In the script ./configure, AST_EXT_LIB_CHECK checks for external libraries. Some libraries do not specify all their dependencies and require additional shared libraries. In AST_EXT_LIB_CHECK, this is the fifth parameter. However, if a library is specified there, it must exist on the platform, because ./configure tries to compile/link/execute a small app using those statements. For example, the library libdl.so is Linux specific and does not exist on BSD-like platforms. Furthermore, no supported platform/version was found, which still (ever?) requires those additional libraries. Therefore, they were simply removed. Finally, this change adds the error code ESTRPIPE to the channel driver chan_alsa for those platforms which lack it, again for example NetBSD. ASTERISK-27720 Change-Id: I3b21f2135f6cbfac7590ccdc2df753257f426e0b
1044 lines
27 KiB
C
1044 lines
27 KiB
C
/*
|
|
* 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>
|
|
*
|
|
* \ingroup channel_drivers
|
|
*/
|
|
|
|
/*! \li \ref chan_alsa.c uses the configuration file \ref alsa.conf
|
|
* \addtogroup configuration_file
|
|
*/
|
|
|
|
/*! \page alsa.conf alsa.conf
|
|
* \verbinclude alsa.conf.sample
|
|
*/
|
|
|
|
/*** MODULEINFO
|
|
<depend>alsa</depend>
|
|
<support_level>extended</support_level>
|
|
***/
|
|
|
|
#include "asterisk.h"
|
|
|
|
#include <errno.h>
|
|
#ifndef ESTRPIPE
|
|
#define ESTRPIPE EPIPE
|
|
#endif
|
|
#include <fcntl.h>
|
|
#include <sys/ioctl.h>
|
|
#include <sys/time.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/channel.h"
|
|
#include "asterisk/module.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 "asterisk/poll-compat.h"
|
|
#include "asterisk/stasis_channels.h"
|
|
#include "asterisk/format_cache.h"
|
|
|
|
/*! Global jitterbuffer configuration - by default, jb is disabled
|
|
* \note Values shown here match the defaults shown in alsa.conf.sample */
|
|
static struct ast_jb_conf default_jbconf = {
|
|
.flags = 0,
|
|
.max_size = 200,
|
|
.resync_threshold = 1000,
|
|
.impl = "fixed",
|
|
.target_extra = 40,
|
|
};
|
|
static struct ast_jb_conf global_jbconf;
|
|
|
|
#define DEBUG 0
|
|
/* Which device to use */
|
|
#define ALSA_INDEV "default"
|
|
#define ALSA_OUTDEV "default"
|
|
#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 char indevname[50] = ALSA_INDEV;
|
|
static char outdevname[50] = ALSA_OUTDEV;
|
|
|
|
static int silencesuppression = 0;
|
|
static int silencethreshold = 1000;
|
|
|
|
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 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];
|
|
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. */
|
|
|
|
#define MAX_BUFFER_SIZE 100
|
|
|
|
/* File descriptors for sound device */
|
|
static int readdev = -1;
|
|
static int writedev = -1;
|
|
|
|
static int autoanswer = 1;
|
|
static int mute = 0;
|
|
static int noaudiocapture = 0;
|
|
|
|
static struct ast_channel *alsa_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause);
|
|
static int alsa_digit(struct ast_channel *c, char digit, unsigned int duration);
|
|
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, const 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 struct ast_channel_tech alsa_tech = {
|
|
.type = "Console",
|
|
.description = tdesc,
|
|
.requester = alsa_request,
|
|
.send_digit_end = 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 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;
|
|
snd_pcm_uframes_t buffer_size = 0;
|
|
unsigned int rate = DESIRED_RATE;
|
|
snd_pcm_uframes_t start_threshold, stop_threshold;
|
|
|
|
err = snd_pcm_open(&handle, dev, stream, SND_PCM_NONBLOCK);
|
|
if (err < 0) {
|
|
ast_log(LOG_ERROR, "snd_pcm_open failed: %s\n", snd_strerror(err));
|
|
return NULL;
|
|
} else {
|
|
ast_debug(1, "Opening device %s in %s mode\n", dev, (stream == SND_PCM_STREAM_CAPTURE) ? "read" : "write");
|
|
}
|
|
|
|
hwparams = ast_alloca(snd_pcm_hw_params_sizeof());
|
|
memset(hwparams, 0, snd_pcm_hw_params_sizeof());
|
|
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 %u\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(%lu frames) is bad: %s\n", period_size, snd_strerror(err));
|
|
else {
|
|
ast_debug(1, "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 %lu: %s\n", buffer_size, snd_strerror(err));
|
|
else {
|
|
ast_debug(1, "Buffer size is set to %d frames\n", err);
|
|
}
|
|
|
|
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));
|
|
|
|
swparams = ast_alloca(snd_pcm_sw_params_sizeof());
|
|
memset(swparams, 0, snd_pcm_sw_params_sizeof());
|
|
snd_pcm_sw_params_current(handle, swparams);
|
|
|
|
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));
|
|
|
|
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));
|
|
|
|
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_debug(1, "Can't handle more than one device\n");
|
|
}
|
|
|
|
snd_pcm_poll_descriptors(handle, &pfd, err);
|
|
ast_debug(1, "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)
|
|
{
|
|
if (!noaudiocapture) {
|
|
alsa.icard = alsa_card_init(indevname, SND_PCM_STREAM_CAPTURE);
|
|
if (!alsa.icard) {
|
|
ast_log(LOG_ERROR, "Problem opening alsa capture device\n");
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
alsa.ocard = alsa_card_init(outdevname, SND_PCM_STREAM_PLAYBACK);
|
|
|
|
if (!alsa.ocard) {
|
|
ast_log(LOG_ERROR, "Problem opening ALSA playback device\n");
|
|
return -1;
|
|
}
|
|
|
|
return writedev;
|
|
}
|
|
|
|
static int alsa_digit(struct ast_channel *c, char digit, unsigned int duration)
|
|
{
|
|
ast_mutex_lock(&alsalock);
|
|
ast_verbose(" << Console Received digit %c of duration %u ms >> \n",
|
|
digit, duration);
|
|
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_channel_trylock(alsa.owner)) {
|
|
DEADLOCK_AVOIDANCE(&alsalock);
|
|
}
|
|
}
|
|
|
|
static int alsa_call(struct ast_channel *c, const char *dest, int timeout)
|
|
{
|
|
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");
|
|
if (mute) {
|
|
ast_verbose( " << Muted >> \n" );
|
|
}
|
|
grab_owner();
|
|
if (alsa.owner) {
|
|
f.subclass.integer = AST_CONTROL_ANSWER;
|
|
ast_queue_frame(alsa.owner, &f);
|
|
ast_channel_unlock(alsa.owner);
|
|
}
|
|
} else {
|
|
ast_verbose(" << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
|
|
grab_owner();
|
|
if (alsa.owner) {
|
|
f.subclass.integer = AST_CONTROL_RINGING;
|
|
ast_queue_frame(alsa.owner, &f);
|
|
ast_channel_unlock(alsa.owner);
|
|
ast_indicate(alsa.owner, AST_CONTROL_RINGING);
|
|
}
|
|
}
|
|
if (!noaudiocapture) {
|
|
snd_pcm_prepare(alsa.icard);
|
|
snd_pcm_start(alsa.icard);
|
|
}
|
|
ast_mutex_unlock(&alsalock);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int alsa_answer(struct ast_channel *c)
|
|
{
|
|
ast_mutex_lock(&alsalock);
|
|
ast_verbose(" << Console call has been answered >> \n");
|
|
ast_setstate(c, AST_STATE_UP);
|
|
if (!noaudiocapture) {
|
|
snd_pcm_prepare(alsa.icard);
|
|
snd_pcm_start(alsa.icard);
|
|
}
|
|
ast_mutex_unlock(&alsalock);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int alsa_hangup(struct ast_channel *c)
|
|
{
|
|
ast_mutex_lock(&alsalock);
|
|
ast_channel_tech_pvt_set(c, NULL);
|
|
alsa.owner = NULL;
|
|
ast_verbose(" << Hangup on console >> \n");
|
|
ast_module_unref(ast_module_info->self);
|
|
hookstate = 0;
|
|
if (!noaudiocapture) {
|
|
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 res = 0;
|
|
/* size_t frames = 0; */
|
|
snd_pcm_state_t state;
|
|
|
|
ast_mutex_lock(&alsalock);
|
|
|
|
/* 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.ptr, f->datalen);
|
|
len += f->datalen;
|
|
state = snd_pcm_state(alsa.ocard);
|
|
if (state == SND_PCM_STATE_XRUN)
|
|
snd_pcm_prepare(alsa.ocard);
|
|
while ((res = snd_pcm_writei(alsa.ocard, sizbuf, len / 2)) == -EAGAIN) {
|
|
usleep(1);
|
|
}
|
|
if (res == -EPIPE) {
|
|
#if DEBUG
|
|
ast_debug(1, "XRUN write\n");
|
|
#endif
|
|
snd_pcm_prepare(alsa.ocard);
|
|
while ((res = snd_pcm_writei(alsa.ocard, sizbuf, len / 2)) == -EAGAIN) {
|
|
usleep(1);
|
|
}
|
|
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);
|
|
|
|
return res >= 0 ? 0 : 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;
|
|
|
|
ast_mutex_lock(&alsalock);
|
|
f.frametype = AST_FRAME_NULL;
|
|
f.subclass.integer = 0;
|
|
f.samples = 0;
|
|
f.datalen = 0;
|
|
f.data.ptr = NULL;
|
|
f.offset = 0;
|
|
f.src = "Console";
|
|
f.mallocd = 0;
|
|
f.delivery.tv_sec = 0;
|
|
f.delivery.tv_usec = 0;
|
|
|
|
if (noaudiocapture) {
|
|
/* Return null frame to asterisk*/
|
|
ast_mutex_unlock(&alsalock);
|
|
return &f;
|
|
}
|
|
|
|
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));
|
|
}
|
|
|
|
/* Return NULL frame on error */
|
|
if (r < 0) {
|
|
ast_mutex_unlock(&alsalock);
|
|
return &f;
|
|
}
|
|
|
|
/* Update positions */
|
|
readpos += r;
|
|
left -= r;
|
|
|
|
if (readpos >= FRAME_SIZE) {
|
|
/* A real frame */
|
|
readpos = 0;
|
|
left = FRAME_SIZE;
|
|
if (ast_channel_state(chan) != AST_STATE_UP) {
|
|
/* Don't transmit unless it's up */
|
|
ast_mutex_unlock(&alsalock);
|
|
return &f;
|
|
}
|
|
if (mute) {
|
|
/* Don't transmit if muted */
|
|
ast_mutex_unlock(&alsalock);
|
|
return &f;
|
|
}
|
|
|
|
f.frametype = AST_FRAME_VOICE;
|
|
f.subclass.format = ast_format_slin;
|
|
f.samples = FRAME_SIZE;
|
|
f.datalen = FRAME_SIZE * 2;
|
|
f.data.ptr = buf;
|
|
f.offset = AST_FRIENDLY_OFFSET;
|
|
f.src = "Console";
|
|
f.mallocd = 0;
|
|
|
|
}
|
|
ast_mutex_unlock(&alsalock);
|
|
|
|
return &f;
|
|
}
|
|
|
|
static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
|
|
{
|
|
struct chan_alsa_pvt *p = ast_channel_tech_pvt(newchan);
|
|
|
|
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:
|
|
case AST_CONTROL_CONGESTION:
|
|
case AST_CONTROL_RINGING:
|
|
case AST_CONTROL_INCOMPLETE:
|
|
case AST_CONTROL_PVT_CAUSE_CODE:
|
|
case -1:
|
|
res = -1; /* Ask for inband indications */
|
|
break;
|
|
case AST_CONTROL_PROGRESS:
|
|
case AST_CONTROL_PROCEEDING:
|
|
case AST_CONTROL_VIDUPDATE:
|
|
case AST_CONTROL_SRCUPDATE:
|
|
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, ast_channel_name(chan));
|
|
res = -1;
|
|
}
|
|
|
|
ast_mutex_unlock(&alsalock);
|
|
|
|
return res;
|
|
}
|
|
|
|
static struct ast_channel *alsa_new(struct chan_alsa_pvt *p, int state, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor)
|
|
{
|
|
struct ast_channel *tmp = NULL;
|
|
|
|
if (!(tmp = ast_channel_alloc(1, state, 0, 0, "", p->exten, p->context, assignedids, requestor, 0, "ALSA/%s", indevname)))
|
|
return NULL;
|
|
|
|
ast_channel_stage_snapshot(tmp);
|
|
|
|
ast_channel_tech_set(tmp, &alsa_tech);
|
|
ast_channel_set_fd(tmp, 0, readdev);
|
|
ast_channel_set_readformat(tmp, ast_format_slin);
|
|
ast_channel_set_writeformat(tmp, ast_format_slin);
|
|
ast_channel_nativeformats_set(tmp, alsa_tech.capabilities);
|
|
|
|
ast_channel_tech_pvt_set(tmp, p);
|
|
if (!ast_strlen_zero(p->context))
|
|
ast_channel_context_set(tmp, p->context);
|
|
if (!ast_strlen_zero(p->exten))
|
|
ast_channel_exten_set(tmp, p->exten);
|
|
if (!ast_strlen_zero(language))
|
|
ast_channel_language_set(tmp, language);
|
|
p->owner = tmp;
|
|
ast_module_ref(ast_module_info->self);
|
|
ast_jb_configure(tmp, &global_jbconf);
|
|
|
|
ast_channel_stage_snapshot_done(tmp);
|
|
ast_channel_unlock(tmp);
|
|
|
|
if (state != AST_STATE_DOWN) {
|
|
if (ast_pbx_start(tmp)) {
|
|
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", ast_channel_name(tmp));
|
|
ast_hangup(tmp);
|
|
tmp = NULL;
|
|
}
|
|
}
|
|
|
|
return tmp;
|
|
}
|
|
|
|
static struct ast_channel *alsa_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
|
|
{
|
|
struct ast_channel *tmp = NULL;
|
|
|
|
if (ast_format_cap_iscompatible_format(cap, ast_format_slin) == AST_FORMAT_CMP_NOT_EQUAL) {
|
|
struct ast_str *codec_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
|
|
ast_log(LOG_NOTICE, "Asked to get a channel of format '%s'\n", ast_format_cap_get_names(cap, &codec_buf));
|
|
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 if (!(tmp = alsa_new(&alsa, AST_STATE_DOWN, assignedids, requestor))) {
|
|
ast_log(LOG_WARNING, "Unable to create new ALSA channel\n");
|
|
}
|
|
|
|
ast_mutex_unlock(&alsalock);
|
|
|
|
return tmp;
|
|
}
|
|
|
|
static char *console_autoanswer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
char *res = CLI_SUCCESS;
|
|
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "console autoanswer [on|off]";
|
|
e->usage =
|
|
"Usage: console 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";
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
if ((a->argc != 2) && (a->argc != 3))
|
|
return CLI_SHOWUSAGE;
|
|
|
|
ast_mutex_lock(&alsalock);
|
|
if (a->argc == 2) {
|
|
ast_cli(a->fd, "Auto answer is %s.\n", autoanswer ? "on" : "off");
|
|
} else {
|
|
if (!strcasecmp(a->argv[2], "on"))
|
|
autoanswer = -1;
|
|
else if (!strcasecmp(a->argv[2], "off"))
|
|
autoanswer = 0;
|
|
else
|
|
res = CLI_SHOWUSAGE;
|
|
}
|
|
ast_mutex_unlock(&alsalock);
|
|
|
|
return res;
|
|
}
|
|
|
|
static char *console_answer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
char *res = CLI_SUCCESS;
|
|
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "console answer";
|
|
e->usage =
|
|
"Usage: console answer\n"
|
|
" Answers an incoming call on the console (ALSA) channel.\n";
|
|
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
if (a->argc != 2)
|
|
return CLI_SHOWUSAGE;
|
|
|
|
ast_mutex_lock(&alsalock);
|
|
|
|
if (!alsa.owner) {
|
|
ast_cli(a->fd, "No one is calling us\n");
|
|
res = CLI_FAILURE;
|
|
} else {
|
|
if (mute) {
|
|
ast_verbose( " << Muted >> \n" );
|
|
}
|
|
hookstate = 1;
|
|
grab_owner();
|
|
if (alsa.owner) {
|
|
ast_queue_control(alsa.owner, AST_CONTROL_ANSWER);
|
|
ast_channel_unlock(alsa.owner);
|
|
}
|
|
}
|
|
|
|
if (!noaudiocapture) {
|
|
snd_pcm_prepare(alsa.icard);
|
|
snd_pcm_start(alsa.icard);
|
|
}
|
|
|
|
ast_mutex_unlock(&alsalock);
|
|
|
|
return res;
|
|
}
|
|
|
|
static char *console_sendtext(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
int tmparg = 3;
|
|
char *res = CLI_SUCCESS;
|
|
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "console send text";
|
|
e->usage =
|
|
"Usage: console send text <message>\n"
|
|
" Sends a text message for display on the remote terminal.\n";
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
if (a->argc < 3)
|
|
return CLI_SHOWUSAGE;
|
|
|
|
ast_mutex_lock(&alsalock);
|
|
|
|
if (!alsa.owner) {
|
|
ast_cli(a->fd, "No channel active\n");
|
|
res = CLI_FAILURE;
|
|
} else {
|
|
struct ast_frame f = { AST_FRAME_TEXT };
|
|
char text2send[256] = "";
|
|
|
|
while (tmparg < a->argc) {
|
|
strncat(text2send, a->argv[tmparg++], sizeof(text2send) - strlen(text2send) - 1);
|
|
strncat(text2send, " ", sizeof(text2send) - strlen(text2send) - 1);
|
|
}
|
|
|
|
text2send[strlen(text2send) - 1] = '\n';
|
|
f.data.ptr = text2send;
|
|
f.datalen = strlen(text2send) + 1;
|
|
grab_owner();
|
|
if (alsa.owner) {
|
|
ast_queue_frame(alsa.owner, &f);
|
|
ast_queue_control(alsa.owner, AST_CONTROL_ANSWER);
|
|
ast_channel_unlock(alsa.owner);
|
|
}
|
|
}
|
|
|
|
ast_mutex_unlock(&alsalock);
|
|
|
|
return res;
|
|
}
|
|
|
|
static char *console_hangup(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
char *res = CLI_SUCCESS;
|
|
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "console hangup";
|
|
e->usage =
|
|
"Usage: console hangup\n"
|
|
" Hangs up any call currently placed on the console.\n";
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
|
|
if (a->argc != 2)
|
|
return CLI_SHOWUSAGE;
|
|
|
|
ast_mutex_lock(&alsalock);
|
|
|
|
if (!alsa.owner && !hookstate) {
|
|
ast_cli(a->fd, "No call to hangup\n");
|
|
res = CLI_FAILURE;
|
|
} else {
|
|
hookstate = 0;
|
|
grab_owner();
|
|
if (alsa.owner) {
|
|
ast_queue_hangup_with_cause(alsa.owner, AST_CAUSE_NORMAL_CLEARING);
|
|
ast_channel_unlock(alsa.owner);
|
|
}
|
|
}
|
|
|
|
ast_mutex_unlock(&alsalock);
|
|
|
|
return res;
|
|
}
|
|
|
|
static char *console_dial(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
char tmp[256], *tmp2;
|
|
char *mye, *myc;
|
|
const char *d;
|
|
char *res = CLI_SUCCESS;
|
|
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "console dial";
|
|
e->usage =
|
|
"Usage: console dial [extension[@context]]\n"
|
|
" Dials a given extension (and context if specified)\n";
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
if ((a->argc != 2) && (a->argc != 3))
|
|
return CLI_SHOWUSAGE;
|
|
|
|
ast_mutex_lock(&alsalock);
|
|
|
|
if (alsa.owner) {
|
|
if (a->argc == 3) {
|
|
if (alsa.owner) {
|
|
for (d = a->argv[2]; *d; d++) {
|
|
struct ast_frame f = { .frametype = AST_FRAME_DTMF, .subclass.integer = *d };
|
|
|
|
ast_queue_frame(alsa.owner, &f);
|
|
}
|
|
}
|
|
} else {
|
|
ast_cli(a->fd, "You're already in a call. You can use this only to dial digits until you hangup\n");
|
|
res = CLI_FAILURE;
|
|
}
|
|
} else {
|
|
mye = exten;
|
|
myc = context;
|
|
if (a->argc == 3) {
|
|
char *stringp = NULL;
|
|
|
|
ast_copy_string(tmp, a->argv[2], sizeof(tmp));
|
|
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)) {
|
|
ast_copy_string(alsa.exten, mye, sizeof(alsa.exten));
|
|
ast_copy_string(alsa.context, myc, sizeof(alsa.context));
|
|
hookstate = 1;
|
|
alsa_new(&alsa, AST_STATE_RINGING, NULL, NULL);
|
|
} else
|
|
ast_cli(a->fd, "No such extension '%s' in context '%s'\n", mye, myc);
|
|
}
|
|
|
|
ast_mutex_unlock(&alsalock);
|
|
|
|
return res;
|
|
}
|
|
|
|
static char *console_mute(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
int toggle = 0;
|
|
char *res = CLI_SUCCESS;
|
|
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "console {mute|unmute} [toggle]";
|
|
e->usage =
|
|
"Usage: console {mute|unmute} [toggle]\n"
|
|
" Mute/unmute the microphone.\n";
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
|
|
if (a->argc > 3) {
|
|
return CLI_SHOWUSAGE;
|
|
}
|
|
|
|
if (a->argc == 3) {
|
|
if (strcasecmp(a->argv[2], "toggle"))
|
|
return CLI_SHOWUSAGE;
|
|
toggle = 1;
|
|
}
|
|
|
|
if (a->argc < 2) {
|
|
return CLI_SHOWUSAGE;
|
|
}
|
|
|
|
if (!strcasecmp(a->argv[1], "mute")) {
|
|
mute = toggle ? !mute : 1;
|
|
} else if (!strcasecmp(a->argv[1], "unmute")) {
|
|
mute = toggle ? !mute : 0;
|
|
} else {
|
|
return CLI_SHOWUSAGE;
|
|
}
|
|
|
|
ast_cli(a->fd, "Console mic is %s\n", mute ? "off" : "on");
|
|
|
|
return res;
|
|
}
|
|
|
|
static struct ast_cli_entry cli_alsa[] = {
|
|
AST_CLI_DEFINE(console_answer, "Answer an incoming console call"),
|
|
AST_CLI_DEFINE(console_hangup, "Hangup a call on the console"),
|
|
AST_CLI_DEFINE(console_dial, "Dial an extension on the console"),
|
|
AST_CLI_DEFINE(console_sendtext, "Send text to the remote device"),
|
|
AST_CLI_DEFINE(console_autoanswer, "Sets/displays autoanswer"),
|
|
AST_CLI_DEFINE(console_mute, "Disable/Enable mic input"),
|
|
};
|
|
|
|
static int unload_module(void)
|
|
{
|
|
ast_channel_unregister(&alsa_tech);
|
|
ast_cli_unregister_multiple(cli_alsa, ARRAY_LEN(cli_alsa));
|
|
|
|
if (alsa.icard)
|
|
snd_pcm_close(alsa.icard);
|
|
if (alsa.ocard)
|
|
snd_pcm_close(alsa.ocard);
|
|
if (alsa.owner)
|
|
ast_softhangup(alsa.owner, AST_SOFTHANGUP_APPUNLOAD);
|
|
if (alsa.owner)
|
|
return -1;
|
|
|
|
ao2_cleanup(alsa_tech.capabilities);
|
|
alsa_tech.capabilities = NULL;
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*!
|
|
* \brief Load the module
|
|
*
|
|
* Module loading including tests for configuration or dependencies.
|
|
* This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE,
|
|
* or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails
|
|
* tests return AST_MODULE_LOAD_FAILURE. If the module can not load the
|
|
* configuration file or other non-critical problem return
|
|
* AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS.
|
|
*/
|
|
static int load_module(void)
|
|
{
|
|
struct ast_config *cfg;
|
|
struct ast_variable *v;
|
|
struct ast_flags config_flags = { 0 };
|
|
|
|
if (!(alsa_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
ast_format_cap_append(alsa_tech.capabilities, ast_format_slin, 0);
|
|
|
|
/* 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, config_flags))) {
|
|
ast_log(LOG_ERROR, "Unable to read ALSA configuration file %s. Aborting.\n", config);
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
} else if (cfg == CONFIG_STATUS_FILEINVALID) {
|
|
ast_log(LOG_ERROR, "%s is in an invalid format. Aborting.\n", config);
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
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, "mute")) {
|
|
mute = ast_true(v->value);
|
|
} else if (!strcasecmp(v->name, "noaudiocapture")) {
|
|
noaudiocapture = 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);
|
|
|
|
if (soundcard_init() < 0) {
|
|
ast_verb(2, "No sound card detected -- console channel will be unavailable\n");
|
|
ast_verb(2, "Turn off ALSA support by adding 'noload=chan_alsa.so' in /etc/asterisk/modules.conf\n");
|
|
unload_module();
|
|
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
if (ast_channel_register(&alsa_tech)) {
|
|
ast_log(LOG_ERROR, "Unable to register channel class 'Console'\n");
|
|
unload_module();
|
|
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
ast_cli_register_multiple(cli_alsa, ARRAY_LEN(cli_alsa));
|
|
|
|
return AST_MODULE_LOAD_SUCCESS;
|
|
}
|
|
|
|
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "ALSA Console Channel Driver",
|
|
.support_level = AST_MODULE_SUPPORT_EXTENDED,
|
|
.load = load_module,
|
|
.unload = unload_module,
|
|
.load_pri = AST_MODPRI_CHANNEL_DRIVER,
|
|
);
|