diff options
author | kpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-03-03 18:13:26 +0000 |
---|---|---|
committer | kpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-03-03 18:13:26 +0000 |
commit | 67da2f8263b4e9bb5522fa59b27e143381d69774 (patch) | |
tree | e69baf2b1594606e9ea2e80d26d691a10bd23831 /1.2-netsec/channels/chan_alsa.c | |
parent | 187ac8fdb51443812933047136b96b5a532dd857 (diff) |
Creating tag for the release of asterisk-1.2.5
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.2.5@11747 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to '1.2-netsec/channels/chan_alsa.c')
-rw-r--r-- | 1.2-netsec/channels/chan_alsa.c | 1140 |
1 files changed, 1140 insertions, 0 deletions
diff --git a/1.2-netsec/channels/chan_alsa.c b/1.2-netsec/channels/chan_alsa.c new file mode 100644 index 000000000..6f0d36070 --- /dev/null +++ b/1.2-netsec/channels/chan_alsa.c @@ -0,0 +1,1140 @@ +/* + * 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 + * + * \par See also + * \arg Config_alsa + * + * \ingroup channel_drivers + */ + + +#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.h" + +ASTERISK_FILE_VERSION(__FILE__, "$Revision$") + +#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 "busy.h" +#include "ringtone.h" +#include "ring10.h" +#include "answer.h" + +#ifdef ALSA_MONITOR +#include "alsa-monitor.h" +#endif + +#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 int block = O_NONBLOCK; */ +static char indevname[50] = ALSA_INDEV; +static char outdevname[50] = ALSA_OUTDEV; + +#if 0 +static struct timeval lasttime; +#endif + +static int usecnt; +static int silencesuppression = 0; +static int silencethreshold = 1000; + +AST_MUTEX_DEFINE_STATIC(usecnt_lock); +AST_MUTEX_DEFINE_STATIC(alsalock); + +static const char type[] = "Console"; +static const char desc[] = "ALSA Console Channel Driver"; +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 int hookstate=0; + +static short silence[FRAME_SIZE] = {0, }; + +struct sound { + int ind; + short *data; + int datalen; + int samplen; + int silencelen; + int repeat; +}; + +static struct sound sounds[] = { + { AST_CONTROL_RINGING, ringtone, sizeof(ringtone)/2, 16000, 32000, 1 }, + { AST_CONTROL_BUSY, busy, sizeof(busy)/2, 4000, 4000, 1 }, + { AST_CONTROL_CONGESTION, busy, sizeof(busy)/2, 2000, 2000, 1 }, + { AST_CONTROL_RING, ring10, sizeof(ring10)/2, 16000, 32000, 1 }, + { AST_CONTROL_ANSWER, answer, sizeof(answer)/2, 2200, 0, 0 }, +}; + +/* Sound command pipe */ +static int sndcmd[2]; + +static struct chan_alsa_pvt { + /* We only have one ALSA structure -- near sighted perhaps, but it + keeps this driver as simple as possible -- as it should be. */ + struct ast_channel *owner; + char exten[AST_MAX_EXTENSION]; + char context[AST_MAX_CONTEXT]; +#if 0 + snd_pcm_t *card; +#endif + snd_pcm_t *icard, *ocard; + +} alsa; + +/* Number of buffers... Each is FRAMESIZE/8 ms long. For example + with 160 sample frames, and a buffer size of 3, we have a 60ms buffer, + usually plenty. */ + +pthread_t sthread; + +#define MAX_BUFFER_SIZE 100 + +/* File descriptors for sound device */ +static int readdev = -1; +static int writedev = -1; + +static int autoanswer = 1; + +static int cursound = -1; +static int sampsent = 0; +static int silencelen=0; +static int offset=0; +static int nosound=0; + +/* ZZ */ +static struct ast_channel *alsa_request(const char *type, int format, void *data, int *cause); +static int alsa_digit(struct ast_channel *c, char digit); +static int alsa_text(struct ast_channel *c, const char *text); +static int alsa_hangup(struct ast_channel *c); +static int alsa_answer(struct ast_channel *c); +static struct ast_frame *alsa_read(struct ast_channel *chan); +static int alsa_call(struct ast_channel *c, char *dest, int timeout); +static int alsa_write(struct ast_channel *chan, struct ast_frame *f); +static int alsa_indicate(struct ast_channel *chan, int cond); +static int alsa_fixup(struct ast_channel *oldchan, struct ast_channel *newchan); + +static const struct ast_channel_tech alsa_tech = { + .type = type, + .description = tdesc, + .capabilities = AST_FORMAT_SLINEAR, + .requester = alsa_request, + .send_digit = alsa_digit, + .send_text = alsa_text, + .hangup = alsa_hangup, + .answer = alsa_answer, + .read = alsa_read, + .call = alsa_call, + .write = alsa_write, + .indicate = alsa_indicate, + .fixup = alsa_fixup, +}; + +static int send_sound(void) +{ + short myframe[FRAME_SIZE]; + int total = FRAME_SIZE; + short *frame = NULL; + int amt=0; + int res; + int myoff; + snd_pcm_state_t state; + + if (cursound > -1) { + res = total; + if (sampsent < sounds[cursound].samplen) { + myoff=0; + while(total) { + amt = total; + if (amt > (sounds[cursound].datalen - offset)) + amt = sounds[cursound].datalen - offset; + memcpy(myframe + myoff, sounds[cursound].data + offset, amt * 2); + total -= amt; + offset += amt; + sampsent += amt; + myoff += amt; + if (offset >= sounds[cursound].datalen) + offset = 0; + } + /* Set it up for silence */ + if (sampsent >= sounds[cursound].samplen) + silencelen = sounds[cursound].silencelen; + frame = myframe; + } else { + if (silencelen > 0) { + frame = silence; + silencelen -= res; + } else { + if (sounds[cursound].repeat) { + /* Start over */ + sampsent = 0; + offset = 0; + } else { + cursound = -1; + nosound = 0; + } + return 0; + } + } + + if (res == 0 || !frame) { + return 0; + } +#ifdef ALSA_MONITOR + alsa_monitor_write((char *)frame, res * 2); +#endif + state = snd_pcm_state(alsa.ocard); + if (state == SND_PCM_STATE_XRUN) { + snd_pcm_prepare(alsa.ocard); + } + res = snd_pcm_writei(alsa.ocard, frame, res); + if (res > 0) + return 0; + return 0; + } + return 0; +} + +static void *sound_thread(void *unused) +{ + fd_set rfds; + fd_set wfds; + int max; + int res; + for(;;) { + FD_ZERO(&rfds); + FD_ZERO(&wfds); + max = sndcmd[0]; + FD_SET(sndcmd[0], &rfds); + if (cursound > -1) { + FD_SET(writedev, &wfds); + if (writedev > max) + max = writedev; + } +#ifdef ALSA_MONITOR + if (!alsa.owner) { + FD_SET(readdev, &rfds); + if (readdev > max) + max = readdev; + } +#endif + res = ast_select(max + 1, &rfds, &wfds, NULL, NULL); + if (res < 1) { + ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno)); + continue; + } +#ifdef ALSA_MONITOR + if (FD_ISSET(readdev, &rfds)) { + /* Keep the pipe going with read audio */ + snd_pcm_state_t state; + short buf[FRAME_SIZE]; + int r; + + state = snd_pcm_state(alsa.ocard); + if (state == SND_PCM_STATE_XRUN) { + snd_pcm_prepare(alsa.ocard); + } + r = snd_pcm_readi(alsa.icard, buf, FRAME_SIZE); + if (r == -EPIPE) { +#if DEBUG + ast_log(LOG_ERROR, "XRUN read\n"); +#endif + snd_pcm_prepare(alsa.icard); + } else if (r == -ESTRPIPE) { + ast_log(LOG_ERROR, "-ESTRPIPE\n"); + snd_pcm_prepare(alsa.icard); + } else if (r < 0) { + ast_log(LOG_ERROR, "Read error: %s\n", snd_strerror(r)); + } else + alsa_monitor_read((char *)buf, r * 2); + } +#endif + if (FD_ISSET(sndcmd[0], &rfds)) { + read(sndcmd[0], &cursound, sizeof(cursound)); + silencelen = 0; + offset = 0; + sampsent = 0; + } + if (FD_ISSET(writedev, &wfds)) + if (send_sound()) + ast_log(LOG_WARNING, "Failed to write sound\n"); + } + /* Never reached */ + return NULL; +} + +static snd_pcm_t *alsa_card_init(char *dev, snd_pcm_stream_t stream) +{ + int err; + int direction; + snd_pcm_t *handle = NULL; + snd_pcm_hw_params_t *hwparams = NULL; + snd_pcm_sw_params_t *swparams = NULL; + struct pollfd pfd; + snd_pcm_uframes_t period_size = PERIOD_FRAMES * 4; + /* int period_bytes = 0; */ + snd_pcm_uframes_t buffer_size = 0; + + unsigned int rate = DESIRED_RATE; +#if 0 + unsigned int per_min = 1; +#endif + /* unsigned int per_max = 8; */ + snd_pcm_uframes_t start_threshold, stop_threshold; + + err = snd_pcm_open(&handle, dev, stream, O_NONBLOCK); + if (err < 0) { + ast_log(LOG_ERROR, "snd_pcm_open failed: %s\n", snd_strerror(err)); + return NULL; + } else { + ast_log(LOG_DEBUG, "Opening device %s in %s mode\n", dev, (stream == SND_PCM_STREAM_CAPTURE) ? "read" : "write"); + } + + snd_pcm_hw_params_alloca(&hwparams); + snd_pcm_hw_params_any(handle, hwparams); + + err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); + if (err < 0) { + ast_log(LOG_ERROR, "set_access failed: %s\n", snd_strerror(err)); + } + + err = snd_pcm_hw_params_set_format(handle, hwparams, format); + if (err < 0) { + ast_log(LOG_ERROR, "set_format failed: %s\n", snd_strerror(err)); + } + + err = snd_pcm_hw_params_set_channels(handle, hwparams, 1); + if (err < 0) { + ast_log(LOG_ERROR, "set_channels failed: %s\n", snd_strerror(err)); + } + + direction = 0; + err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &rate, &direction); + if (rate != DESIRED_RATE) { + ast_log(LOG_WARNING, "Rate not correct, requested %d, got %d\n", DESIRED_RATE, rate); + } + + direction = 0; + err = snd_pcm_hw_params_set_period_size_near(handle, hwparams, &period_size, &direction); + if (err < 0) { + ast_log(LOG_ERROR, "period_size(%ld frames) is bad: %s\n", period_size, snd_strerror(err)); + } else { + ast_log(LOG_DEBUG, "Period size is %d\n", err); + } + + buffer_size = 4096 * 2; /* period_size * 16; */ + err = snd_pcm_hw_params_set_buffer_size_near(handle, hwparams, &buffer_size); + if (err < 0) { + ast_log(LOG_WARNING, "Problem setting buffer size of %ld: %s\n", buffer_size, snd_strerror(err)); + } else { + ast_log(LOG_DEBUG, "Buffer size is set to %d frames\n", err); + } + +#if 0 + direction = 0; + err = snd_pcm_hw_params_set_periods_min(handle, hwparams, &per_min, &direction); + if (err < 0) { + ast_log(LOG_ERROR, "periods_min: %s\n", snd_strerror(err)); + } + + err = snd_pcm_hw_params_set_periods_max(handle, hwparams, &per_max, 0); + if (err < 0) { + ast_log(LOG_ERROR, "periods_max: %s\n", snd_strerror(err)); + } +#endif + + err = snd_pcm_hw_params(handle, hwparams); + if (err < 0) { + ast_log(LOG_ERROR, "Couldn't set the new hw params: %s\n", snd_strerror(err)); + } + + snd_pcm_sw_params_alloca(&swparams); + snd_pcm_sw_params_current(handle, swparams); + +#if 1 + if (stream == SND_PCM_STREAM_PLAYBACK) { + start_threshold = period_size; + } else { + start_threshold = 1; + } + + err = snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold); + if (err < 0) { + ast_log(LOG_ERROR, "start threshold: %s\n", snd_strerror(err)); + } +#endif + +#if 1 + if (stream == SND_PCM_STREAM_PLAYBACK) { + stop_threshold = buffer_size; + } else { + stop_threshold = buffer_size; + } + err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, stop_threshold); + if (err < 0) { + ast_log(LOG_ERROR, "stop threshold: %s\n", snd_strerror(err)); + } +#endif +#if 0 + err = snd_pcm_sw_params_set_xfer_align(handle, swparams, PERIOD_FRAMES); + if (err < 0) { + ast_log(LOG_ERROR, "Unable to set xfer alignment: %s\n", snd_strerror(err)); + } +#endif + +#if 0 + err = snd_pcm_sw_params_set_silence_threshold(handle, swparams, silencethreshold); + if (err < 0) { + ast_log(LOG_ERROR, "Unable to set silence threshold: %s\n", snd_strerror(err)); + } +#endif + err = snd_pcm_sw_params(handle, swparams); + if (err < 0) { + ast_log(LOG_ERROR, "sw_params: %s\n", snd_strerror(err)); + } + + err = snd_pcm_poll_descriptors_count(handle); + if (err <= 0) { + ast_log(LOG_ERROR, "Unable to get a poll descriptors count, error is %s\n", snd_strerror(err)); + } + + if (err != 1) { + ast_log(LOG_DEBUG, "Can't handle more than one device\n"); + } + + snd_pcm_poll_descriptors(handle, &pfd, err); + ast_log(LOG_DEBUG, "Acquired fd %d from the poll descriptor\n", pfd.fd); + + if (stream == SND_PCM_STREAM_CAPTURE) + readdev = pfd.fd; + else + writedev = pfd.fd; + + return handle; +} + +static int soundcard_init(void) +{ + alsa.icard = alsa_card_init(indevname, SND_PCM_STREAM_CAPTURE); + alsa.ocard = alsa_card_init(outdevname, SND_PCM_STREAM_PLAYBACK); + + if (!alsa.icard || !alsa.ocard) { + ast_log(LOG_ERROR, "Problem opening alsa I/O devices\n"); + return -1; + } + + return readdev; +} + +static int alsa_digit(struct ast_channel *c, char digit) +{ + ast_mutex_lock(&alsalock); + ast_verbose( " << Console Received digit %c >> \n", digit); + ast_mutex_unlock(&alsalock); + return 0; +} + +static int alsa_text(struct ast_channel *c, const char *text) +{ + ast_mutex_lock(&alsalock); + ast_verbose( " << Console Received text %s >> \n", text); + ast_mutex_unlock(&alsalock); + return 0; +} + +static void grab_owner(void) +{ + while(alsa.owner && ast_mutex_trylock(&alsa.owner->lock)) { + ast_mutex_unlock(&alsalock); + usleep(1); + ast_mutex_lock(&alsalock); + } +} + +static int alsa_call(struct ast_channel *c, char *dest, int timeout) +{ + int res = 3; + struct ast_frame f = { AST_FRAME_CONTROL }; + ast_mutex_lock(&alsalock); + ast_verbose( " << Call placed to '%s' on console >> \n", dest); + if (autoanswer) { + ast_verbose( " << Auto-answered >> \n" ); + grab_owner(); + if (alsa.owner) { + f.subclass = AST_CONTROL_ANSWER; + ast_queue_frame(alsa.owner, &f); + ast_mutex_unlock(&alsa.owner->lock); + } + } else { + ast_verbose( " << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n"); + grab_owner(); + if (alsa.owner) { + f.subclass = AST_CONTROL_RINGING; + ast_queue_frame(alsa.owner, &f); + ast_mutex_unlock(&alsa.owner->lock); + } + write(sndcmd[1], &res, sizeof(res)); + } + snd_pcm_prepare(alsa.icard); + snd_pcm_start(alsa.icard); + ast_mutex_unlock(&alsalock); + return 0; +} + +static void answer_sound(void) +{ + int res; + nosound = 1; + res = 4; + write(sndcmd[1], &res, sizeof(res)); + +} + +static int alsa_answer(struct ast_channel *c) +{ + ast_mutex_lock(&alsalock); + ast_verbose( " << Console call has been answered >> \n"); + answer_sound(); + ast_setstate(c, AST_STATE_UP); + cursound = -1; + snd_pcm_prepare(alsa.icard); + snd_pcm_start(alsa.icard); + ast_mutex_unlock(&alsalock); + return 0; +} + +static int alsa_hangup(struct ast_channel *c) +{ + int res; + ast_mutex_lock(&alsalock); + cursound = -1; + c->tech_pvt = NULL; + alsa.owner = NULL; + ast_verbose( " << Hangup on console >> \n"); + ast_mutex_lock(&usecnt_lock); + usecnt--; + ast_mutex_unlock(&usecnt_lock); + if (hookstate) { + if (autoanswer) { + hookstate = 0; + } else { + /* Congestion noise */ + res = 2; + write(sndcmd[1], &res, sizeof(res)); + hookstate = 0; + } + } + snd_pcm_drop(alsa.icard); + ast_mutex_unlock(&alsalock); + return 0; +} + +static int alsa_write(struct ast_channel *chan, struct ast_frame *f) +{ + static char sizbuf[8000]; + static int sizpos = 0; + int len = sizpos; + int pos; + int res = 0; + /* size_t frames = 0; */ + snd_pcm_state_t state; + /* Immediately return if no sound is enabled */ + if (nosound) + return 0; + ast_mutex_lock(&alsalock); + /* Stop any currently playing sound */ + if (cursound != -1) { + snd_pcm_drop(alsa.ocard); + snd_pcm_prepare(alsa.ocard); + cursound = -1; + } + + + /* We have to digest the frame in 160-byte portions */ + if (f->datalen > sizeof(sizbuf) - sizpos) { + ast_log(LOG_WARNING, "Frame too large\n"); + res = -1; + } else { + memcpy(sizbuf + sizpos, f->data, f->datalen); + len += f->datalen; + pos = 0; +#ifdef ALSA_MONITOR + alsa_monitor_write(sizbuf, len); +#endif + state = snd_pcm_state(alsa.ocard); + if (state == SND_PCM_STATE_XRUN) { + snd_pcm_prepare(alsa.ocard); + } + res = snd_pcm_writei(alsa.ocard, sizbuf, len/2); + if (res == -EPIPE) { +#if DEBUG + ast_log(LOG_DEBUG, "XRUN write\n"); +#endif + snd_pcm_prepare(alsa.ocard); + res = snd_pcm_writei(alsa.ocard, sizbuf, len/2); + if (res != len/2) { + ast_log(LOG_ERROR, "Write error: %s\n", snd_strerror(res)); + res = -1; + } else if (res < 0) { + ast_log(LOG_ERROR, "Write error %s\n", snd_strerror(res)); + res = -1; + } + } else { + if (res == -ESTRPIPE) { + ast_log(LOG_ERROR, "You've got some big problems\n"); + } else if (res < 0) + ast_log(LOG_NOTICE, "Error %d on write\n", res); + } + } + ast_mutex_unlock(&alsalock); + if (res > 0) + res = 0; + return res; +} + + +static struct ast_frame *alsa_read(struct ast_channel *chan) +{ + static struct ast_frame f; + static short __buf[FRAME_SIZE + AST_FRIENDLY_OFFSET/2]; + short *buf; + static int readpos = 0; + static int left = FRAME_SIZE; + snd_pcm_state_t state; + int r = 0; + int off = 0; + + ast_mutex_lock(&alsalock); + /* Acknowledge any pending cmd */ + f.frametype = AST_FRAME_NULL; + f.subclass = 0; + f.samples = 0; + f.datalen = 0; + f.data = NULL; + f.offset = 0; + f.src = type; + f.mallocd = 0; + f.delivery.tv_sec = 0; + f.delivery.tv_usec = 0; + + state = snd_pcm_state(alsa.icard); + if ((state != SND_PCM_STATE_PREPARED) && + (state != SND_PCM_STATE_RUNNING)) { + snd_pcm_prepare(alsa.icard); + } + + buf = __buf + AST_FRIENDLY_OFFSET/2; + + r = snd_pcm_readi(alsa.icard, buf + readpos, left); + if (r == -EPIPE) { +#if DEBUG + ast_log(LOG_ERROR, "XRUN read\n"); +#endif + snd_pcm_prepare(alsa.icard); + } else if (r == -ESTRPIPE) { + ast_log(LOG_ERROR, "-ESTRPIPE\n"); + snd_pcm_prepare(alsa.icard); + } else if (r < 0) { + ast_log(LOG_ERROR, "Read error: %s\n", snd_strerror(r)); + } else if (r >= 0) { + off -= r; + } + /* Update positions */ + readpos += r; + left -= r; + + if (readpos >= FRAME_SIZE) { + /* A real frame */ + readpos = 0; + left = FRAME_SIZE; + if (chan->_state != AST_STATE_UP) { + /* Don't transmit unless it's up */ + ast_mutex_unlock(&alsalock); + return &f; + } + f.frametype = AST_FRAME_VOICE; + f.subclass = AST_FORMAT_SLINEAR; + f.samples = FRAME_SIZE; + f.datalen = FRAME_SIZE * 2; + f.data = buf; + f.offset = AST_FRIENDLY_OFFSET; + f.src = type; + 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) +{ + 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: + res = 0; + break; + case -1: + res = -1; + break; + case AST_CONTROL_VIDUPDATE: + res = -1; + break; + default: + ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name); + res = -1; + } + if (res > -1) { + write(sndcmd[1], &res, sizeof(res)); + } + ast_mutex_unlock(&alsalock); + return res; +} + +static struct ast_channel *alsa_new(struct chan_alsa_pvt *p, int state) +{ + struct ast_channel *tmp; + tmp = ast_channel_alloc(1); + if (tmp) { + tmp->tech = &alsa_tech; + snprintf(tmp->name, sizeof(tmp->name), "ALSA/%s", indevname); + tmp->type = type; + 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_copy_string(tmp->language, language, sizeof(tmp->language)); + p->owner = tmp; + ast_setstate(tmp, state); + ast_mutex_lock(&usecnt_lock); + usecnt++; + ast_mutex_unlock(&usecnt_lock); + ast_update_use_count(); + if (state != AST_STATE_DOWN) { + if (ast_pbx_start(tmp)) { + ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name); + ast_hangup(tmp); + tmp = NULL; + } + } + } + return tmp; +} + +static struct ast_channel *alsa_request(const char *type, int format, void *data, int *cause) +{ + int oldformat = format; + struct ast_channel *tmp=NULL; + format &= AST_FORMAT_SLINEAR; + if (!format) { + ast_log(LOG_NOTICE, "Asked to get a channel of format '%d'\n", oldformat); + return NULL; + } + ast_mutex_lock(&alsalock); + if (alsa.owner) { + ast_log(LOG_NOTICE, "Already have a call on the ALSA channel\n"); + *cause = AST_CAUSE_BUSY; + } else { + tmp= alsa_new(&alsa, AST_STATE_DOWN); + if (!tmp) { + ast_log(LOG_WARNING, "Unable to create new ALSA channel\n"); + } + } + ast_mutex_unlock(&alsalock); + return tmp; +} + +static int console_autoanswer(int fd, int argc, char *argv[]) +{ + int res = RESULT_SUCCESS;; + if ((argc != 1) && (argc != 2)) + return RESULT_SHOWUSAGE; + ast_mutex_lock(&alsalock); + if (argc == 1) { + ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off"); + } else { + if (!strcasecmp(argv[1], "on")) + autoanswer = -1; + else if (!strcasecmp(argv[1], "off")) + autoanswer = 0; + else + res = RESULT_SHOWUSAGE; + } + ast_mutex_unlock(&alsalock); + return res; +} + +static char *autoanswer_complete(char *line, 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 strdup("on"); + case 1: + if (!ast_strlen_zero(word) && !strncasecmp(word, "off", MIN(strlen(word), 3))) + return strdup("off"); + default: + return NULL; + } + return NULL; +} + +static char autoanswer_usage[] = +"Usage: autoanswer [on|off]\n" +" Enables or disables autoanswer feature. If used without\n" +" argument, displays the current on/off status of autoanswer.\n" +" The default value of autoanswer is in 'alsa.conf'.\n"; + +static int console_answer(int fd, int argc, char *argv[]) +{ + int res = RESULT_SUCCESS; + if (argc != 1) + return RESULT_SHOWUSAGE; + ast_mutex_lock(&alsalock); + if (!alsa.owner) { + ast_cli(fd, "No one is calling us\n"); + res = RESULT_FAILURE; + } else { + hookstate = 1; + cursound = -1; + grab_owner(); + if (alsa.owner) { + struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER }; + ast_queue_frame(alsa.owner, &f); + ast_mutex_unlock(&alsa.owner->lock); + } + answer_sound(); + } + snd_pcm_prepare(alsa.icard); + snd_pcm_start(alsa.icard); + ast_mutex_unlock(&alsalock); + return RESULT_SUCCESS; +} + +static char sendtext_usage[] = +"Usage: send text <message>\n" +" Sends a text message for display on the remote terminal.\n"; + +static int console_sendtext(int fd, int argc, char *argv[]) +{ + int tmparg = 2; + int res = RESULT_SUCCESS; + if (argc < 2) + return RESULT_SHOWUSAGE; + ast_mutex_lock(&alsalock); + if (!alsa.owner) { + ast_cli(fd, "No one is calling us\n"); + res = RESULT_FAILURE; + } else { + struct ast_frame f = { AST_FRAME_TEXT, 0 }; + char text2send[256] = ""; + text2send[0] = '\0'; + while(tmparg < argc) { + strncat(text2send, argv[tmparg++], sizeof(text2send) - strlen(text2send) - 1); + strncat(text2send, " ", sizeof(text2send) - strlen(text2send) - 1); + } + text2send[strlen(text2send) - 1] = '\n'; + f.data = text2send; + f.datalen = strlen(text2send) + 1; + grab_owner(); + if (alsa.owner) { + ast_queue_frame(alsa.owner, &f); + f.frametype = AST_FRAME_CONTROL; + f.subclass = AST_CONTROL_ANSWER; + f.data = NULL; + f.datalen = 0; + ast_queue_frame(alsa.owner, &f); + ast_mutex_unlock(&alsa.owner->lock); + } + } + ast_mutex_unlock(&alsalock); + return res; +} + +static char answer_usage[] = +"Usage: answer\n" +" Answers an incoming call on the console (ALSA) channel.\n"; + +static int console_hangup(int fd, int argc, char *argv[]) +{ + int res = RESULT_SUCCESS; + if (argc != 1) + return RESULT_SHOWUSAGE; + cursound = -1; + ast_mutex_lock(&alsalock); + if (!alsa.owner && !hookstate) { + ast_cli(fd, "No call to hangup up\n"); + res = RESULT_FAILURE; + } else { + hookstate = 0; + grab_owner(); + if (alsa.owner) { + ast_queue_hangup(alsa.owner); + ast_mutex_unlock(&alsa.owner->lock); + } + } + ast_mutex_unlock(&alsalock); + return res; +} + +static char hangup_usage[] = +"Usage: hangup\n" +" Hangs up any call currently placed on the console.\n"; + + +static int console_dial(int fd, int argc, char *argv[]) +{ + char tmp[256], *tmp2; + char *mye, *myc; + char *d; + int res = RESULT_SUCCESS; + if ((argc != 1) && (argc != 2)) + return RESULT_SHOWUSAGE; + ast_mutex_lock(&alsalock); + if (alsa.owner) { + if (argc == 2) { + d = argv[1]; + grab_owner(); + if (alsa.owner) { + struct ast_frame f = { AST_FRAME_DTMF }; + while(*d) { + f.subclass = *d; + ast_queue_frame(alsa.owner, &f); + d++; + } + ast_mutex_unlock(&alsa.owner->lock); + } + } else { + ast_cli(fd, "You're already in a call. You can use this only to dial digits until you hangup\n"); + res = RESULT_FAILURE; + } + } else { + mye = exten; + myc = context; + if (argc == 2) { + char *stringp=NULL; + strncpy(tmp, argv[1], sizeof(tmp)-1); + stringp=tmp; + strsep(&stringp, "@"); + tmp2 = strsep(&stringp, "@"); + if (!ast_strlen_zero(tmp)) + mye = tmp; + if (!ast_strlen_zero(tmp2)) + myc = tmp2; + } + if (ast_exists_extension(NULL, myc, mye, 1, NULL)) { + strncpy(alsa.exten, mye, sizeof(alsa.exten)-1); + strncpy(alsa.context, myc, sizeof(alsa.context)-1); + hookstate = 1; + alsa_new(&alsa, AST_STATE_RINGING); + } else + ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc); + } + ast_mutex_unlock(&alsalock); + return res; +} + +static char dial_usage[] = +"Usage: dial [extension[@context]]\n" +" Dials a given extension (and context if specified)\n"; + + +static struct ast_cli_entry myclis[] = { + { { "answer", NULL }, console_answer, "Answer an incoming console call", answer_usage }, + { { "hangup", NULL }, console_hangup, "Hangup a call on the console", hangup_usage }, + { { "dial", NULL }, console_dial, "Dial an extension on the console", dial_usage }, + { { "send", "text", NULL }, console_sendtext, "Send text to the remote device", sendtext_usage }, + { { "autoanswer", NULL }, console_autoanswer, "Sets/displays autoanswer", autoanswer_usage, autoanswer_complete } +}; + +int load_module() +{ + int res; + int x; + struct ast_config *cfg; + struct ast_variable *v; + if ((cfg = ast_config_load(config))) { + v = ast_variable_browse(cfg, "general"); + while(v) { + 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")) + strncpy(context, v->value, sizeof(context)-1); + else if (!strcasecmp(v->name, "language")) + strncpy(language, v->value, sizeof(language)-1); + else if (!strcasecmp(v->name, "extension")) + strncpy(exten, v->value, sizeof(exten)-1); + else if (!strcasecmp(v->name, "input_device")) + strncpy(indevname, v->value, sizeof(indevname)-1); + else if (!strcasecmp(v->name, "output_device")) + strncpy(outdevname, v->value, sizeof(outdevname)-1); + v=v->next; + } + 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 '%s'\n", type); + return -1; + } + for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++) + ast_cli_register(myclis + x); + ast_pthread_create(&sthread, NULL, sound_thread, NULL); +#ifdef ALSA_MONITOR + if (alsa_monitor_start()) { + ast_log(LOG_ERROR, "Problem starting Monitoring\n"); + } +#endif + return 0; +} + + + +int unload_module() +{ + int x; + + ast_channel_unregister(&alsa_tech); + for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++) + ast_cli_unregister(myclis + x); + if (alsa.icard) + snd_pcm_close(alsa.icard); + if (alsa.ocard) + snd_pcm_close(alsa.ocard); + if (sndcmd[0] > 0) { + close(sndcmd[0]); + close(sndcmd[1]); + } + if (alsa.owner) + ast_softhangup(alsa.owner, AST_SOFTHANGUP_APPUNLOAD); + if (alsa.owner) + return -1; + return 0; +} + +char *description() +{ + return (char *) desc; +} + +int usecount() +{ + return usecnt; +} + +char *key() +{ + return ASTERISK_GPL_KEY; +} |