2 * Asterisk -- An open source telephony toolkit.
4 * Copyright (C) 2011, Digium, Inc.
6 * Joshua Colp <jcolp@digium.com>
7 * David Vossel <dvossel@digium.com>
9 * See http://www.asterisk.org for more information about
10 * the Asterisk project. Please do not directly contact
11 * any of the maintainers of this project for assistance;
12 * the project provides a web site, mailing lists and IRC
13 * channels for your use.
15 * This program is free software, distributed under the terms of
16 * the GNU General Public License Version 2. See the LICENSE file
17 * at the top of the source tree.
22 * \brief Multi-party software based channel mixing
24 * \author Joshua Colp <jcolp@digium.com>
25 * \author David Vossel <dvossel@digium.com>
32 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
42 #include "asterisk/module.h"
43 #include "asterisk/channel.h"
44 #include "asterisk/bridging.h"
45 #include "asterisk/bridging_technology.h"
46 #include "asterisk/frame.h"
47 #include "asterisk/options.h"
48 #include "asterisk/logger.h"
49 #include "asterisk/slinfactory.h"
50 #include "asterisk/astobj2.h"
51 #include "asterisk/timing.h"
52 #include "asterisk/translate.h"
54 #define MAX_DATALEN 8096
56 /*! \brief Interval at which mixing will take place. Valid options are 10, 20, and 40. */
57 #define DEFAULT_SOFTMIX_INTERVAL 20
59 /*! \brief Size of the buffer used for sample manipulation */
60 #define SOFTMIX_DATALEN(rate, interval) ((rate/50) * (interval / 10))
62 /*! \brief Number of samples we are dealing with */
63 #define SOFTMIX_SAMPLES(rate, interval) (SOFTMIX_DATALEN(rate, interval) / 2)
65 /*! \brief Number of mixing iterations to perform between gathering statistics. */
66 #define SOFTMIX_STAT_INTERVAL 100
68 /* This is the threshold in ms at which a channel's own audio will stop getting
69 * mixed out its own write audio stream because it is not talking. */
70 #define DEFAULT_SOFTMIX_SILENCE_THRESHOLD 2500
71 #define DEFAULT_SOFTMIX_TALKING_THRESHOLD 160
73 #define DEFAULT_ENERGY_HISTORY_LEN 150
75 struct video_follow_talker_data {
76 /*! audio energy history */
77 int energy_history[DEFAULT_ENERGY_HISTORY_LEN];
78 /*! The current slot being used in the history buffer, this
79 * increments and wraps around */
80 int energy_history_cur_slot;
81 /*! The current energy sum used for averages. */
83 /*! The current energy average */
87 /*! \brief Structure which contains per-channel mixing information */
88 struct softmix_channel {
89 /*! Lock to protect this structure */
91 /*! Factory which contains audio read in from the channel */
92 struct ast_slinfactory factory;
93 /*! Frame that contains mixed audio to be written out to the channel */
94 struct ast_frame write_frame;
95 /*! Frame that contains mixed audio read from the channel */
96 struct ast_frame read_frame;
97 /*! DSP for detecting silence */
99 /*! Bit used to indicate if a channel is talking or not. This affects how
100 * the channel's audio is mixed back to it. */
102 /*! Bit used to indicate that the channel provided audio for this mixing interval */
104 /*! Bit used to indicate that a frame is available to be written out to the channel */
106 /*! Buffer containing final mixed audio from all sources */
107 short final_buf[MAX_DATALEN];
108 /*! Buffer containing only the audio from the channel */
109 short our_buf[MAX_DATALEN];
110 /*! Data pertaining to talker mode for video conferencing */
111 struct video_follow_talker_data video_talker;
114 struct softmix_bridge_data {
115 struct ast_timer *timer;
116 unsigned int internal_rate;
117 unsigned int internal_mixing_interval;
120 struct softmix_stats {
121 /*! Each index represents a sample rate used above the internal rate. */
122 unsigned int sample_rates[16];
123 /*! Each index represents the number of channels using the same index in the sample_rates array. */
124 unsigned int num_channels[16];
125 /*! the number of channels above the internal sample rate */
126 unsigned int num_above_internal_rate;
127 /*! the number of channels at the internal sample rate */
128 unsigned int num_at_internal_rate;
129 /*! the absolute highest sample rate supported by any channel in the bridge */
130 unsigned int highest_supported_rate;
131 /*! Is the sample rate locked by the bridge, if so what is that rate.*/
132 unsigned int locked_rate;
135 struct softmix_mixing_array {
141 struct softmix_translate_helper_entry {
142 int num_times_requested; /*!< Once this entry is no longer requested, free the trans_pvt
143 and re-init if it was usable. */
144 struct ast_format dst_format; /*!< The destination format for this helper */
145 struct ast_trans_pvt *trans_pvt; /*!< the translator for this slot. */
146 struct ast_frame *out_frame; /*!< The output frame from the last translation */
147 AST_LIST_ENTRY(softmix_translate_helper_entry) entry;
150 struct softmix_translate_helper {
151 struct ast_format slin_src; /*!< the source format expected for all the translators */
152 AST_LIST_HEAD_NOLOCK(, softmix_translate_helper_entry) entries;
155 static struct softmix_translate_helper_entry *softmix_translate_helper_entry_alloc(struct ast_format *dst)
157 struct softmix_translate_helper_entry *entry;
158 if (!(entry = ast_calloc(1, sizeof(*entry)))) {
161 ast_format_copy(&entry->dst_format, dst);
165 static void *softmix_translate_helper_free_entry(struct softmix_translate_helper_entry *entry)
167 if (entry->trans_pvt) {
168 ast_translator_free_path(entry->trans_pvt);
170 if (entry->out_frame) {
171 ast_frfree(entry->out_frame);
177 static void softmix_translate_helper_init(struct softmix_translate_helper *trans_helper, unsigned int sample_rate)
179 memset(trans_helper, 0, sizeof(*trans_helper));
180 ast_format_set(&trans_helper->slin_src, ast_format_slin_by_rate(sample_rate), 0);
183 static void softmix_translate_helper_destroy(struct softmix_translate_helper *trans_helper)
185 struct softmix_translate_helper_entry *entry;
187 while ((entry = AST_LIST_REMOVE_HEAD(&trans_helper->entries, entry))) {
188 softmix_translate_helper_free_entry(entry);
192 static void softmix_translate_helper_change_rate(struct softmix_translate_helper *trans_helper, unsigned int sample_rate)
194 struct softmix_translate_helper_entry *entry;
196 ast_format_set(&trans_helper->slin_src, ast_format_slin_by_rate(sample_rate), 0);
197 AST_LIST_TRAVERSE_SAFE_BEGIN(&trans_helper->entries, entry, entry) {
198 if (entry->trans_pvt) {
199 ast_translator_free_path(entry->trans_pvt);
200 if (!(entry->trans_pvt = ast_translator_build_path(&entry->dst_format, &trans_helper->slin_src))) {
201 AST_LIST_REMOVE_CURRENT(entry);
202 entry = softmix_translate_helper_free_entry(entry);
206 AST_LIST_TRAVERSE_SAFE_END;
211 * \brief Get the next available audio on the softmix channel's read stream
212 * and determine if it should be mixed out or not on the write stream.
214 * \retval pointer to buffer containing the exact number of samples requested on success.
215 * \retval NULL if no samples are present
217 static int16_t *softmix_process_read_audio(struct softmix_channel *sc, unsigned int num_samples)
219 if ((ast_slinfactory_available(&sc->factory) >= num_samples) &&
220 ast_slinfactory_read(&sc->factory, sc->our_buf, num_samples)) {
230 * \brief Process a softmix channel's write audio
232 * \details This function will remove the channel's talking from its own audio if present and
233 * possibly even do the channel's write translation for it depending on how many other
234 * channels use the same write format.
236 static void softmix_process_write_audio(struct softmix_translate_helper *trans_helper,
237 struct ast_format *raw_write_fmt,
238 struct softmix_channel *sc)
240 struct softmix_translate_helper_entry *entry = NULL;
243 /* If we provided audio that was not determined to be silence,
244 * then take it out while in slinear format. */
245 if (sc->have_audio && sc->talking) {
246 for (i = 0; i < sc->write_frame.samples; i++) {
247 ast_slinear_saturated_subtract(&sc->final_buf[i], &sc->our_buf[i]);
249 /* do not do any special write translate optimization if we had to make
250 * a special mix for them to remove their own audio. */
254 AST_LIST_TRAVERSE(&trans_helper->entries, entry, entry) {
255 if (ast_format_cmp(&entry->dst_format, raw_write_fmt) == AST_FORMAT_CMP_EQUAL) {
256 entry->num_times_requested++;
260 if (!entry->trans_pvt && (entry->num_times_requested > 1)) {
261 entry->trans_pvt = ast_translator_build_path(&entry->dst_format, &trans_helper->slin_src);
263 if (entry->trans_pvt && !entry->out_frame) {
264 entry->out_frame = ast_translate(entry->trans_pvt, &sc->write_frame, 0);
266 if (entry->out_frame && (entry->out_frame->datalen < MAX_DATALEN)) {
267 ast_format_copy(&sc->write_frame.subclass.format, &entry->out_frame->subclass.format);
268 memcpy(sc->final_buf, entry->out_frame->data.ptr, entry->out_frame->datalen);
269 sc->write_frame.datalen = entry->out_frame->datalen;
270 sc->write_frame.samples = entry->out_frame->samples;
275 /* add new entry into list if this format destination was not matched. */
276 if (!entry && (entry = softmix_translate_helper_entry_alloc(raw_write_fmt))) {
277 AST_LIST_INSERT_HEAD(&trans_helper->entries, entry, entry);
281 static void softmix_translate_helper_cleanup(struct softmix_translate_helper *trans_helper)
283 struct softmix_translate_helper_entry *entry = NULL;
284 AST_LIST_TRAVERSE(&trans_helper->entries, entry, entry) {
285 if (entry->out_frame) {
286 ast_frfree(entry->out_frame);
287 entry->out_frame = NULL;
289 entry->num_times_requested = 0;
293 static void softmix_bridge_data_destroy(void *obj)
295 struct softmix_bridge_data *softmix_data = obj;
296 ast_timer_close(softmix_data->timer);
299 /*! \brief Function called when a bridge is created */
300 static int softmix_bridge_create(struct ast_bridge *bridge)
302 struct softmix_bridge_data *softmix_data;
304 if (!(softmix_data = ao2_alloc(sizeof(*softmix_data), softmix_bridge_data_destroy))) {
307 if (!(softmix_data->timer = ast_timer_open())) {
308 ao2_ref(softmix_data, -1);
312 /* start at 8khz, let it grow from there */
313 softmix_data->internal_rate = 8000;
314 softmix_data->internal_mixing_interval = DEFAULT_SOFTMIX_INTERVAL;
316 bridge->bridge_pvt = softmix_data;
320 /*! \brief Function called when a bridge is destroyed */
321 static int softmix_bridge_destroy(struct ast_bridge *bridge)
323 struct softmix_bridge_data *softmix_data = bridge->bridge_pvt;
324 if (!bridge->bridge_pvt) {
327 ao2_ref(softmix_data, -1);
328 bridge->bridge_pvt = NULL;
332 static void set_softmix_bridge_data(int rate, int interval, struct ast_bridge_channel *bridge_channel, int reset)
334 struct softmix_channel *sc = bridge_channel->bridge_pvt;
335 unsigned int channel_read_rate = ast_format_rate(&bridge_channel->chan->rawreadformat);
337 ast_mutex_lock(&sc->lock);
339 ast_slinfactory_destroy(&sc->factory);
340 ast_dsp_free(sc->dsp);
342 /* Setup read/write frame parameters */
343 sc->write_frame.frametype = AST_FRAME_VOICE;
344 ast_format_set(&sc->write_frame.subclass.format, ast_format_slin_by_rate(rate), 0);
345 sc->write_frame.data.ptr = sc->final_buf;
346 sc->write_frame.datalen = SOFTMIX_DATALEN(rate, interval);
347 sc->write_frame.samples = SOFTMIX_SAMPLES(rate, interval);
349 sc->read_frame.frametype = AST_FRAME_VOICE;
350 ast_format_set(&sc->read_frame.subclass.format, ast_format_slin_by_rate(channel_read_rate), 0);
351 sc->read_frame.data.ptr = sc->our_buf;
352 sc->read_frame.datalen = SOFTMIX_DATALEN(channel_read_rate, interval);
353 sc->read_frame.samples = SOFTMIX_SAMPLES(channel_read_rate, interval);
356 ast_slinfactory_init_with_format(&sc->factory, &sc->write_frame.subclass.format);
358 /* set new read and write formats on channel. */
359 ast_set_read_format(bridge_channel->chan, &sc->read_frame.subclass.format);
360 ast_set_write_format(bridge_channel->chan, &sc->write_frame.subclass.format);
362 /* set up new DSP. This is on the read side only right before the read frame enters the smoother. */
363 sc->dsp = ast_dsp_new_with_rate(channel_read_rate);
364 /* we want to aggressively detect silence to avoid feedback */
365 if (bridge_channel->tech_args.talking_threshold) {
366 ast_dsp_set_threshold(sc->dsp, bridge_channel->tech_args.talking_threshold);
368 ast_dsp_set_threshold(sc->dsp, DEFAULT_SOFTMIX_TALKING_THRESHOLD);
371 ast_mutex_unlock(&sc->lock);
374 /*! \brief Function called when a channel is joined into the bridge */
375 static int softmix_bridge_join(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
377 struct softmix_channel *sc = NULL;
378 struct softmix_bridge_data *softmix_data = bridge->bridge_pvt;
380 /* Create a new softmix_channel structure and allocate various things on it */
381 if (!(sc = ast_calloc(1, sizeof(*sc)))) {
385 /* Can't forget the lock */
386 ast_mutex_init(&sc->lock);
388 /* Can't forget to record our pvt structure within the bridged channel structure */
389 bridge_channel->bridge_pvt = sc;
391 set_softmix_bridge_data(softmix_data->internal_rate,
392 softmix_data->internal_mixing_interval ? softmix_data->internal_mixing_interval : DEFAULT_SOFTMIX_INTERVAL,
398 /*! \brief Function called when a channel leaves the bridge */
399 static int softmix_bridge_leave(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
401 struct softmix_channel *sc = bridge_channel->bridge_pvt;
403 if (!(bridge_channel->bridge_pvt)) {
406 bridge_channel->bridge_pvt = NULL;
408 /* Drop mutex lock */
409 ast_mutex_destroy(&sc->lock);
411 /* Drop the factory */
412 ast_slinfactory_destroy(&sc->factory);
415 ast_dsp_free(sc->dsp);
417 /* Eep! drop ourselves */
425 * \brief If the bridging core passes DTMF to us, then they want it to be distributed out to all memebers. Do that here.
427 static void softmix_pass_dtmf(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame)
429 struct ast_bridge_channel *tmp;
430 AST_LIST_TRAVERSE(&bridge->channels, tmp, entry) {
431 if (tmp == bridge_channel) {
434 ast_write(tmp->chan, frame);
438 static void softmix_pass_video(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame)
440 struct ast_bridge_channel *tmp;
441 AST_LIST_TRAVERSE(&bridge->channels, tmp, entry) {
442 if (tmp->suspended) {
445 ast_write(tmp->chan, frame);
449 /*! \brief Function called when a channel writes a frame into the bridge */
450 static enum ast_bridge_write_result softmix_bridge_write(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame)
452 struct softmix_channel *sc = bridge_channel->bridge_pvt;
453 struct softmix_bridge_data *softmix_data = bridge->bridge_pvt;
454 int totalsilence = 0;
456 int silence_threshold = bridge_channel->tech_args.silence_threshold ?
457 bridge_channel->tech_args.silence_threshold :
458 DEFAULT_SOFTMIX_SILENCE_THRESHOLD;
459 char update_talking = -1; /* if this is set to 0 or 1, tell the bridge that the channel has started or stopped talking. */
460 int res = AST_BRIDGE_WRITE_SUCCESS;
462 /* Only accept audio frames, all others are unsupported */
463 if (frame->frametype == AST_FRAME_DTMF_END || frame->frametype == AST_FRAME_DTMF_BEGIN) {
464 softmix_pass_dtmf(bridge, bridge_channel, frame);
465 goto bridge_write_cleanup;
466 } else if (frame->frametype != AST_FRAME_VOICE && frame->frametype != AST_FRAME_VIDEO) {
467 res = AST_BRIDGE_WRITE_UNSUPPORTED;
468 goto bridge_write_cleanup;
469 } else if (frame->datalen == 0) {
470 goto bridge_write_cleanup;
473 /* Determine if this video frame should be distributed or not */
474 if (frame->frametype == AST_FRAME_VIDEO) {
475 switch (bridge->video_mode.mode) {
476 case AST_BRIDGE_VIDEO_MODE_NONE:
478 case AST_BRIDGE_VIDEO_MODE_SINGLE_SRC:
479 if (ast_bridge_is_video_src(bridge, bridge_channel->chan)) {
480 softmix_pass_video(bridge, bridge_channel, frame);
483 case AST_BRIDGE_VIDEO_MODE_TALKER_SRC:
484 ast_mutex_lock(&sc->lock);
485 ast_bridge_update_talker_src_video_mode(bridge, bridge_channel->chan, sc->video_talker.energy_average, ast_format_get_video_mark(&frame->subclass.format));
486 ast_mutex_unlock(&sc->lock);
487 if (ast_bridge_is_video_src(bridge, bridge_channel->chan)) {
488 softmix_pass_video(bridge, bridge_channel, frame);
492 goto bridge_write_cleanup;
495 /* If we made it here, we are going to write the frame into the conference */
496 ast_mutex_lock(&sc->lock);
497 ast_dsp_silence_with_energy(sc->dsp, frame, &totalsilence, &cur_energy);
499 if (bridge->video_mode.mode == AST_BRIDGE_VIDEO_MODE_TALKER_SRC) {
500 int cur_slot = sc->video_talker.energy_history_cur_slot;
501 sc->video_talker.energy_accum -= sc->video_talker.energy_history[cur_slot];
502 sc->video_talker.energy_accum += cur_energy;
503 sc->video_talker.energy_history[cur_slot] = cur_energy;
504 sc->video_talker.energy_average = sc->video_talker.energy_accum / DEFAULT_ENERGY_HISTORY_LEN;
505 sc->video_talker.energy_history_cur_slot++;
506 if (sc->video_talker.energy_history_cur_slot == DEFAULT_ENERGY_HISTORY_LEN) {
507 sc->video_talker.energy_history_cur_slot = 0; /* wrap around */
511 if (totalsilence < silence_threshold) {
515 sc->talking = 1; /* tell the write process we have audio to be mixed out */
523 /* Before adding audio in, make sure we haven't fallen behind. If audio has fallen
524 * behind 4 times the amount of samples mixed on every iteration of the mixer, Re-sync
525 * the audio by flushing the buffer before adding new audio in. */
526 if (ast_slinfactory_available(&sc->factory) > (4 * SOFTMIX_SAMPLES(softmix_data->internal_rate, softmix_data->internal_mixing_interval))) {
527 ast_slinfactory_flush(&sc->factory);
530 /* If a frame was provided add it to the smoother, unless drop silence is enabled and this frame
531 * is not determined to be talking. */
532 if (!(bridge_channel->tech_args.drop_silence && !sc->talking) &&
533 (frame->frametype == AST_FRAME_VOICE && ast_format_is_slinear(&frame->subclass.format))) {
534 ast_slinfactory_feed(&sc->factory, frame);
537 /* If a frame is ready to be written out, do so */
538 if (sc->have_frame) {
539 ast_write(bridge_channel->chan, &sc->write_frame);
544 ast_mutex_unlock(&sc->lock);
546 if (update_talking != -1) {
547 ast_bridge_notify_talking(bridge, bridge_channel, update_talking);
552 bridge_write_cleanup:
553 /* Even though the frame is not being written into the conference because it is not audio,
554 * we should use this opportunity to check to see if a frame is ready to be written out from
555 * the conference to the channel. */
556 ast_mutex_lock(&sc->lock);
557 if (sc->have_frame) {
558 ast_write(bridge_channel->chan, &sc->write_frame);
561 ast_mutex_unlock(&sc->lock);
566 /*! \brief Function called when the channel's thread is poked */
567 static int softmix_bridge_poke(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
569 struct softmix_channel *sc = bridge_channel->bridge_pvt;
571 ast_mutex_lock(&sc->lock);
573 if (sc->have_frame) {
574 ast_write(bridge_channel->chan, &sc->write_frame);
578 ast_mutex_unlock(&sc->lock);
583 static void gather_softmix_stats(struct softmix_stats *stats,
584 const struct softmix_bridge_data *softmix_data,
585 struct ast_bridge_channel *bridge_channel)
587 int channel_native_rate;
589 /* Gather stats about channel sample rates. */
590 channel_native_rate = MAX(ast_format_rate(&bridge_channel->chan->rawwriteformat),
591 ast_format_rate(&bridge_channel->chan->rawreadformat));
593 if (channel_native_rate > stats->highest_supported_rate) {
594 stats->highest_supported_rate = channel_native_rate;
596 if (channel_native_rate > softmix_data->internal_rate) {
597 for (i = 0; i < ARRAY_LEN(stats->sample_rates); i++) {
598 if (stats->sample_rates[i] == channel_native_rate) {
599 stats->num_channels[i]++;
601 } else if (!stats->sample_rates[i]) {
602 stats->sample_rates[i] = channel_native_rate;
603 stats->num_channels[i]++;
607 stats->num_above_internal_rate++;
608 } else if (channel_native_rate == softmix_data->internal_rate) {
609 stats->num_at_internal_rate++;
614 * \brief Analyse mixing statistics and change bridges internal rate
617 * \retval 0, no changes to internal rate
618 * \ratval 1, internal rate was changed, update all the channels on the next mixing iteration.
620 static unsigned int analyse_softmix_stats(struct softmix_stats *stats, struct softmix_bridge_data *softmix_data)
623 /* Re-adjust the internal bridge sample rate if
624 * 1. The bridge's internal sample rate is locked in at a sample
625 * rate other than the current sample rate being used.
626 * 2. two or more channels support a higher sample rate
627 * 3. no channels support the current sample rate or a higher rate
629 if (stats->locked_rate) {
630 /* if the rate is locked by the bridge, only update it if it differs
631 * from the current rate we are using. */
632 if (softmix_data->internal_rate != stats->locked_rate) {
633 softmix_data->internal_rate = stats->locked_rate;
634 ast_debug(1, " Bridge is locked in at sample rate %d\n", softmix_data->internal_rate);
637 } else if (stats->num_above_internal_rate >= 2) {
638 /* the highest rate is just used as a starting point */
639 unsigned int best_rate = stats->highest_supported_rate;
642 for (i = 0; i < ARRAY_LEN(stats->num_channels); i++) {
643 if (stats->num_channels[i]) {
646 /* best_rate starts out being the first sample rate
647 * greater than the internal sample rate that 2 or
648 * more channels support. */
649 if (stats->num_channels[i] >= 2 && (best_index == -1)) {
650 best_rate = stats->sample_rates[i];
652 /* If it has been detected that multiple rates above
653 * the internal rate are present, compare those rates
654 * to each other and pick the highest one two or more
655 * channels support. */
656 } else if (((best_index != -1) &&
657 (stats->num_channels[i] >= 2) &&
658 (stats->sample_rates[best_index] < stats->sample_rates[i]))) {
659 best_rate = stats->sample_rates[i];
661 /* It is possible that multiple channels exist with native sample
662 * rates above the internal sample rate, but none of those channels
663 * have the same rate in common. In this case, the lowest sample
664 * rate among those channels is picked. Over time as additional
665 * statistic runs are made the internal sample rate number will
666 * adjust to the most optimal sample rate, but it may take multiple
668 } else if (best_index == -1) {
669 best_rate = MIN(best_rate, stats->sample_rates[i]);
673 ast_debug(1, " Bridge changed from %d To %d\n", softmix_data->internal_rate, best_rate);
674 softmix_data->internal_rate = best_rate;
676 } else if (!stats->num_at_internal_rate && !stats->num_above_internal_rate) {
677 /* In this case, the highest supported rate is actually lower than the internal rate */
678 softmix_data->internal_rate = stats->highest_supported_rate;
679 ast_debug(1, " Bridge changed from %d to %d\n", softmix_data->internal_rate, stats->highest_supported_rate);
685 static int softmix_mixing_array_init(struct softmix_mixing_array *mixing_array, unsigned int starting_num_entries)
687 memset(mixing_array, 0, sizeof(*mixing_array));
688 mixing_array->max_num_entries = starting_num_entries;
689 if (!(mixing_array->buffers = ast_calloc(mixing_array->max_num_entries, sizeof(int16_t *)))) {
690 ast_log(LOG_NOTICE, "Failed to allocate softmix mixing structure. \n");
696 static void softmix_mixing_array_destroy(struct softmix_mixing_array *mixing_array)
698 ast_free(mixing_array->buffers);
701 static int softmix_mixing_array_grow(struct softmix_mixing_array *mixing_array, unsigned int num_entries)
704 /* give it some room to grow since memory is cheap but allocations can be expensive */
705 mixing_array->max_num_entries = num_entries;
706 if (!(tmp = ast_realloc(mixing_array->buffers, (mixing_array->max_num_entries * sizeof(int16_t *))))) {
707 ast_log(LOG_NOTICE, "Failed to re-allocate softmix mixing structure. \n");
710 mixing_array->buffers = tmp;
714 /*! \brief Function which acts as the mixing thread */
715 static int softmix_bridge_thread(struct ast_bridge *bridge)
717 struct softmix_stats stats = { { 0 }, };
718 struct softmix_mixing_array mixing_array;
719 struct softmix_bridge_data *softmix_data = bridge->bridge_pvt;
720 struct ast_timer *timer;
721 struct softmix_translate_helper trans_helper;
722 int16_t buf[MAX_DATALEN] = { 0, };
723 unsigned int stat_iteration_counter = 0; /* counts down, gather stats at zero and reset. */
725 int update_all_rates = 0; /* set this when the internal sample rate has changed */
729 if (!(softmix_data = bridge->bridge_pvt)) {
730 goto softmix_cleanup;
733 ao2_ref(softmix_data, 1);
734 timer = softmix_data->timer;
735 timingfd = ast_timer_fd(timer);
736 softmix_translate_helper_init(&trans_helper, softmix_data->internal_rate);
737 ast_timer_set_rate(timer, (1000 / softmix_data->internal_mixing_interval));
739 /* Give the mixing array room to grow, memory is cheap but allocations are expensive. */
740 if (softmix_mixing_array_init(&mixing_array, bridge->num + 10)) {
741 ast_log(LOG_NOTICE, "Failed to allocate softmix mixing structure. \n");
742 goto softmix_cleanup;
745 while (!bridge->stop && !bridge->refresh && bridge->array_num) {
746 struct ast_bridge_channel *bridge_channel = NULL;
748 enum ast_format_id cur_slin_id = ast_format_slin_by_rate(softmix_data->internal_rate);
749 unsigned int softmix_samples = SOFTMIX_SAMPLES(softmix_data->internal_rate, softmix_data->internal_mixing_interval);
750 unsigned int softmix_datalen = SOFTMIX_DATALEN(softmix_data->internal_rate, softmix_data->internal_mixing_interval);
752 if (softmix_datalen > MAX_DATALEN) {
753 /* This should NEVER happen, but if it does we need to know about it. Almost
754 * all the memcpys used during this process depend on this assumption. Rather
755 * than checking this over and over again through out the code, this single
756 * verification is done on each iteration. */
757 ast_log(LOG_WARNING, "Conference mixing error, requested mixing length greater than mixing buffer.\n");
758 goto softmix_cleanup;
761 /* Grow the mixing array buffer as participants are added. */
762 if (mixing_array.max_num_entries < bridge->num && softmix_mixing_array_grow(&mixing_array, bridge->num + 5)) {
763 goto softmix_cleanup;
766 /* init the number of buffers stored in the mixing array to 0.
767 * As buffers are added for mixing, this number is incremented. */
768 mixing_array.used_entries = 0;
770 /* These variables help determine if a rate change is required */
771 if (!stat_iteration_counter) {
772 memset(&stats, 0, sizeof(stats));
773 stats.locked_rate = bridge->internal_sample_rate;
776 /* If the sample rate has changed, update the translator helper */
777 if (update_all_rates) {
778 softmix_translate_helper_change_rate(&trans_helper, softmix_data->internal_rate);
781 /* Go through pulling audio from each factory that has it available */
782 AST_LIST_TRAVERSE(&bridge->channels, bridge_channel, entry) {
783 struct softmix_channel *sc = bridge_channel->bridge_pvt;
785 /* Update the sample rate to match the bridge's native sample rate if necessary. */
786 if (update_all_rates) {
787 set_softmix_bridge_data(softmix_data->internal_rate, softmix_data->internal_mixing_interval, bridge_channel, 1);
790 /* If stat_iteration_counter is 0, then collect statistics during this mixing interation */
791 if (!stat_iteration_counter) {
792 gather_softmix_stats(&stats, softmix_data, bridge_channel);
795 /* if the channel is suspended, don't check for audio, but still gather stats */
796 if (bridge_channel->suspended) {
800 /* Try to get audio from the factory if available */
801 ast_mutex_lock(&sc->lock);
802 if ((mixing_array.buffers[mixing_array.used_entries] = softmix_process_read_audio(sc, softmix_samples))) {
803 mixing_array.used_entries++;
805 ast_mutex_unlock(&sc->lock);
808 /* mix it like crazy */
809 memset(buf, 0, softmix_datalen);
810 for (i = 0; i < mixing_array.used_entries; i++) {
811 for (x = 0; x < softmix_samples; x++) {
812 ast_slinear_saturated_add(buf + x, mixing_array.buffers[i] + x);
816 /* Next step go through removing the channel's own audio and creating a good frame... */
817 AST_LIST_TRAVERSE(&bridge->channels, bridge_channel, entry) {
818 struct softmix_channel *sc = bridge_channel->bridge_pvt;
820 if (bridge_channel->suspended) {
824 ast_mutex_lock(&sc->lock);
826 /* Make SLINEAR write frame from local buffer */
827 if (sc->write_frame.subclass.format.id != cur_slin_id) {
828 ast_format_set(&sc->write_frame.subclass.format, cur_slin_id, 0);
830 sc->write_frame.datalen = softmix_datalen;
831 sc->write_frame.samples = softmix_samples;
832 memcpy(sc->final_buf, buf, softmix_datalen);
834 /* process the softmix channel's new write audio */
835 softmix_process_write_audio(&trans_helper, &bridge_channel->chan->rawwriteformat, sc);
837 /* The frame is now ready for use... */
840 ast_mutex_unlock(&sc->lock);
842 /* Poke bridged channel thread just in case */
843 pthread_kill(bridge_channel->thread, SIGURG);
846 update_all_rates = 0;
847 if (!stat_iteration_counter) {
848 update_all_rates = analyse_softmix_stats(&stats, softmix_data);
849 stat_iteration_counter = SOFTMIX_STAT_INTERVAL;
851 stat_iteration_counter--;
854 /* cleanup any translation frame data from the previous mixing iteration. */
855 softmix_translate_helper_cleanup(&trans_helper);
856 /* Wait for the timing source to tell us to wake up and get things done */
857 ast_waitfor_n_fd(&timingfd, 1, &timeout, NULL);
858 ast_timer_ack(timer, 1);
861 /* make sure to detect mixing interval changes if they occur. */
862 if (bridge->internal_mixing_interval && (bridge->internal_mixing_interval != softmix_data->internal_mixing_interval)) {
863 softmix_data->internal_mixing_interval = bridge->internal_mixing_interval;
864 ast_timer_set_rate(timer, (1000 / softmix_data->internal_mixing_interval));
865 update_all_rates = 1; /* if the interval changes, the rates must be adjusted as well just to be notified new interval.*/
872 softmix_translate_helper_destroy(&trans_helper);
873 softmix_mixing_array_destroy(&mixing_array);
875 ao2_ref(softmix_data, -1);
880 static struct ast_bridge_technology softmix_bridge = {
882 .capabilities = AST_BRIDGE_CAPABILITY_MULTIMIX | AST_BRIDGE_CAPABILITY_THREAD | AST_BRIDGE_CAPABILITY_MULTITHREADED | AST_BRIDGE_CAPABILITY_OPTIMIZE | AST_BRIDGE_CAPABILITY_VIDEO,
883 .preference = AST_BRIDGE_PREFERENCE_LOW,
884 .create = softmix_bridge_create,
885 .destroy = softmix_bridge_destroy,
886 .join = softmix_bridge_join,
887 .leave = softmix_bridge_leave,
888 .write = softmix_bridge_write,
889 .thread = softmix_bridge_thread,
890 .poke = softmix_bridge_poke,
893 static int unload_module(void)
895 ast_format_cap_destroy(softmix_bridge.format_capabilities);
896 return ast_bridge_technology_unregister(&softmix_bridge);
899 static int load_module(void)
901 struct ast_format tmp;
902 if (!(softmix_bridge.format_capabilities = ast_format_cap_alloc())) {
903 return AST_MODULE_LOAD_DECLINE;
905 ast_format_cap_add(softmix_bridge.format_capabilities, ast_format_set(&tmp, AST_FORMAT_SLINEAR, 0));
906 return ast_bridge_technology_register(&softmix_bridge);
909 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Multi-party software based channel mixing");