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>
*
* \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"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#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"
Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-18 02:28:55 +00:00
#include "asterisk/poll-compat.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_channel *requestor, const char *data, int *cause);
2007-01-19 18:06:03 +00:00
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 %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_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 %ld: %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;
}
2007-01-19 18:06:03 +00:00
static int alsa_digit(struct ast_channel *c, char digit, unsigned int duration)
{
ast_mutex_lock(&alsalock);
2007-01-19 18:06:03 +00:00
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));
}
/* 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;
ast_format_set(&f.subclass.format, AST_FORMAT_SLINEAR, 0);
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:
Merged revisions 335078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines Merged revisions 335064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-09 16:28:23 +00:00
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:
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
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 char *linkedid)
{
struct ast_channel *tmp = NULL;
if (!(tmp = ast_channel_alloc(1, state, 0, 0, "", p->exten, p->context, linkedid, 0, "ALSA/%s", indevname)))
return NULL;
ast_channel_tech_set(tmp, &alsa_tech);
ast_channel_set_fd(tmp, 0, readdev);
ast_format_set(ast_channel_readformat(tmp), AST_FORMAT_SLINEAR, 0);
ast_format_set(ast_channel_writeformat(tmp), AST_FORMAT_SLINEAR, 0);
ast_format_cap_add(ast_channel_nativeformats(tmp), ast_channel_writeformat(tmp));
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);
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(tmp)) {
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
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_channel *requestor, const char *data, int *cause)
{
struct ast_format tmpfmt;
char buf[256];
struct ast_channel *tmp = NULL;
ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 0);
if (!(ast_format_cap_iscompatible(cap, &tmpfmt))) {
ast_log(LOG_NOTICE, "Asked to get a channel of format '%s'\n", ast_getformatname_multiple(buf, sizeof(buf), cap));
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, requestor ? ast_channel_linkedid(requestor) : NULL))) {
ast_log(LOG_WARNING, "Unable to create new ALSA channel\n");
}
ast_mutex_unlock(&alsalock);
return tmp;
}
static char *autoanswer_complete(const char *line, const char *word, int pos, int state)
{
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 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";
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 autoanswer_complete(a->line, a->word, a->pos, a->n);
}
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);
} 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"),
};
/*!
* \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 };
struct ast_format tmpfmt;
if (!(alsa_tech.capabilities = ast_format_cap_alloc())) {
return AST_MODULE_LOAD_DECLINE;
}
ast_format_cap_add(alsa_tech.capabilities, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR, 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");
return AST_MODULE_LOAD_DECLINE;
}
if (ast_channel_register(&alsa_tech)) {
ast_log(LOG_ERROR, "Unable to register channel class 'Console'\n");
return AST_MODULE_LOAD_FAILURE;
}
ast_cli_register_multiple(cli_alsa, ARRAY_LEN(cli_alsa));
return AST_MODULE_LOAD_SUCCESS;
}
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;
alsa_tech.capabilities = ast_format_cap_destroy(alsa_tech.capabilities);
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "ALSA Console Channel Driver",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_CHANNEL_DRIVER,
);