From 7f23115ad2faeee58865afbec6bc11a43210fde7 Mon Sep 17 00:00:00 2001 From: David Vossel Date: Thu, 21 Apr 2011 18:11:40 +0000 Subject: New HD ConfBridge conferencing application. Includes a new highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8khz-192khz. Review: https://reviewboard.asterisk.org/r/1147/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314598 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- bridges/bridge_builtin_features.c | 6 +- bridges/bridge_softmix.c | 712 ++++++++++++++++++++++++++++++-------- 2 files changed, 563 insertions(+), 155 deletions(-) (limited to 'bridges') diff --git a/bridges/bridge_builtin_features.c b/bridges/bridge_builtin_features.c index 67da6489b..2b21933e2 100644 --- a/bridges/bridge_builtin_features.c +++ b/bridges/bridge_builtin_features.c @@ -198,12 +198,12 @@ static int feature_attended_transfer(struct ast_bridge *bridge, struct ast_bridg ast_bridge_features_enable(&caller_features, AST_BRIDGE_BUILTIN_HANGUP, (attended_transfer && !ast_strlen_zero(attended_transfer->complete) ? attended_transfer->complete : "*1"), NULL); ast_bridge_features_hook(&caller_features, (attended_transfer && !ast_strlen_zero(attended_transfer->threeway) ? attended_transfer->threeway : "*2"), - attended_threeway_transfer, NULL); + attended_threeway_transfer, NULL, NULL); ast_bridge_features_hook(&caller_features, (attended_transfer && !ast_strlen_zero(attended_transfer->abort) ? attended_transfer->abort : "*3"), - attended_abort_transfer, NULL); + attended_abort_transfer, NULL, NULL); /* But for the caller we want to join the bridge in a blocking fashion so we don't spin around in this function doing nothing while waiting */ - attended_bridge_result = ast_bridge_join(attended_bridge, bridge_channel->chan, NULL, &caller_features); + attended_bridge_result = ast_bridge_join(attended_bridge, bridge_channel->chan, NULL, &caller_features, NULL); /* Since the above returned the caller features structure is of no more use */ ast_bridge_features_cleanup(&caller_features); diff --git a/bridges/bridge_softmix.c b/bridges/bridge_softmix.c index 1ac2780de..eb476932f 100644 --- a/bridges/bridge_softmix.c +++ b/bridges/bridge_softmix.c @@ -1,9 +1,10 @@ /* * Asterisk -- An open source telephony toolkit. * - * Copyright (C) 2007, Digium, Inc. + * Copyright (C) 2011, Digium, Inc. * * Joshua Colp + * David Vossel * * See http://www.asterisk.org for more information about * the Asterisk project. Please do not directly contact @@ -21,12 +22,9 @@ * \brief Multi-party software based channel mixing * * \author Joshua Colp + * \author David Vossel * * \ingroup bridges - * - * \todo This bridge operates in 8 kHz mode unless a define is uncommented. - * This needs to be improved so the bridge moves between the dominant codec as needed depending - * on channels present in the bridge and transcoding capabilities. */ #include "asterisk.h" @@ -51,20 +49,26 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #include "asterisk/slinfactory.h" #include "asterisk/astobj2.h" #include "asterisk/timing.h" +#include "asterisk/translate.h" -#define MAX_DATALEN 3840 +#define MAX_DATALEN 8096 /*! \brief Interval at which mixing will take place. Valid options are 10, 20, and 40. */ -#define SOFTMIX_INTERVAL 20 +#define DEFAULT_SOFTMIX_INTERVAL 20 /*! \brief Size of the buffer used for sample manipulation */ -#define SOFTMIX_DATALEN(rate) ((rate/50) * (SOFTMIX_INTERVAL / 10)) +#define SOFTMIX_DATALEN(rate, interval) ((rate/50) * (interval / 10)) /*! \brief Number of samples we are dealing with */ -#define SOFTMIX_SAMPLES(rate) (SOFTMIX_DATALEN(rate) / 2) +#define SOFTMIX_SAMPLES(rate, interval) (SOFTMIX_DATALEN(rate, interval) / 2) + +/*! \brief Number of mixing iterations to perform between gathering statistics. */ +#define SOFTMIX_STAT_INTERVAL 100 -/*! \brief Define used to turn on 16 kHz audio support */ -/* #define SOFTMIX_16_SUPPORT */ +/* This is the threshold in ms at which a channel's own audio will stop getting + * mixed out its own write audio stream because it is not talking. */ +#define DEFAULT_SOFTMIX_SILENCE_THRESHOLD 2500 +#define DEFAULT_SOFTMIX_TALKING_THRESHOLD 160 /*! \brief Structure which contains per-channel mixing information */ struct softmix_channel { @@ -73,7 +77,14 @@ struct softmix_channel { /*! Factory which contains audio read in from the channel */ struct ast_slinfactory factory; /*! Frame that contains mixed audio to be written out to the channel */ - struct ast_frame frame; + struct ast_frame write_frame; + /*! Frame that contains mixed audio read from the channel */ + struct ast_frame read_frame; + /*! DSP for detecting silence */ + struct ast_dsp *dsp; + /*! Bit used to indicate if a channel is talking or not. This affects how + * the channel's audio is mixed back to it. */ + int talking:1; /*! Bit used to indicate that the channel provided audio for this mixing interval */ int have_audio:1; /*! Bit used to indicate that a frame is available to be written out to the channel */ @@ -87,66 +98,268 @@ struct softmix_channel { struct softmix_bridge_data { struct ast_timer *timer; unsigned int internal_rate; + unsigned int internal_mixing_interval; +}; + +struct softmix_stats { + /*! Each index represents a sample rate used above the internal rate. */ + unsigned int sample_rates[16]; + /*! Each index represents the number of channels using the same index in the sample_rates array. */ + unsigned int num_channels[16]; + /*! the number of channels above the internal sample rate */ + unsigned int num_above_internal_rate; + /*! the number of channels at the internal sample rate */ + unsigned int num_at_internal_rate; + /*! the absolute highest sample rate supported by any channel in the bridge */ + unsigned int highest_supported_rate; + /*! Is the sample rate locked by the bridge, if so what is that rate.*/ + unsigned int locked_rate; }; +struct softmix_mixing_array { + int max_num_entries; + int used_entries; + int16_t **buffers; +}; + +struct softmix_translate_helper_entry { + int num_times_requested; /*!< Once this entry is no longer requested, free the trans_pvt + and re-init if it was usable. */ + struct ast_format dst_format; /*!< The destination format for this helper */ + struct ast_trans_pvt *trans_pvt; /*!< the translator for this slot. */ + struct ast_frame *out_frame; /*!< The output frame from the last translation */ + AST_LIST_ENTRY(softmix_translate_helper_entry) entry; +}; + +struct softmix_translate_helper { + struct ast_format slin_src; /*!< the source format expected for all the translators */ + AST_LIST_HEAD_NOLOCK(, softmix_translate_helper_entry) entries; +}; + +static struct softmix_translate_helper_entry *softmix_translate_helper_entry_alloc(struct ast_format *dst) +{ + struct softmix_translate_helper_entry *entry; + if (!(entry = ast_calloc(1, sizeof(*entry)))) { + return NULL; + } + ast_format_copy(&entry->dst_format, dst); + return entry; +} + +static void *softmix_translate_helper_free_entry(struct softmix_translate_helper_entry *entry) +{ + if (entry->trans_pvt) { + ast_translator_free_path(entry->trans_pvt); + } + if (entry->out_frame) { + ast_frfree(entry->out_frame); + } + ast_free(entry); + return NULL; +} + +static void softmix_translate_helper_init(struct softmix_translate_helper *trans_helper, unsigned int sample_rate) +{ + memset(trans_helper, 0, sizeof(*trans_helper)); + ast_format_set(&trans_helper->slin_src, ast_format_slin_by_rate(sample_rate), 0); +} + +static void softmix_translate_helper_destroy(struct softmix_translate_helper *trans_helper) +{ + struct softmix_translate_helper_entry *entry; + + while ((entry = AST_LIST_REMOVE_HEAD(&trans_helper->entries, entry))) { + softmix_translate_helper_free_entry(entry); + } +} + +static void softmix_translate_helper_change_rate(struct softmix_translate_helper *trans_helper, unsigned int sample_rate) +{ + struct softmix_translate_helper_entry *entry; + + ast_format_set(&trans_helper->slin_src, ast_format_slin_by_rate(sample_rate), 0); + AST_LIST_TRAVERSE_SAFE_BEGIN(&trans_helper->entries, entry, entry) { + if (entry->trans_pvt) { + ast_translator_free_path(entry->trans_pvt); + if (!(entry->trans_pvt = ast_translator_build_path(&entry->dst_format, &trans_helper->slin_src))) { + AST_LIST_REMOVE_CURRENT(entry); + entry = softmix_translate_helper_free_entry(entry); + } + } + } + AST_LIST_TRAVERSE_SAFE_END; +} + +/*! + * \internal + * \brief Get the next available audio on the softmix channel's read stream + * and determine if it should be mixed out or not on the write stream. + * + * \retval pointer to buffer containing the exact number of samples requested on success. + * \retval NULL if no samples are present + */ +static int16_t *softmix_process_read_audio(struct softmix_channel *sc, unsigned int num_samples) +{ + if ((ast_slinfactory_available(&sc->factory) >= num_samples) && + ast_slinfactory_read(&sc->factory, sc->our_buf, num_samples)) { + sc->have_audio = 1; + return sc->our_buf; + } + sc->have_audio = 0; + return NULL; +} + +/*! + * \internal + * \brief Process a softmix channel's write audio + * + * \details This function will remove the channel's talking from its own audio if present and + * possibly even do the channel's write translation for it depending on how many other + * channels use the same write format. + */ +static void softmix_process_write_audio(struct softmix_translate_helper *trans_helper, + struct ast_format *raw_write_fmt, + struct softmix_channel *sc) +{ + struct softmix_translate_helper_entry *entry = NULL; + int i; + + /* If we provided audio that was not determined to be silence, + * then take it out while in slinear format. */ + if (sc->have_audio && sc->talking) { + for (i = 0; i < sc->write_frame.samples; i++) { + ast_slinear_saturated_subtract(&sc->final_buf[i], &sc->our_buf[i]); + } + /* do not do any special write translate optimization if we had to make + * a special mix for them to remove their own audio. */ + return; + } + + AST_LIST_TRAVERSE(&trans_helper->entries, entry, entry) { + if (ast_format_cmp(&entry->dst_format, raw_write_fmt) == AST_FORMAT_CMP_EQUAL) { + entry->num_times_requested++; + } else { + continue; + } + if (!entry->trans_pvt && (entry->num_times_requested > 1)) { + entry->trans_pvt = ast_translator_build_path(&entry->dst_format, &trans_helper->slin_src); + } + if (entry->trans_pvt && !entry->out_frame) { + entry->out_frame = ast_translate(entry->trans_pvt, &sc->write_frame, 0); + } + if (entry->out_frame && (entry->out_frame->datalen < MAX_DATALEN)) { + ast_format_copy(&sc->write_frame.subclass.format, &entry->out_frame->subclass.format); + memcpy(sc->final_buf, entry->out_frame->data.ptr, entry->out_frame->datalen); + sc->write_frame.datalen = entry->out_frame->datalen; + sc->write_frame.samples = entry->out_frame->samples; + } + break; + } + + /* add new entry into list if this format destination was not matched. */ + if (!entry && (entry = softmix_translate_helper_entry_alloc(raw_write_fmt))) { + AST_LIST_INSERT_HEAD(&trans_helper->entries, entry, entry); + } +} + +static void softmix_translate_helper_cleanup(struct softmix_translate_helper *trans_helper) +{ + struct softmix_translate_helper_entry *entry = NULL; + AST_LIST_TRAVERSE(&trans_helper->entries, entry, entry) { + if (entry->out_frame) { + ast_frfree(entry->out_frame); + entry->out_frame = NULL; + } + entry->num_times_requested = 0; + } +} + +static void softmix_bridge_data_destroy(void *obj) +{ + struct softmix_bridge_data *softmix_data = obj; + ast_timer_close(softmix_data->timer); +} + /*! \brief Function called when a bridge is created */ static int softmix_bridge_create(struct ast_bridge *bridge) { - struct softmix_bridge_data *bridge_data; + struct softmix_bridge_data *softmix_data; - if (!(bridge_data = ast_calloc(1, sizeof(*bridge_data)))) { + if (!(softmix_data = ao2_alloc(sizeof(*softmix_data), softmix_bridge_data_destroy))) { return -1; } - if (!(bridge_data->timer = ast_timer_open())) { - ast_free(bridge_data); + if (!(softmix_data->timer = ast_timer_open())) { + ao2_ref(softmix_data, -1); return -1; } /* start at 8khz, let it grow from there */ - bridge_data->internal_rate = 8000; + softmix_data->internal_rate = 8000; + softmix_data->internal_mixing_interval = DEFAULT_SOFTMIX_INTERVAL; - bridge->bridge_pvt = bridge_data; + bridge->bridge_pvt = softmix_data; return 0; } /*! \brief Function called when a bridge is destroyed */ static int softmix_bridge_destroy(struct ast_bridge *bridge) { - struct softmix_bridge_data *bridge_data = bridge->bridge_pvt; + struct softmix_bridge_data *softmix_data = bridge->bridge_pvt; if (!bridge->bridge_pvt) { return -1; } - ast_timer_close(bridge_data->timer); - ast_free(bridge_data); + ao2_ref(softmix_data, -1); + bridge->bridge_pvt = NULL; return 0; } -static void set_softmix_bridge_data(int rate, struct ast_bridge_channel *bridge_channel, int reset) +static void set_softmix_bridge_data(int rate, int interval, struct ast_bridge_channel *bridge_channel, int reset) { struct softmix_channel *sc = bridge_channel->bridge_pvt; + unsigned int channel_read_rate = ast_format_rate(&bridge_channel->chan->rawreadformat); + + ast_mutex_lock(&sc->lock); if (reset) { ast_slinfactory_destroy(&sc->factory); + ast_dsp_free(sc->dsp); } - /* Setup frame parameters */ - sc->frame.frametype = AST_FRAME_VOICE; - - ast_format_set(&sc->frame.subclass.format, ast_format_slin_by_rate(rate), 0); - sc->frame.data.ptr = sc->final_buf; - sc->frame.datalen = SOFTMIX_DATALEN(rate); - sc->frame.samples = SOFTMIX_SAMPLES(rate); + /* Setup read/write frame parameters */ + sc->write_frame.frametype = AST_FRAME_VOICE; + ast_format_set(&sc->write_frame.subclass.format, ast_format_slin_by_rate(rate), 0); + sc->write_frame.data.ptr = sc->final_buf; + sc->write_frame.datalen = SOFTMIX_DATALEN(rate, interval); + sc->write_frame.samples = SOFTMIX_SAMPLES(rate, interval); + + sc->read_frame.frametype = AST_FRAME_VOICE; + ast_format_set(&sc->read_frame.subclass.format, ast_format_slin_by_rate(channel_read_rate), 0); + sc->read_frame.data.ptr = sc->our_buf; + sc->read_frame.datalen = SOFTMIX_DATALEN(channel_read_rate, interval); + sc->read_frame.samples = SOFTMIX_SAMPLES(channel_read_rate, interval); /* Setup smoother */ - ast_slinfactory_init_with_format(&sc->factory, &sc->frame.subclass.format); + ast_slinfactory_init_with_format(&sc->factory, &sc->write_frame.subclass.format); + + /* set new read and write formats on channel. */ + ast_set_read_format(bridge_channel->chan, &sc->read_frame.subclass.format); + ast_set_write_format(bridge_channel->chan, &sc->write_frame.subclass.format); + + /* set up new DSP. This is on the read side only right before the read frame enters the smoother. */ + sc->dsp = ast_dsp_new_with_rate(channel_read_rate); + /* we want to aggressively detect silence to avoid feedback */ + if (bridge_channel->tech_args.talking_threshold) { + ast_dsp_set_threshold(sc->dsp, bridge_channel->tech_args.talking_threshold); + } else { + ast_dsp_set_threshold(sc->dsp, DEFAULT_SOFTMIX_TALKING_THRESHOLD); + } - ast_set_read_format(bridge_channel->chan, &sc->frame.subclass.format); - ast_set_write_format(bridge_channel->chan, &sc->frame.subclass.format); + ast_mutex_unlock(&sc->lock); } /*! \brief Function called when a channel is joined into the bridge */ static int softmix_bridge_join(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel) { struct softmix_channel *sc = NULL; - struct softmix_bridge_data *bridge_data = bridge->bridge_pvt; + struct softmix_bridge_data *softmix_data = bridge->bridge_pvt; /* Create a new softmix_channel structure and allocate various things on it */ if (!(sc = ast_calloc(1, sizeof(*sc)))) { @@ -159,7 +372,9 @@ static int softmix_bridge_join(struct ast_bridge *bridge, struct ast_bridge_chan /* Can't forget to record our pvt structure within the bridged channel structure */ bridge_channel->bridge_pvt = sc; - set_softmix_bridge_data(bridge_data->internal_rate, bridge_channel, 0); + set_softmix_bridge_data(softmix_data->internal_rate, + softmix_data->internal_mixing_interval ? softmix_data->internal_mixing_interval : DEFAULT_SOFTMIX_INTERVAL, + bridge_channel, 0); return 0; } @@ -169,44 +384,102 @@ static int softmix_bridge_leave(struct ast_bridge *bridge, struct ast_bridge_cha { struct softmix_channel *sc = bridge_channel->bridge_pvt; + if (!(bridge_channel->bridge_pvt)) { + return 0; + } + bridge_channel->bridge_pvt = NULL; + /* Drop mutex lock */ ast_mutex_destroy(&sc->lock); /* Drop the factory */ ast_slinfactory_destroy(&sc->factory); + /* Drop the DSP */ + ast_dsp_free(sc->dsp); + /* Eep! drop ourselves */ ast_free(sc); return 0; } +/*! + * \internal + * \brief If the bridging core passes DTMF to us, then they want it to be distributed out to all memebers. Do that here. + */ +static void softmix_pass_dtmf(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame) +{ + struct ast_bridge_channel *tmp; + AST_LIST_TRAVERSE(&bridge->channels, tmp, entry) { + if (tmp == bridge_channel) { + continue; + } + ast_write(tmp->chan, frame); + } +} + /*! \brief Function called when a channel writes a frame into the bridge */ static enum ast_bridge_write_result softmix_bridge_write(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame) { struct softmix_channel *sc = bridge_channel->bridge_pvt; + struct softmix_bridge_data *softmix_data = bridge->bridge_pvt; + int totalsilence = 0; + int silence_threshold = bridge_channel->tech_args.silence_threshold ? + bridge_channel->tech_args.silence_threshold : + DEFAULT_SOFTMIX_SILENCE_THRESHOLD; + char update_talking = -1; /* if this is set to 0 or 1, tell the bridge that the channel has started or stopped talking. */ /* Only accept audio frames, all others are unsupported */ - if (frame->frametype != AST_FRAME_VOICE) { + if (frame->frametype == AST_FRAME_DTMF_END || frame->frametype == AST_FRAME_DTMF_BEGIN) { + softmix_pass_dtmf(bridge, bridge_channel, frame); + return AST_BRIDGE_WRITE_SUCCESS; + } else if (frame->frametype != AST_FRAME_VOICE) { return AST_BRIDGE_WRITE_UNSUPPORTED; } ast_mutex_lock(&sc->lock); - /* If a frame was provided add it to the smoother */ - if (frame->frametype == AST_FRAME_VOICE && ast_format_is_slinear(&frame->subclass.format)) { + ast_dsp_silence(sc->dsp, frame, &totalsilence); + if (totalsilence < silence_threshold) { + if (!sc->talking) { + update_talking = 1; + } + sc->talking = 1; /* tell the write process we have audio to be mixed out */ + } else { + if (sc->talking) { + update_talking = 0; + } + sc->talking = 0; + } + + /* Before adding audio in, make sure we haven't fallen behind. If audio has fallen + * behind 4 times the amount of samples mixed on every iteration of the mixer, Re-sync + * the audio by flushing the buffer before adding new audio in. */ + if (ast_slinfactory_available(&sc->factory) > (4 * SOFTMIX_SAMPLES(softmix_data->internal_rate, softmix_data->internal_mixing_interval))) { + ast_slinfactory_flush(&sc->factory); + } + + /* If a frame was provided add it to the smoother, unless drop silence is enabled and this frame + * is not determined to be talking. */ + if (!(bridge_channel->tech_args.drop_silence && !sc->talking) && + (frame->frametype == AST_FRAME_VOICE && ast_format_is_slinear(&frame->subclass.format))) { ast_slinfactory_feed(&sc->factory, frame); } /* If a frame is ready to be written out, do so */ if (sc->have_frame) { - ast_write(bridge_channel->chan, &sc->frame); + ast_write(bridge_channel->chan, &sc->write_frame); sc->have_frame = 0; } /* Alllll done */ ast_mutex_unlock(&sc->lock); + if (update_talking != -1) { + ast_bridge_notify_talking(bridge, bridge_channel, update_talking); + } + return AST_BRIDGE_WRITE_SUCCESS; } @@ -218,7 +491,7 @@ static int softmix_bridge_poke(struct ast_bridge *bridge, struct ast_bridge_chan ast_mutex_lock(&sc->lock); if (sc->have_frame) { - ast_write(bridge_channel->chan, &sc->frame); + ast_write(bridge_channel->chan, &sc->write_frame); sc->have_frame = 0; } @@ -227,167 +500,306 @@ static int softmix_bridge_poke(struct ast_bridge *bridge, struct ast_bridge_chan return 0; } +static void gather_softmix_stats(struct softmix_stats *stats, + const struct softmix_bridge_data *softmix_data, + struct ast_bridge_channel *bridge_channel) +{ + int channel_native_rate; + int i; + /* Gather stats about channel sample rates. */ + channel_native_rate = MAX(ast_format_rate(&bridge_channel->chan->rawwriteformat), + ast_format_rate(&bridge_channel->chan->rawreadformat)); + + if (channel_native_rate > stats->highest_supported_rate) { + stats->highest_supported_rate = channel_native_rate; + } + if (channel_native_rate > softmix_data->internal_rate) { + for (i = 0; i < ARRAY_LEN(stats->sample_rates); i++) { + if (stats->sample_rates[i] == channel_native_rate) { + stats->num_channels[i]++; + break; + } else if (!stats->sample_rates[i]) { + stats->sample_rates[i] = channel_native_rate; + stats->num_channels[i]++; + break; + } + } + stats->num_above_internal_rate++; + } else if (channel_native_rate == softmix_data->internal_rate) { + stats->num_at_internal_rate++; + } +} +/*! + * \internal + * \brief Analyse mixing statistics and change bridges internal rate + * if necessary. + * + * \retval 0, no changes to internal rate + * \ratval 1, internal rate was changed, update all the channels on the next mixing iteration. + */ +static unsigned int analyse_softmix_stats(struct softmix_stats *stats, struct softmix_bridge_data *softmix_data) +{ + int i; + /* Re-adjust the internal bridge sample rate if + * 1. The bridge's internal sample rate is locked in at a sample + * rate other than the current sample rate being used. + * 2. two or more channels support a higher sample rate + * 3. no channels support the current sample rate or a higher rate + */ + if (stats->locked_rate) { + /* if the rate is locked by the bridge, only update it if it differs + * from the current rate we are using. */ + if (softmix_data->internal_rate != stats->locked_rate) { + softmix_data->internal_rate = stats->locked_rate; + ast_debug(1, " Bridge is locked in at sample rate %d\n", softmix_data->internal_rate); + return 1; + } + } else if (stats->num_above_internal_rate >= 2) { + /* the highest rate is just used as a starting point */ + unsigned int best_rate = stats->highest_supported_rate; + int best_index = -1; + + for (i = 0; i < ARRAY_LEN(stats->num_channels); i++) { + if (stats->num_channels[i]) { + break; + } + /* best_rate starts out being the first sample rate + * greater than the internal sample rate that 2 or + * more channels support. */ + if (stats->num_channels[i] >= 2 && (best_index == -1)) { + best_rate = stats->sample_rates[i]; + best_index = i; + /* If it has been detected that multiple rates above + * the internal rate are present, compare those rates + * to each other and pick the highest one two or more + * channels support. */ + } else if (((best_index != -1) && + (stats->num_channels[i] >= 2) && + (stats->sample_rates[best_index] < stats->sample_rates[i]))) { + best_rate = stats->sample_rates[i]; + best_index = i; + /* It is possible that multiple channels exist with native sample + * rates above the internal sample rate, but none of those channels + * have the same rate in common. In this case, the lowest sample + * rate among those channels is picked. Over time as additional + * statistic runs are made the internal sample rate number will + * adjust to the most optimal sample rate, but it may take multiple + * iterations. */ + } else if (best_index == -1) { + best_rate = MIN(best_rate, stats->sample_rates[i]); + } + } + + ast_debug(1, " Bridge changed from %d To %d\n", softmix_data->internal_rate, best_rate); + softmix_data->internal_rate = best_rate; + return 1; + } else if (!stats->num_at_internal_rate && !stats->num_above_internal_rate) { + /* In this case, the highest supported rate is actually lower than the internal rate */ + softmix_data->internal_rate = stats->highest_supported_rate; + ast_debug(1, " Bridge changed from %d to %d\n", softmix_data->internal_rate, stats->highest_supported_rate); + return 1; + } + return 0; +} + +static int softmix_mixing_array_init(struct softmix_mixing_array *mixing_array, unsigned int starting_num_entries) +{ + memset(mixing_array, 0, sizeof(*mixing_array)); + mixing_array->max_num_entries = starting_num_entries; + if (!(mixing_array->buffers = ast_calloc(mixing_array->max_num_entries, sizeof(int16_t *)))) { + ast_log(LOG_NOTICE, "Failed to allocate softmix mixing structure. \n"); + return -1; + } + return 0; +} + +static void softmix_mixing_array_destroy(struct softmix_mixing_array *mixing_array) +{ + ast_free(mixing_array->buffers); +} + +static int softmix_mixing_array_grow(struct softmix_mixing_array *mixing_array, unsigned int num_entries) +{ + int16_t **tmp; + /* give it some room to grow since memory is cheap but allocations can be expensive */ + mixing_array->max_num_entries = num_entries; + if (!(tmp = ast_realloc(mixing_array->buffers, (mixing_array->max_num_entries * sizeof(int16_t *))))) { + ast_log(LOG_NOTICE, "Failed to re-allocate softmix mixing structure. \n"); + return -1; + } + mixing_array->buffers = tmp; + return 0; +} + /*! \brief Function which acts as the mixing thread */ static int softmix_bridge_thread(struct ast_bridge *bridge) { - struct { - /*! Each index represents a sample rate used above the internal rate. */ - unsigned int sample_rates[8]; - /*! Each index represents the number of channels using the same index in the sample_rates array. */ - unsigned int num_channels[8]; - /*! the number of channels above the internal sample rate */ - unsigned int num_above_internal_rate; - /*! the number of channels at the internal sample rate */ - unsigned int num_at_internal_rate; - /*! the absolute highest sample rate supported by any channel in the bridge */ - unsigned int highest_supported_rate; - } stats; - struct softmix_bridge_data *bridge_data = bridge->bridge_pvt; - struct ast_timer *timer = bridge_data->timer; - int timingfd = ast_timer_fd(timer); + struct softmix_stats stats = { { 0 }, }; + struct softmix_mixing_array mixing_array; + struct softmix_bridge_data *softmix_data = bridge->bridge_pvt; + struct ast_timer *timer; + struct softmix_translate_helper trans_helper; + int16_t buf[MAX_DATALEN] = { 0, }; + unsigned int stat_iteration_counter = 0; /* counts down, gather stats at zero and reset. */ + int timingfd; int update_all_rates = 0; /* set this when the internal sample rate has changed */ - int i; + int i, x; + int res = -1; - ast_timer_set_rate(timer, (1000 / SOFTMIX_INTERVAL)); + if (!(softmix_data = bridge->bridge_pvt)) { + goto softmix_cleanup; + } + + ao2_ref(softmix_data, 1); + timer = softmix_data->timer; + timingfd = ast_timer_fd(timer); + softmix_translate_helper_init(&trans_helper, softmix_data->internal_rate); + ast_timer_set_rate(timer, (1000 / softmix_data->internal_mixing_interval)); + + /* Give the mixing array room to grow, memory is cheap but allocations are expensive. */ + if (softmix_mixing_array_init(&mixing_array, bridge->num + 10)) { + ast_log(LOG_NOTICE, "Failed to allocate softmix mixing structure. \n"); + goto softmix_cleanup; + } while (!bridge->stop && !bridge->refresh && bridge->array_num) { struct ast_bridge_channel *bridge_channel = NULL; - short buf[MAX_DATALEN] = {0, }; int timeout = -1; + enum ast_format_id cur_slin_id = ast_format_slin_by_rate(softmix_data->internal_rate); + unsigned int softmix_samples = SOFTMIX_SAMPLES(softmix_data->internal_rate, softmix_data->internal_mixing_interval); + unsigned int softmix_datalen = SOFTMIX_DATALEN(softmix_data->internal_rate, softmix_data->internal_mixing_interval); + + if (softmix_datalen > MAX_DATALEN) { + /* This should NEVER happen, but if it does we need to know about it. Almost + * all the memcpys used during this process depend on this assumption. Rather + * than checking this over and over again through out the code, this single + * verification is done on each iteration. */ + ast_log(LOG_WARNING, "Conference mixing error, requested mixing length greater than mixing buffer.\n"); + goto softmix_cleanup; + } - /* these variables help determine if a rate change is required */ - memset(&stats, 0, sizeof(stats)); - stats.highest_supported_rate = 8000; + /* Grow the mixing array buffer as participants are added. */ + if (mixing_array.max_num_entries < bridge->num && softmix_mixing_array_grow(&mixing_array, bridge->num + 5)) { + goto softmix_cleanup; + } + + /* init the number of buffers stored in the mixing array to 0. + * As buffers are added for mixing, this number is incremented. */ + mixing_array.used_entries = 0; + + /* These variables help determine if a rate change is required */ + if (!stat_iteration_counter) { + memset(&stats, 0, sizeof(stats)); + stats.locked_rate = bridge->internal_sample_rate; + } + + /* If the sample rate has changed, update the translator helper */ + if (update_all_rates) { + softmix_translate_helper_change_rate(&trans_helper, softmix_data->internal_rate); + } /* Go through pulling audio from each factory that has it available */ AST_LIST_TRAVERSE(&bridge->channels, bridge_channel, entry) { struct softmix_channel *sc = bridge_channel->bridge_pvt; - int channel_native_rate; - - ast_mutex_lock(&sc->lock); + /* Update the sample rate to match the bridge's native sample rate if necessary. */ if (update_all_rates) { - set_softmix_bridge_data(bridge_data->internal_rate, bridge_channel, 1); + set_softmix_bridge_data(softmix_data->internal_rate, softmix_data->internal_mixing_interval, bridge_channel, 1); } - /* Try to get audio from the factory if available */ - if (ast_slinfactory_available(&sc->factory) >= SOFTMIX_SAMPLES(bridge_data->internal_rate) && - ast_slinfactory_read(&sc->factory, sc->our_buf, SOFTMIX_SAMPLES(bridge_data->internal_rate))) { - short *data1, *data2; - int i; - - /* Put into the local final buffer */ - for (i = 0, data1 = buf, data2 = sc->our_buf; i < SOFTMIX_DATALEN(bridge_data->internal_rate); i++, data1++, data2++) - ast_slinear_saturated_add(data1, data2); - /* Yay we have our own audio */ - sc->have_audio = 1; - } else { - /* Awww we don't have audio ;( */ - sc->have_audio = 0; + /* If stat_iteration_counter is 0, then collect statistics during this mixing interation */ + if (!stat_iteration_counter) { + gather_softmix_stats(&stats, softmix_data, bridge_channel); } - /* Gather stats about channel sample rates. */ - channel_native_rate = MAX(ast_format_rate(&bridge_channel->chan->rawwriteformat), - ast_format_rate(&bridge_channel->chan->rawreadformat)); - - if (channel_native_rate > stats.highest_supported_rate) { - stats.highest_supported_rate = channel_native_rate; - } - if (channel_native_rate > bridge_data->internal_rate) { - for (i = 0; i < ARRAY_LEN(stats.sample_rates); i++) { - if (stats.sample_rates[i] == channel_native_rate) { - stats.num_channels[i]++; - break; - } else if (!stats.sample_rates[i]) { - stats.sample_rates[i] = channel_native_rate; - stats.num_channels[i]++; - break; - } - } - stats.num_above_internal_rate++; - } else if (channel_native_rate == bridge_data->internal_rate) { - stats.num_at_internal_rate++; + /* if the channel is suspended, don't check for audio, but still gather stats */ + if (bridge_channel->suspended) { + continue; } + /* Try to get audio from the factory if available */ + ast_mutex_lock(&sc->lock); + if ((mixing_array.buffers[mixing_array.used_entries] = softmix_process_read_audio(sc, softmix_samples))) { + mixing_array.used_entries++; + } ast_mutex_unlock(&sc->lock); } + /* mix it like crazy */ + memset(buf, 0, softmix_datalen); + for (i = 0; i < mixing_array.used_entries; i++) { + for (x = 0; x < softmix_samples; x++) { + ast_slinear_saturated_add(buf + x, mixing_array.buffers[i] + x); + } + } + /* Next step go through removing the channel's own audio and creating a good frame... */ AST_LIST_TRAVERSE(&bridge->channels, bridge_channel, entry) { struct softmix_channel *sc = bridge_channel->bridge_pvt; - int i = 0; - /* Copy from local final buffer to our final buffer */ - memcpy(sc->final_buf, buf, sizeof(sc->final_buf)); + if (bridge_channel->suspended) { + continue; + } - /* If we provided audio then take it out */ - if (sc->have_audio) { - for (i = 0; i < SOFTMIX_DATALEN(bridge_data->internal_rate); i++) { - ast_slinear_saturated_subtract(&sc->final_buf[i], &sc->our_buf[i]); - } + ast_mutex_lock(&sc->lock); + + /* Make SLINEAR write frame from local buffer */ + if (sc->write_frame.subclass.format.id != cur_slin_id) { + ast_format_set(&sc->write_frame.subclass.format, cur_slin_id, 0); } + sc->write_frame.datalen = softmix_datalen; + sc->write_frame.samples = softmix_samples; + memcpy(sc->final_buf, buf, softmix_datalen); + + /* process the softmix channel's new write audio */ + softmix_process_write_audio(&trans_helper, &bridge_channel->chan->rawwriteformat, sc); /* The frame is now ready for use... */ sc->have_frame = 1; + ast_mutex_unlock(&sc->lock); + /* Poke bridged channel thread just in case */ pthread_kill(bridge_channel->thread, SIGURG); } - /* Re-adjust the internal bridge sample rate if - * 1. two or more channels support a higher sample rate - * 2. no channels support the current sample rate or a higher rate - */ - if (stats.num_above_internal_rate >= 2) { - /* the highest rate is just used as a starting point */ - unsigned int best_rate = stats.highest_supported_rate; - int best_index = -1; - - /* 1. pick the best sample rate two or more channels support - * 2. if two or more channels do not support the same rate, pick the - * lowest sample rate that is still above the internal rate. */ - for (i = 0; ((i < ARRAY_LEN(stats.num_channels)) && stats.num_channels[i]); i++) { - if ((stats.num_channels[i] >= 2 && (best_index == -1)) || - ((best_index != -1) && - (stats.num_channels[i] >= 2) && - (stats.sample_rates[best_index] < stats.sample_rates[i]))) { - - best_rate = stats.sample_rates[i]; - best_index = i; - } else if (best_index == -1) { - best_rate = MIN(best_rate, stats.sample_rates[i]); - } - } - - ast_debug(1, " Bridge changed from %d To %d\n", bridge_data->internal_rate, best_rate); - bridge_data->internal_rate = best_rate; - update_all_rates = 1; - } else if (!stats.num_at_internal_rate && !stats.num_above_internal_rate) { - update_all_rates = 1; - /* in this case, the highest supported rate is actually lower than the internal rate */ - bridge_data->internal_rate = stats.highest_supported_rate; - ast_debug(1, " Bridge changed from %d to %d\n", bridge_data->internal_rate, stats.highest_supported_rate); - update_all_rates = 1; - } else { - update_all_rates = 0; + update_all_rates = 0; + if (!stat_iteration_counter) { + update_all_rates = analyse_softmix_stats(&stats, softmix_data); + stat_iteration_counter = SOFTMIX_STAT_INTERVAL; } + stat_iteration_counter--; ao2_unlock(bridge); - + /* cleanup any translation frame data from the previous mixing iteration. */ + softmix_translate_helper_cleanup(&trans_helper); /* Wait for the timing source to tell us to wake up and get things done */ ast_waitfor_n_fd(&timingfd, 1, &timeout, NULL); - ast_timer_ack(timer, 1); - ao2_lock(bridge); + + /* make sure to detect mixing interval changes if they occur. */ + if (bridge->internal_mixing_interval && (bridge->internal_mixing_interval != softmix_data->internal_mixing_interval)) { + softmix_data->internal_mixing_interval = bridge->internal_mixing_interval; + ast_timer_set_rate(timer, (1000 / softmix_data->internal_mixing_interval)); + update_all_rates = 1; /* if the interval changes, the rates must be adjusted as well just to be notified new interval.*/ + } } - return 0; + res = 0; + +softmix_cleanup: + softmix_translate_helper_destroy(&trans_helper); + softmix_mixing_array_destroy(&mixing_array); + if (softmix_data) { + ao2_ref(softmix_data, -1); + } + return res; } static struct ast_bridge_technology softmix_bridge = { .name = "softmix", - .capabilities = AST_BRIDGE_CAPABILITY_MULTIMIX | AST_BRIDGE_CAPABILITY_THREAD | AST_BRIDGE_CAPABILITY_MULTITHREADED, + .capabilities = AST_BRIDGE_CAPABILITY_MULTIMIX | AST_BRIDGE_CAPABILITY_THREAD | AST_BRIDGE_CAPABILITY_MULTITHREADED | AST_BRIDGE_CAPABILITY_OPTIMIZE, .preference = AST_BRIDGE_PREFERENCE_LOW, .create = softmix_bridge_create, .destroy = softmix_bridge_destroy, @@ -410,11 +822,7 @@ static int load_module(void) if (!(softmix_bridge.format_capabilities = ast_format_cap_alloc())) { return AST_MODULE_LOAD_DECLINE; } -#ifdef SOFTMIX_16_SUPPORT - ast_format_cap_add(softmix_bridge.format_capabilities, ast_format_set(&tmp, AST_FORMAT_SLINEAR16, 0)); -#else ast_format_cap_add(softmix_bridge.format_capabilities, ast_format_set(&tmp, AST_FORMAT_SLINEAR, 0)); -#endif return ast_bridge_technology_register(&softmix_bridge); } -- cgit v1.2.3