/* * Asterisk -- An open source telephony toolkit. * * Copyright (C) 1999 - 2007, Digium, Inc. * * Joshua Colp * * 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 */ /*** MODULEINFO core ***/ #include "asterisk.h" #include #include "asterisk/channel.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" #include "asterisk/format_cache.h" #define AST_AUDIOHOOK_SYNC_TOLERANCE 100 /*!< Tolerance in milliseconds for audiohooks synchronization */ #define AST_AUDIOHOOK_SMALL_QUEUE_TOLERANCE 100 /*!< When small queue is enabled, this is the maximum amount of audio that can remain queued at a time. */ #define AST_AUDIOHOOK_LONG_QUEUE_TOLERANCE 500 /*!< Otheriwise we still don't want the queue to grow indefinitely */ #define DEFAULT_INTERNAL_SAMPLE_RATE 8000 struct ast_audiohook_translate { struct ast_trans_pvt *trans_pvt; struct ast_format *format; }; struct ast_audiohook_list { /* If all the audiohooks in this list are capable * of processing slinear at any sample rate, this * variable will be set and the sample rate will * be preserved during ast_audiohook_write_list()*/ int native_slin_compatible; int list_internal_samp_rate;/*!< Internal sample rate used when writing to the 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; }; static int audiohook_set_internal_rate(struct ast_audiohook *audiohook, int rate, int reset) { struct ast_format *slin; if (audiohook->hook_internal_samp_rate == rate) { return 0; } audiohook->hook_internal_samp_rate = rate; slin = ast_format_cache_get_slin_by_rate(rate); /* Setup the factories that are needed for this audiohook type */ switch (audiohook->type) { case AST_AUDIOHOOK_TYPE_SPY: case AST_AUDIOHOOK_TYPE_WHISPER: if (reset) { ast_slinfactory_destroy(&audiohook->read_factory); ast_slinfactory_destroy(&audiohook->write_factory); } ast_slinfactory_init_with_format(&audiohook->read_factory, slin); ast_slinfactory_init_with_format(&audiohook->write_factory, slin); break; default: break; } return 0; } int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source, enum ast_audiohook_init_flags init_flags) { /* 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); audiohook->init_flags = init_flags; /* initialize internal rate at 8khz, this will adjust if necessary */ audiohook_set_internal_rate(audiohook, DEFAULT_INTERNAL_SAMPLE_RATE, 0); /* Since we are just starting out... this audiohook is new */ ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_NEW); return 0; } int ast_audiohook_destroy(struct ast_audiohook *audiohook) { /* Drop the factories used by this audiohook type */ switch (audiohook->type) { case AST_AUDIOHOOK_TYPE_SPY: case AST_AUDIOHOOK_TYPE_WHISPER: ast_slinfactory_destroy(&audiohook->read_factory); 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); ao2_cleanup(audiohook->format); /* Lock and trigger be gone! */ ast_cond_destroy(&audiohook->trigger); ast_mutex_destroy(&audiohook->lock); return 0; } #define SHOULD_MUTE(hook, dir) \ ((ast_test_flag(hook, AST_AUDIOHOOK_MUTE_READ) && (dir == AST_AUDIOHOOK_DIRECTION_READ)) || \ (ast_test_flag(hook, AST_AUDIOHOOK_MUTE_WRITE) && (dir == AST_AUDIOHOOK_DIRECTION_WRITE)) || \ (ast_test_flag(hook, AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE) == (AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE))) 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); struct ast_slinfactory *other_factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->write_factory : &audiohook->read_factory); struct timeval *rwtime = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_time : &audiohook->write_time), previous_time = *rwtime; int our_factory_samples; int our_factory_ms; int other_factory_samples; int other_factory_ms; /* Update last feeding time to be current */ *rwtime = ast_tvnow(); our_factory_samples = ast_slinfactory_available(factory); our_factory_ms = ast_tvdiff_ms(*rwtime, previous_time) + (our_factory_samples / (audiohook->hook_internal_samp_rate / 1000)); other_factory_samples = ast_slinfactory_available(other_factory); other_factory_ms = other_factory_samples / (audiohook->hook_internal_samp_rate / 1000); if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC) && (our_factory_ms - other_factory_ms > AST_AUDIOHOOK_SYNC_TOLERANCE)) { ast_debug(4, "Flushing audiohook %p so it remains in sync\n", audiohook); ast_slinfactory_flush(factory); ast_slinfactory_flush(other_factory); } if (ast_test_flag(audiohook, AST_AUDIOHOOK_SMALL_QUEUE) && ((our_factory_ms > AST_AUDIOHOOK_SMALL_QUEUE_TOLERANCE) || (other_factory_ms > AST_AUDIOHOOK_SMALL_QUEUE_TOLERANCE))) { ast_debug(4, "Audiohook %p has stale audio in its factories. Flushing them both\n", audiohook); ast_slinfactory_flush(factory); ast_slinfactory_flush(other_factory); } else if ((our_factory_ms > AST_AUDIOHOOK_LONG_QUEUE_TOLERANCE) || (other_factory_ms > AST_AUDIOHOOK_LONG_QUEUE_TOLERANCE)) { ast_debug(4, "Audiohook %p has stale audio in its factories. Flushing them both\n", audiohook); ast_slinfactory_flush(factory); ast_slinfactory_flush(other_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 */ if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_READ) && (direction == AST_AUDIOHOOK_DIRECTION_READ)) { ast_cond_signal(&audiohook->trigger); } else if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_WRITE) && (direction == AST_AUDIOHOOK_DIRECTION_WRITE)) { ast_cond_signal(&audiohook->trigger); } else if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC)) { ast_cond_signal(&audiohook->trigger); } 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.format = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate), .data.ptr = 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 (SHOULD_MUTE(audiohook, direction)) { /* Swap frame data for zeros if mute is required */ ast_frame_clear(&frame); } else if (vol) { /* If a volume adjustment needs to be applied apply it */ 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, struct ast_frame **read_reference, struct ast_frame **write_reference) { int count; int usable_read; int usable_write; short adjust_value; short buf1[samples]; short buf2[samples]; short *read_buf = NULL; short *write_buf = NULL; struct ast_frame frame = { .frametype = AST_FRAME_VOICE, .datalen = sizeof(buf1), .samples = samples, }; /* Make sure both factories have the required samples */ usable_read = (ast_slinfactory_available(&audiohook->read_factory) >= samples ? 1 : 0); usable_write = (ast_slinfactory_available(&audiohook->write_factory) >= samples ? 1 : 0); if (!usable_read && !usable_write) { /* If both factories are unusable bail out */ ast_debug(3, "Read factory %p and write factory %p both fail to provide %zu samples\n", &audiohook->read_factory, &audiohook->write_factory, samples); return NULL; } /* If we want to provide only a read factory make sure we aren't waiting for other audio */ if (usable_read && !usable_write && (ast_tvdiff_ms(ast_tvnow(), audiohook->write_time) < (samples/8)*2)) { ast_debug(3, "Write factory %p was pretty quick last time, waiting for them.\n", &audiohook->write_factory); return NULL; } /* If we want to provide only a write factory make sure we aren't waiting for other audio */ if (usable_write && !usable_read && (ast_tvdiff_ms(ast_tvnow(), audiohook->read_time) < (samples/8)*2)) { ast_debug(3, "Read factory %p was pretty quick last time, waiting for them.\n", &audiohook->read_factory); return NULL; } /* Start with the read factory... if there are enough samples, read them in */ if (usable_read) { if (ast_slinfactory_read(&audiohook->read_factory, buf1, samples)) { read_buf = buf1; if ((ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_READ))) { /* Clear the frame data if we are muting */ memset(buf1, 0, sizeof(buf1)); } else if (audiohook->options.read_volume) { /* Adjust read volume if need be */ 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 { ast_debug(1, "Failed to get %d samples from read factory %p\n", (int)samples, &audiohook->read_factory); } /* Move on to the write factory... if there are enough samples, read them in */ if (usable_write) { if (ast_slinfactory_read(&audiohook->write_factory, buf2, samples)) { write_buf = buf2; if ((ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_WRITE))) { /* Clear the frame data if we are muting */ memset(buf2, 0, sizeof(buf2)); } else if (audiohook->options.write_volume) { /* Adjust write volume if need be */ 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 { ast_debug(3, "Failed to get %d samples from write factory %p\n", (int)samples, &audiohook->write_factory); } frame.subclass.format = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate); /* Should we substitute silence if one side lacks audio? */ if ((ast_test_flag(audiohook, AST_AUDIOHOOK_SUBSTITUTE_SILENCE))) { if (read_reference && !read_buf && write_buf) { read_buf = buf1; memset(buf1, 0, sizeof(buf1)); } else if (write_reference && read_buf && !write_buf) { write_buf = buf2; memset(buf2, 0, sizeof(buf2)); } } /* Basically we figure out which buffer to use... and if mixing can be done here */ if (read_buf && read_reference) { frame.data.ptr = read_buf; *read_reference = ast_frdup(&frame); } if (write_buf && write_reference) { frame.data.ptr = write_buf; *write_reference = ast_frdup(&frame); } /* Make the correct buffer part of the built frame, so it gets duplicated. */ if (read_buf) { frame.data.ptr = read_buf; if (write_buf) { for (count = 0; count < samples; count++) { ast_slinear_saturated_add(read_buf++, write_buf++); } } } else if (write_buf) { frame.data.ptr = write_buf; } else { return NULL; } /* Yahoo, a combined copy of the audio! */ return ast_frdup(&frame); } static struct ast_frame *audiohook_read_frame_helper(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, struct ast_format *format, struct ast_frame **read_reference, struct ast_frame **write_reference) { struct ast_frame *read_frame = NULL, *final_frame = NULL; struct ast_format *slin; /* * Update the rate if compatibility mode is turned off or if it is * turned on and the format rate is higher than the current rate. * * This makes it so any unnecessary rate switching/resetting does * not take place and also any associated audiohook_list's internal * sample rate maintains the highest sample rate between hooks. */ if (!ast_test_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE) || (ast_test_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE) && ast_format_get_sample_rate(format) > audiohook->hook_internal_samp_rate)) { audiohook_set_internal_rate(audiohook, ast_format_get_sample_rate(format), 1); } /* If the sample rate of the requested format differs from that of the underlying audiohook * sample rate determine how many samples we actually need to get from the audiohook. This * needs to occur as the signed linear factory stores them at the rate of the audiohook. * We do this by determining the duration of audio they've requested and then determining * how many samples that would be in the audiohook format. */ if (ast_format_get_sample_rate(format) != audiohook->hook_internal_samp_rate) { samples = (audiohook->hook_internal_samp_rate / 1000) * (samples / (ast_format_get_sample_rate(format) / 1000)); } if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ? audiohook_read_frame_both(audiohook, samples, read_reference, write_reference) : audiohook_read_frame_single(audiohook, samples, direction)))) { return NULL; } slin = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate); /* If they don't want signed linear back out, we'll have to send it through the translation path */ if (ast_format_cmp(format, slin) != AST_FORMAT_CMP_EQUAL) { /* Rebuild translation path if different format then previously */ if (ast_format_cmp(format, audiohook->format) == AST_FORMAT_CMP_NOT_EQUAL) { 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, slin))) { ast_frfree(read_frame); return NULL; } ao2_replace(audiohook->format, format); } /* 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; } struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, struct ast_format *format) { return audiohook_read_frame_helper(audiohook, samples, direction, format, NULL, NULL); } struct ast_frame *ast_audiohook_read_frame_all(struct ast_audiohook *audiohook, size_t samples, struct ast_format *format, struct ast_frame **read_frame, struct ast_frame **write_frame) { return audiohook_read_frame_helper(audiohook, samples, AST_AUDIOHOOK_DIRECTION_BOTH, format, read_frame, write_frame); } static void audiohook_list_set_samplerate_compatibility(struct ast_audiohook_list *audiohook_list) { struct ast_audiohook *ah = NULL; /* * Anytime the samplerate compatibility is set (attach/remove an audiohook) the * list's internal sample rate needs to be reset so that the next time processing * through write_list, if needed, it will get updated to the correct rate. * * A list's internal rate always chooses the higher between its own rate and a * given rate. If the current rate is being driven by an audiohook that wanted a * higher rate then when this audiohook is removed the list's rate would remain * at that level when it should be lower, and with no way to lower it since any * rate compared against it would be lower. * * By setting it back to the lowest rate it can recalulate the new highest rate. */ audiohook_list->list_internal_samp_rate = DEFAULT_INTERNAL_SAMPLE_RATE; audiohook_list->native_slin_compatible = 1; AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, ah, list) { if (!(ah->init_flags & AST_AUDIOHOOK_MANIPULATE_ALL_RATES)) { audiohook_list->native_slin_compatible = 0; return; } } } int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook) { ast_channel_lock(chan); /* Don't allow an audiohook to be attached to a channel that is already hung up. * The hang up process is what actually notifies the audiohook that it should * stop. */ if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_ZOMBIE)) { ast_channel_unlock(chan); return -1; } if (!ast_channel_audiohooks(chan)) { struct ast_audiohook_list *ahlist; /* Whoops... allocate a new structure */ if (!(ahlist = ast_calloc(1, sizeof(*ahlist)))) { ast_channel_unlock(chan); return -1; } ast_channel_audiohooks_set(chan, ahlist); AST_LIST_HEAD_INIT_NOLOCK(&ast_channel_audiohooks(chan)->spy_list); AST_LIST_HEAD_INIT_NOLOCK(&ast_channel_audiohooks(chan)->whisper_list); AST_LIST_HEAD_INIT_NOLOCK(&ast_channel_audiohooks(chan)->manipulate_list); /* This sample rate will adjust as necessary when writing to the list. */ ast_channel_audiohooks(chan)->list_internal_samp_rate = DEFAULT_INTERNAL_SAMPLE_RATE; } /* Drop into respective list */ if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY) { AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->spy_list, audiohook, list); } else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER) { AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->whisper_list, audiohook, list); } else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE) { AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->manipulate_list, audiohook, list); } /* * Initialize the audiohook's rate to the default. If it needs to be, * it will get updated later. */ audiohook_set_internal_rate(audiohook, DEFAULT_INTERNAL_SAMPLE_RATE, 1); audiohook_list_set_samplerate_compatibility(ast_channel_audiohooks(chan)); /* Change status over to running since it is now attached */ ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_RUNNING); if (ast_channel_is_bridged(chan)) { ast_channel_set_unbridged_nolock(chan, 1); } ast_channel_unlock(chan); return 0; } void ast_audiohook_update_status(struct ast_audiohook *audiohook, enum ast_audiohook_status status) { ast_audiohook_lock(audiohook); if (audiohook->status != AST_AUDIOHOOK_STATUS_DONE) { audiohook->status = status; ast_cond_signal(&audiohook->trigger); } ast_audiohook_unlock(audiohook); } int ast_audiohook_detach(struct ast_audiohook *audiohook) { if (audiohook->status == AST_AUDIOHOOK_STATUS_NEW || audiohook->status == AST_AUDIOHOOK_STATUS_DONE) { return 0; } ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN); while (audiohook->status != AST_AUDIOHOOK_STATUS_DONE) { ast_audiohook_trigger_wait(audiohook); } return 0; } void ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list) { int i; struct ast_audiohook *audiohook; if (!audiohook_list) { return; } /* Drop any spies */ while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->spy_list, list))) { ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE); } /* Drop any whispering sources */ while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->whisper_list, list))) { ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE); } /* Drop any manipulators */ while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->manipulate_list, list))) { ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE); audiohook->manipulate_callback(audiohook, NULL, NULL, 0); } /* 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); ao2_cleanup(audiohook_list->in_translate[i].format); } if (audiohook_list->out_translate[i].trans_pvt) { ast_translator_free_path(audiohook_list->out_translate[i].trans_pvt); ao2_cleanup(audiohook_list->in_translate[i].format); } } /* Free ourselves */ ast_free(audiohook_list); } /*! \brief find an audiohook based on its source * \param audiohook_list The list of audiohooks to search in * \param source The source of the audiohook we wish to find * \return corresponding audiohook * \retval NULL if it cannot be found */ 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; } static void audiohook_move(struct ast_channel *old_chan, struct ast_channel *new_chan, struct ast_audiohook *audiohook) { enum ast_audiohook_status oldstatus; /* By locking both channels and the audiohook, we can assure that * another thread will not have a chance to read the audiohook's status * as done, even though ast_audiohook_remove signals the trigger * condition. */ ast_audiohook_lock(audiohook); oldstatus = audiohook->status; ast_audiohook_remove(old_chan, audiohook); ast_audiohook_attach(new_chan, audiohook); audiohook->status = oldstatus; ast_audiohook_unlock(audiohook); } void ast_audiohook_move_by_source(struct ast_channel *old_chan, struct ast_channel *new_chan, const char *source) { struct ast_audiohook *audiohook; if (!ast_channel_audiohooks(old_chan)) { return; } audiohook = find_audiohook_by_source(ast_channel_audiohooks(old_chan), source); if (!audiohook) { return; } audiohook_move(old_chan, new_chan, audiohook); } void ast_audiohook_move_all(struct ast_channel *old_chan, struct ast_channel *new_chan) { struct ast_audiohook *audiohook; struct ast_audiohook_list *audiohook_list; audiohook_list = ast_channel_audiohooks(old_chan); if (!audiohook_list) { return; } AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) { audiohook_move(old_chan, new_chan, audiohook); } AST_LIST_TRAVERSE_SAFE_END; AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) { audiohook_move(old_chan, new_chan, audiohook); } AST_LIST_TRAVERSE_SAFE_END; AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) { audiohook_move(old_chan, new_chan, audiohook); } AST_LIST_TRAVERSE_SAFE_END; } 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 (!ast_channel_audiohooks(chan)) { ast_channel_unlock(chan); return -1; } audiohook = find_audiohook_by_source(ast_channel_audiohooks(chan), source); ast_channel_unlock(chan); if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE) { ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN); } return (audiohook ? 0 : -1); } int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audiohook) { ast_channel_lock(chan); if (!ast_channel_audiohooks(chan)) { ast_channel_unlock(chan); return -1; } if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY) { AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->spy_list, audiohook, list); } else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER) { AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->whisper_list, audiohook, list); } else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE) { AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->manipulate_list, audiohook, list); } audiohook_list_set_samplerate_compatibility(ast_channel_audiohooks(chan)); ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE); if (ast_channel_is_bridged(chan)) { ast_channel_set_unbridged_nolock(chan, 1); } ast_channel_unlock(chan); return 0; } /*! \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 frame on success * \retval 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; int removed = 0; 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(list); removed = 1; ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE); ast_audiohook_unlock(audiohook); audiohook->manipulate_callback(audiohook, NULL, NULL, 0); if (ast_channel_is_bridged(chan)) { ast_channel_set_unbridged_nolock(chan, 1); } 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; /* if an audiohook got removed, reset samplerate compatibility */ if (removed) { audiohook_list_set_samplerate_compatibility(audiohook_list); } return frame; } static struct ast_frame *audiohook_list_translate_to_slin(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_frame *new_frame = frame; struct ast_format *slin; /* * If we are capable of sample rates other that 8khz, update the internal * audiohook_list's rate and higher sample rate audio arrives. If native * slin compatibility is turned on all audiohooks in the list will be * updated as well during read/write processing. */ audiohook_list->list_internal_samp_rate = MAX(ast_format_get_sample_rate(frame->subclass.format), audiohook_list->list_internal_samp_rate); slin = ast_format_cache_get_slin_by_rate(audiohook_list->list_internal_samp_rate); if (ast_format_cmp(frame->subclass.format, slin) == AST_FORMAT_CMP_EQUAL) { return new_frame; } if (!in_translate->format || ast_format_cmp(frame->subclass.format, in_translate->format) != AST_FORMAT_CMP_EQUAL) { struct ast_trans_pvt *new_trans; new_trans = ast_translator_build_path(slin, frame->subclass.format); if (!new_trans) { return NULL; } if (in_translate->trans_pvt) { ast_translator_free_path(in_translate->trans_pvt); } in_translate->trans_pvt = new_trans; ao2_replace(in_translate->format, frame->subclass.format); } if (!(new_frame = ast_translate(in_translate->trans_pvt, frame, 0))) { return NULL; } return new_frame; } static struct ast_frame *audiohook_list_translate_to_native(struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *slin_frame, struct ast_format *outformat) { struct ast_audiohook_translate *out_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->out_translate[0] : &audiohook_list->out_translate[1]); struct ast_frame *outframe = NULL; if (ast_format_cmp(slin_frame->subclass.format, outformat) == AST_FORMAT_CMP_NOT_EQUAL) { /* rebuild translators if necessary */ if (ast_format_cmp(out_translate->format, outformat) == AST_FORMAT_CMP_NOT_EQUAL) { if (out_translate->trans_pvt) { ast_translator_free_path(out_translate->trans_pvt); } if (!(out_translate->trans_pvt = ast_translator_build_path(outformat, slin_frame->subclass.format))) { return NULL; } ao2_replace(out_translate->format, outformat); } /* translate back to the format the frame came in as. */ if (!(outframe = ast_translate(out_translate->trans_pvt, slin_frame, 0))) { return NULL; } } return outframe; } /*! *\brief Set the audiohook's internal sample rate to the audiohook_list's rate, * but only when native slin compatibility is turned on. * * \param audiohook_list audiohook_list data object * \param audiohook the audiohook to update * \param rate the current max internal sample rate */ static void audiohook_list_set_hook_rate(struct ast_audiohook_list *audiohook_list, struct ast_audiohook *audiohook, int *rate) { /* The rate should always be the max between itself and the hook */ if (audiohook->hook_internal_samp_rate > *rate) { *rate = audiohook->hook_internal_samp_rate; } /* * If native slin compatibility is turned on then update the audiohook * with the audiohook_list's current rate. Note, the audiohook's rate is * set to the audiohook_list's rate and not the given rate. If there is * a change in rate the hook's rate is changed on its next check. */ if (audiohook_list->native_slin_compatible) { ast_set_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE); audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1); } else { ast_clear_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE); } } /*! * \brief Pass an AUDIO frame off to be handled by the audiohook core * * \details * This function has 3 ast_frames and 3 parts to handle each. At the beginning of this * function all 3 frames, start_frame, middle_frame, and end_frame point to the initial * input frame. * * Part_1: Translate the start_frame into SLINEAR audio if it is not already in that * format. The result of this part is middle_frame is guaranteed to be in * SLINEAR format for Part_2. * Part_2: Send middle_frame off to spies and manipulators. At this point middle_frame is * either a new frame as result of the translation, or points directly to the start_frame * because no translation to SLINEAR audio was required. * Part_3: Translate end_frame's audio back into the format of start frame if necessary. This * is only necessary if manipulation of middle_frame occurred. * * \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 frame on success * \retval 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_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame; struct ast_audiohook *audiohook = NULL; int samples; int middle_frame_manipulated = 0; int removed = 0; int internal_sample_rate; /* ---Part_1. translate start_frame to SLINEAR if necessary. */ if (!(middle_frame = audiohook_list_translate_to_slin(audiohook_list, direction, start_frame))) { return frame; } /* If the translation resulted in an interpolated frame then immediately return as audiohooks * rely on actual media being present to do things. */ if (!middle_frame->data.ptr) { if (middle_frame != start_frame) { ast_frfree(middle_frame); } return start_frame; } samples = middle_frame->samples; /* * While processing each audiohook check to see if the internal sample rate needs * to be adjusted (it should be the highest rate specified between formats and * hooks). The given audiohook_list's internal sample rate is then set to the * updated value before returning. * * If slin compatibility mode is turned on then an audiohook's internal sample * rate is set to its audiohook_list's rate. If an audiohook_list's rate is * adjusted during this pass then the change is picked up by the audiohooks * on the next pass. */ internal_sample_rate = audiohook_list->list_internal_samp_rate; /* ---Part_2: Send middle_frame to spy and manipulator lists. middle_frame is guaranteed to be SLINEAR here.*/ /* 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(list); removed = 1; ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE); ast_audiohook_unlock(audiohook); if (ast_channel_is_bridged(chan)) { ast_channel_set_unbridged_nolock(chan, 1); } continue; } audiohook_list_set_hook_rate(audiohook_list, audiohook, &internal_sample_rate); 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 (!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) { struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory); ast_audiohook_lock(audiohook); if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) { AST_LIST_REMOVE_CURRENT(list); removed = 1; ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE); ast_audiohook_unlock(audiohook); if (ast_channel_is_bridged(chan)) { ast_channel_set_unbridged_nolock(chan, 1); } continue; } audiohook_list_set_hook_rate(audiohook_list, audiohook, &internal_sample_rate); if (ast_slinfactory_available(factory) >= samples && ast_slinfactory_read(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.ptr, data2 = combine_buf; i < samples; i++, data1++, data2++) { ast_slinear_saturated_add(data1, data2); } middle_frame_manipulated = 1; } /* 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(list); removed = 1; ast_audiohook_update_status(audiohook, 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); if (ast_channel_is_bridged(chan)) { ast_channel_set_unbridged_nolock(chan, 1); } continue; } audiohook_list_set_hook_rate(audiohook_list, audiohook, &internal_sample_rate); /* * Feed in frame to manipulation. */ if (!audiohook->manipulate_callback(audiohook, chan, middle_frame, direction)) { /* * XXX FAILURES ARE IGNORED XXX * If the manipulation fails then the frame will be returned in its original state. * Since there are potentially more manipulator callbacks in the list, no action should * be taken here to exit early. */ middle_frame_manipulated = 1; } ast_audiohook_unlock(audiohook); } AST_LIST_TRAVERSE_SAFE_END; } /* ---Part_3: Decide what to do with the end_frame (whether to transcode or not) */ if (middle_frame_manipulated) { if (!(end_frame = audiohook_list_translate_to_native(audiohook_list, direction, middle_frame, start_frame->subclass.format))) { /* translation failed, so just pass back the input frame */ end_frame = start_frame; } } else { end_frame = start_frame; } /* clean up our middle_frame if required */ if (middle_frame != end_frame) { ast_frfree(middle_frame); middle_frame = NULL; } /* Before returning, if an audiohook got removed, reset samplerate compatibility */ if (removed) { audiohook_list_set_samplerate_compatibility(audiohook_list); } else { /* * Set the audiohook_list's rate to the updated rate. Note that if a hook * was removed then the list's internal rate is reset to the default. */ audiohook_list->list_internal_samp_rate = internal_sample_rate; } return end_frame; } int ast_audiohook_write_list_empty(struct ast_audiohook_list *audiohook_list) { return !audiohook_list || (AST_LIST_EMPTY(&audiohook_list->spy_list) && AST_LIST_EMPTY(&audiohook_list->whisper_list) && AST_LIST_EMPTY(&audiohook_list->manipulate_list)); } 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 wait; struct timespec ts; wait = ast_tvadd(ast_tvnow(), ast_samp2tv(50000, 1000)); ts.tv_sec = wait.tv_sec; ts.tv_nsec = wait.tv_usec * 1000; ast_cond_timedwait(&audiohook->trigger, &audiohook->lock, &ts); return; } /* Count number of channel audiohooks by type, regardless of type */ int ast_channel_audiohook_count_by_source(struct ast_channel *chan, const char *source, enum ast_audiohook_type type) { int count = 0; struct ast_audiohook *ah = NULL; if (!ast_channel_audiohooks(chan)) { return -1; } switch (type) { case AST_AUDIOHOOK_TYPE_SPY: AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->spy_list, ah, list) { if (!strcmp(ah->source, source)) { count++; } } break; case AST_AUDIOHOOK_TYPE_WHISPER: AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->whisper_list, ah, list) { if (!strcmp(ah->source, source)) { count++; } } break; case AST_AUDIOHOOK_TYPE_MANIPULATE: AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->manipulate_list, ah, list) { if (!strcmp(ah->source, source)) { count++; } } break; default: ast_debug(1, "Invalid audiohook type supplied, (%u)\n", type); return -1; } return count; } /* Count number of channel audiohooks by type that are running */ int ast_channel_audiohook_count_by_source_running(struct ast_channel *chan, const char *source, enum ast_audiohook_type type) { int count = 0; struct ast_audiohook *ah = NULL; if (!ast_channel_audiohooks(chan)) return -1; switch (type) { case AST_AUDIOHOOK_TYPE_SPY: AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->spy_list, ah, list) { if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING)) count++; } break; case AST_AUDIOHOOK_TYPE_WHISPER: AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->whisper_list, ah, list) { if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING)) count++; } break; case AST_AUDIOHOOK_TYPE_MANIPULATE: AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->manipulate_list, ah, list) { if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING)) count++; } break; default: ast_debug(1, "Invalid audiohook type supplied, (%u)\n", type); return -1; } return count; } /*! \brief Audiohook volume adjustment structure */ struct audiohook_volume { struct ast_audiohook audiohook; /*!< Audiohook attached to the channel */ int read_adjustment; /*!< Value to adjust frames read from the channel by */ int write_adjustment; /*!< Value to adjust frames written to the channel by */ }; /*! \brief Callback used to destroy the audiohook volume datastore * \param data Volume information structure */ static void audiohook_volume_destroy(void *data) { struct audiohook_volume *audiohook_volume = data; /* Destroy the audiohook as it is no longer in use */ ast_audiohook_destroy(&audiohook_volume->audiohook); /* Finally free ourselves, we are of no more use */ ast_free(audiohook_volume); return; } /*! \brief Datastore used to store audiohook volume information */ static const struct ast_datastore_info audiohook_volume_datastore = { .type = "Volume", .destroy = audiohook_volume_destroy, }; /*! \brief Helper function which actually gets called by audiohooks to perform the adjustment * \param audiohook Audiohook attached to the channel * \param chan Channel we are attached to * \param frame Frame of audio we want to manipulate * \param direction Direction the audio came in from * \retval 0 on success * \retval -1 on failure */ static int audiohook_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 audiohook_volume *audiohook_volume = NULL; int *gain = NULL; /* If the audiohook is shutting down don't even bother */ if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) { return 0; } /* Try to find the datastore containg adjustment information, if we can't just bail out */ if (!(datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) { return 0; } audiohook_volume = datastore->data; /* Based on direction grab the appropriate adjustment value */ if (direction == AST_AUDIOHOOK_DIRECTION_READ) { gain = &audiohook_volume->read_adjustment; } else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) { gain = &audiohook_volume->write_adjustment; } /* If an adjustment value is present modify the frame */ if (gain && *gain) { ast_frame_adjust_volume(frame, *gain); } return 0; } /*! \brief Helper function which finds and optionally creates an audiohook_volume_datastore datastore on a channel * \param chan Channel to look on * \param create Whether to create the datastore if not found * \return audiohook_volume structure on success * \retval NULL on failure */ static struct audiohook_volume *audiohook_volume_get(struct ast_channel *chan, int create) { struct ast_datastore *datastore = NULL; struct audiohook_volume *audiohook_volume = NULL; /* If we are able to find the datastore return the contents (which is actually an audiohook_volume structure) */ if ((datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) { return datastore->data; } /* If we are not allowed to create a datastore or if we fail to create a datastore, bail out now as we have nothing for them */ if (!create || !(datastore = ast_datastore_alloc(&audiohook_volume_datastore, NULL))) { return NULL; } /* Create a new audiohook_volume structure to contain our adjustments and audiohook */ if (!(audiohook_volume = ast_calloc(1, sizeof(*audiohook_volume)))) { ast_datastore_free(datastore); return NULL; } /* Setup our audiohook structure so we can manipulate the audio */ ast_audiohook_init(&audiohook_volume->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Volume", AST_AUDIOHOOK_MANIPULATE_ALL_RATES); audiohook_volume->audiohook.manipulate_callback = audiohook_volume_callback; /* Attach the audiohook_volume blob to the datastore and attach to the channel */ datastore->data = audiohook_volume; ast_channel_datastore_add(chan, datastore); /* All is well... put the audiohook into motion */ ast_audiohook_attach(chan, &audiohook_volume->audiohook); return audiohook_volume; } int ast_audiohook_volume_set(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume) { struct audiohook_volume *audiohook_volume = NULL; /* Attempt to find the audiohook volume information, but only create it if we are not setting the adjustment value to zero */ if (!(audiohook_volume = audiohook_volume_get(chan, (volume ? 1 : 0)))) { return -1; } /* Now based on the direction set the proper value */ if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) { audiohook_volume->read_adjustment = volume; } if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) { audiohook_volume->write_adjustment = volume; } return 0; } int ast_audiohook_volume_get(struct ast_channel *chan, enum ast_audiohook_direction direction) { struct audiohook_volume *audiohook_volume = NULL; int adjustment = 0; /* Attempt to find the audiohook volume information, but do not create it as we only want to look at the values */ if (!(audiohook_volume = audiohook_volume_get(chan, 0))) { return 0; } /* Grab the adjustment value based on direction given */ if (direction == AST_AUDIOHOOK_DIRECTION_READ) { adjustment = audiohook_volume->read_adjustment; } else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) { adjustment = audiohook_volume->write_adjustment; } return adjustment; } int ast_audiohook_volume_adjust(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume) { struct audiohook_volume *audiohook_volume = NULL; /* Attempt to find the audiohook volume information, and create an audiohook if none exists */ if (!(audiohook_volume = audiohook_volume_get(chan, 1))) { return -1; } /* Based on the direction change the specific adjustment value */ if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) { audiohook_volume->read_adjustment += volume; } if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) { audiohook_volume->write_adjustment += volume; } return 0; } int ast_audiohook_set_mute(struct ast_channel *chan, const char *source, enum ast_audiohook_flags flag, int clear) { struct ast_audiohook *audiohook = NULL; ast_channel_lock(chan); /* Ensure the channel has audiohooks on it */ if (!ast_channel_audiohooks(chan)) { ast_channel_unlock(chan); return -1; } audiohook = find_audiohook_by_source(ast_channel_audiohooks(chan), source); if (audiohook) { if (clear) { ast_clear_flag(audiohook, flag); } else { ast_set_flag(audiohook, flag); } } ast_channel_unlock(chan); return (audiohook ? 0 : -1); }