS" }, + { "LINEOUTR_DCS", NULL, "CLK_SYS" }, + { "Left Input Mux", "IN1L", "IN1L" }, { "Left Input Mux", "IN2L", "IN2L" }, { "Left Input Mux", "IN3L", "IN3L" }, @@ -1059,10 +1067,11 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, WM8903_CLOCK_RATES_2, + WM8903_CLK_SYS_ENA, WM8903_CLK_SYS_ENA); wm8903_run_sequence(codec, 32); - reg = snd_soc_read(codec, WM8903_CLOCK_RATES_2); - reg &= ~WM8903_CLK_SYS_ENA; - snd_soc_write(codec, WM8903_CLOCK_RATES_2, reg); + snd_soc_update_bits(codec, WM8903_CLOCK_RATES_2, + WM8903_CLK_SYS_ENA, 0); break; } -- cgit v1.2.3 From e12adab00222817213fcdc68c5fd6ee2e5dfb247 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Feb 2011 17:42:57 +0000 Subject: ASoC: Fix WM8903 DAC mute default The WM8903 register map does not mute the DAC by default at startup so we need to explicitly do so. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 389d2e8088f..119abceb1d8 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1870,9 +1870,9 @@ static int wm8903_probe(struct snd_soc_codec *codec) snd_soc_write(codec, WM8903_ANALOGUE_OUT3_RIGHT, val); /* Enable DAC soft mute by default */ - val = snd_soc_read(codec, WM8903_DAC_DIGITAL_1); - val |= WM8903_DAC_MUTEMODE; - snd_soc_write(codec, WM8903_DAC_DIGITAL_1, val); + snd_soc_update_bits(codec, WM8903_DAC_DIGITAL_1, + WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE, + WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE); snd_soc_add_controls(codec, wm8903_snd_controls, ARRAY_SIZE(wm8903_snd_controls)); -- cgit v1.2.3 From fa9879edebdaad4cfcd2dbe3eaa2ba0dc4f0a262 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 9 Feb 2011 14:44:17 +0530 Subject: ASoC: add support for multiple jack types This patch adds soc-jack support for adding voltage zones and for detecting jack type Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Signed-off-by: Mark Brown --- sound/soc/soc-jack.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ac5a5bc7375..99dbaf756b4 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -37,6 +37,7 @@ int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, { jack->codec = codec; INIT_LIST_HEAD(&jack->pins); + INIT_LIST_HEAD(&jack->jack_zones); BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier); return snd_jack_new(codec->card->snd_card, id, type, &jack->jack); @@ -111,6 +112,51 @@ out: } EXPORT_SYMBOL_GPL(snd_soc_jack_report); +/** + * snd_soc_jack_add_zones - Associate voltage zones with jack + * + * @jack: ASoC jack + * @count: Number of zones + * @zone: Array of zones + * + * After this function has been called the zones specified in the + * array will be associated with the jack. + */ +int snd_soc_jack_add_zones(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_zone *zones) +{ + int i; + + for (i = 0; i < count; i++) { + INIT_LIST_HEAD(&zones[i].list); + list_add(&(zones[i].list), &jack->jack_zones); + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_add_zones); + +/** + * snd_soc_jack_get_type - Based on the mic bias value, this function returns + * the type of jack from the zones delcared in the jack type + * + * @micbias_voltage: mic bias voltage at adc channel when jack is plugged in + * + * Based on the mic bias value passed, this function helps identify + * the type of jack from the already delcared jack zones + */ +int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage) +{ + struct snd_soc_jack_zone *zone; + + list_for_each_entry(zone, &jack->jack_zones, list) { + if (micbias_voltage >= zone->min_mv && + micbias_voltage < zone->max_mv) + return zone->jack_type; + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_get_type); + /** * snd_soc_jack_add_pins - Associate DAPM pins with an ASoC jack * -- cgit v1.2.3 From c5b6a9feaeb0fa0e39e3fc10f9bf8cc8de498739 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Feb 2011 20:14:42 +0000 Subject: ASoC: Actively manage WM8903 DC servo configuration Explicitly cache the DC servo offsets for digital paths in the driver, allowing them to be preserved over suspend and resume, and ensure that we recalibrate analogue outputs paths when they are in use so that we cover any changes in the input offset. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 123 ++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/wm8903.h | 8 +++ 2 files changed, 127 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 119abceb1d8..2f912276a8f 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -223,6 +223,9 @@ struct wm8903_priv { int fs; int deemph; + int dcs_pending; + int dcs_cache[4]; + /* Reference count */ int class_w_users; @@ -248,6 +251,10 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re case WM8903_WRITE_SEQUENCER_4: case WM8903_POWER_MANAGEMENT_3: case WM8903_POWER_MANAGEMENT_2: + case WM8903_DC_SERVO_READBACK_1: + case WM8903_DC_SERVO_READBACK_2: + case WM8903_DC_SERVO_READBACK_3: + case WM8903_DC_SERVO_READBACK_4: return 1; default: @@ -315,6 +322,103 @@ static int wm8903_cp_event(struct snd_soc_dapm_widget *w, return 0; } +static int wm8903_dcs_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + wm8903->dcs_pending |= 1 << w->shift; + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, WM8903_DC_SERVO_0, + 1 << w->shift, 0); + break; + } + + return 0; +} + +#define WM8903_DCS_MODE_WRITE_STOP 0 +#define WM8903_DCS_MODE_START_STOP 2 + +static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, + enum snd_soc_dapm_type event, int subseq) +{ + struct snd_soc_codec *codec = container_of(dapm, + struct snd_soc_codec, dapm); + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + int dcs_mode = WM8903_DCS_MODE_WRITE_STOP; + int i, val; + + /* Complete any pending DC servo starts */ + if (wm8903->dcs_pending) { + dev_dbg(codec->dev, "Starting DC servo for %x\n", + wm8903->dcs_pending); + + /* If we've no cached values then we need to do startup */ + for (i = 0; i < ARRAY_SIZE(wm8903->dcs_cache); i++) { + if (!(wm8903->dcs_pending & (1 << i))) + continue; + + if (wm8903->dcs_cache[i]) { + dev_dbg(codec->dev, + "Restore DC servo %d value %x\n", + 3 - i, wm8903->dcs_cache[i]); + + snd_soc_write(codec, WM8903_DC_SERVO_4 + i, + wm8903->dcs_cache[i] & 0xff); + } else { + dev_dbg(codec->dev, + "Calibrate DC servo %d\n", 3 - i); + dcs_mode = WM8903_DCS_MODE_START_STOP; + } + } + + /* Don't trust the cache for analogue */ + if (wm8903->class_w_users) + dcs_mode = WM8903_DCS_MODE_START_STOP; + + snd_soc_update_bits(codec, WM8903_DC_SERVO_2, + WM8903_DCS_MODE_MASK, dcs_mode); + + snd_soc_update_bits(codec, WM8903_DC_SERVO_0, + WM8903_DCS_ENA_MASK, wm8903->dcs_pending); + + switch (dcs_mode) { + case WM8903_DCS_MODE_WRITE_STOP: + break; + + case WM8903_DCS_MODE_START_STOP: + msleep(270); + + /* Cache the measured offsets for digital */ + if (wm8903->class_w_users) + break; + + for (i = 0; i < ARRAY_SIZE(wm8903->dcs_cache); i++) { + if (!(wm8903->dcs_pending & (1 << i))) + continue; + + val = snd_soc_read(codec, + WM8903_DC_SERVO_READBACK_1 + i); + dev_dbg(codec->dev, "DC servo %d: %x\n", + 3 - i, val); + wm8903->dcs_cache[i] = val; + } + break; + + default: + pr_warn("DCS mode %d delay not set\n", dcs_mode); + break; + } + + wm8903->dcs_pending = 0; + } +} + /* * When used with DAC outputs only the WM8903 charge pump supports * operation in class W mode, providing very low power consumption @@ -847,10 +951,15 @@ SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0, SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPL_DCS", 3, WM8903_DC_SERVO_0, 3, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPR_DCS", 3, WM8903_DC_SERVO_0, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("LINEOUTL_DCS", 3, WM8903_DC_SERVO_0, 1, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("LINEOUTR_DCS", 3, WM8903_DC_SERVO_0, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DCS Master", WM8903_DC_SERVO_0, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPL_DCS", 3, SND_SOC_NOPM, 3, 0, wm8903_dcs_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_S("HPR_DCS", 3, SND_SOC_NOPM, 2, 0, wm8903_dcs_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_S("LINEOUTL_DCS", 3, SND_SOC_NOPM, 1, 0, wm8903_dcs_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_S("LINEOUTR_DCS", 3, SND_SOC_NOPM, 0, 0, wm8903_dcs_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA("Left Speaker PGA", WM8903_POWER_MANAGEMENT_5, 1, 0, NULL, 0), @@ -979,6 +1088,11 @@ static const struct snd_soc_dapm_route intercon[] = { { "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" }, { "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" }, + { "HPL_DCS", NULL, "DCS Master" }, + { "HPR_DCS", NULL, "DCS Master" }, + { "LINEOUTL_DCS", NULL, "DCS Master" }, + { "LINEOUTR_DCS", NULL, "DCS Master" }, + { "HPL_DCS", NULL, "HPL_ENA_DLY" }, { "HPR_DCS", NULL, "HPR_ENA_DLY" }, { "LINEOUTL_DCS", NULL, "LINEOUTL_ENA_DLY" }, @@ -1901,6 +2015,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8903 = { .reg_word_size = sizeof(u16), .reg_cache_default = wm8903_reg_defaults, .volatile_register = wm8903_volatile_register, + .seq_notifier = wm8903_seq_notifier, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h index e8490f3edd0..9ef394ae0dd 100644 --- a/sound/soc/codecs/wm8903.h +++ b/sound/soc/codecs/wm8903.h @@ -75,6 +75,14 @@ extern int wm8903_mic_detect(struct snd_soc_codec *codec, #define WM8903_ANALOGUE_SPK_OUTPUT_CONTROL_0 0x41 #define WM8903_DC_SERVO_0 0x43 #define WM8903_DC_SERVO_2 0x45 +#define WM8903_DC_SERVO_4 0x47 +#define WM8903_DC_SERVO_5 0x48 +#define WM8903_DC_SERVO_6 0x49 +#define WM8903_DC_SERVO_7 0x4A +#define WM8903_DC_SERVO_READBACK_1 0x51 +#define WM8903_DC_SERVO_READBACK_2 0x52 +#define WM8903_DC_SERVO_READBACK_3 0x53 +#define WM8903_DC_SERVO_READBACK_4 0x54 #define WM8903_ANALOGUE_HP_0 0x5A #define WM8903_ANALOGUE_LINEOUT_0 0x5E #define WM8903_CHARGE_PUMP_0 0x62 -- cgit v1.2.3 From 66daaa59d5f0310238de183918e13062428fb59f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 10 Feb 2011 13:32:58 +0000 Subject: ASoC: Convert WM8903 bias management to use snd_soc_update_bits() Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 17 +++++++---------- 1 file changed, 7 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 2f912276a8f..b88c6165dd2 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1139,15 +1139,13 @@ static int wm8903_add_widgets(struct snd_soc_codec *codec) static int wm8903_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; - switch (level) { case SND_SOC_BIAS_ON: + break; case SND_SOC_BIAS_PREPARE: - reg = snd_soc_read(codec, WM8903_VMID_CONTROL_0); - reg &= ~(WM8903_VMID_RES_MASK); - reg |= WM8903_VMID_RES_50K; - snd_soc_write(codec, WM8903_VMID_CONTROL_0, reg); + snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, + WM8903_VMID_RES_MASK, + WM8903_VMID_RES_50K); break; case SND_SOC_BIAS_STANDBY: @@ -1174,10 +1172,9 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, WM8903_CP_DYN_V); } - reg = snd_soc_read(codec, WM8903_VMID_CONTROL_0); - reg &= ~(WM8903_VMID_RES_MASK); - reg |= WM8903_VMID_RES_250K; - snd_soc_write(codec, WM8903_VMID_CONTROL_0, reg); + snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, + WM8903_VMID_RES_MASK, + WM8903_VMID_RES_250K); break; case SND_SOC_BIAS_OFF: -- cgit v1.2.3 From 22f226dd1496a0fa470e64a66e2da474f34eebf8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 10 Feb 2011 14:01:38 +0000 Subject: ASoC: Don't use write sequencer to power up WM8903 The write sequencer sequencer sequence takes longer than is desirable as it brings up a full playback path which is not required at this point. Open coding the sequence cuts the startup time by two thirds. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 73 +++++++++++++++++++++++++++++++++++------------ 1 file changed, 54 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b88c6165dd2..e203a3ec655 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -297,15 +297,6 @@ static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start) return 0; } -static void wm8903_sync_reg_cache(struct snd_soc_codec *codec, u16 *cache) -{ - int i; - - /* There really ought to be something better we can do here :/ */ - for (i = 0; i < ARRAY_SIZE(wm8903_reg_defaults); i++) - cache[i] = codec->hw_read(codec, i); -} - static void wm8903_reset(struct snd_soc_codec *codec) { snd_soc_write(codec, WM8903_SW_RESET_AND_ID, 0); @@ -1142,6 +1133,7 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: break; + case SND_SOC_BIAS_PREPARE: snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, WM8903_VMID_RES_MASK, @@ -1150,16 +1142,59 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_write(codec, WM8903_CLOCK_RATES_2, - WM8903_CLK_SYS_ENA); - - /* Change DC servo dither level in startup sequence */ - snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, 0x11); - snd_soc_write(codec, WM8903_WRITE_SEQUENCER_1, 0x1257); - snd_soc_write(codec, WM8903_WRITE_SEQUENCER_2, 0x2); - - wm8903_run_sequence(codec, 0); - wm8903_sync_reg_cache(codec, codec->reg_cache); + snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0, + WM8903_POBCTRL | WM8903_ISEL_MASK | + WM8903_STARTUP_BIAS_ENA | + WM8903_BIAS_ENA, + WM8903_POBCTRL | + (2 << WM8903_ISEL_SHIFT) | + WM8903_STARTUP_BIAS_ENA); + + snd_soc_update_bits(codec, + WM8903_ANALOGUE_SPK_OUTPUT_CONTROL_0, + WM8903_SPK_DISCHARGE, + WM8903_SPK_DISCHARGE); + + msleep(33); + + snd_soc_update_bits(codec, WM8903_POWER_MANAGEMENT_5, + WM8903_SPKL_ENA | WM8903_SPKR_ENA, + WM8903_SPKL_ENA | WM8903_SPKR_ENA); + + snd_soc_update_bits(codec, + WM8903_ANALOGUE_SPK_OUTPUT_CONTROL_0, + WM8903_SPK_DISCHARGE, 0); + + snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, + WM8903_VMID_TIE_ENA | + WM8903_BUFIO_ENA | + WM8903_VMID_IO_ENA | + WM8903_VMID_SOFT_MASK | + WM8903_VMID_RES_MASK | + WM8903_VMID_BUF_ENA, + WM8903_VMID_TIE_ENA | + WM8903_BUFIO_ENA | + WM8903_VMID_IO_ENA | + (2 << WM8903_VMID_SOFT_SHIFT) | + WM8903_VMID_RES_250K | + WM8903_VMID_BUF_ENA); + + msleep(129); + + snd_soc_update_bits(codec, WM8903_POWER_MANAGEMENT_5, + WM8903_SPKL_ENA | WM8903_SPKR_ENA, + 0); + + snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, + WM8903_VMID_SOFT_MASK, 0); + + snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, + WM8903_VMID_RES_MASK, + WM8903_VMID_RES_50K); + + snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0, + WM8903_BIAS_ENA | WM8903_POBCTRL, + WM8903_BIAS_ENA); /* By default no bypass paths are enabled so * enable Class W support. -- cgit v1.2.3 From b4d06f456dcf956761e2c927e62b03861f07dbbf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 10 Feb 2011 14:20:49 +0000 Subject: ASoC: Use explicit sequence for WM8903 bias off This makes no real difference compared to the write sequencer sequence that was previously used but can run without a clock being provided. Also remove the write sequencer support code as this was the last use of it. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 66 +++++++++++++++-------------------------------- 1 file changed, 21 insertions(+), 45 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index e203a3ec655..c7b52a04fe1 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -229,8 +229,6 @@ struct wm8903_priv { /* Reference count */ int class_w_users; - struct completion wseq; - struct snd_soc_jack *mic_jack; int mic_det; int mic_short; @@ -262,41 +260,6 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re } } -static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start) -{ - u16 reg[5]; - struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - - BUG_ON(start > 48); - - /* Enable the sequencer if it's not already on */ - reg[0] = snd_soc_read(codec, WM8903_WRITE_SEQUENCER_0); - snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, - reg[0] | WM8903_WSEQ_ENA); - - dev_dbg(codec->dev, "Starting sequence at %d\n", start); - - snd_soc_write(codec, WM8903_WRITE_SEQUENCER_3, - start | WM8903_WSEQ_START); - - /* Wait for it to complete. If we have the interrupt wired up then - * that will break us out of the poll early. - */ - do { - wait_for_completion_timeout(&wm8903->wseq, - msecs_to_jiffies(10)); - - reg[4] = snd_soc_read(codec, WM8903_WRITE_SEQUENCER_4); - } while (reg[4] & WM8903_WSEQ_BUSY); - - dev_dbg(codec->dev, "Sequence complete\n"); - - /* Disable the sequencer again if we enabled it */ - snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, reg[0]); - - return 0; -} - static void wm8903_reset(struct snd_soc_codec *codec) { snd_soc_write(codec, WM8903_SW_RESET_AND_ID, 0); @@ -1213,11 +1176,26 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: - snd_soc_update_bits(codec, WM8903_CLOCK_RATES_2, - WM8903_CLK_SYS_ENA, WM8903_CLK_SYS_ENA); - wm8903_run_sequence(codec, 32); - snd_soc_update_bits(codec, WM8903_CLOCK_RATES_2, - WM8903_CLK_SYS_ENA, 0); + snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0, + WM8903_BIAS_ENA, 0); + + snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, + WM8903_VMID_SOFT_MASK, + 2 << WM8903_VMID_SOFT_SHIFT); + + snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, + WM8903_VMID_BUF_ENA, 0); + + msleep(290); + + snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0, + WM8903_VMID_TIE_ENA | WM8903_BUFIO_ENA | + WM8903_VMID_IO_ENA | WM8903_VMID_RES_MASK | + WM8903_VMID_SOFT_MASK | + WM8903_VMID_BUF_ENA, 0); + + snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0, + WM8903_STARTUP_BIAS_ENA, 0); break; } @@ -1670,8 +1648,7 @@ static irqreturn_t wm8903_irq(int irq, void *data) int_val = snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1) & mask; if (int_val & WM8903_WSEQ_BUSY_EINT) { - dev_dbg(codec->dev, "Write sequencer done\n"); - complete(&wm8903->wseq); + dev_warn(codec->dev, "Write sequencer done\n"); } /* @@ -1918,7 +1895,6 @@ static int wm8903_probe(struct snd_soc_codec *codec) u16 val; wm8903->codec = codec; - init_completion(&wm8903->wseq); ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); if (ret != 0) { -- cgit v1.2.3 From 7ae7434086f5b106021276e88b8ef49debf30aa8 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 10 Feb 2011 12:58:01 +0530 Subject: ASoC: mid-x86: Use the soc-jack apis for jack type detection This patch modifies the mfld_machine to use the new jack apis for adding the voltage zones for jack type detection. It also modifed TI sn95031 codec driver to use these new apis Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 4 +--- sound/soc/mid-x86/mfld_machine.c | 12 ++++++++++++ 2 files changed, 13 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 4cc00177ee3..d0b78020671 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -782,9 +782,7 @@ static int sn95031_get_headset_state(struct snd_soc_jack *mfld_jack) { int micbias = sn95031_get_mic_bias(mfld_jack->codec); - /* Defaulting to HEADSET for now. - * will change after adding soc-jack detection apis */ - int jack_type = SND_JACK_HEADSET; + int jack_type = snd_soc_jack_get_type(mfld_jack, micbias); pr_debug("jack type detected = %d\n", jack_type); if (jack_type == SND_JACK_HEADSET) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 45a00670802..96487fb8d26 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -74,6 +74,12 @@ static struct snd_soc_jack_pin mfld_jack_pins[] = { }, }; +/* jack detection voltage zones */ +static struct snd_soc_jack_zone mfld_zones[] = { + {MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE}, + {MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET}, +}; + /* sound card controls */ static const char *headset_switch_text[] = {"Earpiece", "Headset"}; @@ -264,6 +270,12 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) pr_err("adding jack pins failed\n"); return ret_val; } + ret_val = snd_soc_jack_add_zones(&mfld_jack, + ARRAY_SIZE(mfld_zones), mfld_zones); + if (ret_val) { + pr_err("adding jack zones failed\n"); + return ret_val; + } /* we want to check if anything is inserted at boot, * so send a fake event to codec and it will read adc -- cgit v1.2.3 From 535787b6ae081171a5e7dbf0158ef9fa56d59dc8 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Thu, 10 Feb 2011 17:22:23 +0200 Subject: ASoC: Allow use sleeping gpio in soc-jack It is safe to use sleeping gpio in snd_soc_jack_gpio_detect as it is not called from interrupt context. This avoids WARN_ON from __gpio_get_value if sleeping gpio is registered for jack. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 99dbaf756b4..4579ee090bb 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -240,7 +240,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) int enable; int report; - enable = gpio_get_value(gpio->gpio); + enable = gpio_get_value_cansleep(gpio->gpio); if (gpio->invert) enable = !enable; -- cgit v1.2.3 From 41b5f9b349617a42696b6783a86cfa8226044c6f Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 10 Feb 2011 15:37:16 -0700 Subject: ASoC: Tegra: Harmony: Implement mic detection * Add jack definition for mic jack * Request wm8903 to enable mic detection * Force mic bias on, since it's required for mic detection Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/harmony.c | 25 ++++++++++++++++++++++--- 1 file changed, 22 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index 61befcc281a..cb95e534485 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -43,6 +43,8 @@ #include #include +#include "../codecs/wm8903.h" + #include "tegra_das.h" #include "tegra_i2s.h" #include "tegra_pcm.h" @@ -142,6 +144,15 @@ static struct snd_soc_jack_gpio harmony_hp_jack_gpios[] = { } }; +static struct snd_soc_jack harmony_mic_jack; + +static struct snd_soc_jack_pin harmony_mic_jack_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, +}; + static int harmony_event_int_spk(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { @@ -206,9 +217,6 @@ static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_add_routes(dapm, harmony_audio_map, ARRAY_SIZE(harmony_audio_map)); - snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_sync(dapm); - harmony_hp_jack_gpios[0].gpio = pdata->gpio_hp_det; snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, &harmony_hp_jack); @@ -219,6 +227,17 @@ static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(harmony_hp_jack_gpios), harmony_hp_jack_gpios); + snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, + &harmony_mic_jack); + snd_soc_jack_add_pins(&harmony_mic_jack, + ARRAY_SIZE(harmony_mic_jack_pins), + harmony_mic_jack_pins); + wm8903_mic_detect(codec, &harmony_mic_jack, SND_JACK_MICROPHONE, 0); + + snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); + + snd_soc_dapm_sync(dapm); + return 0; } -- cgit v1.2.3 From 0d6cdca7199fbc93a8f29369f2b1e750d346ac8f Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 10 Feb 2011 15:37:17 -0700 Subject: ASoC: Harmony: Call snd_soc_dapm_nc_pin Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/harmony.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index cb95e534485..cd5d90ae33e 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -236,6 +236,11 @@ static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); + snd_soc_dapm_nc_pin(dapm, "IN3L"); + snd_soc_dapm_nc_pin(dapm, "IN3R"); + snd_soc_dapm_nc_pin(dapm, "LINEOUTL"); + snd_soc_dapm_nc_pin(dapm, "LINEOUTR"); + snd_soc_dapm_sync(dapm); return 0; -- cgit v1.2.3 From bf1b1328360d53a41cdf3f8639a5f38cb9537f1c Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 10 Feb 2011 15:37:18 -0700 Subject: ASoC: Tegra: Harmony: Explicitly set mic enables Harmony has both an external mic (a regular mic jack) and an internal mic (a 0.1" two-pin header on the board). The external mic is connected to the WM8903's IN1L pin, and is supported by the current driver. The internal mic is connected to the WM8903's IN1R pin, and is not supported by the current driver. It appears that no Harmony systems were shipped with any internal mic connected; users were expected to provide their own. This makes the internal mic connection less interesting. The WM8903's Mic Bias signal is used for both of these mics. For each mic, a GPIO drives a transistor which gates whether the mic bias signal is actively connected to that mic, or isolated from it. The dual use of the mic bias for both mics makes a general-purpose complete implementation of mic detection using the mic bias complex. So, for simplicity, the internal mic is currently ignored by the driver. This patch configures the relevant GPIOs to enable the mic bias connection to the external mic, and disable the mic bias connection to the internal mic. Note that in practice, this is the default state if these GPIOs aren't configured. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/harmony.c | 34 +++++++++++++++++++++++++++++++--- 1 file changed, 31 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index cd5d90ae33e..66512426317 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -52,10 +52,14 @@ #define DRV_NAME "tegra-snd-harmony" +#define GPIO_SPKR_EN BIT(0) +#define GPIO_INT_MIC_EN BIT(1) +#define GPIO_EXT_MIC_EN BIT(2) + struct tegra_harmony { struct tegra_asoc_utils_data util_data; struct harmony_audio_platform_data *pdata; - int gpio_spkr_en_requested; + int gpio_requested; }; static int harmony_asoc_hw_params(struct snd_pcm_substream *substream, @@ -202,10 +206,30 @@ static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd) dev_err(card->dev, "cannot get spkr_en gpio\n"); return ret; } - harmony->gpio_spkr_en_requested = 1; + harmony->gpio_requested |= GPIO_SPKR_EN; gpio_direction_output(pdata->gpio_spkr_en, 0); + ret = gpio_request(pdata->gpio_int_mic_en, "int_mic_en"); + if (ret) { + dev_err(card->dev, "cannot get int_mic_en gpio\n"); + return ret; + } + harmony->gpio_requested |= GPIO_INT_MIC_EN; + + /* Disable int mic; enable signal is active-high */ + gpio_direction_output(pdata->gpio_int_mic_en, 0); + + ret = gpio_request(pdata->gpio_ext_mic_en, "ext_mic_en"); + if (ret) { + dev_err(card->dev, "cannot get ext_mic_en gpio\n"); + return ret; + } + harmony->gpio_requested |= GPIO_EXT_MIC_EN; + + /* Enable ext mic; enable signal is active-low */ + gpio_direction_output(pdata->gpio_ext_mic_en, 0); + ret = snd_soc_add_controls(codec, harmony_controls, ARRAY_SIZE(harmony_controls)); if (ret < 0) @@ -330,7 +354,11 @@ static int __devexit tegra_snd_harmony_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&harmony->util_data); - if (harmony->gpio_spkr_en_requested) + if (harmony->gpio_requested & GPIO_EXT_MIC_EN) + gpio_free(pdata->gpio_ext_mic_en); + if (harmony->gpio_requested & GPIO_INT_MIC_EN) + gpio_free(pdata->gpio_int_mic_en); + if (harmony->gpio_requested & GPIO_SPKR_EN) gpio_free(pdata->gpio_spkr_en); kfree(harmony); -- cgit v1.2.3 From 8eb34207c8cf90bc991f0141f7d3fa614429a00b Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Thu, 10 Feb 2011 15:37:19 -0700 Subject: ASoC: Tegra: Add MODULE_ALIAS With the appropriate MODULE_ALIAS in place, the audio modules will be automatically loaded; there is no longer a need for manual modprobes. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/harmony.c | 1 + sound/soc/tegra/tegra_das.c | 1 + sound/soc/tegra/tegra_i2s.c | 1 + sound/soc/tegra/tegra_pcm.c | 5 ++++- 4 files changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index 66512426317..8585957477e 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -390,3 +390,4 @@ module_exit(snd_tegra_harmony_exit); MODULE_AUTHOR("Stephen Warren "); MODULE_DESCRIPTION("Harmony machine ASoC driver"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c index 01eb9c9301d..9f24ef73f2c 100644 --- a/sound/soc/tegra/tegra_das.c +++ b/sound/soc/tegra/tegra_das.c @@ -262,3 +262,4 @@ module_exit(tegra_das_modexit); MODULE_AUTHOR("Stephen Warren "); MODULE_DESCRIPTION("Tegra DAS driver"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 870ee361f75..4f5e2c90b02 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -500,3 +500,4 @@ module_exit(snd_tegra_i2s_exit); MODULE_AUTHOR("Stephen Warren "); MODULE_DESCRIPTION("Tegra I2S ASoC driver"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 663ea9fa0ca..40540b175f5 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -39,6 +39,8 @@ #include "tegra_pcm.h" +#define DRV_NAME "tegra-pcm-audio" + static const struct snd_pcm_hardware tegra_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -377,7 +379,7 @@ static int __devexit tegra_pcm_platform_remove(struct platform_device *pdev) static struct platform_driver tegra_pcm_driver = { .driver = { - .name = "tegra-pcm-audio", + .name = DRV_NAME, .owner = THIS_MODULE, }, .probe = tegra_pcm_platform_probe, @@ -399,3 +401,4 @@ module_exit(snd_tegra_pcm_exit); MODULE_AUTHOR("Stephen Warren "); MODULE_DESCRIPTION("Tegra PCM ASoC driver"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); -- cgit v1.2.3 From 905f6952c5bc8126f1d82c2eb8a699271080b57e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 11 Feb 2011 14:39:13 +0000 Subject: ASoC: Warn if WM8903 platform data is used to enable microphone IRQ The WM8903 interrupts are clear on read so if the WM8903 detection is enabled from platform data when the IRQ is in use (rather than using a direct signal from a GPIO) status may be lost during startup. Help users spot this misconfiguration by adding a WARN_ON(). Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8903.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index c7b52a04fe1..f656a000b36 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1917,12 +1917,26 @@ static int wm8903_probe(struct snd_soc_codec *codec) /* Set up GPIOs and microphone detection */ if (pdata) { + bool mic_gpio = false; + for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { if (pdata->gpio_cfg[i] == WM8903_GPIO_NO_CONFIG) continue; snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i, pdata->gpio_cfg[i] & 0xffff); + + val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK) + >> WM8903_GP1_FN_SHIFT; + + switch (val) { + case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT: + case WM8903_GPn_FN_MICBIAS_SHORT_DETECT: + mic_gpio = true; + break; + default: + break; + } } snd_soc_write(codec, WM8903_MIC_BIAS_CONTROL_0, @@ -1933,6 +1947,14 @@ static int wm8903_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0, WM8903_WSEQ_ENA, WM8903_WSEQ_ENA); + /* If microphone detection is enabled by pdata but + * detected via IRQ then interrupts can be lost before + * the machine driver has set up microphone detection + * IRQs as the IRQs are clear on read. The detection + * will be enabled when the machine driver configures. + */ + WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA)); + wm8903->mic_delay = pdata->micdet_delay; } -- cgit v1.2.3 From d8ec598e5d0d66e68184664073e06b432e3dd140 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 14 Feb 2011 17:20:20 +0200 Subject: ASoC: omap: rx51: Use gpio_request_one to configure tvout_sel gpio Just slight cleanup to be sync with upcoming change. Signed-off-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/rx51.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 2d7d4115cb7..9411969d6f7 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -394,10 +394,10 @@ static int __init rx51_soc_init(void) if (!machine_is_nokia_rx51()) return -ENODEV; - err = gpio_request(RX51_TVOUT_SEL_GPIO, "tvout_sel"); + err = gpio_request_one(RX51_TVOUT_SEL_GPIO, + GPIOF_DIR_OUT | GPIOF_INIT_LOW, "tvout_sel"); if (err) goto err_gpio_tvout_sel; - gpio_direction_output(RX51_TVOUT_SEL_GPIO, 0); rx51_snd_device = platform_device_alloc("soc-audio", -1); if (!rx51_snd_device) { -- cgit v1.2.3 From 31164c7cf1190729292126b2d6d27d792adfcec5 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 14 Feb 2011 17:20:21 +0200 Subject: ASoC: omap: rx51: Add headset support This patch adds support for headset microphone in Nokia RX-51/N900. The mic signal from audio jack is routed to codec A LINE1L via two switches and the mic bias is coming from codec B part. First switch is the tv-out switch that is already supported and the second switch selects between voltage detection circuit and codecs. As there is no use for voltage detection at the moment the second switch is connected statically to codecs in rx51_soc_init. Headset can be active when control "Jack Function" is set to "Headset". Signed-off-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/rx51.c | 25 +++++++++++++++++++++++-- 1 file changed, 23 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 9411969d6f7..9860018cc2e 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -40,6 +40,7 @@ #define RX51_TVOUT_SEL_GPIO 40 #define RX51_JACK_DETECT_GPIO 177 +#define RX51_ECI_SW_GPIO 182 /* * REVISIT: TWL4030 GPIO base in RX-51. Now statically defined to 192. This * gpio is reserved in arch/arm/mach-omap2/board-rx51-peripherals.c @@ -50,6 +51,7 @@ enum { RX51_JACK_DISABLED, RX51_JACK_TVOUT, /* tv-out with stereo output */ RX51_JACK_HP, /* headphone: stereo output, no mic */ + RX51_JACK_HS, /* headset: stereo output with mic */ }; static int rx51_spk_func; @@ -59,11 +61,15 @@ static int rx51_jack_func; static void rx51_ext_control(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = &codec->dapm; - int hp = 0, tvout = 0; + int hp = 0, hs = 0, tvout = 0; switch (rx51_jack_func) { case RX51_JACK_TVOUT: tvout = 1; + hp = 1; + break; + case RX51_JACK_HS: + hs = 1; case RX51_JACK_HP: hp = 1; break; @@ -81,6 +87,10 @@ static void rx51_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); else snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); + if (hs) + snd_soc_dapm_enable_pin(dapm, "HS Mic"); + else + snd_soc_dapm_disable_pin(dapm, "HS Mic"); gpio_set_value(RX51_TVOUT_SEL_GPIO, tvout); @@ -240,6 +250,7 @@ static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event), SND_SOC_DAPM_MIC("DMic", NULL), SND_SOC_DAPM_HP("Headphone Jack", rx51_hp_event), + SND_SOC_DAPM_MIC("HS Mic", NULL), }; static const struct snd_soc_dapm_widget aic34_dapm_widgetsb[] = { @@ -259,11 +270,14 @@ static const struct snd_soc_dapm_route audio_map[] = { static const struct snd_soc_dapm_route audio_mapb[] = { {"b LINE2R", NULL, "MONO_LOUT"}, {"Earphone", NULL, "b HPLOUT"}, + + {"LINE1L", NULL, "b Mic Bias 2.5V"}, + {"b Mic Bias 2.5V", NULL, "HS Mic"} }; static const char *spk_function[] = {"Off", "On"}; static const char *input_function[] = {"ADC", "Digital Mic"}; -static const char *jack_function[] = {"Off", "TV-OUT", "Headphone"}; +static const char *jack_function[] = {"Off", "TV-OUT", "Headphone", "Headset"}; static const struct soc_enum rx51_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), @@ -398,6 +412,10 @@ static int __init rx51_soc_init(void) GPIOF_DIR_OUT | GPIOF_INIT_LOW, "tvout_sel"); if (err) goto err_gpio_tvout_sel; + err = gpio_request_one(RX51_ECI_SW_GPIO, + GPIOF_DIR_OUT | GPIOF_INIT_HIGH, "eci_sw"); + if (err) + goto err_gpio_eci_sw; rx51_snd_device = platform_device_alloc("soc-audio", -1); if (!rx51_snd_device) { @@ -415,6 +433,8 @@ static int __init rx51_soc_init(void) err2: platform_device_put(rx51_snd_device); err1: + gpio_free(RX51_ECI_SW_GPIO); +err_gpio_eci_sw: gpio_free(RX51_TVOUT_SEL_GPIO); err_gpio_tvout_sel: @@ -427,6 +447,7 @@ static void __exit rx51_soc_exit(void) rx51_av_jack_gpios); platform_device_unregister(rx51_snd_device); + gpio_free(RX51_ECI_SW_GPIO); gpio_free(RX51_TVOUT_SEL_GPIO); } -- cgit v1.2.3 From 178406195755293bf63799ae41ed0c92968ded8c Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 14 Feb 2011 17:20:22 +0200 Subject: ASoC: omap: rx51: Report headset insertion instead of video out cable It is more usefull to report headset instead of video out cable in response to jack insertion as this is more usual use-case and because now the headset feature is supported. Automatic accessory detection is not possible at the moment so most sensible static accessory type have to be used. Signed-off-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/rx51.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 9860018cc2e..e28f089b38d 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -240,7 +240,7 @@ static struct snd_soc_jack_gpio rx51_av_jack_gpios[] = { { .gpio = RX51_JACK_DETECT_GPIO, .name = "avdet-gpio", - .report = SND_JACK_VIDEOOUT, + .report = SND_JACK_HEADSET, .invert = 1, .debounce_time = 200, }, @@ -331,7 +331,8 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) /* AV jack detection */ err = snd_soc_jack_new(codec, "AV Jack", - SND_JACK_VIDEOOUT, &rx51_av_jack); + SND_JACK_HEADSET | SND_JACK_VIDEOOUT, + &rx51_av_jack); if (err) return err; err = snd_soc_jack_add_gpios(&rx51_av_jack, -- cgit v1.2.3 From 1461d0630ef74073e308a79834ba09e1bd2df08e Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 15 Feb 2011 18:28:51 +0530 Subject: ASoC: sn95031: make playback rails depend on actual pins they control This patch makes the codec playback rails (headset and speaker) depend on actual pins they control. This enables better power management of the codec Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index d0b78020671..9c3db0d7b75 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -519,6 +519,8 @@ static const struct snd_soc_dapm_widget sn95031_dapm_widgets[] = { static const struct snd_soc_dapm_route sn95031_audio_map[] = { /* headset and earpiece map */ + { "HPOUTL", NULL, "Headset Rail"}, + { "HPOUTR", NULL, "Headset Rail"}, { "HPOUTL", NULL, "Headset Left Playback" }, { "HPOUTR", NULL, "Headset Right Playback" }, { "EPOUT", NULL, "Earpiece Playback" }, @@ -527,18 +529,16 @@ static const struct snd_soc_dapm_route sn95031_audio_map[] = { { "Earpiece Playback", NULL, "Headset Left Filter"}, { "Headset Left Filter", NULL, "HSDAC Left"}, { "Headset Right Filter", NULL, "HSDAC Right"}, - { "HSDAC Left", NULL, "Headset Rail"}, - { "HSDAC Right", NULL, "Headset Rail"}, /* speaker map */ + { "IHFOUTL", NULL, "Speaker Rail"}, + { "IHFOUTR", NULL, "Speaker Rail"}, { "IHFOUTL", "NULL", "Speaker Left Playback"}, { "IHFOUTR", "NULL", "Speaker Right Playback"}, { "Speaker Left Playback", NULL, "Speaker Left Filter"}, { "Speaker Right Playback", NULL, "Speaker Right Filter"}, { "Speaker Left Filter", NULL, "IHFDAC Left"}, { "Speaker Right Filter", NULL, "IHFDAC Right"}, - { "IHFDAC Left", NULL, "Speaker Rail"}, - { "IHFDAC Right", NULL, "Speaker Rail"}, /* vibra map */ { "VIB1OUT", NULL, "Vibra1 Playback"}, -- cgit v1.2.3 From a62ffc92e8332135c3dbc3351c7e90031830e2f7 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 15 Feb 2011 18:28:52 +0530 Subject: ASoC: sn95031: fix the DMIC path routing This patch makes the DMIC dynamically connect to TX Mux, earlier code had erroneously made this as static path Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 48 +++++++++++++++++++++++----------------------- 1 file changed, 24 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 9c3db0d7b75..5eb39c7ed96 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -602,30 +602,30 @@ static const struct snd_soc_dapm_route sn95031_audio_map[] = { { "Txpath2 Capture Route", "ADC Right", "ADC Right"}, { "Txpath3 Capture Route", "ADC Right", "ADC Right"}, { "Txpath4 Capture Route", "ADC Right", "ADC Right"}, - { "Txpath1 Capture Route", NULL, "DMIC1"}, - { "Txpath2 Capture Route", NULL, "DMIC1"}, - { "Txpath3 Capture Route", NULL, "DMIC1"}, - { "Txpath4 Capture Route", NULL, "DMIC1"}, - { "Txpath1 Capture Route", NULL, "DMIC2"}, - { "Txpath2 Capture Route", NULL, "DMIC2"}, - { "Txpath3 Capture Route", NULL, "DMIC2"}, - { "Txpath4 Capture Route", NULL, "DMIC2"}, - { "Txpath1 Capture Route", NULL, "DMIC3"}, - { "Txpath2 Capture Route", NULL, "DMIC3"}, - { "Txpath3 Capture Route", NULL, "DMIC3"}, - { "Txpath4 Capture Route", NULL, "DMIC3"}, - { "Txpath1 Capture Route", NULL, "DMIC4"}, - { "Txpath2 Capture Route", NULL, "DMIC4"}, - { "Txpath3 Capture Route", NULL, "DMIC4"}, - { "Txpath4 Capture Route", NULL, "DMIC4"}, - { "Txpath1 Capture Route", NULL, "DMIC5"}, - { "Txpath2 Capture Route", NULL, "DMIC5"}, - { "Txpath3 Capture Route", NULL, "DMIC5"}, - { "Txpath4 Capture Route", NULL, "DMIC5"}, - { "Txpath1 Capture Route", NULL, "DMIC6"}, - { "Txpath2 Capture Route", NULL, "DMIC6"}, - { "Txpath3 Capture Route", NULL, "DMIC6"}, - { "Txpath4 Capture Route", NULL, "DMIC6"}, + { "Txpath1 Capture Route", "DMIC1", "DMIC1"}, + { "Txpath2 Capture Route", "DMIC1", "DMIC1"}, + { "Txpath3 Capture Route", "DMIC1", "DMIC1"}, + { "Txpath4 Capture Route", "DMIC1", "DMIC1"}, + { "Txpath1 Capture Route", "DMIC2", "DMIC2"}, + { "Txpath2 Capture Route", "DMIC2", "DMIC2"}, + { "Txpath3 Capture Route", "DMIC2", "DMIC2"}, + { "Txpath4 Capture Route", "DMIC2", "DMIC2"}, + { "Txpath1 Capture Route", "DMIC3", "DMIC3"}, + { "Txpath2 Capture Route", "DMIC3", "DMIC3"}, + { "Txpath3 Capture Route", "DMIC3", "DMIC3"}, + { "Txpath4 Capture Route", "DMIC3", "DMIC3"}, + { "Txpath1 Capture Route", "DMIC4", "DMIC4"}, + { "Txpath2 Capture Route", "DMIC4", "DMIC4"}, + { "Txpath3 Capture Route", "DMIC4", "DMIC4"}, + { "Txpath4 Capture Route", "DMIC4", "DMIC4"}, + { "Txpath1 Capture Route", "DMIC5", "DMIC5"}, + { "Txpath2 Capture Route", "DMIC5", "DMIC5"}, + { "Txpath3 Capture Route", "DMIC5", "DMIC5"}, + { "Txpath4 Capture Route", "DMIC5", "DMIC5"}, + { "Txpath1 Capture Route", "DMIC6", "DMIC6"}, + { "Txpath2 Capture Route", "DMIC6", "DMIC6"}, + { "Txpath3 Capture Route", "DMIC6", "DMIC6"}, + { "Txpath4 Capture Route", "DMIC6", "DMIC6"}, /* tx path */ { "TX1 Enable", NULL, "Txpath1 Capture Route"}, -- cgit v1.2.3 From 65e9625e1f86658ee869420713be3afb9a75debd Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 15 Feb 2011 18:28:53 +0530 Subject: ASoC: sn95031: fix the amic tlv scale The tlv scale is defined as (min, step, mute). The mute is not supported here so put the value to 0 Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 5eb39c7ed96..2a30eae1881 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -359,7 +359,7 @@ static const struct snd_kcontrol_new sn95031_input4_mux_control = static const char *sn95031_micmode_text[] = {"Single Ended", "Differential"}; /* 0dB to 30dB in 10dB steps */ -static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 30); +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 0); static const struct soc_enum sn95031_micmode1_enum = SOC_ENUM_SINGLE(SN95031_MICAMP1, 1, 2, sn95031_micmode_text); -- cgit v1.2.3 From d58198b943c7af6975dec869d75b15dd66d1495c Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 15 Feb 2011 18:28:55 +0530 Subject: ASoC: mfld_machine: make use of soc_register_card API This patch removes the old method of soc-audio device creation in mfld machine and makes use of new soc_register_card API to register the card Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/mid-x86/mfld_machine.c | 20 ++++++-------------- 1 file changed, 6 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 96487fb8d26..429aa1be2cf 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -392,25 +392,17 @@ static int __devinit snd_mfld_mc_probe(struct platform_device *pdev) pr_err("cannot register IRQ\n"); goto unalloc; } - /* create soc device */ - mc_drv_ctx->socdev = platform_device_alloc("soc-audio", -1); - if (!mc_drv_ctx->socdev) { - pr_err("soc-audio device allocation failed\n"); - ret_val = -ENOMEM; - goto freeirq; - } - platform_set_drvdata(mc_drv_ctx->socdev, &snd_soc_card_mfld); - ret_val = platform_device_add(mc_drv_ctx->socdev); + /* register the soc card */ + snd_soc_card_mfld.dev = &pdev->dev; + ret_val = snd_soc_register_card(&snd_soc_card_mfld); if (ret_val) { - pr_err("Unable to add soc-audio device, err %d\n", ret_val); - goto unregister; + pr_debug("snd_soc_register_card failed %d\n", ret_val); + goto freeirq; } platform_set_drvdata(pdev, mc_drv_ctx); pr_debug("successfully exited probe\n"); return ret_val; -unregister: - platform_device_put(mc_drv_ctx->socdev); freeirq: free_irq(irq, mc_drv_ctx); unalloc: @@ -424,7 +416,7 @@ static int __devexit snd_mfld_mc_remove(struct platform_device *pdev) pr_debug("snd_mfld_mc_remove called\n"); free_irq(platform_get_irq(pdev, 0), mc_drv_ctx); - platform_device_unregister(mc_drv_ctx->socdev); + snd_soc_unregister_card(&snd_soc_card_mfld); kfree(mc_drv_ctx); platform_set_drvdata(pdev, NULL); return 0; -- cgit v1.2.3 From 5b499f8bf376ecd1575c0cc2095555cf382063ae Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 15 Feb 2011 18:28:54 +0530 Subject: ASoC: sst_platform: fix the pulseaudio error Pulseaudio doesnt work with current driver and it was root caused to absense of hw_params function and malloc_pages in it. This patch adds this and allows pa to work fine with these drivers Signed-off-by: Vinod Koul Signed-off-by: Harsha Priya Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/mid-x86/sst_platform.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 96e6e9c9c5f..ee2c22475a7 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -365,6 +365,14 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer return stream->stream_info.buffer_ptr; } +static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); + + return 0; +} static struct snd_pcm_ops sst_platform_ops = { .open = sst_platform_open, @@ -373,6 +381,7 @@ static struct snd_pcm_ops sst_platform_ops = { .prepare = sst_platform_pcm_prepare, .trigger = sst_platform_pcm_trigger, .pointer = sst_platform_pcm_pointer, + .hw_params = sst_platform_pcm_hw_params, }; static void sst_pcm_free(struct snd_pcm *pcm) -- cgit v1.2.3 From 4a8d929d142ae594d8a5f0e1efba1d278d07bd8b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Feb 2011 14:57:17 -0800 Subject: ASoC: Fix missing space in WM8994 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index d78abeceaed..c0adbf8098e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -526,7 +526,7 @@ static int wm8994_get_retune_mobile_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct wm8994_priv *wm8994 =snd_soc_codec_get_drvdata(codec); + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int block = wm8994_get_retune_mobile_block(kcontrol->id.name); ucontrol->value.enumerated.item[0] = wm8994->retune_mobile_cfg[block]; -- cgit v1.2.3 From d1118aaad2f5671d252e8dc3b07944d77099835d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 16 Feb 2011 19:24:39 -0800 Subject: ASoC: Remove export of snd_soc_dapm_stream_event() The only thing that should ever be calling this is soc-core and that is built as part of the same module so doesn't need the export. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1bd24d52150..3bee4b20a7c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2263,7 +2263,6 @@ int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, mutex_unlock(&codec->mutex); return 0; } -EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); /** * snd_soc_dapm_enable_pin - enable pin. -- cgit v1.2.3 From 5a9f91ca7994bd6a7c696fd397716da3bb440921 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Feb 2011 12:05:46 -0800 Subject: ASoC: Log wm_hubs DC servo operation code when reporting a timeout Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm_hubs.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 613df5db0b3..a5c1556e32e 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -82,7 +82,8 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) } while (reg & op && count < 400); if (reg & op) - dev_err(codec->dev, "Timed out waiting for DC Servo\n"); + dev_err(codec->dev, "Timed out waiting for DC Servo %x\n", + op); } /* -- cgit v1.2.3 From 7887ab3a274dc5f1d1d94ca0cd41ae495d01f94f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Feb 2011 16:35:55 -0800 Subject: ASoC: Allow GPIO jack detection to be configured as a wake source Some systems wish to use jacks as wake sources. Provide a wake flag in the GPIO configuration which causes the driver to enable the IRQ as a wake source. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-jack.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 4579ee090bb..1382251ed2a 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -330,6 +330,14 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, if (ret) goto err; + if (gpios[i].wake) { + ret = set_irq_wake(gpio_to_irq(gpios[i].gpio), 1); + if (ret != 0) + printk(KERN_ERR + "Failed to mark GPIO %d as wake source: %d\n", + gpios[i].gpio, ret); + } + #ifdef CONFIG_GPIO_SYSFS /* Expose GPIO value over sysfs for diagnostic purposes */ gpio_export(gpios[i].gpio, false); -- cgit v1.2.3 From 40d2f1592ac262f13487b12cb47057e65b947bce Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 18 Feb 2011 14:47:02 -0800 Subject: ASoC: Mark WM8958 microphone detection registers readable So they show up in codec_reg. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994-tables.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c index 68e9b024dd4..42b3248ba2f 100644 --- a/sound/soc/codecs/wm8994-tables.c +++ b/sound/soc/codecs/wm8994-tables.c @@ -209,9 +209,9 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x0000, 0x0000 }, /* R205 */ { 0x0000, 0x0000 }, /* R206 */ { 0x0000, 0x0000 }, /* R207 */ - { 0x0000, 0x0000 }, /* R208 */ - { 0x0000, 0x0000 }, /* R209 */ - { 0x0000, 0x0000 }, /* R210 */ + { 0xFFFF, 0xFFFF }, /* R208 */ + { 0xFFFF, 0xFFFF }, /* R209 */ + { 0xFFFF, 0xFFFF }, /* R210 */ { 0x0000, 0x0000 }, /* R211 */ { 0x0000, 0x0000 }, /* R212 */ { 0x0000, 0x0000 }, /* R213 */ -- cgit v1.2.3 From 4baafdd76bafc51699a924d2bc1317f50b4ea75f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 18 Feb 2011 15:05:53 -0800 Subject: ASoC: Hook wm_hubs micbiases up to CLK_SYS The microphone detection functionality requires a clock to work. In any non-detection case where the MICBIAS is enabled CLK_SYS will be needed anyway so there is no negative impact on power consumption. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index a5c1556e32e..7b6b3c18e29 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -675,6 +675,9 @@ SND_SOC_DAPM_OUTPUT("LINEOUT2N"), }; static const struct snd_soc_dapm_route analogue_routes[] = { + { "MICBIAS1", NULL, "CLK_SYS" }, + { "MICBIAS2", NULL, "CLK_SYS" }, + { "IN1L PGA", "IN1LP Switch", "IN1LP" }, { "IN1L PGA", "IN1LN Switch", "IN1LN" }, -- cgit v1.2.3 From 9d7e584b3fb4d9f581b83c0916e10b91de78e663 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 21 Feb 2011 14:57:22 +0200 Subject: ASoC: omap: rx51: Add FM transmitter support Si4713 FM transmitter on Nokia RX-51/N900 is connected to same Line out signals of TLV320AIC34 than TPA6130 headphone amplifier. This patch adds route to transmitter and "FM Transmitter" control to keep route active when needed. Signed-off-by: Jarkko Nikula Acked-by: Mark Brown Acked-by: Peter Ujfalusi Signed-off-by: Liam Girdwood --- sound/soc/omap/rx51.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index e28f089b38d..055e447b2ed 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -251,6 +251,7 @@ static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = { SND_SOC_DAPM_MIC("DMic", NULL), SND_SOC_DAPM_HP("Headphone Jack", rx51_hp_event), SND_SOC_DAPM_MIC("HS Mic", NULL), + SND_SOC_DAPM_LINE("FM Transmitter", NULL), }; static const struct snd_soc_dapm_widget aic34_dapm_widgetsb[] = { @@ -262,6 +263,8 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Ext Spk", NULL, "HPROUT"}, {"Headphone Jack", NULL, "LLOUT"}, {"Headphone Jack", NULL, "RLOUT"}, + {"FM Transmitter", NULL, "LLOUT"}, + {"FM Transmitter", NULL, "RLOUT"}, {"DMic Rate 64", NULL, "Mic Bias 2V"}, {"Mic Bias 2V", NULL, "DMic"}, @@ -292,6 +295,7 @@ static const struct snd_kcontrol_new aic34_rx51_controls[] = { rx51_get_input, rx51_set_input), SOC_ENUM_EXT("Jack Function", rx51_enum[2], rx51_get_jack, rx51_set_jack), + SOC_DAPM_PIN_SWITCH("FM Transmitter"), }; static const struct snd_kcontrol_new aic34_rx51_controlsb[] = { -- cgit v1.2.3 From ed5a4c472346e4bc1de37f84bb8d2aca26e59538 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 18 Feb 2011 11:12:42 -0800 Subject: ASoC: Remove card from snd_soc_dapm_set_bias_level() We can get the card from the DAPM context so don't bother passing it as an argument. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3bee4b20a7c..1a932bbe825 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -125,17 +125,17 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( /** * snd_soc_dapm_set_bias_level - set the bias level for the system - * @card: audio device + * @dapm: DAPM context * @level: level to configure * * Configure the bias (power) levels for the SoC audio device. * * Returns 0 for success else error. */ -static int snd_soc_dapm_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, +static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { + struct snd_soc_card *card = dapm->card; int ret = 0; switch (level) { @@ -1112,7 +1112,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) list_for_each_entry(d, &dapm->card->dapm_list, list) { if (d->dev_power && d->bias_level == SND_SOC_BIAS_OFF) { - ret = snd_soc_dapm_set_bias_level(card, d, + ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY); if (ret != 0) dev_err(d->dev, @@ -1122,7 +1122,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) /* If we're changing to all on or all off then prepare */ if ((d->dev_power && d->bias_level == SND_SOC_BIAS_STANDBY) || (!d->dev_power && d->bias_level == SND_SOC_BIAS_ON)) { - ret = snd_soc_dapm_set_bias_level(card, d, + ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE); if (ret != 0) dev_err(d->dev, @@ -1141,7 +1141,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) list_for_each_entry(d, &dapm->card->dapm_list, list) { /* If we just powered the last thing off drop to standby bias */ if (d->bias_level == SND_SOC_BIAS_PREPARE && !d->dev_power) { - ret = snd_soc_dapm_set_bias_level(card, d, + ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY); if (ret != 0) dev_err(d->dev, @@ -1152,7 +1152,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) /* If we're in standby and can support bias off then do that */ if (d->bias_level == SND_SOC_BIAS_STANDBY && d->idle_bias_off) { - ret = snd_soc_dapm_set_bias_level(card, d, + ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_OFF); if (ret != 0) dev_err(d->dev, @@ -1161,7 +1161,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) /* If we just powered up then move to active bias */ if (d->bias_level == SND_SOC_BIAS_PREPARE && d->dev_power) { - ret = snd_soc_dapm_set_bias_level(card, d, + ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_ON); if (ret != 0) dev_err(d->dev, @@ -2439,9 +2439,9 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) * standby. */ if (powerdown) { - snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_PREPARE); + snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE); dapm_seq_run(dapm, &down_list, 0, false); - snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_STANDBY); + snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY); } } @@ -2454,7 +2454,7 @@ void snd_soc_dapm_shutdown(struct snd_soc_card *card) list_for_each_entry(codec, &card->codec_dev_list, list) { soc_dapm_shutdown_codec(&codec->dapm); - snd_soc_dapm_set_bias_level(card, &codec->dapm, SND_SOC_BIAS_OFF); + snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF); } } -- cgit v1.2.3 From 9d0624a740b773d961fe0dc454d748c60d892b71 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 18 Feb 2011 11:49:43 -0800 Subject: ASoC: Run bias level changes for all DAPM contexts in parallel As bias level changes can be quite time consuming and the bias changes for multiple devices aren't strongly tied to each other (if anything it can be advantageous to bring different devices up together) we can improve the state transition time for multi-component systems by running the bias level changes for all the devices in parallel. This is very simple to achieve using the kernel async functionality so use that to schedule the work. This should have no practical effect for the overwhelming majority of systems which have a single DAPM context - we'll bounce into another thread to do the bias level change but otherwise everything will happen in exactly the same order as it did before. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 118 +++++++++++++++++++++++++++++---------------------- 1 file changed, 67 insertions(+), 51 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1a932bbe825..200ae7cd2f0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -32,6 +32,7 @@ #include #include #include +#include #include #include #include @@ -1014,7 +1015,62 @@ static void dapm_widget_update(struct snd_soc_dapm_context *dapm) } } +/* Async callback run prior to DAPM sequences - brings to _PREPARE if + * they're changing state. + */ +static void dapm_pre_sequence_async(void *data, async_cookie_t cookie) +{ + struct snd_soc_dapm_context *d = data; + int ret; + + if (d->dev_power && d->bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY); + if (ret != 0) + dev_err(d->dev, + "Failed to turn on bias: %d\n", ret); + } + + /* If we're changing to all on or all off then prepare */ + if ((d->dev_power && d->bias_level == SND_SOC_BIAS_STANDBY) || + (!d->dev_power && d->bias_level == SND_SOC_BIAS_ON)) { + ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE); + if (ret != 0) + dev_err(d->dev, + "Failed to prepare bias: %d\n", ret); + } +} + +/* Async callback run prior to DAPM sequences - brings to their final + * state. + */ +static void dapm_post_sequence_async(void *data, async_cookie_t cookie) +{ + struct snd_soc_dapm_context *d = data; + int ret; + + /* If we just powered the last thing off drop to standby bias */ + if (d->bias_level == SND_SOC_BIAS_PREPARE && !d->dev_power) { + ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY); + if (ret != 0) + dev_err(d->dev, "Failed to apply standby bias: %d\n", + ret); + } + /* If we're in standby and can support bias off then do that */ + if (d->bias_level == SND_SOC_BIAS_STANDBY && d->idle_bias_off) { + ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_OFF); + if (ret != 0) + dev_err(d->dev, "Failed to turn off bias: %d\n", ret); + } + + /* If we just powered up then move to active bias */ + if (d->bias_level == SND_SOC_BIAS_PREPARE && d->dev_power) { + ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_ON); + if (ret != 0) + dev_err(d->dev, "Failed to apply active bias: %d\n", + ret); + } +} /* * Scan each dapm widget for complete audio path. @@ -1032,7 +1088,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) struct snd_soc_dapm_context *d; LIST_HEAD(up_list); LIST_HEAD(down_list); - int ret = 0; + LIST_HEAD(async_domain); int power; trace_snd_soc_dapm_start(card); @@ -1110,25 +1166,11 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } } - list_for_each_entry(d, &dapm->card->dapm_list, list) { - if (d->dev_power && d->bias_level == SND_SOC_BIAS_OFF) { - ret = snd_soc_dapm_set_bias_level(d, - SND_SOC_BIAS_STANDBY); - if (ret != 0) - dev_err(d->dev, - "Failed to turn on bias: %d\n", ret); - } - - /* If we're changing to all on or all off then prepare */ - if ((d->dev_power && d->bias_level == SND_SOC_BIAS_STANDBY) || - (!d->dev_power && d->bias_level == SND_SOC_BIAS_ON)) { - ret = snd_soc_dapm_set_bias_level(d, - SND_SOC_BIAS_PREPARE); - if (ret != 0) - dev_err(d->dev, - "Failed to prepare bias: %d\n", ret); - } - } + /* Run all the bias changes in parallel */ + list_for_each_entry(d, &dapm->card->dapm_list, list) + async_schedule_domain(dapm_pre_sequence_async, d, + &async_domain); + async_synchronize_full_domain(&async_domain); /* Power down widgets first; try to avoid amplifying pops. */ dapm_seq_run(dapm, &down_list, event, false); @@ -1138,37 +1180,11 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) /* Now power up. */ dapm_seq_run(dapm, &up_list, event, true); - list_for_each_entry(d, &dapm->card->dapm_list, list) { - /* If we just powered the last thing off drop to standby bias */ - if (d->bias_level == SND_SOC_BIAS_PREPARE && !d->dev_power) { - ret = snd_soc_dapm_set_bias_level(d, - SND_SOC_BIAS_STANDBY); - if (ret != 0) - dev_err(d->dev, - "Failed to apply standby bias: %d\n", - ret); - } - - /* If we're in standby and can support bias off then do that */ - if (d->bias_level == SND_SOC_BIAS_STANDBY && - d->idle_bias_off) { - ret = snd_soc_dapm_set_bias_level(d, - SND_SOC_BIAS_OFF); - if (ret != 0) - dev_err(d->dev, - "Failed to turn off bias: %d\n", ret); - } - - /* If we just powered up then move to active bias */ - if (d->bias_level == SND_SOC_BIAS_PREPARE && d->dev_power) { - ret = snd_soc_dapm_set_bias_level(d, - SND_SOC_BIAS_ON); - if (ret != 0) - dev_err(d->dev, - "Failed to apply active bias: %d\n", - ret); - } - } + /* Run all the bias changes in parallel */ + list_for_each_entry(d, &dapm->card->dapm_list, list) + async_schedule_domain(dapm_post_sequence_async, d, + &async_domain); + async_synchronize_full_domain(&async_domain); pop_dbg(dapm->dev, card->pop_time, "DAPM sequencing finished, waiting %dms\n", card->pop_time); -- cgit v1.2.3 From 7d700ac8d91f63f25cb58edeba06caddc65d85b0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 18 Feb 2011 16:57:22 -0800 Subject: ASoC: Mark WM8958 microphone bias registers as readable Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994-tables.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c index 42b3248ba2f..61dfd91c6c7 100644 --- a/sound/soc/codecs/wm8994-tables.c +++ b/sound/soc/codecs/wm8994-tables.c @@ -62,8 +62,8 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x00FF, 0x00FF }, /* R58 - MICBIAS */ { 0x000F, 0x000F }, /* R59 - LDO 1 */ { 0x0007, 0x0007 }, /* R60 - LDO 2 */ - { 0x0000, 0x0000 }, /* R61 */ - { 0x0000, 0x0000 }, /* R62 */ + { 0xFFFF, 0xFFFF }, /* R61 */ + { 0xFFFF, 0xFFFF }, /* R62 */ { 0x0000, 0x0000 }, /* R63 */ { 0x0000, 0x0000 }, /* R64 */ { 0x0000, 0x0000 }, /* R65 */ -- cgit v1.2.3 From 9b7c525dfaa9a1b5f01db1f3a1edc50bbb6eb739 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 17 Feb 2011 20:05:44 -0800 Subject: ASoC: Support WM8958 direct microphone detection IRQ Allow direct routing of the WM8958 microphone detection signal to a GPIO to be used, saving the need to demux the interrupt. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 57 +++++++++++++++++++++++++++++------------------ 1 file changed, 35 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b23e91027d6..1ad6e3db780 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -104,6 +104,7 @@ struct wm8994_priv { void *jack_cb_data; bool jack_is_mic; bool jack_is_video; + int micdet_irq; int revision; struct wm8994_pdata *pdata; @@ -3102,6 +3103,12 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->pdata = dev_get_platdata(codec->dev->parent); wm8994->codec = codec; + if (wm8994->pdata && wm8994->pdata->micdet_irq) + wm8994->micdet_irq = wm8994->pdata->micdet_irq; + else if (wm8994->pdata && wm8994->pdata->irq_base) + wm8994->micdet_irq = wm8994->pdata->irq_base + + WM8994_IRQ_MIC1_DET; + pm_runtime_enable(codec->dev); pm_runtime_resume(codec->dev); @@ -3150,14 +3157,17 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (control->type) { case WM8994: - ret = wm8994_request_irq(codec->control_data, - WM8994_IRQ_MIC1_DET, - wm8994_mic_irq, "Mic 1 detect", - wm8994); - if (ret != 0) - dev_warn(codec->dev, - "Failed to request Mic1 detect IRQ: %d\n", - ret); + if (wm8994->micdet_irq) { + ret = request_threaded_irq(wm8994->micdet_irq, NULL, + wm8994_mic_irq, + IRQF_TRIGGER_RISING, + "Mic1 detect", + wm8994); + if (ret != 0) + dev_warn(codec->dev, + "Failed to request Mic1 detect IRQ: %d\n", + ret); + } ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, @@ -3188,15 +3198,17 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; case WM8958: - ret = wm8994_request_irq(codec->control_data, - WM8994_IRQ_MIC1_DET, - wm8958_mic_irq, "Mic detect", - wm8994); - if (ret != 0) - dev_warn(codec->dev, - "Failed to request Mic detect IRQ: %d\n", - ret); - break; + if (wm8994->micdet_irq) { + ret = request_threaded_irq(wm8994->micdet_irq, NULL, + wm8958_mic_irq, + IRQF_TRIGGER_RISING, + "Mic detect", + wm8994); + if (ret != 0) + dev_warn(codec->dev, + "Failed to request Mic detect IRQ: %d\n", + ret); + } } /* Remember if AIFnLRCLK is configured as a GPIO. This should be @@ -3328,7 +3340,8 @@ err_irq: wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_SHRT, wm8994); wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET, wm8994); wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, wm8994); - wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET, wm8994); + if (wm8994->micdet_irq) + free_irq(wm8994->micdet_irq, wm8994); err: kfree(wm8994); return ret; @@ -3345,8 +3358,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) switch (control->type) { case WM8994: - wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_SHRT, - wm8994); + if (wm8994->micdet_irq) + free_irq(wm8994->micdet_irq, wm8994); wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET, wm8994); wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, @@ -3356,8 +3369,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) break; case WM8958: - wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET, - wm8994); + if (wm8994->micdet_irq) + free_irq(wm8994->micdet_irq, wm8994); break; } kfree(wm8994->retune_mobile_texts); -- cgit v1.2.3 From 48e028eccabc9c246bfad175262582a1ce34a316 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 21 Feb 2011 17:11:59 -0800 Subject: ASoC: Support configuration of WM8958 microphone bias analogue parameters The WM8958 has a different microphone bias architecture to WM8994 so needs different configuration to WM8994. Support this in platform data. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1ad6e3db780..9b9c15ffb7d 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2855,6 +2855,13 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) else snd_soc_add_controls(wm8994->codec, wm8994_eq_controls, ARRAY_SIZE(wm8994_eq_controls)); + + for (i = 0; i < ARRAY_SIZE(pdata->micbias); i++) { + if (pdata->micbias[i]) { + snd_soc_write(codec, WM8958_MICBIAS1 + i, + pdata->micbias[i] & 0xffff); + } + } } /** -- cgit v1.2.3 From 864c4bd2487619564acd75fdcf1a4349992e9090 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 21 Feb 2011 20:51:13 -0800 Subject: ASoC: Simplify default WM8958 jack detection code The default WM8958 jack detection handler implements a full set of buttons and also support for video detection. Support for multi-button jacks is fairly system specific and will usually require some tuning for headsets so simplify the implementation to only report a simple short to ground button, leaving multi-button headsets to be handled by system specific code. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 38 ++++---------------------------------- 1 file changed, 4 insertions(+), 34 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 9b9c15ffb7d..0dc14115f10 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -102,8 +102,6 @@ struct wm8994_priv { wm8958_micdet_cb jack_cb; void *jack_cb_data; - bool jack_is_mic; - bool jack_is_video; int micdet_irq; int revision; @@ -2972,46 +2970,18 @@ static void wm8958_default_micdet(u16 status, void *data) int report = 0; /* If nothing present then clear our statuses */ - if (!(status & WM8958_MICD_STS)) { - wm8994->jack_is_video = false; - wm8994->jack_is_mic = false; + if (!(status & WM8958_MICD_STS)) goto done; - } - - /* Assume anything over 475 ohms is a microphone and remember - * that we've seen one (since buttons override it) */ - if (status & 0x600) - wm8994->jack_is_mic = true; - if (wm8994->jack_is_mic) - report |= SND_JACK_MICROPHONE; - /* Video has an impedence of approximately 75 ohms; assume - * this isn't used as a button and remember it since buttons - * override it. */ - if (status & 0x40) - wm8994->jack_is_video = true; - if (wm8994->jack_is_video) - report |= SND_JACK_VIDEOOUT; + report = SND_JACK_MICROPHONE; /* Everything else is buttons; just assign slots */ - if (status & 0x4) + if (status & 0x1c0) report |= SND_JACK_BTN_0; - if (status & 0x8) - report |= SND_JACK_BTN_1; - if (status & 0x10) - report |= SND_JACK_BTN_2; - if (status & 0x20) - report |= SND_JACK_BTN_3; - if (status & 0x80) - report |= SND_JACK_BTN_4; - if (status & 0x100) - report |= SND_JACK_BTN_5; done: snd_soc_jack_report(wm8994->micdet[0].jack, report, - SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | - SND_JACK_BTN_3 | SND_JACK_BTN_4 | SND_JACK_BTN_5 | - SND_JACK_MICROPHONE | SND_JACK_VIDEOOUT); + SND_JACK_BTN_0 | SND_JACK_MICROPHONE); } /** -- cgit v1.2.3 From 1d2c27f94142e4ee9248dac1c93e001b707f3b74 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 22 Feb 2011 12:42:35 -0800 Subject: ASoC: Pass the jack to jack notifiers We're currently not passing anything and this will make the card and so on more discoverable. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 1382251ed2a..fcab80b36a3 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -101,7 +101,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) } /* Report before the DAPM sync to help users updating micbias status */ - blocking_notifier_call_chain(&jack->notifier, status, NULL); + blocking_notifier_call_chain(&jack->notifier, status, jack); snd_soc_dapm_sync(dapm); -- cgit v1.2.3 From 2031c0645c8e2f3935254bf201b4f6f00dd0790d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 24 Feb 2011 20:26:02 +0000 Subject: ASoC: Remove -codec suffix from WM9081 driver Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm9081.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 5c224dd917d..b62e98942cf 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1363,7 +1363,7 @@ MODULE_DEVICE_TABLE(i2c, wm9081_i2c_id); static struct i2c_driver wm9081_i2c_driver = { .driver = { - .name = "wm9081-codec", + .name = "wm9081", .owner = THIS_MODULE, }, .probe = wm9081_i2c_probe, -- cgit v1.2.3 From 67f5ed6e716bf9b574c8cbe04765171de5d4d0fe Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Thu, 24 Feb 2011 17:09:32 +0000 Subject: ASoC: soc-dapm: Include quotes around contents in debugfs entries Sometimes the name of the control switch of a dapm route contains spaces which makes it impossible to distinguish it from the source widget. Add quotes around the names of the widgets to makes these parsable. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 200ae7cd2f0..b8e6ab739de 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1242,7 +1242,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (p->connect) ret += snprintf(buf + ret, PAGE_SIZE - ret, - " in %s %s\n", + " in \"%s\" \"%s\"\n", p->name ? p->name : "static", p->source->name); } @@ -1252,7 +1252,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (p->connect) ret += snprintf(buf + ret, PAGE_SIZE - ret, - " out %s %s\n", + " out \"%s\" \"%s\"\n", p->name ? p->name : "static", p->sink->name); } -- cgit v1.2.3 From a3cd50deef7ba7df506af5347764a00b4621d7a7 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 22 Feb 2011 17:23:56 -0700 Subject: ASoC: Tegra: Move utilities to separate module The utilities will be required by every machine driver. Including the utility object directly into every machine driver causes a build failure if the modules are actually built into the kernel, since each will define the symbols exported by the utility file. Solve this by moving the utility object into a separate module. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/Makefile | 3 ++- sound/soc/tegra/tegra_asoc_utils.c | 6 ++++++ 2 files changed, 8 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index dfd2ab9d975..fd183d3ab4f 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -2,13 +2,14 @@ snd-soc-tegra-das-objs := tegra_das.o snd-soc-tegra-pcm-objs := tegra_pcm.o snd-soc-tegra-i2s-objs := tegra_i2s.o +snd-soc-tegra-utils-objs += tegra_asoc_utils.o +obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-utils.o obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-das.o obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-pcm.o obj-$(CONFIG_SND_TEGRA_SOC_I2S) += snd-soc-tegra-i2s.o # Tegra machine Support snd-soc-tegra-harmony-objs := harmony.o -snd-soc-tegra-harmony-objs += tegra_asoc_utils.o obj-$(CONFIG_SND_TEGRA_SOC_HARMONY) += snd-soc-tegra-harmony.o diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index cb4fc13c7d2..52f0a3f9ce4 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -101,6 +101,7 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, return 0; } +EXPORT_SYMBOL_GPL(tegra_asoc_utils_set_rate); int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, struct device *dev) @@ -139,6 +140,7 @@ err_put_pll_a: err: return ret; } +EXPORT_SYMBOL_GPL(tegra_asoc_utils_init); void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data) { @@ -146,4 +148,8 @@ void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data) clk_put(data->clk_pll_a_out0); clk_put(data->clk_pll_a); } +EXPORT_SYMBOL_GPL(tegra_asoc_utils_fini); +MODULE_AUTHOR("Stephen Warren "); +MODULE_DESCRIPTION("Tegra ASoC utility code"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From ceade6c81bd913b23bffb3d80909ea430850c79b Mon Sep 17 00:00:00 2001 From: "Justin P. Mattock" Date: Thu, 24 Feb 2011 22:16:37 -0800 Subject: ASoC: remove one to many l's in the word The patch below removes an extra "l" in the word. Signed-off-by: Justin P. Mattock Acked-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/samsung/dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 9bce1df1f0d..5cb3b880f0d 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -310,7 +310,7 @@ dma_pointer(struct snd_pcm_substream *substream) /* we seem to be getting the odd error from the pcm library due * to out-of-bounds pointers. this is maybe due to the dma engine * not having loaded the new values for the channel before being - * callled... (todo - fix ) + * called... (todo - fix ) */ if (res >= snd_pcm_lib_buffer_bytes(substream)) { -- cgit v1.2.3 From 9b34e6cc3bc2bf826c078c93c81b46f6c08da25f Mon Sep 17 00:00:00 2001 From: Zeng Zhaoming Date: Thu, 24 Feb 2011 02:08:21 +0800 Subject: ASoC: Add Freescale SGTL5000 codec support Add Freescale SGTL5000 codec support. Supported features: - line-in and mic input - headphone and line-out output - line-in bypass ADC and DAC to headphone - 16, 20, 24, 32 bit audio - 8 ~ 96k sample rates Signed-off-by: Zeng Zhaoming Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sgtl5000.c | 1512 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sgtl5000.h | 402 ++++++++++++ 4 files changed, 1921 insertions(+) create mode 100644 sound/soc/codecs/sgtl5000.c create mode 100644 sound/soc/codecs/sgtl5000.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e239345a4d5..c04da187129 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -33,6 +33,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 + select SND_SOC_SGTL5000 if I2C select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C @@ -182,6 +183,10 @@ config SND_SOC_MAX98088 config SND_SOC_PCM3008 tristate +#Freescale sgtl5000 codec +config SND_SOC_SGTL5000 + tristate + config SND_SOC_SN95031 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ae10507dd2e..3bbb08c512d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -19,6 +19,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-objs := spdif_transciever.o @@ -103,6 +104,7 @@ obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c new file mode 100644 index 00000000000..9eb206e852d --- /dev/null +++ b/sound/soc/codecs/sgtl5000.c @@ -0,0 +1,1512 @@ +/* + * sgtl5000.c -- SGTL5000 ALSA SoC Audio driver + * + * Copyright 2010-2011 Freescale Semiconductor, Inc. All Rights Reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "sgtl5000.h" + +#define SGTL5000_DAP_REG_OFFSET 0x0100 +#define SGTL5000_MAX_REG_OFFSET 0x013A + +/* default value of sgtl5000 registers except DAP */ +static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET >> 1] = { + 0xa011, /* 0x0000, CHIP_ID. 11 stand for revison 17 */ + 0x0000, /* 0x0002, CHIP_DIG_POWER. */ + 0x0008, /* 0x0004, CHIP_CKL_CTRL */ + 0x0010, /* 0x0006, CHIP_I2S_CTRL */ + 0x0000, /* 0x0008, reserved */ + 0x0008, /* 0x000A, CHIP_SSS_CTRL */ + 0x0000, /* 0x000C, reserved */ + 0x020c, /* 0x000E, CHIP_ADCDAC_CTRL */ + 0x3c3c, /* 0x0010, CHIP_DAC_VOL */ + 0x0000, /* 0x0012, reserved */ + 0x015f, /* 0x0014, CHIP_PAD_STRENGTH */ + 0x0000, /* 0x0016, reserved */ + 0x0000, /* 0x0018, reserved */ + 0x0000, /* 0x001A, reserved */ + 0x0000, /* 0x001E, reserved */ + 0x0000, /* 0x0020, CHIP_ANA_ADC_CTRL */ + 0x1818, /* 0x0022, CHIP_ANA_HP_CTRL */ + 0x0111, /* 0x0024, CHIP_ANN_CTRL */ + 0x0000, /* 0x0026, CHIP_LINREG_CTRL */ + 0x0000, /* 0x0028, CHIP_REF_CTRL */ + 0x0000, /* 0x002A, CHIP_MIC_CTRL */ + 0x0000, /* 0x002C, CHIP_LINE_OUT_CTRL */ + 0x0404, /* 0x002E, CHIP_LINE_OUT_VOL */ + 0x7060, /* 0x0030, CHIP_ANA_POWER */ + 0x5000, /* 0x0032, CHIP_PLL_CTRL */ + 0x0000, /* 0x0034, CHIP_CLK_TOP_CTRL */ + 0x0000, /* 0x0036, CHIP_ANA_STATUS */ + 0x0000, /* 0x0038, reserved */ + 0x0000, /* 0x003A, CHIP_ANA_TEST2 */ + 0x0000, /* 0x003C, CHIP_SHORT_CTRL */ + 0x0000, /* reserved */ +}; + +/* default value of dap registers */ +static const u16 sgtl5000_dap_regs[] = { + 0x0000, /* 0x0100, DAP_CONTROL */ + 0x0000, /* 0x0102, DAP_PEQ */ + 0x0040, /* 0x0104, DAP_BASS_ENHANCE */ + 0x051f, /* 0x0106, DAP_BASS_ENHANCE_CTRL */ + 0x0000, /* 0x0108, DAP_AUDIO_EQ */ + 0x0040, /* 0x010A, DAP_SGTL_SURROUND */ + 0x0000, /* 0x010C, DAP_FILTER_COEF_ACCESS */ + 0x0000, /* 0x010E, DAP_COEF_WR_B0_MSB */ + 0x0000, /* 0x0110, DAP_COEF_WR_B0_LSB */ + 0x0000, /* 0x0112, reserved */ + 0x0000, /* 0x0114, reserved */ + 0x002f, /* 0x0116, DAP_AUDIO_EQ_BASS_BAND0 */ + 0x002f, /* 0x0118, DAP_AUDIO_EQ_BAND0 */ + 0x002f, /* 0x011A, DAP_AUDIO_EQ_BAND2 */ + 0x002f, /* 0x011C, DAP_AUDIO_EQ_BAND3 */ + 0x002f, /* 0x011E, DAP_AUDIO_EQ_TREBLE_BAND4 */ + 0x8000, /* 0x0120, DAP_MAIN_CHAN */ + 0x0000, /* 0x0122, DAP_MIX_CHAN */ + 0x0510, /* 0x0124, DAP_AVC_CTRL */ + 0x1473, /* 0x0126, DAP_AVC_THRESHOLD */ + 0x0028, /* 0x0128, DAP_AVC_ATTACK */ + 0x0050, /* 0x012A, DAP_AVC_DECAY */ + 0x0000, /* 0x012C, DAP_COEF_WR_B1_MSB */ + 0x0000, /* 0x012E, DAP_COEF_WR_B1_LSB */ + 0x0000, /* 0x0130, DAP_COEF_WR_B2_MSB */ + 0x0000, /* 0x0132, DAP_COEF_WR_B2_LSB */ + 0x0000, /* 0x0134, DAP_COEF_WR_A1_MSB */ + 0x0000, /* 0x0136, DAP_COEF_WR_A1_LSB */ + 0x0000, /* 0x0138, DAP_COEF_WR_A2_MSB */ + 0x0000, /* 0x013A, DAP_COEF_WR_A2_LSB */ +}; + +/* regulator supplies for sgtl5000, VDDD is an optional external supply */ +enum sgtl5000_regulator_supplies { + VDDA, + VDDIO, + VDDD, + SGTL5000_SUPPLY_NUM +}; + +/* vddd is optional supply */ +static const char *supply_names[SGTL5000_SUPPLY_NUM] = { + "VDDA", + "VDDIO", + "VDDD" +}; + +#define LDO_CONSUMER_NAME "VDDD_LDO" +#define LDO_VOLTAGE 1200000 + +static struct regulator_consumer_supply ldo_consumer[] = { + REGULATOR_SUPPLY(LDO_CONSUMER_NAME, NULL), +}; + +struct regulator_init_data ldo_init_data = { + .constraints = { + .min_uV = 850000, + .max_uV = 1600000, + .valid_modes_mask = REGULATOR_MODE_NORMAL, + .valid_ops_mask = REGULATOR_CHANGE_STATUS, + }, + .num_consumer_supplies = 1, + .consumer_supplies = &ldo_consumer[0], +}; + +/* + * sgtl5000 internal ldo regulator, + * enabled when VDDD not provided + */ +struct ldo_regulator { + struct regulator_desc desc; + struct regulator_dev *dev; + int voltage; + void *codec_data; + bool enabled; +}; + +/* sgtl5000 private structure in codec */ +struct sgtl5000_priv { + int sysclk; /* sysclk rate */ + int master; /* i2s master or not */ + int fmt; /* i2s data format */ + struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM]; + struct ldo_regulator *ldo; +}; + +/* + * mic_bias power on/off share the same register bits with + * output impedance of mic bias, when power on mic bias, we + * need reclaim it to impedance value. + * 0x0 = Powered off + * 0x1 = 2Kohm + * 0x2 = 4Kohm + * 0x3 = 8Kohm + */ +static int mic_bias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* change mic bias resistor to 4Kohm */ + snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, + SGTL5000_BIAS_R_4k, SGTL5000_BIAS_R_4k); + break; + + case SND_SOC_DAPM_PRE_PMD: + /* + * SGTL5000_BIAS_R_8k as mask to clean the two bits + * of mic bias and output impedance + */ + snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, + SGTL5000_BIAS_R_8k, 0); + break; + } + return 0; +} + +/* + * using codec assist to small pop, hp_powerup or lineout_powerup + * should stay setting until vag_powerup is fully ramped down, + * vag fully ramped down require 400ms. + */ +static int small_pop_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, 0); + msleep(400); + break; + default: + break; + } + + return 0; +} + +/* input sources for ADC */ +static const char *adc_mux_text[] = { + "MIC_IN", "LINE_IN" +}; + +static const struct soc_enum adc_enum = +SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 2, 2, adc_mux_text); + +static const struct snd_kcontrol_new adc_mux = +SOC_DAPM_ENUM("Capture Mux", adc_enum); + +/* input sources for DAC */ +static const char *dac_mux_text[] = { + "DAC", "LINE_IN" +}; + +static const struct soc_enum dac_enum = +SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 6, 2, dac_mux_text); + +static const struct snd_kcontrol_new dac_mux = +SOC_DAPM_ENUM("Headphone Mux", dac_enum); + +static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("LINE_IN"), + SND_SOC_DAPM_INPUT("MIC_IN"), + + SND_SOC_DAPM_OUTPUT("HP_OUT"), + SND_SOC_DAPM_OUTPUT("LINE_OUT"), + + SND_SOC_DAPM_MICBIAS_E("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0, + mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0, + small_pop_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0, + small_pop_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux), + SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux), + + /* aif for i2s input */ + SND_SOC_DAPM_AIF_IN("AIFIN", "Playback", + 0, SGTL5000_CHIP_DIG_POWER, + 0, 0), + + /* aif for i2s output */ + SND_SOC_DAPM_AIF_OUT("AIFOUT", "Capture", + 0, SGTL5000_CHIP_DIG_POWER, + 1, 0), + + SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), + + SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), +}; + +/* routes for sgtl5000 */ +static const struct snd_soc_dapm_route audio_map[] = { + {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ + {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ + + {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ + {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ + + {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ + {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ + {"LO", NULL, "DAC"}, /* dac --> line_out */ + + {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */ + {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */ + + {"LINE_OUT", NULL, "LO"}, + {"HP_OUT", NULL, "HP"}, +}; + +/* custom function to fetch info of PCM playback volume */ +static int dac_info_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0xfc - 0x3c; + return 0; +} + +/* + * custom function to get of PCM playback volume + * + * dac volume register + * 15-------------8-7--------------0 + * | R channel vol | L channel vol | + * ------------------------------- + * + * PCM volume with 0.5017 dB steps from 0 to -90 dB + * + * register values map to dB + * 0x3B and less = Reserved + * 0x3C = 0 dB + * 0x3D = -0.5 dB + * 0xF0 = -90 dB + * 0xFC and greater = Muted + * + * register value map to userspace value + * + * register value 0x3c(0dB) 0xf0(-90dB)0xfc + * ------------------------------ + * userspace value 0xc0 0 + */ +static int dac_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg; + int l; + int r; + + reg = snd_soc_read(codec, SGTL5000_CHIP_DAC_VOL); + + /* get left channel volume */ + l = (reg & SGTL5000_DAC_VOL_LEFT_MASK) >> SGTL5000_DAC_VOL_LEFT_SHIFT; + + /* get right channel volume */ + r = (reg & SGTL5000_DAC_VOL_RIGHT_MASK) >> SGTL5000_DAC_VOL_RIGHT_SHIFT; + + /* make sure value fall in (0x3c,0xfc) */ + l = clamp(l, 0x3c, 0xfc); + r = clamp(r, 0x3c, 0xfc); + + /* invert it and map to userspace value */ + l = 0xfc - l; + r = 0xfc - r; + + ucontrol->value.integer.value[0] = l; + ucontrol->value.integer.value[1] = r; + + return 0; +} + +/* + * custom function to put of PCM playback volume + * + * dac volume register + * 15-------------8-7--------------0 + * | R channel vol | L channel vol | + * ------------------------------- + * + * PCM volume with 0.5017 dB steps from 0 to -90 dB + * + * register values map to dB + * 0x3B and less = Reserved + * 0x3C = 0 dB + * 0x3D = -0.5 dB + * 0xF0 = -90 dB + * 0xFC and greater = Muted + * + * userspace value map to register value + * + * userspace value 0xc0 0 + * ------------------------------ + * register value 0x3c(0dB) 0xf0(-90dB)0xfc + */ +static int dac_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg; + int l; + int r; + + l = ucontrol->value.integer.value[0]; + r = ucontrol->value.integer.value[1]; + + /* make sure userspace volume fall in (0, 0xfc-0x3c) */ + l = clamp(l, 0, 0xfc - 0x3c); + r = clamp(r, 0, 0xfc - 0x3c); + + /* invert it, get the value can be set to register */ + l = 0xfc - l; + r = 0xfc - r; + + /* shift to get the register value */ + reg = l << SGTL5000_DAC_VOL_LEFT_SHIFT | + r << SGTL5000_DAC_VOL_RIGHT_SHIFT; + + snd_soc_write(codec, SGTL5000_CHIP_DAC_VOL, reg); + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0); + +/* tlv for mic gain, 0db 20db 30db 40db */ +static const unsigned int mic_gain_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0), +}; + +/* tlv for hp volume, -51.5db to 12.0db, step .5db */ +static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0); + +static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { + /* SOC_DOUBLE_S8_TLV with invert */ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = dac_info_volsw, + .get = dac_get_volsw, + .put = dac_put_volsw, + }, + + SOC_DOUBLE("Capture Volume", SGTL5000_CHIP_ANA_ADC_CTRL, 0, 4, 0xf, 0), + SOC_SINGLE_TLV("Capture Attenuate Switch (-6dB)", + SGTL5000_CHIP_ANA_ADC_CTRL, + 8, 2, 0, capture_6db_attenuate), + SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0), + + SOC_DOUBLE_TLV("Headphone Playback Volume", + SGTL5000_CHIP_ANA_HP_CTRL, + 0, 8, + 0x7f, 1, + headphone_volume), + SOC_SINGLE("Headphone Playback ZC Switch", SGTL5000_CHIP_ANA_CTRL, + 5, 1, 0), + + SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, + 0, 4, 0, mic_gain_tlv), +}; + +/* mute the codec used by alsa core */ +static int sgtl5000_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 adcdac_ctrl = SGTL5000_DAC_MUTE_LEFT | SGTL5000_DAC_MUTE_RIGHT; + + snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL, + adcdac_ctrl, mute ? adcdac_ctrl : 0); + + return 0; +} + +/* set codec format */ +static int sgtl5000_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + u16 i2sctl = 0; + + sgtl5000->master = 0; + /* + * i2s clock and frame master setting. + * ONLY support: + * - clock and frame slave, + * - clock and frame master + */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBM_CFM: + i2sctl |= SGTL5000_I2S_MASTER; + sgtl5000->master = 1; + break; + default: + return -EINVAL; + } + + /* setting i2s data format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + i2sctl |= SGTL5000_I2S_MODE_PCM; + break; + case SND_SOC_DAIFMT_DSP_B: + i2sctl |= SGTL5000_I2S_MODE_PCM; + i2sctl |= SGTL5000_I2S_LRALIGN; + break; + case SND_SOC_DAIFMT_I2S: + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + break; + case SND_SOC_DAIFMT_RIGHT_J: + i2sctl |= SGTL5000_I2S_MODE_RJ; + i2sctl |= SGTL5000_I2S_LRPOL; + break; + case SND_SOC_DAIFMT_LEFT_J: + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + i2sctl |= SGTL5000_I2S_LRALIGN; + break; + default: + return -EINVAL; + } + + sgtl5000->fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + /* Clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + i2sctl |= SGTL5000_I2S_SCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, SGTL5000_CHIP_I2S_CTRL, i2sctl); + + return 0; +} + +/* set codec sysclk */ +static int sgtl5000_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case SGTL5000_SYSCLK: + sgtl5000->sysclk = freq; + break; + default: + return -EINVAL; + } + + return 0; +} + +/* + * set clock according to i2s frame clock, + * sgtl5000 provide 2 clock sources. + * 1. sys_mclk. sample freq can only configure to + * 1/256, 1/384, 1/512 of sys_mclk. + * 2. pll. can derive any audio clocks. + * + * clock setting rules: + * 1. in slave mode, only sys_mclk can use. + * 2. as constraint by sys_mclk, sample freq should + * set to 32k, 44.1k and above. + * 3. using sys_mclk prefer to pll to save power. + */ +static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + int clk_ctl = 0; + int sys_fs; /* sample freq */ + + /* + * sample freq should be divided by frame clock, + * if frame clock lower than 44.1khz, sample feq should set to + * 32khz or 44.1khz. + */ + switch (frame_rate) { + case 8000: + case 16000: + sys_fs = 32000; + break; + case 11025: + case 22050: + sys_fs = 44100; + break; + default: + sys_fs = frame_rate; + break; + } + + /* set divided factor of frame clock */ + switch (sys_fs / frame_rate) { + case 4: + clk_ctl |= SGTL5000_RATE_MODE_DIV_4 << SGTL5000_RATE_MODE_SHIFT; + break; + case 2: + clk_ctl |= SGTL5000_RATE_MODE_DIV_2 << SGTL5000_RATE_MODE_SHIFT; + break; + case 1: + clk_ctl |= SGTL5000_RATE_MODE_DIV_1 << SGTL5000_RATE_MODE_SHIFT; + break; + default: + return -EINVAL; + } + + /* set the sys_fs according to frame rate */ + switch (sys_fs) { + case 32000: + clk_ctl |= SGTL5000_SYS_FS_32k << SGTL5000_SYS_FS_SHIFT; + break; + case 44100: + clk_ctl |= SGTL5000_SYS_FS_44_1k << SGTL5000_SYS_FS_SHIFT; + break; + case 48000: + clk_ctl |= SGTL5000_SYS_FS_48k << SGTL5000_SYS_FS_SHIFT; + break; + case 96000: + clk_ctl |= SGTL5000_SYS_FS_96k << SGTL5000_SYS_FS_SHIFT; + break; + default: + dev_err(codec->dev, "frame rate %d not supported\n", + frame_rate); + return -EINVAL; + } + + /* + * calculate the divider of mclk/sample_freq, + * factor of freq =96k can only be 256, since mclk in range (12m,27m) + */ + switch (sgtl5000->sysclk / sys_fs) { + case 256: + clk_ctl |= SGTL5000_MCLK_FREQ_256FS << + SGTL5000_MCLK_FREQ_SHIFT; + break; + case 384: + clk_ctl |= SGTL5000_MCLK_FREQ_384FS << + SGTL5000_MCLK_FREQ_SHIFT; + break; + case 512: + clk_ctl |= SGTL5000_MCLK_FREQ_512FS << + SGTL5000_MCLK_FREQ_SHIFT; + break; + default: + /* if mclk not satisify the divider, use pll */ + if (sgtl5000->master) { + clk_ctl |= SGTL5000_MCLK_FREQ_PLL << + SGTL5000_MCLK_FREQ_SHIFT; + } else { + dev_err(codec->dev, + "PLL not supported in slave mode\n"); + return -EINVAL; + } + } + + /* if using pll, please check manual 6.4.2 for detail */ + if ((clk_ctl & SGTL5000_MCLK_FREQ_MASK) == SGTL5000_MCLK_FREQ_PLL) { + u64 out, t; + int div2; + int pll_ctl; + unsigned int in, int_div, frac_div; + + if (sgtl5000->sysclk > 17000000) { + div2 = 1; + in = sgtl5000->sysclk / 2; + } else { + div2 = 0; + in = sgtl5000->sysclk; + } + if (sys_fs == 44100) + out = 180633600; + else + out = 196608000; + t = do_div(out, in); + int_div = out; + t *= 2048; + do_div(t, in); + frac_div = t; + pll_ctl = int_div << SGTL5000_PLL_INT_DIV_SHIFT | + frac_div << SGTL5000_PLL_FRAC_DIV_SHIFT; + + snd_soc_write(codec, SGTL5000_CHIP_PLL_CTRL, pll_ctl); + if (div2) + snd_soc_update_bits(codec, + SGTL5000_CHIP_CLK_TOP_CTRL, + SGTL5000_INPUT_FREQ_DIV2, + SGTL5000_INPUT_FREQ_DIV2); + else + snd_soc_update_bits(codec, + SGTL5000_CHIP_CLK_TOP_CTRL, + SGTL5000_INPUT_FREQ_DIV2, + 0); + + /* power up pll */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, + SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP); + } else { + /* power down pll */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, + 0); + } + + /* if using pll, clk_ctrl must be set after pll power up */ + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); + + return 0; +} + +/* + * Set PCM DAI bit size and sample rate. + * input: params_rate, params_fmt + */ +static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + int channels = params_channels(params); + int i2s_ctl = 0; + int stereo; + int ret; + + /* sysclk should already set */ + if (!sgtl5000->sysclk) { + dev_err(codec->dev, "%s: set sysclk first!\n", __func__); + return -EFAULT; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + stereo = SGTL5000_DAC_STEREO; + else + stereo = SGTL5000_ADC_STEREO; + + /* set mono to save power */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, stereo, + channels == 1 ? 0 : stereo); + + /* set codec clock base on lrclk */ + ret = sgtl5000_set_clock(codec, params_rate(params)); + if (ret) + return ret; + + /* set i2s data format */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J) + return -EINVAL; + i2s_ctl |= SGTL5000_I2S_DLEN_16 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_32FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + i2s_ctl |= SGTL5000_I2S_DLEN_20 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S24_LE: + i2s_ctl |= SGTL5000_I2S_DLEN_24 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S32_LE: + if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J) + return -EINVAL; + i2s_ctl |= SGTL5000_I2S_DLEN_32 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, SGTL5000_CHIP_I2S_CTRL, i2s_ctl, i2s_ctl); + + return 0; +} + +static int ldo_regulator_is_enabled(struct regulator_dev *dev) +{ + struct ldo_regulator *ldo = rdev_get_drvdata(dev); + + return ldo->enabled; +} + +static int ldo_regulator_enable(struct regulator_dev *dev) +{ + struct ldo_regulator *ldo = rdev_get_drvdata(dev); + struct snd_soc_codec *codec = (struct snd_soc_codec *)ldo->codec_data; + int reg; + + if (ldo_regulator_is_enabled(dev)) + return 0; + + /* set regulator value firstly */ + reg = (1600 - ldo->voltage / 1000) / 50; + reg = clamp(reg, 0x0, 0xf); + + /* amend the voltage value, unit: uV */ + ldo->voltage = (1600 - reg * 50) * 1000; + + /* set voltage to register */ + snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, + (0x1 << 4) - 1, reg); + + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINEREG_D_POWERUP, + SGTL5000_LINEREG_D_POWERUP); + + /* when internal ldo enabled, simple digital power can be disabled */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINREG_SIMPLE_POWERUP, + 0); + + ldo->enabled = 1; + return 0; +} + +static int ldo_regulator_disable(struct regulator_dev *dev) +{ + struct ldo_regulator *ldo = rdev_get_drvdata(dev); + struct snd_soc_codec *codec = (struct snd_soc_codec *)ldo->codec_data; + + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINEREG_D_POWERUP, + 0); + + /* clear voltage info */ + snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, + (0x1 << 4) - 1, 0); + + ldo->enabled = 0; + + return 0; +} + +static int ldo_regulator_get_voltage(struct regulator_dev *dev) +{ + struct ldo_regulator *ldo = rdev_get_drvdata(dev); + + return ldo->voltage; +} + +static struct regulator_ops ldo_regulator_ops = { + .is_enabled = ldo_regulator_is_enabled, + .enable = ldo_regulator_enable, + .disable = ldo_regulator_disable, + .get_voltage = ldo_regulator_get_voltage, +}; + +static int ldo_regulator_register(struct snd_soc_codec *codec, + struct regulator_init_data *init_data, + int voltage) +{ + struct ldo_regulator *ldo; + + ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL); + + if (!ldo) { + dev_err(codec->dev, "failed to allocate ldo_regulator\n"); + return -ENOMEM; + } + + ldo->desc.name = kstrdup(dev_name(codec->dev), GFP_KERNEL); + if (!ldo->desc.name) { + kfree(ldo); + dev_err(codec->dev, "failed to allocate decs name memory\n"); + return -ENOMEM; + } + + ldo->desc.type = REGULATOR_VOLTAGE; + ldo->desc.owner = THIS_MODULE; + ldo->desc.ops = &ldo_regulator_ops; + ldo->desc.n_voltages = 1; + + ldo->codec_data = codec; + ldo->voltage = voltage; + + ldo->dev = regulator_register(&ldo->desc, codec->dev, + init_data, ldo); + if (IS_ERR(ldo->dev)) { + dev_err(codec->dev, "failed to register regulator\n"); + kfree(ldo->desc.name); + kfree(ldo); + + return PTR_ERR(ldo->dev); + } + + return 0; +} + +static int ldo_regulator_remove(struct snd_soc_codec *codec) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + struct ldo_regulator *ldo = sgtl5000->ldo; + + if (!ldo) + return 0; + + regulator_unregister(ldo->dev); + kfree(ldo->desc.name); + kfree(ldo); + + return 0; +} + +/* + * set dac bias + * common state changes: + * startup: + * off --> standby --> prepare --> on + * standby --> prepare --> on + * + * stop: + * on --> prepare --> standby + */ +static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable( + ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) + return ret; + udelay(10); + } + + break; + case SND_SOC_BIAS_OFF: + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +#define SGTL5000_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai_ops sgtl5000_ops = { + .hw_params = sgtl5000_pcm_hw_params, + .digital_mute = sgtl5000_digital_mute, + .set_fmt = sgtl5000_set_dai_fmt, + .set_sysclk = sgtl5000_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver sgtl5000_dai = { + .name = "sgtl5000", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + /* + * only support 8~48K + 96K, + * TODO modify hw_param to support more + */ + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_96000, + .formats = SGTL5000_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_96000, + .formats = SGTL5000_FORMATS, + }, + .ops = &sgtl5000_ops, + .symmetric_rates = 1, +}; + +static int sgtl5000_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case SGTL5000_CHIP_ID: + case SGTL5000_CHIP_ADCDAC_CTRL: + case SGTL5000_CHIP_ANA_STATUS: + return 1; + } + + return 0; +} + +#ifdef CONFIG_SUSPEND +static int sgtl5000_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +/* + * restore all sgtl5000 registers, + * since a big hole between dap and regular registers, + * we will restore them respectively. + */ +static int sgtl5000_restore_regs(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i; + int regular_regs = SGTL5000_CHIP_SHORT_CTRL >> 1; + + /* restore regular registers */ + for (i = 0; i < regular_regs; i++) { + int reg = i << 1; + + /* this regs depends on the others */ + if (reg == SGTL5000_CHIP_ANA_POWER || + reg == SGTL5000_CHIP_CLK_CTRL || + reg == SGTL5000_CHIP_LINREG_CTRL || + reg == SGTL5000_CHIP_LINE_OUT_CTRL || + reg == SGTL5000_CHIP_CLK_CTRL) + continue; + + snd_soc_write(codec, reg, cache[i]); + } + + /* restore dap registers */ + for (i = SGTL5000_DAP_REG_OFFSET >> 1; + i < SGTL5000_MAX_REG_OFFSET >> 1; i++) { + int reg = i << 1; + + snd_soc_write(codec, reg, cache[i]); + } + + /* + * restore power and other regs according + * to set_power() and set_clock() + */ + snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, + cache[SGTL5000_CHIP_LINREG_CTRL >> 1]); + + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, + cache[SGTL5000_CHIP_ANA_POWER >> 1]); + + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, + cache[SGTL5000_CHIP_CLK_CTRL >> 1]); + + snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, + cache[SGTL5000_CHIP_REF_CTRL >> 1]); + + snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, + cache[SGTL5000_CHIP_LINE_OUT_CTRL >> 1]); + return 0; +} + +static int sgtl5000_resume(struct snd_soc_codec *codec) +{ + /* Bring the codec back up to standby to enable regulators */ + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Restore registers by cached in memory */ + sgtl5000_restore_regs(codec); + return 0; +} +#else +#define sgtl5000_suspend NULL +#define sgtl5000_resume NULL +#endif /* CONFIG_SUSPEND */ + +/* + * sgtl5000 has 3 internal power supplies: + * 1. VAG, normally set to vdda/2 + * 2. chargepump, set to different value + * according to voltage of vdda and vddio + * 3. line out VAG, normally set to vddio/2 + * + * and should be set according to: + * 1. vddd provided by external or not + * 2. vdda and vddio voltage value. > 3.1v or not + * 3. chip revision >=0x11 or not. If >=0x11, not use external vddd. + */ +static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) +{ + int vddd; + int vdda; + int vddio; + u16 ana_pwr; + u16 lreg_ctrl; + int vag; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + vdda = regulator_get_voltage(sgtl5000->supplies[VDDA].consumer); + vddio = regulator_get_voltage(sgtl5000->supplies[VDDIO].consumer); + vddd = regulator_get_voltage(sgtl5000->supplies[VDDD].consumer); + + vdda = vdda / 1000; + vddio = vddio / 1000; + vddd = vddd / 1000; + + if (vdda <= 0 || vddio <= 0 || vddd < 0) { + dev_err(codec->dev, "regulator voltage not set correctly\n"); + + return -EINVAL; + } + + /* according to datasheet, maximum voltage of supplies */ + if (vdda > 3600 || vddio > 3600 || vddd > 1980) { + dev_err(codec->dev, + "exceed max voltage vdda %dmv vddio %dma vddd %dma\n", + vdda, vddio, vddd); + + return -EINVAL; + } + + /* reset value */ + ana_pwr = snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER); + ana_pwr |= SGTL5000_DAC_STEREO | + SGTL5000_ADC_STEREO | + SGTL5000_REFTOP_POWERUP; + lreg_ctrl = snd_soc_read(codec, SGTL5000_CHIP_LINREG_CTRL); + + if (vddio < 3100 && vdda < 3100) { + /* enable internal oscillator used for charge pump */ + snd_soc_update_bits(codec, SGTL5000_CHIP_CLK_TOP_CTRL, + SGTL5000_INT_OSC_EN, + SGTL5000_INT_OSC_EN); + /* Enable VDDC charge pump */ + ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP; + } else if (vddio >= 3100 && vdda >= 3100) { + /* + * if vddio and vddd > 3.1v, + * charge pump should be clean before set ana_pwr + */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VDDC_CHRGPMP_POWERUP, 0); + + /* VDDC use VDDIO rail */ + lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD; + lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO << + SGTL5000_VDDC_MAN_ASSN_SHIFT; + } + + snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, lreg_ctrl); + + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, ana_pwr); + + /* set voltage to register */ + snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, + (0x1 << 4) - 1, 0x8); + + /* + * if vddd linear reg has been enabled, + * simple digital supply should be clear to get + * proper VDDD voltage. + */ + if (ana_pwr & SGTL5000_LINEREG_D_POWERUP) + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINREG_SIMPLE_POWERUP, + 0); + else + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINREG_SIMPLE_POWERUP | + SGTL5000_STARTUP_POWERUP, + 0); + + /* + * set ADC/DAC VAG to vdda / 2, + * should stay in range (0.8v, 1.575v) + */ + vag = vdda / 2; + if (vag <= SGTL5000_ANA_GND_BASE) + vag = 0; + else if (vag >= SGTL5000_ANA_GND_BASE + SGTL5000_ANA_GND_STP * + (SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT)) + vag = SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT; + else + vag = (vag - SGTL5000_ANA_GND_BASE) / SGTL5000_ANA_GND_STP; + + snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, + vag << SGTL5000_ANA_GND_SHIFT, + vag << SGTL5000_ANA_GND_SHIFT); + + /* set line out VAG to vddio / 2, in range (0.8v, 1.675v) */ + vag = vddio / 2; + if (vag <= SGTL5000_LINE_OUT_GND_BASE) + vag = 0; + else if (vag >= SGTL5000_LINE_OUT_GND_BASE + + SGTL5000_LINE_OUT_GND_STP * SGTL5000_LINE_OUT_GND_MAX) + vag = SGTL5000_LINE_OUT_GND_MAX; + else + vag = (vag - SGTL5000_LINE_OUT_GND_BASE) / + SGTL5000_LINE_OUT_GND_STP; + + snd_soc_update_bits(codec, SGTL5000_CHIP_LINE_OUT_CTRL, + vag << SGTL5000_LINE_OUT_GND_SHIFT | + SGTL5000_LINE_OUT_CURRENT_360u << + SGTL5000_LINE_OUT_CURRENT_SHIFT, + vag << SGTL5000_LINE_OUT_GND_SHIFT | + SGTL5000_LINE_OUT_CURRENT_360u << + SGTL5000_LINE_OUT_CURRENT_SHIFT); + + return 0; +} + +static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) +{ + u16 reg; + int ret; + int rev; + int i; + int external_vddd = 0; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++) + sgtl5000->supplies[i].supply = supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (!ret) + external_vddd = 1; + else { + /* set internal ldo to 1.2v */ + int voltage = LDO_VOLTAGE; + + ret = ldo_regulator_register(codec, &ldo_init_data, voltage); + if (ret) { + dev_err(codec->dev, + "Failed to register vddd internal supplies: %d\n", + ret); + return ret; + } + + sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME; + + ret = regulator_bulk_get(codec->dev, + ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + + if (ret) { + ldo_regulator_remove(codec); + dev_err(codec->dev, + "Failed to request supplies: %d\n", ret); + + return ret; + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) + goto err_regulator_free; + + /* wait for all power rails bring up */ + udelay(10); + + /* read chip information */ + reg = snd_soc_read(codec, SGTL5000_CHIP_ID); + if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != + SGTL5000_PARTID_PART_ID) { + dev_err(codec->dev, + "Device with ID register %x is not a sgtl5000\n", reg); + ret = -ENODEV; + goto err_regulator_disable; + } + + rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; + dev_info(codec->dev, "sgtl5000 revision %d\n", rev); + + /* + * workaround for revision 0x11 and later, + * roll back to use internal LDO + */ + if (external_vddd && rev >= 0x11) { + int voltage = LDO_VOLTAGE; + /* disable all regulator first */ + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + /* free VDDD regulator */ + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + + ret = ldo_regulator_register(codec, &ldo_init_data, voltage); + if (ret) + return ret; + + sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME; + + ret = regulator_bulk_get(codec->dev, + ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) { + ldo_regulator_remove(codec); + dev_err(codec->dev, + "Failed to request supplies: %d\n", ret); + + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) + goto err_regulator_free; + + /* wait for all power rails bring up */ + udelay(10); + } + + return 0; + +err_regulator_disable: + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); +err_regulator_free: + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (external_vddd) + ldo_regulator_remove(codec); + return ret; + +} + +static int sgtl5000_probe(struct snd_soc_codec *codec) +{ + int ret; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + /* setup i2c data ops */ + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + ret = sgtl5000_enable_regulators(codec); + if (ret) + return ret; + + /* power up sgtl5000 */ + ret = sgtl5000_set_power_regs(codec); + if (ret) + goto err; + + /* enable small pop, introduce 400ms delay in turning off */ + snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, + SGTL5000_SMALL_POP, + SGTL5000_SMALL_POP); + + /* disable short cut detector */ + snd_soc_write(codec, SGTL5000_CHIP_SHORT_CTRL, 0); + + /* + * set i2s as default input of sound switch + * TODO: add sound switch to control and dapm widge. + */ + snd_soc_write(codec, SGTL5000_CHIP_SSS_CTRL, + SGTL5000_DAC_SEL_I2S_IN << SGTL5000_DAC_SEL_SHIFT); + snd_soc_write(codec, SGTL5000_CHIP_DIG_POWER, + SGTL5000_ADC_EN | SGTL5000_DAC_EN); + + /* enable dac volume ramp by default */ + snd_soc_write(codec, SGTL5000_CHIP_ADCDAC_CTRL, + SGTL5000_DAC_VOL_RAMP_EN | + SGTL5000_DAC_MUTE_RIGHT | + SGTL5000_DAC_MUTE_LEFT); + + snd_soc_write(codec, SGTL5000_CHIP_PAD_STRENGTH, 0x015f); + + snd_soc_write(codec, SGTL5000_CHIP_ANA_CTRL, + SGTL5000_HP_ZCD_EN | + SGTL5000_ADC_ZCD_EN); + + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + + /* + * disable DAP + * TODO: + * Enable DAP in kcontrol and dapm. + */ + snd_soc_write(codec, SGTL5000_DAP_CTRL, 0); + + /* leading to standby state */ + ret = sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (ret) + goto err; + + snd_soc_add_controls(codec, sgtl5000_snd_controls, + ARRAY_SIZE(sgtl5000_snd_controls)); + + snd_soc_dapm_new_controls(&codec->dapm, sgtl5000_dapm_widgets, + ARRAY_SIZE(sgtl5000_dapm_widgets)); + + snd_soc_dapm_add_routes(&codec->dapm, audio_map, + ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(&codec->dapm); + + return 0; + +err: + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + ldo_regulator_remove(codec); + + return ret; +} + +static int sgtl5000_remove(struct snd_soc_codec *codec) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); + + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + ldo_regulator_remove(codec); + + return 0; +} + +struct snd_soc_codec_driver sgtl5000_driver = { + .probe = sgtl5000_probe, + .remove = sgtl5000_remove, + .suspend = sgtl5000_suspend, + .resume = sgtl5000_resume, + .set_bias_level = sgtl5000_set_bias_level, + .reg_cache_size = ARRAY_SIZE(sgtl5000_regs), + .reg_word_size = sizeof(u16), + .reg_cache_step = 2, + .reg_cache_default = sgtl5000_regs, + .volatile_register = sgtl5000_volatile_register, +}; + +static __devinit int sgtl5000_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct sgtl5000_priv *sgtl5000; + int ret; + + sgtl5000 = kzalloc(sizeof(struct sgtl5000_priv), GFP_KERNEL); + if (!sgtl5000) + return -ENOMEM; + + /* + * copy DAP default values to default value array. + * sgtl5000 register space has a big hole, merge it + * at init phase makes life easy. + * FIXME: should we drop 'const' of sgtl5000_regs? + */ + memcpy((void *)(&sgtl5000_regs[0] + (SGTL5000_DAP_REG_OFFSET >> 1)), + sgtl5000_dap_regs, + SGTL5000_MAX_REG_OFFSET - SGTL5000_DAP_REG_OFFSET); + + i2c_set_clientdata(client, sgtl5000); + + ret = snd_soc_register_codec(&client->dev, + &sgtl5000_driver, &sgtl5000_dai, 1); + if (ret) { + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + kfree(sgtl5000); + return ret; + } + + return 0; +} + +static __devexit int sgtl5000_i2c_remove(struct i2c_client *client) +{ + struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + kfree(sgtl5000); + return 0; +} + +static const struct i2c_device_id sgtl5000_id[] = { + {"sgtl5000", 0}, + {}, +}; + +MODULE_DEVICE_TABLE(i2c, sgtl5000_id); + +static struct i2c_driver sgtl5000_i2c_driver = { + .driver = { + .name = "sgtl5000", + .owner = THIS_MODULE, + }, + .probe = sgtl5000_i2c_probe, + .remove = __devexit_p(sgtl5000_i2c_remove), + .id_table = sgtl5000_id, +}; + +static int __init sgtl5000_modinit(void) +{ + return i2c_add_driver(&sgtl5000_i2c_driver); +} +module_init(sgtl5000_modinit); + +static void __exit sgtl5000_exit(void) +{ + i2c_del_driver(&sgtl5000_i2c_driver); +} +module_exit(sgtl5000_exit); + +MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Zeng Zhaoming "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h new file mode 100644 index 00000000000..c29312ab801 --- /dev/null +++ b/sound/soc/codecs/sgtl5000.h @@ -0,0 +1,402 @@ +/* + * sgtl5000.h - SGTL5000 audio codec interface + * + * Copyright 2010-2011 Freescale Semiconductor, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _SGTL5000_H +#define _SGTL5000_H + +#include + +/* + * Register values. + */ +#define SGTL5000_CHIP_ID 0x0000 +#define SGTL5000_CHIP_DIG_POWER 0x0002 +#define SGTL5000_CHIP_CLK_CTRL 0x0004 +#define SGTL5000_CHIP_I2S_CTRL 0x0006 +#define SGTL5000_CHIP_SSS_CTRL 0x000a +#define SGTL5000_CHIP_ADCDAC_CTRL 0x000e +#define SGTL5000_CHIP_DAC_VOL 0x0010 +#define SGTL5000_CHIP_PAD_STRENGTH 0x0014 +#define SGTL5000_CHIP_ANA_ADC_CTRL 0x0020 +#define SGTL5000_CHIP_ANA_HP_CTRL 0x0022 +#define SGTL5000_CHIP_ANA_CTRL 0x0024 +#define SGTL5000_CHIP_LINREG_CTRL 0x0026 +#define SGTL5000_CHIP_REF_CTRL 0x0028 +#define SGTL5000_CHIP_MIC_CTRL 0x002a +#define SGTL5000_CHIP_LINE_OUT_CTRL 0x002c +#define SGTL5000_CHIP_LINE_OUT_VOL 0x002e +#define SGTL5000_CHIP_ANA_POWER 0x0030 +#define SGTL5000_CHIP_PLL_CTRL 0x0032 +#define SGTL5000_CHIP_CLK_TOP_CTRL 0x0034 +#define SGTL5000_CHIP_ANA_STATUS 0x0036 +#define SGTL5000_CHIP_SHORT_CTRL 0x003c +#define SGTL5000_CHIP_ANA_TEST2 0x003a +#define SGTL5000_DAP_CTRL 0x0100 +#define SGTL5000_DAP_PEQ 0x0102 +#define SGTL5000_DAP_BASS_ENHANCE 0x0104 +#define SGTL5000_DAP_BASS_ENHANCE_CTRL 0x0106 +#define SGTL5000_DAP_AUDIO_EQ 0x0108 +#define SGTL5000_DAP_SURROUND 0x010a +#define SGTL5000_DAP_FLT_COEF_ACCESS 0x010c +#define SGTL5000_DAP_COEF_WR_B0_MSB 0x010e +#define SGTL5000_DAP_COEF_WR_B0_LSB 0x0110 +#define SGTL5000_DAP_EQ_BASS_BAND0 0x0116 +#define SGTL5000_DAP_EQ_BASS_BAND1 0x0118 +#define SGTL5000_DAP_EQ_BASS_BAND2 0x011a +#define SGTL5000_DAP_EQ_BASS_BAND3 0x011c +#define SGTL5000_DAP_EQ_BASS_BAND4 0x011e +#define SGTL5000_DAP_MAIN_CHAN 0x0120 +#define SGTL5000_DAP_MIX_CHAN 0x0122 +#define SGTL5000_DAP_AVC_CTRL 0x0124 +#define SGTL5000_DAP_AVC_THRESHOLD 0x0126 +#define SGTL5000_DAP_AVC_ATTACK 0x0128 +#define SGTL5000_DAP_AVC_DECAY 0x012a +#define SGTL5000_DAP_COEF_WR_B1_MSB 0x012c +#define SGTL5000_DAP_COEF_WR_B1_LSB 0x012e +#define SGTL5000_DAP_COEF_WR_B2_MSB 0x0130 +#define SGTL5000_DAP_COEF_WR_B2_LSB 0x0132 +#define SGTL5000_DAP_COEF_WR_A1_MSB 0x0134 +#define SGTL5000_DAP_COEF_WR_A1_LSB 0x0136 +#define SGTL5000_DAP_COEF_WR_A2_MSB 0x0138 +#define SGTL5000_DAP_COEF_WR_A2_LSB 0x013a + +/* + * Field Definitions. + */ + +/* + * SGTL5000_CHIP_ID + */ +#define SGTL5000_PARTID_MASK 0xff00 +#define SGTL5000_PARTID_SHIFT 8 +#define SGTL5000_PARTID_WIDTH 8 +#define SGTL5000_PARTID_PART_ID 0xa0 +#define SGTL5000_REVID_MASK 0x00ff +#define SGTL5000_REVID_SHIFT 0 +#define SGTL5000_REVID_WIDTH 8 + +/* + * SGTL5000_CHIP_DIG_POWER + */ +#define SGTL5000_ADC_EN 0x0040 +#define SGTL5000_DAC_EN 0x0020 +#define SGTL5000_DAP_POWERUP 0x0010 +#define SGTL5000_I2S_OUT_POWERUP 0x0002 +#define SGTL5000_I2S_IN_POWERUP 0x0001 + +/* + * SGTL5000_CHIP_CLK_CTRL + */ +#define SGTL5000_RATE_MODE_MASK 0x0030 +#define SGTL5000_RATE_MODE_SHIFT 4 +#define SGTL5000_RATE_MODE_WIDTH 2 +#define SGTL5000_RATE_MODE_DIV_1 0 +#define SGTL5000_RATE_MODE_DIV_2 1 +#define SGTL5000_RATE_MODE_DIV_4 2 +#define SGTL5000_RATE_MODE_DIV_6 3 +#define SGTL5000_SYS_FS_MASK 0x000c +#define SGTL5000_SYS_FS_SHIFT 2 +#define SGTL5000_SYS_FS_WIDTH 2 +#define SGTL5000_SYS_FS_32k 0x0 +#define SGTL5000_SYS_FS_44_1k 0x1 +#define SGTL5000_SYS_FS_48k 0x2 +#define SGTL5000_SYS_FS_96k 0x3 +#define SGTL5000_MCLK_FREQ_MASK 0x0003 +#define SGTL5000_MCLK_FREQ_SHIFT 0 +#define SGTL5000_MCLK_FREQ_WIDTH 2 +#define SGTL5000_MCLK_FREQ_256FS 0x0 +#define SGTL5000_MCLK_FREQ_384FS 0x1 +#define SGTL5000_MCLK_FREQ_512FS 0x2 +#define SGTL5000_MCLK_FREQ_PLL 0x3 + +/* + * SGTL5000_CHIP_I2S_CTRL + */ +#define SGTL5000_I2S_SCLKFREQ_MASK 0x0100 +#define SGTL5000_I2S_SCLKFREQ_SHIFT 8 +#define SGTL5000_I2S_SCLKFREQ_WIDTH 1 +#define SGTL5000_I2S_SCLKFREQ_64FS 0x0 +#define SGTL5000_I2S_SCLKFREQ_32FS 0x1 /* Not for RJ mode */ +#define SGTL5000_I2S_MASTER 0x0080 +#define SGTL5000_I2S_SCLK_INV 0x0040 +#define SGTL5000_I2S_DLEN_MASK 0x0030 +#define SGTL5000_I2S_DLEN_SHIFT 4 +#define SGTL5000_I2S_DLEN_WIDTH 2 +#define SGTL5000_I2S_DLEN_32 0x0 +#define SGTL5000_I2S_DLEN_24 0x1 +#define SGTL5000_I2S_DLEN_20 0x2 +#define SGTL5000_I2S_DLEN_16 0x3 +#define SGTL5000_I2S_MODE_MASK 0x000c +#define SGTL5000_I2S_MODE_SHIFT 2 +#define SGTL5000_I2S_MODE_WIDTH 2 +#define SGTL5000_I2S_MODE_I2S_LJ 0x0 +#define SGTL5000_I2S_MODE_RJ 0x1 +#define SGTL5000_I2S_MODE_PCM 0x2 +#define SGTL5000_I2S_LRALIGN 0x0002 +#define SGTL5000_I2S_LRPOL 0x0001 /* set for which mode */ + +/* + * SGTL5000_CHIP_SSS_CTRL + */ +#define SGTL5000_DAP_MIX_LRSWAP 0x4000 +#define SGTL5000_DAP_LRSWAP 0x2000 +#define SGTL5000_DAC_LRSWAP 0x1000 +#define SGTL5000_I2S_OUT_LRSWAP 0x0400 +#define SGTL5000_DAP_MIX_SEL_MASK 0x0300 +#define SGTL5000_DAP_MIX_SEL_SHIFT 8 +#define SGTL5000_DAP_MIX_SEL_WIDTH 2 +#define SGTL5000_DAP_MIX_SEL_ADC 0x0 +#define SGTL5000_DAP_MIX_SEL_I2S_IN 0x1 +#define SGTL5000_DAP_SEL_MASK 0x00c0 +#define SGTL5000_DAP_SEL_SHIFT 6 +#define SGTL5000_DAP_SEL_WIDTH 2 +#define SGTL5000_DAP_SEL_ADC 0x0 +#define SGTL5000_DAP_SEL_I2S_IN 0x1 +#define SGTL5000_DAC_SEL_MASK 0x0030 +#define SGTL5000_DAC_SEL_SHIFT 4 +#define SGTL5000_DAC_SEL_WIDTH 2 +#define SGTL5000_DAC_SEL_ADC 0x0 +#define SGTL5000_DAC_SEL_I2S_IN 0x1 +#define SGTL5000_DAC_SEL_DAP 0x3 +#define SGTL5000_I2S_OUT_SEL_MASK 0x0003 +#define SGTL5000_I2S_OUT_SEL_SHIFT 0 +#define SGTL5000_I2S_OUT_SEL_WIDTH 2 +#define SGTL5000_I2S_OUT_SEL_ADC 0x0 +#define SGTL5000_I2S_OUT_SEL_I2S_IN 0x1 +#define SGTL5000_I2S_OUT_SEL_DAP 0x3 + +/* + * SGTL5000_CHIP_ADCDAC_CTRL + */ +#define SGTL5000_VOL_BUSY_DAC_RIGHT 0x2000 +#define SGTL5000_VOL_BUSY_DAC_LEFT 0x1000 +#define SGTL5000_DAC_VOL_RAMP_EN 0x0200 +#define SGTL5000_DAC_VOL_RAMP_EXPO 0x0100 +#define SGTL5000_DAC_MUTE_RIGHT 0x0008 +#define SGTL5000_DAC_MUTE_LEFT 0x0004 +#define SGTL5000_ADC_HPF_FREEZE 0x0002 +#define SGTL5000_ADC_HPF_BYPASS 0x0001 + +/* + * SGTL5000_CHIP_DAC_VOL + */ +#define SGTL5000_DAC_VOL_RIGHT_MASK 0xff00 +#define SGTL5000_DAC_VOL_RIGHT_SHIFT 8 +#define SGTL5000_DAC_VOL_RIGHT_WIDTH 8 +#define SGTL5000_DAC_VOL_LEFT_MASK 0x00ff +#define SGTL5000_DAC_VOL_LEFT_SHIFT 0 +#define SGTL5000_DAC_VOL_LEFT_WIDTH 8 + +/* + * SGTL5000_CHIP_PAD_STRENGTH + */ +#define SGTL5000_PAD_I2S_LRCLK_MASK 0x0300 +#define SGTL5000_PAD_I2S_LRCLK_SHIFT 8 +#define SGTL5000_PAD_I2S_LRCLK_WIDTH 2 +#define SGTL5000_PAD_I2S_SCLK_MASK 0x00c0 +#define SGTL5000_PAD_I2S_SCLK_SHIFT 6 +#define SGTL5000_PAD_I2S_SCLK_WIDTH 2 +#define SGTL5000_PAD_I2S_DOUT_MASK 0x0030 +#define SGTL5000_PAD_I2S_DOUT_SHIFT 4 +#define SGTL5000_PAD_I2S_DOUT_WIDTH 2 +#define SGTL5000_PAD_I2C_SDA_MASK 0x000c +#define SGTL5000_PAD_I2C_SDA_SHIFT 2 +#define SGTL5000_PAD_I2C_SDA_WIDTH 2 +#define SGTL5000_PAD_I2C_SCL_MASK 0x0003 +#define SGTL5000_PAD_I2C_SCL_SHIFT 0 +#define SGTL5000_PAD_I2C_SCL_WIDTH 2 + +/* + * SGTL5000_CHIP_ANA_ADC_CTRL + */ +#define SGTL5000_ADC_VOL_M6DB 0x0100 +#define SGTL5000_ADC_VOL_RIGHT_MASK 0x00f0 +#define SGTL5000_ADC_VOL_RIGHT_SHIFT 4 +#define SGTL5000_ADC_VOL_RIGHT_WIDTH 4 +#define SGTL5000_ADC_VOL_LEFT_MASK 0x000f +#define SGTL5000_ADC_VOL_LEFT_SHIFT 0 +#define SGTL5000_ADC_VOL_LEFT_WIDTH 4 + +/* + * SGTL5000_CHIP_ANA_HP_CTRL + */ +#define SGTL5000_HP_VOL_RIGHT_MASK 0x7f00 +#define SGTL5000_HP_VOL_RIGHT_SHIFT 8 +#define SGTL5000_HP_VOL_RIGHT_WIDTH 7 +#define SGTL5000_HP_VOL_LEFT_MASK 0x007f +#define SGTL5000_HP_VOL_LEFT_SHIFT 0 +#define SGTL5000_HP_VOL_LEFT_WIDTH 7 + +/* + * SGTL5000_CHIP_ANA_CTRL + */ +#define SGTL5000_LINE_OUT_MUTE 0x0100 +#define SGTL5000_HP_SEL_MASK 0x0040 +#define SGTL5000_HP_SEL_SHIFT 6 +#define SGTL5000_HP_SEL_WIDTH 1 +#define SGTL5000_HP_SEL_DAC 0x0 +#define SGTL5000_HP_SEL_LINE_IN 0x1 +#define SGTL5000_HP_ZCD_EN 0x0020 +#define SGTL5000_HP_MUTE 0x0010 +#define SGTL5000_ADC_SEL_MASK 0x0004 +#define SGTL5000_ADC_SEL_SHIFT 2 +#define SGTL5000_ADC_SEL_WIDTH 1 +#define SGTL5000_ADC_SEL_MIC 0x0 +#define SGTL5000_ADC_SEL_LINE_IN 0x1 +#define SGTL5000_ADC_ZCD_EN 0x0002 +#define SGTL5000_ADC_MUTE 0x0001 + +/* + * SGTL5000_CHIP_LINREG_CTRL + */ +#define SGTL5000_VDDC_MAN_ASSN_MASK 0x0040 +#define SGTL5000_VDDC_MAN_ASSN_SHIFT 6 +#define SGTL5000_VDDC_MAN_ASSN_WIDTH 1 +#define SGTL5000_VDDC_MAN_ASSN_VDDA 0x0 +#define SGTL5000_VDDC_MAN_ASSN_VDDIO 0x1 +#define SGTL5000_VDDC_ASSN_OVRD 0x0020 +#define SGTL5000_LINREG_VDDD_MASK 0x000f +#define SGTL5000_LINREG_VDDD_SHIFT 0 +#define SGTL5000_LINREG_VDDD_WIDTH 4 + +/* + * SGTL5000_CHIP_REF_CTRL + */ +#define SGTL5000_ANA_GND_MASK 0x01f0 +#define SGTL5000_ANA_GND_SHIFT 4 +#define SGTL5000_ANA_GND_WIDTH 5 +#define SGTL5000_ANA_GND_BASE 800 /* mv */ +#define SGTL5000_ANA_GND_STP 25 /*mv */ +#define SGTL5000_BIAS_CTRL_MASK 0x000e +#define SGTL5000_BIAS_CTRL_SHIFT 1 +#define SGTL5000_BIAS_CTRL_WIDTH 3 +#define SGTL5000_SMALL_POP 0x0001 + +/* + * SGTL5000_CHIP_MIC_CTRL + */ +#define SGTL5000_BIAS_R_MASK 0x0200 +#define SGTL5000_BIAS_R_SHIFT 8 +#define SGTL5000_BIAS_R_WIDTH 2 +#define SGTL5000_BIAS_R_off 0x0 +#define SGTL5000_BIAS_R_2K 0x1 +#define SGTL5000_BIAS_R_4k 0x2 +#define SGTL5000_BIAS_R_8k 0x3 +#define SGTL5000_BIAS_VOLT_MASK 0x0070 +#define SGTL5000_BIAS_VOLT_SHIFT 4 +#define SGTL5000_BIAS_VOLT_WIDTH 3 +#define SGTL5000_MIC_GAIN_MASK 0x0003 +#define SGTL5000_MIC_GAIN_SHIFT 0 +#define SGTL5000_MIC_GAIN_WIDTH 2 + +/* + * SGTL5000_CHIP_LINE_OUT_CTRL + */ +#define SGTL5000_LINE_OUT_CURRENT_MASK 0x0f00 +#define SGTL5000_LINE_OUT_CURRENT_SHIFT 8 +#define SGTL5000_LINE_OUT_CURRENT_WIDTH 4 +#define SGTL5000_LINE_OUT_CURRENT_180u 0x0 +#define SGTL5000_LINE_OUT_CURRENT_270u 0x1 +#define SGTL5000_LINE_OUT_CURRENT_360u 0x3 +#define SGTL5000_LINE_OUT_CURRENT_450u 0x7 +#define SGTL5000_LINE_OUT_CURRENT_540u 0xf +#define SGTL5000_LINE_OUT_GND_MASK 0x003f +#define SGTL5000_LINE_OUT_GND_SHIFT 0 +#define SGTL5000_LINE_OUT_GND_WIDTH 6 +#define SGTL5000_LINE_OUT_GND_BASE 800 /* mv */ +#define SGTL5000_LINE_OUT_GND_STP 25 +#define SGTL5000_LINE_OUT_GND_MAX 0x23 + +/* + * SGTL5000_CHIP_LINE_OUT_VOL + */ +#define SGTL5000_LINE_OUT_VOL_RIGHT_MASK 0x1f00 +#define SGTL5000_LINE_OUT_VOL_RIGHT_SHIFT 8 +#define SGTL5000_LINE_OUT_VOL_RIGHT_WIDTH 5 +#define SGTL5000_LINE_OUT_VOL_LEFT_MASK 0x001f +#define SGTL5000_LINE_OUT_VOL_LEFT_SHIFT 0 +#define SGTL5000_LINE_OUT_VOL_LEFT_WIDTH 5 + +/* + * SGTL5000_CHIP_ANA_POWER + */ +#define SGTL5000_DAC_STEREO 0x4000 +#define SGTL5000_LINREG_SIMPLE_POWERUP 0x2000 +#define SGTL5000_STARTUP_POWERUP 0x1000 +#define SGTL5000_VDDC_CHRGPMP_POWERUP 0x0800 +#define SGTL5000_PLL_POWERUP 0x0400 +#define SGTL5000_LINEREG_D_POWERUP 0x0200 +#define SGTL5000_VCOAMP_POWERUP 0x0100 +#define SGTL5000_VAG_POWERUP 0x0080 +#define SGTL5000_ADC_STEREO 0x0040 +#define SGTL5000_REFTOP_POWERUP 0x0020 +#define SGTL5000_HP_POWERUP 0x0010 +#define SGTL5000_DAC_POWERUP 0x0008 +#define SGTL5000_CAPLESS_HP_POWERUP 0x0004 +#define SGTL5000_ADC_POWERUP 0x0002 +#define SGTL5000_LINE_OUT_POWERUP 0x0001 + +/* + * SGTL5000_CHIP_PLL_CTRL + */ +#define SGTL5000_PLL_INT_DIV_MASK 0xf800 +#define SGTL5000_PLL_INT_DIV_SHIFT 11 +#define SGTL5000_PLL_INT_DIV_WIDTH 5 +#define SGTL5000_PLL_FRAC_DIV_MASK 0x0700 +#define SGTL5000_PLL_FRAC_DIV_SHIFT 0 +#define SGTL5000_PLL_FRAC_DIV_WIDTH 11 + +/* + * SGTL5000_CHIP_CLK_TOP_CTRL + */ +#define SGTL5000_INT_OSC_EN 0x0800 +#define SGTL5000_INPUT_FREQ_DIV2 0x0008 + +/* + * SGTL5000_CHIP_ANA_STATUS + */ +#define SGTL5000_HP_LRSHORT 0x0200 +#define SGTL5000_CAPLESS_SHORT 0x0100 +#define SGTL5000_PLL_LOCKED 0x0010 + +/* + * SGTL5000_CHIP_SHORT_CTRL + */ +#define SGTL5000_LVLADJR_MASK 0x7000 +#define SGTL5000_LVLADJR_SHIFT 12 +#define SGTL5000_LVLADJR_WIDTH 3 +#define SGTL5000_LVLADJL_MASK 0x0700 +#define SGTL5000_LVLADJL_SHIFT 8 +#define SGTL5000_LVLADJL_WIDTH 3 +#define SGTL5000_LVLADJC_MASK 0x0070 +#define SGTL5000_LVLADJC_SHIFT 4 +#define SGTL5000_LVLADJC_WIDTH 3 +#define SGTL5000_LR_SHORT_MOD_MASK 0x000c +#define SGTL5000_LR_SHORT_MOD_SHIFT 2 +#define SGTL5000_LR_SHORT_MOD_WIDTH 2 +#define SGTL5000_CM_SHORT_MOD_MASK 0x0003 +#define SGTL5000_CM_SHORT_MOD_SHIFT 0 +#define SGTL5000_CM_SHORT_MOD_WIDTH 2 + +/* + *SGTL5000_CHIP_ANA_TEST2 + */ +#define SGTL5000_MONO_DAC 0x1000 + +/* + * SGTL5000_DAP_CTRL + */ +#define SGTL5000_DAP_MIX_EN 0x0010 +#define SGTL5000_DAP_EN 0x0001 + +#define SGTL5000_SYSCLK 0x00 +#define SGTL5000_LRCLK 0x01 + +#endif -- cgit v1.2.3 From b3111a9aa8823e360f20e3ed9fb106d757b89704 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 25 Feb 2011 12:25:18 +0000 Subject: ASoC: Move WM2000 to dev_pm_ops There's a general move to use dev_pm_ops rather than bus specific functions in order to facilitate work on the PM core. Do this conversion to WM2000. The driver ought to be updated to work better in a multi-component model but the mechanical conversion ensures that we avoid blocking PM core work until that happens. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm2000.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 80ddf4fd23d..a3b9cbb20ee 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -836,24 +836,25 @@ static void wm2000_i2c_shutdown(struct i2c_client *i2c) } #ifdef CONFIG_PM -static int wm2000_i2c_suspend(struct i2c_client *i2c, pm_message_t mesg) +static int wm2000_i2c_suspend(struct device *dev) { + struct i2c_client *i2c = to_i2c_client(dev); struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); return wm2000_anc_transition(wm2000, ANC_OFF); } -static int wm2000_i2c_resume(struct i2c_client *i2c) +static int wm2000_i2c_resume(struct device *dev) { + struct i2c_client *i2c = to_i2c_client(dev); struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); return wm2000_anc_set_mode(wm2000); } -#else -#define wm2000_i2c_suspend NULL -#define wm2000_i2c_resume NULL #endif +static SIMPLE_DEV_PM_OPS(wm2000_pm, wm2000_i2c_suspend, wm2000_i2c_resume); + static const struct i2c_device_id wm2000_i2c_id[] = { { "wm2000", 0 }, { } @@ -864,11 +865,10 @@ static struct i2c_driver wm2000_i2c_driver = { .driver = { .name = "wm2000", .owner = THIS_MODULE, + .pm = &wm2000_pm, }, .probe = wm2000_i2c_probe, .remove = __devexit_p(wm2000_i2c_remove), - .suspend = wm2000_i2c_suspend, - .resume = wm2000_i2c_resume, .shutdown = wm2000_i2c_shutdown, .id_table = wm2000_i2c_id, }; -- cgit v1.2.3 From 573f89b5f39a8b7b4f0a21a7567dd1121430a3c3 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Eric=20B=C3=A9nard?= Date: Fri, 25 Feb 2011 13:48:15 +0100 Subject: ASoC: eukrea-tlv320: add MBIMXSD51 support MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Signed-off-by: Eric Bénard Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/Kconfig | 3 ++- sound/soc/imx/eukrea-tlv320.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 642270a635e..9eeb8f0d67e 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -44,7 +44,8 @@ config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" depends on MACH_EUKREA_MBIMX27_BASEBOARD \ || MACH_EUKREA_MBIMXSD25_BASEBOARD \ - || MACH_EUKREA_MBIMXSD35_BASEBOARD + || MACH_EUKREA_MBIMXSD35_BASEBOARD \ + || MACH_EUKREA_MBIMXSD51_BASEBOARD select SND_SOC_TLV320AIC23 select SND_MXC_SOC_SSI select SND_MXC_SOC_FIQ diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c index e20c9e1457c..a08e822e015 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/imx/eukrea-tlv320.c @@ -98,7 +98,8 @@ static int __init eukrea_tlv320_init(void) int ret; if (!machine_is_eukrea_cpuimx27() && !machine_is_eukrea_cpuimx25sd() - && !machine_is_eukrea_cpuimx35sd()) + && !machine_is_eukrea_cpuimx35sd() + && !machine_is_eukrea_cpuimx51sd()) /* return happy. We might run on a totally different machine */ return 0; -- cgit v1.2.3 From 5f83df9a6192b3197a58beb66908b3732cb5a670 Mon Sep 17 00:00:00 2001 From: Zeng Zhaoming Date: Mon, 28 Feb 2011 03:45:21 +0800 Subject: ASoC: remove unnecessary header including in SGTL5000 codec driver Remove unnecessary headers: - mach/hardware.h in sgtl5000.c - linux/i2c.h in sgtl5000.h Signed-off-by: Zeng Zhaoming Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 1 - sound/soc/codecs/sgtl5000.h | 2 -- 2 files changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 9eb206e852d..b528f971f3f 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -27,7 +27,6 @@ #include #include #include -#include #include "sgtl5000.h" diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index c29312ab801..eec3ab368f3 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -11,8 +11,6 @@ #ifndef _SGTL5000_H #define _SGTL5000_H -#include - /* * Register values. */ -- cgit v1.2.3 From 61a142b7e4b5c4cce1b4ea52a829984959120089 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 28 Feb 2011 14:33:01 +0000 Subject: ASoC: Staticise non-exported symbols in SGTL5000 Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index b528f971f3f..b7e97c02689 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -124,7 +124,7 @@ static struct regulator_consumer_supply ldo_consumer[] = { REGULATOR_SUPPLY(LDO_CONSUMER_NAME, NULL), }; -struct regulator_init_data ldo_init_data = { +static struct regulator_init_data ldo_init_data = { .constraints = { .min_uV = 850000, .max_uV = 1600000, @@ -946,7 +946,7 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE) -struct snd_soc_dai_ops sgtl5000_ops = { +static struct snd_soc_dai_ops sgtl5000_ops = { .hw_params = sgtl5000_pcm_hw_params, .digital_mute = sgtl5000_digital_mute, .set_fmt = sgtl5000_set_dai_fmt, @@ -1421,7 +1421,7 @@ static int sgtl5000_remove(struct snd_soc_codec *codec) return 0; } -struct snd_soc_codec_driver sgtl5000_driver = { +static struct snd_soc_codec_driver sgtl5000_driver = { .probe = sgtl5000_probe, .remove = sgtl5000_remove, .suspend = sgtl5000_suspend, -- cgit v1.2.3 From e1412e636eeb84023b79e525909b366a9b28a185 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 22 Feb 2011 20:16:34 -0700 Subject: ASoC: Tegra: Fix error handling in DMA channel alloc tegra_dma_allocate_channel() returns NULL on errors, not an error pointer. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 40540b175f5..3c271f95358 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -161,8 +161,8 @@ static int tegra_pcm_open(struct snd_pcm_substream *substream) prtd->dma_req[1].dev = prtd; prtd->dma_chan = tegra_dma_allocate_channel(TEGRA_DMA_MODE_ONESHOT); - if (IS_ERR(prtd->dma_chan)) { - ret = PTR_ERR(prtd->dma_chan); + if (prtd->dma_chan == NULL) { + ret = -ENOMEM; goto err; } -- cgit v1.2.3 From b462c6e69a26dd534d6372ed65a6fc7c01073883 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Tue, 1 Mar 2011 12:54:39 +0000 Subject: ASoC: WM8994: Ensure MICBIAS is provided with a clock The patch 'ASoC: WM8994: Improve Playback Robustness' did not handle this case properly. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b17ac1971b0..125bfb6eb24 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1102,6 +1102,13 @@ static int adc_mux_ev(struct snd_soc_dapm_widget *w, return 0; } +static int micbias_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + late_enable_ev(w, kcontrol, event); + return 0; +} + static int dac_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1439,6 +1446,10 @@ SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), SND_SOC_DAPM_INPUT("Clock"), +SND_SOC_DAPM_MICBIAS("MICBIAS", WM8994_MICBIAS, 2, 0), +SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -1754,6 +1765,8 @@ static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { { "AIF2DACDAT", NULL, "AIF1DACDAT" }, { "AIF1ADCDAT", NULL, "AIF2ADCDAT" }, { "AIF2ADCDAT", NULL, "AIF1ADCDAT" }, + { "MICBIAS", NULL, "CLK_SYS" }, + { "MICBIAS", NULL, "MICBIAS Supply" }, }; static const struct snd_soc_dapm_route wm8994_intercon[] = { -- cgit v1.2.3 From 0a93421b6adf8ba127b3eafc4c16e3a14017e2ae Mon Sep 17 00:00:00 2001 From: Javier Martin Date: Tue, 1 Mar 2011 15:02:06 +0100 Subject: ASoC: Fix burstsize and DSP_B format problems in imx-ssi. When choosing IMX_DMA flag, burtsizes are set to its default value (0) which leads to driver malfunction. Change them to 4. DSP_B interface needs additional flag to match DSP_B formats as described in several codecs as wm8741 and aic3205. Signed-off-by: Javier Martin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-ssi.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 30894ea7f33..bc92ec62000 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -108,7 +108,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) break; case SND_SOC_DAIFMT_DSP_B: /* data on rising edge of bclk, frame high with data */ - strcr |= SSI_STCR_TFSL; + strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0; break; case SND_SOC_DAIFMT_DSP_A: /* data on rising edge of bclk, frame high 1clk before data */ @@ -656,6 +656,9 @@ static int imx_ssi_probe(struct platform_device *pdev) ssi->dma_params_rx.dma_addr = res->start + SSI_SRX0; ssi->dma_params_tx.dma_addr = res->start + SSI_STX0; + ssi->dma_params_tx.burstsize = 4; + ssi->dma_params_rx.burstsize = 4; + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0"); if (res) ssi->dma_params_tx.dma = res->start; -- cgit v1.2.3 From 49542656ade68b4d4952feec6a4d508fd32be6f1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 24 Feb 2011 20:25:45 +0000 Subject: ASoC: Remove module probe announcements from CODEC drivers Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/ak4104.c | 1 - sound/soc/codecs/cs4270.c | 2 -- 2 files changed, 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index c27f8f59dc6..cbf0b6d400b 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -294,7 +294,6 @@ static struct spi_driver ak4104_spi_driver = { static int __init ak4104_init(void) { - pr_info("Asahi Kasei AK4104 ALSA SoC Codec Driver\n"); return spi_register_driver(&ak4104_spi_driver); } module_init(ak4104_init); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index c0fccadaea9..65f578ff611 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -743,8 +743,6 @@ static struct i2c_driver cs4270_i2c_driver = { static int __init cs4270_init(void) { - pr_info("Cirrus Logic CS4270 ALSA SoC Codec Driver\n"); - return i2c_add_driver(&cs4270_i2c_driver); } module_init(cs4270_init); -- cgit v1.2.3 From 4a5f7bda8fe9d0ed08ed4c5beb5dc3fa62f09d05 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 1 Mar 2011 20:10:46 +0000 Subject: ASoC: Add platform data for WM9081 IRQ pin configuration The WM9081 IRQ output can be either active high or active low and can support either CMOS or open drain modes. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm9081.c | 29 +++++++++++++++++++---------- 1 file changed, 19 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 2103623a077..7883f3ed797 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -167,7 +167,7 @@ struct wm9081_priv { int fll_fref; int fll_fout; int tdm_width; - struct wm9081_retune_mobile_config *retune; + struct wm9081_pdata pdata; }; static int wm9081_volatile_register(struct snd_soc_codec *codec, unsigned int reg) @@ -1082,21 +1082,22 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream, aif4 |= wm9081->bclk / wm9081->fs; /* Apply a ReTune Mobile configuration if it's in use */ - if (wm9081->retune) { - struct wm9081_retune_mobile_config *retune = wm9081->retune; + if (wm9081->pdata.num_retune_configs) { + struct wm9081_pdata *pdata = &wm9081->pdata; struct wm9081_retune_mobile_setting *s; int eq1; best = 0; - best_val = abs(retune->configs[0].rate - wm9081->fs); - for (i = 0; i < retune->num_configs; i++) { - cur_val = abs(retune->configs[i].rate - wm9081->fs); + best_val = abs(pdata->retune_configs[0].rate - wm9081->fs); + for (i = 0; i < pdata->num_retune_configs; i++) { + cur_val = abs(pdata->retune_configs[i].rate - + wm9081->fs); if (cur_val < best_val) { best_val = cur_val; best = i; } } - s = &retune->configs[best]; + s = &pdata->retune_configs[best]; dev_dbg(codec->dev, "ReTune Mobile %s tuned for %dHz\n", s->name, s->rate); @@ -1255,6 +1256,14 @@ static int wm9081_probe(struct snd_soc_codec *codec) return ret; } + reg = 0; + if (wm9081->pdata.irq_high) + reg |= WM9081_IRQ_POL; + if (!wm9081->pdata.irq_cmos) + reg |= WM9081_IRQ_OP_CTRL; + snd_soc_update_bits(codec, WM9081_INTERRUPT_CONTROL, + WM9081_IRQ_POL | WM9081_IRQ_OP_CTRL, reg); + wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Enable zero cross by default */ @@ -1266,7 +1275,7 @@ static int wm9081_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm9081_snd_controls, ARRAY_SIZE(wm9081_snd_controls)); - if (!wm9081->retune) { + if (!wm9081->pdata.num_retune_configs) { dev_dbg(codec->dev, "No ReTune Mobile data, using normal EQ\n"); snd_soc_add_controls(codec, wm9081_eq_controls, @@ -1343,8 +1352,8 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, wm9081->control_data = i2c; if (dev_get_platdata(&i2c->dev)) - memcpy(&wm9081->retune, dev_get_platdata(&i2c->dev), - sizeof(wm9081->retune)); + memcpy(&wm9081->pdata, dev_get_platdata(&i2c->dev), + sizeof(wm9081->pdata)); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9081, &wm9081_dai, 1); -- cgit v1.2.3 From 8959c910884e8faf7987391d194d508e74904c16 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Tue, 1 Mar 2011 12:54:39 +0000 Subject: ASoC: WM8994: Ensure MICBIAS is provided with a clock The patch 'ASoC: WM8994: Improve Playback Robustness' did not handle this case properly. Signed-off-by: Dimitris Papastamos Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 4afbe3b2e44..38bfff7d209 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1103,6 +1103,13 @@ static int adc_mux_ev(struct snd_soc_dapm_widget *w, return 0; } +static int micbias_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + late_enable_ev(w, kcontrol, event); + return 0; +} + static int dac_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1440,6 +1447,10 @@ SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), SND_SOC_DAPM_INPUT("Clock"), +SND_SOC_DAPM_MICBIAS("MICBIAS", WM8994_MICBIAS, 2, 0), +SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), @@ -1755,6 +1766,8 @@ static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { { "AIF2DACDAT", NULL, "AIF1DACDAT" }, { "AIF1ADCDAT", NULL, "AIF2ADCDAT" }, { "AIF2ADCDAT", NULL, "AIF1ADCDAT" }, + { "MICBIAS", NULL, "CLK_SYS" }, + { "MICBIAS", NULL, "MICBIAS Supply" }, }; static const struct snd_soc_dapm_route wm8994_intercon[] = { -- cgit v1.2.3 From c8fb034ccd38ecce61564119bcd56ce6e8e97a80 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Mar 2011 11:01:18 +0000 Subject: ASoC: Fix broken bitfield definitions in WM8978 Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8978.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 4bbc3442703..8dfb0a0da67 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -145,18 +145,18 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { SOC_SINGLE("DAC Playback Limiter Threshold", WM8978_DAC_LIMITER_2, 4, 7, 0), SOC_SINGLE("DAC Playback Limiter Boost", - WM8978_DAC_LIMITER_2, 0, 15, 0), + WM8978_DAC_LIMITER_2, 0, 12, 0), SOC_ENUM("ALC Enable Switch", alc1), SOC_SINGLE("ALC Capture Min Gain", WM8978_ALC_CONTROL_1, 0, 7, 0), SOC_SINGLE("ALC Capture Max Gain", WM8978_ALC_CONTROL_1, 3, 7, 0), - SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 7, 0), + SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 10, 0), SOC_SINGLE("ALC Capture Target", WM8978_ALC_CONTROL_2, 0, 15, 0), SOC_ENUM("ALC Capture Mode", alc3), - SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 15, 0), - SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 15, 0), + SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 10, 0), + SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 10, 0), SOC_SINGLE("ALC Capture Noise Gate Switch", WM8978_NOISE_GATE, 3, 1, 0), SOC_SINGLE("ALC Capture Noise Gate Threshold", @@ -211,8 +211,10 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1), /* DAC / ADC oversampling */ - SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, 8, 1, 0), - SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, 8, 1, 0), + SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, + 5, 1, 0), + SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, + 5, 1, 0), }; /* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */ -- cgit v1.2.3 From 1916a2aae52b8cb8f992599204ce06c0accd08e2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Mar 2011 11:04:10 +0000 Subject: ASoC: Add TLV information for WM8978 DAC limiter Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8978.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 7ce4f49a67c..85e3e630e76 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -93,6 +93,7 @@ static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1); +static const DECLARE_TLV_DB_SCALE(limiter_tlv, 0, 100, 0); static const struct snd_kcontrol_new wm8978_snd_controls[] = { @@ -144,8 +145,8 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { SOC_SINGLE("DAC Playback Limiter Threshold", WM8978_DAC_LIMITER_2, 4, 7, 0), - SOC_SINGLE("DAC Playback Limiter Boost", - WM8978_DAC_LIMITER_2, 0, 12, 0), + SOC_SINGLE_TLV("DAC Playback Limiter Volume", + WM8978_DAC_LIMITER_2, 0, 12, 0, limiter_tlv), SOC_ENUM("ALC Enable Switch", alc1), SOC_SINGLE("ALC Capture Min Gain", WM8978_ALC_CONTROL_1, 0, 7, 0), -- cgit v1.2.3 From 12ea2c782e8bd98bcbf88165b2fd1610d1b9a81d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Mar 2011 18:17:32 +0000 Subject: ASoC: Get the card directly from the DAPM context Rather than indirecting through the CODEC we can look the card up directly from the card pointer in the DAPM context. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b8e6ab739de..8240ab853d7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -368,7 +368,7 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, int i, ret = 0; size_t name_len; struct snd_soc_dapm_path *path; - struct snd_card *card = dapm->codec->card->snd_card; + struct snd_card *card = dapm->card->snd_card; /* add kcontrol */ for (i = 0; i < w->num_kcontrols; i++) { @@ -430,7 +430,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, { struct snd_soc_dapm_path *path = NULL; struct snd_kcontrol *kcontrol; - struct snd_card *card = dapm->codec->card->snd_card; + struct snd_card *card = dapm->card->snd_card; int ret = 0; if (!w->num_kcontrols) { @@ -480,7 +480,7 @@ static inline void dapm_clear_walk(struct snd_soc_dapm_context *dapm) */ static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget) { - int level = snd_power_get_state(widget->dapm->codec->card->snd_card); + int level = snd_power_get_state(widget->dapm->card->snd_card); switch (level) { case SNDRV_CTL_POWER_D3hot: @@ -1083,7 +1083,7 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) */ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) { - struct snd_soc_card *card = dapm->codec->card; + struct snd_soc_card *card = dapm->card; struct snd_soc_dapm_widget *w; struct snd_soc_dapm_context *d; LIST_HEAD(up_list); -- cgit v1.2.3 From 88e8b9a84b41be8ea37cf3bbe192e72f84747f66 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Mar 2011 18:18:24 +0000 Subject: ASoC: Check for a CODEC before dereferencing in DAPM A CODEC pointer is optional (and is checked for in most contexts within DAPM) - add checks to the few places where it was missed. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8240ab853d7..570db8819d9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1517,7 +1517,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, char prefixed_source[80]; int ret = 0; - if (dapm->codec->name_prefix) { + if (dapm->codec && dapm->codec->name_prefix) { snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s", dapm->codec->name_prefix, route->sink); sink = prefixed_sink; @@ -2167,14 +2167,14 @@ int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, return -ENOMEM; name_len = strlen(widget->name) + 1; - if (dapm->codec->name_prefix) + if (dapm->codec && dapm->codec->name_prefix) name_len += 1 + strlen(dapm->codec->name_prefix); w->name = kmalloc(name_len, GFP_KERNEL); if (w->name == NULL) { kfree(w); return -ENOMEM; } - if (dapm->codec->name_prefix) + if (dapm->codec && dapm->codec->name_prefix) snprintf(w->name, name_len, "%s %s", dapm->codec->name_prefix, widget->name); else -- cgit v1.2.3 From e37a4970cd7ab6aec9e848cd3c355fd47fd18afd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Mar 2011 18:21:57 +0000 Subject: ASoC: Add a per-card DAPM context This means that rather than adding the board specific DAPM widgets to a random CODEC DAPM context they can be added to the card itself which is a bit cleaner. Previously there only was one DAPM context and it was tied to the single supported CODEC. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 64befac3f9c..24bfc3ff8e1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1837,6 +1837,11 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } card->snd_card->dev = card->dev; + card->dapm.bias_level = SND_SOC_BIAS_OFF; + card->dapm.dev = card->dev; + card->dapm.card = card; + list_add(&card->dapm.list, &card->dapm_list); + #ifdef CONFIG_PM_SLEEP /* deferred resume work */ INIT_WORK(&card->deferred_resume_work, soc_resume_deferred); @@ -1867,6 +1872,14 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } } + card->dapm.debugfs_dapm = debugfs_create_dir("dapm", + card->debugfs_card_root); + if (!card->dapm.debugfs_dapm) + printk(KERN_WARNING + "Failed to create card DAPM debugfs directory\n"); + + snd_soc_dapm_debugfs_init(&card->dapm); + snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), "%s", card->name); snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), -- cgit v1.2.3 From b8ad29debd7401d257da923480d32838172c431a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Mar 2011 18:35:51 +0000 Subject: ASoC: Allow card DAPM widgets and routes to be set up at registration These will be added after all devices are registered and allow most DAI init functions in machine drivers to be replaced by simple data. Regular controls are not supported as the registration function still works in terms of CODECs. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 24bfc3ff8e1..6a2839c1844 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1872,6 +1872,13 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } } + if (card->dapm_widgets) + snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets, + card->num_dapm_widgets); + if (card->dapm_routes) + snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, + card->num_dapm_routes); + card->dapm.debugfs_dapm = debugfs_create_dir("dapm", card->debugfs_card_root); if (!card->dapm.debugfs_dapm) -- cgit v1.2.3 From 28e9ad921d3b7defd8940a3e30e8241c8ed734db Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Mar 2011 18:36:34 +0000 Subject: ASoC: Add a late_probe() callback to cards This is run after the DAPM widgets and routes are added, allowing setup of things like jacks using the routes. The main card probe() is run before anything else so can't be used for this purpose. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6a2839c1844..8926d38fc5a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1892,6 +1892,15 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), "%s", card->name); + if (card->late_probe) { + ret = card->late_probe(card); + if (ret < 0) { + dev_err(card->dev, "%s late_probe() failed: %d\n", + card->name, ret); + goto probe_aux_dev_err; + } + } + ret = snd_card_register(card->snd_card); if (ret < 0) { printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name); -- cgit v1.2.3 From 1d471cd1261a44a3b28350bef7e5113a4609c106 Mon Sep 17 00:00:00 2001 From: Javier Martin Date: Wed, 2 Mar 2011 14:52:32 +0100 Subject: ASoC: Add TI tlv320aic32x4 codec support. This patch adds support for tlv320aic3205 and tlv320aic3254 codecs. It doesn't include miniDSP support for aic3254. Signed-off-by: Javier Martin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic32x4.c | 794 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.h | 143 +++++++ 4 files changed, 943 insertions(+) create mode 100644 sound/soc/codecs/tlv320aic32x4.c create mode 100644 sound/soc/codecs/tlv320aic32x4.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c04da187129..82a46309ded 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -40,6 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER + select SND_SOC_TVL320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -206,6 +207,9 @@ config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE depends on SPI +config SND_SOC_TVL320AIC32X4 + tristate + config SND_SOC_TLV320AIC3X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3bbb08c512d..b43f9d418c9 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -28,6 +28,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320dac33-objs := tlv320dac33.o snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o @@ -112,6 +113,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_TVL320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c new file mode 100644 index 00000000000..ee82e389603 --- /dev/null +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -0,0 +1,794 @@ +/* + * linux/sound/soc/codecs/tlv320aic32x4.c + * + * Copyright 2011 Vista Silicon S.L. + * + * Author: Javier Martin + * + * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, + * MA 02110-1301, USA. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "tlv320aic32x4.h" + +struct aic32x4_rate_divs { + u32 mclk; + u32 rate; + u8 p_val; + u8 pll_j; + u16 pll_d; + u16 dosr; + u8 ndac; + u8 mdac; + u8 aosr; + u8 nadc; + u8 madc; + u8 blck_N; +}; + +struct aic32x4_priv { + u32 sysclk; + s32 master; + u8 page_no; + void *control_data; + u32 power_cfg; + u32 micpga_routing; + bool swapdacs; +}; + +/* 0dB min, 1dB steps */ +static DECLARE_TLV_DB_SCALE(tlv_step_1, 0, 100, 0); +/* 0dB min, 0.5dB steps */ +static DECLARE_TLV_DB_SCALE(tlv_step_0_5, 0, 50, 0); + +static const struct snd_kcontrol_new aic32x4_snd_controls[] = { + SOC_DOUBLE_R_TLV("PCM Playback Volume", AIC32X4_LDACVOL, + AIC32X4_RDACVOL, 0, 0x30, 0, tlv_step_0_5), + SOC_DOUBLE_R_TLV("HP Driver Gain Volume", AIC32X4_HPLGAIN, + AIC32X4_HPRGAIN, 0, 0x1D, 0, tlv_step_1), + SOC_DOUBLE_R_TLV("LO Driver Gain Volume", AIC32X4_LOLGAIN, + AIC32X4_LORGAIN, 0, 0x1D, 0, tlv_step_1), + SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN, + AIC32X4_HPRGAIN, 6, 0x01, 1), + SOC_DOUBLE_R("LO DAC Playback Switch", AIC32X4_LOLGAIN, + AIC32X4_LORGAIN, 6, 0x01, 1), + SOC_DOUBLE_R("Mic PGA Switch", AIC32X4_LMICPGAVOL, + AIC32X4_RMICPGAVOL, 7, 0x01, 1), + + SOC_SINGLE("ADCFGA Left Mute Switch", AIC32X4_ADCFGA, 7, 1, 0), + SOC_SINGLE("ADCFGA Right Mute Switch", AIC32X4_ADCFGA, 3, 1, 0), + + SOC_DOUBLE_R_TLV("ADC Level Volume", AIC32X4_LADCVOL, + AIC32X4_RADCVOL, 0, 0x28, 0, tlv_step_0_5), + SOC_DOUBLE_R_TLV("PGA Level Volume", AIC32X4_LMICPGAVOL, + AIC32X4_RMICPGAVOL, 0, 0x5f, 0, tlv_step_0_5), + + SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0), + + SOC_SINGLE("AGC Left Switch", AIC32X4_LAGC1, 7, 1, 0), + SOC_SINGLE("AGC Right Switch", AIC32X4_RAGC1, 7, 1, 0), + SOC_DOUBLE_R("AGC Target Level", AIC32X4_LAGC1, AIC32X4_RAGC1, + 4, 0x07, 0), + SOC_DOUBLE_R("AGC Gain Hysteresis", AIC32X4_LAGC1, AIC32X4_RAGC1, + 0, 0x03, 0), + SOC_DOUBLE_R("AGC Hysteresis", AIC32X4_LAGC2, AIC32X4_RAGC2, + 6, 0x03, 0), + SOC_DOUBLE_R("AGC Noise Threshold", AIC32X4_LAGC2, AIC32X4_RAGC2, + 1, 0x1F, 0), + SOC_DOUBLE_R("AGC Max PGA", AIC32X4_LAGC3, AIC32X4_RAGC3, + 0, 0x7F, 0), + SOC_DOUBLE_R("AGC Attack Time", AIC32X4_LAGC4, AIC32X4_RAGC4, + 3, 0x1F, 0), + SOC_DOUBLE_R("AGC Decay Time", AIC32X4_LAGC5, AIC32X4_RAGC5, + 3, 0x1F, 0), + SOC_DOUBLE_R("AGC Noise Debounce", AIC32X4_LAGC6, AIC32X4_RAGC6, + 0, 0x1F, 0), + SOC_DOUBLE_R("AGC Signal Debounce", AIC32X4_LAGC7, AIC32X4_RAGC7, + 0, 0x0F, 0), +}; + +static const struct aic32x4_rate_divs aic32x4_divs[] = { + /* 8k rate */ + {AIC32X4_FREQ_12000000, 8000, 1, 7, 6800, 768, 5, 3, 128, 5, 18, 24}, + {AIC32X4_FREQ_24000000, 8000, 2, 7, 6800, 768, 15, 1, 64, 45, 4, 24}, + {AIC32X4_FREQ_25000000, 8000, 2, 7, 3728, 768, 15, 1, 64, 45, 4, 24}, + /* 11.025k rate */ + {AIC32X4_FREQ_12000000, 11025, 1, 7, 5264, 512, 8, 2, 128, 8, 8, 16}, + {AIC32X4_FREQ_24000000, 11025, 2, 7, 5264, 512, 16, 1, 64, 32, 4, 16}, + /* 16k rate */ + {AIC32X4_FREQ_12000000, 16000, 1, 7, 6800, 384, 5, 3, 128, 5, 9, 12}, + {AIC32X4_FREQ_24000000, 16000, 2, 7, 6800, 384, 15, 1, 64, 18, 5, 12}, + {AIC32X4_FREQ_25000000, 16000, 2, 7, 3728, 384, 15, 1, 64, 18, 5, 12}, + /* 22.05k rate */ + {AIC32X4_FREQ_12000000, 22050, 1, 7, 5264, 256, 4, 4, 128, 4, 8, 8}, + {AIC32X4_FREQ_24000000, 22050, 2, 7, 5264, 256, 16, 1, 64, 16, 4, 8}, + {AIC32X4_FREQ_25000000, 22050, 2, 7, 2253, 256, 16, 1, 64, 16, 4, 8}, + /* 32k rate */ + {AIC32X4_FREQ_12000000, 32000, 1, 7, 1680, 192, 2, 7, 64, 2, 21, 6}, + {AIC32X4_FREQ_24000000, 32000, 2, 7, 1680, 192, 7, 2, 64, 7, 6, 6}, + /* 44.1k rate */ + {AIC32X4_FREQ_12000000, 44100, 1, 7, 5264, 128, 2, 8, 128, 2, 8, 4}, + {AIC32X4_FREQ_24000000, 44100, 2, 7, 5264, 128, 8, 2, 64, 8, 4, 4}, + {AIC32X4_FREQ_25000000, 44100, 2, 7, 2253, 128, 8, 2, 64, 8, 4, 4}, + /* 48k rate */ + {AIC32X4_FREQ_12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4}, + {AIC32X4_FREQ_24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4}, + {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4} +}; + +static const struct snd_kcontrol_new hpl_output_mixer_controls[] = { + SOC_DAPM_SINGLE("L_DAC Switch", AIC32X4_HPLROUTE, 3, 1, 0), + SOC_DAPM_SINGLE("IN1_L Switch", AIC32X4_HPLROUTE, 2, 1, 0), +}; + +static const struct snd_kcontrol_new hpr_output_mixer_controls[] = { + SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_HPRROUTE, 3, 1, 0), + SOC_DAPM_SINGLE("IN1_R Switch", AIC32X4_HPRROUTE, 2, 1, 0), +}; + +static const struct snd_kcontrol_new lol_output_mixer_controls[] = { + SOC_DAPM_SINGLE("L_DAC Switch", AIC32X4_LOLROUTE, 3, 1, 0), +}; + +static const struct snd_kcontrol_new lor_output_mixer_controls[] = { + SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_LORROUTE, 3, 1, 0), +}; + +static const struct snd_kcontrol_new left_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN1_L P Switch", AIC32X4_LMICPGAPIN, 6, 1, 0), + SOC_DAPM_SINGLE("IN2_L P Switch", AIC32X4_LMICPGAPIN, 4, 1, 0), + SOC_DAPM_SINGLE("IN3_L P Switch", AIC32X4_LMICPGAPIN, 2, 1, 0), +}; + +static const struct snd_kcontrol_new right_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN1_R P Switch", AIC32X4_RMICPGAPIN, 6, 1, 0), + SOC_DAPM_SINGLE("IN2_R P Switch", AIC32X4_RMICPGAPIN, 4, 1, 0), + SOC_DAPM_SINGLE("IN3_R P Switch", AIC32X4_RMICPGAPIN, 2, 1, 0), +}; + +static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", AIC32X4_DACSETUP, 7, 0), + SND_SOC_DAPM_MIXER("HPL Output Mixer", SND_SOC_NOPM, 0, 0, + &hpl_output_mixer_controls[0], + ARRAY_SIZE(hpl_output_mixer_controls)), + SND_SOC_DAPM_PGA("HPL Power", AIC32X4_OUTPWRCTL, 5, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("LOL Output Mixer", SND_SOC_NOPM, 0, 0, + &lol_output_mixer_controls[0], + ARRAY_SIZE(lol_output_mixer_controls)), + SND_SOC_DAPM_PGA("LOL Power", AIC32X4_OUTPWRCTL, 3, 0, NULL, 0), + + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", AIC32X4_DACSETUP, 6, 0), + SND_SOC_DAPM_MIXER("HPR Output Mixer", SND_SOC_NOPM, 0, 0, + &hpr_output_mixer_controls[0], + ARRAY_SIZE(hpr_output_mixer_controls)), + SND_SOC_DAPM_PGA("HPR Power", AIC32X4_OUTPWRCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_MIXER("LOR Output Mixer", SND_SOC_NOPM, 0, 0, + &lor_output_mixer_controls[0], + ARRAY_SIZE(lor_output_mixer_controls)), + SND_SOC_DAPM_PGA("LOR Power", AIC32X4_OUTPWRCTL, 2, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Left Input Mixer", SND_SOC_NOPM, 0, 0, + &left_input_mixer_controls[0], + ARRAY_SIZE(left_input_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Input Mixer", SND_SOC_NOPM, 0, 0, + &right_input_mixer_controls[0], + ARRAY_SIZE(right_input_mixer_controls)), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", AIC32X4_ADCSETUP, 6, 0), + SND_SOC_DAPM_MICBIAS("Mic Bias", AIC32X4_MICBIAS, 6, 0), + + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("LOL"), + SND_SOC_DAPM_OUTPUT("LOR"), + SND_SOC_DAPM_INPUT("IN1_L"), + SND_SOC_DAPM_INPUT("IN1_R"), + SND_SOC_DAPM_INPUT("IN2_L"), + SND_SOC_DAPM_INPUT("IN2_R"), + SND_SOC_DAPM_INPUT("IN3_L"), + SND_SOC_DAPM_INPUT("IN3_R"), +}; + +static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { + /* Left Output */ + {"HPL Output Mixer", "L_DAC Switch", "Left DAC"}, + {"HPL Output Mixer", "IN1_L Switch", "IN1_L"}, + + {"HPL Power", NULL, "HPL Output Mixer"}, + {"HPL", NULL, "HPL Power"}, + + {"LOL Output Mixer", "L_DAC Switch", "Left DAC"}, + + {"LOL Power", NULL, "LOL Output Mixer"}, + {"LOL", NULL, "LOL Power"}, + + /* Right Output */ + {"HPR Output Mixer", "R_DAC Switch", "Right DAC"}, + {"HPR Output Mixer", "IN1_R Switch", "IN1_R"}, + + {"HPR Power", NULL, "HPR Output Mixer"}, + {"HPR", NULL, "HPR Power"}, + + {"LOR Output Mixer", "R_DAC Switch", "Right DAC"}, + + {"LOR Power", NULL, "LOR Output Mixer"}, + {"LOR", NULL, "LOR Power"}, + + /* Left input */ + {"Left Input Mixer", "IN1_L P Switch", "IN1_L"}, + {"Left Input Mixer", "IN2_L P Switch", "IN2_L"}, + {"Left Input Mixer", "IN3_L P Switch", "IN3_L"}, + + {"Left ADC", NULL, "Left Input Mixer"}, + + /* Right Input */ + {"Right Input Mixer", "IN1_R P Switch", "IN1_R"}, + {"Right Input Mixer", "IN2_R P Switch", "IN2_R"}, + {"Right Input Mixer", "IN3_R P Switch", "IN3_R"}, + + {"Right ADC", NULL, "Right Input Mixer"}, +}; + +static inline int aic32x4_change_page(struct snd_soc_codec *codec, + unsigned int new_page) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 data[2]; + int ret; + + data[0] = 0x00; + data[1] = new_page & 0xff; + + ret = codec->hw_write(codec->control_data, data, 2); + if (ret == 2) { + aic32x4->page_no = new_page; + return 0; + } else { + return ret; + } +} + +static int aic32x4_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + unsigned int page = reg / 128; + unsigned int fixed_reg = reg % 128; + u8 data[2]; + int ret; + + /* A write to AIC32X4_PSEL is really a non-explicit page change */ + if (reg == AIC32X4_PSEL) + return aic32x4_change_page(codec, val); + + if (aic32x4->page_no != page) { + ret = aic32x4_change_page(codec, page); + if (ret != 0) + return ret; + } + + data[0] = fixed_reg & 0xff; + data[1] = val & 0xff; + + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static unsigned int aic32x4_read(struct snd_soc_codec *codec, unsigned int reg) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + unsigned int page = reg / 128; + unsigned int fixed_reg = reg % 128; + int ret; + + if (aic32x4->page_no != page) { + ret = aic32x4_change_page(codec, page); + if (ret != 0) + return ret; + } + return i2c_smbus_read_byte_data(codec->control_data, fixed_reg & 0xff); +} + +static inline int aic32x4_get_divs(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(aic32x4_divs); i++) { + if ((aic32x4_divs[i].rate == rate) + && (aic32x4_divs[i].mclk == mclk)) { + return i; + } + } + printk(KERN_ERR "aic32x4: master clock and sample rate is not supported\n"); + return -EINVAL; +} + +static int aic32x4_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, aic32x4_dapm_widgets, + ARRAY_SIZE(aic32x4_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, aic32x4_dapm_routes, + ARRAY_SIZE(aic32x4_dapm_routes)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case AIC32X4_FREQ_12000000: + case AIC32X4_FREQ_24000000: + case AIC32X4_FREQ_25000000: + aic32x4->sysclk = freq; + return 0; + } + printk(KERN_ERR "aic32x4: invalid frequency to set DAI system clock\n"); + return -EINVAL; +} + +static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 iface_reg_1; + u8 iface_reg_2; + u8 iface_reg_3; + + iface_reg_1 = snd_soc_read(codec, AIC32X4_IFACE1); + iface_reg_1 = iface_reg_1 & ~(3 << 6 | 3 << 2); + iface_reg_2 = snd_soc_read(codec, AIC32X4_IFACE2); + iface_reg_2 = 0; + iface_reg_3 = snd_soc_read(codec, AIC32X4_IFACE3); + iface_reg_3 = iface_reg_3 & ~(1 << 3); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + aic32x4->master = 1; + iface_reg_1 |= AIC32X4_BCLKMASTER | AIC32X4_WCLKMASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + aic32x4->master = 0; + break; + default: + printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg_1 |= (AIC32X4_DSP_MODE << AIC32X4_PLLJ_SHIFT); + iface_reg_3 |= (1 << 3); /* invert bit clock */ + iface_reg_2 = 0x01; /* add offset 1 */ + break; + case SND_SOC_DAIFMT_DSP_B: + iface_reg_1 |= (AIC32X4_DSP_MODE << AIC32X4_PLLJ_SHIFT); + iface_reg_3 |= (1 << 3); /* invert bit clock */ + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface_reg_1 |= + (AIC32X4_RIGHT_JUSTIFIED_MODE << AIC32X4_PLLJ_SHIFT); + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg_1 |= + (AIC32X4_LEFT_JUSTIFIED_MODE << AIC32X4_PLLJ_SHIFT); + break; + default: + printk(KERN_ERR "aic32x4: invalid DAI interface format\n"); + return -EINVAL; + } + + snd_soc_write(codec, AIC32X4_IFACE1, iface_reg_1); + snd_soc_write(codec, AIC32X4_IFACE2, iface_reg_2); + snd_soc_write(codec, AIC32X4_IFACE3, iface_reg_3); + return 0; +} + +static int aic32x4_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 data; + int i; + + i = aic32x4_get_divs(aic32x4->sysclk, params_rate(params)); + if (i < 0) { + printk(KERN_ERR "aic32x4: sampling rate not supported\n"); + return i; + } + + /* Use PLL as CODEC_CLKIN and DAC_MOD_CLK as BDIV_CLKIN */ + snd_soc_write(codec, AIC32X4_CLKMUX, AIC32X4_PLLCLKIN); + snd_soc_write(codec, AIC32X4_IFACE3, AIC32X4_DACMOD2BCLK); + + /* We will fix R value to 1 and will make P & J=K.D as varialble */ + data = snd_soc_read(codec, AIC32X4_PLLPR); + data &= ~(7 << 4); + snd_soc_write(codec, AIC32X4_PLLPR, + (data | (aic32x4_divs[i].p_val << 4) | 0x01)); + + snd_soc_write(codec, AIC32X4_PLLJ, aic32x4_divs[i].pll_j); + + snd_soc_write(codec, AIC32X4_PLLDMSB, (aic32x4_divs[i].pll_d >> 8)); + snd_soc_write(codec, AIC32X4_PLLDLSB, + (aic32x4_divs[i].pll_d & 0xff)); + + /* NDAC divider value */ + data = snd_soc_read(codec, AIC32X4_NDAC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_NDAC, data | aic32x4_divs[i].ndac); + + /* MDAC divider value */ + data = snd_soc_read(codec, AIC32X4_MDAC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_MDAC, data | aic32x4_divs[i].mdac); + + /* DOSR MSB & LSB values */ + snd_soc_write(codec, AIC32X4_DOSRMSB, aic32x4_divs[i].dosr >> 8); + snd_soc_write(codec, AIC32X4_DOSRLSB, + (aic32x4_divs[i].dosr & 0xff)); + + /* NADC divider value */ + data = snd_soc_read(codec, AIC32X4_NADC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_NADC, data | aic32x4_divs[i].nadc); + + /* MADC divider value */ + data = snd_soc_read(codec, AIC32X4_MADC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_MADC, data | aic32x4_divs[i].madc); + + /* AOSR value */ + snd_soc_write(codec, AIC32X4_AOSR, aic32x4_divs[i].aosr); + + /* BCLK N divider */ + data = snd_soc_read(codec, AIC32X4_BCLKN); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_BCLKN, data | aic32x4_divs[i].blck_N); + + data = snd_soc_read(codec, AIC32X4_IFACE1); + data = data & ~(3 << 4); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + data |= (AIC32X4_WORD_LEN_20BITS << AIC32X4_DOSRMSB_SHIFT); + break; + case SNDRV_PCM_FORMAT_S24_LE: + data |= (AIC32X4_WORD_LEN_24BITS << AIC32X4_DOSRMSB_SHIFT); + break; + case SNDRV_PCM_FORMAT_S32_LE: + data |= (AIC32X4_WORD_LEN_32BITS << AIC32X4_DOSRMSB_SHIFT); + break; + } + snd_soc_write(codec, AIC32X4_IFACE1, data); + + return 0; +} + +static int aic32x4_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 dac_reg; + + dac_reg = snd_soc_read(codec, AIC32X4_DACMUTE) & ~AIC32X4_MUTEON; + if (mute) + snd_soc_write(codec, AIC32X4_DACMUTE, dac_reg | AIC32X4_MUTEON); + else + snd_soc_write(codec, AIC32X4_DACMUTE, dac_reg); + return 0; +} + +static int aic32x4_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 value; + + switch (level) { + case SND_SOC_BIAS_ON: + if (aic32x4->master) { + /* Switch on PLL */ + value = snd_soc_read(codec, AIC32X4_PLLPR); + snd_soc_write(codec, AIC32X4_PLLPR, + (value | AIC32X4_PLLEN)); + + /* Switch on NDAC Divider */ + value = snd_soc_read(codec, AIC32X4_NDAC); + snd_soc_write(codec, AIC32X4_NDAC, + value | AIC32X4_NDACEN); + + /* Switch on MDAC Divider */ + value = snd_soc_read(codec, AIC32X4_MDAC); + snd_soc_write(codec, AIC32X4_MDAC, + value | AIC32X4_MDACEN); + + /* Switch on NADC Divider */ + value = snd_soc_read(codec, AIC32X4_NADC); + snd_soc_write(codec, AIC32X4_NADC, + value | AIC32X4_MDACEN); + + /* Switch on MADC Divider */ + value = snd_soc_read(codec, AIC32X4_MADC); + snd_soc_write(codec, AIC32X4_MADC, + value | AIC32X4_MDACEN); + + /* Switch on BCLK_N Divider */ + value = snd_soc_read(codec, AIC32X4_BCLKN); + snd_soc_write(codec, AIC32X4_BCLKN, + value | AIC32X4_BCLKEN); + } + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (aic32x4->master) { + /* Switch off PLL */ + value = snd_soc_read(codec, AIC32X4_PLLPR); + snd_soc_write(codec, AIC32X4_PLLPR, + (value & ~AIC32X4_PLLEN)); + + /* Switch off NDAC Divider */ + value = snd_soc_read(codec, AIC32X4_NDAC); + snd_soc_write(codec, AIC32X4_NDAC, + value & ~AIC32X4_NDACEN); + + /* Switch off MDAC Divider */ + value = snd_soc_read(codec, AIC32X4_MDAC); + snd_soc_write(codec, AIC32X4_MDAC, + value & ~AIC32X4_MDACEN); + + /* Switch off NADC Divider */ + value = snd_soc_read(codec, AIC32X4_NADC); + snd_soc_write(codec, AIC32X4_NADC, + value & ~AIC32X4_NDACEN); + + /* Switch off MADC Divider */ + value = snd_soc_read(codec, AIC32X4_MADC); + snd_soc_write(codec, AIC32X4_MADC, + value & ~AIC32X4_MDACEN); + value = snd_soc_read(codec, AIC32X4_BCLKN); + + /* Switch off BCLK_N Divider */ + snd_soc_write(codec, AIC32X4_BCLKN, + value & ~AIC32X4_BCLKEN); + } + break; + case SND_SOC_BIAS_OFF: + break; + } + codec->bias_level = level; + return 0; +} + +#define AIC32X4_RATES SNDRV_PCM_RATE_8000_48000 +#define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops aic32x4_ops = { + .hw_params = aic32x4_hw_params, + .digital_mute = aic32x4_mute, + .set_fmt = aic32x4_set_dai_fmt, + .set_sysclk = aic32x4_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver aic32x4_dai = { + .name = "tlv320aic32x4-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AIC32X4_RATES, + .formats = AIC32X4_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AIC32X4_RATES, + .formats = AIC32X4_FORMATS,}, + .ops = &aic32x4_ops, + .symmetric_rates = 1, +}; + +static int aic32x4_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int aic32x4_resume(struct snd_soc_codec *codec) +{ + aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int aic32x4_probe(struct snd_soc_codec *codec) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u32 tmp_reg; + + codec->hw_write = (hw_write_t) i2c_master_send; + codec->control_data = aic32x4->control_data; + + snd_soc_write(codec, AIC32X4_RESET, 0x01); + + /* Power platform configuration */ + if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) { + snd_soc_write(codec, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN | + AIC32X4_MICBIAS_2075V); + } + if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) { + snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE); + } + if (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) { + snd_soc_write(codec, AIC32X4_LDOCTL, AIC32X4_LDOCTLEN); + } + tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE); + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) { + tmp_reg |= AIC32X4_LDOIN_18_36; + } + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) { + tmp_reg |= AIC32X4_LDOIN2HP; + } + snd_soc_write(codec, AIC32X4_CMMODE, tmp_reg); + + /* Do DACs need to be swapped? */ + if (aic32x4->swapdacs) { + snd_soc_write(codec, AIC32X4_DACSETUP, AIC32X4_LDAC2RCHN | AIC32X4_RDAC2LCHN); + } else { + snd_soc_write(codec, AIC32X4_DACSETUP, AIC32X4_LDAC2LCHN | AIC32X4_RDAC2RCHN); + } + + /* Mic PGA routing */ + if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) { + snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K); + } + if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) { + snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K); + } + + aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + snd_soc_add_controls(codec, aic32x4_snd_controls, + ARRAY_SIZE(aic32x4_snd_controls)); + aic32x4_add_widgets(codec); + + return 0; +} + +static int aic32x4_remove(struct snd_soc_codec *codec) +{ + aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { + .read = aic32x4_read, + .write = aic32x4_write, + .probe = aic32x4_probe, + .remove = aic32x4_remove, + .suspend = aic32x4_suspend, + .resume = aic32x4_resume, + .set_bias_level = aic32x4_set_bias_level, +}; + +static __devinit int aic32x4_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct aic32x4_pdata *pdata = i2c->dev.platform_data; + struct aic32x4_priv *aic32x4; + int ret; + + aic32x4 = kzalloc(sizeof(struct aic32x4_priv), GFP_KERNEL); + if (aic32x4 == NULL) + return -ENOMEM; + + aic32x4->control_data = i2c; + i2c_set_clientdata(i2c, aic32x4); + + if (pdata) { + aic32x4->power_cfg = pdata->power_cfg; + aic32x4->swapdacs = pdata->swapdacs; + aic32x4->micpga_routing = pdata->micpga_routing; + } else { + aic32x4->power_cfg = 0; + aic32x4->swapdacs = false; + aic32x4->micpga_routing = 0; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_aic32x4, &aic32x4_dai, 1); + if (ret < 0) + kfree(aic32x4); + return ret; +} + +static __devexit int aic32x4_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id aic32x4_i2c_id[] = { + { "tlv320aic32x4", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id); + +static struct i2c_driver aic32x4_i2c_driver = { + .driver = { + .name = "tlv320aic32x4", + .owner = THIS_MODULE, + }, + .probe = aic32x4_i2c_probe, + .remove = __devexit_p(aic32x4_i2c_remove), + .id_table = aic32x4_i2c_id, +}; + +static int __init aic32x4_modinit(void) +{ + int ret = 0; + + ret = i2c_add_driver(&aic32x4_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register aic32x4 I2C driver: %d\n", + ret); + } + return ret; +} +module_init(aic32x4_modinit); + +static void __exit aic32x4_exit(void) +{ + i2c_del_driver(&aic32x4_i2c_driver); +} +module_exit(aic32x4_exit); + +MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver"); +MODULE_AUTHOR("Javier Martin "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h new file mode 100644 index 00000000000..aae2b244039 --- /dev/null +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -0,0 +1,143 @@ +/* + * tlv320aic32x4.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + + +#ifndef _TLV320AIC32X4_H +#define _TLV320AIC32X4_H + +/* tlv320aic32x4 register space (in decimal to match datasheet) */ + +#define AIC32X4_PAGE1 128 + +#define AIC32X4_PSEL 0 +#define AIC32X4_RESET 1 +#define AIC32X4_CLKMUX 4 +#define AIC32X4_PLLPR 5 +#define AIC32X4_PLLJ 6 +#define AIC32X4_PLLDMSB 7 +#define AIC32X4_PLLDLSB 8 +#define AIC32X4_NDAC 11 +#define AIC32X4_MDAC 12 +#define AIC32X4_DOSRMSB 13 +#define AIC32X4_DOSRLSB 14 +#define AIC32X4_NADC 18 +#define AIC32X4_MADC 19 +#define AIC32X4_AOSR 20 +#define AIC32X4_CLKMUX2 25 +#define AIC32X4_CLKOUTM 26 +#define AIC32X4_IFACE1 27 +#define AIC32X4_IFACE2 28 +#define AIC32X4_IFACE3 29 +#define AIC32X4_BCLKN 30 +#define AIC32X4_IFACE4 31 +#define AIC32X4_IFACE5 32 +#define AIC32X4_IFACE6 33 +#define AIC32X4_DOUTCTL 53 +#define AIC32X4_DINCTL 54 +#define AIC32X4_DACSPB 60 +#define AIC32X4_ADCSPB 61 +#define AIC32X4_DACSETUP 63 +#define AIC32X4_DACMUTE 64 +#define AIC32X4_LDACVOL 65 +#define AIC32X4_RDACVOL 66 +#define AIC32X4_ADCSETUP 81 +#define AIC32X4_ADCFGA 82 +#define AIC32X4_LADCVOL 83 +#define AIC32X4_RADCVOL 84 +#define AIC32X4_LAGC1 86 +#define AIC32X4_LAGC2 87 +#define AIC32X4_LAGC3 88 +#define AIC32X4_LAGC4 89 +#define AIC32X4_LAGC5 90 +#define AIC32X4_LAGC6 91 +#define AIC32X4_LAGC7 92 +#define AIC32X4_RAGC1 94 +#define AIC32X4_RAGC2 95 +#define AIC32X4_RAGC3 96 +#define AIC32X4_RAGC4 97 +#define AIC32X4_RAGC5 98 +#define AIC32X4_RAGC6 99 +#define AIC32X4_RAGC7 100 +#define AIC32X4_PWRCFG (AIC32X4_PAGE1 + 1) +#define AIC32X4_LDOCTL (AIC32X4_PAGE1 + 2) +#define AIC32X4_OUTPWRCTL (AIC32X4_PAGE1 + 9) +#define AIC32X4_CMMODE (AIC32X4_PAGE1 + 10) +#define AIC32X4_HPLROUTE (AIC32X4_PAGE1 + 12) +#define AIC32X4_HPRROUTE (AIC32X4_PAGE1 + 13) +#define AIC32X4_LOLROUTE (AIC32X4_PAGE1 + 14) +#define AIC32X4_LORROUTE (AIC32X4_PAGE1 + 15) +#define AIC32X4_HPLGAIN (AIC32X4_PAGE1 + 16) +#define AIC32X4_HPRGAIN (AIC32X4_PAGE1 + 17) +#define AIC32X4_LOLGAIN (AIC32X4_PAGE1 + 18) +#define AIC32X4_LORGAIN (AIC32X4_PAGE1 + 19) +#define AIC32X4_HEADSTART (AIC32X4_PAGE1 + 20) +#define AIC32X4_MICBIAS (AIC32X4_PAGE1 + 51) +#define AIC32X4_LMICPGAPIN (AIC32X4_PAGE1 + 52) +#define AIC32X4_LMICPGANIN (AIC32X4_PAGE1 + 54) +#define AIC32X4_RMICPGAPIN (AIC32X4_PAGE1 + 55) +#define AIC32X4_RMICPGANIN (AIC32X4_PAGE1 + 57) +#define AIC32X4_FLOATINGINPUT (AIC32X4_PAGE1 + 58) +#define AIC32X4_LMICPGAVOL (AIC32X4_PAGE1 + 59) +#define AIC32X4_RMICPGAVOL (AIC32X4_PAGE1 + 60) + +#define AIC32X4_FREQ_12000000 12000000 +#define AIC32X4_FREQ_24000000 24000000 +#define AIC32X4_FREQ_25000000 25000000 + +#define AIC32X4_WORD_LEN_16BITS 0x00 +#define AIC32X4_WORD_LEN_20BITS 0x01 +#define AIC32X4_WORD_LEN_24BITS 0x02 +#define AIC32X4_WORD_LEN_32BITS 0x03 + +#define AIC32X4_I2S_MODE 0x00 +#define AIC32X4_DSP_MODE 0x01 +#define AIC32X4_RIGHT_JUSTIFIED_MODE 0x02 +#define AIC32X4_LEFT_JUSTIFIED_MODE 0x03 + +#define AIC32X4_AVDDWEAKDISABLE 0x08 +#define AIC32X4_LDOCTLEN 0x01 + +#define AIC32X4_LDOIN_18_36 0x01 +#define AIC32X4_LDOIN2HP 0x02 + +#define AIC32X4_DACSPBLOCK_MASK 0x1f +#define AIC32X4_ADCSPBLOCK_MASK 0x1f + +#define AIC32X4_PLLJ_SHIFT 6 +#define AIC32X4_DOSRMSB_SHIFT 4 + +#define AIC32X4_PLLCLKIN 0x03 + +#define AIC32X4_MICBIAS_LDOIN 0x08 +#define AIC32X4_MICBIAS_2075V 0x60 + +#define AIC32X4_LMICPGANIN_IN2R_10K 0x10 +#define AIC32X4_RMICPGANIN_IN1L_10K 0x10 + +#define AIC32X4_LMICPGAVOL_NOGAIN 0x80 +#define AIC32X4_RMICPGAVOL_NOGAIN 0x80 + +#define AIC32X4_BCLKMASTER 0x08 +#define AIC32X4_WCLKMASTER 0x04 +#define AIC32X4_PLLEN (0x01 << 7) +#define AIC32X4_NDACEN (0x01 << 7) +#define AIC32X4_MDACEN (0x01 << 7) +#define AIC32X4_NADCEN (0x01 << 7) +#define AIC32X4_MADCEN (0x01 << 7) +#define AIC32X4_BCLKEN (0x01 << 7) +#define AIC32X4_DACEN (0x03 << 6) +#define AIC32X4_RDAC2LCHN (0x02 << 2) +#define AIC32X4_LDAC2RCHN (0x02 << 4) +#define AIC32X4_LDAC2LCHN (0x01 << 4) +#define AIC32X4_RDAC2RCHN (0x01 << 2) + +#define AIC32X4_SSTEP2WCLK 0x01 +#define AIC32X4_MUTEON 0x0C +#define AIC32X4_DACMOD2BCLK 0x01 + +#endif /* _TLV320AIC32X4_H */ -- cgit v1.2.3 From 20d660653a488c1c88db6fe51c2459e00cb79230 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Mar 2011 11:07:24 +0000 Subject: ASoC: Fix outdated API usage in tlv320aic32x4 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/tlv320aic32x4.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index ee82e389603..e93b9d1ae1d 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -340,13 +340,13 @@ static inline int aic32x4_get_divs(int mclk, int rate) static int aic32x4_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, aic32x4_dapm_widgets, - ARRAY_SIZE(aic32x4_dapm_widgets)); + snd_soc_dapm_new_controls(&codec->dapm, aic32x4_dapm_widgets, + ARRAY_SIZE(aic32x4_dapm_widgets)); - snd_soc_dapm_add_routes(codec, aic32x4_dapm_routes, + snd_soc_dapm_add_routes(&codec->dapm, aic32x4_dapm_routes, ARRAY_SIZE(aic32x4_dapm_routes)); - snd_soc_dapm_new_widgets(codec); + snd_soc_dapm_new_widgets(&codec->dapm); return 0; } @@ -602,7 +602,7 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } -- cgit v1.2.3 From a2721fd9fa6d4181fd66c58316d3893b597ba118 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Mar 2011 11:09:25 +0000 Subject: ASoC: Add missing debugfs conditionals Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8926d38fc5a..be34f6b9538 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1879,6 +1879,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); +#ifdef CONFIG_DEBUG_FS card->dapm.debugfs_dapm = debugfs_create_dir("dapm", card->debugfs_card_root); if (!card->dapm.debugfs_dapm) @@ -1886,6 +1887,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) "Failed to create card DAPM debugfs directory\n"); snd_soc_dapm_debugfs_init(&card->dapm); +#endif snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), "%s", card->name); -- cgit v1.2.3 From 573f26e3c36ca7036d117bc89d498856073e7284 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 4 Mar 2011 15:18:18 +0800 Subject: ASoC: tlv320dac33: add MODULE_DEVICE_TABLE The device table is required to load modules based on modaliases. Signed-off-by: Axel Lin Acked-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 71d7be8ac48..00b6d87e7bd 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1615,6 +1615,7 @@ static const struct i2c_device_id tlv320dac33_i2c_id[] = { }, { }, }; +MODULE_DEVICE_TABLE(i2c, tlv320dac33_i2c_id); static struct i2c_driver tlv320dac33_i2c_driver = { .driver = { -- cgit v1.2.3 From 79a54ea1ede0f028e5b0be1016bff8f49326a265 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 4 Mar 2011 15:22:03 +0800 Subject: ASoC: Constify i2c_device_id table Signed-off-by: Axel Lin Acked-by: Alexander Sverdlin Acked-by: Timur Tabi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 65f578ff611..0206a17d728 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -719,7 +719,7 @@ static int cs4270_i2c_remove(struct i2c_client *i2c_client) /* * cs4270_id - I2C device IDs supported by this driver */ -static struct i2c_device_id cs4270_id[] = { +static const struct i2c_device_id cs4270_id[] = { {"cs4270", 0}, {} }; diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 1791796216c..41a3ee6f614 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -555,7 +555,7 @@ static struct spi_driver cs4271_spi_driver = { #endif /* defined(CONFIG_SPI_MASTER) */ #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -static struct i2c_device_id cs4271_i2c_id[] = { +static const struct i2c_device_id cs4271_i2c_id[] = { {"cs4271", 0}, {} }; -- cgit v1.2.3 From 841a451f687807110c247ee01b27dab862221d7f Mon Sep 17 00:00:00 2001 From: Javier Martin Date: Mon, 7 Mar 2011 08:47:09 +0100 Subject: ASoC: Add machine driver for Visstrim_M10 board. Visstrim_M10 boards have an external tlcv320aic3205 codec attached to SSI1. This driver glues together both interfaces. External amplifier is not supported in this first version. Signed-off-by: Javier Martin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/Kconfig | 10 +++ sound/soc/imx/Makefile | 2 + sound/soc/imx/mx27vis-aic32x4.c | 137 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 149 insertions(+) create mode 100644 sound/soc/imx/mx27vis-aic32x4.c (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 9eeb8f0d67e..d8f130d39dd 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -30,6 +30,16 @@ config SND_MXC_SOC_WM1133_EV1 Enable support for audio on the i.MX31ADS with the WM1133-EV1 PMIC board with WM8835x fitted. +config SND_SOC_MX27VIS_AIC32X4 + tristate "SoC audio support for Visstrim M10 boards" + depends on MACH_IMX27_VISSTRIM_M10 + select SND_SOC_TVL320AIC32X4 + select SND_MXC_SOC_SSI + select SND_MXC_SOC_MX2 + help + Say Y if you want to add support for SoC audio on Visstrim SM10 + board with TLV320AIC32X4 codec. + config SND_SOC_PHYCORE_AC97 tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" depends on MACH_PCM043 || MACH_PCA100 diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index b67fc02a4ec..d6d609ba7e2 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -10,8 +10,10 @@ obj-$(CONFIG_SND_MXC_SOC_MX2) += snd-soc-imx-mx2.o # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o snd-soc-phycore-ac97-objs := phycore-ac97.o +snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o +obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c new file mode 100644 index 00000000000..054110b91d4 --- /dev/null +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -0,0 +1,137 @@ +/* + * mx27vis-aic32x4.c + * + * Copyright 2011 Vista Silicon S.L. + * + * Author: Javier Martin + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, + * MA 02110-1301, USA. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "../codecs/tlv320aic32x4.h" +#include "imx-ssi.h" + +static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret; + u32 dai_format; + + dai_format = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + /* set codec DAI configuration */ + snd_soc_dai_set_fmt(codec_dai, dai_format); + + /* set cpu DAI configuration */ + snd_soc_dai_set_fmt(cpu_dai, dai_format); + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + 25000000, SND_SOC_CLOCK_OUT); + if (ret) { + pr_err("%s: failed setting codec sysclk\n", __func__); + return ret; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, + SND_SOC_CLOCK_IN); + if (ret) { + pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops mx27vis_aic32x4_snd_ops = { + .hw_params = mx27vis_aic32x4_hw_params, +}; + +static struct snd_soc_dai_link mx27vis_aic32x4_dai = { + .name = "tlv320aic32x4", + .stream_name = "TLV320AIC32X4", + .codec_dai_name = "tlv320aic32x4-hifi", + .platform_name = "imx-pcm-audio.0", + .codec_name = "tlv320aic32x4.0-0018", + .cpu_dai_name = "imx-ssi.0", + .ops = &mx27vis_aic32x4_snd_ops, +}; + +static struct snd_soc_card mx27vis_aic32x4 = { + .name = "visstrim_m10-audio", + .dai_link = &mx27vis_aic32x4_dai, + .num_links = 1, +}; + +static struct platform_device *mx27vis_aic32x4_snd_device; + +static int __init mx27vis_aic32x4_init(void) +{ + int ret; + + mx27vis_aic32x4_snd_device = platform_device_alloc("soc-audio", -1); + if (!mx27vis_aic32x4_snd_device) + return -ENOMEM; + + platform_set_drvdata(mx27vis_aic32x4_snd_device, &mx27vis_aic32x4); + ret = platform_device_add(mx27vis_aic32x4_snd_device); + + if (ret) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + platform_device_put(mx27vis_aic32x4_snd_device); + } + + /* Connect SSI0 as clock slave to SSI1 external pins */ + mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, + MXC_AUDMUX_V1_PCR_SYN | + MXC_AUDMUX_V1_PCR_TFSDIR | + MXC_AUDMUX_V1_PCR_TCLKDIR | + MXC_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1) | + MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1) + ); + mxc_audmux_v1_configure_port(MX27_AUDMUX_PPCR1_SSI_PINS_1, + MXC_AUDMUX_V1_PCR_SYN | + MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) + ); + + return ret; +} + +static void __exit mx27vis_aic32x4_exit(void) +{ + platform_device_unregister(mx27vis_aic32x4_snd_device); +} + +module_init(mx27vis_aic32x4_init); +module_exit(mx27vis_aic32x4_exit); + +MODULE_AUTHOR("Javier Martin "); +MODULE_DESCRIPTION("ALSA SoC AIC32X4 mx27 visstrim"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 004c52e009e3c585eb0692740b714867fd2b3555 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 7 Mar 2011 08:04:54 +0100 Subject: ASoC: neo1973_wm8753: Remove scenario management code. It has been proven to be inflexible to do scenario management in kernel space. Since actual neo1973 board support has not been merged in mainline and this patch has been in the neo1973 tree for some time now it should be safe to remove this functionality without breaking existing userspace. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/samsung/neo1973_wm8753.c | 148 +++---------------------------------- 1 file changed, 11 insertions(+), 137 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index d3cd6888a81..fe929467f93 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -37,17 +37,6 @@ #include "dma.h" #include "s3c24xx-i2s.h" -/* define the scenarios */ -#define NEO_AUDIO_OFF 0 -#define NEO_GSM_CALL_AUDIO_HANDSET 1 -#define NEO_GSM_CALL_AUDIO_HEADSET 2 -#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3 -#define NEO_STEREO_TO_SPEAKERS 4 -#define NEO_STEREO_TO_HEADPHONES 5 -#define NEO_CAPTURE_HANDSET 6 -#define NEO_CAPTURE_HEADSET 7 -#define NEO_CAPTURE_BLUETOOTH 8 - static struct snd_soc_card neo1973; static struct i2c_client *i2c; @@ -224,113 +213,6 @@ static struct snd_soc_ops neo1973_voice_ops = { .hw_free = neo1973_voice_hw_free, }; -static int neo1973_scenario; - -static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = neo1973_scenario; - return 0; -} - -static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - pr_debug("Entered %s\n", __func__); - - switch (neo1973_scenario) { - case NEO_AUDIO_OFF: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_HANDSET: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_enable_pin(dapm, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_HEADSET: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line In"); - snd_soc_dapm_enable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_BLUETOOTH: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_STEREO_TO_SPEAKERS: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_STEREO_TO_HEADPHONES: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_CAPTURE_HANDSET: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_enable_pin(dapm, "Call Mic"); - break; - case NEO_CAPTURE_HEADSET: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_enable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_CAPTURE_BLUETOOTH: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - default: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - } - - snd_soc_dapm_sync(dapm); - - return 0; -} - -static int neo1973_set_scenario(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - - pr_debug("Entered %s\n", __func__); - - if (neo1973_scenario == ucontrol->value.integer.value[0]) - return 0; - - neo1973_scenario = ucontrol->value.integer.value[0]; - set_scenario_endpoints(codec, neo1973_scenario); - return 1; -} - static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; static void lm4857_write_regs(void) @@ -454,26 +336,16 @@ static const struct soc_enum lm4857_mode_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode), }; -static const char *neo_scenarios[] = { - "Off", - "GSM Handset", - "GSM Headset", - "GSM Bluetooth", - "Speakers", - "Headphones", - "Capture Handset", - "Capture Headset", - "Capture Bluetooth" -}; - -static const struct soc_enum neo_scenario_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios), -}; - static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { + SOC_DAPM_PIN_SWITCH("Audio Out"), + SOC_DAPM_PIN_SWITCH("GSM Line Out"), + SOC_DAPM_PIN_SWITCH("GSM Line In"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Call Mic"), + SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, lm4857_get_reg, lm4857_set_reg, stereo_tlv), SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, @@ -482,8 +354,6 @@ static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { lm4857_get_reg, lm4857_set_reg, mono_tlv), SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0], lm4857_get_mode, lm4857_set_mode), - SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0], - neo1973_get_scenario, neo1973_set_scenario), SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0, lm4857_get_reg, lm4857_set_reg), SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0, @@ -520,7 +390,11 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(wm8753_dapm_widgets)); /* set endpoints to default mode */ - set_scenario_endpoints(codec, NEO_AUDIO_OFF); + snd_soc_dapm_disable_pin(dapm, "Audio Out"); + snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); + snd_soc_dapm_disable_pin(dapm, "GSM Line In"); + snd_soc_dapm_disable_pin(dapm, "Headset Mic"); + snd_soc_dapm_disable_pin(dapm, "Call Mic"); /* add neo1973 specific controls */ err = snd_soc_add_controls(codec, wm8753_neo1973_controls, -- cgit v1.2.3 From 9b0a25f0386f9775b69c46ced9b5632b649f00ba Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 7 Mar 2011 08:04:55 +0100 Subject: ASoC: neo1973_wm8753: Move lm4857 specefic code to its own module This patch moves the code for the lm4857 AMP from the neo1973_wm8753 sound board driver to its own module. The lm4857 is a generic AMP IC and not specific to the neo1973. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/lm4857.c | 276 +++++++++++++++++++++++++++++++++++++ sound/soc/samsung/Kconfig | 1 + sound/soc/samsung/lm4857.h | 32 ----- sound/soc/samsung/neo1973_wm8753.c | 269 +++++++++--------------------------- 6 files changed, 347 insertions(+), 236 deletions(-) create mode 100644 sound/soc/codecs/lm4857.c delete mode 100644 sound/soc/samsung/lm4857.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 82a46309ded..37035e6984b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -347,6 +347,9 @@ config SND_SOC_WM9713 tristate # Amp +config SND_SOC_LM4857 + tristate + config SND_SOC_MAX9877 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b43f9d418c9..0663d22e86b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -77,6 +77,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o snd-soc-jz4740-codec-objs := jz4740.o # Amp +snd-soc-lm4857-objs := lm4857.o snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o snd-soc-wm2000-objs := wm2000.o @@ -161,6 +162,7 @@ obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o # Amp +obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c new file mode 100644 index 00000000000..72de47e5d04 --- /dev/null +++ b/sound/soc/codecs/lm4857.c @@ -0,0 +1,276 @@ +/* + * LM4857 AMP driver + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com + * Copyright 2011 Lars-Peter Clausen + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include +#include +#include +#include + +#include +#include +#include + +struct lm4857 { + struct i2c_client *i2c; + uint8_t mode; +}; + +static const uint8_t lm4857_default_regs[] = { + 0x00, 0x00, 0x00, 0x00, +}; + +/* The register offsets in the cache array */ +#define LM4857_MVOL 0 +#define LM4857_LVOL 1 +#define LM4857_RVOL 2 +#define LM4857_CTRL 3 + +/* the shifts required to set these bits */ +#define LM4857_3D 5 +#define LM4857_WAKEUP 5 +#define LM4857_EPGAIN 4 + +static int lm4857_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + uint8_t data; + int ret; + + ret = snd_soc_cache_write(codec, reg, value); + if (ret < 0) + return ret; + + data = (reg << 6) | value; + ret = i2c_master_send(codec->control_data, &data, 1); + if (ret != 1) { + dev_err(codec->dev, "Failed to write register: %d\n", ret); + return ret; + } + + return 0; +} + +static unsigned int lm4857_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + unsigned int val; + int ret; + + ret = snd_soc_cache_read(codec, reg, &val); + if (ret) + return -1; + + return val; +} + +static int lm4857_get_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = lm4857->mode; + + return 0; +} + +static int lm4857_set_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); + uint8_t value = ucontrol->value.integer.value[0]; + + lm4857->mode = value; + + if (codec->dapm.bias_level == SND_SOC_BIAS_ON) + snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, value + 6); + + return 1; +} + +static int lm4857_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, lm4857->mode + 6); + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, 0); + break; + default: + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +static const char *lm4857_mode[] = { + "Earpiece", + "Loudspeaker", + "Loudspeaker + Headphone", + "Headphone", +}; + +static const struct soc_enum lm4857_mode_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode); + +static const struct snd_soc_dapm_widget lm4857_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("IN"), + + SND_SOC_DAPM_OUTPUT("LS"), + SND_SOC_DAPM_OUTPUT("HP"), + SND_SOC_DAPM_OUTPUT("EP"), +}; + +static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); +static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); + +static const struct snd_kcontrol_new lm4857_controls[] = { + SOC_SINGLE_TLV("Left Playback Volume", LM4857_LVOL, 0, 31, 0, + stereo_tlv), + SOC_SINGLE_TLV("Right Playback Volume", LM4857_RVOL, 0, 31, 0, + stereo_tlv), + SOC_SINGLE_TLV("Mono Playback Volume", LM4857_MVOL, 0, 31, 0, + mono_tlv), + SOC_SINGLE("Spk 3D Playback Switch", LM4857_LVOL, LM4857_3D, 1, 0), + SOC_SINGLE("HP 3D Playback Switch", LM4857_RVOL, LM4857_3D, 1, 0), + SOC_SINGLE("Fast Wakeup Playback Switch", LM4857_CTRL, + LM4857_WAKEUP, 1, 0), + SOC_SINGLE("Earpiece 6dB Playback Switch", LM4857_CTRL, + LM4857_EPGAIN, 1, 0), + + SOC_ENUM_EXT("Mode", lm4857_mode_enum, + lm4857_get_mode, lm4857_set_mode), +}; + +/* There is a demux inbetween the the input signal and the output signals. + * Currently there is no easy way to model it in ASoC and since it does not make + * much of a difference in practice simply connect the input direclty to the + * outputs. */ +static const struct snd_soc_dapm_route lm4857_routes[] = { + {"LS", NULL, "IN"}, + {"HP", NULL, "IN"}, + {"EP", NULL, "IN"}, +}; + +static int lm4857_probe(struct snd_soc_codec *codec) +{ + struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + codec->control_data = lm4857->i2c; + + ret = snd_soc_add_controls(codec, lm4857_controls, + ARRAY_SIZE(lm4857_controls)); + if (ret) + return ret; + + ret = snd_soc_dapm_new_controls(dapm, lm4857_dapm_widgets, + ARRAY_SIZE(lm4857_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, lm4857_routes, + ARRAY_SIZE(lm4857_routes)); + if (ret) + return ret; + + snd_soc_dapm_new_widgets(dapm); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_lm4857 = { + .write = lm4857_write, + .read = lm4857_read, + .probe = lm4857_probe, + .reg_cache_size = ARRAY_SIZE(lm4857_default_regs), + .reg_word_size = sizeof(uint8_t), + .reg_cache_default = lm4857_default_regs, + .set_bias_level = lm4857_set_bias_level, +}; + +static int __devinit lm4857_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct lm4857 *lm4857; + int ret; + + lm4857 = kzalloc(sizeof(*lm4857), GFP_KERNEL); + if (!lm4857) + return -ENOMEM; + + i2c_set_clientdata(i2c, lm4857); + + lm4857->i2c = i2c; + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0); + + if (ret) { + kfree(lm4857); + return ret; + } + + return 0; +} + +static int __devexit lm4857_i2c_remove(struct i2c_client *i2c) +{ + struct lm4857 *lm4857 = i2c_get_clientdata(i2c); + + snd_soc_unregister_codec(&i2c->dev); + kfree(lm4857); + + return 0; +} + +static const struct i2c_device_id lm4857_i2c_id[] = { + { "lm4857", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, lm4857_i2c_id); + +static struct i2c_driver lm4857_i2c_driver = { + .driver = { + .name = "lm4857", + .owner = THIS_MODULE, + }, + .probe = lm4857_i2c_probe, + .remove = __devexit_p(lm4857_i2c_remove), + .id_table = lm4857_i2c_id, +}; + +static int __init lm4857_init(void) +{ + return i2c_add_driver(&lm4857_i2c_driver); +} +module_init(lm4857_init); + +static void __exit lm4857_exit(void) +{ + i2c_del_driver(&lm4857_i2c_driver); +} +module_exit(lm4857_exit); + +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_DESCRIPTION("LM4857 amplifier driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index a6a6b5fa2f2..ba78e26e20f 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -39,6 +39,7 @@ config SND_SOC_SAMSUNG_NEO1973_WM8753 depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA01 select SND_S3C24XX_I2S select SND_SOC_WM8753 + select SND_SOC_LM4857 help Say Y if you want to add support for SoC audio on smdk2440 with the WM8753. diff --git a/sound/soc/samsung/lm4857.h b/sound/soc/samsung/lm4857.h deleted file mode 100644 index 0cf5b7011d6..00000000000 --- a/sound/soc/samsung/lm4857.h +++ /dev/null @@ -1,32 +0,0 @@ -/* - * lm4857.h -- ALSA Soc Audio Layer - * - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * Revision history - * 18th Jun 2007 Initial version. - */ - -#ifndef LM4857_H_ -#define LM4857_H_ - -/* The register offsets in the cache array */ -#define LM4857_MVOL 0 -#define LM4857_LVOL 1 -#define LM4857_RVOL 2 -#define LM4857_CTRL 3 - -/* the shifts required to set these bits */ -#define LM4857_3D 5 -#define LM4857_WAKEUP 5 -#define LM4857_EPGAIN 4 - -#endif /*LM4857_H_*/ - diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index fe929467f93..7761827314b 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -17,11 +17,9 @@ #include #include #include -#include #include #include #include -#include #include #include @@ -33,12 +31,10 @@ #include #include "../codecs/wm8753.h" -#include "lm4857.h" #include "dma.h" #include "s3c24xx-i2s.h" static struct snd_soc_card neo1973; -static struct i2c_client *i2c; static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -213,85 +209,7 @@ static struct snd_soc_ops neo1973_voice_ops = { .hw_free = neo1973_voice_hw_free, }; -static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; - -static void lm4857_write_regs(void) -{ - pr_debug("Entered %s\n", __func__); - - if (i2c_master_send(i2c, lm4857_regs, 4) != 4) - printk(KERN_ERR "lm4857: i2c write failed\n"); -} - -static int lm4857_get_reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int reg = mc->reg; - int shift = mc->shift; - int mask = mc->max; - - pr_debug("Entered %s\n", __func__); - - ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask; - return 0; -} - -static int lm4857_set_reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int reg = mc->reg; - int shift = mc->shift; - int mask = mc->max; - - if (((lm4857_regs[reg] >> shift) & mask) == - ucontrol->value.integer.value[0]) - return 0; - - lm4857_regs[reg] &= ~(mask << shift); - lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift; - lm4857_write_regs(); - return 1; -} - -static int lm4857_get_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - u8 value = lm4857_regs[LM4857_CTRL] & 0x0F; - - pr_debug("Entered %s\n", __func__); - - if (value) - value -= 5; - - ucontrol->value.integer.value[0] = value; - return 0; -} - -static int lm4857_set_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - u8 value = ucontrol->value.integer.value[0]; - - pr_debug("Entered %s\n", __func__); - - if (value) - value += 5; - - if ((lm4857_regs[LM4857_CTRL] & 0x0F) == value) - return 0; - - lm4857_regs[LM4857_CTRL] &= 0xF0; - lm4857_regs[LM4857_CTRL] |= value; - lm4857_write_regs(); - return 1; -} - static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { - SND_SOC_DAPM_LINE("Audio Out", NULL), SND_SOC_DAPM_LINE("GSM Line Out", NULL), SND_SOC_DAPM_LINE("GSM Line In", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -299,12 +217,7 @@ static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { }; -static const struct snd_soc_dapm_route dapm_routes[] = { - - /* Connections to the lm4857 amp */ - {"Audio Out", NULL, "LOUT1"}, - {"Audio Out", NULL, "ROUT1"}, - +static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = { /* Connections to the GSM Module */ {"GSM Line Out", NULL, "MONO1"}, {"GSM Line Out", NULL, "MONO2"}, @@ -324,46 +237,14 @@ static const struct snd_soc_dapm_route dapm_routes[] = { {"ACIN", NULL, "ACOP"}, }; -static const char *lm4857_mode[] = { - "Off", - "Call Speaker", - "Stereo Speakers", - "Stereo Speakers + Headphones", - "Headphones" -}; - -static const struct soc_enum lm4857_mode_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode), -}; - -static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); -static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); - -static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { - SOC_DAPM_PIN_SWITCH("Audio Out"), +static const struct snd_kcontrol_new neo1973_wm8753_controls[] = { SOC_DAPM_PIN_SWITCH("GSM Line Out"), SOC_DAPM_PIN_SWITCH("GSM Line In"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Call Mic"), - - SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg, stereo_tlv), - SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg, stereo_tlv), - SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg, mono_tlv), - SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0], - lm4857_get_mode, lm4857_set_mode), - SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Fast Wakeup Playback Switch", LM4857_CTRL, 5, 1, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Earpiece 6dB Playback Switch", LM4857_CTRL, 4, 1, 0, - lm4857_get_reg, lm4857_set_reg), }; + /* * This is an example machine initialisation for a wm8753 connected to a * neo1973 II. It is missing logic to detect hp/mic insertions and logic @@ -387,29 +268,66 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) /* Add neo1973 specific widgets */ snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets, - ARRAY_SIZE(wm8753_dapm_widgets)); + ARRAY_SIZE(wm8753_dapm_widgets)); /* set endpoints to default mode */ - snd_soc_dapm_disable_pin(dapm, "Audio Out"); snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); snd_soc_dapm_disable_pin(dapm, "GSM Line In"); snd_soc_dapm_disable_pin(dapm, "Headset Mic"); snd_soc_dapm_disable_pin(dapm, "Call Mic"); /* add neo1973 specific controls */ - err = snd_soc_add_controls(codec, wm8753_neo1973_controls, - ARRAY_SIZE(8753_neo1973_controls)); + err = snd_soc_add_controls(codec, neo1973_wm8753_controls, + ARRAY_SIZE(neo1973_wm8753_controls)); if (err < 0) return err; /* set up neo1973 specific audio routes */ - err = snd_soc_dapm_add_routes(dapm, dapm_routes, - ARRAY_SIZE(dapm_routes)); + err = snd_soc_dapm_add_routes(dapm, neo1973_wm8753_routes, + ARRAY_SIZE(neo1973_wm8753_routes)); snd_soc_dapm_sync(dapm); return 0; } +static const struct snd_soc_dapm_route neo1973_lm4857_routes[] = { + {"Amp IN", NULL, "ROUT1"}, + {"Amp IN", NULL, "LOUT1"}, + + {"Handset Spk", NULL, "Amp EP"}, + {"Stereo Out", NULL, "Amp LS"}, + {"Headphone", NULL, "Amp HP"}, +}; + +static const struct snd_soc_dapm_widget neo1973_lm4857_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Handset Spk", NULL), + SND_SOC_DAPM_SPK("Stereo Out", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), +}; + +static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) +{ + int ret; + + ret = snd_soc_dapm_new_controls(dapm, neo1973_lm4857_dapm_widgets, + ARRAY_SIZE(neo1973_lm4857_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, neo1973_lm4857_routes, + ARRAY_SIZE(neo1973_lm4857_routes)); + if (ret) + return ret; + + snd_soc_dapm_ignore_suspend(dapm, "Stereo Out"); + snd_soc_dapm_ignore_suspend(dapm, "Handset Spk"); + snd_soc_dapm_ignore_suspend(dapm, "Headphone"); + + snd_soc_dapm_sync(dapm); + + return 0; +} + /* * BT Codec DAI */ @@ -449,83 +367,29 @@ static struct snd_soc_dai_link neo1973_dai[] = { }, }; -static struct snd_soc_card neo1973 = { - .name = "neo1973", - .dai_link = neo1973_dai, - .num_links = ARRAY_SIZE(neo1973_dai), +static struct snd_soc_aux_dev neo1973_aux_devs[] = { + { + .name = "lm4857", + .codec_name = "lm4857.0-007c", + .init = neo1973_lm4857_init, + }, }; -static int lm4857_i2c_probe(struct i2c_client *client, - const struct i2c_device_id *id) -{ - pr_debug("Entered %s\n", __func__); - - i2c = client; - - lm4857_write_regs(); - return 0; -} - -static int lm4857_i2c_remove(struct i2c_client *client) -{ - pr_debug("Entered %s\n", __func__); - - i2c = NULL; - - return 0; -} - -static u8 lm4857_state; - -static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) -{ - pr_debug("Entered %s\n", __func__); - - dev_dbg(&dev->dev, "lm4857_suspend\n"); - lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf; - if (lm4857_state) { - lm4857_regs[LM4857_CTRL] &= 0xf0; - lm4857_write_regs(); - } - return 0; -} - -static int lm4857_resume(struct i2c_client *dev) -{ - pr_debug("Entered %s\n", __func__); - - if (lm4857_state) { - lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f); - lm4857_write_regs(); - } - return 0; -} - -static void lm4857_shutdown(struct i2c_client *dev) -{ - pr_debug("Entered %s\n", __func__); - - dev_dbg(&dev->dev, "lm4857_shutdown\n"); - lm4857_regs[LM4857_CTRL] &= 0xf0; - lm4857_write_regs(); -} - -static const struct i2c_device_id lm4857_i2c_id[] = { - { "neo1973_lm4857", 0 }, - { } +static struct snd_soc_codec_conf neo1973_codec_conf[] = { + { + .dev_name = "lm4857.0-007c", + .name_prefix = "Amp", + }, }; -static struct i2c_driver lm4857_i2c_driver = { - .driver = { - .name = "LM4857 I2C Amp", - .owner = THIS_MODULE, - }, - .suspend = lm4857_suspend, - .resume = lm4857_resume, - .shutdown = lm4857_shutdown, - .probe = lm4857_i2c_probe, - .remove = lm4857_i2c_remove, - .id_table = lm4857_i2c_id, +static struct snd_soc_card neo1973 = { + .name = "neo1973", + .dai_link = neo1973_dai, + .num_links = ARRAY_SIZE(neo1973_dai), + .aux_dev = neo1973_aux_devs, + .num_aux_devs = ARRAY_SIZE(neo1973_aux_devs), + .codec_conf = neo1973_codec_conf, + .num_configs = ARRAY_SIZE(neo1973_codec_conf), }; static struct platform_device *neo1973_snd_device; @@ -554,8 +418,6 @@ static int __init neo1973_init(void) return ret; } - ret = i2c_add_driver(&lm4857_i2c_driver); - if (ret != 0) platform_device_unregister(neo1973_snd_device); @@ -566,7 +428,6 @@ static void __exit neo1973_exit(void) { pr_debug("Entered %s\n", __func__); - i2c_del_driver(&lm4857_i2c_driver); platform_device_unregister(neo1973_snd_device); } -- cgit v1.2.3 From da70a3e926b9d8456240ede505c8854d89c90e49 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 7 Mar 2011 08:04:56 +0100 Subject: ASoC: neo1973_gta02_wm8753: Remove lm4853_{set,get}_state This patch drops the lm4853_{set,get}_state functions and the "Amp State Switch" control. Those were noops which existed to maintain alsa state file compatibility. Since the control names have changed due to internal changes in the ASoC core and state file compatibility was broken anyway it makes sense to drop them. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/samsung/neo1973_gta02_wm8753.c | 55 ++++---------------------------- 1 file changed, 7 insertions(+), 48 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c index 95ebf812b14..b007dd7902e 100644 --- a/sound/soc/samsung/neo1973_gta02_wm8753.c +++ b/sound/soc/samsung/neo1973_gta02_wm8753.c @@ -194,47 +194,14 @@ static struct snd_soc_ops neo1973_gta02_voice_ops = { .hw_free = neo1973_gta02_voice_hw_free, }; -#define LM4853_AMP 1 -#define LM4853_SPK 2 - -static u8 lm4853_state; - -/* This has no effect, it exists only to maintain compatibility with - * existing ALSA state files. - */ -static int lm4853_set_state(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - int val = ucontrol->value.integer.value[0]; - - if (val) - lm4853_state |= LM4853_AMP; - else - lm4853_state &= ~LM4853_AMP; - - return 0; -} - -static int lm4853_get_state(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = lm4853_state & LM4853_AMP; - - return 0; -} +static int gta02_speaker_enabled; static int lm4853_set_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - int val = ucontrol->value.integer.value[0]; - - if (val) { - lm4853_state |= LM4853_SPK; - gpio_set_value(GTA02_GPIO_HP_IN, 0); - } else { - lm4853_state &= ~LM4853_SPK; - gpio_set_value(GTA02_GPIO_HP_IN, 1); - } + gta02_speaker_enabled = ucontrol->value.integer.value[0]; + + gpio_set_value(GTA02_GPIO_HP_IN, !gta02_speaker_enabled); return 0; } @@ -242,14 +209,12 @@ static int lm4853_set_spk(struct snd_kcontrol *kcontrol, static int lm4853_get_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = (lm4853_state & LM4853_SPK) >> 1; - + ucontrol->value.integer.value[0] = gta02_speaker_enabled; return 0; } static int lm4853_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, - int event) + struct snd_kcontrol *k, int event) { gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(event)); @@ -304,13 +269,7 @@ static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = { SOC_DAPM_PIN_SWITCH("Handset Mic"), SOC_DAPM_PIN_SWITCH("Handset Spk"), - /* This has no effect, it exists only to maintain compatibility with - * existing ALSA state files. - */ - SOC_SINGLE_EXT("Amp State Switch", 6, 0, 1, 0, - lm4853_get_state, - lm4853_set_state), - SOC_SINGLE_EXT("Amp Spk Switch", 7, 0, 1, 0, + SOC_SINGLE_BOOL_EXT("Amp Spk Switch", 0, lm4853_get_spk, lm4853_set_spk), }; -- cgit v1.2.3 From b7874e4490aa201be2f904ffd8bb88a7b9970c6a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 7 Mar 2011 08:04:57 +0100 Subject: ASoC: neo1973_gta02_wm8753: Use gpio_request_array to request gpios Using gpio_request_array instead of requesting and setting up each gpio by hand makes the code more readable and more compact. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/samsung/neo1973_gta02_wm8753.c | 52 +++++++++++--------------------- 1 file changed, 17 insertions(+), 35 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c index b007dd7902e..28d8448e959 100644 --- a/sound/soc/samsung/neo1973_gta02_wm8753.c +++ b/sound/soc/samsung/neo1973_gta02_wm8753.c @@ -370,6 +370,11 @@ static struct snd_soc_card neo1973_gta02 = { .num_links = ARRAY_SIZE(neo1973_gta02_dai), }; +static const struct gpio neo1973_gta02_gpios[] = { + { GTA02_GPIO_HP_IN, GPIOF_OUT_INIT_HIGH, "GTA02_HP_IN" }, + { GTA02_GPIO_AMP_SHUT, GPIOF_OUT_INIT_HIGH, "GTA02_AMP_SHUT" }, +}; + static struct platform_device *neo1973_gta02_snd_device; static int __init neo1973_gta02_init(void) @@ -382,9 +387,16 @@ static int __init neo1973_gta02_init(void) return -ENODEV; } + ret = gpio_request_array(neo1973_gta02_gpios, + ARRAY_SIZE(neo1973_gta02_gpios)); + if (ret) + return ret; + neo1973_gta02_snd_device = platform_device_alloc("soc-audio", -1); - if (!neo1973_gta02_snd_device) - return -ENOMEM; + if (!neo1973_gta02_snd_device) { + ret = -ENOMEM; + goto err_gpio_free; + } /* register bluetooth DAI here */ ret = snd_soc_register_dai(&neo1973_gta02_snd_device->dev, &bt_dai); @@ -397,43 +409,14 @@ static int __init neo1973_gta02_init(void) if (ret) goto err_unregister_dai; - /* Initialise GPIOs used by amp */ - ret = gpio_request(GTA02_GPIO_HP_IN, "GTA02_HP_IN"); - if (ret) { - pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_HP_IN); - goto err_del_device; - } - - ret = gpio_direction_output(GTA02_GPIO_HP_IN, 1); - if (ret) { - pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_HP_IN); - goto err_free_gpio_hp_in; - } - - ret = gpio_request(GTA02_GPIO_AMP_SHUT, "GTA02_AMP_SHUT"); - if (ret) { - pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_AMP_SHUT); - goto err_free_gpio_hp_in; - } - - ret = gpio_direction_output(GTA02_GPIO_AMP_SHUT, 1); - if (ret) { - pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_AMP_SHUT); - goto err_free_gpio_amp_shut; - } - return 0; -err_free_gpio_amp_shut: - gpio_free(GTA02_GPIO_AMP_SHUT); -err_free_gpio_hp_in: - gpio_free(GTA02_GPIO_HP_IN); -err_del_device: - platform_device_del(neo1973_gta02_snd_device); err_unregister_dai: snd_soc_unregister_dai(&neo1973_gta02_snd_device->dev); err_put_device: platform_device_put(neo1973_gta02_snd_device); +err_gpio_free: + gpio_free_array(neo1973_gta02_gpios, ARRAY_SIZE(neo1973_gta02_gpios)); return ret; } module_init(neo1973_gta02_init); @@ -442,8 +425,7 @@ static void __exit neo1973_gta02_exit(void) { snd_soc_unregister_dai(&neo1973_gta02_snd_device->dev); platform_device_unregister(neo1973_gta02_snd_device); - gpio_free(GTA02_GPIO_HP_IN); - gpio_free(GTA02_GPIO_AMP_SHUT); + gpio_free_array(neo1973_gta02_gpios, ARRAY_SIZE(neo1973_gta02_gpios)); } module_exit(neo1973_gta02_exit); -- cgit v1.2.3 From f5c4ffbd65892829f7ec503a89ea24eb0fc952dc Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 7 Mar 2011 08:04:58 +0100 Subject: ASoC: Samsung: Merge neo1937_wm8753 and neo1973_gta02_wm8753 sound board driver The neo1973(GTA01) and neo1973_gta02(GTA02) have a very similar audio hardware setup. They both use the same codec with the same routing to the gsm modem and bluetooth chip. But they do use different AMPs though and there are some minor differences in the speaker setup. As a result most of the code of those two drivers is identical. So from a maintenance point of view it makes sense to merge them into a single driver. It also reduces the size of kernel images supporting both the GTA01 and GTA02. As a side-effect of this merge the GTA01 for example gains support for routing audio to and from the bluetooth DAI. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/samsung/Kconfig | 19 +- sound/soc/samsung/Makefile | 2 - sound/soc/samsung/neo1973_gta02_wm8753.c | 435 ------------------------------- sound/soc/samsung/neo1973_wm8753.c | 262 ++++++++++++++----- 4 files changed, 196 insertions(+), 522 deletions(-) delete mode 100644 sound/soc/samsung/neo1973_gta02_wm8753.c (limited to 'sound') diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index ba78e26e20f..c3014e82157 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -35,24 +35,15 @@ config SND_SAMSUNG_I2S tristate config SND_SOC_SAMSUNG_NEO1973_WM8753 - tristate "SoC I2S Audio support for NEO1973 - WM8753" - depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA01 + tristate "Audio support for Openmoko Neo1973 Smartphones (GTA01/GTA02)" + depends on SND_SOC_SAMSUNG && (MACH_NEO1973_GTA01 || MACH_NEO1973_GTA02) select SND_S3C24XX_I2S select SND_SOC_WM8753 - select SND_SOC_LM4857 + select SND_SOC_LM4857 if MACH_NEO1973_GTA01 help - Say Y if you want to add support for SoC audio on smdk2440 - with the WM8753. + Say Y here to enable audio support for the Openmoko Neo1973 + Smartphones. -config SND_SOC_SAMSUNG_NEO1973_GTA02_WM8753 - tristate "Audio support for the Openmoko Neo FreeRunner (GTA02)" - depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA02 - select SND_S3C24XX_I2S - select SND_SOC_WM8753 - help - This driver provides audio support for the Openmoko Neo FreeRunner - smartphone. - config SND_SOC_SAMSUNG_JIVE_WM8750 tristate "SoC I2S Audio support for Jive" depends on SND_SOC_SAMSUNG && MACH_JIVE diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 705d4e8a672..294dec05c26 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -20,7 +20,6 @@ obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-i2s.o # S3C24XX Machine Support snd-soc-jive-wm8750-objs := jive_wm8750.o snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o -snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o @@ -38,7 +37,6 @@ snd-soc-smdk-spdif-objs := smdk_spdif.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o -obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o obj-$(CONFIG_SND_SOC_SAMSUNG_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o obj-$(CONFIG_SND_SOC_SAMSUNG_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c deleted file mode 100644 index 28d8448e959..00000000000 --- a/sound/soc/samsung/neo1973_gta02_wm8753.c +++ /dev/null @@ -1,435 +0,0 @@ -/* - * neo1973_gta02_wm8753.c -- SoC audio for Openmoko Freerunner(GTA02) - * - * Copyright 2007 Openmoko Inc - * Author: Graeme Gregory - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * Copyright 2009 Wolfson Microelectronics - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include - -#include - -#include -#include -#include - -#include "../codecs/wm8753.h" -#include "s3c24xx-i2s.h" - -static struct snd_soc_card neo1973_gta02; - -static int neo1973_gta02_hifi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int pll_out = 0, bclk = 0; - int ret = 0; - unsigned long iis_clkrate; - - iis_clkrate = s3c24xx_i2s_get_clockrate(); - - switch (params_rate(params)) { - case 8000: - case 16000: - pll_out = 12288000; - break; - case 48000: - bclk = WM8753_BCLK_DIV_4; - pll_out = 12288000; - break; - case 96000: - bclk = WM8753_BCLK_DIV_2; - pll_out = 12288000; - break; - case 11025: - bclk = WM8753_BCLK_DIV_16; - pll_out = 11289600; - break; - case 22050: - bclk = WM8753_BCLK_DIV_8; - pll_out = 11289600; - break; - case 44100: - bclk = WM8753_BCLK_DIV_4; - pll_out = 11289600; - break; - case 88200: - bclk = WM8753_BCLK_DIV_2; - pll_out = 11289600; - break; - } - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set MCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, - S3C2410_IISMOD_32FS); - if (ret < 0) - return ret; - - /* set codec BCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(codec_dai, - WM8753_BCLKDIV, bclk); - if (ret < 0) - return ret; - - /* set prescaler division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - S3C24XX_PRESCALE(4, 4)); - if (ret < 0) - return ret; - - /* codec PLL input is PCLK/4 */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, - iis_clkrate / 4, pll_out); - if (ret < 0) - return ret; - - return 0; -} - -static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - - /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); -} - -/* - * Neo1973 WM8753 HiFi DAI opserations. - */ -static struct snd_soc_ops neo1973_gta02_hifi_ops = { - .hw_params = neo1973_gta02_hifi_hw_params, - .hw_free = neo1973_gta02_hifi_hw_free, -}; - -static int neo1973_gta02_voice_hw_params( - struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - unsigned int pcmdiv = 0; - int ret = 0; - unsigned long iis_clkrate; - - iis_clkrate = s3c24xx_i2s_get_clockrate(); - - if (params_rate(params) != 8000) - return -EINVAL; - if (params_channels(params) != 1) - return -EINVAL; - - pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */ - - /* todo: gg check mode (DSP_B) against CSR datasheet */ - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, - 12288000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set codec PCM division for sample rate */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, - pcmdiv); - if (ret < 0) - return ret; - - /* configure and enable PLL for 12.288MHz output */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, - iis_clkrate / 4, 12288000); - if (ret < 0) - return ret; - - return 0; -} - -static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - - /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); -} - -static struct snd_soc_ops neo1973_gta02_voice_ops = { - .hw_params = neo1973_gta02_voice_hw_params, - .hw_free = neo1973_gta02_voice_hw_free, -}; - -static int gta02_speaker_enabled; - -static int lm4853_set_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - gta02_speaker_enabled = ucontrol->value.integer.value[0]; - - gpio_set_value(GTA02_GPIO_HP_IN, !gta02_speaker_enabled); - - return 0; -} - -static int lm4853_get_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = gta02_speaker_enabled; - return 0; -} - -static int lm4853_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(event)); - - return 0; -} - -static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Stereo Out", lm4853_event), - SND_SOC_DAPM_LINE("GSM Line Out", NULL), - SND_SOC_DAPM_LINE("GSM Line In", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Handset Mic", NULL), - SND_SOC_DAPM_SPK("Handset Spk", NULL), -}; - - -/* example machine audio_mapnections */ -static const struct snd_soc_dapm_route audio_map[] = { - - /* Connections to the lm4853 amp */ - {"Stereo Out", NULL, "LOUT1"}, - {"Stereo Out", NULL, "ROUT1"}, - - /* Connections to the GSM Module */ - {"GSM Line Out", NULL, "MONO1"}, - {"GSM Line Out", NULL, "MONO2"}, - {"RXP", NULL, "GSM Line In"}, - {"RXN", NULL, "GSM Line In"}, - - /* Connections to Headset */ - {"MIC1", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Headset Mic"}, - - /* Call Mic */ - {"MIC2", NULL, "Mic Bias"}, - {"MIC2N", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Handset Mic"}, - - /* Call Speaker */ - {"Handset Spk", NULL, "LOUT2"}, - {"Handset Spk", NULL, "ROUT2"}, - - /* Connect the ALC pins */ - {"ACIN", NULL, "ACOP"}, -}; - -static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = { - SOC_DAPM_PIN_SWITCH("Stereo Out"), - SOC_DAPM_PIN_SWITCH("GSM Line Out"), - SOC_DAPM_PIN_SWITCH("GSM Line In"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Handset Mic"), - SOC_DAPM_PIN_SWITCH("Handset Spk"), - - SOC_SINGLE_BOOL_EXT("Amp Spk Switch", 0, - lm4853_get_spk, - lm4853_set_spk), -}; - -/* - * This is an example machine initialisation for a wm8753 connected to a - * neo1973 GTA02. - */ -static int neo1973_gta02_wm8753_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - int err; - - /* set up NC codec pins */ - snd_soc_dapm_nc_pin(dapm, "OUT3"); - snd_soc_dapm_nc_pin(dapm, "OUT4"); - snd_soc_dapm_nc_pin(dapm, "LINE1"); - snd_soc_dapm_nc_pin(dapm, "LINE2"); - - /* Add neo1973 gta02 specific widgets */ - snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets, - ARRAY_SIZE(wm8753_dapm_widgets)); - - /* add neo1973 gta02 specific controls */ - err = snd_soc_add_controls(codec, wm8753_neo1973_gta02_controls, - ARRAY_SIZE(wm8753_neo1973_gta02_controls)); - - if (err < 0) - return err; - - /* set up neo1973 gta02 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - /* set endpoints to default off mode */ - snd_soc_dapm_disable_pin(dapm, "Stereo Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Handset Mic"); - snd_soc_dapm_disable_pin(dapm, "Handset Spk"); - - /* allow audio paths from the GSM modem to run during suspend */ - snd_soc_dapm_ignore_suspend(dapm, "Stereo Out"); - snd_soc_dapm_ignore_suspend(dapm, "GSM Line Out"); - snd_soc_dapm_ignore_suspend(dapm, "GSM Line In"); - snd_soc_dapm_ignore_suspend(dapm, "Headset Mic"); - snd_soc_dapm_ignore_suspend(dapm, "Handset Mic"); - snd_soc_dapm_ignore_suspend(dapm, "Handset Spk"); - - snd_soc_dapm_sync(dapm); - - return 0; -} - -/* - * BT Codec DAI - */ -static struct snd_soc_dai_driver bt_dai = { - .name = "bluetooth-dai", - .playback = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, -}; - -static struct snd_soc_dai_link neo1973_gta02_dai[] = { -{ /* Hifi Playback - for similatious use with voice below */ - .name = "WM8753", - .stream_name = "WM8753 HiFi", - .cpu_dai_name = "s3c24xx-iis", - .codec_dai_name = "wm8753-hifi", - .init = neo1973_gta02_wm8753_init, - .platform_name = "samsung-audio", - .codec_name = "wm8753-codec.0-001a", - .ops = &neo1973_gta02_hifi_ops, -}, -{ /* Voice via BT */ - .name = "Bluetooth", - .stream_name = "Voice", - .cpu_dai_name = "bluetooth-dai", - .codec_dai_name = "wm8753-voice", - .ops = &neo1973_gta02_voice_ops, - .codec_name = "wm8753-codec.0-001a", - .platform_name = "samsung-audio", -}, -}; - -static struct snd_soc_card neo1973_gta02 = { - .name = "neo1973-gta02", - .dai_link = neo1973_gta02_dai, - .num_links = ARRAY_SIZE(neo1973_gta02_dai), -}; - -static const struct gpio neo1973_gta02_gpios[] = { - { GTA02_GPIO_HP_IN, GPIOF_OUT_INIT_HIGH, "GTA02_HP_IN" }, - { GTA02_GPIO_AMP_SHUT, GPIOF_OUT_INIT_HIGH, "GTA02_AMP_SHUT" }, -}; - -static struct platform_device *neo1973_gta02_snd_device; - -static int __init neo1973_gta02_init(void) -{ - int ret; - - if (!machine_is_neo1973_gta02()) { - printk(KERN_INFO - "Only GTA02 is supported by this ASoC driver\n"); - return -ENODEV; - } - - ret = gpio_request_array(neo1973_gta02_gpios, - ARRAY_SIZE(neo1973_gta02_gpios)); - if (ret) - return ret; - - neo1973_gta02_snd_device = platform_device_alloc("soc-audio", -1); - if (!neo1973_gta02_snd_device) { - ret = -ENOMEM; - goto err_gpio_free; - } - - /* register bluetooth DAI here */ - ret = snd_soc_register_dai(&neo1973_gta02_snd_device->dev, &bt_dai); - if (ret) - goto err_put_device; - - platform_set_drvdata(neo1973_gta02_snd_device, &neo1973_gta02); - ret = platform_device_add(neo1973_gta02_snd_device); - - if (ret) - goto err_unregister_dai; - - return 0; - -err_unregister_dai: - snd_soc_unregister_dai(&neo1973_gta02_snd_device->dev); -err_put_device: - platform_device_put(neo1973_gta02_snd_device); -err_gpio_free: - gpio_free_array(neo1973_gta02_gpios, ARRAY_SIZE(neo1973_gta02_gpios)); - return ret; -} -module_init(neo1973_gta02_init); - -static void __exit neo1973_gta02_exit(void) -{ - snd_soc_unregister_dai(&neo1973_gta02_snd_device->dev); - platform_device_unregister(neo1973_gta02_snd_device); - gpio_free_array(neo1973_gta02_gpios, ARRAY_SIZE(neo1973_gta02_gpios)); -} -module_exit(neo1973_gta02_exit); - -/* Module information */ -MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org"); -MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 GTA02"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 7761827314b..37cfbb8ca39 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -1,41 +1,32 @@ /* - * neo1973_wm8753.c -- SoC audio for Neo1973 + * neo1973_wm8753.c -- SoC audio for Openmoko Neo1973 and Freerunner devices * + * Copyright 2007 Openmoko Inc + * Author: Graeme Gregory * Copyright 2007 Wolfson Microelectronics PLC. * Author: Graeme Gregory * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com + * Copyright 2009 Wolfson Microelectronics * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. - * */ #include -#include -#include -#include #include -#include -#include +#include + #include #include -#include -#include -#include -#include -#include - #include +#include #include "../codecs/wm8753.h" -#include "dma.h" #include "s3c24xx-i2s.h" -static struct snd_soc_card neo1973; - static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -46,8 +37,6 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, int ret = 0; unsigned long iis_clkrate; - pr_debug("Entered %s\n", __func__); - iis_clkrate = s3c24xx_i2s_get_clockrate(); switch (params_rate(params)) { @@ -132,8 +121,6 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - pr_debug("Entered %s\n", __func__); - /* disable the PLL */ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); } @@ -155,8 +142,6 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, int ret = 0; unsigned long iis_clkrate; - pr_debug("Entered %s\n", __func__); - iis_clkrate = s3c24xx_i2s_get_clockrate(); if (params_rate(params) != 8000) @@ -198,8 +183,6 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - pr_debug("Entered %s\n", __func__); - /* disable the PLL */ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); } @@ -209,14 +192,15 @@ static struct snd_soc_ops neo1973_voice_ops = { .hw_free = neo1973_voice_hw_free, }; -static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { +/* Shared routes and controls */ + +static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = { SND_SOC_DAPM_LINE("GSM Line Out", NULL), SND_SOC_DAPM_LINE("GSM Line In", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Call Mic", NULL), + SND_SOC_DAPM_MIC("Handset Mic", NULL), }; - static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = { /* Connections to the GSM Module */ {"GSM Line Out", NULL, "MONO1"}, @@ -231,7 +215,7 @@ static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = { /* Call Mic */ {"MIC2", NULL, "Mic Bias"}, {"MIC2N", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Call Mic"}, + {"Mic Bias", NULL, "Handset Mic"}, /* Connect the ALC pins */ {"ACIN", NULL, "ACOP"}, @@ -241,55 +225,157 @@ static const struct snd_kcontrol_new neo1973_wm8753_controls[] = { SOC_DAPM_PIN_SWITCH("GSM Line Out"), SOC_DAPM_PIN_SWITCH("GSM Line In"), SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Call Mic"), + SOC_DAPM_PIN_SWITCH("Handset Mic"), }; +/* GTA02 specific routes and controlls */ + +#ifdef CONFIG_MACH_NEO1973_GTA02 + +static int gta02_speaker_enabled; + +static int lm4853_set_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + gta02_speaker_enabled = ucontrol->value.integer.value[0]; + + gpio_set_value(GTA02_GPIO_HP_IN, !gta02_speaker_enabled); + + return 0; +} + +static int lm4853_get_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = gta02_speaker_enabled; + return 0; +} + +static int lm4853_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(event)); + + return 0; +} + +static const struct snd_soc_dapm_route neo1973_gta02_routes[] = { + /* Connections to the amp */ + {"Stereo Out", NULL, "LOUT1"}, + {"Stereo Out", NULL, "ROUT1"}, + + /* Call Speaker */ + {"Handset Spk", NULL, "LOUT2"}, + {"Handset Spk", NULL, "ROUT2"}, +}; + +static const struct snd_kcontrol_new neo1973_gta02_wm8753_controls[] = { + SOC_DAPM_PIN_SWITCH("Handset Spk"), + SOC_DAPM_PIN_SWITCH("Stereo Out"), + + SOC_SINGLE_BOOL_EXT("Amp Spk Switch", 0, + lm4853_get_spk, + lm4853_set_spk), +}; + +static const struct snd_soc_dapm_widget neo1973_gta02_wm8753_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Handset Spk", NULL), + SND_SOC_DAPM_SPK("Stereo Out", lm4853_event), +}; + +static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + ret = snd_soc_dapm_new_controls(dapm, neo1973_gta02_wm8753_dapm_widgets, + ARRAY_SIZE(neo1973_gta02_wm8753_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, neo1973_gta02_routes, + ARRAY_SIZE(neo1973_gta02_routes)); + if (ret) + return ret; + + ret = snd_soc_add_controls(codec, neo1973_gta02_wm8753_controls, + ARRAY_SIZE(neo1973_gta02_wm8753_controls)); + if (ret) + return ret; + + snd_soc_dapm_disable_pin(dapm, "Stereo Out"); + snd_soc_dapm_disable_pin(dapm, "Handset Spk"); + snd_soc_dapm_ignore_suspend(dapm, "Stereo Out"); + snd_soc_dapm_ignore_suspend(dapm, "Handset Spk"); + + return 0; +} + +#else +static int neo1973_gta02_wm8753_init(struct snd_soc_code *codec) { return 0; } +#endif -/* - * This is an example machine initialisation for a wm8753 connected to a - * neo1973 II. It is missing logic to detect hp/mic insertions and logic - * to re-route the audio in such an event. - */ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int err; - - pr_debug("Entered %s\n", __func__); + int ret; /* set up NC codec pins */ - snd_soc_dapm_nc_pin(dapm, "LOUT2"); - snd_soc_dapm_nc_pin(dapm, "ROUT2"); + if (machine_is_neo1973_gta01()) { + snd_soc_dapm_nc_pin(dapm, "LOUT2"); + snd_soc_dapm_nc_pin(dapm, "ROUT2"); + } snd_soc_dapm_nc_pin(dapm, "OUT3"); snd_soc_dapm_nc_pin(dapm, "OUT4"); snd_soc_dapm_nc_pin(dapm, "LINE1"); snd_soc_dapm_nc_pin(dapm, "LINE2"); /* Add neo1973 specific widgets */ - snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets, - ARRAY_SIZE(wm8753_dapm_widgets)); - - /* set endpoints to default mode */ - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); + ret = snd_soc_dapm_new_controls(dapm, neo1973_wm8753_dapm_widgets, + ARRAY_SIZE(neo1973_wm8753_dapm_widgets)); + if (ret) + return ret; /* add neo1973 specific controls */ - err = snd_soc_add_controls(codec, neo1973_wm8753_controls, - ARRAY_SIZE(neo1973_wm8753_controls)); - if (err < 0) - return err; + ret = snd_soc_add_controls(codec, neo1973_wm8753_controls, + ARRAY_SIZE(neo1973_wm8753_controls)); + if (ret) + return ret; /* set up neo1973 specific audio routes */ - err = snd_soc_dapm_add_routes(dapm, neo1973_wm8753_routes, + ret = snd_soc_dapm_add_routes(dapm, neo1973_wm8753_routes, ARRAY_SIZE(neo1973_wm8753_routes)); + if (ret) + return ret; + + /* set endpoints to default off mode */ + snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); + snd_soc_dapm_disable_pin(dapm, "GSM Line In"); + snd_soc_dapm_disable_pin(dapm, "Headset Mic"); + snd_soc_dapm_disable_pin(dapm, "Handset Mic"); + + /* allow audio paths from the GSM modem to run during suspend */ + snd_soc_dapm_ignore_suspend(dapm, "GSM Line Out"); + snd_soc_dapm_ignore_suspend(dapm, "GSM Line In"); + snd_soc_dapm_ignore_suspend(dapm, "Headset Mic"); + snd_soc_dapm_ignore_suspend(dapm, "Handset Mic"); + + if (machine_is_neo1973_gta02()) { + ret = neo1973_gta02_wm8753_init(codec); + if (ret) + return ret; + } snd_soc_dapm_sync(dapm); + return 0; } +/* GTA01 specific controlls */ + +#ifdef CONFIG_MACH_NEO1973_GTA01 + static const struct snd_soc_dapm_route neo1973_lm4857_routes[] = { {"Amp IN", NULL, "ROUT1"}, {"Amp IN", NULL, "LOUT1"}, @@ -328,10 +414,14 @@ static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) return 0; } +#else +static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) { return 0; }; +#endif + /* * BT Codec DAI */ -static struct snd_soc_dai bt_dai = { +static struct snd_soc_dai_driver bt_dai = { .name = "bluetooth-dai", .playback = { .channels_min = 1, @@ -382,6 +472,15 @@ static struct snd_soc_codec_conf neo1973_codec_conf[] = { }, }; +#ifdef CONFIG_MACH_NEO1973_GTA02 +static const struct gpio neo1973_gta02_gpios[] = { + { GTA02_GPIO_HP_IN, GPIOF_OUT_INIT_HIGH, "GTA02_HP_IN" }, + { GTA02_GPIO_AMP_SHUT, GPIOF_OUT_INIT_HIGH, "GTA02_AMP_SHUT" }, +}; +#else +static const struct gpio neo1973_gta02_gpios[] = {}; +#endif + static struct snd_soc_card neo1973 = { .name = "neo1973", .dai_link = neo1973_dai, @@ -398,43 +497,64 @@ static int __init neo1973_init(void) { int ret; - pr_debug("Entered %s\n", __func__); - - if (!machine_is_neo1973_gta01()) { - printk(KERN_INFO - "Only GTA01 hardware supported by ASoC driver\n"); + if (!machine_is_neo1973_gta01() && !machine_is_neo1973_gta02()) return -ENODEV; + + if (machine_is_neo1973_gta02()) { + neo1973.name = "neo1973gta02"; + neo1973.num_aux_devs = 0; + + ret = gpio_request_array(neo1973_gta02_gpios, + ARRAY_SIZE(neo1973_gta02_gpios)); + if (ret) + return ret; } neo1973_snd_device = platform_device_alloc("soc-audio", -1); - if (!neo1973_snd_device) - return -ENOMEM; + if (!neo1973_snd_device) { + ret = -ENOMEM; + goto err_gpio_free; + } + + /* register bluetooth DAI here */ + ret = snd_soc_register_dai(&neo1973_snd_device->dev, &bt_dai); + if (ret) + goto err_put_device; platform_set_drvdata(neo1973_snd_device, &neo1973); ret = platform_device_add(neo1973_snd_device); - if (ret) { - platform_device_put(neo1973_snd_device); - return ret; - } + if (ret) + goto err_unregister_dai; - if (ret != 0) - platform_device_unregister(neo1973_snd_device); + return 0; +err_unregister_dai: + snd_soc_unregister_dai(&neo1973_snd_device->dev); +err_put_device: + platform_device_put(neo1973_snd_device); +err_gpio_free: + if (machine_is_neo1973_gta02()) { + gpio_free_array(neo1973_gta02_gpios, + ARRAY_SIZE(neo1973_gta02_gpios)); + } return ret; } +module_init(neo1973_init); static void __exit neo1973_exit(void) { - pr_debug("Entered %s\n", __func__); - + snd_soc_unregister_dai(&neo1973_snd_device->dev); platform_device_unregister(neo1973_snd_device); -} -module_init(neo1973_init); + if (machine_is_neo1973_gta02()) { + gpio_free_array(neo1973_gta02_gpios, + ARRAY_SIZE(neo1973_gta02_gpios)); + } +} module_exit(neo1973_exit); /* Module information */ MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org, www.openmoko.org"); -MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973"); +MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 and Frerunner"); MODULE_LICENSE("GPL"); -- cgit v1.2.3 From a077ff9034897232ab4208f55880221390bd6877 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 7 Mar 2011 08:04:59 +0100 Subject: ASoC: Add driver for the dfbmcs320 bluetooth module This patch adds a codec driver for the dfbmcs320 bluetooth module, which is used on the neo1973 boards. The patch also modifies the neo1937_wm8753 sound board driver to use the new driver instead of registering the bluetooth DAI manually. Previously there was a name mismatch between the bluetooth DAI and the bluetooth DAI link and the sound card was not instantiated, with this patch the issue is no longer present and sound support works again. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 5 ++- sound/soc/codecs/Makefile | 2 ++ sound/soc/codecs/dfbmcs320.c | 72 ++++++++++++++++++++++++++++++++++++++ sound/soc/samsung/Kconfig | 1 + sound/soc/samsung/neo1973_wm8753.c | 35 ++++-------------- 5 files changed, 86 insertions(+), 29 deletions(-) create mode 100644 sound/soc/codecs/dfbmcs320.c (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 37035e6984b..212859cf19a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -29,6 +29,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C + select SND_SOC_DFBMCS320 select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C @@ -175,6 +176,9 @@ config SND_SOC_L3 config SND_SOC_DA7210 tristate +config SND_SOC_DFBMCS320 + tristate + config SND_SOC_DMIC tristate @@ -361,4 +365,3 @@ config SND_SOC_WM2000 config SND_SOC_WM9090 tristate - diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 0663d22e86b..ebb059c0eef 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o +snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o @@ -101,6 +102,7 @@ obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o +obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o diff --git a/sound/soc/codecs/dfbmcs320.c b/sound/soc/codecs/dfbmcs320.c new file mode 100644 index 00000000000..704bbde6573 --- /dev/null +++ b/sound/soc/codecs/dfbmcs320.c @@ -0,0 +1,72 @@ +/* + * Driver for the DFBM-CS320 bluetooth module + * Copyright 2011 Lars-Peter Clausen + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include +#include +#include + +#include + +static struct snd_soc_dai_driver dfbmcs320_dai = { + .name = "dfbmcs320-pcm", + .playback = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_dfbmcs320; + +static int __devinit dfbmcs320_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_dfbmcs320, + &dfbmcs320_dai, 1); +} + +static int __devexit dfbmcs320_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + + return 0; +} + +static struct platform_driver dfmcs320_driver = { + .driver = { + .name = "dfbmcs320", + .owner = THIS_MODULE, + }, + .probe = dfbmcs320_probe, + .remove = __devexit_p(dfbmcs320_remove), +}; + +static int __init dfbmcs320_init(void) +{ + return platform_driver_register(&dfmcs320_driver); +} +module_init(dfbmcs320_init); + +static void __exit dfbmcs320_exit(void) +{ + platform_driver_unregister(&dfmcs320_driver); +} +module_exit(dfbmcs320_exit); + +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_DESCRIPTION("ASoC DFBM-CS320 bluethooth module driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index c3014e82157..a08237acc53 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -40,6 +40,7 @@ config SND_SOC_SAMSUNG_NEO1973_WM8753 select SND_S3C24XX_I2S select SND_SOC_WM8753 select SND_SOC_LM4857 if MACH_NEO1973_GTA01 + select SND_SOC_DFBMCS320 help Say Y here to enable audio support for the Openmoko Neo1973 Smartphones. diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 37cfbb8ca39..78bfdb3f5d7 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -418,23 +418,6 @@ static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) { return 0; }; #endif -/* - * BT Codec DAI - */ -static struct snd_soc_dai_driver bt_dai = { - .name = "bluetooth-dai", - .playback = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, -}; - static struct snd_soc_dai_link neo1973_dai[] = { { /* Hifi Playback - for similatious use with voice below */ .name = "WM8753", @@ -450,7 +433,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { .name = "Bluetooth", .stream_name = "Voice", .platform_name = "samsung-audio", - .cpu_dai_name = "bluetooth-dai", + .cpu_dai_name = "dfbmcs320-pcm", .codec_dai_name = "wm8753-voice", .codec_name = "wm8753-codec.0-001a", .ops = &neo1973_voice_ops, @@ -458,6 +441,10 @@ static struct snd_soc_dai_link neo1973_dai[] = { }; static struct snd_soc_aux_dev neo1973_aux_devs[] = { + { + .name = "dfbmcs320", + .codec_name = "dfbmcs320.0", + }, { .name = "lm4857", .codec_name = "lm4857.0-007c", @@ -502,7 +489,7 @@ static int __init neo1973_init(void) if (machine_is_neo1973_gta02()) { neo1973.name = "neo1973gta02"; - neo1973.num_aux_devs = 0; + neo1973.num_aux_devs = 1; ret = gpio_request_array(neo1973_gta02_gpios, ARRAY_SIZE(neo1973_gta02_gpios)); @@ -516,21 +503,14 @@ static int __init neo1973_init(void) goto err_gpio_free; } - /* register bluetooth DAI here */ - ret = snd_soc_register_dai(&neo1973_snd_device->dev, &bt_dai); - if (ret) - goto err_put_device; - platform_set_drvdata(neo1973_snd_device, &neo1973); ret = platform_device_add(neo1973_snd_device); if (ret) - goto err_unregister_dai; + goto err_put_device; return 0; -err_unregister_dai: - snd_soc_unregister_dai(&neo1973_snd_device->dev); err_put_device: platform_device_put(neo1973_snd_device); err_gpio_free: @@ -544,7 +524,6 @@ module_init(neo1973_init); static void __exit neo1973_exit(void) { - snd_soc_unregister_dai(&neo1973_snd_device->dev); platform_device_unregister(neo1973_snd_device); if (machine_is_neo1973_gta02()) { -- cgit v1.2.3 From 9b74c7d6ba8b95b4feb8c17b2b38d72eb4240ea4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Mar 2011 12:23:10 +0000 Subject: ASoC: Add LM4857 to SND_SOC_ALL_CODECS Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 212859cf19a..10f11dd98c2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -31,6 +31,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7210 if I2C select SND_SOC_DFBMCS320 select SND_SOC_JZ4740_CODEC if SOC_JZ4740 + select SND_SOC_LM4857 if I2C select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 -- cgit v1.2.3 From 383f8465d1579635d7b7df5d7a1d1966b6ffb868 Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin Date: Mon, 7 Mar 2011 20:29:36 +0300 Subject: ASoC: Extend range of supported sample rates for CS4271 CODEC. Extend range of supported sample rates for CS4271 CODEC. Signed-off-by: Alexander Sverdlin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 41a3ee6f614..538e814136d 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -33,6 +33,7 @@ #define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +#define CS4271_PCM_RATES SNDRV_PCM_RATE_8000_192000 /* * CS4271 registers @@ -392,14 +393,14 @@ static struct snd_soc_dai_driver cs4271_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = CS4271_PCM_RATES, .formats = CS4271_PCM_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = CS4271_PCM_RATES, .formats = CS4271_PCM_FORMATS, }, .ops = &cs4271_dai_ops, -- cgit v1.2.3 From f9c540456ab662a3b5d79c1fa93c51305a97fdf3 Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin Date: Mon, 7 Mar 2011 20:29:58 +0300 Subject: ASoC: Remove warnings in ep93xx-i2s.c Remove warnings in ep93xx-i2s.c Signed-off-by: Alexander Sverdlin Acked-by: Ryan Mallon Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/ep93xx/ep93xx-i2s.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index fff579a1c13..4d3c1389d3d 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -314,10 +314,12 @@ static int ep93xx_i2s_suspend(struct snd_soc_dai *dai) struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); if (!dai->active) - return; + return 0; ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_PLAYBACK); ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_CAPTURE); + + return 0; } static int ep93xx_i2s_resume(struct snd_soc_dai *dai) @@ -325,10 +327,12 @@ static int ep93xx_i2s_resume(struct snd_soc_dai *dai) struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); if (!dai->active) - return; + return 0; ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_PLAYBACK); ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_CAPTURE); + + return 0; } #else #define ep93xx_i2s_suspend NULL -- cgit v1.2.3 From 89b95ac09e408b5d88a8b3792dc76c863e45fb31 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Mar 2011 16:38:44 +0000 Subject: ASoC: Add DAPM widget and path data to CODEC driver structure Allow a slight simplification of CODEC drivers by allowing DAPM routes and widgets to be provided in a table. They will be instantiated at the end of CODEC probe. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index be34f6b9538..c12f2bd23a4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1464,6 +1464,7 @@ static int soc_probe_codec(struct snd_soc_card *card, struct snd_soc_codec *codec) { int ret = 0; + const struct snd_soc_codec_driver *driver = codec->driver; codec->card = card; codec->dapm.card = card; @@ -1472,8 +1473,8 @@ static int soc_probe_codec(struct snd_soc_card *card, if (!try_module_get(codec->dev->driver->owner)) return -ENODEV; - if (codec->driver->probe) { - ret = codec->driver->probe(codec); + if (driver->probe) { + ret = driver->probe(codec); if (ret < 0) { dev_err(codec->dev, "asoc: failed to probe CODEC %s: %d\n", @@ -1482,6 +1483,13 @@ static int soc_probe_codec(struct snd_soc_card *card, } } + if (driver->dapm_widgets) + snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, + driver->num_dapm_widgets); + if (driver->dapm_routes) + snd_soc_dapm_add_routes(&codec->dapm, driver->dapm_routes, + driver->num_dapm_routes); + soc_init_codec_debugfs(codec); /* mark codec as probed and add to card codec list */ -- cgit v1.2.3 From 149c7b441bc66b37325f0c494bfe9f303484bd88 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Mar 2011 16:39:50 +0000 Subject: ASoC: Use data based init for WM9081 DAPM Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm9081.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 7883f3ed797..8b1b2c9ab59 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -762,7 +762,7 @@ SND_SOC_DAPM_SUPPLY("TOCLK", WM9081_CLOCK_CONTROL_3, 2, 0, NULL, 0), }; -static const struct snd_soc_dapm_route audio_paths[] = { +static const struct snd_soc_dapm_route wm9081_audio_paths[] = { { "DAC", NULL, "CLK_SYS" }, { "DAC", NULL, "CLK_DSP" }, @@ -1232,7 +1232,6 @@ static struct snd_soc_dai_driver wm9081_dai = { static int wm9081_probe(struct snd_soc_codec *codec) { struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; u16 reg; @@ -1282,10 +1281,6 @@ static int wm9081_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm9081_eq_controls)); } - snd_soc_dapm_new_controls(dapm, wm9081_dapm_widgets, - ARRAY_SIZE(wm9081_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths)); - return ret; } @@ -1334,6 +1329,10 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9081 = { .reg_word_size = sizeof(u16), .reg_cache_default = wm9081_reg_defaults, .volatile_register = wm9081_volatile_register, + .dapm_widgets = wm9081_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm9081_dapm_widgets), + .dapm_routes = wm9081_audio_paths, + .num_dapm_routes = ARRAY_SIZE(wm9081_audio_paths), }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -- cgit v1.2.3 From b993f92b99288d4b3a1a1237f3e40fa6460e4b47 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Mar 2011 16:42:20 +0000 Subject: ASoC: Fix section mismatch warnings in WM8994 Annoying as the __devinitdata is actually correct. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994-tables.c | 2 +- sound/soc/codecs/wm8994.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c index 61dfd91c6c7..a87adbd05ee 100644 --- a/sound/soc/codecs/wm8994-tables.c +++ b/sound/soc/codecs/wm8994-tables.c @@ -1573,7 +1573,7 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x03C3, 0x03C3 }, /* R1569 - Sidetone */ }; -const __devinitdata u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { +const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x8994, /* R0 - Software Reset */ 0x0000, /* R1 - Power Management (1) */ 0x6000, /* R2 - Power Management (2) */ diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 0c355bfc88f..999b8851226 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -43,6 +43,6 @@ struct wm8994_access_mask { }; extern const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE]; -extern const __devinitdata u16 wm8994_reg_defaults[WM8994_CACHE_SIZE]; +extern const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE]; #endif -- cgit v1.2.3 From ec4ee52a8f5fb5b8e235ae9f02589d60d54740cc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Mar 2011 20:58:11 +0000 Subject: ASoC: Provide CODEC clocking operations and API calls When multi component systems use DAIless amplifiers which require clocking configuration it is at best hard to use the current clocking API as this requires a DAI even though the device may not even have one. Address this by adding set_sysclk() and set_pll() operations and APIs for CODECs. In order to avoid issues with devices which could be used either with or without DAIs make the DAI variants call through to their CODEC counterparts if there is no DAI specific operation. Converting over entirely would create problems for multi-DAI devices which offer per-DAI clocking setup. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c12f2bd23a4..c2ec6cb0563 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3064,11 +3064,33 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, { if (dai->driver && dai->driver->ops->set_sysclk) return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir); + else if (dai->codec && dai->codec->driver->set_sysclk) + return dai->codec->driver->set_sysclk(dai->codec, clk_id, + freq, dir); else return -EINVAL; } EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); +/** + * snd_soc_codec_set_sysclk - configure CODEC system or master clock. + * @codec: CODEC + * @clk_id: DAI specific clock ID + * @freq: new clock frequency in Hz + * @dir: new clock direction - input/output. + * + * Configures the CODEC master (MCLK) or system (SYSCLK) clocking. + */ +int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, + unsigned int freq, int dir) +{ + if (codec->driver->set_sysclk) + return codec->driver->set_sysclk(codec, clk_id, freq, dir); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_sysclk); + /** * snd_soc_dai_set_clkdiv - configure DAI clock dividers. * @dai: DAI @@ -3105,11 +3127,35 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, if (dai->driver && dai->driver->ops->set_pll) return dai->driver->ops->set_pll(dai, pll_id, source, freq_in, freq_out); + else if (dai->codec && dai->codec->driver->set_pll) + return dai->codec->driver->set_pll(dai->codec, pll_id, source, + freq_in, freq_out); else return -EINVAL; } EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); +/* + * snd_soc_codec_set_pll - configure codec PLL. + * @codec: CODEC + * @pll_id: DAI specific PLL ID + * @source: DAI specific source for the PLL + * @freq_in: PLL input clock frequency in Hz + * @freq_out: requested PLL output clock frequency in Hz + * + * Configures and enables PLL to generate output clock based on input clock. + */ +int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + if (codec->driver->set_pll) + return codec->driver->set_pll(codec, pll_id, source, + freq_in, freq_out); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); + /** * snd_soc_dai_set_fmt - configure DAI hardware audio format. * @dai: DAI -- cgit v1.2.3 From 63d24b79b66ea5d5b04dadb9e92f31c0141948fb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 7 Mar 2011 20:59:45 +0000 Subject: ASoC: Convert WM9081 SYSCLK configuration to be device wide Also respace the CODEC ops a bit for legibility. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm9081.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 8b1b2c9ab59..effaf75cacc 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1140,10 +1140,9 @@ static int wm9081_digital_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } -static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai, +static int wm9081_set_sysclk(struct snd_soc_codec *codec, int clk_id, unsigned int freq, int dir) { - struct snd_soc_codec *codec = codec_dai->codec; struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); switch (clk_id) { @@ -1208,7 +1207,6 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, static struct snd_soc_dai_ops wm9081_dai_ops = { .hw_params = wm9081_hw_params, - .set_sysclk = wm9081_set_sysclk, .set_fmt = wm9081_set_dai_fmt, .digital_mute = wm9081_digital_mute, .set_tdm_slot = wm9081_set_tdm_slot, @@ -1324,11 +1322,15 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9081 = { .remove = wm9081_remove, .suspend = wm9081_suspend, .resume = wm9081_resume, + + .set_sysclk = wm9081_set_sysclk, .set_bias_level = wm9081_set_bias_level, + .reg_cache_size = ARRAY_SIZE(wm9081_reg_defaults), .reg_word_size = sizeof(u16), .reg_cache_default = wm9081_reg_defaults, .volatile_register = wm9081_volatile_register, + .dapm_widgets = wm9081_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm9081_dapm_widgets), .dapm_routes = wm9081_audio_paths, -- cgit v1.2.3 From 4f333b20aef4459a24590fa8545e974f8f137f62 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 8 Mar 2011 00:11:15 +0000 Subject: ASoC: Refactor symmetric_rates check to reduce indentation Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c2ec6cb0563..65ff61211a8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -500,20 +500,20 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - if (codec_dai->driver->symmetric_rates || cpu_dai->driver->symmetric_rates || - rtd->dai_link->symmetric_rates) { - dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", - rtd->rate); - - ret = snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - rtd->rate, - rtd->rate); - if (ret < 0) { - dev_err(&rtd->dev, - "Unable to apply rate symmetry constraint: %d\n", ret); - return ret; - } + if (!codec_dai->driver->symmetric_rates && + !cpu_dai->driver->symmetric_rates && + !rtd->dai_link->symmetric_rates) + return 0; + + dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); + + ret = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + rtd->rate, rtd->rate); + if (ret < 0) { + dev_err(&rtd->dev, + "Unable to apply rate symmetry constraint: %d\n", ret); + return ret; } return 0; -- cgit v1.2.3 From c4ef87867b42bd1fa7d6dacaa28bf07cf741a724 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 8 Mar 2011 00:17:56 +0000 Subject: ASoC: Warn rather than set a silly constraint when we can't do symmetry Symmetric rate configuration can fail if the second stream starting tries to apply the symmetric constraint before the first stream has got far enough to pick a rate. Rather than try to enforce a nonsensical rate of 0Hz log a warning and allow the application to carry on. Things might go wrong later on but the user will know about it and there's unlikely to be lasting damage. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 65ff61211a8..db3075dd11f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -505,6 +505,16 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) !rtd->dai_link->symmetric_rates) return 0; + /* This can happen if multiple streams are starting simultaneously - + * the second can need to get its constraints before the first has + * picked a rate. Complain and allow the application to carry on. + */ + if (!rtd->rate) { + dev_warn(&rtd->dev, + "Not enforcing symmetric_rates due to race\n"); + return 0; + } + dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); ret = snd_pcm_hw_constraint_minmax(substream->runtime, -- cgit v1.2.3 From efb7ac3f9c28fcb379c51f987b63174f727b7453 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 8 Mar 2011 17:23:24 +0000 Subject: ASoC: Fix prefixing of DAPM controls by factoring prefix into snd_soc_cnew() Currently will ignore prefixes when creating DAPM controls. Since currently all control creation goes through snd_soc_cnew() we can fix this by factoring the prefixing into that function. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 45 +++++++++++++++++++++++++++++++-------------- sound/soc/soc-dapm.c | 16 ++++++++++++++-- 2 files changed, 45 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index db3075dd11f..17efacdb248 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2344,22 +2344,45 @@ EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); * @_template: control template * @data: control private data * @long_name: control long name + * @prefix: control name prefix * * Create a new mixer control from a template control. * * Returns 0 for success, else error. */ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, - void *data, char *long_name) + void *data, char *long_name, + const char *prefix) { struct snd_kcontrol_new template; + struct snd_kcontrol *kcontrol; + char *name = NULL; + int name_len; memcpy(&template, _template, sizeof(template)); - if (long_name) - template.name = long_name; template.index = 0; - return snd_ctl_new1(&template, data); + if (!long_name) + long_name = template.name; + + if (prefix) { + name_len = strlen(long_name) + strlen(prefix) + 2; + name = kmalloc(name_len, GFP_ATOMIC); + if (!name) + return NULL; + + snprintf(name, name_len, "%s %s", prefix, long_name); + + template.name = name; + } else { + template.name = long_name; + } + + kcontrol = snd_ctl_new1(&template, data); + + kfree(name); + + return kcontrol; } EXPORT_SYMBOL_GPL(snd_soc_cnew); @@ -2378,22 +2401,16 @@ int snd_soc_add_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls) { struct snd_card *card = codec->card->snd_card; - char prefixed_name[44], *name; int err, i; for (i = 0; i < num_controls; i++) { const struct snd_kcontrol_new *control = &controls[i]; - if (codec->name_prefix) { - snprintf(prefixed_name, sizeof(prefixed_name), "%s %s", - codec->name_prefix, control->name); - name = prefixed_name; - } else { - name = control->name; - } - err = snd_ctl_add(card, snd_soc_cnew(control, codec, name)); + err = snd_ctl_add(card, snd_soc_cnew(control, codec, + control->name, + codec->name_prefix)); if (err < 0) { dev_err(codec->dev, "%s: Failed to add %s: %d\n", - codec->name, name, err); + codec->name, control->name, err); return err; } } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 570db8819d9..a6fb85d4641 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -369,6 +369,12 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, size_t name_len; struct snd_soc_dapm_path *path; struct snd_card *card = dapm->card->snd_card; + const char *prefix; + + if (dapm->codec) + prefix = dapm->codec->name_prefix; + else + prefix = NULL; /* add kcontrol */ for (i = 0; i < w->num_kcontrols; i++) { @@ -409,7 +415,7 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, path->long_name[name_len - 1] = '\0'; path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w, - path->long_name); + path->long_name, prefix); ret = snd_ctl_add(card, path->kcontrol); if (ret < 0) { dev_err(dapm->dev, @@ -431,6 +437,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_path *path = NULL; struct snd_kcontrol *kcontrol; struct snd_card *card = dapm->card->snd_card; + const char *prefix; int ret = 0; if (!w->num_kcontrols) { @@ -438,7 +445,12 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, return -EINVAL; } - kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name); + if (dapm->codec) + prefix = dapm->codec->name_prefix; + else + prefix = NULL; + + kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name, prefix); ret = snd_ctl_add(card, kcontrol); if (ret < 0) -- cgit v1.2.3 From 62f75aafdf180554b4fad29ff1f3827b151d39db Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Tue, 8 Mar 2011 14:39:24 +0300 Subject: ASoC: sgtl5000: use after free in ldo_regulator_register() The "ldo" variable was dereferenced after free on the error path. Signed-off-by: Dan Carpenter Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index b7e97c02689..1f7217f703e 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -875,11 +875,13 @@ static int ldo_regulator_register(struct snd_soc_codec *codec, ldo->dev = regulator_register(&ldo->desc, codec->dev, init_data, ldo); if (IS_ERR(ldo->dev)) { + int ret = PTR_ERR(ldo->dev); + dev_err(codec->dev, "failed to register regulator\n"); kfree(ldo->desc.name); kfree(ldo); - return PTR_ERR(ldo->dev); + return ret; } return 0; -- cgit v1.2.3 From fa4d1f595a89234647b34a7f507436be873091bc Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 9 Mar 2011 11:25:00 +0200 Subject: ASoC: omap: rx51: Enable McBSP2 sidetone McBSP sidetone is needed in telephony applications. McBSP sidetone is a configurable FIR filter that forms a loopback from McBSP input to output. This patch enables the McBSP2 sidetone ALSA controls so that it can be used on Nokia RX-51/N900. Sidetone feature can be tested with following commands: (set up codec input and output paths) # Enable and configure sidetone amixer -D hw:0 set 'McBSP2 Sidetone' on amixer set -D hw:0 'McBSP2 Sidetone Channel 0' 32767 echo 32767 >/sys/devices/platform/omap-mcbsp.2/st_taps # Do not loop audio via CPU arecord -f dat >/dev/null |aplay /dev/zero Signed-off-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/rx51.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 055e447b2ed..d0986220eff 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -331,6 +331,10 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) return err; snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42); + err = omap_mcbsp_st_add_controls(codec, 1); + if (err < 0) + return err; + snd_soc_dapm_sync(dapm); /* AV jack detection */ -- cgit v1.2.3 From 378a90f4540cc113e6ef36861ae914b0c63700a0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 8 Mar 2011 18:52:08 +0000 Subject: ASoC: Simplify WM9081 speaker startup by using widgets for spaker output Now we have a register write minimisation code in DAPM we don't need to worry about the ordering of the enable and disable of the PGA and the output stage. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm9081.c | 32 ++++++-------------------------- 1 file changed, 6 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index effaf75cacc..55cdf298202 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -389,27 +389,6 @@ SOC_DAPM_SINGLE("IN2 Switch", WM9081_ANALOGUE_MIXER, 2, 1, 0), SOC_DAPM_SINGLE("Playback Switch", WM9081_ANALOGUE_MIXER, 4, 1, 0), }; -static int speaker_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - unsigned int reg = snd_soc_read(codec, WM9081_POWER_MANAGEMENT); - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - reg |= WM9081_SPK_ENA; - break; - - case SND_SOC_DAPM_PRE_PMD: - reg &= ~WM9081_SPK_ENA; - break; - } - - snd_soc_write(codec, WM9081_POWER_MANAGEMENT, reg); - - return 0; -} - struct _fll_div { u16 fll_fratio; u16 fll_outdiv; @@ -747,9 +726,8 @@ SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0, SND_SOC_DAPM_PGA("LINEOUT PGA", WM9081_POWER_MANAGEMENT, 4, 0, NULL, 0), -SND_SOC_DAPM_PGA_E("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0, - speaker_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("LINEOUT"), SND_SOC_DAPM_OUTPUT("SPKN"), @@ -780,8 +758,10 @@ static const struct snd_soc_dapm_route wm9081_audio_paths[] = { { "Speaker PGA", NULL, "TOCLK" }, { "Speaker PGA", NULL, "CLK_SYS" }, - { "SPKN", NULL, "Speaker PGA" }, - { "SPKP", NULL, "Speaker PGA" }, + { "Speaker", NULL, "Speaker PGA" }, + + { "SPKN", NULL, "Speaker" }, + { "SPKP", NULL, "Speaker" }, }; static int wm9081_set_bias_level(struct snd_soc_codec *codec, -- cgit v1.2.3 From 823dba5191220fc94b83dc0b3f2178ff0842e294 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 2 Mar 2011 11:01:18 +0000 Subject: ASoC: Fix broken bitfield definitions in WM8978 Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8978.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 4bbc3442703..8dfb0a0da67 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -145,18 +145,18 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { SOC_SINGLE("DAC Playback Limiter Threshold", WM8978_DAC_LIMITER_2, 4, 7, 0), SOC_SINGLE("DAC Playback Limiter Boost", - WM8978_DAC_LIMITER_2, 0, 15, 0), + WM8978_DAC_LIMITER_2, 0, 12, 0), SOC_ENUM("ALC Enable Switch", alc1), SOC_SINGLE("ALC Capture Min Gain", WM8978_ALC_CONTROL_1, 0, 7, 0), SOC_SINGLE("ALC Capture Max Gain", WM8978_ALC_CONTROL_1, 3, 7, 0), - SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 7, 0), + SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 10, 0), SOC_SINGLE("ALC Capture Target", WM8978_ALC_CONTROL_2, 0, 15, 0), SOC_ENUM("ALC Capture Mode", alc3), - SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 15, 0), - SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 15, 0), + SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 10, 0), + SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 10, 0), SOC_SINGLE("ALC Capture Noise Gate Switch", WM8978_NOISE_GATE, 3, 1, 0), SOC_SINGLE("ALC Capture Noise Gate Threshold", @@ -211,8 +211,10 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1), /* DAC / ADC oversampling */ - SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, 8, 1, 0), - SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, 8, 1, 0), + SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, + 5, 1, 0), + SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, + 5, 1, 0), }; /* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */ -- cgit v1.2.3 From 28e868081086c495c897a48c50d2d5187ef677d2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 8 Mar 2011 19:29:53 +0000 Subject: ASoC: Use the correct DAPM context when cleaning up final widget set Now we've got multi-component we need to make sure that the DAPM context (and hence register I/O context) we use to apply the pending updates at the end of a DAPM sequence is the one we were processing rather than the one that was used to initate the state change. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 25e54230cc6..1790f83ee66 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -941,7 +941,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, } if (!list_empty(&pending)) - dapm_seq_run_coalesced(dapm, &pending); + dapm_seq_run_coalesced(cur_dapm, &pending); } static void dapm_widget_update(struct snd_soc_dapm_context *dapm) -- cgit v1.2.3 From 3e5ff4dfa5a0a5627235a245665035597f050d86 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Mar 2011 11:33:09 +0000 Subject: ASoC: Fix double addition of prefixes due to widget prefixing We're not only prefixing all controls, we're also prefixing the widget names in the runtime data. This causes us to add the prefix twice - once when using the widget name to generate the control name and once when adding the control. Really we shouldn't be prefixing the widget names at all, the matching code should be handing this as we always know which DAPM context a widget came from and always display the widget name in terms of a DAPM context. However, we're quite close to the merge window and that's relatively invasive. Signed-off-by: Mark Brown Reported-by: Jarkko Nikula Acked-by: Jarkko Nikula Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 29 ++++++++++++++++++++++++++--- 1 file changed, 26 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a6fb85d4641..e981a8339f8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -366,7 +366,7 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_widget *w) { int i, ret = 0; - size_t name_len; + size_t name_len, prefix_len; struct snd_soc_dapm_path *path; struct snd_card *card = dapm->card->snd_card; const char *prefix; @@ -376,6 +376,11 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, else prefix = NULL; + if (prefix) + prefix_len = strlen(prefix) + 1; + else + prefix_len = 0; + /* add kcontrol */ for (i = 0; i < w->num_kcontrols; i++) { @@ -403,8 +408,15 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, switch (w->id) { default: + /* The control will get a prefix from + * the control creation process but + * we're also using the same prefix + * for widgets so cut the prefix off + * the front of the widget name. + */ snprintf(path->long_name, name_len, "%s %s", - w->name, w->kcontrols[i].name); + w->name + prefix_len, + w->kcontrols[i].name); break; case snd_soc_dapm_mixer_named_ctl: snprintf(path->long_name, name_len, "%s", @@ -438,6 +450,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, struct snd_kcontrol *kcontrol; struct snd_card *card = dapm->card->snd_card; const char *prefix; + size_t prefix_len; int ret = 0; if (!w->num_kcontrols) { @@ -450,7 +463,17 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, else prefix = NULL; - kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name, prefix); + if (prefix) + prefix_len = strlen(prefix) + 1; + else + prefix_len = 0; + + /* The control will get a prefix from the control creation + * process but we're also using the same prefix for widgets so + * cut the prefix off the front of the widget name. + */ + kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name + prefix_len, + prefix); ret = snd_ctl_add(card, kcontrol); if (ret < 0) -- cgit v1.2.3 From 5c3a12e96c9f3158602b2feac8583ae35c10b80e Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin Date: Mon, 7 Mar 2011 20:29:45 +0300 Subject: ASoC: Manage mode and rate bits correctly for CS4271 CODEC. Manage mode and rate bits correctly, according to datasheet in CS4271 CODEC. This is done to make capture work properly. Signed-off-by: Alexander Sverdlin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 83 ++++++++++++++++++++++++++++++----------------- 1 file changed, 53 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 538e814136d..083aab96ca8 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -168,27 +168,6 @@ struct cs4271_private { int gpio_disable; }; -struct cs4271_clk_cfg { - unsigned int ratio; /* MCLK / sample rate */ - u8 speed_mode; /* codec speed mode: 1x, 2x, 4x */ - u8 mclk_master; /* ratio bit mask for Master mode */ - u8 mclk_slave; /* ratio bit mask for Slave mode */ -}; - -static struct cs4271_clk_cfg cs4271_clk_tab[] = { - {64, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, - {96, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, - {128, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, - {192, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, - {256, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, - {384, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, - {512, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_2, CS4271_MODE1_DIV_1}, - {768, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3}, - {1024, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3} -}; - -#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab) - /* * @freq is the desired MCLK rate * MCLK rate should (c) be the sample rate, multiplied by one of the @@ -297,6 +276,45 @@ static int cs4271_put_deemph(struct snd_kcontrol *kcontrol, return cs4271_set_deemph(codec); } +struct cs4271_clk_cfg { + bool master; /* codec mode */ + u8 speed_mode; /* codec speed mode: 1x, 2x, 4x */ + unsigned short ratio; /* MCLK / sample rate */ + u8 ratio_mask; /* ratio bit mask for Master mode */ +}; + +static struct cs4271_clk_cfg cs4271_clk_tab[] = { + {1, CS4271_MODE1_MODE_1X, 256, CS4271_MODE1_DIV_1}, + {1, CS4271_MODE1_MODE_1X, 384, CS4271_MODE1_DIV_15}, + {1, CS4271_MODE1_MODE_1X, 512, CS4271_MODE1_DIV_2}, + {1, CS4271_MODE1_MODE_1X, 768, CS4271_MODE1_DIV_3}, + {1, CS4271_MODE1_MODE_2X, 128, CS4271_MODE1_DIV_1}, + {1, CS4271_MODE1_MODE_2X, 192, CS4271_MODE1_DIV_15}, + {1, CS4271_MODE1_MODE_2X, 256, CS4271_MODE1_DIV_2}, + {1, CS4271_MODE1_MODE_2X, 384, CS4271_MODE1_DIV_3}, + {1, CS4271_MODE1_MODE_4X, 64, CS4271_MODE1_DIV_1}, + {1, CS4271_MODE1_MODE_4X, 96, CS4271_MODE1_DIV_15}, + {1, CS4271_MODE1_MODE_4X, 128, CS4271_MODE1_DIV_2}, + {1, CS4271_MODE1_MODE_4X, 192, CS4271_MODE1_DIV_3}, + {0, CS4271_MODE1_MODE_1X, 256, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_1X, 384, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_1X, 512, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_1X, 768, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_1X, 1024, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_2X, 128, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_2X, 192, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_2X, 256, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_2X, 384, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_2X, 512, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_4X, 64, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_4X, 96, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_4X, 128, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_4X, 192, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_4X, 256, CS4271_MODE1_DIV_2}, +}; + +#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab) + static int cs4271_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -308,23 +326,28 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, unsigned int ratio, val; cs4271->rate = params_rate(params); + + /* Configure DAC */ + if (cs4271->rate < 50000) + val = CS4271_MODE1_MODE_1X; + else if (cs4271->rate < 100000) + val = CS4271_MODE1_MODE_2X; + else + val = CS4271_MODE1_MODE_4X; + ratio = cs4271->mclk / cs4271->rate; for (i = 0; i < CS4171_NR_RATIOS; i++) - if (cs4271_clk_tab[i].ratio == ratio) + if ((cs4271_clk_tab[i].master == cs4271->master) && + (cs4271_clk_tab[i].speed_mode == val) && + (cs4271_clk_tab[i].ratio == ratio)) break; - if ((i == CS4171_NR_RATIOS) || ((ratio == 1024) && cs4271->master)) { + if (i == CS4171_NR_RATIOS) { dev_err(codec->dev, "Invalid sample rate\n"); return -EINVAL; } - /* Configure DAC */ - val = cs4271_clk_tab[i].speed_mode; - - if (cs4271->master) - val |= cs4271_clk_tab[i].mclk_master; - else - val |= cs4271_clk_tab[i].mclk_slave; + val |= cs4271_clk_tab[i].ratio_mask; ret = snd_soc_update_bits(codec, CS4271_MODE1, CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val); -- cgit v1.2.3 From 6d33cd771b779abd8faeaf63821c200ba36a63b1 Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin Date: Mon, 7 Mar 2011 20:29:53 +0300 Subject: ASoC: EDB93xx: Manage I2S rates according to datasheet for CS4271 CODEC. Manage I2S rates according to datasheet for CS4271 CODEC in EDB93xx machine driver. Signed-off-by: Alexander Sverdlin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/ep93xx/edb93xx.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c index b270085227f..d3aa15119d2 100644 --- a/sound/soc/ep93xx/edb93xx.c +++ b/sound/soc/ep93xx/edb93xx.c @@ -41,17 +41,17 @@ static int edb93xx_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int err; + unsigned int mclk_rate; unsigned int rate = params_rate(params); + /* - * We set LRCLK equal to `rate' and SCLK = LRCLK * 64, - * because our sample size is 32 bit * 2 channels. - * I2S standard permits us to transmit more bits than - * the codec uses. - * MCLK = SCLK * 4 is the best recommended value, - * but we have to fall back to ratio 2 for higher - * sample rates. + * According to CS4271 datasheet we use MCLK/LRCK=256 for + * rates below 50kHz and 128 for higher sample rates */ - unsigned int mclk_rate = rate * 64 * ((rate <= 48000) ? 4 : 2); + if (rate < 50000) + mclk_rate = rate * 64 * 4; + else + mclk_rate = rate * 64 * 2; err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | -- cgit v1.2.3 From 5cbf7e49de3276c82705fc303d305ce170793ef6 Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin Date: Mon, 7 Mar 2011 20:30:36 +0300 Subject: ASoC: Improve EP93xx I2S clocks management. Improve EP93xx I2S clocks management. Some freqs values are set not exact as they requested for MCLK and original code was not able to find divisors for SCLK and LRCLK. This code just picks up nearest value from 3 possible variants. This patch makes 44100 and 192000 rates working and fixes capture function (by selecting SCLK/LRCLK=64 where possible). All other rates should work as before. Signed-off-by: Alexander Sverdlin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/ep93xx/ep93xx-i2s.c | 19 +++++++++---------- 1 file changed, 9 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 4d3c1389d3d..207d5893d97 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -242,7 +242,7 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream, { struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); unsigned word_len, div, sdiv, lrdiv; - int found = 0, err; + int err; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -275,15 +275,14 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream, * the codec uses. */ div = clk_get_rate(info->mclk) / params_rate(params); - for (sdiv = 2; sdiv <= 4; sdiv += 2) - for (lrdiv = 64; lrdiv <= 128; lrdiv <<= 1) - if (sdiv * lrdiv == div) { - found = 1; - goto out; - } -out: - if (!found) - return -EINVAL; + sdiv = 4; + if (div > (256 + 512) / 2) { + lrdiv = 128; + } else { + lrdiv = 64; + if (div < (128 + 256) / 2) + sdiv = 2; + } err = clk_set_rate(info->sclk, clk_get_rate(info->mclk) / sdiv); if (err) -- cgit v1.2.3 From 4cfeb695e29d69570496354384f32b862795a47c Mon Sep 17 00:00:00 2001 From: Alexander Sverdlin Date: Mon, 7 Mar 2011 20:30:12 +0300 Subject: ASoC: Enable 192kHz sample rate for EP93xx. Enable 192kHz sample rate for EP93xx. Signed-off-by: Alexander Sverdlin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/ep93xx/ep93xx-i2s.c | 4 ++-- sound/soc/ep93xx/ep93xx-pcm.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 207d5893d97..042f4e93746 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -355,13 +355,13 @@ static struct snd_soc_dai_driver ep93xx_i2s_dai = { .playback = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = EP93XX_I2S_FORMATS, }, .capture = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = EP93XX_I2S_FORMATS, }, .ops = &ep93xx_i2s_dai_ops, diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index 06670776f64..a456e491155 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -35,9 +35,9 @@ static const struct snd_pcm_hardware ep93xx_pcm_hardware = { SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = SNDRV_PCM_RATE_8000_192000, .rate_min = SNDRV_PCM_RATE_8000, - .rate_max = SNDRV_PCM_RATE_96000, + .rate_max = SNDRV_PCM_RATE_192000, .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | -- cgit v1.2.3 From 0627bd2575a30a83901b79d7bcf2ca1fa09fbb8b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Mar 2011 19:09:17 +0000 Subject: ASoC: Fix typo in late revision WM8994 DAC2R name Without this fix the driver won't instantiate properly on relevant devices. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 4afbe3b2e44..d92673314f4 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1418,7 +1418,7 @@ SND_SOC_DAPM_DAC_E("DAC1R", NULL, SND_SOC_NOPM, 0, 0, static const struct snd_soc_dapm_widget wm8994_dac_widgets[] = { SND_SOC_DAPM_DAC("DAC2L", NULL, WM8994_POWER_MANAGEMENT_5, 3, 0), -SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0), +SND_SOC_DAPM_DAC("DAC2R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0), SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0), SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0), }; -- cgit v1.2.3 From 7c2de633863fcd46537d9ddbf5a9701f48225268 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 9 Mar 2011 19:10:15 +0000 Subject: ASoC: Ensure WM8958 gets all WM8994 late revision widgets Without this fix the driver won't instantiate properly on relevant devices. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index d92673314f4..c6c958ee5d5 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3325,6 +3325,12 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8958: snd_soc_add_controls(codec, wm8958_snd_controls, ARRAY_SIZE(wm8958_snd_controls)); + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, + ARRAY_SIZE(wm8994_lateclk_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets, + ARRAY_SIZE(wm8994_adc_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets, + ARRAY_SIZE(wm8994_dac_widgets)); snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets, ARRAY_SIZE(wm8958_dapm_widgets)); break; @@ -3350,6 +3356,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) } break; case WM8958: + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon, + ARRAY_SIZE(wm8994_lateclk_intercon)); snd_soc_dapm_add_routes(dapm, wm8958_intercon, ARRAY_SIZE(wm8958_intercon)); break; -- cgit v1.2.3 From 3cbdd7533148f00444013700af89548b8cf32646 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 29 Aug 2008 16:09:01 +0200 Subject: ALSA: Add snd_ctl_activate_id() Added a new API function snd_ctl_activate_id() for activate / inactivate the control element dynamically. Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/core/control.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index 9ce00ed20fb..db51e4e6498 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -465,6 +465,52 @@ error: return ret; } +/** + * snd_ctl_activate_id - activate/inactivate the control of the given id + * @card: the card instance + * @id: the control id to activate/inactivate + * @active: non-zero to activate + * + * Finds the control instance with the given id, and activate or + * inactivate the control together with notification, if changed. + * + * Returns 0 if unchanged, 1 if changed, or a negative error code on failure. + */ +int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, + int active) +{ + struct snd_kcontrol *kctl; + struct snd_kcontrol_volatile *vd; + unsigned int index_offset; + int ret; + + down_write(&card->controls_rwsem); + kctl = snd_ctl_find_id(card, id); + if (kctl == NULL) { + ret = -ENOENT; + goto unlock; + } + index_offset = snd_ctl_get_ioff(kctl, &kctl->id); + vd = &kctl->vd[index_offset]; + ret = 0; + if (active) { + if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_INACTIVE)) + goto unlock; + vd->access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + } else { + if (vd->access & SNDRV_CTL_ELEM_ACCESS_INACTIVE) + goto unlock; + vd->access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; + } + ret = 1; + unlock: + up_write(&card->controls_rwsem); + if (ret > 0) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, id); + return ret; +} +EXPORT_SYMBOL_GPL(snd_ctl_activate_id); + /** * snd_ctl_rename_id - replace the id of a control on the card * @card: the card instance -- cgit v1.2.3 From 0e45cab64449660fe83bb71208ab43b8d22a5648 Mon Sep 17 00:00:00 2001 From: Christian Glindkamp Date: Wed, 9 Mar 2011 11:20:04 +0100 Subject: ASoC: Add MAX9850 codec driver This patch adds ASoC support for the MAX9850 codec with headphone amplifier. Supported features: - Playback - 16, 20 and 24 bit audio - 8k - 48k sample rates - DAPM Signed-off-by: Christian Glindkamp Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9850.c | 389 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max9850.h | 38 +++++ 4 files changed, 433 insertions(+) create mode 100644 sound/soc/codecs/max9850.c create mode 100644 sound/soc/codecs/max9850.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 10f11dd98c2..d63c1754e05 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -33,6 +33,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_LM4857 if I2C select SND_SOC_MAX98088 if I2C + select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 select SND_SOC_SGTL5000 if I2C @@ -186,6 +187,9 @@ config SND_SOC_DMIC config SND_SOC_MAX98088 tristate +config SND_SOC_MAX9850 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ebb059c0eef..379bc55f072 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -19,6 +19,7 @@ snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o +snd-soc-max9850-objs := max9850.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o @@ -107,6 +108,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o +obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c new file mode 100644 index 00000000000..6d2c5a37550 --- /dev/null +++ b/sound/soc/codecs/max9850.c @@ -0,0 +1,389 @@ +/* + * max9850.c -- codec driver for max9850 + * + * Copyright (C) 2011 taskit GmbH + * + * Author: Christian Glindkamp + * + * Initial development of this code was funded by + * MICRONIC Computer Systeme GmbH, http://www.mcsberlin.de/ + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "max9850.h" + +struct max9850_priv { + unsigned int sysclk; +}; + +/* max9850 register cache */ +static const u8 max9850_reg[MAX9850_CACHEREGNUM] = { + 0x00, 0x00, 0x0c, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 +}; + +/* these registers are not used at the moment but provided for the sake of + * completeness */ +static int max9850_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case MAX9850_STATUSA: + case MAX9850_STATUSB: + return 1; + default: + return 0; + } +} + +static const unsigned int max9850_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0x18, 0x1f, TLV_DB_SCALE_ITEM(-7450, 400, 0), + 0x20, 0x33, TLV_DB_SCALE_ITEM(-4150, 200, 0), + 0x34, 0x37, TLV_DB_SCALE_ITEM(-150, 100, 0), + 0x38, 0x3f, TLV_DB_SCALE_ITEM(250, 50, 0), +}; + +static const struct snd_kcontrol_new max9850_controls[] = { +SOC_SINGLE_TLV("Headphone Volume", MAX9850_VOLUME, 0, 0x3f, 1, max9850_tlv), +SOC_SINGLE("Headphone Switch", MAX9850_VOLUME, 7, 1, 1), +SOC_SINGLE("Mono Switch", MAX9850_GENERAL_PURPOSE, 2, 1, 0), +}; + +static const struct snd_kcontrol_new max9850_mixer_controls[] = { + SOC_DAPM_SINGLE("Line In Switch", MAX9850_ENABLE, 1, 1, 0), +}; + +static const struct snd_soc_dapm_widget max9850_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("Charge Pump 1", MAX9850_ENABLE, 4, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("Charge Pump 2", MAX9850_ENABLE, 5, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MCLK", MAX9850_ENABLE, 6, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("SHDN", MAX9850_ENABLE, 7, 0, NULL, 0), +SND_SOC_DAPM_MIXER_NAMED_CTL("Output Mixer", MAX9850_ENABLE, 2, 0, + &max9850_mixer_controls[0], + ARRAY_SIZE(max9850_mixer_controls)), +SND_SOC_DAPM_PGA("Headphone Output", MAX9850_ENABLE, 3, 0, NULL, 0), +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", MAX9850_ENABLE, 0, 0), +SND_SOC_DAPM_OUTPUT("OUTL"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("OUTR"), +SND_SOC_DAPM_OUTPUT("HPR"), +SND_SOC_DAPM_MIXER("Line Input", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_INPUT("INL"), +SND_SOC_DAPM_INPUT("INR"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* output mixer */ + {"Output Mixer", NULL, "DAC"}, + {"Output Mixer", "Line In Switch", "Line Input"}, + + /* outputs */ + {"Headphone Output", NULL, "Output Mixer"}, + {"HPL", NULL, "Headphone Output"}, + {"HPR", NULL, "Headphone Output"}, + {"OUTL", NULL, "Output Mixer"}, + {"OUTR", NULL, "Output Mixer"}, + + /* inputs */ + {"Line Input", NULL, "INL"}, + {"Line Input", NULL, "INR"}, + + /* supplies */ + {"Output Mixer", NULL, "Charge Pump 1"}, + {"Output Mixer", NULL, "Charge Pump 2"}, + {"Output Mixer", NULL, "SHDN"}, + {"DAC", NULL, "MCLK"}, +}; + +static int max9850_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec); + u64 lrclk_div; + u8 sf, da; + + if(!max9850->sysclk) + return -EINVAL; + + /* lrclk_div = 2^22 * rate / iclk with iclk = mclk / sf */ + sf = (snd_soc_read(codec, MAX9850_CLOCK) >> 2) + 1; + lrclk_div = (1 << 22); + lrclk_div *= params_rate(params); + lrclk_div *= sf; + do_div(lrclk_div, max9850->sysclk); + + snd_soc_write(codec, MAX9850_LRCLK_MSB, (lrclk_div >> 8) & 0x7f); + snd_soc_write(codec, MAX9850_LRCLK_LSB, lrclk_div & 0xff); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + da = 0; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + da = 0x2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + da = 0x3; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, MAX9850_DIGITAL_AUDIO, 0x3, da); + + return 0; +} + +static int max9850_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec); + + /* calculate mclk -> iclk divider */ + if (freq <= 13000000) + snd_soc_write(codec, MAX9850_CLOCK, 0x0); + else if (freq <= 26000000) + snd_soc_write(codec, MAX9850_CLOCK, 0x4); + else if (freq <= 40000000) + snd_soc_write(codec, MAX9850_CLOCK, 0x8); + else + return -EINVAL; + + max9850->sysclk = freq; + return 0; +} + +static int max9850_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 da = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + da |= MAX9850_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + da |= MAX9850_DLY; + break; + case SND_SOC_DAIFMT_RIGHT_J: + da |= MAX9850_RTJ; + break; + case SND_SOC_DAIFMT_LEFT_J: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + da |= MAX9850_BCINV | MAX9850_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + da |= MAX9850_BCINV; + break; + case SND_SOC_DAIFMT_NB_IF: + da |= MAX9850_INV; + break; + default: + return -EINVAL; + } + + /* set da */ + snd_soc_write(codec, MAX9850_DIGITAL_AUDIO, da); + + return 0; +} + +static int max9850_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = snd_soc_cache_sync(codec); + if (ret) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + break; + case SND_SOC_BIAS_OFF: + break; + } + codec->dapm.bias_level = level; + return 0; +} + +#define MAX9850_RATES SNDRV_PCM_RATE_8000_48000 + +#define MAX9850_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops max9850_dai_ops = { + .hw_params = max9850_hw_params, + .set_sysclk = max9850_set_dai_sysclk, + .set_fmt = max9850_set_dai_fmt, +}; + +static struct snd_soc_dai_driver max9850_dai = { + .name = "max9850-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX9850_RATES, + .formats = MAX9850_FORMATS + }, + .ops = &max9850_dai_ops, +}; + +#ifdef CONFIG_PM +static int max9850_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + max9850_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int max9850_resume(struct snd_soc_codec *codec) +{ + max9850_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define max9850_suspend NULL +#define max9850_resume NULL +#endif + +static int max9850_probe(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* enable zero-detect */ + snd_soc_update_bits(codec, MAX9850_GENERAL_PURPOSE, 1, 1); + /* enable slew-rate control */ + snd_soc_update_bits(codec, MAX9850_VOLUME, 0x40, 0x40); + /* set slew-rate 125ms */ + snd_soc_update_bits(codec, MAX9850_CHARGE_PUMP, 0xff, 0xc0); + + snd_soc_dapm_new_controls(dapm, max9850_dapm_widgets, + ARRAY_SIZE(max9850_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); + + snd_soc_add_controls(codec, max9850_controls, + ARRAY_SIZE(max9850_controls)); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_max9850 = { + .probe = max9850_probe, + .suspend = max9850_suspend, + .resume = max9850_resume, + .set_bias_level = max9850_set_bias_level, + .reg_cache_size = ARRAY_SIZE(max9850_reg), + .reg_word_size = sizeof(u8), + .reg_cache_default = max9850_reg, + .volatile_register = max9850_volatile_register, +}; + +static int __devinit max9850_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max9850_priv *max9850; + int ret; + + max9850 = kzalloc(sizeof(struct max9850_priv), GFP_KERNEL); + if (max9850 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, max9850); + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_max9850, &max9850_dai, 1); + if (ret < 0) + kfree(max9850); + return ret; +} + +static __devexit int max9850_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id max9850_i2c_id[] = { + { "max9850", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max9850_i2c_id); + +static struct i2c_driver max9850_i2c_driver = { + .driver = { + .name = "max9850", + .owner = THIS_MODULE, + }, + .probe = max9850_i2c_probe, + .remove = __devexit_p(max9850_i2c_remove), + .id_table = max9850_i2c_id, +}; + +static int __init max9850_init(void) +{ + return i2c_add_driver(&max9850_i2c_driver); +} +module_init(max9850_init); + +static void __exit max9850_exit(void) +{ + i2c_del_driver(&max9850_i2c_driver); +} +module_exit(max9850_exit); + +MODULE_AUTHOR("Christian Glindkamp "); +MODULE_DESCRIPTION("ASoC MAX9850 codec driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max9850.h b/sound/soc/codecs/max9850.h new file mode 100644 index 00000000000..72b1ddb04b0 --- /dev/null +++ b/sound/soc/codecs/max9850.h @@ -0,0 +1,38 @@ +/* + * max9850.h -- codec driver for max9850 + * + * Copyright (C) 2011 taskit GmbH + * Author: Christian Glindkamp + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef _MAX9850_H +#define _MAX9850_H + +#define MAX9850_STATUSA 0x00 +#define MAX9850_STATUSB 0x01 +#define MAX9850_VOLUME 0x02 +#define MAX9850_GENERAL_PURPOSE 0x03 +#define MAX9850_INTERRUPT 0x04 +#define MAX9850_ENABLE 0x05 +#define MAX9850_CLOCK 0x06 +#define MAX9850_CHARGE_PUMP 0x07 +#define MAX9850_LRCLK_MSB 0x08 +#define MAX9850_LRCLK_LSB 0x09 +#define MAX9850_DIGITAL_AUDIO 0x0a + +#define MAX9850_CACHEREGNUM 11 + +/* MAX9850_DIGITAL_AUDIO */ +#define MAX9850_MASTER (1<<7) +#define MAX9850_INV (1<<6) +#define MAX9850_BCINV (1<<5) +#define MAX9850_DLY (1<<3) +#define MAX9850_RTJ (1<<2) + +#endif -- cgit v1.2.3 From 5f3822c48a96e50ee4183b38e72581428eff8308 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Fri, 11 Mar 2011 11:05:37 +0200 Subject: ASoC: PXA: z2: Mute internal speaker when headphones are connected Signed-off-by: Vasily Khoruzhick Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/pxa/z2.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 3ceaef68e01..b0b6c8d7c2f 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -95,6 +95,11 @@ static struct snd_soc_jack_pin hs_jack_pins[] = { .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE, }, + { + .pin = "Ext Spk", + .mask = SND_JACK_HEADPHONE, + .invert = 1 + }, }; /* Headset jack detection gpios */ -- cgit v1.2.3 From 64c25a92e865f06ad8782fbdaa1e2a97d50acf73 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Fri, 11 Mar 2011 11:05:38 +0200 Subject: ASoC: PXA: Z2: Fix codec pin name MONO was renamed to MONO1. Signed-off-by: Vasily Khoruzhick Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/pxa/z2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index b0b6c8d7c2f..d69d9fc3223 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -152,7 +152,7 @@ static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "LINPUT3"); snd_soc_dapm_disable_pin(dapm, "RINPUT3"); snd_soc_dapm_disable_pin(dapm, "OUT3"); - snd_soc_dapm_disable_pin(dapm, "MONO"); + snd_soc_dapm_disable_pin(dapm, "MONO1"); /* Add z2 specific widgets */ snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets, -- cgit v1.2.3 From 27380fb83079bc7bd644e1115bb001dfdcec307f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 11 Mar 2011 12:07:31 +0000 Subject: ASoC: Fix spacing in MAX8950 Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/max9850.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 6d2c5a37550..208d2ee6185 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -118,7 +118,7 @@ static int max9850_hw_params(struct snd_pcm_substream *substream, u64 lrclk_div; u8 sf, da; - if(!max9850->sysclk) + if (!max9850->sysclk) return -EINVAL; /* lrclk_div = 2^22 * rate / iclk with iclk = mclk / sf */ -- cgit v1.2.3