+++ /dev/null
-/*
- * 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>
- <defaultenabled>no</defaultenabled>
- <support_level>deprecated</support_level>
- <replacement>chan_console</replacement>
- <deprecated_in>19</deprecated_in>
- <removed_in>21</removed_in>
- ***/
-
-#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_DEPRECATED,
- .load = load_module,
- .unload = unload_module,
- .load_pri = AST_MODPRI_CHANNEL_DRIVER,
-);
projects / thirdparty / asterisk.git / commitdiff
