On Tue, Jun 9, 2009 at 4:07 AM, Takashi Iwai<tiwai@xxxxxxx> wrote:
> At Mon, 8 Jun 2009 15:56:38 -0400,
> Robert Krakora wrote:
>>
>> Takashi:
>>
>> I applied your Speex echo cancellation patch. I had to modify it
>> since the 'buf_out' buffer was not being malloc'd. Attached are the
>> files affected by your patch. The echo cancellation seems to work
>> pretty well. However, turning on 'denoise' seems to cause audio to
>> drop in and out at a very periodic rate. Do you have any example of
>> how to set up all of the parameters? A tutorial or something? Thanks
>> again.
>
> No idea, I just wrote it up blindly :)
> Can the preprocessor be dropped, possibly, like the patch below?
>
>
> Takashi
>
> ---
> diff --git a/doc/speexdsp.txt b/doc/speexdsp.txt
> index 875fc19..5b5e5a0 100644
> --- a/doc/speexdsp.txt
> +++ b/doc/speexdsp.txt
> @@ -12,7 +12,7 @@ using libspeex DSP API. You can use the plugin with the plugin type
>
> Then record like
>
> - % arecord -fdat -c1 -Dplug:speex foo.wav
> + % arecord -fdat -c1 -Dplug:my_pcm foo.wav
>
> so that you'll get 48kHz mono stream with the denoising effect.
>
> @@ -44,6 +44,15 @@ The following parameters can be set optionally:
>
> A boolean value to enable/disable dereverb function. Default is no.
>
> +* echo
> +
> + A boolean value to enable/disable echo-cancellation function.
> + Default is no.
> +
> +* filter_length
> +
> + Number of samples of echo to cancel. As default it's 256.
> +
> For example, you can enable agc like
>
> pcm.my_pcm {
> diff --git a/speex/pcm_speex.c b/speex/pcm_speex.c
> index 7bb9213..757a400 100644
> --- a/speex/pcm_speex.c
> +++ b/speex/pcm_speex.c
> @@ -1,5 +1,5 @@
> /*
> - * Speex preprocess plugin
> + * Speex DSP plugin
> *
> * Copyright (c) 2009 by Takashi Iwai <tiwai@xxxxxxx>
> *
> @@ -21,12 +21,15 @@
> #include <alsa/asoundlib.h>
> #include <alsa/pcm_external.h>
> #include <speex/speex_preprocess.h>
> +#include <speex/speex_echo.h>
>
> -/* preprocessing parameters */
> +/* DSP parameters */
> struct spx_parms {
> int frames;
> int denoise;
> int agc;
> + int echo;
> + int filter_length;
> float agc_level;
> int dereverb;
> float dereverb_decay;
> @@ -38,7 +41,9 @@ typedef struct {
> struct spx_parms parms;
> /* instance and intermedate buffer */
> SpeexPreprocessState *state;
> + SpeexEchoState *echo_state;
> short *buf;
> + short *outbuf;
> /* running states */
> unsigned int filled;
> unsigned int processed;
> @@ -64,6 +69,18 @@ spx_transfer(snd_pcm_extplug_t *ext,
> short *src = area_addr(src_areas, src_offset);
> short *dst = area_addr(dst_areas, dst_offset);
> unsigned int count = size;
> + short *databuf;
> +
> + if (!spx->state && !spx->echo_state) {
> + /* no DSP processing */
> + memcpy(dst, src, count * 2);
> + return size;
> + }
> +
> + if (spx->echo_state)
> + databuf = spx->outbuf;
> + else
> + databuf = spx->buf;
>
> while (count > 0) {
> unsigned int chunk;
> @@ -72,14 +89,20 @@ spx_transfer(snd_pcm_extplug_t *ext,
> else
> chunk = count;
> if (spx->processed)
> - memcpy(dst, spx->buf + spx->filled, chunk * 2);
> + memcpy(dst, databuf + spx->filled, chunk * 2);
> else
> memset(dst, 0, chunk * 2);
> dst += chunk;
> memcpy(spx->buf + spx->filled, src, chunk * 2);
> spx->filled += chunk;
> if (spx->filled == spx->parms.frames) {
> - speex_preprocess_run(spx->state, spx->buf);
> + if (spx->echo_state)
> + speex_echo_capture(spx->echo_state, spx->buf,
> + spx->outbuf);
> + if (spx->state)
> + speex_preprocess_run(spx->state, databuf);
> + if (spx->echo_state)
> + speex_echo_playback(spx->echo_state, databuf);
> spx->processed = 1;
> spx->filled = 0;
> }
> @@ -94,6 +117,9 @@ static int spx_init(snd_pcm_extplug_t *ext)
> {
> snd_pcm_speex_t *spx = (snd_pcm_speex_t *)ext;
>
> + spx->filled = 0;
> + spx->processed = 0;
> +
> if (!spx->buf) {
> spx->buf = malloc(spx->parms.frames * 2);
> if (!spx->buf)
> @@ -101,12 +127,43 @@ static int spx_init(snd_pcm_extplug_t *ext)
> }
> memset(spx->buf, 0, spx->parms.frames * 2);
>
> - if (spx->state)
> + if (!spx->outbuf) {
> + spx->outbuf = malloc(spx->parms.frames * 2);
> + if (!spx->outbuf)
> + return -ENOMEM;
> + }
> + memset(spx->outbuf, 0, spx->parms.frames * 2);
> +
> + if (spx->state) {
> speex_preprocess_state_destroy(spx->state);
> + spx->state = NULL;
> + }
> + if (spx->echo_state) {
> + speex_echo_state_destroy(spx->echo_state);
> + spx->echo_state = NULL;
> + }
> +
> + if (spx->parms.echo) {
> + spx->echo_state = speex_echo_state_init(spx->parms.frames,
> + spx->parms.filter_length);
> + if (!spx->echo_state)
> + return -EIO;
> + speex_echo_ctl(spx->echo_state, SPEEX_ECHO_SET_SAMPLING_RATE,
> + &spx->ext.rate);
> + }
> +
> + /* no preprocessor? */
> + if (!spx->parms.denoise && !spx->parms.agc && !spx->parms.dereverb)
> + return 0;
> +
> spx->state = speex_preprocess_state_init(spx->parms.frames,
> spx->ext.rate);
> if (!spx->state)
> return -EIO;
> + if (spx->echo_state)
> + speex_preprocess_ctl(spx->state,
> + SPEEX_PREPROCESS_SET_ECHO_STATE,
> + spx->echo_state);
>
> speex_preprocess_ctl(spx->state, SPEEX_PREPROCESS_SET_DENOISE,
> &spx->parms.denoise);
> @@ -120,18 +177,18 @@ static int spx_init(snd_pcm_extplug_t *ext)
> &spx->parms.dereverb_decay);
> speex_preprocess_ctl(spx->state, SPEEX_PREPROCESS_SET_DEREVERB_LEVEL,
> &spx->parms.dereverb_level);
> -
> - spx->filled = 0;
> - spx->processed = 0;
> return 0;
> }
>
> static int spx_close(snd_pcm_extplug_t *ext)
> {
> snd_pcm_speex_t *spx = (snd_pcm_speex_t *)ext;
> + free(spx->outbuf);
> free(spx->buf);
> if (spx->state)
> speex_preprocess_state_destroy(spx->state);
> + if (spx->echo_state)
> + speex_echo_state_destroy(spx->echo_state);
> return 0;
> }
>
> @@ -205,6 +262,8 @@ SND_PCM_PLUGIN_DEFINE_FUNC(speex)
> .dereverb = 0,
> .dereverb_decay = 0,
> .dereverb_level = 0,
> + .echo = 0,
> + .filter_length = 256,
> };
>
> snd_config_for_each(i, next, conf) {
> @@ -242,6 +301,13 @@ SND_PCM_PLUGIN_DEFINE_FUNC(speex)
> &parms.dereverb_level);
> if (err)
> goto ok;
> + err = get_bool_parm(n, id, "echo", &parms.echo);
> + if (err)
> + goto ok;
> + err = get_int_parm(n, id, "filter_length",
> + &parms.filter_length);
> + if (err)
> + goto ok;
> SNDERR("Unknown field %s", id);
> err = -EINVAL;
> ok:
> @@ -259,7 +325,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(speex)
> return -ENOMEM;
>
> spx->ext.version = SND_PCM_EXTPLUG_VERSION;
> - spx->ext.name = "Speex Denoise Plugin";
> + spx->ext.name = "Speex DSP Plugin";
> spx->ext.callback = &speex_callback;
> spx->ext.private_data = spx;
> spx->parms = parms;
>
>
Takashi:
Finally, I have your patch applied successfully and all seems to be
working well. See the attached files.
Best Regards,
--
Rob Krakora
Senior Software Engineer
MessageNet Systems
101 East Carmel Dr. Suite 105
Carmel, IN 46032
(317)566-1677 Ext. 206
(317)663-0808 Fax
/* * Asterisk -- An open source telephony toolkit. * * Copyright (C) 1999 - 2005, Digium, Inc. * * By Matthew Fredrickson <creslin@xxxxxxxxxx> * * 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@xxxxxxxxxx> * * \par See also * \arg Config_alsa * * \ingroup channel_drivers */ /*** MODULEINFO <depend>asound</depend> ***/ #include "asterisk.h" ASTERISK_FILE_VERSION(__FILE__, "$Revision: 167095 $") #include <unistd.h> #include <fcntl.h> #include <errno.h> #include <sys/ioctl.h> #include <sys/time.h> #include <string.h> #include <stdlib.h> #include <stdio.h> #define ALSA_PCM_NEW_HW_PARAMS_API #define ALSA_PCM_NEW_SW_PARAMS_API #include <alsa/asoundlib.h> #include "asterisk/frame.h" #include "asterisk/logger.h" #include "asterisk/channel.h" #include "asterisk/module.h" #include "asterisk/options.h" #include "asterisk/pbx.h" #include "asterisk/config.h" #include "asterisk/cli.h" #include "asterisk/utils.h" #include "asterisk/causes.h" #include "asterisk/endian.h" #include "asterisk/stringfields.h" #include "asterisk/abstract_jb.h" #include "asterisk/musiconhold.h" #include "busy_tone.h" #include "ring_tone.h" #include "ring10.h" #include "answer.h" #ifdef ALSA_MONITOR #include "alsa-monitor.h" #endif /*! Global jitterbuffer configuration - by default, jb is disabled */ static struct ast_jb_conf default_jbconf = { .flags = 0, .max_size = -1, .resync_threshold = -1, .impl = "" }; static struct ast_jb_conf global_jbconf; #define DEBUG 0 /* Which device to use */ #define ALSA_INDEV "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; #if 0 static struct timeval lasttime; #endif 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 short silence[FRAME_SIZE] = { 0, }; struct sound { int ind; short *data; int datalen; int samplen; int silencelen; int repeat; }; static struct sound sounds[] = { {AST_CONTROL_RINGING, ringtone, sizeof(ringtone) / 2, 16000, 32000, 1}, {AST_CONTROL_BUSY, busy, sizeof(busy) / 2, 4000, 4000, 1}, {AST_CONTROL_CONGESTION, busy, sizeof(busy) / 2, 2000, 2000, 1}, {AST_CONTROL_RING, ring10, sizeof(ring10) / 2, 16000, 32000, 1}, {AST_CONTROL_ANSWER, answer, sizeof(answer) / 2, 2200, 0, 0}, }; /* Sound command pipe */ static int sndcmd[2]; static struct chan_alsa_pvt { /* We only have one ALSA structure -- near sighted perhaps, but it keeps this driver as simple as possible -- as it should be. */ struct ast_channel *owner; char exten[AST_MAX_EXTENSION]; char context[AST_MAX_CONTEXT]; #if 0 snd_pcm_t *card; #endif snd_pcm_t *icard, *ocard; } alsa; /* Number of buffers... Each is FRAMESIZE/8 ms long. For example with 160 sample frames, and a buffer size of 3, we have a 60ms buffer, usually plenty. */ pthread_t sthread; #define MAX_BUFFER_SIZE 100 /* File descriptors for sound device */ static int readdev = -1; static int writedev = -1; static int autoanswer = 1; static int cursound = -1; static int sampsent = 0; static int silencelen = 0; static int offset = 0; static int nosound = 0; /* ZZ */ static struct ast_channel *alsa_request(const char *type, int format, void *data, int *cause); static int alsa_digit(struct ast_channel *c, char digit, 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, char *dest, int timeout); static int alsa_write(struct ast_channel *chan, struct ast_frame *f); static int alsa_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen); static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan); static const struct ast_channel_tech alsa_tech = { .type = "Console", .description = tdesc, .capabilities = AST_FORMAT_SLINEAR, .requester = alsa_request, .send_digit_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 int send_sound(void) { short myframe[FRAME_SIZE]; int total = FRAME_SIZE; short *frame = NULL; int amt = 0, res, myoff; snd_pcm_state_t state; if (cursound == -1) return 0; res = total; if (sampsent < sounds[cursound].samplen) { myoff = 0; while (total) { amt = total; if (amt > (sounds[cursound].datalen - offset)) amt = sounds[cursound].datalen - offset; memcpy(myframe + myoff, sounds[cursound].data + offset, amt * 2); total -= amt; offset += amt; sampsent += amt; myoff += amt; if (offset >= sounds[cursound].datalen) offset = 0; } /* Set it up for silence */ if (sampsent >= sounds[cursound].samplen) silencelen = sounds[cursound].silencelen; frame = myframe; } else { if (silencelen > 0) { frame = silence; silencelen -= res; } else { if (sounds[cursound].repeat) { /* Start over */ sampsent = 0; offset = 0; } else { cursound = -1; nosound = 0; } return 0; } } if (res == 0 || !frame) return 0; #ifdef ALSA_MONITOR alsa_monitor_write((char *) frame, res * 2); #endif if (alsa.ocard > 0) { state = snd_pcm_state(alsa.ocard); if (state == SND_PCM_STATE_XRUN) snd_pcm_prepare(alsa.ocard); while ((res = snd_pcm_writei(alsa.ocard, frame, res)) == -EAGAIN) { usleep(1); } if (res > 0) return 0; } return 0; } static void *sound_thread(void *unused) { fd_set rfds; fd_set wfds; int max, res; for (;;) { FD_ZERO(&rfds); FD_ZERO(&wfds); max = sndcmd[0]; FD_SET(sndcmd[0], &rfds); if (cursound > -1 && writedev > -1) { FD_SET(writedev, &wfds); if (writedev > max) max = writedev; } #ifdef ALSA_MONITOR if (!alsa.owner && readdev > -1) { FD_SET(readdev, &rfds); if (readdev > max) max = readdev; } #endif res = ast_select(max + 1, &rfds, &wfds, NULL, NULL); if (res < 1) { ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno)); continue; } #ifdef ALSA_MONITOR if (readdev > -1 && FD_ISSET(readdev, &rfds)) { /* Keep the pipe going with read audio */ snd_pcm_state_t state; short buf[FRAME_SIZE]; int r; if (alsa.ocard > 0) { state = snd_pcm_state(alsa.ocard); if (state == SND_PCM_STATE_XRUN) { snd_pcm_prepare(alsa.ocard); } } r = snd_pcm_readi(alsa.icard, buf, FRAME_SIZE); if (r == -EPIPE) { #if DEBUG ast_log(LOG_ERROR, "XRUN read\n"); #endif snd_pcm_prepare(alsa.icard); } else if (r == -ESTRPIPE) { ast_log(LOG_ERROR, "-ESTRPIPE\n"); snd_pcm_prepare(alsa.icard); } else if (r < 0) { ast_log(LOG_ERROR, "Read error: %s\n", snd_strerror(r)); } else alsa_monitor_read((char *) buf, r * 2); } #endif if (FD_ISSET(sndcmd[0], &rfds)) { if (read(sndcmd[0], &cursound, sizeof(cursound)) < 0) { ast_log(LOG_WARNING, "read() failed: %s\n", strerror(errno)); } silencelen = 0; offset = 0; sampsent = 0; } if (writedev > -1 && FD_ISSET(writedev, &wfds)) if (send_sound()) ast_log(LOG_WARNING, "Failed to write sound\n"); } /* Never reached */ return NULL; } static snd_pcm_t *alsa_card_init(char *dev, snd_pcm_stream_t stream) { int err; int direction; snd_pcm_t *handle = NULL; snd_pcm_hw_params_t *hwparams = NULL; snd_pcm_sw_params_t *swparams = NULL; struct pollfd pfd; snd_pcm_uframes_t period_size = PERIOD_FRAMES * 4; /* int period_bytes = 0; */ snd_pcm_uframes_t buffer_size = 0; unsigned int rate = DESIRED_RATE; #if 0 unsigned int per_min = 1; #endif /* unsigned int per_max = 8; */ snd_pcm_uframes_t start_threshold, stop_threshold; err = snd_pcm_open(&handle, dev, stream, SND_PCM_NONBLOCK); if (err < 0) { ast_log(LOG_ERROR, "snd_pcm_open failed: %s\n", snd_strerror(err)); return NULL; } else ast_log(LOG_DEBUG, "Opening device %s in %s mode\n", dev, (stream == SND_PCM_STREAM_CAPTURE) ? "read" : "write"); hwparams = 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_log(LOG_DEBUG, "Period size is %d\n", err); buffer_size = 4096 * 2; /* period_size * 16; */ err = snd_pcm_hw_params_set_buffer_size_near(handle, hwparams, &buffer_size); if (err < 0) ast_log(LOG_WARNING, "Problem setting buffer size of %ld: %s\n", buffer_size, snd_strerror(err)); else ast_log(LOG_DEBUG, "Buffer size is set to %d frames\n", err); #if 0 direction = 0; err = snd_pcm_hw_params_set_periods_min(handle, hwparams, &per_min, &direction); if (err < 0) ast_log(LOG_ERROR, "periods_min: %s\n", snd_strerror(err)); err = snd_pcm_hw_params_set_periods_max(handle, hwparams, &per_max, 0); if (err < 0) ast_log(LOG_ERROR, "periods_max: %s\n", snd_strerror(err)); #endif err = snd_pcm_hw_params(handle, hwparams); if (err < 0) ast_log(LOG_ERROR, "Couldn't set the new hw params: %s\n", snd_strerror(err)); swparams = alloca(snd_pcm_sw_params_sizeof()); memset(swparams, 0, snd_pcm_sw_params_sizeof()); snd_pcm_sw_params_current(handle, swparams); #if 1 if (stream == SND_PCM_STREAM_PLAYBACK) start_threshold = period_size; else start_threshold = 1; err = snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold); if (err < 0) ast_log(LOG_ERROR, "start threshold: %s\n", snd_strerror(err)); #endif #if 1 if (stream == SND_PCM_STREAM_PLAYBACK) stop_threshold = buffer_size; else stop_threshold = buffer_size; err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, stop_threshold); if (err < 0) ast_log(LOG_ERROR, "stop threshold: %s\n", snd_strerror(err)); #endif #if 0 err = snd_pcm_sw_params_set_xfer_align(handle, swparams, PERIOD_FRAMES); if (err < 0) ast_log(LOG_ERROR, "Unable to set xfer alignment: %s\n", snd_strerror(err)); #endif #if 0 err = snd_pcm_sw_params_set_silence_threshold(handle, swparams, silencethreshold); if (err < 0) ast_log(LOG_ERROR, "Unable to set silence threshold: %s\n", snd_strerror(err)); #endif err = snd_pcm_sw_params(handle, swparams); if (err < 0) ast_log(LOG_ERROR, "sw_params: %s\n", snd_strerror(err)); err = snd_pcm_poll_descriptors_count(handle); if (err <= 0) ast_log(LOG_ERROR, "Unable to get a poll descriptors count, error is %s\n", snd_strerror(err)); if (err != 1) ast_log(LOG_DEBUG, "Can't handle more than one device\n"); snd_pcm_poll_descriptors(handle, &pfd, err); ast_log(LOG_DEBUG, "Acquired fd %d from the poll descriptor\n", pfd.fd); if (stream == SND_PCM_STREAM_CAPTURE) readdev = pfd.fd; else writedev = pfd.fd; return handle; } static int soundcard_init(void) { alsa.icard = alsa_card_init(indevname, SND_PCM_STREAM_CAPTURE); alsa.ocard = alsa_card_init(outdevname, SND_PCM_STREAM_PLAYBACK); if (!alsa.icard && !alsa.ocard) { ast_log(LOG_ERROR, "Problem opening alsa I/O devices\n"); return -1; } return 0; } 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_mutex_trylock(&alsa.owner->lock)) { DEADLOCK_AVOIDANCE(&alsalock); } } static int alsa_call(struct ast_channel *c, char *dest, int timeout) { int res = 3; struct ast_frame f = { AST_FRAME_CONTROL }; ast_mutex_lock(&alsalock); ast_verbose(" << Call placed to '%s' on console >> \n", dest); if (autoanswer) { ast_verbose(" << Auto-answered >> \n"); grab_owner(); if (alsa.owner) { f.subclass = AST_CONTROL_ANSWER; ast_queue_frame(alsa.owner, &f); ast_mutex_unlock(&alsa.owner->lock); } } else { ast_verbose(" << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n"); grab_owner(); if (alsa.owner) { f.subclass = AST_CONTROL_RINGING; ast_queue_frame(alsa.owner, &f); ast_mutex_unlock(&alsa.owner->lock); } if (write(sndcmd[1], &res, sizeof(res)) < 0) { ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno)); } } if (alsa.icard > 0) { snd_pcm_prepare(alsa.icard); snd_pcm_start(alsa.icard); } ast_mutex_unlock(&alsalock); return 0; } static void answer_sound(void) { int res; nosound = 1; res = 4; if (write(sndcmd[1], &res, sizeof(res)) < 0) { ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno)); } } static int alsa_answer(struct ast_channel *c) { ast_mutex_lock(&alsalock); ast_verbose(" << Console call has been answered >> \n"); answer_sound(); ast_setstate(c, AST_STATE_UP); cursound = -1; if (alsa.icard > 0) { snd_pcm_prepare(alsa.icard); snd_pcm_start(alsa.icard); } ast_mutex_unlock(&alsalock); return 0; } static int alsa_hangup(struct ast_channel *c) { int res; ast_mutex_lock(&alsalock); cursound = -1; c->tech_pvt = NULL; alsa.owner = NULL; ast_verbose(" << Hangup on console >> \n"); ast_module_unref(ast_module_info->self); if (hookstate) { hookstate = 0; if (!autoanswer) { /* Congestion noise */ res = 2; if (write(sndcmd[1], &res, sizeof(res)) < 0) { ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno)); } } } if (alsa.icard > 0) { snd_pcm_drop(alsa.icard); } ast_mutex_unlock(&alsalock); return 0; } static int alsa_write(struct ast_channel *chan, struct ast_frame *f) { static char sizbuf[8000]; static int sizpos = 0; int len = sizpos; int pos; int res = 0; /* size_t frames = 0; */ snd_pcm_state_t state; /* Immediately return if no sound is enabled */ if (nosound || !alsa.ocard) return 0; ast_mutex_lock(&alsalock); /* Stop any currently playing sound */ if (cursound != -1) { snd_pcm_drop(alsa.ocard); snd_pcm_prepare(alsa.ocard); cursound = -1; } /* We have to digest the frame in 160-byte portions */ if (f->datalen > sizeof(sizbuf) - sizpos) { ast_log(LOG_WARNING, "Frame too large\n"); res = -1; } else { memcpy(sizbuf + sizpos, f->data, f->datalen); len += f->datalen; pos = 0; #ifdef ALSA_MONITOR alsa_monitor_write(sizbuf, len); #endif state = snd_pcm_state(alsa.ocard); if (state == SND_PCM_STATE_XRUN) snd_pcm_prepare(alsa.ocard); while ((res = snd_pcm_writei(alsa.ocard, sizbuf, len / 2)) == -EAGAIN) { usleep(1); } if (res == -EPIPE) { #if DEBUG ast_log(LOG_DEBUG, "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); if (res > 0) res = 0; return res; } static struct ast_frame *alsa_read(struct ast_channel *chan) { static struct ast_frame f; static short __buf[FRAME_SIZE + AST_FRIENDLY_OFFSET / 2]; short *buf; static int readpos = 0; static int left = FRAME_SIZE; snd_pcm_state_t state; int r = 0; int off = 0; ast_mutex_lock(&alsalock); /* Acknowledge any pending cmd */ f.frametype = AST_FRAME_NULL; f.subclass = 0; f.samples = 0; f.datalen = 0; f.data = NULL; f.offset = 0; f.src = "Console"; f.mallocd = 0; f.delivery.tv_sec = 0; f.delivery.tv_usec = 0; if (alsa.icard > 0) { state = snd_pcm_state(alsa.icard); if ((state != SND_PCM_STATE_PREPARED) && (state != SND_PCM_STATE_RUNNING)) { snd_pcm_prepare(alsa.icard); } buf = __buf + AST_FRIENDLY_OFFSET / 2; r = snd_pcm_readi(alsa.icard, buf + readpos, left); if (r == -EPIPE) { #if DEBUG ast_log(LOG_ERROR, "XRUN read\n"); #endif snd_pcm_prepare(alsa.icard); } else if (r == -ESTRPIPE) { ast_log(LOG_ERROR, "-ESTRPIPE\n"); snd_pcm_prepare(alsa.icard); } else if (r < 0) { ast_log(LOG_ERROR, "Read error: %s\n", snd_strerror(r)); } else if (r >= 0) { off -= r; } /* Update positions */ readpos += r; left -= r; if (readpos >= FRAME_SIZE) { /* A real frame */ readpos = 0; left = FRAME_SIZE; if (chan->_state != AST_STATE_UP) { /* Don't transmit unless it's up */ ast_mutex_unlock(&alsalock); return &f; } f.frametype = AST_FRAME_VOICE; f.subclass = AST_FORMAT_SLINEAR; f.samples = FRAME_SIZE; f.datalen = FRAME_SIZE * 2; f.data = buf; f.offset = AST_FRIENDLY_OFFSET; f.src = "Console"; f.mallocd = 0; #ifdef ALSA_MONITOR alsa_monitor_read((char *) buf, FRAME_SIZE * 2); #endif } } ast_mutex_unlock(&alsalock); return &f; } static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan) { struct chan_alsa_pvt *p = newchan->tech_pvt; ast_mutex_lock(&alsalock); p->owner = newchan; ast_mutex_unlock(&alsalock); return 0; } static int alsa_indicate(struct ast_channel *chan, int cond, const void *data, size_t datalen) { int res = 0; ast_mutex_lock(&alsalock); switch (cond) { case AST_CONTROL_BUSY: res = 1; break; case AST_CONTROL_CONGESTION: res = 2; break; case AST_CONTROL_RINGING: case AST_CONTROL_PROGRESS: break; case -1: res = -1; break; case AST_CONTROL_VIDUPDATE: res = -1; break; case AST_CONTROL_HOLD: ast_verbose(" << Console Has Been Placed on Hold >> \n"); ast_moh_start(chan, data, mohinterpret); break; case AST_CONTROL_UNHOLD: ast_verbose(" << Console Has Been Retrieved from Hold >> \n"); ast_moh_stop(chan); break; case AST_CONTROL_SRCUPDATE: break; default: ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name); res = -1; } if (res > -1) { if (write(sndcmd[1], &res, sizeof(res)) < 0) { ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno)); } } ast_mutex_unlock(&alsalock); return res; } static struct ast_channel *alsa_new(struct chan_alsa_pvt *p, int state) { struct ast_channel *tmp = NULL; if (!(tmp = ast_channel_alloc(1, state, 0, 0, "", p->exten, p->context, 0, "ALSA/%s", indevname))) return NULL; tmp->tech = &alsa_tech; tmp->fds[0] = readdev; tmp->nativeformats = AST_FORMAT_SLINEAR; tmp->readformat = AST_FORMAT_SLINEAR; tmp->writeformat = AST_FORMAT_SLINEAR; tmp->tech_pvt = p; if (!ast_strlen_zero(p->context)) ast_copy_string(tmp->context, p->context, sizeof(tmp->context)); if (!ast_strlen_zero(p->exten)) ast_copy_string(tmp->exten, p->exten, sizeof(tmp->exten)); if (!ast_strlen_zero(language)) ast_string_field_set(tmp, language, language); p->owner = tmp; ast_module_ref(ast_module_info->self); ast_jb_configure(tmp, &global_jbconf); if (state != AST_STATE_DOWN) { if (ast_pbx_start(tmp)) { ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name); ast_hangup(tmp); tmp = NULL; } } return tmp; } static struct ast_channel *alsa_request(const char *type, int format, void *data, int *cause) { int oldformat = format; struct ast_channel *tmp = NULL; format &= AST_FORMAT_SLINEAR; if (!format) { ast_log(LOG_NOTICE, "Asked to get a channel of format '%d'\n", oldformat); return NULL; } ast_mutex_lock(&alsalock); if (alsa.owner) { ast_log(LOG_NOTICE, "Already have a call on the ALSA channel\n"); *cause = AST_CAUSE_BUSY; } else if (!(tmp = alsa_new(&alsa, AST_STATE_DOWN))) ast_log(LOG_WARNING, "Unable to create new ALSA channel\n"); ast_mutex_unlock(&alsalock); return tmp; } static int console_autoanswer_deprecated(int fd, int argc, char *argv[]) { int res = RESULT_SUCCESS; if ((argc != 1) && (argc != 2)) return RESULT_SHOWUSAGE; ast_mutex_lock(&alsalock); if (argc == 1) { ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off"); } else { if (!strcasecmp(argv[1], "on")) autoanswer = -1; else if (!strcasecmp(argv[1], "off")) autoanswer = 0; else res = RESULT_SHOWUSAGE; } ast_mutex_unlock(&alsalock); return res; } static int console_autoanswer(int fd, int argc, char *argv[]) { int res = RESULT_SUCCESS;; if ((argc != 2) && (argc != 3)) return RESULT_SHOWUSAGE; ast_mutex_lock(&alsalock); if (argc == 2) { ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off"); } else { if (!strcasecmp(argv[2], "on")) autoanswer = -1; else if (!strcasecmp(argv[2], "off")) autoanswer = 0; else res = RESULT_SHOWUSAGE; } ast_mutex_unlock(&alsalock); return res; } static char *autoanswer_complete(const char *line, const char *word, int pos, int state) { #ifndef MIN #define MIN(a,b) ((a) < (b) ? (a) : (b)) #endif switch (state) { case 0: if (!ast_strlen_zero(word) && !strncasecmp(word, "on", MIN(strlen(word), 2))) return ast_strdup("on"); case 1: if (!ast_strlen_zero(word) && !strncasecmp(word, "off", MIN(strlen(word), 3))) return ast_strdup("off"); default: return NULL; } return NULL; } static const char autoanswer_usage[] = "Usage: 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"; static int console_answer_deprecated(int fd, int argc, char *argv[]) { int res = RESULT_SUCCESS; if (argc != 1) return RESULT_SHOWUSAGE; ast_mutex_lock(&alsalock); if (!alsa.owner) { ast_cli(fd, "No one is calling us\n"); res = RESULT_FAILURE; } else { hookstate = 1; cursound = -1; grab_owner(); if (alsa.owner) { struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER }; ast_queue_frame(alsa.owner, &f); ast_mutex_unlock(&alsa.owner->lock); } answer_sound(); } if (alsa.icard > 0) { snd_pcm_prepare(alsa.icard); snd_pcm_start(alsa.icard); } ast_mutex_unlock(&alsalock); return RESULT_SUCCESS; } static int console_answer(int fd, int argc, char *argv[]) { int res = RESULT_SUCCESS; if (argc != 2) return RESULT_SHOWUSAGE; ast_mutex_lock(&alsalock); if (!alsa.owner) { ast_cli(fd, "No one is calling us\n"); res = RESULT_FAILURE; } else { hookstate = 1; cursound = -1; grab_owner(); if (alsa.owner) { struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER }; ast_queue_frame(alsa.owner, &f); ast_mutex_unlock(&alsa.owner->lock); } answer_sound(); } if (alsa.icard > 0) { snd_pcm_prepare(alsa.icard); snd_pcm_start(alsa.icard); } ast_mutex_unlock(&alsalock); return RESULT_SUCCESS; } static char sendtext_usage[] = "Usage: console send text <message>\n" " Sends a text message for display on the remote terminal.\n"; static int console_sendtext_deprecated(int fd, int argc, char *argv[]) { int tmparg = 2; int res = RESULT_SUCCESS; if (argc < 2) return RESULT_SHOWUSAGE; ast_mutex_lock(&alsalock); if (!alsa.owner) { ast_cli(fd, "No one is calling us\n"); res = RESULT_FAILURE; } else { struct ast_frame f = { AST_FRAME_TEXT, 0 }; char text2send[256] = ""; text2send[0] = '\0'; while (tmparg < argc) { strncat(text2send, argv[tmparg++], sizeof(text2send) - strlen(text2send) - 1); strncat(text2send, " ", sizeof(text2send) - strlen(text2send) - 1); } text2send[strlen(text2send) - 1] = '\n'; f.data = text2send; f.datalen = strlen(text2send) + 1; grab_owner(); if (alsa.owner) { ast_queue_frame(alsa.owner, &f); f.frametype = AST_FRAME_CONTROL; f.subclass = AST_CONTROL_ANSWER; f.data = NULL; f.datalen = 0; ast_queue_frame(alsa.owner, &f); ast_mutex_unlock(&alsa.owner->lock); } } ast_mutex_unlock(&alsalock); return res; } static int console_sendtext(int fd, int argc, char *argv[]) { int tmparg = 3; int res = RESULT_SUCCESS; if (argc < 3) return RESULT_SHOWUSAGE; ast_mutex_lock(&alsalock); if (!alsa.owner) { ast_cli(fd, "No one is calling us\n"); res = RESULT_FAILURE; } else { struct ast_frame f = { AST_FRAME_TEXT, 0 }; char text2send[256] = ""; text2send[0] = '\0'; while (tmparg < argc) { strncat(text2send, argv[tmparg++], sizeof(text2send) - strlen(text2send) - 1); strncat(text2send, " ", sizeof(text2send) - strlen(text2send) - 1); } text2send[strlen(text2send) - 1] = '\n'; f.data = text2send; f.datalen = strlen(text2send) + 1; grab_owner(); if (alsa.owner) { ast_queue_frame(alsa.owner, &f); f.frametype = AST_FRAME_CONTROL; f.subclass = AST_CONTROL_ANSWER; f.data = NULL; f.datalen = 0; ast_queue_frame(alsa.owner, &f); ast_mutex_unlock(&alsa.owner->lock); } } ast_mutex_unlock(&alsalock); return res; } static char answer_usage[] = "Usage: console answer\n" " Answers an incoming call on the console (ALSA) channel.\n"; static int console_hangup_deprecated(int fd, int argc, char *argv[]) { int res = RESULT_SUCCESS; if (argc != 1) return RESULT_SHOWUSAGE; cursound = -1; ast_mutex_lock(&alsalock); if (!alsa.owner && !hookstate) { ast_cli(fd, "No call to hangup up\n"); res = RESULT_FAILURE; } else { hookstate = 0; grab_owner(); if (alsa.owner) { ast_queue_hangup(alsa.owner); ast_mutex_unlock(&alsa.owner->lock); } } ast_mutex_unlock(&alsalock); return res; } static int console_hangup(int fd, int argc, char *argv[]) { int res = RESULT_SUCCESS; if (argc != 2) return RESULT_SHOWUSAGE; cursound = -1; ast_mutex_lock(&alsalock); if (!alsa.owner && !hookstate) { ast_cli(fd, "No call to hangup up\n"); res = RESULT_FAILURE; } else { hookstate = 0; grab_owner(); if (alsa.owner) { ast_queue_hangup(alsa.owner); ast_mutex_unlock(&alsa.owner->lock); } } ast_mutex_unlock(&alsalock); return res; } static char hangup_usage[] = "Usage: console hangup\n" " Hangs up any call currently placed on the console.\n"; static int console_dial_deprecated(int fd, int argc, char *argv[]) { char tmp[256], *tmp2; char *mye, *myc; char *d; int res = RESULT_SUCCESS; if ((argc != 1) && (argc != 2)) return RESULT_SHOWUSAGE; ast_mutex_lock(&alsalock); if (alsa.owner) { if (argc == 2) { d = argv[1]; grab_owner(); if (alsa.owner) { struct ast_frame f = { AST_FRAME_DTMF }; while (*d) { f.subclass = *d; ast_queue_frame(alsa.owner, &f); d++; } ast_mutex_unlock(&alsa.owner->lock); } } else { ast_cli(fd, "You're already in a call. You can use this only to dial digits until you hangup\n"); res = RESULT_FAILURE; } } else { mye = exten; myc = context; if (argc == 2) { char *stringp = NULL; ast_copy_string(tmp, argv[1], 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); } else ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc); } ast_mutex_unlock(&alsalock); return res; } static int console_dial(int fd, int argc, char *argv[]) { char tmp[256], *tmp2; char *mye, *myc; char *d; int res = RESULT_SUCCESS; if ((argc != 2) && (argc != 3)) return RESULT_SHOWUSAGE; ast_mutex_lock(&alsalock); if (alsa.owner) { if (argc == 3) { d = argv[2]; grab_owner(); if (alsa.owner) { struct ast_frame f = { AST_FRAME_DTMF }; while (*d) { f.subclass = *d; ast_queue_frame(alsa.owner, &f); d++; } ast_mutex_unlock(&alsa.owner->lock); } } else { ast_cli(fd, "You're already in a call. You can use this only to dial digits until you hangup\n"); res = RESULT_FAILURE; } } else { mye = exten; myc = context; if (argc == 3) { char *stringp = NULL; ast_copy_string(tmp, 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); } else ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc); } ast_mutex_unlock(&alsalock); return res; } static char dial_usage[] = "Usage: console dial [extension[@context]]\n" " Dials a given extension (and context if specified)\n"; static struct ast_cli_entry cli_alsa_answer_deprecated = { { "answer", NULL }, console_answer_deprecated, NULL, NULL }; static struct ast_cli_entry cli_alsa_hangup_deprecated = { { "hangup", NULL }, console_hangup_deprecated, NULL, NULL }; static struct ast_cli_entry cli_alsa_dial_deprecated = { { "dial", NULL }, console_dial_deprecated, NULL, NULL }; static struct ast_cli_entry cli_alsa_send_text_deprecated = { { "send", "text", NULL }, console_sendtext_deprecated, NULL, NULL }; static struct ast_cli_entry cli_alsa_autoanswer_deprecated = { { "autoanswer", NULL }, console_autoanswer_deprecated, NULL, NULL, autoanswer_complete }; static struct ast_cli_entry cli_alsa[] = { { { "console", "answer", NULL }, console_answer, "Answer an incoming console call", answer_usage, NULL, &cli_alsa_answer_deprecated }, { { "console", "hangup", NULL }, console_hangup, "Hangup a call on the console", hangup_usage, NULL, &cli_alsa_hangup_deprecated }, { { "console", "dial", NULL }, console_dial, "Dial an extension on the console", dial_usage, NULL, &cli_alsa_dial_deprecated }, { { "console", "send", "text", NULL }, console_sendtext, "Send text to the remote device", sendtext_usage, NULL, &cli_alsa_send_text_deprecated }, { { "console", "autoanswer", NULL }, console_autoanswer, "Sets/displays autoanswer", autoanswer_usage, autoanswer_complete, &cli_alsa_autoanswer_deprecated }, }; static int load_module(void) { int res; struct ast_config *cfg; struct ast_variable *v; /* Copy the default jb config over global_jbconf */ memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf)); strcpy(mohinterpret, "default"); if ((cfg = ast_config_load(config))) { v = ast_variable_browse(cfg, "general"); for (; v; v = v->next) { /* handle jb conf */ if (!ast_jb_read_conf(&global_jbconf, v->name, v->value)) continue; if (!strcasecmp(v->name, "autoanswer")) autoanswer = ast_true(v->value); else if (!strcasecmp(v->name, "silencesuppression")) silencesuppression = ast_true(v->value); else if (!strcasecmp(v->name, "silencethreshold")) silencethreshold = atoi(v->value); else if (!strcasecmp(v->name, "context")) ast_copy_string(context, v->value, sizeof(context)); else if (!strcasecmp(v->name, "language")) ast_copy_string(language, v->value, sizeof(language)); else if (!strcasecmp(v->name, "extension")) ast_copy_string(exten, v->value, sizeof(exten)); else if (!strcasecmp(v->name, "input_device")) ast_copy_string(indevname, v->value, sizeof(indevname)); else if (!strcasecmp(v->name, "output_device")) ast_copy_string(outdevname, v->value, sizeof(outdevname)); else if (!strcasecmp(v->name, "mohinterpret")) ast_copy_string(mohinterpret, v->value, sizeof(mohinterpret)); } ast_config_destroy(cfg); } res = pipe(sndcmd); if (res) { ast_log(LOG_ERROR, "Unable to create pipe\n"); return -1; } res = soundcard_init(); if (res < 0) { if (option_verbose > 1) { ast_verbose(VERBOSE_PREFIX_2 "No sound card detected -- console channel will be unavailable\n"); ast_verbose(VERBOSE_PREFIX_2 "Turn off ALSA support by adding 'noload=chan_alsa.so' in /etc/asterisk/modules.conf\n"); } return 0; } res = ast_channel_register(&alsa_tech); if (res < 0) { ast_log(LOG_ERROR, "Unable to register channel class 'Console'\n"); return -1; } ast_cli_register_multiple(cli_alsa, sizeof(cli_alsa) / sizeof(struct ast_cli_entry)); ast_pthread_create_background(&sthread, NULL, sound_thread, NULL); #ifdef ALSA_MONITOR if (alsa_monitor_start()) ast_log(LOG_ERROR, "Problem starting Monitoring\n"); #endif return 0; } static int unload_module(void) { ast_channel_unregister(&alsa_tech); ast_cli_unregister_multiple(cli_alsa, sizeof(cli_alsa) / sizeof(struct ast_cli_entry)); if (alsa.icard > 0) snd_pcm_close(alsa.icard); if (alsa.ocard > 0) snd_pcm_close(alsa.ocard); if (sndcmd[0] > 0) { close(sndcmd[0]); close(sndcmd[1]); } if (alsa.owner) ast_softhangup(alsa.owner, AST_SOFTHANGUP_APPUNLOAD); if (alsa.owner) return -1; return 0; } AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "ALSA Console Channel Driver");
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