diff options
-rw-r--r-- | apps/app_chanspy.c | 146 | ||||
-rw-r--r-- | apps/app_mixmonitor.c | 135 | ||||
-rw-r--r-- | funcs/func_volume.c | 163 | ||||
-rw-r--r-- | include/asterisk/audiohook.h | 185 | ||||
-rw-r--r-- | include/asterisk/channel.h | 7 | ||||
-rw-r--r-- | include/asterisk/chanspy.h | 150 | ||||
-rw-r--r-- | include/asterisk/slinfactory.h | 1 | ||||
-rw-r--r-- | main/Makefile | 2 | ||||
-rw-r--r-- | main/audiohook.c | 625 | ||||
-rw-r--r-- | main/channel.c | 602 | ||||
-rw-r--r-- | main/slinfactory.c | 18 |
11 files changed, 1117 insertions, 917 deletions
diff --git a/apps/app_chanspy.c b/apps/app_chanspy.c index edfd5290d..bb1510bd2 100644 --- a/apps/app_chanspy.c +++ b/apps/app_chanspy.c @@ -40,7 +40,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #include "asterisk/file.h" #include "asterisk/logger.h" #include "asterisk/channel.h" -#include "asterisk/chanspy.h" +#include "asterisk/audiohook.h" #include "asterisk/features.h" #include "asterisk/options.h" #include "asterisk/app.h" @@ -166,7 +166,8 @@ AST_APP_OPTIONS(spy_opts, { struct chanspy_translation_helper { /* spy data */ - struct ast_channel_spy spy; + struct ast_audiohook spy_audiohook; + struct ast_audiohook whisper_audiohook; int fd; int volfactor; }; @@ -185,15 +186,18 @@ static void spy_release(struct ast_channel *chan, void *data) static int spy_generate(struct ast_channel *chan, void *data, int len, int samples) { struct chanspy_translation_helper *csth = data; - struct ast_frame *f; + struct ast_frame *f = NULL; - if (csth->spy.status != CHANSPY_RUNNING) + ast_audiohook_lock(&csth->spy_audiohook); + if (csth->spy_audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING) { /* Channel is already gone more than likely */ + ast_audiohook_unlock(&csth->spy_audiohook); return -1; + } - ast_mutex_lock(&csth->spy.lock); - f = ast_channel_spy_read_frame(&csth->spy, samples); - ast_mutex_unlock(&csth->spy.lock); + f = ast_audiohook_read_frame(&csth->spy_audiohook, samples, AST_AUDIOHOOK_DIRECTION_BOTH, AST_FORMAT_SLINEAR); + + ast_audiohook_unlock(&csth->spy_audiohook); if (!f) return 0; @@ -217,16 +221,14 @@ static struct ast_generator spygen = { .generate = spy_generate, }; -static int start_spying(struct ast_channel *chan, struct ast_channel *spychan, struct ast_channel_spy *spy) +static int start_spying(struct ast_channel *chan, struct ast_channel *spychan, struct ast_audiohook *audiohook) { - int res; - struct ast_channel *peer; + int res = 0; + struct ast_channel *peer = NULL; ast_log(LOG_NOTICE, "Attaching %s to %s\n", spychan->name, chan->name); - ast_channel_lock(chan); - res = ast_channel_spy_add(chan, spy); - ast_channel_unlock(chan); + res = ast_audiohook_attach(chan, audiohook); if (!res && ast_test_flag(chan, AST_FLAG_NBRIDGE) && (peer = ast_bridged_channel(chan))) ast_softhangup(peer, AST_SOFTHANGUP_UNBRIDGE); @@ -234,35 +236,6 @@ static int start_spying(struct ast_channel *chan, struct ast_channel *spychan, s return res; } -/* Map 'volume' levels from -4 through +4 into - decibel (dB) settings for channel drivers -*/ -static signed char volfactor_map[] = { - -24, - -18, - -12, - -6, - 0, - 6, - 12, - 18, - 24, -}; - -/* attempt to set the desired gain adjustment via the channel driver; - if successful, clear it out of the csth structure so the - generator will not attempt to do the adjustment itself -*/ -static void set_volume(struct ast_channel *chan, struct chanspy_translation_helper *csth) -{ - signed char volume_adjust = volfactor_map[csth->volfactor + 4]; - - if (!ast_channel_setoption(chan, AST_OPTION_TXGAIN, &volume_adjust, sizeof(volume_adjust), 0)) - csth->volfactor = 0; - csth->spy.read_vol_adjustment = csth->volfactor; - csth->spy.write_vol_adjustment = csth->volfactor; -} - static int channel_spy(struct ast_channel *chan, struct ast_channel *spyee, int *volfactor, int fd, const struct ast_flags *flags, char *exitcontext) { @@ -280,49 +253,17 @@ static int channel_spy(struct ast_channel *chan, struct ast_channel *spyee, int ast_verb(2, "Spying on channel %s\n", name); memset(&csth, 0, sizeof(csth)); - ast_set_flag(&csth.spy, CHANSPY_FORMAT_AUDIO); - ast_set_flag(&csth.spy, CHANSPY_TRIGGER_NONE); - if (!ast_test_flag(flags, OPTION_READONLY)) - ast_set_flag(&csth.spy, CHANSPY_MIXAUDIO); - csth.spy.type = "ChanSpy"; - csth.spy.status = CHANSPY_RUNNING; - csth.spy.read_queue.format = AST_FORMAT_SLINEAR; - csth.spy.write_queue.format = AST_FORMAT_SLINEAR; - ast_mutex_init(&csth.spy.lock); - csth.volfactor = *volfactor; - set_volume(chan, &csth); - if (csth.volfactor) { - ast_set_flag(&csth.spy, CHANSPY_READ_VOLADJUST); - csth.spy.read_vol_adjustment = csth.volfactor; - ast_set_flag(&csth.spy, CHANSPY_WRITE_VOLADJUST); - csth.spy.write_vol_adjustment = csth.volfactor; - } - csth.fd = fd; + + ast_audiohook_init(&csth.spy_audiohook, AST_AUDIOHOOK_TYPE_SPY, "ChanSpy"); - if (start_spying(spyee, chan, &csth.spy)) { - ast_mutex_destroy(&csth.spy.lock); + if (start_spying(spyee, chan, &csth.spy_audiohook)) { + ast_audiohook_destroy(&csth.spy_audiohook); return 0; } if (ast_test_flag(flags, OPTION_WHISPER)) { - struct ast_filestream *beepstream; - int old_write_format = 0; - - ast_channel_whisper_start(csth.spy.chan); - old_write_format = chan->writeformat; - if ((beepstream = ast_openstream_full(chan, "beep", chan->language, 1))) { - struct ast_frame *f; - - while ((f = ast_readframe(beepstream))) { - ast_channel_whisper_feed(csth.spy.chan, f); - ast_frfree(f); - } - - ast_closestream(beepstream); - chan->stream = NULL; - } - if (old_write_format) - ast_set_write_format(chan, old_write_format); + ast_audiohook_init(&csth.whisper_audiohook, AST_AUDIOHOOK_TYPE_WHISPER, "ChanSpy"); + start_spying(spyee, chan, &csth.whisper_audiohook); } if (ast_test_flag(flags, OPTION_PRIVATE)) @@ -344,21 +285,20 @@ static int channel_spy(struct ast_channel *chan, struct ast_channel *spyee, int has arrived, since the spied-on channel could have gone away while we were waiting */ - while ((res = ast_waitfor(chan, -1) > -1) && - csth.spy.status == CHANSPY_RUNNING && - csth.spy.chan) { + while ((res = ast_waitfor(chan, -1) > -1) && csth.spy_audiohook.status == AST_AUDIOHOOK_STATUS_RUNNING) { if (!(f = ast_read(chan)) || ast_check_hangup(chan)) { running = -1; break; } - if (ast_test_flag(flags, OPTION_WHISPER) && - (f->frametype == AST_FRAME_VOICE)) { - ast_channel_whisper_feed(csth.spy.chan, f); + if (ast_test_flag(flags, OPTION_WHISPER) && f->frametype == AST_FRAME_VOICE) { + ast_audiohook_lock(&csth.whisper_audiohook); + ast_audiohook_write_frame(&csth.whisper_audiohook, AST_AUDIOHOOK_DIRECTION_WRITE, f); + ast_audiohook_unlock(&csth.whisper_audiohook); ast_frfree(f); continue; } - + res = (f->frametype == AST_FRAME_DTMF) ? f->subclass : 0; ast_frfree(f); if (!res) @@ -401,37 +341,25 @@ static int channel_spy(struct ast_channel *chan, struct ast_channel *spyee, int if (*volfactor > 4) *volfactor = -4; ast_verb(3, "Setting spy volume on %s to %d\n", chan->name, *volfactor); - csth.volfactor = *volfactor; - set_volume(chan, &csth); - if (csth.volfactor) { - ast_set_flag(&csth.spy, CHANSPY_READ_VOLADJUST); - csth.spy.read_vol_adjustment = csth.volfactor; - ast_set_flag(&csth.spy, CHANSPY_WRITE_VOLADJUST); - csth.spy.write_vol_adjustment = csth.volfactor; - } else { - ast_clear_flag(&csth.spy, CHANSPY_READ_VOLADJUST); - ast_clear_flag(&csth.spy, CHANSPY_WRITE_VOLADJUST); - } } } - if (ast_test_flag(flags, OPTION_WHISPER) && csth.spy.chan) - ast_channel_whisper_stop(csth.spy.chan); - if (ast_test_flag(flags, OPTION_PRIVATE)) ast_channel_stop_silence_generator(chan, silgen); else ast_deactivate_generator(chan); - csth.spy.status = CHANSPY_DONE; - - /* If a channel still exists on our spy structure then we need to remove ourselves */ - if (csth.spy.chan) { - ast_channel_lock(csth.spy.chan); - ast_channel_spy_remove(csth.spy.chan, &csth.spy); - ast_channel_unlock(csth.spy.chan); + if (ast_test_flag(flags, OPTION_WHISPER)) { + ast_audiohook_lock(&csth.whisper_audiohook); + ast_audiohook_detach(&csth.whisper_audiohook); + ast_audiohook_unlock(&csth.whisper_audiohook); + ast_audiohook_destroy(&csth.whisper_audiohook); } - ast_channel_spy_free(&csth.spy); + + ast_audiohook_lock(&csth.spy_audiohook); + ast_audiohook_detach(&csth.spy_audiohook); + ast_audiohook_unlock(&csth.spy_audiohook); + ast_audiohook_destroy(&csth.spy_audiohook); if (option_verbose >= 2) ast_verbose(VERBOSE_PREFIX_2 "Done Spying on channel %s\n", name); diff --git a/apps/app_mixmonitor.c b/apps/app_mixmonitor.c index cd532b321..614adcba1 100644 --- a/apps/app_mixmonitor.c +++ b/apps/app_mixmonitor.c @@ -45,7 +45,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #include "asterisk/file.h" #include "asterisk/logger.h" #include "asterisk/channel.h" -#include "asterisk/chanspy.h" +#include "asterisk/audiohook.h" #include "asterisk/pbx.h" #include "asterisk/module.h" #include "asterisk/lock.h" @@ -93,7 +93,7 @@ struct module_symbols *me; static const char *mixmonitor_spy_type = "MixMonitor"; struct mixmonitor { - struct ast_channel_spy spy; + struct ast_audiohook audiohook; char *filename; char *post_process; char *name; @@ -123,17 +123,15 @@ AST_APP_OPTIONS(mixmonitor_opts, { AST_APP_OPTION_ARG('W', MUXFLAG_VOLUME, OPT_ARG_VOLUME), }); -static int startmon(struct ast_channel *chan, struct ast_channel_spy *spy) +static int startmon(struct ast_channel *chan, struct ast_audiohook *audiohook) { - struct ast_channel *peer; - int res; + struct ast_channel *peer = NULL; + int res = 0; if (!chan) return -1; - ast_channel_lock(chan); - res = ast_channel_spy_add(chan, spy); - ast_channel_unlock(chan); + ast_audiohook_attach(chan, audiohook); if (!res && ast_test_flag(chan, AST_FLAG_NBRIDGE) && (peer = ast_bridged_channel(chan))) ast_softhangup(peer, AST_SOFTHANGUP_UNBRIDGE); @@ -146,7 +144,6 @@ static int startmon(struct ast_channel *chan, struct ast_channel_spy *spy) static void *mixmonitor_thread(void *obj) { struct mixmonitor *mixmonitor = obj; - struct ast_frame *f = NULL; struct ast_filestream *fs = NULL; unsigned int oflags; char *ext; @@ -155,59 +152,48 @@ static void *mixmonitor_thread(void *obj) if (option_verbose > 1) ast_verbose(VERBOSE_PREFIX_2 "Begin MixMonitor Recording %s\n", mixmonitor->name); - ast_mutex_lock(&mixmonitor->spy.lock); + ast_audiohook_lock(&mixmonitor->audiohook); - while (mixmonitor->spy.chan) { - struct ast_frame *next; - int write; + while (mixmonitor->audiohook.status == AST_AUDIOHOOK_STATUS_RUNNING) { + struct ast_frame *fr = NULL; - ast_channel_spy_trigger_wait(&mixmonitor->spy); - - if (!mixmonitor->spy.chan || mixmonitor->spy.status != CHANSPY_RUNNING) + ast_audiohook_trigger_wait(&mixmonitor->audiohook); + + if (mixmonitor->audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING) break; - - while (1) { - if (!(f = ast_channel_spy_read_frame(&mixmonitor->spy, SAMPLES_PER_FRAME))) - break; - - write = (!ast_test_flag(mixmonitor, MUXFLAG_BRIDGED) || - ast_bridged_channel(mixmonitor->spy.chan)); - - /* it is possible for ast_channel_spy_read_frame() to return a chain - of frames if a queue flush was necessary, so process them - */ - for (; f; f = next) { - next = AST_LIST_NEXT(f, frame_list); - if (write && errflag == 0) { - if (!fs) { - /* Determine creation flags and filename plus extension for filestream */ - oflags = O_CREAT | O_WRONLY; - oflags |= ast_test_flag(mixmonitor, MUXFLAG_APPEND) ? O_APPEND : O_TRUNC; - - if ((ext = strrchr(mixmonitor->filename, '.'))) - *(ext++) = '\0'; - else - ext = "raw"; - - /* Move onto actually creating the filestream */ - if (!(fs = ast_writefile(mixmonitor->filename, ext, NULL, oflags, 0, 0644))) { - ast_log(LOG_ERROR, "Cannot open %s.%s\n", mixmonitor->filename, ext); - errflag = 1; - } - - } - if (fs) - ast_writestream(fs, f); - } - ast_frame_free(f, 0); + + if (!(fr = ast_audiohook_read_frame(&mixmonitor->audiohook, SAMPLES_PER_FRAME, AST_AUDIOHOOK_DIRECTION_BOTH, AST_FORMAT_SLINEAR))) + continue; + + /* Initialize the file if not already done so */ + if (!fs && !errflag) { + oflags = O_CREAT | O_WRONLY; + oflags |= ast_test_flag(mixmonitor, MUXFLAG_APPEND) ? O_APPEND : O_TRUNC; + + if ((ext = strrchr(mixmonitor->filename, '.'))) + *(ext++) = '\0'; + else + ext = "raw"; + + if (!(fs = ast_writefile(mixmonitor->filename, ext, NULL, oflags, 0, 0644))) { + ast_log(LOG_ERROR, "Cannot open %s.%s\n", mixmonitor->filename, ext); + errflag = 1; } } + + /* Write out frame */ + if (fs) + ast_writestream(fs, fr); + + /* All done! free it. */ + ast_frame_free(fr, 0); + } - ast_mutex_unlock(&mixmonitor->spy.lock); + ast_audiohook_detach(&mixmonitor->audiohook); + ast_audiohook_unlock(&mixmonitor->audiohook); + ast_audiohook_destroy(&mixmonitor->audiohook); - ast_channel_spy_free(&mixmonitor->spy); - if (option_verbose > 1) ast_verbose(VERBOSE_PREFIX_2 "End MixMonitor Recording %s\n", mixmonitor->name); @@ -270,27 +256,17 @@ static void launch_monitor_thread(struct ast_channel *chan, const char *filename strcpy(mixmonitor->filename, filename); /* Setup the actual spy before creating our thread */ - ast_set_flag(&mixmonitor->spy, CHANSPY_FORMAT_AUDIO); - ast_set_flag(&mixmonitor->spy, CHANSPY_MIXAUDIO); - mixmonitor->spy.type = mixmonitor_spy_type; - mixmonitor->spy.status = CHANSPY_RUNNING; - mixmonitor->spy.read_queue.format = AST_FORMAT_SLINEAR; - mixmonitor->spy.write_queue.format = AST_FORMAT_SLINEAR; - if (readvol) { - ast_set_flag(&mixmonitor->spy, CHANSPY_READ_VOLADJUST); - mixmonitor->spy.read_vol_adjustment = readvol; - } - if (writevol) { - ast_set_flag(&mixmonitor->spy, CHANSPY_WRITE_VOLADJUST); - mixmonitor->spy.write_vol_adjustment = writevol; + if (ast_audiohook_init(&mixmonitor->audiohook, AST_AUDIOHOOK_TYPE_SPY, mixmonitor_spy_type)) { + free(mixmonitor); + return; } - ast_mutex_init(&mixmonitor->spy.lock); - if (startmon(chan, &mixmonitor->spy)) { + ast_set_flag(&mixmonitor->audiohook, AST_AUDIOHOOK_TRIGGER_WRITE); + + if (startmon(chan, &mixmonitor->audiohook)) { ast_log(LOG_WARNING, "Unable to add '%s' spy to channel '%s'\n", - mixmonitor->spy.type, chan->name); - /* Since we couldn't add ourselves - bail out! */ - ast_mutex_destroy(&mixmonitor->spy.lock); + mixmonitor_spy_type, chan->name); + ast_audiohook_destroy(&mixmonitor->audiohook); ast_free(mixmonitor); return; } @@ -382,9 +358,7 @@ static int mixmonitor_exec(struct ast_channel *chan, void *data) static int stop_mixmonitor_exec(struct ast_channel *chan, void *data) { - ast_channel_lock(chan); - ast_channel_spy_stop_by_type(chan, mixmonitor_spy_type); - ast_channel_unlock(chan); + ast_audiohook_detach_source(chan, mixmonitor_spy_type); return 0; } @@ -400,12 +374,13 @@ static int mixmonitor_cli(int fd, int argc, char **argv) return RESULT_SUCCESS; } - if (!strcasecmp(argv[1], "start")) + if (!strcasecmp(argv[1], "start")) { mixmonitor_exec(chan, argv[3]); - else if (!strcasecmp(argv[1], "stop")) - ast_channel_spy_stop_by_type(chan, mixmonitor_spy_type); - - ast_channel_unlock(chan); + ast_channel_unlock(chan); + } else { + ast_channel_unlock(chan); + ast_audiohook_detach_source(chan, mixmonitor_spy_type); + } return RESULT_SUCCESS; } diff --git a/funcs/func_volume.c b/funcs/func_volume.c new file mode 100644 index 000000000..79fc13f30 --- /dev/null +++ b/funcs/func_volume.c @@ -0,0 +1,163 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 2007, Digium, Inc. + * + * Joshua Colp <jcolp@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 Technology independent volume control + * + * \author Joshua Colp <jcolp@digium.com> + * + * \ingroup functions + * + */ + +#include "asterisk.h" + +ASTERISK_FILE_VERSION(__FILE__, "$Revision$") + +#include <stdlib.h> + +#include "asterisk/module.h" +#include "asterisk/channel.h" +#include "asterisk/pbx.h" +#include "asterisk/utils.h" +#include "asterisk/linkedlists.h" +#include "asterisk/audiohook.h" + +struct volume_information { + struct ast_audiohook audiohook; + int tx_gain; + int rx_gain; +}; + +static void destroy_callback(void *data) +{ + struct volume_information *vi = data; + + /* Destroy the audiohook, and destroy ourselves */ + ast_audiohook_destroy(&vi->audiohook); + free(vi); + + return; +} + +/*! \brief Static structure for datastore information */ +static const struct ast_datastore_info volume_datastore = { + .type = "volume", + .destroy = destroy_callback +}; + +static int volume_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction) +{ + struct ast_datastore *datastore = NULL; + struct volume_information *vi = NULL; + int *gain = NULL; + + /* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */ + if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) + return 0; + + /* Grab datastore which contains our gain information */ + if (!(datastore = ast_channel_datastore_find(chan, &volume_datastore, NULL))) + return 0; + + vi = datastore->data; + + /* If this is DTMF then allow them to increase/decrease the gains */ + if (frame->frametype == AST_FRAME_DTMF) { + /* Only use DTMF coming from the source... not going to it */ + if (direction != AST_AUDIOHOOK_DIRECTION_READ) + return 0; + if (frame->subclass == '*') { + vi->tx_gain += 1; + vi->rx_gain += 1; + } else if (frame->subclass == '#') { + vi->tx_gain -= 1; + vi->rx_gain -= 1; + } + } else if (frame->frametype == AST_FRAME_VOICE) { + /* Based on direction of frame grab the gain, and confirm it is applicable */ + if (!(gain = (direction == AST_AUDIOHOOK_DIRECTION_READ) ? &vi->rx_gain : &vi->tx_gain) || !*gain) + return 0; + /* Apply gain to frame... easy as pi */ + ast_frame_adjust_volume(frame, *gain); + } + + return 0; +} + +static int volume_write(struct ast_channel *chan, const char *cmd, char *data, const char *value) +{ + struct ast_datastore *datastore = NULL; + struct volume_information *vi = NULL; + int is_new = 0; + + if (!(datastore = ast_channel_datastore_find(chan, &volume_datastore, NULL))) { + /* Allocate a new datastore to hold the reference to this volume and audiohook information */ + if (!(datastore = ast_channel_datastore_alloc(&volume_datastore, NULL))) + return 0; + if (!(vi = ast_calloc(1, sizeof(*vi)))) { + ast_channel_datastore_free(datastore); + return 0; + } + ast_audiohook_init(&vi->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Volume"); + vi->audiohook.manipulate_callback = volume_callback; + ast_set_flag(&vi->audiohook, AST_AUDIOHOOK_WANTS_DTMF); + is_new = 1; + } else { + vi = datastore->data; + } + + /* Adjust gain on volume information structure */ + if (!strcasecmp(data, "tx")) + vi->tx_gain = atoi(value); + else if (!strcasecmp(data, "rx")) + vi->rx_gain = atoi(value); + + if (is_new) { + datastore->data = vi; + ast_channel_datastore_add(chan, datastore); + ast_audiohook_attach(chan, &vi->audiohook); + } + + return 0; +} + +static struct ast_custom_function volume_function = { + .name = "VOLUME", + .synopsis = "Set the TX or RX volume of a channel", + .syntax = "VOLUME(TX|RX)", + .desc = + " The VOLUME function can be used to increase or decrease the tx or\n" + "rx gain of any channel. For example:\n" + " Set(VOLUME(TX)=3)\n" + " Set(VOLUME(RX)=2)\n", + .write = volume_write, +}; + +static int unload_module(void) +{ + return ast_custom_function_unregister(&volume_function); +} + +static int load_module(void) +{ + return ast_custom_function_register(&volume_function); +} + +AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Technology independent volume control"); diff --git a/include/asterisk/audiohook.h b/include/asterisk/audiohook.h new file mode 100644 index 000000000..a374a630a --- /dev/null +++ b/include/asterisk/audiohook.h @@ -0,0 +1,185 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 1999 - 2007, Digium, Inc. + * + * Joshua Colp <jcolp@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 Audiohooks Architecture + */ + +#ifndef _ASTERISK_AUDIOHOOK_H +#define _ASTERISK_AUDIOHOOK_H + +#if defined(__cplusplus) || defined(c_plusplus) +extern "C" { +#endif + +#include "asterisk/slinfactory.h" + +enum ast_audiohook_type { + AST_AUDIOHOOK_TYPE_SPY = 0, /*!< Audiohook wants to receive audio */ + AST_AUDIOHOOK_TYPE_WHISPER, /*!< Audiohook wants to provide audio to be mixed with existing audio */ + AST_AUDIOHOOK_TYPE_MANIPULATE, /*!< Audiohook wants to manipulate the audio */ +}; + +enum ast_audiohook_status { + AST_AUDIOHOOK_STATUS_NEW = 0, /*!< Audiohook was just created, not in use yet */ + AST_AUDIOHOOK_STATUS_RUNNING, /*!< Audiohook is running on a channel */ + AST_AUDIOHOOK_STATUS_SHUTDOWN, /*!< Audiohook is being shutdown */ + AST_AUDIOHOOK_STATUS_DONE, /*!< Audiohook has shutdown and is not running on a channel any longer */ +}; + +enum ast_audiohook_direction { + AST_AUDIOHOOK_DIRECTION_READ = 0, /*!< Reading audio in */ + AST_AUDIOHOOK_DIRECTION_WRITE, /*!< Writing audio out */ + AST_AUDIOHOOK_DIRECTION_BOTH, /*!< Both reading audio in and writing audio out */ +}; + +enum ast_audiohook_flags { + AST_AUDIOHOOK_TRIGGER_MODE = (3 << 0), /*!< When audiohook should be triggered to do something */ + AST_AUDIOHOOK_TRIGGER_READ = (1 << 0), /*!< Audiohook wants to be triggered when reading audio in */ + AST_AUDIOHOOK_TRIGGER_WRITE = (2 << 0), /*!< Audiohook wants to be triggered when writing audio out */ + AST_AUDIOHOOK_WANTS_DTMF = (1 << 1), /*!< Audiohook also wants to receive DTMF frames */ +}; + +struct ast_audiohook; + +/*! \brief Callback function for manipulate audiohook type + * \param audiohook Audiohook structure + * \param chan Channel + * \param frame Frame of audio to manipulate + * \param direction Direction frame came from + * \return Returns 0 on success, -1 on failure + * \note An audiohook does not have any reference to a private data structure for manipulate types. It is up to the manipulate callback to store this data + * via it's own method. An example would be datastores. + */ +typedef int (*ast_audiohook_manipulate_callback)(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction); + +struct ast_audiohook_options { + int read_volume; /*!< Volume adjustment on frames read from the channel the hook is on */ + int write_volume; /*!< Volume adjustment on frames written to the channel the hook is on */ +}; + +struct ast_audiohook { + ast_mutex_t lock; /*!< Lock that protects the audiohook structure */ + ast_cond_t trigger; /*!< Trigger condition (if enabled) */ + enum ast_audiohook_type type; /*!< Type of audiohook */ + enum ast_audiohook_status status; /*!< Status of the audiohook */ + const char *source; /*!< Who this audiohook ultimately belongs to */ + unsigned int flags; /*!< Flags on the audiohook */ + struct ast_slinfactory read_factory; /*!< Factory where frames read from the channel, or read from the whisper source will go through */ + struct ast_slinfactory write_factory; /*!< Factory where frames written to the channel will go through */ + int format; /*!< Format translation path is setup as */ + struct ast_trans_pvt *trans_pvt; /*!< Translation path for reading frames */ + ast_audiohook_manipulate_callback manipulate_callback; /*!< Manipulation callback */ + struct ast_audiohook_options options; /*!< Applicable options */ + AST_LIST_ENTRY(ast_audiohook) list; /*!< Linked list information */ +}; + +struct ast_audiohook_list; + +/*! \brief Initialize an audiohook structure + * \param audiohook Audiohook structure + * \param type Type of audiohook to initialize this as + * \param source Who is initializing this audiohook + * \return Returns 0 on success, -1 on failure + */ +int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source); + +/*! \brief Destroys an audiohook structure + * \param audiohook Audiohook structure + * \return Returns 0 on success, -1 on failure + */ +int ast_audiohook_destroy(struct ast_audiohook *audiohook); + +/*! \brief Writes a frame into the audiohook structure + * \param audiohook Audiohook structure + * \param direction Direction the audio frame came from + * \param frame Frame to write in + * \return Returns 0 on success, -1 on failure + */ +int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohook_direction direction, struct ast_frame *frame); + +/*! \brief Reads a frame in from the audiohook structure + * \param audiohook Audiohook structure + * \param samples Number of samples wanted + * \param direction Direction the audio frame came from + * \param format Format of frame remote side wants back + * \return Returns frame on success, NULL on failure + */ +struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, int format); + +/*! \brief Attach audiohook to channel + * \param chan Channel + * \param audiohook Audiohook structure + * \return Returns 0 on success, -1 on failure + */ +int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook); + +/*! \brief Detach audiohook from channel + * \param audiohook Audiohook structure + * \return Returns 0 on success, -1 on failure + */ +int ast_audiohook_detach(struct ast_audiohook *audiohook); + +/*! \brief Detach audiohooks from list and destroy said list + * \param audiohook_list List of audiohooks + * \return Returns 0 on success, -1 on failure + */ +int ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list); + +/*! \brief Detach specified source audiohook from channel + * \param chan Channel to detach from + * \param source Name of source to detach + * \return Returns 0 on success, -1 on failure + */ +int ast_audiohook_detach_source(struct ast_channel *chan, const char *source); + +/*! \brief Pass a frame off to be handled by the audiohook core + * \param chan Channel that the list is coming off of + * \param audiohook_list List of audiohooks + * \param direction Direction frame is coming in from + * \param frame The frame itself + * \return Return frame on success, NULL on failure + */ +struct ast_frame *ast_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame); + +/*! \brief Wait for audiohook trigger to be triggered + * \param audiohook Audiohook to wait on + */ +void ast_audiohook_trigger_wait(struct ast_audiohook *audiohook); + +/*! \brief Lock an audiohook + * \param audiohook Audiohook structure + */ +static inline int ast_audiohook_lock(struct ast_audiohook *audiohook) +{ + return ast_mutex_lock(&audiohook->lock); +} + +/*! \brief Unlock an audiohook + * \param audiohook Audiohook structure + */ +static inline int ast_audiohook_unlock(struct ast_audiohook *audiohook) +{ + return ast_mutex_unlock(&audiohook->lock); +} + +#if defined(__cplusplus) || defined(c_plusplus) +} +#endif + +#endif /* _ASTERISK_AUDIOHOOK_H */ diff --git a/include/asterisk/channel.h b/include/asterisk/channel.h index 3e63259aa..39f2c636d 100644 --- a/include/asterisk/channel.h +++ b/include/asterisk/channel.h @@ -316,9 +316,6 @@ struct ast_channel_tech { int (* func_channel_write)(struct ast_channel *chan, const char *function, char *data, const char *value); }; -struct ast_channel_spy_list; /*!< \todo Add explanation here */ -struct ast_channel_whisper_buffer; /*!< \todo Add explanation here */ - /*! * The high bit of the frame count is used as a debug marker, so * increments of the counters must be done with care. @@ -481,8 +478,8 @@ struct ast_channel { int rawreadformat; /*!< Raw read format */ int rawwriteformat; /*!< Raw write format */ - struct ast_channel_spy_list *spies; /*!< Chan Spy stuff */ - struct ast_channel_whisper_buffer *whisper; /*!< Whisper Paging buffer */ + struct ast_audiohook_list *audiohooks; + AST_LIST_ENTRY(ast_channel) chan_list; /*!< For easy linking */ struct ast_jb jb; /*!< The jitterbuffer state */ diff --git a/include/asterisk/chanspy.h b/include/asterisk/chanspy.h deleted file mode 100644 index 8550210d0..000000000 --- a/include/asterisk/chanspy.h +++ /dev/null @@ -1,150 +0,0 @@ -/* - * Asterisk -- An open source telephony toolkit. - * - * Copyright (C) 1999 - 2006, Digium, Inc. - * - * Mark Spencer <markster@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 Asterisk PBX channel spy definitions - */ - -#ifndef _ASTERISK_CHANSPY_H -#define _ASTERISK_CHANSPY_H - -#if defined(__cplusplus) || defined(c_plusplus) -extern "C" { -#endif - -#include "asterisk/linkedlists.h" - -enum chanspy_states { - CHANSPY_NEW = 0, /*!< spy not yet operating */ - CHANSPY_RUNNING = 1, /*!< normal operation, spy is still operating */ - CHANSPY_DONE = 2, /*!< spy is stopped and already removed from channel */ - CHANSPY_STOP = 3, /*!< spy requested to stop, still attached to channel */ -}; - -enum chanspy_flags { - CHANSPY_MIXAUDIO = (1 << 0), - CHANSPY_READ_VOLADJUST = (1 << 1), - CHANSPY_WRITE_VOLADJUST = (1 << 2), - CHANSPY_FORMAT_AUDIO = (1 << 3), - CHANSPY_TRIGGER_MODE = (3 << 4), - CHANSPY_TRIGGER_READ = (1 << 4), - CHANSPY_TRIGGER_WRITE = (2 << 4), - CHANSPY_TRIGGER_NONE = (3 << 4), - CHANSPY_TRIGGER_FLUSH = (1 << 6), -}; - -struct ast_channel_spy_queue { - AST_LIST_HEAD_NOLOCK(, ast_frame) list; - unsigned int samples; - unsigned int format; -}; - -struct ast_channel_spy { - AST_LIST_ENTRY(ast_channel_spy) list; - ast_mutex_t lock; - ast_cond_t trigger; - struct ast_channel *chan; - struct ast_channel_spy_queue read_queue; - struct ast_channel_spy_queue write_queue; - unsigned int flags; - enum chanspy_states status; - const char *type; - /* The volume adjustment values are very straightforward: - positive values cause the samples to be multiplied by that amount - negative values cause the samples to be divided by the absolute value of that amount - */ - int read_vol_adjustment; - int write_vol_adjustment; -}; - -/*! - \brief Adds a spy to a channel, to begin receiving copies of the channel's audio frames. - \param chan The channel to add the spy to. - \param spy A pointer to ast_channel_spy structure describing how the spy is to be used. - \return 0 for success, non-zero for failure - - Note: This function performs no locking; you must hold the channel's lock before - calling this function. - */ -int ast_channel_spy_add(struct ast_channel *chan, struct ast_channel_spy *spy); - -/*! - \brief Remove a spy from a channel. - \param chan The channel to remove the spy from - \param spy The spy to be removed - \return nothing - - Note: This function performs no locking; you must hold the channel's lock before - calling this function. - */ -void ast_channel_spy_remove(struct ast_channel *chan, struct ast_channel_spy *spy); - -/*! - \brief Free a spy. - \param spy The spy to free - \return nothing - - Note: This function MUST NOT be called with the spy locked. -*/ -void ast_channel_spy_free(struct ast_channel_spy *spy); - -/*! - \brief Find all spies of a particular type on a channel and stop them. - \param chan The channel to operate on - \param type A character string identifying the type of spies to be stopped - \return nothing - - Note: This function performs no locking; you must hold the channel's lock before - calling this function. - */ -void ast_channel_spy_stop_by_type(struct ast_channel *chan, const char *type); - -/*! - \brief Read one (or more) frames of audio from a channel being spied upon. - \param spy The spy to operate on - \param samples The number of audio samples to read - \return NULL for failure, one ast_frame pointer, or a chain of ast_frame pointers - - This function can return multiple frames if the spy structure needs to be 'flushed' - due to mismatched queue lengths, or if the spy structure is configured to return - unmixed audio (in which case each call to this function will return a frame of audio - from each side of channel). - - Note: This function performs no locking; you must hold the spy's lock before calling - this function. You must <b>not</b> hold the channel's lock at the same time. - */ -struct ast_frame *ast_channel_spy_read_frame(struct ast_channel_spy *spy, unsigned int samples); - -/*! - \brief Efficiently wait until audio is available for a spy, or an exception occurs. - \param spy The spy to wait on - \return nothing - - Note: The locking rules for this function are non-obvious... first, you must <b>not</b> - hold the channel's lock when calling this function. Second, you must hold the spy's lock - before making the function call; while the function runs the lock will be released, and - when the trigger event occurs, the lock will be re-obtained. This means that when control - returns to your code, you will again hold the spy's lock. - */ -void ast_channel_spy_trigger_wait(struct ast_channel_spy *spy); - -#if defined(__cplusplus) || defined(c_plusplus) -} -#endif - -#endif /* _ASTERISK_CHANSPY_H */ diff --git a/include/asterisk/slinfactory.h b/include/asterisk/slinfactory.h index b81817d6b..3ab42d283 100644 --- a/include/asterisk/slinfactory.h +++ b/include/asterisk/slinfactory.h @@ -46,6 +46,7 @@ void ast_slinfactory_destroy(struct ast_slinfactory *sf); int ast_slinfactory_feed(struct ast_slinfactory *sf, struct ast_frame *f); int ast_slinfactory_read(struct ast_slinfactory *sf, short *buf, size_t samples); unsigned int ast_slinfactory_available(const struct ast_slinfactory *sf); +void ast_slinfactory_flush(struct ast_slinfactory *sf); #if defined(__cplusplus) || defined(c_plusplus) } diff --git a/main/Makefile b/main/Makefile index f3c1b99f9..ba780b281 100644 --- a/main/Makefile +++ b/main/Makefile @@ -26,7 +26,7 @@ OBJS= io.o sched.o logger.o frame.o loader.o config.o channel.o \ utils.o plc.o jitterbuf.o dnsmgr.o devicestate.o \ netsock.o slinfactory.o ast_expr2.o ast_expr2f.o \ cryptostub.o sha1.o http.o fixedjitterbuf.o abstract_jb.o \ - strcompat.o threadstorage.o dial.o event.o adsistub.o + strcompat.o threadstorage.o dial.o event.o adsistub.o audiohook.o # we need to link in the objects statically, not as a library, because # otherwise modules will not have them available if none of the static diff --git a/main/audiohook.c b/main/audiohook.c new file mode 100644 index 000000000..a7600a356 --- /dev/null +++ b/main/audiohook.c @@ -0,0 +1,625 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 1999 - 2007, Digium, Inc. + * + * Joshua Colp <jcolp@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 Audiohooks Architecture + * + * \author Joshua Colp <jcolp@digium.com> + */ + +#include "asterisk.h" + +ASTERISK_FILE_VERSION(__FILE__, "$Revision$") + +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <signal.h> +#include <errno.h> +#include <unistd.h> + +#include "asterisk/logger.h" +#include "asterisk/channel.h" +#include "asterisk/options.h" +#include "asterisk/utils.h" +#include "asterisk/lock.h" +#include "asterisk/linkedlists.h" +#include "asterisk/audiohook.h" +#include "asterisk/slinfactory.h" +#include "asterisk/frame.h" +#include "asterisk/translate.h" + +struct ast_audiohook_translate { + struct ast_trans_pvt *trans_pvt; + int format; +}; + +struct ast_audiohook_list { + struct ast_audiohook_translate in_translate[2]; + struct ast_audiohook_translate out_translate[2]; + AST_LIST_HEAD_NOLOCK(, ast_audiohook) spy_list; + AST_LIST_HEAD_NOLOCK(, ast_audiohook) whisper_list; + AST_LIST_HEAD_NOLOCK(, ast_audiohook) manipulate_list; +}; + +/*! \brief Initialize an audiohook structure + * \param audiohook Audiohook structure + * \return Returns 0 on success, -1 on failure + */ +int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source) +{ + /* Need to keep the type and source */ + audiohook->type = type; + audiohook->source = source; + + /* Initialize lock that protects our audiohook */ + ast_mutex_init(&audiohook->lock); + ast_cond_init(&audiohook->trigger, NULL); + + /* Setup the factories that are needed for this audiohook type */ + switch (type) { + case AST_AUDIOHOOK_TYPE_SPY: + ast_slinfactory_init(&audiohook->read_factory); + case AST_AUDIOHOOK_TYPE_WHISPER: + ast_slinfactory_init(&audiohook->write_factory); + break; + default: + break; + } + + /* Since we are just starting out... this audiohook is new */ + audiohook->status = AST_AUDIOHOOK_STATUS_NEW; + + return 0; +} + +/*! \brief Destroys an audiohook structure + * \param audiohook Audiohook structure + * \return Returns 0 on success, -1 on failure + */ +int ast_audiohook_destroy(struct ast_audiohook *audiohook) +{ + /* Drop the factories used by this audiohook type */ + switch (audiohook->type) { + case AST_AUDIOHOOK_TYPE_SPY: + ast_slinfactory_destroy(&audiohook->read_factory); + case AST_AUDIOHOOK_TYPE_WHISPER: + ast_slinfactory_destroy(&audiohook->write_factory); + break; + default: + break; + } + + /* Destroy translation path if present */ + if (audiohook->trans_pvt) + ast_translator_free_path(audiohook->trans_pvt); + + /* Lock and trigger be gone! */ + ast_cond_destroy(&audiohook->trigger); + ast_mutex_destroy(&audiohook->lock); + + return 0; +} + +/*! \brief Writes a frame into the audiohook structure + * \param audiohook Audiohook structure + * \param direction Direction the audio frame came from + * \param frame Frame to write in + * \return Returns 0 on success, -1 on failure + */ +int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohook_direction direction, struct ast_frame *frame) +{ + struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory); + + /* Write frame out to respective factory */ + ast_slinfactory_feed(factory, frame); + + /* If we need to notify the respective handler of this audiohook, do so */ + switch (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE)) { + case AST_AUDIOHOOK_TRIGGER_READ: + if (direction == AST_AUDIOHOOK_DIRECTION_READ) + ast_cond_signal(&audiohook->trigger); + break; + case AST_AUDIOHOOK_TRIGGER_WRITE: + if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) + ast_cond_signal(&audiohook->trigger); + break; + default: + break; + } + + return 0; +} + +static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction) +{ + struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory); + int vol = (direction == AST_AUDIOHOOK_DIRECTION_READ ? audiohook->options.read_volume : audiohook->options.write_volume); + short buf[samples]; + struct ast_frame frame = { + .frametype = AST_FRAME_VOICE, + .subclass = AST_FORMAT_SLINEAR, + .data = buf, + .datalen = sizeof(buf), + .samples = samples, + }; + + /* Ensure the factory is able to give us the samples we want */ + if (samples > ast_slinfactory_available(factory)) + return NULL; + + /* Read data in from factory */ + if (!ast_slinfactory_read(factory, buf, samples)) + return NULL; + + /* If a volume adjustment needs to be applied apply it */ + if (vol) + ast_frame_adjust_volume(&frame, vol); + + return ast_frdup(&frame); +} + +static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audiohook, size_t samples) +{ + int i = 0; + short buf1[samples], buf2[samples], *read_buf = NULL, *write_buf = NULL, *final_buf = NULL, *data1 = NULL, *data2 = NULL; + struct ast_frame frame = { + .frametype = AST_FRAME_VOICE, + .subclass = AST_FORMAT_SLINEAR, + .data = NULL, + .datalen = sizeof(buf1), + .samples = samples, + }; + + /* Start with the read factory... if there are enough samples, read them in */ + if (ast_slinfactory_available(&audiohook->read_factory) >= samples) { + if (ast_slinfactory_read(&audiohook->read_factory, buf1, samples)) + read_buf = buf1; + /* Adjust read volume if need be */ + if (audiohook->options.read_volume) { + int count = 0; + short adjust_value = abs(audiohook->options.read_volume); + for (count = 0; count < samples; count++) { + if (audiohook->options.read_volume > 0) + ast_slinear_saturated_multiply(&buf1[count], &adjust_value); + else if (audiohook->options.read_volume < 0) + ast_slinear_saturated_divide(&buf1[count], &adjust_value); + } + } + } else if (option_debug) + ast_log(LOG_DEBUG, "Failed to get %zd samples from read factory %p\n", samples, &audiohook->read_factory); + + /* Move on to the write factory... if there are enough samples, read them in */ + if (ast_slinfactory_available(&audiohook->write_factory) >= samples) { + if (ast_slinfactory_read(&audiohook->write_factory, buf2, samples)) + write_buf = buf2; + /* Adjust write volume if need be */ + if (audiohook->options.write_volume) { + int count = 0; + short adjust_value = abs(audiohook->options.write_volume); + for (count = 0; count < samples; count++) { + if (audiohook->options.write_volume > 0) + ast_slinear_saturated_multiply(&buf2[count], &adjust_value); + else if (audiohook->options.write_volume < 0) + ast_slinear_saturated_divide(&buf2[count], &adjust_value); + } + } + } else if (option_debug) + ast_log(LOG_DEBUG, "Failed to get %zd samples from write factory %p\n", samples, &audiohook->write_factory); + + /* Basically we figure out which buffer to use... and if mixing can be done here */ + if (!read_buf && !write_buf) + return NULL; + else if (read_buf && write_buf) { + for (i = 0, data1 = read_buf, data2 = write_buf; i < samples; i++, data1++, data2++) + ast_slinear_saturated_add(data1, data2); + final_buf = buf1; + } else if (read_buf) + final_buf = buf1; + else if (write_buf) + final_buf = buf2; + + /* Make the final buffer part of the frame, so it gets duplicated fine */ + frame.data = final_buf; + + /* Yahoo, a combined copy of the audio! */ + return ast_frdup(&frame); +} + +/*! \brief Reads a frame in from the audiohook structure + * \param audiohook Audiohook structure + * \param samples Number of samples wanted + * \param direction Direction the audio frame came from + * \param format Format of frame remote side wants back + * \return Returns frame on success, NULL on failure + */ +struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, int format) +{ + struct ast_frame *read_frame = NULL, *final_frame = NULL; + + if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ? audiohook_read_frame_both(audiohook, samples) : audiohook_read_frame_single(audiohook, samples, direction)))) + return NULL; + + /* If they don't want signed linear back out, we'll have to send it through the translation path */ + if (format != AST_FORMAT_SLINEAR) { + /* Rebuild translation path if different format then previously */ + if (audiohook->format != format) { + if (audiohook->trans_pvt) { + ast_translator_free_path(audiohook->trans_pvt); + audiohook->trans_pvt = NULL; + } + /* Setup new translation path for this format... if we fail we can't very well return signed linear so free the frame and return nothing */ + if (!(audiohook->trans_pvt = ast_translator_build_path(format, AST_FORMAT_SLINEAR))) { + ast_frfree(read_frame); + return NULL; + } + } + /* Convert to requested format, and allow the read in frame to be freed */ + final_frame = ast_translate(audiohook->trans_pvt, read_frame, 1); + } else { + final_frame = read_frame; + } + + return final_frame; +} + +/*! \brief Attach audiohook to channel + * \param chan Channel + * \param audiohook Audiohook structure + * \return Returns 0 on success, -1 on failure + */ +int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook) +{ + ast_channel_lock(chan); + + if (!chan->audiohooks) { + /* Whoops... allocate a new structure */ + if (!(chan->audiohooks = ast_calloc(1, sizeof(*chan->audiohooks)))) { + ast_channel_unlock(chan); + return -1; + } + AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->spy_list); + AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->whisper_list); + AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->manipulate_list); + } + + /* Drop into respective list */ + if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY) + AST_LIST_INSERT_TAIL(&chan->audiohooks->spy_list, audiohook, list); + else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER) + AST_LIST_INSERT_TAIL(&chan->audiohooks->whisper_list, audiohook, list); + else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE) + AST_LIST_INSERT_TAIL(&chan->audiohooks->manipulate_list, audiohook, list); + + /* Change status over to running since it is now attached */ + audiohook->status = AST_AUDIOHOOK_STATUS_RUNNING; + + ast_channel_unlock(chan); + + return 0; +} + +/*! \brief Detach audiohook from channel + * \param audiohook Audiohook structure + * \return Returns 0 on success, -1 on failure + */ +int ast_audiohook_detach(struct ast_audiohook *audiohook) +{ + if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) + return 0; + + audiohook->status = AST_AUDIOHOOK_STATUS_SHUTDOWN; + + while (audiohook->status != AST_AUDIOHOOK_STATUS_DONE) + ast_audiohook_trigger_wait(audiohook); + + return 0; +} + +/*! \brief Detach audiohooks from list and destroy said list + * \param audiohook_list List of audiohooks + * \return Returns 0 on success, -1 on failure + */ +int ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list) +{ + int i = 0; + struct ast_audiohook *audiohook = NULL; + + /* Drop any spies */ + AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) { + ast_audiohook_lock(audiohook); + AST_LIST_REMOVE_CURRENT(&audiohook_list->spy_list, list); + audiohook->status = AST_AUDIOHOOK_STATUS_DONE; + ast_cond_signal(&audiohook->trigger); + ast_audiohook_unlock(audiohook); + } + AST_LIST_TRAVERSE_SAFE_END + + /* Drop any whispering sources */ + AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) { + ast_audiohook_lock(audiohook); + AST_LIST_REMOVE_CURRENT(&audiohook_list->whisper_list, list); + audiohook->status = AST_AUDIOHOOK_STATUS_DONE; + ast_cond_signal(&audiohook->trigger); + ast_audiohook_unlock(audiohook); + } + AST_LIST_TRAVERSE_SAFE_END + + /* Drop any manipulaters */ + AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) { + ast_audiohook_lock(audiohook); + ast_mutex_lock(&audiohook->lock); + AST_LIST_REMOVE_CURRENT(&audiohook_list->manipulate_list, list); + audiohook->status = AST_AUDIOHOOK_STATUS_DONE; + ast_audiohook_unlock(audiohook); + audiohook->manipulate_callback(audiohook, NULL, NULL, 0); + } + AST_LIST_TRAVERSE_SAFE_END + + /* Drop translation paths if present */ + for (i = 0; i < 2; i++) { + if (audiohook_list->in_translate[i].trans_pvt) + ast_translator_free_path(audiohook_list->in_translate[i].trans_pvt); + if (audiohook_list->out_translate[i].trans_pvt) + ast_translator_free_path(audiohook_list->out_translate[i].trans_pvt); + } + + /* Free ourselves */ + ast_free(audiohook_list); + + return 0; +} + +static struct ast_audiohook *find_audiohook_by_source(struct ast_audiohook_list *audiohook_list, const char *source) +{ + struct ast_audiohook *audiohook = NULL; + + AST_LIST_TRAVERSE(&audiohook_list->spy_list, audiohook, list) { + if (!strcasecmp(audiohook->source, source)) + return audiohook; + } + + AST_LIST_TRAVERSE(&audiohook_list->whisper_list, audiohook, list) { + if (!strcasecmp(audiohook->source, source)) + return audiohook; + } + + AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, audiohook, list) { + if (!strcasecmp(audiohook->source, source)) + return audiohook; + } + + return NULL; +} + +/*! \brief Detach specified source audiohook from channel + * \param chan Channel to detach from + * \param source Name of source to detach + * \return Returns 0 on success, -1 on failure + */ +int ast_audiohook_detach_source(struct ast_channel *chan, const char *source) +{ + struct ast_audiohook *audiohook = NULL; + + ast_channel_lock(chan); + + /* Ensure the channel has audiohooks on it */ + if (!chan->audiohooks) { + ast_channel_unlock(chan); + return -1; + } + + audiohook = find_audiohook_by_source(chan->audiohooks, source); + + ast_channel_unlock(chan); + + if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE) + audiohook->status = AST_AUDIOHOOK_STATUS_SHUTDOWN; + + return (audiohook ? 0 : -1); +} + +/*! \brief Pass a DTMF frame off to be handled by the audiohook core + * \param chan Channel that the list is coming off of + * \param audiohook_list List of audiohooks + * \param direction Direction frame is coming in from + * \param frame The frame itself + * \return Return frame on success, NULL on failure + */ +static struct ast_frame *dtmf_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame) +{ + struct ast_audiohook *audiohook = NULL; + + AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) { + ast_audiohook_lock(audiohook); + if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) { + AST_LIST_REMOVE_CURRENT(&audiohook_list->manipulate_list, list); + audiohook->status = AST_AUDIOHOOK_STATUS_DONE; + ast_audiohook_unlock(audiohook); + audiohook->manipulate_callback(audiohook, NULL, NULL, 0); + continue; + } + if (ast_test_flag(audiohook, AST_AUDIOHOOK_WANTS_DTMF)) + audiohook->manipulate_callback(audiohook, chan, frame, direction); + ast_audiohook_unlock(audiohook); + } + AST_LIST_TRAVERSE_SAFE_END + + return frame; +} + +/*! \brief Pass an AUDIO frame off to be handled by the audiohook core + * \param chan Channel that the list is coming off of + * \param audiohook_list List of audiohooks + * \param direction Direction frame is coming in from + * \param frame The frame itself + * \return Return frame on success, NULL on failure + */ +static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame) +{ + struct ast_audiohook_translate *in_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->in_translate[0] : &audiohook_list->in_translate[1]); + struct ast_audiohook_translate *out_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->out_translate[0] : &audiohook_list->out_translate[1]); + struct ast_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame; + struct ast_audiohook *audiohook = NULL; + int samples = frame->samples; + + /* If the frame coming in is not signed linear we have to send it through the in_translate path */ + if (frame->subclass != AST_FORMAT_SLINEAR) { + if (in_translate->format != frame->subclass) { + if (in_translate->trans_pvt) + ast_translator_free_path(in_translate->trans_pvt); + if (!(in_translate->trans_pvt = ast_translator_build_path(AST_FORMAT_SLINEAR, frame->subclass))) + return frame; + in_translate->format = frame->subclass; + } + if (!(middle_frame = ast_translate(in_translate->trans_pvt, frame, 0))) + return frame; + } + + /* Queue up signed linear frame to each spy */ + AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) { + ast_audiohook_lock(audiohook); + if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) { + AST_LIST_REMOVE_CURRENT(&audiohook_list->spy_list, list); + audiohook->status = AST_AUDIOHOOK_STATUS_DONE; + ast_cond_signal(&audiohook->trigger); + ast_audiohook_unlock(audiohook); + continue; + } + ast_audiohook_write_frame(audiohook, direction, middle_frame); + ast_audiohook_unlock(audiohook); + } + AST_LIST_TRAVERSE_SAFE_END + + /* If this frame is being written out to the channel then we need to use whisper sources */ + if (direction == AST_AUDIOHOOK_DIRECTION_WRITE && !AST_LIST_EMPTY(&audiohook_list->whisper_list)) { + int i = 0; + short read_buf[samples], combine_buf[samples], *data1 = NULL, *data2 = NULL; + memset(&combine_buf, 0, sizeof(combine_buf)); + AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) { + ast_audiohook_lock(audiohook); + if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) { + AST_LIST_REMOVE_CURRENT(&audiohook_list->whisper_list, list); + audiohook->status = AST_AUDIOHOOK_STATUS_DONE; + ast_cond_signal(&audiohook->trigger); + ast_audiohook_unlock(audiohook); + continue; + } + if (ast_slinfactory_available(&audiohook->write_factory) >= samples && ast_slinfactory_read(&audiohook->write_factory, read_buf, samples)) { + /* Take audio from this whisper source and combine it into our main buffer */ + for (i = 0, data1 = combine_buf, data2 = read_buf; i < samples; i++, data1++, data2++) + ast_slinear_saturated_add(data1, data2); + } + ast_audiohook_unlock(audiohook); + } + AST_LIST_TRAVERSE_SAFE_END + /* We take all of the combined whisper sources and combine them into the audio being written out */ + for (i = 0, data1 = middle_frame->data, data2 = combine_buf; i < samples; i++, data1++, data2++) + ast_slinear_saturated_add(data1, data2); + end_frame = middle_frame; + } + + /* Pass off frame to manipulate audiohooks */ + if (!AST_LIST_EMPTY(&audiohook_list->manipulate_list)) { + AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) { + ast_audiohook_lock(audiohook); + if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) { + AST_LIST_REMOVE_CURRENT(&audiohook_list->manipulate_list, list); + audiohook->status = AST_AUDIOHOOK_STATUS_DONE; + ast_audiohook_unlock(audiohook); + /* We basically drop all of our links to the manipulate audiohook and prod it to do it's own destructive things */ + audiohook->manipulate_callback(audiohook, chan, NULL, direction); + continue; + } + /* Feed in frame to manipulation */ + audiohook->manipulate_callback(audiohook, chan, middle_frame, direction); + ast_audiohook_unlock(audiohook); + } + AST_LIST_TRAVERSE_SAFE_END + end_frame = middle_frame; + } + + /* Now we figure out what to do with our end frame (whether to transcode or not) */ + if (middle_frame == end_frame) { + /* Middle frame was modified and became the end frame... let's see if we need to transcode */ + if (end_frame->subclass != start_frame->subclass) { + if (out_translate->format != start_frame->subclass) { + if (out_translate->trans_pvt) + ast_translator_free_path(out_translate->trans_pvt); + if (!(out_translate->trans_pvt = ast_translator_build_path(start_frame->subclass, AST_FORMAT_SLINEAR))) { + /* We can't transcode this... drop our middle frame and return the original */ + ast_frfree(middle_frame); + return start_frame; + } + out_translate->format = start_frame->subclass; + } + /* Transcode from our middle (signed linear) frame to new format of the frame that came in */ + if (!(end_frame = ast_translate(out_translate->trans_pvt, middle_frame, 0))) { + /* Failed to transcode the frame... drop it and return the original */ + ast_frfree(middle_frame); + return start_frame; + } + /* Here's the scoop... middle frame is no longer of use to us */ + ast_frfree(middle_frame); + } + /* Yay let's rid ourselves of the start frame */ + ast_frfree(start_frame); + } else { + /* No frame was modified, we can just drop our middle frame and pass the frame we got in out */ + ast_frfree(middle_frame); + } + + return end_frame; +} + +/*! \brief Pass a frame off to be handled by the audiohook core + * \param chan Channel that the list is coming off of + * \param audiohook_list List of audiohooks + * \param direction Direction frame is coming in from + * \param frame The frame itself + * \return Return frame on success, NULL on failure + */ +struct ast_frame *ast_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame) +{ + /* Pass off frame to it's respective list write function */ + if (frame->frametype == AST_FRAME_VOICE) + return audio_audiohook_write_list(chan, audiohook_list, direction, frame); + else if (frame->frametype == AST_FRAME_DTMF) + return dtmf_audiohook_write_list(chan, audiohook_list, direction, frame); + else + return frame; +} + + +/*! \brief Wait for audiohook trigger to be triggered + * \param audiohook Audiohook to wait on + */ +void ast_audiohook_trigger_wait(struct ast_audiohook *audiohook) +{ + struct timeval tv; + struct timespec ts; + + tv = ast_tvadd(ast_tvnow(), ast_samp2tv(50000, 1000)); + ts.tv_sec = tv.tv_sec; + ts.tv_nsec = tv.tv_usec * 1000; + + ast_cond_timedwait(&audiohook->trigger, &audiohook->lock, &ts); + + return; +} diff --git a/main/channel.c b/main/channel.c index 5c8de5694..3f3ce9466 100644 --- a/main/channel.c +++ b/main/channel.c @@ -44,7 +44,6 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #include "asterisk/sched.h" #include "asterisk/options.h" #include "asterisk/channel.h" -#include "asterisk/chanspy.h" #include "asterisk/musiconhold.h" #include "asterisk/logger.h" #include "asterisk/say.h" @@ -66,27 +65,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #include "asterisk/sha1.h" #include "asterisk/threadstorage.h" #include "asterisk/slinfactory.h" - -struct channel_spy_trans { - int last_format; - struct ast_trans_pvt *path; -}; - -/*! \brief List of SPY structures -*/ -struct ast_channel_spy_list { - struct channel_spy_trans read_translator; - struct channel_spy_trans write_translator; - AST_LIST_HEAD_NOLOCK(, ast_channel_spy) list; -}; - -/*! \brief Definition of the Whisper buffer */ -struct ast_channel_whisper_buffer { - ast_mutex_t lock; - struct ast_slinfactory sf; - unsigned int original_format; - struct ast_trans_pvt *path; -}; +#include "asterisk/audiohook.h" /* uncomment if you have problems with 'monitoring' synchronized files */ #if 0 @@ -1121,10 +1100,6 @@ void ast_channel_free(struct ast_channel *chan) if (chan->music_state) ast_moh_cleanup(chan); - /* if someone is whispering on the channel, stop them */ - if (chan->whisper) - ast_channel_whisper_stop(chan); - /* Free translators */ if (chan->readtrans) ast_translator_free_path(chan->readtrans); @@ -1281,176 +1256,6 @@ struct ast_datastore *ast_channel_datastore_find(struct ast_channel *chan, const return datastore; } -int ast_channel_spy_add(struct ast_channel *chan, struct ast_channel_spy *spy) -{ - /* Link the owner channel to the spy */ - spy->chan = chan; - - if (!ast_test_flag(spy, CHANSPY_FORMAT_AUDIO)) { - ast_log(LOG_WARNING, "Could not add channel spy '%s' to channel '%s', only audio format spies are supported.\n", - spy->type, chan->name); - return -1; - } - - if (ast_test_flag(spy, CHANSPY_READ_VOLADJUST) && (spy->read_queue.format != AST_FORMAT_SLINEAR)) { - ast_log(LOG_WARNING, "Cannot provide volume adjustment on '%s' format spies\n", - ast_getformatname(spy->read_queue.format)); - return -1; - } - - if (ast_test_flag(spy, CHANSPY_WRITE_VOLADJUST) && (spy->write_queue.format != AST_FORMAT_SLINEAR)) { - ast_log(LOG_WARNING, "Cannot provide volume adjustment on '%s' format spies\n", - ast_getformatname(spy->write_queue.format)); - return -1; - } - - if (ast_test_flag(spy, CHANSPY_MIXAUDIO) && - ((spy->read_queue.format != AST_FORMAT_SLINEAR) || - (spy->write_queue.format != AST_FORMAT_SLINEAR))) { - ast_log(LOG_WARNING, "Cannot provide audio mixing on '%s'-'%s' format spies\n", - ast_getformatname(spy->read_queue.format), ast_getformatname(spy->write_queue.format)); - return -1; - } - - if (!chan->spies) { - if (!(chan->spies = ast_calloc(1, sizeof(*chan->spies)))) { - return -1; - } - - AST_LIST_HEAD_INIT_NOLOCK(&chan->spies->list); - AST_LIST_INSERT_HEAD(&chan->spies->list, spy, list); - } else { - AST_LIST_INSERT_TAIL(&chan->spies->list, spy, list); - } - - if (ast_test_flag(spy, CHANSPY_TRIGGER_MODE) != CHANSPY_TRIGGER_NONE) { - ast_cond_init(&spy->trigger, NULL); - ast_set_flag(spy, CHANSPY_TRIGGER_READ); - ast_clear_flag(spy, CHANSPY_TRIGGER_WRITE); - } - - ast_debug(1, "Spy %s added to channel %s\n", - spy->type, chan->name); - - return 0; -} - -/* Clean up a channel's spy information */ -static void spy_cleanup(struct ast_channel *chan) -{ - if (!AST_LIST_EMPTY(&chan->spies->list)) - return; - if (chan->spies->read_translator.path) - ast_translator_free_path(chan->spies->read_translator.path); - if (chan->spies->write_translator.path) - ast_translator_free_path(chan->spies->write_translator.path); - ast_free(chan->spies); - chan->spies = NULL; - return; -} - -/* Detach a spy from it's channel */ -static void spy_detach(struct ast_channel_spy *spy, struct ast_channel *chan) -{ - /* We only need to poke them if they aren't already done */ - if (spy->status != CHANSPY_DONE) { - ast_mutex_lock(&spy->lock); - /* Indicate to the spy to stop */ - spy->status = CHANSPY_STOP; - spy->chan = NULL; - /* Poke the spy if needed */ - if (ast_test_flag(spy, CHANSPY_TRIGGER_MODE) != CHANSPY_TRIGGER_NONE) - ast_cond_signal(&spy->trigger); - ast_mutex_unlock(&spy->lock); - } - - /* Print it out while we still have a lock so the structure can't go away (if signalled above) */ - ast_debug(1, "Spy %s removed from channel %s\n", spy->type, chan->name); - - return; -} - -void ast_channel_spy_stop_by_type(struct ast_channel *chan, const char *type) -{ - struct ast_channel_spy *spy = NULL; - - if (!chan->spies) - return; - - AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->spies->list, spy, list) { - if ((spy->type == type) && (spy->status == CHANSPY_RUNNING)) { - AST_LIST_REMOVE_CURRENT(&chan->spies->list, list); - spy_detach(spy, chan); - } - } - AST_LIST_TRAVERSE_SAFE_END - spy_cleanup(chan); -} - -void ast_channel_spy_trigger_wait(struct ast_channel_spy *spy) -{ - struct timeval tv; - struct timespec ts; - - tv = ast_tvadd(ast_tvnow(), ast_samp2tv(50000, 1000)); - ts.tv_sec = tv.tv_sec; - ts.tv_nsec = tv.tv_usec * 1000; - - ast_cond_timedwait(&spy->trigger, &spy->lock, &ts); -} - -void ast_channel_spy_remove(struct ast_channel *chan, struct ast_channel_spy *spy) -{ - if (!chan->spies) - return; - - AST_LIST_REMOVE(&chan->spies->list, spy, list); - spy_detach(spy, chan); - spy_cleanup(chan); -} - -void ast_channel_spy_free(struct ast_channel_spy *spy) -{ - struct ast_frame *f = NULL; - - if (spy->status == CHANSPY_DONE) - return; - - /* Switch status to done in case we get called twice */ - spy->status = CHANSPY_DONE; - - /* Drop any frames in the queue */ - while ((f = AST_LIST_REMOVE_HEAD(&spy->write_queue.list, frame_list))) - ast_frfree(f); - while ((f = AST_LIST_REMOVE_HEAD(&spy->read_queue.list, frame_list))) - ast_frfree(f); - - /* Destroy the condition if in use */ - if (ast_test_flag(spy, CHANSPY_TRIGGER_MODE) != CHANSPY_TRIGGER_NONE) - ast_cond_destroy(&spy->trigger); - - /* Destroy our mutex since it is no longer in use */ - ast_mutex_destroy(&spy->lock); - - return; -} - -static void detach_spies(struct ast_channel *chan) -{ - struct ast_channel_spy *spy = NULL; - - if (!chan->spies) - return; - - AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->spies->list, spy, list) { - AST_LIST_REMOVE_CURRENT(&chan->spies->list, list); - spy_detach(spy, chan); - } - AST_LIST_TRAVERSE_SAFE_END - - spy_cleanup(chan); -} - /*! \brief Softly hangup a channel, don't lock */ int ast_softhangup_nolock(struct ast_channel *chan, int cause) { @@ -1474,124 +1279,6 @@ int ast_softhangup(struct ast_channel *chan, int cause) return res; } -enum spy_direction { - SPY_READ, - SPY_WRITE, -}; - -#define SPY_QUEUE_SAMPLE_LIMIT 4000 /* half of one second */ - -static void queue_frame_to_spies(struct ast_channel *chan, struct ast_frame *f, enum spy_direction dir) -{ - struct ast_frame *translated_frame = NULL; - struct ast_channel_spy *spy; - struct channel_spy_trans *trans; - - trans = (dir == SPY_READ) ? &chan->spies->read_translator : &chan->spies->write_translator; - - AST_LIST_TRAVERSE(&chan->spies->list, spy, list) { - struct ast_channel_spy_queue *queue; - struct ast_frame *duped_fr; - - if (spy->status != CHANSPY_RUNNING) - continue; - - ast_mutex_lock(&spy->lock); - - queue = (dir == SPY_READ) ? &spy->read_queue : &spy->write_queue; - - if ((queue->format == AST_FORMAT_SLINEAR) && (f->subclass != AST_FORMAT_SLINEAR)) { - if (!translated_frame) { - if (trans->path && (trans->last_format != f->subclass)) { - ast_translator_free_path(trans->path); - trans->path = NULL; - } - if (!trans->path) { - ast_debug(1, "Building translator from %s to SLINEAR for spies on channel %s\n", - ast_getformatname(f->subclass), chan->name); - if ((trans->path = ast_translator_build_path(AST_FORMAT_SLINEAR, f->subclass)) == NULL) { - ast_log(LOG_WARNING, "Cannot build a path from %s to %s\n", - ast_getformatname(f->subclass), ast_getformatname(AST_FORMAT_SLINEAR)); - ast_mutex_unlock(&spy->lock); - continue; - } else { - trans->last_format = f->subclass; - } - } - if (!(translated_frame = ast_translate(trans->path, f, 0))) { - ast_log(LOG_ERROR, "Translation to %s failed, dropping frame for spies\n", - ast_getformatname(AST_FORMAT_SLINEAR)); - ast_mutex_unlock(&spy->lock); - break; - } - } - duped_fr = ast_frdup(translated_frame); - } else if (f->subclass != queue->format) { - ast_log(LOG_WARNING, "Spy '%s' on channel '%s' wants format '%s', but frame is '%s', dropping\n", - spy->type, chan->name, - ast_getformatname(queue->format), ast_getformatname(f->subclass)); - ast_mutex_unlock(&spy->lock); - continue; - } else - duped_fr = ast_frdup(f); - - AST_LIST_INSERT_TAIL(&queue->list, duped_fr, frame_list); - - queue->samples += f->samples; - - if (queue->samples > SPY_QUEUE_SAMPLE_LIMIT) { - if (ast_test_flag(spy, CHANSPY_TRIGGER_MODE) != CHANSPY_TRIGGER_NONE) { - switch (ast_test_flag(spy, CHANSPY_TRIGGER_MODE)) { - case CHANSPY_TRIGGER_READ: - if (dir == SPY_WRITE) { - ast_set_flag(spy, CHANSPY_TRIGGER_WRITE); - ast_clear_flag(spy, CHANSPY_TRIGGER_READ); - ast_debug(1, "Switching spy '%s' on '%s' to write-trigger mode\n", - spy->type, chan->name); - } - break; - case CHANSPY_TRIGGER_WRITE: - if (dir == SPY_READ) { - ast_set_flag(spy, CHANSPY_TRIGGER_READ); - ast_clear_flag(spy, CHANSPY_TRIGGER_WRITE); - ast_debug(1, "Switching spy '%s' on '%s' to read-trigger mode\n", - spy->type, chan->name); - } - break; - } - ast_debug(1, "Triggering queue flush for spy '%s' on '%s'\n", - spy->type, chan->name); - ast_set_flag(spy, CHANSPY_TRIGGER_FLUSH); - ast_cond_signal(&spy->trigger); - } else { - ast_debug(1, "Spy '%s' on channel '%s' %s queue too long, dropping frames\n", - spy->type, chan->name, (dir == SPY_READ) ? "read" : "write"); - while (queue->samples > SPY_QUEUE_SAMPLE_LIMIT) { - struct ast_frame *drop = AST_LIST_REMOVE_HEAD(&queue->list, frame_list); - queue->samples -= drop->samples; - ast_frfree(drop); - } - } - } else { - switch (ast_test_flag(spy, CHANSPY_TRIGGER_MODE)) { - case CHANSPY_TRIGGER_READ: - if (dir == SPY_READ) - ast_cond_signal(&spy->trigger); - break; - case CHANSPY_TRIGGER_WRITE: - if (dir == SPY_WRITE) - ast_cond_signal(&spy->trigger); - break; - } - } - - ast_mutex_unlock(&spy->lock); - } - - if (translated_frame) - ast_frfree(translated_frame); -} - static void free_translation(struct ast_channel *clone) { if (clone->writetrans) @@ -1614,7 +1301,10 @@ int ast_hangup(struct ast_channel *chan) if someone is going to masquerade as us */ ast_channel_lock(chan); - detach_spies(chan); /* get rid of spies */ + if (chan->audiohooks) { + ast_audiohook_detach_list(chan->audiohooks); + chan->audiohooks = NULL; + } if (chan->masq) { if (ast_do_masquerade(chan)) @@ -2310,6 +2000,8 @@ static struct ast_frame *__ast_read(struct ast_channel *chan, int dropaudio) chan->emulate_dtmf_duration = AST_DEFAULT_EMULATE_DTMF_DURATION; ast_log(LOG_DTMF, "DTMF begin emulation of '%c' with duration %u queued on %s\n", f->subclass, chan->emulate_dtmf_duration, chan->name); } + if (chan->audiohooks) + f = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_READ, f); } else { struct timeval now = ast_tvnow(); if (ast_test_flag(chan, AST_FLAG_IN_DTMF)) { @@ -2331,6 +2023,8 @@ static struct ast_frame *__ast_read(struct ast_channel *chan, int dropaudio) ast_log(LOG_DTMF, "DTMF end passthrough '%c' on %s\n", f->subclass, chan->name); chan->dtmf_tv = now; } + if (chan->audiohooks) + f = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_READ, f); } break; case AST_FRAME_DTMF_BEGIN: @@ -2392,6 +2086,8 @@ static struct ast_frame *__ast_read(struct ast_channel *chan, int dropaudio) f->subclass = chan->emulate_dtmf_digit; f->len = ast_tvdiff_ms(now, chan->dtmf_tv); chan->dtmf_tv = now; + if (chan->audiohooks) + f = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_READ, f); ast_log(LOG_DTMF, "DTMF end emulation of '%c' queued on %s\n", f->subclass, chan->name); } else { /* Drop voice frames while we're still in the middle of the digit */ @@ -2406,9 +2102,9 @@ static struct ast_frame *__ast_read(struct ast_channel *chan, int dropaudio) ast_frfree(f); f = &ast_null_frame; } else if ((f->frametype == AST_FRAME_VOICE)) { - if (chan->spies) - queue_frame_to_spies(chan, f, SPY_READ); - + /* Send frame to audiohooks if present */ + if (chan->audiohooks) + f = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_READ, f); if (chan->monitor && chan->monitor->read_stream ) { /* XXX what does this do ? */ #ifndef MONITOR_CONSTANT_DELAY @@ -2746,6 +2442,8 @@ int ast_write(struct ast_channel *chan, struct ast_frame *fr) chan->tech->indicate(chan, fr->subclass, fr->data, fr->datalen); break; case AST_FRAME_DTMF_BEGIN: + if (chan->audiohooks) + fr = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_WRITE, fr); send_dtmf_event(chan, "Sent", fr->subclass, "Yes", "No"); ast_clear_flag(chan, AST_FLAG_BLOCKING); ast_channel_unlock(chan); @@ -2754,6 +2452,8 @@ int ast_write(struct ast_channel *chan, struct ast_frame *fr) CHECK_BLOCKING(chan); break; case AST_FRAME_DTMF_END: + if (chan->audiohooks) + fr = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_WRITE, fr); send_dtmf_event(chan, "Sent", fr->subclass, "No", "Yes"); ast_clear_flag(chan, AST_FLAG_BLOCKING); ast_channel_unlock(chan); @@ -2787,42 +2487,21 @@ int ast_write(struct ast_channel *chan, struct ast_frame *fr) if (chan->tech->write == NULL) break; /*! \todo XXX should return 0 maybe ? */ - /* If someone is whispering on this channel then we must ensure that we are always getting signed linear frames */ - if (ast_test_flag(chan, AST_FLAG_WHISPER)) { - if (fr->subclass == AST_FORMAT_SLINEAR) - f = fr; - else { - ast_mutex_lock(&chan->whisper->lock); - if (chan->writeformat != AST_FORMAT_SLINEAR) { - /* Rebuild the translation path and set our write format back to signed linear */ - chan->whisper->original_format = chan->writeformat; - ast_set_write_format(chan, AST_FORMAT_SLINEAR); - if (chan->whisper->path) - ast_translator_free_path(chan->whisper->path); - chan->whisper->path = ast_translator_build_path(AST_FORMAT_SLINEAR, chan->whisper->original_format); - } - /* Translate frame using the above translation path */ - f = (chan->whisper->path) ? ast_translate(chan->whisper->path, fr, 0) : fr; - ast_mutex_unlock(&chan->whisper->lock); - } - } else { - /* If the frame is in the raw write format, then it's easy... just use the frame - otherwise we will have to translate */ - if (fr->subclass == chan->rawwriteformat) - f = fr; - else - f = (chan->writetrans) ? ast_translate(chan->writetrans, fr, 0) : fr; - } + /* If audiohooks are present, write the frame out */ + if (chan->audiohooks) + fr = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_WRITE, fr); + + /* If the frame is in the raw write format, then it's easy... just use the frame - otherwise we will have to translate */ + if (fr->subclass == chan->rawwriteformat) + f = fr; + else + f = (chan->writetrans) ? ast_translate(chan->writetrans, fr, 0) : fr; - /* If we have no frame of audio, then we have to bail out */ - if (f == NULL) { + if (!f) { res = 0; break; } - /* If spies are on the channel then queue the frame out to them */ - if (chan->spies) - queue_frame_to_spies(chan, f, SPY_WRITE); - /* If Monitor is running on this channel, then we have to write frames out there too */ if (chan->monitor && chan->monitor->write_stream) { /* XXX must explain this code */ @@ -2849,30 +2528,6 @@ int ast_write(struct ast_channel *chan, struct ast_frame *fr) ast_log(LOG_WARNING, "Failed to write data to channel monitor write stream\n"); } } - - /* Finally the good part! Write this out to the channel */ - if (ast_test_flag(chan, AST_FLAG_WHISPER)) { - /* frame is assumed to be in SLINEAR, since that is - required for whisper mode */ - ast_frame_adjust_volume(f, -2); - if (ast_slinfactory_available(&chan->whisper->sf) >= f->samples) { - short buf[f->samples]; - struct ast_frame whisper = { - .frametype = AST_FRAME_VOICE, - .subclass = AST_FORMAT_SLINEAR, - .data = buf, - .datalen = sizeof(buf), - .samples = f->samples, - }; - - ast_mutex_lock(&chan->whisper->lock); - if (ast_slinfactory_read(&chan->whisper->sf, buf, f->samples)) - ast_frame_slinear_sum(f, &whisper); - ast_mutex_unlock(&chan->whisper->lock); - } - /* and now put it through the regular translator */ - f = (chan->writetrans) ? ast_translate(chan->writetrans, f, 0) : f; - } res = f ? chan->tech->write(chan, f) : 0; break; case AST_FRAME_NULL: @@ -3460,8 +3115,6 @@ int ast_do_masquerade(struct ast_channel *original) void *t_pvt; struct ast_callerid tmpcid; struct ast_channel *clone = original->masq; - struct ast_channel_spy_list *spy_list = NULL; - struct ast_channel_spy *spy = NULL; struct ast_cdr *cdr; int rformat = original->readformat; int wformat = original->writeformat; @@ -3547,27 +3200,6 @@ int ast_do_masquerade(struct ast_channel *original) original->rawwriteformat = clone->rawwriteformat; clone->rawwriteformat = x; - /* Swap the spies */ - spy_list = original->spies; - original->spies = clone->spies; - clone->spies = spy_list; - - /* Update channel on respective spy lists if present */ - if (original->spies) { - AST_LIST_TRAVERSE(&original->spies->list, spy, list) { - ast_mutex_lock(&spy->lock); - spy->chan = original; - ast_mutex_unlock(&spy->lock); - } - } - if (clone->spies) { - AST_LIST_TRAVERSE(&clone->spies->list, spy, list) { - ast_mutex_lock(&spy->lock); - spy->chan = clone; - ast_mutex_unlock(&spy->lock); - } - } - /* Save any pending frames on both sides. Start by counting * how many we're going to need... */ x = 0; @@ -3632,15 +3264,6 @@ int ast_do_masquerade(struct ast_channel *original) ast_app_group_update(clone, original); - /* move any whisperer over */ - ast_channel_whisper_stop(original); - if (ast_test_flag(clone, AST_FLAG_WHISPER)) { - original->whisper = clone->whisper; - ast_set_flag(original, AST_FLAG_WHISPER); - clone->whisper = NULL; - ast_clear_flag(clone, AST_FLAG_WHISPER); - } - /* Move data stores over */ if (AST_LIST_FIRST(&clone->datastores)) AST_LIST_INSERT_TAIL(&original->datastores, AST_LIST_FIRST(&clone->datastores), entry); @@ -4152,7 +3775,8 @@ enum ast_bridge_result ast_channel_bridge(struct ast_channel *c0, struct ast_cha (config->timelimit == 0) && (c0->tech->bridge == c1->tech->bridge) && !nativefailed && !c0->monitor && !c1->monitor && - !c0->spies && !c1->spies && !ast_test_flag(&(config->features_callee),AST_FEATURE_REDIRECT) && + !c0->audiohooks && !c1->audiohooks && + !ast_test_flag(&(config->features_callee),AST_FEATURE_REDIRECT) && !ast_test_flag(&(config->features_caller),AST_FEATURE_REDIRECT) ) { /* Looks like they share a bridge method and nothing else is in the way */ ast_set_flag(c0, AST_FLAG_NBRIDGE); @@ -4525,129 +4149,6 @@ void ast_set_variables(struct ast_channel *chan, struct ast_variable *vars) pbx_builtin_setvar_helper(chan, cur->name, cur->value); } -static void copy_data_from_queue(struct ast_channel_spy_queue *queue, short *buf, unsigned int samples) -{ - struct ast_frame *f; - int tocopy; - int bytestocopy; - - while (samples) { - if (!(f = AST_LIST_FIRST(&queue->list))) { - ast_log(LOG_ERROR, "Ran out of frames before buffer filled!\n"); - break; - } - - tocopy = (f->samples > samples) ? samples : f->samples; - bytestocopy = ast_codec_get_len(queue->format, tocopy); - memcpy(buf, f->data, bytestocopy); - samples -= tocopy; - buf += tocopy; - f->samples -= tocopy; - f->data += bytestocopy; - f->datalen -= bytestocopy; - f->offset += bytestocopy; - queue->samples -= tocopy; - - if (!f->samples) - ast_frfree(AST_LIST_REMOVE_HEAD(&queue->list, frame_list)); - } -} - -struct ast_frame *ast_channel_spy_read_frame(struct ast_channel_spy *spy, unsigned int samples) -{ - struct ast_frame *result; - /* buffers are allocated to hold SLINEAR, which is the largest format */ - short read_buf[samples]; - short write_buf[samples]; - struct ast_frame *read_frame; - struct ast_frame *write_frame; - int need_dup; - struct ast_frame stack_read_frame = { .frametype = AST_FRAME_VOICE, - .subclass = spy->read_queue.format, - .data = read_buf, - .samples = samples, - .datalen = ast_codec_get_len(spy->read_queue.format, samples), - }; - struct ast_frame stack_write_frame = { .frametype = AST_FRAME_VOICE, - .subclass = spy->write_queue.format, - .data = write_buf, - .samples = samples, - .datalen = ast_codec_get_len(spy->write_queue.format, samples), - }; - - /* if a flush has been requested, dump everything in whichever queue is larger */ - if (ast_test_flag(spy, CHANSPY_TRIGGER_FLUSH)) { - if (spy->read_queue.samples > spy->write_queue.samples) { - if (ast_test_flag(spy, CHANSPY_READ_VOLADJUST)) { - AST_LIST_TRAVERSE(&spy->read_queue.list, result, frame_list) - ast_frame_adjust_volume(result, spy->read_vol_adjustment); - } - result = AST_LIST_FIRST(&spy->read_queue.list); - AST_LIST_HEAD_SET_NOLOCK(&spy->read_queue.list, NULL); - spy->read_queue.samples = 0; - } else { - if (ast_test_flag(spy, CHANSPY_WRITE_VOLADJUST)) { - AST_LIST_TRAVERSE(&spy->write_queue.list, result, frame_list) - ast_frame_adjust_volume(result, spy->write_vol_adjustment); - } - result = AST_LIST_FIRST(&spy->write_queue.list); - AST_LIST_HEAD_SET_NOLOCK(&spy->write_queue.list, NULL); - spy->write_queue.samples = 0; - } - ast_clear_flag(spy, CHANSPY_TRIGGER_FLUSH); - return result; - } - - if ((spy->read_queue.samples < samples) || (spy->write_queue.samples < samples)) - return NULL; - - /* short-circuit if both head frames have exactly what we want */ - if ((AST_LIST_FIRST(&spy->read_queue.list)->samples == samples) && - (AST_LIST_FIRST(&spy->write_queue.list)->samples == samples)) { - read_frame = AST_LIST_REMOVE_HEAD(&spy->read_queue.list, frame_list); - write_frame = AST_LIST_REMOVE_HEAD(&spy->write_queue.list, frame_list); - - spy->read_queue.samples -= samples; - spy->write_queue.samples -= samples; - - need_dup = 0; - } else { - copy_data_from_queue(&spy->read_queue, read_buf, samples); - copy_data_from_queue(&spy->write_queue, write_buf, samples); - - read_frame = &stack_read_frame; - write_frame = &stack_write_frame; - need_dup = 1; - } - - if (ast_test_flag(spy, CHANSPY_READ_VOLADJUST)) - ast_frame_adjust_volume(read_frame, spy->read_vol_adjustment); - - if (ast_test_flag(spy, CHANSPY_WRITE_VOLADJUST)) - ast_frame_adjust_volume(write_frame, spy->write_vol_adjustment); - - if (ast_test_flag(spy, CHANSPY_MIXAUDIO)) { - ast_frame_slinear_sum(read_frame, write_frame); - - if (need_dup) - result = ast_frdup(read_frame); - else { - result = read_frame; - ast_frfree(write_frame); - } - } else { - if (need_dup) { - result = ast_frdup(read_frame); - AST_LIST_NEXT(result, frame_list) = ast_frdup(write_frame); - } else { - result = read_frame; - AST_LIST_NEXT(result, frame_list) = write_frame; - } - } - - return result; -} - static void *silence_generator_alloc(struct ast_channel *chan, void *data) { /* just store the data pointer in the channel structure */ @@ -4894,46 +4395,3 @@ int ast_say_digits_full(struct ast_channel *chan, int num, snprintf(buf, sizeof(buf), "%d", num); return ast_say_digit_str_full(chan, buf, ints, lang, audiofd, ctrlfd); } - -int ast_channel_whisper_start(struct ast_channel *chan) -{ - if (chan->whisper) - return -1; - - if (!(chan->whisper = ast_calloc(1, sizeof(*chan->whisper)))) - return -1; - - ast_mutex_init(&chan->whisper->lock); - ast_slinfactory_init(&chan->whisper->sf); - ast_set_flag(chan, AST_FLAG_WHISPER); - - return 0; -} - -int ast_channel_whisper_feed(struct ast_channel *chan, struct ast_frame *f) -{ - if (!chan->whisper) - return -1; - - ast_mutex_lock(&chan->whisper->lock); - ast_slinfactory_feed(&chan->whisper->sf, f); - ast_mutex_unlock(&chan->whisper->lock); - - return 0; -} - -void ast_channel_whisper_stop(struct ast_channel *chan) -{ - if (!chan->whisper) - return; - - ast_clear_flag(chan, AST_FLAG_WHISPER); - if (chan->whisper->path) - ast_translator_free_path(chan->whisper->path); - if (chan->whisper->original_format && chan->writeformat == AST_FORMAT_SLINEAR) - ast_set_write_format(chan, chan->whisper->original_format); - ast_slinfactory_destroy(&chan->whisper->sf); - ast_mutex_destroy(&chan->whisper->lock); - ast_free(chan->whisper); - chan->whisper = NULL; -} diff --git a/main/slinfactory.c b/main/slinfactory.c index a42b2b213..038fa0d7b 100644 --- a/main/slinfactory.c +++ b/main/slinfactory.c @@ -148,3 +148,21 @@ unsigned int ast_slinfactory_available(const struct ast_slinfactory *sf) { return sf->size; } + +void ast_slinfactory_flush(struct ast_slinfactory *sf) +{ + struct ast_frame *fr = NULL; + + if (sf->trans) { + ast_translator_free_path(sf->trans); + sf->trans = NULL; + } + + while ((fr = AST_LIST_REMOVE_HEAD(&sf->queue, frame_list))) + ast_frfree(fr); + + sf->size = sf->holdlen = 0; + sf->offset = sf->hold; + + return; +} |