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asterisk/channels/chan_alsa.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
*
* By Matthew Fredrickson <creslin@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
* \brief ALSA sound card channel driver
*
* \author Matthew Fredrickson <creslin@digium.com>
*
* \par See also
* \arg Config_alsa
*
* \ingroup channel_drivers
*/
/*** MODULEINFO
<depend>asound</depend>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <unistd.h>
#include <fcntl.h>
#include <errno.h>
#include <sys/ioctl.h>
#include <sys/time.h>
#include <string.h>
#include <stdlib.h>
#include <stdio.h>
#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
#include <alsa/asoundlib.h>
#include "asterisk/frame.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/module.h"
#include "asterisk/options.h"
#include "asterisk/pbx.h"
#include "asterisk/config.h"
#include "asterisk/cli.h"
#include "asterisk/utils.h"
#include "asterisk/causes.h"
#include "asterisk/endian.h"
#include "asterisk/stringfields.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/musiconhold.h"
#include "busy.h"
#include "ringtone.h"
#include "ring10.h"
#include "answer.h"
#ifdef ALSA_MONITOR
#include "alsa-monitor.h"
#endif
/*! Global jitterbuffer configuration - by default, jb is disabled */
static struct ast_jb_conf default_jbconf =
{
.flags = 0,
.max_size = -1,
.resync_threshold = -1,
.impl = ""
};
static struct ast_jb_conf global_jbconf;
#define DEBUG 0
/* Which device to use */
#define ALSA_INDEV "hw:0,0"
#define ALSA_OUTDEV "hw:0,0"
#define DESIRED_RATE 8000
/* Lets use 160 sample frames, just like GSM. */
#define FRAME_SIZE 160
#define PERIOD_FRAMES 80 /* 80 Frames, at 2 bytes each */
/* When you set the frame size, you have to come up with
the right buffer format as well. */
/* 5 64-byte frames = one frame */
#define BUFFER_FMT ((buffersize * 10) << 16) | (0x0006);
/* Don't switch between read/write modes faster than every 300 ms */
#define MIN_SWITCH_TIME 600
#if __BYTE_ORDER == __LITTLE_ENDIAN
static snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
#else
static snd_pcm_format_t format = SND_PCM_FORMAT_S16_BE;
#endif
/* static int block = O_NONBLOCK; */
static char indevname[50] = ALSA_INDEV;
static char outdevname[50] = ALSA_OUTDEV;
#if 0
static struct timeval lasttime;
#endif
static int usecnt;
static int silencesuppression = 0;
static int silencethreshold = 1000;
AST_MUTEX_DEFINE_STATIC(usecnt_lock);
AST_MUTEX_DEFINE_STATIC(alsalock);
static const char tdesc[] = "ALSA Console Channel Driver";
static const char config[] = "alsa.conf";
static char context[AST_MAX_CONTEXT] = "default";
static char language[MAX_LANGUAGE] = "";
static char exten[AST_MAX_EXTENSION] = "s";
static char mohinterpret[MAX_MUSICCLASS];
static int hookstate=0;
static short silence[FRAME_SIZE] = {0, };
struct sound {
int ind;
short *data;
int datalen;
int samplen;
int silencelen;
int repeat;
};
static struct sound sounds[] = {
{ AST_CONTROL_RINGING, ringtone, sizeof(ringtone)/2, 16000, 32000, 1 },
{ AST_CONTROL_BUSY, busy, sizeof(busy)/2, 4000, 4000, 1 },
{ AST_CONTROL_CONGESTION, busy, sizeof(busy)/2, 2000, 2000, 1 },
{ AST_CONTROL_RING, ring10, sizeof(ring10)/2, 16000, 32000, 1 },
{ AST_CONTROL_ANSWER, answer, sizeof(answer)/2, 2200, 0, 0 },
};
/* Sound command pipe */
static int sndcmd[2];
static struct chan_alsa_pvt {
/* We only have one ALSA structure -- near sighted perhaps, but it
keeps this driver as simple as possible -- as it should be. */
struct ast_channel *owner;
char exten[AST_MAX_EXTENSION];
char context[AST_MAX_CONTEXT];
#if 0
snd_pcm_t *card;
#endif
snd_pcm_t *icard, *ocard;
} alsa;
/* Number of buffers... Each is FRAMESIZE/8 ms long. For example
with 160 sample frames, and a buffer size of 3, we have a 60ms buffer,
usually plenty. */
pthread_t sthread;
#define MAX_BUFFER_SIZE 100
/* File descriptors for sound device */
static int readdev = -1;
static int writedev = -1;
static int autoanswer = 1;
static int cursound = -1;
static int sampsent = 0;
static int silencelen=0;
static int offset=0;
static int nosound=0;
/* ZZ */
static struct ast_channel *alsa_request(const char *type, int format, void *data, int *cause);
static int alsa_digit(struct ast_channel *c, char digit);
static int alsa_text(struct ast_channel *c, const char *text);
static int alsa_hangup(struct ast_channel *c);
static int alsa_answer(struct ast_channel *c);
static struct ast_frame *alsa_read(struct ast_channel *chan);
static int alsa_call(struct ast_channel *c, char *dest, int timeout);
static int alsa_write(struct ast_channel *chan, struct ast_frame *f);
static int alsa_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen);
static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
static const struct ast_channel_tech alsa_tech = {
.type = "Console",
.description = tdesc,
.capabilities = AST_FORMAT_SLINEAR,
.requester = alsa_request,
.send_digit = alsa_digit,
.send_text = alsa_text,
.hangup = alsa_hangup,
.answer = alsa_answer,
.read = alsa_read,
.call = alsa_call,
.write = alsa_write,
.indicate = alsa_indicate,
.fixup = alsa_fixup,
};
static int send_sound(void)
{
short myframe[FRAME_SIZE];
int total = FRAME_SIZE;
short *frame = NULL;
int amt=0;
int res;
int myoff;
snd_pcm_state_t state;
if (cursound > -1) {
res = total;
if (sampsent < sounds[cursound].samplen) {
myoff=0;
while(total) {
amt = total;
if (amt > (sounds[cursound].datalen - offset))
amt = sounds[cursound].datalen - offset;
memcpy(myframe + myoff, sounds[cursound].data + offset, amt * 2);
total -= amt;
offset += amt;
sampsent += amt;
myoff += amt;
if (offset >= sounds[cursound].datalen)
offset = 0;
}
/* Set it up for silence */
if (sampsent >= sounds[cursound].samplen)
silencelen = sounds[cursound].silencelen;
frame = myframe;
} else {
if (silencelen > 0) {
frame = silence;
silencelen -= res;
} else {
if (sounds[cursound].repeat) {
/* Start over */
sampsent = 0;
offset = 0;
} else {
cursound = -1;
nosound = 0;
}
return 0;
}
}
if (res == 0 || !frame) {
return 0;
}
#ifdef ALSA_MONITOR
alsa_monitor_write((char *)frame, res * 2);
#endif
state = snd_pcm_state(alsa.ocard);
if (state == SND_PCM_STATE_XRUN) {
snd_pcm_prepare(alsa.ocard);
}
res = snd_pcm_writei(alsa.ocard, frame, res);
if (res > 0)
return 0;
return 0;
}
return 0;
}
static void *sound_thread(void *unused)
{
fd_set rfds;
fd_set wfds;
int max;
int res;
for(;;) {
FD_ZERO(&rfds);
FD_ZERO(&wfds);
max = sndcmd[0];
FD_SET(sndcmd[0], &rfds);
if (cursound > -1) {
FD_SET(writedev, &wfds);
if (writedev > max)
max = writedev;
}
#ifdef ALSA_MONITOR
if (!alsa.owner) {
FD_SET(readdev, &rfds);
if (readdev > max)
max = readdev;
}
#endif
res = ast_select(max + 1, &rfds, &wfds, NULL, NULL);
if (res < 1) {
ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno));
continue;
}
#ifdef ALSA_MONITOR
if (FD_ISSET(readdev, &rfds)) {
/* Keep the pipe going with read audio */
snd_pcm_state_t state;
short buf[FRAME_SIZE];
int r;
state = snd_pcm_state(alsa.ocard);
if (state == SND_PCM_STATE_XRUN) {
snd_pcm_prepare(alsa.ocard);
}
r = snd_pcm_readi(alsa.icard, buf, FRAME_SIZE);
if (r == -EPIPE) {
#if DEBUG
ast_log(LOG_ERROR, "XRUN read\n");
#endif
snd_pcm_prepare(alsa.icard);
} else if (r == -ESTRPIPE) {
ast_log(LOG_ERROR, "-ESTRPIPE\n");
snd_pcm_prepare(alsa.icard);
} else if (r < 0) {
ast_log(LOG_ERROR, "Read error: %s\n", snd_strerror(r));
} else
alsa_monitor_read((char *)buf, r * 2);
}
#endif
if (FD_ISSET(sndcmd[0], &rfds)) {
read(sndcmd[0], &cursound, sizeof(cursound));
silencelen = 0;
offset = 0;
sampsent = 0;
}
if (FD_ISSET(writedev, &wfds))
if (send_sound())
ast_log(LOG_WARNING, "Failed to write sound\n");
}
/* Never reached */
return NULL;
}
static snd_pcm_t *alsa_card_init(char *dev, snd_pcm_stream_t stream)
{
int err;
int direction;
snd_pcm_t *handle = NULL;
snd_pcm_hw_params_t *hwparams = NULL;
snd_pcm_sw_params_t *swparams = NULL;
struct pollfd pfd;
snd_pcm_uframes_t period_size = PERIOD_FRAMES * 4;
/* int period_bytes = 0; */
snd_pcm_uframes_t buffer_size = 0;
unsigned int rate = DESIRED_RATE;
#if 0
unsigned int per_min = 1;
#endif
/* unsigned int per_max = 8; */
snd_pcm_uframes_t start_threshold, stop_threshold;
err = snd_pcm_open(&handle, dev, stream, O_NONBLOCK);
if (err < 0) {
ast_log(LOG_ERROR, "snd_pcm_open failed: %s\n", snd_strerror(err));
return NULL;
} else {
ast_log(LOG_DEBUG, "Opening device %s in %s mode\n", dev, (stream == SND_PCM_STREAM_CAPTURE) ? "read" : "write");
}
snd_pcm_hw_params_alloca(&hwparams);
snd_pcm_hw_params_any(handle, hwparams);
err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
ast_log(LOG_ERROR, "set_access failed: %s\n", snd_strerror(err));
}
err = snd_pcm_hw_params_set_format(handle, hwparams, format);
if (err < 0) {
ast_log(LOG_ERROR, "set_format failed: %s\n", snd_strerror(err));
}
err = snd_pcm_hw_params_set_channels(handle, hwparams, 1);
if (err < 0) {
ast_log(LOG_ERROR, "set_channels failed: %s\n", snd_strerror(err));
}
direction = 0;
err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &rate, &direction);
if (rate != DESIRED_RATE) {
ast_log(LOG_WARNING, "Rate not correct, requested %d, got %d\n", DESIRED_RATE, rate);
}
direction = 0;
err = snd_pcm_hw_params_set_period_size_near(handle, hwparams, &period_size, &direction);
if (err < 0) {
ast_log(LOG_ERROR, "period_size(%ld frames) is bad: %s\n", period_size, snd_strerror(err));
} else {
ast_log(LOG_DEBUG, "Period size is %d\n", err);
}
buffer_size = 4096 * 2; /* period_size * 16; */
err = snd_pcm_hw_params_set_buffer_size_near(handle, hwparams, &buffer_size);
if (err < 0) {
ast_log(LOG_WARNING, "Problem setting buffer size of %ld: %s\n", buffer_size, snd_strerror(err));
} else {
ast_log(LOG_DEBUG, "Buffer size is set to %d frames\n", err);
}
#if 0
direction = 0;
err = snd_pcm_hw_params_set_periods_min(handle, hwparams, &per_min, &direction);
if (err < 0) {
ast_log(LOG_ERROR, "periods_min: %s\n", snd_strerror(err));
}
err = snd_pcm_hw_params_set_periods_max(handle, hwparams, &per_max, 0);
if (err < 0) {
ast_log(LOG_ERROR, "periods_max: %s\n", snd_strerror(err));
}
#endif
err = snd_pcm_hw_params(handle, hwparams);
if (err < 0) {
ast_log(LOG_ERROR, "Couldn't set the new hw params: %s\n", snd_strerror(err));
}
snd_pcm_sw_params_alloca(&swparams);
snd_pcm_sw_params_current(handle, swparams);
#if 1
if (stream == SND_PCM_STREAM_PLAYBACK) {
start_threshold = period_size;
} else {
start_threshold = 1;
}
err = snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold);
if (err < 0) {
ast_log(LOG_ERROR, "start threshold: %s\n", snd_strerror(err));
}
#endif
#if 1
if (stream == SND_PCM_STREAM_PLAYBACK) {
stop_threshold = buffer_size;
} else {
stop_threshold = buffer_size;
}
err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, stop_threshold);
if (err < 0) {
ast_log(LOG_ERROR, "stop threshold: %s\n", snd_strerror(err));
}
#endif
#if 0
err = snd_pcm_sw_params_set_xfer_align(handle, swparams, PERIOD_FRAMES);
if (err < 0) {
ast_log(LOG_ERROR, "Unable to set xfer alignment: %s\n", snd_strerror(err));
}
#endif
#if 0
err = snd_pcm_sw_params_set_silence_threshold(handle, swparams, silencethreshold);
if (err < 0) {
ast_log(LOG_ERROR, "Unable to set silence threshold: %s\n", snd_strerror(err));
}
#endif
err = snd_pcm_sw_params(handle, swparams);
if (err < 0) {
ast_log(LOG_ERROR, "sw_params: %s\n", snd_strerror(err));
}
err = snd_pcm_poll_descriptors_count(handle);
if (err <= 0) {
ast_log(LOG_ERROR, "Unable to get a poll descriptors count, error is %s\n", snd_strerror(err));
}
if (err != 1) {
ast_log(LOG_DEBUG, "Can't handle more than one device\n");
}
snd_pcm_poll_descriptors(handle, &pfd, err);
ast_log(LOG_DEBUG, "Acquired fd %d from the poll descriptor\n", pfd.fd);
if (stream == SND_PCM_STREAM_CAPTURE)
readdev = pfd.fd;
else
writedev = pfd.fd;
return handle;
}
static int soundcard_init(void)
{
alsa.icard = alsa_card_init(indevname, SND_PCM_STREAM_CAPTURE);
alsa.ocard = alsa_card_init(outdevname, SND_PCM_STREAM_PLAYBACK);
if (!alsa.icard || !alsa.ocard) {
ast_log(LOG_ERROR, "Problem opening alsa I/O devices\n");
return -1;
}
return readdev;
}
static int alsa_digit(struct ast_channel *c, char digit)
{
ast_mutex_lock(&alsalock);
ast_verbose( " << Console Received digit %c >> \n", digit);
ast_mutex_unlock(&alsalock);
return 0;
}
static int alsa_text(struct ast_channel *c, const char *text)
{
ast_mutex_lock(&alsalock);
ast_verbose( " << Console Received text %s >> \n", text);
ast_mutex_unlock(&alsalock);
return 0;
}
static void grab_owner(void)
{
while(alsa.owner && ast_mutex_trylock(&alsa.owner->lock)) {
ast_mutex_unlock(&alsalock);
usleep(1);
ast_mutex_lock(&alsalock);
}
}
static int alsa_call(struct ast_channel *c, char *dest, int timeout)
{
int res = 3;
struct ast_frame f = { AST_FRAME_CONTROL };
ast_mutex_lock(&alsalock);
ast_verbose( " << Call placed to '%s' on console >> \n", dest);
if (autoanswer) {
ast_verbose( " << Auto-answered >> \n" );
grab_owner();
if (alsa.owner) {
f.subclass = AST_CONTROL_ANSWER;
ast_queue_frame(alsa.owner, &f);
ast_mutex_unlock(&alsa.owner->lock);
}
} else {
ast_verbose( " << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
grab_owner();
if (alsa.owner) {
f.subclass = AST_CONTROL_RINGING;
ast_queue_frame(alsa.owner, &f);
ast_mutex_unlock(&alsa.owner->lock);
}
write(sndcmd[1], &res, sizeof(res));
}
snd_pcm_prepare(alsa.icard);
snd_pcm_start(alsa.icard);
ast_mutex_unlock(&alsalock);
return 0;
}
static void answer_sound(void)
{
int res;
nosound = 1;
res = 4;
write(sndcmd[1], &res, sizeof(res));
}
static int alsa_answer(struct ast_channel *c)
{
ast_mutex_lock(&alsalock);
ast_verbose( " << Console call has been answered >> \n");
answer_sound();
ast_setstate(c, AST_STATE_UP);
cursound = -1;
snd_pcm_prepare(alsa.icard);
snd_pcm_start(alsa.icard);
ast_mutex_unlock(&alsalock);
return 0;
}
static int alsa_hangup(struct ast_channel *c)
{
int res;
ast_mutex_lock(&alsalock);
cursound = -1;
c->tech_pvt = NULL;
alsa.owner = NULL;
ast_verbose( " << Hangup on console >> \n");
ast_mutex_lock(&usecnt_lock);
usecnt--;
ast_mutex_unlock(&usecnt_lock);
if (hookstate) {
if (autoanswer) {
hookstate = 0;
} else {
/* Congestion noise */
res = 2;
write(sndcmd[1], &res, sizeof(res));
hookstate = 0;
}
}
snd_pcm_drop(alsa.icard);
ast_mutex_unlock(&alsalock);
return 0;
}
static int alsa_write(struct ast_channel *chan, struct ast_frame *f)
{
static char sizbuf[8000];
static int sizpos = 0;
int len = sizpos;
int pos;
int res = 0;
/* size_t frames = 0; */
snd_pcm_state_t state;
/* Immediately return if no sound is enabled */
if (nosound)
return 0;
ast_mutex_lock(&alsalock);
/* Stop any currently playing sound */
if (cursound != -1) {
snd_pcm_drop(alsa.ocard);
snd_pcm_prepare(alsa.ocard);
cursound = -1;
}
/* We have to digest the frame in 160-byte portions */
if (f->datalen > sizeof(sizbuf) - sizpos) {
ast_log(LOG_WARNING, "Frame too large\n");
res = -1;
} else {
memcpy(sizbuf + sizpos, f->data, f->datalen);
len += f->datalen;
pos = 0;
#ifdef ALSA_MONITOR
alsa_monitor_write(sizbuf, len);
#endif
state = snd_pcm_state(alsa.ocard);
if (state == SND_PCM_STATE_XRUN) {
snd_pcm_prepare(alsa.ocard);
}
res = snd_pcm_writei(alsa.ocard, sizbuf, len/2);
if (res == -EPIPE) {
#if DEBUG
ast_log(LOG_DEBUG, "XRUN write\n");
#endif
snd_pcm_prepare(alsa.ocard);
res = snd_pcm_writei(alsa.ocard, sizbuf, len/2);
if (res != len/2) {
ast_log(LOG_ERROR, "Write error: %s\n", snd_strerror(res));
res = -1;
} else if (res < 0) {
ast_log(LOG_ERROR, "Write error %s\n", snd_strerror(res));
res = -1;
}
} else {
if (res == -ESTRPIPE) {
ast_log(LOG_ERROR, "You've got some big problems\n");
} else if (res < 0)
ast_log(LOG_NOTICE, "Error %d on write\n", res);
}
}
ast_mutex_unlock(&alsalock);
if (res > 0)
res = 0;
return res;
}
static struct ast_frame *alsa_read(struct ast_channel *chan)
{
static struct ast_frame f;
static short __buf[FRAME_SIZE + AST_FRIENDLY_OFFSET/2];
short *buf;
static int readpos = 0;
static int left = FRAME_SIZE;
snd_pcm_state_t state;
int r = 0;
int off = 0;
ast_mutex_lock(&alsalock);
/* Acknowledge any pending cmd */
f.frametype = AST_FRAME_NULL;
f.subclass = 0;
f.samples = 0;
f.datalen = 0;
f.data = NULL;
f.offset = 0;
f.src = "Console";
f.mallocd = 0;
f.delivery.tv_sec = 0;
f.delivery.tv_usec = 0;
state = snd_pcm_state(alsa.icard);
if ((state != SND_PCM_STATE_PREPARED) &&
(state != SND_PCM_STATE_RUNNING)) {
snd_pcm_prepare(alsa.icard);
}
buf = __buf + AST_FRIENDLY_OFFSET/2;
r = snd_pcm_readi(alsa.icard, buf + readpos, left);
if (r == -EPIPE) {
#if DEBUG
ast_log(LOG_ERROR, "XRUN read\n");
#endif
snd_pcm_prepare(alsa.icard);
} else if (r == -ESTRPIPE) {
ast_log(LOG_ERROR, "-ESTRPIPE\n");
snd_pcm_prepare(alsa.icard);
} else if (r < 0) {
ast_log(LOG_ERROR, "Read error: %s\n", snd_strerror(r));
} else if (r >= 0) {
off -= r;
}
/* Update positions */
readpos += r;
left -= r;
if (readpos >= FRAME_SIZE) {
/* A real frame */
readpos = 0;
left = FRAME_SIZE;
if (chan->_state != AST_STATE_UP) {
/* Don't transmit unless it's up */
ast_mutex_unlock(&alsalock);
return &f;
}
f.frametype = AST_FRAME_VOICE;
f.subclass = AST_FORMAT_SLINEAR;
f.samples = FRAME_SIZE;
f.datalen = FRAME_SIZE * 2;
f.data = buf;
f.offset = AST_FRIENDLY_OFFSET;
f.src = "Console";
f.mallocd = 0;
#ifdef ALSA_MONITOR
alsa_monitor_read((char *)buf, FRAME_SIZE * 2);
#endif
}
ast_mutex_unlock(&alsalock);
return &f;
}
static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
{
struct chan_alsa_pvt *p = newchan->tech_pvt;
ast_mutex_lock(&alsalock);
p->owner = newchan;
ast_mutex_unlock(&alsalock);
return 0;
}
static int alsa_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen)
{
int res = 0;
ast_mutex_lock(&alsalock);
switch(cond) {
case AST_CONTROL_BUSY:
res = 1;
break;
case AST_CONTROL_CONGESTION:
res = 2;
break;
case AST_CONTROL_RINGING:
break;
case -1:
res = -1;
break;
case AST_CONTROL_VIDUPDATE:
res = -1;
break;
case AST_CONTROL_HOLD:
ast_verbose( " << Console Has Been Placed on Hold >> \n");
ast_moh_start(chan, data, mohinterpret);
break;
case AST_CONTROL_UNHOLD:
ast_verbose( " << Console Has Been Retrieved from Hold >> \n");
ast_moh_stop(chan);
break;
default:
ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name);
res = -1;
}
if (res > -1)
write(sndcmd[1], &res, sizeof(res));
ast_mutex_unlock(&alsalock);
return res;
}
static struct ast_channel *alsa_new(struct chan_alsa_pvt *p, int state)
{
struct ast_channel *tmp;
tmp = ast_channel_alloc(1);
if (tmp) {
tmp->tech = &alsa_tech;
ast_string_field_build(tmp, name, "ALSA/%s", indevname);
tmp->fds[0] = readdev;
tmp->nativeformats = AST_FORMAT_SLINEAR;
tmp->readformat = AST_FORMAT_SLINEAR;
tmp->writeformat = AST_FORMAT_SLINEAR;
tmp->tech_pvt = p;
if (!ast_strlen_zero(p->context))
ast_copy_string(tmp->context, p->context, sizeof(tmp->context));
if (!ast_strlen_zero(p->exten))
ast_copy_string(tmp->exten, p->exten, sizeof(tmp->exten));
if (!ast_strlen_zero(language))
ast_string_field_set(tmp, language, language);
p->owner = tmp;
ast_setstate(tmp, state);
ast_mutex_lock(&usecnt_lock);
usecnt++;
ast_mutex_unlock(&usecnt_lock);
ast_update_use_count();
ast_jb_configure(tmp, &global_jbconf);
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(tmp)) {
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
ast_hangup(tmp);
tmp = NULL;
}
}
}
return tmp;
}
static struct ast_channel *alsa_request(const char *type, int format, void *data, int *cause)
{
int oldformat = format;
struct ast_channel *tmp=NULL;
format &= AST_FORMAT_SLINEAR;
if (!format) {
ast_log(LOG_NOTICE, "Asked to get a channel of format '%d'\n", oldformat);
return NULL;
}
ast_mutex_lock(&alsalock);
if (alsa.owner) {
ast_log(LOG_NOTICE, "Already have a call on the ALSA channel\n");
*cause = AST_CAUSE_BUSY;
} else {
tmp= alsa_new(&alsa, AST_STATE_DOWN);
if (!tmp) {
ast_log(LOG_WARNING, "Unable to create new ALSA channel\n");
}
}
ast_mutex_unlock(&alsalock);
return tmp;
}
static int console_autoanswer(int fd, int argc, char *argv[])
{
int res = RESULT_SUCCESS;;
if ((argc != 1) && (argc != 2))
return RESULT_SHOWUSAGE;
ast_mutex_lock(&alsalock);
if (argc == 1) {
ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off");
} else {
if (!strcasecmp(argv[1], "on"))
autoanswer = -1;
else if (!strcasecmp(argv[1], "off"))
autoanswer = 0;
else
res = RESULT_SHOWUSAGE;
}
ast_mutex_unlock(&alsalock);
return res;
}
static char *autoanswer_complete(const char *line, const char *word, int pos, int state)
{
#ifndef MIN
#define MIN(a,b) ((a) < (b) ? (a) : (b))
#endif
switch(state) {
case 0:
if (!ast_strlen_zero(word) && !strncasecmp(word, "on", MIN(strlen(word), 2)))
return ast_strdup("on");
case 1:
if (!ast_strlen_zero(word) && !strncasecmp(word, "off", MIN(strlen(word), 3)))
return ast_strdup("off");
default:
return NULL;
}
return NULL;
}
static const char autoanswer_usage[] =
"Usage: autoanswer [on|off]\n"
" Enables or disables autoanswer feature. If used without\n"
" argument, displays the current on/off status of autoanswer.\n"
" The default value of autoanswer is in 'alsa.conf'.\n";
static int console_answer(int fd, int argc, char *argv[])
{
int res = RESULT_SUCCESS;
if (argc != 1)
return RESULT_SHOWUSAGE;
ast_mutex_lock(&alsalock);
if (!alsa.owner) {
ast_cli(fd, "No one is calling us\n");
res = RESULT_FAILURE;
} else {
hookstate = 1;
cursound = -1;
grab_owner();
if (alsa.owner) {
struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER };
ast_queue_frame(alsa.owner, &f);
ast_mutex_unlock(&alsa.owner->lock);
}
answer_sound();
}
snd_pcm_prepare(alsa.icard);
snd_pcm_start(alsa.icard);
ast_mutex_unlock(&alsalock);
return RESULT_SUCCESS;
}
static char sendtext_usage[] =
"Usage: send text <message>\n"
" Sends a text message for display on the remote terminal.\n";
static int console_sendtext(int fd, int argc, char *argv[])
{
int tmparg = 2;
int res = RESULT_SUCCESS;
if (argc < 2)
return RESULT_SHOWUSAGE;
ast_mutex_lock(&alsalock);
if (!alsa.owner) {
ast_cli(fd, "No one is calling us\n");
res = RESULT_FAILURE;
} else {
struct ast_frame f = { AST_FRAME_TEXT, 0 };
char text2send[256] = "";
text2send[0] = '\0';
while(tmparg < argc) {
strncat(text2send, argv[tmparg++], sizeof(text2send) - strlen(text2send) - 1);
strncat(text2send, " ", sizeof(text2send) - strlen(text2send) - 1);
}
text2send[strlen(text2send) - 1] = '\n';
f.data = text2send;
f.datalen = strlen(text2send) + 1;
grab_owner();
if (alsa.owner) {
ast_queue_frame(alsa.owner, &f);
f.frametype = AST_FRAME_CONTROL;
f.subclass = AST_CONTROL_ANSWER;
f.data = NULL;
f.datalen = 0;
ast_queue_frame(alsa.owner, &f);
ast_mutex_unlock(&alsa.owner->lock);
}
}
ast_mutex_unlock(&alsalock);
return res;
}
static char answer_usage[] =
"Usage: answer\n"
" Answers an incoming call on the console (ALSA) channel.\n";
static int console_hangup(int fd, int argc, char *argv[])
{
int res = RESULT_SUCCESS;
if (argc != 1)
return RESULT_SHOWUSAGE;
cursound = -1;
ast_mutex_lock(&alsalock);
if (!alsa.owner && !hookstate) {
ast_cli(fd, "No call to hangup up\n");
res = RESULT_FAILURE;
} else {
hookstate = 0;
grab_owner();
if (alsa.owner) {
ast_queue_hangup(alsa.owner);
ast_mutex_unlock(&alsa.owner->lock);
}
}
ast_mutex_unlock(&alsalock);
return res;
}
static char hangup_usage[] =
"Usage: hangup\n"
" Hangs up any call currently placed on the console.\n";
static int console_dial(int fd, int argc, char *argv[])
{
char tmp[256], *tmp2;
char *mye, *myc;
char *d;
int res = RESULT_SUCCESS;
if ((argc != 1) && (argc != 2))
return RESULT_SHOWUSAGE;
ast_mutex_lock(&alsalock);
if (alsa.owner) {
if (argc == 2) {
d = argv[1];
grab_owner();
if (alsa.owner) {
struct ast_frame f = { AST_FRAME_DTMF };
while(*d) {
f.subclass = *d;
ast_queue_frame(alsa.owner, &f);
d++;
}
ast_mutex_unlock(&alsa.owner->lock);
}
} else {
ast_cli(fd, "You're already in a call. You can use this only to dial digits until you hangup\n");
res = RESULT_FAILURE;
}
} else {
mye = exten;
myc = context;
if (argc == 2) {
char *stringp=NULL;
strncpy(tmp, argv[1], sizeof(tmp)-1);
stringp=tmp;
strsep(&stringp, "@");
tmp2 = strsep(&stringp, "@");
if (!ast_strlen_zero(tmp))
mye = tmp;
if (!ast_strlen_zero(tmp2))
myc = tmp2;
}
if (ast_exists_extension(NULL, myc, mye, 1, NULL)) {
strncpy(alsa.exten, mye, sizeof(alsa.exten)-1);
strncpy(alsa.context, myc, sizeof(alsa.context)-1);
hookstate = 1;
alsa_new(&alsa, AST_STATE_RINGING);
} else
ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc);
}
ast_mutex_unlock(&alsalock);
return res;
}
static char dial_usage[] =
"Usage: dial [extension[@context]]\n"
" Dials a given extension (and context if specified)\n";
static struct ast_cli_entry myclis[] = {
{ { "answer", NULL }, console_answer, "Answer an incoming console call", answer_usage },
{ { "hangup", NULL }, console_hangup, "Hangup a call on the console", hangup_usage },
{ { "dial", NULL }, console_dial, "Dial an extension on the console", dial_usage },
{ { "send", "text", NULL }, console_sendtext, "Send text to the remote device", sendtext_usage },
{ { "autoanswer", NULL }, console_autoanswer, "Sets/displays autoanswer", autoanswer_usage, autoanswer_complete }
};
static int load_module(void)
{
int res;
int x;
struct ast_config *cfg;
struct ast_variable *v;
/* Copy the default jb config over global_jbconf */
memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
strcpy(mohinterpret, "default");
if ((cfg = ast_config_load(config))) {
v = ast_variable_browse(cfg, "general");
for (; v; v = v->next) {
/* handle jb conf */
if (!ast_jb_read_conf(&global_jbconf, v->name, v->value))
continue;
if (!strcasecmp(v->name, "autoanswer"))
autoanswer = ast_true(v->value);
else if (!strcasecmp(v->name, "silencesuppression"))
silencesuppression = ast_true(v->value);
else if (!strcasecmp(v->name, "silencethreshold"))
silencethreshold = atoi(v->value);
else if (!strcasecmp(v->name, "context"))
ast_copy_string(context, v->value, sizeof(context));
else if (!strcasecmp(v->name, "language"))
ast_copy_string(language, v->value, sizeof(language));
else if (!strcasecmp(v->name, "extension"))
ast_copy_string(exten, v->value, sizeof(exten));
else if (!strcasecmp(v->name, "input_device"))
ast_copy_string(indevname, v->value, sizeof(indevname));
else if (!strcasecmp(v->name, "output_device"))
ast_copy_string(outdevname, v->value, sizeof(outdevname));
else if (!strcasecmp(v->name, "mohinterpret"))
ast_copy_string(mohinterpret, v->value, sizeof(mohinterpret));
}
ast_config_destroy(cfg);
}
res = pipe(sndcmd);
if (res) {
ast_log(LOG_ERROR, "Unable to create pipe\n");
return -1;
}
res = soundcard_init();
if (res < 0) {
if (option_verbose > 1) {
ast_verbose(VERBOSE_PREFIX_2 "No sound card detected -- console channel will be unavailable\n");
ast_verbose(VERBOSE_PREFIX_2 "Turn off ALSA support by adding 'noload=chan_alsa.so' in /etc/asterisk/modules.conf\n");
}
return 0;
}
res = ast_channel_register(&alsa_tech);
if (res < 0) {
ast_log(LOG_ERROR, "Unable to register channel class 'Console'\n");
return -1;
}
for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
ast_cli_register(myclis + x);
ast_pthread_create(&sthread, NULL, sound_thread, NULL);
#ifdef ALSA_MONITOR
if (alsa_monitor_start()) {
ast_log(LOG_ERROR, "Problem starting Monitoring\n");
}
#endif
return 0;
}
static int unload_module(void)
{
int x;
ast_channel_unregister(&alsa_tech);
for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
ast_cli_unregister(myclis + x);
if (alsa.icard)
snd_pcm_close(alsa.icard);
if (alsa.ocard)
snd_pcm_close(alsa.ocard);
if (sndcmd[0] > 0) {
close(sndcmd[0]);
close(sndcmd[1]);
}
if (alsa.owner)
ast_softhangup(alsa.owner, AST_SOFTHANGUP_APPUNLOAD);
if (alsa.owner)
return -1;
return 0;
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "ALSA Console Channel Driver");
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