Add libs

3 files changed, 553 insertions, 0 deletions

diff --git a/src/libsound/Makefile.am b/src/libsound/Makefile.am
new file mode 100644
index 0000000..afbbb0a
--- /dev/null
+++ b/src/libsound/Makefile.am
@@ -0,0 +1,8 @@
+AM_CPPFLAGS = -Wall -Wextra -g $(all_includes)
+
+noinst_LIBRARIES = libsound.a
+
+libsound_a_SOURCES = \
+	sound_alsa.c
+
+AM_CPPFLAGS += -DHAVE_ALSA
diff --git a/src/libsound/sound.h b/src/libsound/sound.h
new file mode 100644
index 0000000..6440a3c
--- /dev/null
+++ b/src/libsound/sound.h
@@ -0,0 +1,10 @@
+
+enum paging_signal;
+
+void *sound_open(const char *audiodev, double *tx_frequency, double *rx_frequency, int *am, int channels, double paging_frequency, int samplerate, int latspl, double max_deviation, double max_modulation, double modulation_index);
+int sound_start(void *inst);
+void sound_close(void *inst);
+int sound_write(void *inst, sample_t **samples, uint8_t **power, int num, enum paging_signal *paging_signal, int *on, int channels);
+int sound_read(void *inst, sample_t **samples, int num, int channels, double *rf_level_db);
+int sound_get_tosend(void *inst, int latspl);
+
diff --git a/src/libsound/sound_alsa.c b/src/libsound/sound_alsa.c
new file mode 100644
index 0000000..6216b73
--- /dev/null
+++ b/src/libsound/sound_alsa.c
@@ -0,0 +1,535 @@
+/* Sound device access
+ *
+ * (C) 2016 by Andreas Eversberg <jolly@eversberg.eu>
+ * 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 as published by
+ * the Free Software Foundation, either version 3 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, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <stdlib.h>
+#include <stdint.h>
+#include <math.h>
+#include <alsa/asoundlib.h>
+#include "../libsample/sample.h"
+#include "../libdebug/debug.h"
+#ifdef HAVE_MOBILE
+#include "../libmobile/sender.h"
+#else
+#include "sound.h"
+#endif
+
+typedef struct sound {
+	snd_pcm_t *phandle, *chandle;
+	int pchannels, cchannels;
+	int channels;			/* required number of channels */
+	int samplerate;			/* required sample rate */
+	char *audiodev;			/* required device */
+	double spl_deviation;		/* how much deviation is one sample step */
+#ifdef HAVE_MOBILE
+	double paging_phaseshift;	/* phase to shift every sample */
+	double paging_phase;	 	/* current phase */
+	double rx_frequency[2];		/* rx frequency of radio connected to channel */
+	dispmeasparam_t *dmp[2];
+#endif
+} sound_t;
+
+static int set_hw_params(snd_pcm_t *handle, int samplerate, int *channels)
+{
+	snd_pcm_hw_params_t *hw_params = NULL;
+	int rc;
+	unsigned int rrate;
+
+	rc = snd_pcm_hw_params_malloc(&hw_params);
+	if (rc < 0) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "Failed to allocate hw_params! (%s)\n", snd_strerror(rc));
+		goto error;
+	}
+
+	rc = snd_pcm_hw_params_any(handle, hw_params);
+	if (rc < 0) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "cannot initialize hardware parameter structure (%s)\n", snd_strerror(rc));
+		goto error;
+	}
+
+	rc = snd_pcm_hw_params_set_rate_resample(handle, hw_params, 0);
+	if (rc < 0) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "cannot set real hardware rate (%s)\n", snd_strerror(rc));
+		goto error;
+	}
+
+	rc = snd_pcm_hw_params_set_access (handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
+	if (rc < 0) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "cannot set access to interleaved (%s)\n", snd_strerror(rc));
+		goto error;
+	}
+
+	rc = snd_pcm_hw_params_set_format(handle, hw_params, SND_PCM_FORMAT_S16);
+	if (rc < 0) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "cannot set sample format (%s)\n", snd_strerror(rc));
+		goto error;
+	}
+
+	rrate = samplerate;
+	rc = snd_pcm_hw_params_set_rate_near(handle, hw_params, &rrate, 0);
+	if (rc < 0) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "cannot set sample rate (%s)\n", snd_strerror(rc));
+		goto error;
+	}
+	if ((int)rrate != samplerate) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "Rate doesn't match (requested %dHz, get %dHz)\n", samplerate, rrate);
+		rc = -EIO;
+		goto error;
+	}
+
+	*channels = 1;
+	rc = snd_pcm_hw_params_set_channels(handle, hw_params, *channels);
+	if (rc < 0) {
+		*channels = 2;
+		rc = snd_pcm_hw_params_set_channels(handle, hw_params, *channels);
+		if (rc < 0) {
+			PDEBUG(DSOUND, DEBUG_ERROR, "cannot set channel count to 1 nor 2 (%s)\n", snd_strerror(rc));
+			goto error;
+		}
+	}
+
+	rc = snd_pcm_hw_params(handle, hw_params);
+	if (rc < 0) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "cannot set parameters (%s)\n", snd_strerror(rc));
+		goto error;
+	}
+
+	snd_pcm_hw_params_free(hw_params);
+
+	return 0;
+
+error:
+	if (hw_params) {
+		snd_pcm_hw_params_free(hw_params);
+	}
+
+	return rc;
+}
+
+static int dev_open(sound_t *sound)
+{
+	int rc, rc_rec, rc_play;
+
+	rc_play = snd_pcm_open(&sound->phandle, sound->audiodev, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+	rc_rec = snd_pcm_open(&sound->chandle, sound->audiodev, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
+	if (rc_play < 0 && rc_rec < 0) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "Failed to open '%s'! (%s)\n", sound->audiodev, snd_strerror(rc_play));
+		PDEBUG(DSOUND, DEBUG_ERROR, "Run 'aplay -l' to get a list of available cards and devices.\n");
+		PDEBUG(DSOUND, DEBUG_ERROR, "Then use 'hw:<card>:<device>' for audio device.\n");
+		return rc_play;
+	}
+	if (rc_play < 0) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "Failed to open '%s' for playback! (%s) Please select a device that supports both direction audio.\n", sound->audiodev, snd_strerror(rc_play));
+		return rc_play;
+	}
+	if (rc_rec < 0) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "Failed to open '%s' for capture! (%s) Please select a device that supports both direction audio.\n", sound->audiodev, snd_strerror(rc_rec));
+		return rc_rec;
+	}
+
+	rc = set_hw_params(sound->phandle, sound->samplerate, &sound->pchannels);
+	if (rc < 0) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "Failed to set playback hw params\n");
+		return rc;
+	}
+	if (sound->pchannels < sound->channels) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "Sound card only supports %d channel for playback.\n", sound->pchannels);
+		return rc;
+	}
+	PDEBUG(DSOUND, DEBUG_DEBUG, "Playback with %d channels.\n", sound->pchannels);
+
+	rc = set_hw_params(sound->chandle, sound->samplerate, &sound->cchannels);
+	if (rc < 0) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "Failed to set capture hw params\n");
+		return rc;
+	}
+	if (sound->cchannels < sound->channels) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "Sound card only supports %d channel for capture.\n", sound->cchannels);
+		return -EIO;
+	}
+	PDEBUG(DSOUND, DEBUG_DEBUG, "Capture with %d channels.\n", sound->cchannels);
+
+	rc = snd_pcm_prepare(sound->phandle);
+	if (rc < 0) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "cannot prepare audio interface for use (%s)\n", snd_strerror(rc));
+		return rc;
+	}
+
+	rc = snd_pcm_prepare(sound->chandle);
+	if (rc < 0) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "cannot prepare audio interface for use (%s)\n", snd_strerror(rc));
+		return rc;
+	}
+
+	return 0;
+}
+
+static void dev_close(sound_t *sound)
+{
+	if (sound->phandle != NULL)
+		snd_pcm_close(sound->phandle);
+	if (sound->chandle != NULL)
+		snd_pcm_close(sound->chandle);
+}
+
+void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_frequency, double __attribute__((unused)) *rx_frequency, int __attribute__((unused)) *am, int channels, double __attribute__((unused)) paging_frequency, int samplerate, int __attribute((unused)) latspl, double max_deviation, double __attribute__((unused)) max_modulation, double __attribute__((unused)) modulation_index)
+{
+	sound_t *sound;
+	int rc;
+
+	if (channels < 1 || channels > 2) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "Cannot use more than two channels with the same sound card!\n");
+		return NULL;
+	}
+
+	sound = calloc(1, sizeof(sound_t));
+	if (!sound) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "Failed to alloc memory!\n");
+		return NULL;
+	}
+
+	sound->audiodev = strdup(audiodev); // is feed when closed
+	sound->channels = channels;
+	sound->samplerate = samplerate;
+	sound->spl_deviation = max_deviation / 32767.0;
+#ifdef HAVE_MOBILE
+	sound->paging_phaseshift = 1.0 / ((double)samplerate / 1000.0);
+#endif
+
+	rc = dev_open(sound);
+	if (rc < 0)
+		goto error;
+
+#ifdef HAVE_MOBILE
+	if (rx_frequency) {
+		sender_t *sender;
+		int i;
+		for (i = 0; i < channels; i++) {
+			sound->rx_frequency[i] = rx_frequency[i];
+			sender = get_sender_by_empfangsfrequenz(sound->rx_frequency[i]);
+			if (!sender)
+				continue;
+			sound->dmp[i] = display_measurements_add(&sender->dispmeas, "RX Level", "%.1f dB", DISPLAY_MEAS_PEAK, DISPLAY_MEAS_LEFT, -96.0, 0.0, -INFINITY);
+		}
+	}
+#endif
+
+	return sound;
+
+error:
+	sound_close(sound);
+	return NULL;
+}
+
+/* start streaming */
+int sound_start(void *inst)
+{
+	sound_t *sound = (sound_t *)inst;
+	int16_t buff[2];
+
+	/* trigger capturing */
+	snd_pcm_readi(sound->chandle, buff, 1);
+
+	return 0;
+}
+
+void sound_close(void *inst)
+{
+	sound_t *sound = (sound_t *)inst;
+
+	dev_close(sound);
+	free(sound->audiodev);
+	free(sound);
+}
+
+#ifdef HAVE_MOBILE
+static void gen_paging_tone(sound_t *sound, int16_t *samples, int length, enum paging_signal paging_signal, int on)
+{
+	double phaseshift, phase;
+	int i;
+
+	switch (paging_signal) {
+	case PAGING_SIGNAL_NOTONE:
+		/* no tone if paging signal is on */
+		on = !on;
+		/* FALLTHRU */
+	case PAGING_SIGNAL_TONE:
+		/* tone if paging signal is on */
+		if (on) {
+			phaseshift = sound->paging_phaseshift;
+			phase = sound->paging_phase;
+			for (i = 0; i < length; i++) {
+				if (phase < 0.5)
+					*samples++ = 30000;
+				else
+					*samples++ = -30000;
+				phase += phaseshift;
+				if (phase >= 1.0)
+					phase -= 1.0;
+			}
+			sound->paging_phase = phase;
+		} else
+			memset(samples, 0, length << 1);
+		break;
+	case PAGING_SIGNAL_NEGATIVE:
+		/* negative signal if paging signal is on */
+		on = !on;
+		/* FALLTHRU */
+	case PAGING_SIGNAL_POSITIVE:
+		/* positive signal if paging signal is on */
+		if (on)
+			memset(samples, 127, length << 1);
+		else
+			memset(samples, 128, length << 1);
+		break;
+	case PAGING_SIGNAL_NONE:
+		break;
+	}
+}
+#endif
+
+int sound_write(void *inst, sample_t **samples, uint8_t __attribute__((unused)) **power, int num, enum paging_signal __attribute__((unused)) *paging_signal, int __attribute__((unused)) *on, int channels)
+{
+	sound_t *sound = (sound_t *)inst;
+	double spl_deviation = sound->spl_deviation;
+	int32_t value;
+	int16_t buff[num << 1];
+	int rc;
+	int i, ii;
+
+	if (sound->pchannels == 2) {
+		/* two channels */
+#ifdef HAVE_MOBILE
+		if (paging_signal && on && paging_signal[0] != PAGING_SIGNAL_NONE) {
+			int16_t paging[num << 1];
+			gen_paging_tone(sound, paging, num, paging_signal[0], on[0]);
+			for (i = 0, ii = 0; i < num; i++) {
+				value = samples[0][i] / spl_deviation;
+				if (value > 32767)
+					value = 32767;
+				else if (value < -32767)
+					value = -32767;
+				buff[ii++] = value;
+				buff[ii++] = paging[i];
+			}
+		} else
+#endif
+		if (channels == 2) {
+			for (i = 0, ii = 0; i < num; i++) {
+				value = samples[0][i] / spl_deviation;
+				if (value > 32767)
+					value = 32767;
+				else if (value < -32767)
+					value = -32767;
+				buff[ii++] = value;
+				value = samples[1][i] / spl_deviation;
+				if (value > 32767)
+					value = 32767;
+				else if (value < -32767)
+					value = -32767;
+				buff[ii++] = value;
+			}
+		} else {
+			for (i = 0, ii = 0; i < num; i++) {
+				value = samples[0][i] / spl_deviation;
+				if (value > 32767)
+					value = 32767;
+				else if (value < -32767)
+					value = -32767;
+				buff[ii++] = value;
+				buff[ii++] = value;
+			}
+		}
+	} else {
+		/* one channel */
+		for (i = 0, ii = 0; i < num; i++) {
+			value = samples[0][i] / spl_deviation;
+			if (value > 32767)
+				value = 32767;
+			else if (value < -32767)
+				value = -32767;
+			buff[ii++] = value;
+		}
+	}
+	rc = snd_pcm_writei(sound->phandle, buff, num);
+
+	if (rc < 0) {
+		PDEBUG(DSOUND, DEBUG_ERROR, "failed to write audio to interface (%s)\n", snd_strerror(rc));
+		if (rc == -EPIPE) {
+			dev_close(sound);
+			rc = dev_open(sound);
+			if (rc < 0)
+				return rc;
+			sound_start(sound);
+			return -EPIPE; /* indicate what happened */
+		}
+		return rc;
+	}
+
+	if (rc != num)
+		PDEBUG(DSOUND, DEBUG_ERROR, "short write to audio interface, written %d bytes, got %d bytes\n", num, rc);
+
+	return rc;
+}
+
+#define KEEP_FRAMES	8	/* minimum frames not to read, due to bug in ALSA */
+
+int sound_read(void *inst, sample_t **samples, int num, int channels, double __attribute__((unused)) *rf_level_db)
+{
+	sound_t *sound = (sound_t *)inst;
+	double spl_deviation = sound->spl_deviation;
+	int16_t buff[num << 1];
+	int32_t spl;
+	int32_t max[2], a;
+	int in, rc;
+	int i, ii;
+
+	/* make valgrind happy, because snd_pcm_readi() does not seem to initially fill buffer with values */
+	memset(buff, 0, sizeof(buff));
+
+	/* get samples in rx buffer */
+	in = snd_pcm_avail(sound->chandle);
+	/* if not more than KEEP_FRAMES frames available, try next time */
+	if (in <= KEEP_FRAMES)
+		return 0;
+	/* read some frames less than in buffer, because snd_pcm_readi() seems
+	 * to corrupt last frames */
+	in -= KEEP_FRAMES;
+	if (in > num)
+		in = num;
+
+	rc = snd_pcm_readi(sound->chandle, buff, in);
+
+	if (rc < 0) {
+		if (errno == EAGAIN)
+			return 0;
+		PDEBUG(DSOUND, DEBUG_ERROR, "failed to read audio from interface (%s)\n", snd_strerror(rc));
+		/* recover read */
+		if (rc == -EPIPE) {
+			dev_close(sound);
+			rc = dev_open(sound);
+			if (rc < 0)
+				return rc;
+			sound_start(sound);
+			return -EPIPE; /* indicate what happened */
+		}
+		return rc;
+	}
+
+	if (rc == 0)
+		return rc;
+
+	if (sound->cchannels == 2) {
+		if (channels < 2) {
+			for (i = 0, ii = 0; i < rc; i++) {
+				spl = buff[ii++];
+				spl += buff[ii++];
+				a = (spl >= 0) ? spl : -spl;
+				if (i == 0 || a > max[0])
+					max[0] = a;
+				samples[0][i] = (double)spl * spl_deviation;
+			}
+		} else {
+			for (i = 0, ii = 0; i < rc; i++) {
+				spl = buff[ii++];
+				a = (spl >= 0) ? spl : -spl;
+				if (i == 0 || a > max[0])
+					max[0] = a;
+				samples[0][i] = (double)spl * spl_deviation;
+				spl = buff[ii++];
+				a = (spl >= 0) ? spl : -spl;
+				if (i == 0 || a > max[1])
+					max[1] = a;
+				samples[1][i] = (double)spl * spl_deviation;
+			}
+		}
+	} else {
+		for (i = 0, ii = 0; i < rc; i++) {
+			spl = buff[ii++];
+			a = (spl >= 0) ? spl : -spl;
+			if (i == 0 || a > max[0])
+				max[0] = a;
+			samples[0][i] = (double)spl * spl_deviation;
+		}
+	}
+
+#ifdef HAVE_MOBILE
+	sender_t *sender;
+	for (i = 0; i < channels; i++) {
+		sender = get_sender_by_empfangsfrequenz(sound->rx_frequency[i]);
+		if (!sender)
+			continue;
+		display_measurements_update(sound->dmp[i], log10((double)max[i] / 32768.0) * 20, 0.0);
+		if (rf_level_db)
+			rf_level_db[i] = 0.0;
+	}
+#endif
+
+	return rc;
+}
+
+/* 
+ * get playback buffer space
+ *
+ * return number of samples to be sent */
+int sound_get_tosend(void *inst, int latspl)
+{
+	sound_t *sound = (sound_t *)inst;
+	int rc;
+	snd_pcm_sframes_t delay;
+	int tosend;
+
+	rc = snd_pcm_delay(sound->phandle, &delay);
+	if (rc < 0) {
+		if (rc == -32)
+			PDEBUG(DSOUND, DEBUG_ERROR, "Buffer underrun: Please use higher latency and enable real time scheduling\n");
+		else
+			PDEBUG(DSOUND, DEBUG_ERROR, "failed to get delay from interface (%s)\n", snd_strerror(rc));
+		if (rc == -EPIPE) {
+			dev_close(sound);
+			rc = dev_open(sound);
+			if (rc < 0)
+				return rc;
+			sound_start(sound);
+			return -EPIPE; /* indicate what happened */
+		}
+		return rc;
+	}
+
+	tosend = latspl - delay;
+	return tosend;
+}
+
+int sound_is_stereo_capture(void *inst)
+{
+	sound_t *sound = (sound_t *)inst;
+
+	if (sound->cchannels == 2)
+		return 1;
+	return 0;
+}
+
+int sound_is_stereo_playback(void *inst)
+{
+	sound_t *sound = (sound_t *)inst;
+
+	if (sound->pchannels == 2)
+		return 1;
+	return 0;
+}
+