bridges/bridge_softmix.c

   1 /*
   2  * Asterisk -- An open source telephony toolkit.
   3  *
   4  * Copyright (C) 2011, Digium, Inc.
   5  *
   6  * Joshua Colp <jcolp@digium.com>
   7  * David Vossel <dvossel@digium.com>
   8  *
   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.
  14  *
  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.
  18  */
  19
  20 /*! \file
  21  *
  22  * \brief Multi-party software based channel mixing
  23  *
  24  * \author Joshua Colp <jcolp@digium.com>
  25  * \author David Vossel <dvossel@digium.com>
  26  *
  27  * \ingroup bridges
  28  */
  29
  30 /*** MODULEINFO
  31         <support_level>core</support_level>
  32  ***/
  33
  34 #include "asterisk.h"
  35
  36 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
  37
  38 #include <stdio.h>
  39 #include <stdlib.h>
  40 #include <string.h>
  41 #include <sys/time.h>
  42 #include <signal.h>
  43 #include <errno.h>
  44 #include <unistd.h>
  45
  46 #include "asterisk/module.h"
  47 #include "asterisk/channel.h"
  48 #include "asterisk/bridging.h"
  49 #include "asterisk/bridging_technology.h"
  50 #include "asterisk/frame.h"
  51 #include "asterisk/options.h"
  52 #include "asterisk/logger.h"
  53 #include "asterisk/slinfactory.h"
  54 #include "asterisk/astobj2.h"
  55 #include "asterisk/timing.h"
  56 #include "asterisk/translate.h"
  57
  58 #define MAX_DATALEN 8096
  59
  60 /*! \brief Interval at which mixing will take place. Valid options are 10, 20, and 40. */
  61 #define DEFAULT_SOFTMIX_INTERVAL 20
  62
  63 /*! \brief Size of the buffer used for sample manipulation */
  64 #define SOFTMIX_DATALEN(rate, interval) ((rate/50) * (interval / 10))
  65
  66 /*! \brief Number of samples we are dealing with */
  67 #define SOFTMIX_SAMPLES(rate, interval) (SOFTMIX_DATALEN(rate, interval) / 2)
  68
  69 /*! \brief Number of mixing iterations to perform between gathering statistics. */
  70 #define SOFTMIX_STAT_INTERVAL 100
  71
  72 /* This is the threshold in ms at which a channel's own audio will stop getting
  73  * mixed out its own write audio stream because it is not talking. */
  74 #define DEFAULT_SOFTMIX_SILENCE_THRESHOLD 2500
  75 #define DEFAULT_SOFTMIX_TALKING_THRESHOLD 160
  76
  77 #define DEFAULT_ENERGY_HISTORY_LEN 150
  78
  79 struct video_follow_talker_data {
  80         /*! audio energy history */
  81         int energy_history[DEFAULT_ENERGY_HISTORY_LEN];
  82         /*! The current slot being used in the history buffer, this
  83          *  increments and wraps around */
  84         int energy_history_cur_slot;
  85         /*! The current energy sum used for averages. */
  86         int energy_accum;
  87         /*! The current energy average */
  88         int energy_average;
  89 };
  90
  91 /*! \brief Structure which contains per-channel mixing information */
  92 struct softmix_channel {
  93         /*! Lock to protect this structure */
  94         ast_mutex_t lock;
  95         /*! Factory which contains audio read in from the channel */
  96         struct ast_slinfactory factory;
  97         /*! Frame that contains mixed audio to be written out to the channel */
  98         struct ast_frame write_frame;
  99         /*! Frame that contains mixed audio read from the channel */
 100         struct ast_frame read_frame;
 101         /*! DSP for detecting silence */
 102         struct ast_dsp *dsp;
 103         /*! Bit used to indicate if a channel is talking or not. This affects how
 104          * the channel's audio is mixed back to it. */
 105         int talking:1;
 106         /*! Bit used to indicate that the channel provided audio for this mixing interval */
 107         int have_audio:1;
 108         /*! Bit used to indicate that a frame is available to be written out to the channel */
 109         int have_frame:1;
 110         /*! Buffer containing final mixed audio from all sources */
 111         short final_buf[MAX_DATALEN];
 112         /*! Buffer containing only the audio from the channel */
 113         short our_buf[MAX_DATALEN];
 114         /*! Data pertaining to talker mode for video conferencing */
 115         struct video_follow_talker_data video_talker;
 116 };
 117
 118 struct softmix_bridge_data {
 119         struct ast_timer *timer;
 120         unsigned int internal_rate;
 121         unsigned int internal_mixing_interval;
 122 };
 123
 124 struct softmix_stats {
 125                 /*! Each index represents a sample rate used above the internal rate. */
 126                 unsigned int sample_rates[16];
 127                 /*! Each index represents the number of channels using the same index in the sample_rates array.  */
 128                 unsigned int num_channels[16];
 129                 /*! the number of channels above the internal sample rate */
 130                 unsigned int num_above_internal_rate;
 131                 /*! the number of channels at the internal sample rate */
 132                 unsigned int num_at_internal_rate;
 133                 /*! the absolute highest sample rate supported by any channel in the bridge */
 134                 unsigned int highest_supported_rate;
 135                 /*! Is the sample rate locked by the bridge, if so what is that rate.*/
 136                 unsigned int locked_rate;
 137 };
 138
 139 struct softmix_mixing_array {
 140         int max_num_entries;
 141         int used_entries;
 142         int16_t **buffers;
 143 };
 144
 145 struct softmix_translate_helper_entry {
 146         int num_times_requested; /*!< Once this entry is no longer requested, free the trans_pvt
 147                                       and re-init if it was usable. */
 148         struct ast_format dst_format; /*!< The destination format for this helper */
 149         struct ast_trans_pvt *trans_pvt; /*!< the translator for this slot. */
 150         struct ast_frame *out_frame; /*!< The output frame from the last translation */
 151         AST_LIST_ENTRY(softmix_translate_helper_entry) entry;
 152 };
 153
 154 struct softmix_translate_helper {
 155         struct ast_format slin_src; /*!< the source format expected for all the translators */
 156         AST_LIST_HEAD_NOLOCK(, softmix_translate_helper_entry) entries;
 157 };
 158
 159 static struct softmix_translate_helper_entry *softmix_translate_helper_entry_alloc(struct ast_format *dst)
 160 {
 161         struct softmix_translate_helper_entry *entry;
 162         if (!(entry = ast_calloc(1, sizeof(*entry)))) {
 163                 return NULL;
 164         }
 165         ast_format_copy(&entry->dst_format, dst);
 166         return entry;
 167 }
 168
 169 static void *softmix_translate_helper_free_entry(struct softmix_translate_helper_entry *entry)
 170 {
 171         if (entry->trans_pvt) {
 172                 ast_translator_free_path(entry->trans_pvt);
 173         }
 174         if (entry->out_frame) {
 175                 ast_frfree(entry->out_frame);
 176         }
 177         ast_free(entry);
 178         return NULL;
 179 }
 180
 181 static void softmix_translate_helper_init(struct softmix_translate_helper *trans_helper, unsigned int sample_rate)
 182 {
 183         memset(trans_helper, 0, sizeof(*trans_helper));
 184         ast_format_set(&trans_helper->slin_src, ast_format_slin_by_rate(sample_rate), 0);
 185 }
 186
 187 static void softmix_translate_helper_destroy(struct softmix_translate_helper *trans_helper)
 188 {
 189         struct softmix_translate_helper_entry *entry;
 190
 191         while ((entry = AST_LIST_REMOVE_HEAD(&trans_helper->entries, entry))) {
 192                 softmix_translate_helper_free_entry(entry);
 193         }
 194 }
 195
 196 static void softmix_translate_helper_change_rate(struct softmix_translate_helper *trans_helper, unsigned int sample_rate)
 197 {
 198         struct softmix_translate_helper_entry *entry;
 199
 200         ast_format_set(&trans_helper->slin_src, ast_format_slin_by_rate(sample_rate), 0);
 201         AST_LIST_TRAVERSE_SAFE_BEGIN(&trans_helper->entries, entry, entry) {
 202                 if (entry->trans_pvt) {
 203                         ast_translator_free_path(entry->trans_pvt);
 204                         if (!(entry->trans_pvt = ast_translator_build_path(&entry->dst_format, &trans_helper->slin_src))) {
 205                                 AST_LIST_REMOVE_CURRENT(entry);
 206                                 entry = softmix_translate_helper_free_entry(entry);
 207                         }
 208                 }
 209         }
 210         AST_LIST_TRAVERSE_SAFE_END;
 211 }
 212
 213 /*!
 214  * \internal
 215  * \brief Get the next available audio on the softmix channel's read stream
 216  * and determine if it should be mixed out or not on the write stream.
 217  *
 218  * \retval pointer to buffer containing the exact number of samples requested on success.
 219  * \retval NULL if no samples are present
 220  */
 221 static int16_t *softmix_process_read_audio(struct softmix_channel *sc, unsigned int num_samples)
 222 {
 223         if ((ast_slinfactory_available(&sc->factory) >= num_samples) &&
 224                 ast_slinfactory_read(&sc->factory, sc->our_buf, num_samples)) {
 225                 sc->have_audio = 1;
 226                 return sc->our_buf;
 227         }
 228         sc->have_audio = 0;
 229         return NULL;
 230 }
 231
 232 /*!
 233  * \internal
 234  * \brief Process a softmix channel's write audio
 235  *
 236  * \details This function will remove the channel's talking from its own audio if present and
 237  * possibly even do the channel's write translation for it depending on how many other
 238  * channels use the same write format.
 239  */
 240 static void softmix_process_write_audio(struct softmix_translate_helper *trans_helper,
 241         struct ast_format *raw_write_fmt,
 242         struct softmix_channel *sc)
 243 {
 244         struct softmix_translate_helper_entry *entry = NULL;
 245         int i;
 246
 247         /* If we provided audio that was not determined to be silence,
 248          * then take it out while in slinear format. */
 249         if (sc->have_audio && sc->talking) {
 250                 for (i = 0; i < sc->write_frame.samples; i++) {
 251                         ast_slinear_saturated_subtract(&sc->final_buf[i], &sc->our_buf[i]);
 252                 }
 253                 /* do not do any special write translate optimization if we had to make
 254                  * a special mix for them to remove their own audio. */
 255                 return;
 256         }
 257
 258         AST_LIST_TRAVERSE(&trans_helper->entries, entry, entry) {
 259                 if (ast_format_cmp(&entry->dst_format, raw_write_fmt) == AST_FORMAT_CMP_EQUAL) {
 260                         entry->num_times_requested++;
 261                 } else {
 262                         continue;
 263                 }
 264                 if (!entry->trans_pvt && (entry->num_times_requested > 1)) {
 265                         entry->trans_pvt = ast_translator_build_path(&entry->dst_format, &trans_helper->slin_src);
 266                 }
 267                 if (entry->trans_pvt && !entry->out_frame) {
 268                         entry->out_frame = ast_translate(entry->trans_pvt, &sc->write_frame, 0);
 269                 }
 270                 if (entry->out_frame && (entry->out_frame->datalen < MAX_DATALEN)) {
 271                         ast_format_copy(&sc->write_frame.subclass.format, &entry->out_frame->subclass.format);
 272                         memcpy(sc->final_buf, entry->out_frame->data.ptr, entry->out_frame->datalen);
 273                         sc->write_frame.datalen = entry->out_frame->datalen;
 274                         sc->write_frame.samples = entry->out_frame->samples;
 275                 }
 276                 break;
 277         }
 278
 279         /* add new entry into list if this format destination was not matched. */
 280         if (!entry && (entry = softmix_translate_helper_entry_alloc(raw_write_fmt))) {
 281                 AST_LIST_INSERT_HEAD(&trans_helper->entries, entry, entry);
 282         }
 283 }
 284
 285 static void softmix_translate_helper_cleanup(struct softmix_translate_helper *trans_helper)
 286 {
 287         struct softmix_translate_helper_entry *entry = NULL;
 288         AST_LIST_TRAVERSE(&trans_helper->entries, entry, entry) {
 289                 if (entry->out_frame) {
 290                         ast_frfree(entry->out_frame);
 291                         entry->out_frame = NULL;
 292                 }
 293                 entry->num_times_requested = 0;
 294         }
 295 }
 296
 297 static void softmix_bridge_data_destroy(void *obj)
 298 {
 299         struct softmix_bridge_data *softmix_data = obj;
 300         ast_timer_close(softmix_data->timer);
 301 }
 302
 303 /*! \brief Function called when a bridge is created */
 304 static int softmix_bridge_create(struct ast_bridge *bridge)
 305 {
 306         struct softmix_bridge_data *softmix_data;
 307
 308         if (!(softmix_data = ao2_alloc(sizeof(*softmix_data), softmix_bridge_data_destroy))) {
 309                 return -1;
 310         }
 311         if (!(softmix_data->timer = ast_timer_open())) {
 312                 ao2_ref(softmix_data, -1);
 313                 return -1;
 314         }
 315
 316         /* start at 8khz, let it grow from there */
 317         softmix_data->internal_rate = 8000;
 318         softmix_data->internal_mixing_interval = DEFAULT_SOFTMIX_INTERVAL;
 319
 320         bridge->bridge_pvt = softmix_data;
 321         return 0;
 322 }
 323
 324 /*! \brief Function called when a bridge is destroyed */
 325 static int softmix_bridge_destroy(struct ast_bridge *bridge)
 326 {
 327         struct softmix_bridge_data *softmix_data = bridge->bridge_pvt;
 328         if (!bridge->bridge_pvt) {
 329                 return -1;
 330         }
 331         ao2_ref(softmix_data, -1);
 332         bridge->bridge_pvt = NULL;
 333         return 0;
 334 }
 335
 336 static void set_softmix_bridge_data(int rate, int interval, struct ast_bridge_channel *bridge_channel, int reset)
 337 {
 338         struct softmix_channel *sc = bridge_channel->bridge_pvt;
 339         unsigned int channel_read_rate = ast_format_rate(&bridge_channel->chan->rawreadformat);
 340
 341         ast_mutex_lock(&sc->lock);
 342         if (reset) {
 343                 ast_slinfactory_destroy(&sc->factory);
 344                 ast_dsp_free(sc->dsp);
 345         }
 346         /* Setup read/write frame parameters */
 347         sc->write_frame.frametype = AST_FRAME_VOICE;
 348         ast_format_set(&sc->write_frame.subclass.format, ast_format_slin_by_rate(rate), 0);
 349         sc->write_frame.data.ptr = sc->final_buf;
 350         sc->write_frame.datalen = SOFTMIX_DATALEN(rate, interval);
 351         sc->write_frame.samples = SOFTMIX_SAMPLES(rate, interval);
 352
 353         sc->read_frame.frametype = AST_FRAME_VOICE;
 354         ast_format_set(&sc->read_frame.subclass.format, ast_format_slin_by_rate(channel_read_rate), 0);
 355         sc->read_frame.data.ptr = sc->our_buf;
 356         sc->read_frame.datalen = SOFTMIX_DATALEN(channel_read_rate, interval);
 357         sc->read_frame.samples = SOFTMIX_SAMPLES(channel_read_rate, interval);
 358
 359         /* Setup smoother */
 360         ast_slinfactory_init_with_format(&sc->factory, &sc->write_frame.subclass.format);
 361
 362         /* set new read and write formats on channel. */
 363         ast_set_read_format(bridge_channel->chan, &sc->read_frame.subclass.format);
 364         ast_set_write_format(bridge_channel->chan, &sc->write_frame.subclass.format);
 365
 366         /* set up new DSP.  This is on the read side only right before the read frame enters the smoother.  */
 367         sc->dsp = ast_dsp_new_with_rate(channel_read_rate);
 368         /* we want to aggressively detect silence to avoid feedback */
 369         if (bridge_channel->tech_args.talking_threshold) {
 370                 ast_dsp_set_threshold(sc->dsp, bridge_channel->tech_args.talking_threshold);
 371         } else {
 372                 ast_dsp_set_threshold(sc->dsp, DEFAULT_SOFTMIX_TALKING_THRESHOLD);
 373         }
 374
 375         ast_mutex_unlock(&sc->lock);
 376 }
 377
 378 /*! \brief Function called when a channel is joined into the bridge */
 379 static int softmix_bridge_join(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
 380 {
 381         struct softmix_channel *sc = NULL;
 382         struct softmix_bridge_data *softmix_data = bridge->bridge_pvt;
 383
 384         /* Create a new softmix_channel structure and allocate various things on it */
 385         if (!(sc = ast_calloc(1, sizeof(*sc)))) {
 386                 return -1;
 387         }
 388
 389         /* Can't forget the lock */
 390         ast_mutex_init(&sc->lock);
 391
 392         /* Can't forget to record our pvt structure within the bridged channel structure */
 393         bridge_channel->bridge_pvt = sc;
 394
 395         set_softmix_bridge_data(softmix_data->internal_rate,
 396                 softmix_data->internal_mixing_interval ? softmix_data->internal_mixing_interval : DEFAULT_SOFTMIX_INTERVAL,
 397                 bridge_channel, 0);
 398
 399         return 0;
 400 }
 401
 402 /*! \brief Function called when a channel leaves the bridge */
 403 static int softmix_bridge_leave(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
 404 {
 405         struct softmix_channel *sc = bridge_channel->bridge_pvt;
 406
 407         if (!(bridge_channel->bridge_pvt)) {
 408                 return 0;
 409         }
 410         bridge_channel->bridge_pvt = NULL;
 411
 412         /* Drop mutex lock */
 413         ast_mutex_destroy(&sc->lock);
 414
 415         /* Drop the factory */
 416         ast_slinfactory_destroy(&sc->factory);
 417
 418         /* Drop the DSP */
 419         ast_dsp_free(sc->dsp);
 420
 421         /* Eep! drop ourselves */
 422         ast_free(sc);
 423
 424         return 0;
 425 }
 426
 427 /*!
 428  * \internal
 429  * \brief If the bridging core passes DTMF to us, then they want it to be distributed out to all memebers. Do that here.
 430  */
 431 static void softmix_pass_dtmf(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame)
 432 {
 433         struct ast_bridge_channel *tmp;
 434         AST_LIST_TRAVERSE(&bridge->channels, tmp, entry) {
 435                 if (tmp == bridge_channel) {
 436                         continue;
 437                 }
 438                 ast_write(tmp->chan, frame);
 439         }
 440 }
 441
 442 static void softmix_pass_video_top_priority(struct ast_bridge *bridge, struct ast_frame *frame)
 443 {
 444         struct ast_bridge_channel *tmp;
 445         AST_LIST_TRAVERSE(&bridge->channels, tmp, entry) {
 446                 if (tmp->suspended) {
 447                         continue;
 448                 }
 449                 if (ast_bridge_is_video_src(bridge, tmp->chan) == 1) {
 450                         ast_write(tmp->chan, frame);
 451                         break;
 452                 }
 453         }
 454 }
 455
 456 static void softmix_pass_video_all(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame, int echo)
 457 {
 458         struct ast_bridge_channel *tmp;
 459         AST_LIST_TRAVERSE(&bridge->channels, tmp, entry) {
 460                 if (tmp->suspended) {
 461                         continue;
 462                 }
 463                 if ((tmp->chan == bridge_channel->chan) && !echo) {
 464                         continue;
 465                 }
 466                 ast_write(tmp->chan, frame);
 467         }
 468 }
 469
 470 /*! \brief Function called when a channel writes a frame into the bridge */
 471 static enum ast_bridge_write_result softmix_bridge_write(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame)
 472 {
 473         struct softmix_channel *sc = bridge_channel->bridge_pvt;
 474         struct softmix_bridge_data *softmix_data = bridge->bridge_pvt;
 475         int totalsilence = 0;
 476         int cur_energy = 0;
 477         int silence_threshold = bridge_channel->tech_args.silence_threshold ?
 478                 bridge_channel->tech_args.silence_threshold :
 479                 DEFAULT_SOFTMIX_SILENCE_THRESHOLD;
 480         char update_talking = -1;  /* if this is set to 0 or 1, tell the bridge that the channel has started or stopped talking. */
 481         int res = AST_BRIDGE_WRITE_SUCCESS;
 482
 483         /* Only accept audio frames, all others are unsupported */
 484         if (frame->frametype == AST_FRAME_DTMF_END || frame->frametype == AST_FRAME_DTMF_BEGIN) {
 485                 softmix_pass_dtmf(bridge, bridge_channel, frame);
 486                 goto bridge_write_cleanup;
 487         } else if (frame->frametype != AST_FRAME_VOICE && frame->frametype != AST_FRAME_VIDEO) {
 488                 res = AST_BRIDGE_WRITE_UNSUPPORTED;
 489                 goto bridge_write_cleanup;
 490         } else if (frame->datalen == 0) {
 491                 goto bridge_write_cleanup;
 492         }
 493
 494         /* Determine if this video frame should be distributed or not */
 495         if (frame->frametype == AST_FRAME_VIDEO) {
 496                 int num_src = ast_bridge_number_video_src(bridge);
 497                 int video_src_priority = ast_bridge_is_video_src(bridge, bridge_channel->chan);
 498
 499                 switch (bridge->video_mode.mode) {
 500                 case AST_BRIDGE_VIDEO_MODE_NONE:
 501                         break;
 502                 case AST_BRIDGE_VIDEO_MODE_SINGLE_SRC:
 503                         if (video_src_priority == 1) {
 504                                 softmix_pass_video_all(bridge, bridge_channel, frame, 1);
 505                         }
 506                         break;
 507                 case AST_BRIDGE_VIDEO_MODE_TALKER_SRC:
 508                         ast_mutex_lock(&sc->lock);
 509                         ast_bridge_update_talker_src_video_mode(bridge, bridge_channel->chan, sc->video_talker.energy_average, ast_format_get_video_mark(&frame->subclass.format));
 510                         ast_mutex_unlock(&sc->lock);
 511                         if (video_src_priority == 1) {
 512                                 int echo = num_src > 1 ? 0 : 1;
 513                                 softmix_pass_video_all(bridge, bridge_channel, frame, echo);
 514                         } else if (video_src_priority == 2) {
 515                                 softmix_pass_video_top_priority(bridge, frame);
 516                         }
 517                         break;
 518                 }
 519                 goto bridge_write_cleanup;
 520         }
 521
 522         /* If we made it here, we are going to write the frame into the conference */
 523         ast_mutex_lock(&sc->lock);
 524         ast_dsp_silence_with_energy(sc->dsp, frame, &totalsilence, &cur_energy);
 525
 526         if (bridge->video_mode.mode == AST_BRIDGE_VIDEO_MODE_TALKER_SRC) {
 527                 int cur_slot = sc->video_talker.energy_history_cur_slot;
 528                 sc->video_talker.energy_accum -= sc->video_talker.energy_history[cur_slot];
 529                 sc->video_talker.energy_accum += cur_energy;
 530                 sc->video_talker.energy_history[cur_slot] = cur_energy;
 531                 sc->video_talker.energy_average = sc->video_talker.energy_accum / DEFAULT_ENERGY_HISTORY_LEN;
 532                 sc->video_talker.energy_history_cur_slot++;
 533                 if (sc->video_talker.energy_history_cur_slot == DEFAULT_ENERGY_HISTORY_LEN) {
 534                         sc->video_talker.energy_history_cur_slot = 0; /* wrap around */
 535                 }
 536         }
 537
 538         if (totalsilence < silence_threshold) {
 539                 if (!sc->talking) {
 540                         update_talking = 1;
 541                 }
 542                 sc->talking = 1; /* tell the write process we have audio to be mixed out */
 543         } else {
 544                 if (sc->talking) {
 545                         update_talking = 0;
 546                 }
 547                 sc->talking = 0;
 548         }
 549
 550         /* Before adding audio in, make sure we haven't fallen behind. If audio has fallen
 551          * behind 4 times the amount of samples mixed on every iteration of the mixer, Re-sync
 552          * the audio by flushing the buffer before adding new audio in. */
 553         if (ast_slinfactory_available(&sc->factory) > (4 * SOFTMIX_SAMPLES(softmix_data->internal_rate, softmix_data->internal_mixing_interval))) {
 554                 ast_slinfactory_flush(&sc->factory);
 555         }
 556
 557         /* If a frame was provided add it to the smoother, unless drop silence is enabled and this frame
 558          * is not determined to be talking. */
 559         if (!(bridge_channel->tech_args.drop_silence && !sc->talking) &&
 560                 (frame->frametype == AST_FRAME_VOICE && ast_format_is_slinear(&frame->subclass.format))) {
 561                 ast_slinfactory_feed(&sc->factory, frame);
 562         }
 563
 564         /* If a frame is ready to be written out, do so */
 565         if (sc->have_frame) {
 566                 ast_write(bridge_channel->chan, &sc->write_frame);
 567                 sc->have_frame = 0;
 568         }
 569
 570         /* Alllll done */
 571         ast_mutex_unlock(&sc->lock);
 572
 573         if (update_talking != -1) {
 574                 ast_bridge_notify_talking(bridge, bridge_channel, update_talking);
 575         }
 576
 577         return res;
 578
 579 bridge_write_cleanup:
 580         /* Even though the frame is not being written into the conference because it is not audio,
 581          * we should use this opportunity to check to see if a frame is ready to be written out from
 582          * the conference to the channel. */
 583         ast_mutex_lock(&sc->lock);
 584         if (sc->have_frame) {
 585                 ast_write(bridge_channel->chan, &sc->write_frame);
 586                 sc->have_frame = 0;
 587         }
 588         ast_mutex_unlock(&sc->lock);
 589
 590         return res;
 591 }
 592
 593 /*! \brief Function called when the channel's thread is poked */
 594 static int softmix_bridge_poke(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
 595 {
 596         struct softmix_channel *sc = bridge_channel->bridge_pvt;
 597
 598         ast_mutex_lock(&sc->lock);
 599
 600         if (sc->have_frame) {
 601                 ast_write(bridge_channel->chan, &sc->write_frame);
 602                 sc->have_frame = 0;
 603         }
 604
 605         ast_mutex_unlock(&sc->lock);
 606
 607         return 0;
 608 }
 609
 610 static void gather_softmix_stats(struct softmix_stats *stats,
 611         const struct softmix_bridge_data *softmix_data,
 612         struct ast_bridge_channel *bridge_channel)
 613 {
 614         int channel_native_rate;
 615         int i;
 616         /* Gather stats about channel sample rates. */
 617         channel_native_rate = MAX(ast_format_rate(&bridge_channel->chan->rawwriteformat),
 618                 ast_format_rate(&bridge_channel->chan->rawreadformat));
 619
 620         if (channel_native_rate > stats->highest_supported_rate) {
 621                 stats->highest_supported_rate = channel_native_rate;
 622         }
 623         if (channel_native_rate > softmix_data->internal_rate) {
 624                 for (i = 0; i < ARRAY_LEN(stats->sample_rates); i++) {
 625                         if (stats->sample_rates[i] == channel_native_rate) {
 626                                 stats->num_channels[i]++;
 627                                 break;
 628                         } else if (!stats->sample_rates[i]) {
 629                                 stats->sample_rates[i] = channel_native_rate;
 630                                 stats->num_channels[i]++;
 631                                 break;
 632                         }
 633                 }
 634                 stats->num_above_internal_rate++;
 635         } else if (channel_native_rate == softmix_data->internal_rate) {
 636                 stats->num_at_internal_rate++;
 637         }
 638 }
 639 /*!
 640  * \internal
 641  * \brief Analyse mixing statistics and change bridges internal rate
 642  * if necessary.
 643  *
 644  * \retval 0, no changes to internal rate
 645  * \ratval 1, internal rate was changed, update all the channels on the next mixing iteration.
 646  */
 647 static unsigned int analyse_softmix_stats(struct softmix_stats *stats, struct softmix_bridge_data *softmix_data)
 648 {
 649         int i;
 650         /* Re-adjust the internal bridge sample rate if
 651          * 1. The bridge's internal sample rate is locked in at a sample
 652          *    rate other than the current sample rate being used.
 653          * 2. two or more channels support a higher sample rate
 654          * 3. no channels support the current sample rate or a higher rate
 655          */
 656         if (stats->locked_rate) {
 657                 /* if the rate is locked by the bridge, only update it if it differs
 658                  * from the current rate we are using. */
 659                 if (softmix_data->internal_rate != stats->locked_rate) {
 660                         softmix_data->internal_rate = stats->locked_rate;
 661                         ast_debug(1, " Bridge is locked in at sample rate %d\n", softmix_data->internal_rate);
 662                         return 1;
 663                 }
 664         } else if (stats->num_above_internal_rate >= 2) {
 665                 /* the highest rate is just used as a starting point */
 666                 unsigned int best_rate = stats->highest_supported_rate;
 667                 int best_index = -1;
 668
 669                 for (i = 0; i < ARRAY_LEN(stats->num_channels); i++) {
 670                         if (stats->num_channels[i]) {
 671                                 break;
 672                         }
 673                         /* best_rate starts out being the first sample rate
 674                          * greater than the internal sample rate that 2 or
 675                          * more channels support. */
 676                         if (stats->num_channels[i] >= 2 && (best_index == -1)) {
 677                                 best_rate = stats->sample_rates[i];
 678                                 best_index = i;
 679                         /* If it has been detected that multiple rates above
 680                          * the internal rate are present, compare those rates
 681                          * to each other and pick the highest one two or more
 682                          * channels support. */
 683                         } else if (((best_index != -1) &&
 684                                 (stats->num_channels[i] >= 2) &&
 685                                 (stats->sample_rates[best_index] < stats->sample_rates[i]))) {
 686                                 best_rate = stats->sample_rates[i];
 687                                 best_index = i;
 688                         /* It is possible that multiple channels exist with native sample
 689                          * rates above the internal sample rate, but none of those channels
 690                          * have the same rate in common.  In this case, the lowest sample
 691                          * rate among those channels is picked. Over time as additional
 692                          * statistic runs are made the internal sample rate number will
 693                          * adjust to the most optimal sample rate, but it may take multiple
 694                          * iterations. */
 695                         } else if (best_index == -1) {
 696                                 best_rate = MIN(best_rate, stats->sample_rates[i]);
 697                         }
 698                 }
 699
 700                 ast_debug(1, " Bridge changed from %d To %d\n", softmix_data->internal_rate, best_rate);
 701                 softmix_data->internal_rate = best_rate;
 702                 return 1;
 703         } else if (!stats->num_at_internal_rate && !stats->num_above_internal_rate) {
 704                 /* In this case, the highest supported rate is actually lower than the internal rate */
 705                 softmix_data->internal_rate = stats->highest_supported_rate;
 706                 ast_debug(1, " Bridge changed from %d to %d\n", softmix_data->internal_rate, stats->highest_supported_rate);
 707                 return 1;
 708         }
 709         return 0;
 710 }
 711
 712 static int softmix_mixing_array_init(struct softmix_mixing_array *mixing_array, unsigned int starting_num_entries)
 713 {
 714         memset(mixing_array, 0, sizeof(*mixing_array));
 715         mixing_array->max_num_entries = starting_num_entries;
 716         if (!(mixing_array->buffers = ast_calloc(mixing_array->max_num_entries, sizeof(int16_t *)))) {
 717                 ast_log(LOG_NOTICE, "Failed to allocate softmix mixing structure. \n");
 718                 return -1;
 719         }
 720         return 0;
 721 }
 722
 723 static void softmix_mixing_array_destroy(struct softmix_mixing_array *mixing_array)
 724 {
 725         ast_free(mixing_array->buffers);
 726 }
 727
 728 static int softmix_mixing_array_grow(struct softmix_mixing_array *mixing_array, unsigned int num_entries)
 729 {
 730         int16_t **tmp;
 731         /* give it some room to grow since memory is cheap but allocations can be expensive */
 732         mixing_array->max_num_entries = num_entries;
 733         if (!(tmp = ast_realloc(mixing_array->buffers, (mixing_array->max_num_entries * sizeof(int16_t *))))) {
 734                 ast_log(LOG_NOTICE, "Failed to re-allocate softmix mixing structure. \n");
 735                 return -1;
 736         }
 737         mixing_array->buffers = tmp;
 738         return 0;
 739 }
 740
 741 /*! \brief Function which acts as the mixing thread */
 742 static int softmix_bridge_thread(struct ast_bridge *bridge)
 743 {
 744         struct softmix_stats stats = { { 0 }, };
 745         struct softmix_mixing_array mixing_array;
 746         struct softmix_bridge_data *softmix_data = bridge->bridge_pvt;
 747         struct ast_timer *timer;
 748         struct softmix_translate_helper trans_helper;
 749         int16_t buf[MAX_DATALEN] = { 0, };
 750         unsigned int stat_iteration_counter = 0; /* counts down, gather stats at zero and reset. */
 751         int timingfd;
 752         int update_all_rates = 0; /* set this when the internal sample rate has changed */
 753         int i, x;
 754         int res = -1;
 755
 756         if (!(softmix_data = bridge->bridge_pvt)) {
 757                 goto softmix_cleanup;
 758         }
 759
 760         ao2_ref(softmix_data, 1);
 761         timer = softmix_data->timer;
 762         timingfd = ast_timer_fd(timer);
 763         softmix_translate_helper_init(&trans_helper, softmix_data->internal_rate);
 764         ast_timer_set_rate(timer, (1000 / softmix_data->internal_mixing_interval));
 765
 766         /* Give the mixing array room to grow, memory is cheap but allocations are expensive. */
 767         if (softmix_mixing_array_init(&mixing_array, bridge->num + 10)) {
 768                 ast_log(LOG_NOTICE, "Failed to allocate softmix mixing structure. \n");
 769                 goto softmix_cleanup;
 770         }
 771
 772         while (!bridge->stop && !bridge->refresh && bridge->array_num) {
 773                 struct ast_bridge_channel *bridge_channel = NULL;
 774                 int timeout = -1;
 775                 enum ast_format_id cur_slin_id = ast_format_slin_by_rate(softmix_data->internal_rate);
 776                 unsigned int softmix_samples = SOFTMIX_SAMPLES(softmix_data->internal_rate, softmix_data->internal_mixing_interval);
 777                 unsigned int softmix_datalen = SOFTMIX_DATALEN(softmix_data->internal_rate, softmix_data->internal_mixing_interval);
 778
 779                 if (softmix_datalen > MAX_DATALEN) {
 780                         /* This should NEVER happen, but if it does we need to know about it. Almost
 781                          * all the memcpys used during this process depend on this assumption.  Rather
 782                          * than checking this over and over again through out the code, this single
 783                          * verification is done on each iteration. */
 784                         ast_log(LOG_WARNING, "Conference mixing error, requested mixing length greater than mixing buffer.\n");
 785                         goto softmix_cleanup;
 786                 }
 787
 788                 /* Grow the mixing array buffer as participants are added. */
 789                 if (mixing_array.max_num_entries < bridge->num && softmix_mixing_array_grow(&mixing_array, bridge->num + 5)) {
 790                         goto softmix_cleanup;
 791                 }
 792
 793                 /* init the number of buffers stored in the mixing array to 0.
 794                  * As buffers are added for mixing, this number is incremented. */
 795                 mixing_array.used_entries = 0;
 796
 797                 /* These variables help determine if a rate change is required */
 798                 if (!stat_iteration_counter) {
 799                         memset(&stats, 0, sizeof(stats));
 800                         stats.locked_rate = bridge->internal_sample_rate;
 801                 }
 802
 803                 /* If the sample rate has changed, update the translator helper */
 804                 if (update_all_rates) {
 805                         softmix_translate_helper_change_rate(&trans_helper, softmix_data->internal_rate);
 806                 }
 807
 808                 /* Go through pulling audio from each factory that has it available */
 809                 AST_LIST_TRAVERSE(&bridge->channels, bridge_channel, entry) {
 810                         struct softmix_channel *sc = bridge_channel->bridge_pvt;
 811
 812                         /* Update the sample rate to match the bridge's native sample rate if necessary. */
 813                         if (update_all_rates) {
 814                                 set_softmix_bridge_data(softmix_data->internal_rate, softmix_data->internal_mixing_interval, bridge_channel, 1);
 815                         }
 816
 817                         /* If stat_iteration_counter is 0, then collect statistics during this mixing interation */
 818                         if (!stat_iteration_counter) {
 819                                 gather_softmix_stats(&stats, softmix_data, bridge_channel);
 820                         }
 821
 822                         /* if the channel is suspended, don't check for audio, but still gather stats */
 823                         if (bridge_channel->suspended) {
 824                                 continue;
 825                         }
 826
 827                         /* Try to get audio from the factory if available */
 828                         ast_mutex_lock(&sc->lock);
 829                         if ((mixing_array.buffers[mixing_array.used_entries] = softmix_process_read_audio(sc, softmix_samples))) {
 830                                 mixing_array.used_entries++;
 831                         }
 832                         ast_mutex_unlock(&sc->lock);
 833                 }
 834
 835                 /* mix it like crazy */
 836                 memset(buf, 0, softmix_datalen);
 837                 for (i = 0; i < mixing_array.used_entries; i++) {
 838                         for (x = 0; x < softmix_samples; x++) {
 839                                 ast_slinear_saturated_add(buf + x, mixing_array.buffers[i] + x);
 840                         }
 841                 }
 842
 843                 /* Next step go through removing the channel's own audio and creating a good frame... */
 844                 AST_LIST_TRAVERSE(&bridge->channels, bridge_channel, entry) {
 845                         struct softmix_channel *sc = bridge_channel->bridge_pvt;
 846
 847                         if (bridge_channel->suspended) {
 848                                 continue;
 849                         }
 850
 851                         ast_mutex_lock(&sc->lock);
 852
 853                         /* Make SLINEAR write frame from local buffer */
 854                         if (sc->write_frame.subclass.format.id != cur_slin_id) {
 855                                 ast_format_set(&sc->write_frame.subclass.format, cur_slin_id, 0);
 856                         }
 857                         sc->write_frame.datalen = softmix_datalen;
 858                         sc->write_frame.samples = softmix_samples;
 859                         memcpy(sc->final_buf, buf, softmix_datalen);
 860
 861                         /* process the softmix channel's new write audio */
 862                         softmix_process_write_audio(&trans_helper, &bridge_channel->chan->rawwriteformat, sc);
 863
 864                         /* The frame is now ready for use... */
 865                         sc->have_frame = 1;
 866
 867                         ast_mutex_unlock(&sc->lock);
 868
 869                         /* Poke bridged channel thread just in case */
 870                         pthread_kill(bridge_channel->thread, SIGURG);
 871                 }
 872
 873                 update_all_rates = 0;
 874                 if (!stat_iteration_counter) {
 875                         update_all_rates = analyse_softmix_stats(&stats, softmix_data);
 876                         stat_iteration_counter = SOFTMIX_STAT_INTERVAL;
 877                 }
 878                 stat_iteration_counter--;
 879
 880                 ao2_unlock(bridge);
 881                 /* cleanup any translation frame data from the previous mixing iteration. */
 882                 softmix_translate_helper_cleanup(&trans_helper);
 883                 /* Wait for the timing source to tell us to wake up and get things done */
 884                 ast_waitfor_n_fd(&timingfd, 1, &timeout, NULL);
 885                 ast_timer_ack(timer, 1);
 886                 ao2_lock(bridge);
 887
 888                 /* make sure to detect mixing interval changes if they occur. */
 889                 if (bridge->internal_mixing_interval && (bridge->internal_mixing_interval != softmix_data->internal_mixing_interval)) {
 890                         softmix_data->internal_mixing_interval = bridge->internal_mixing_interval;
 891                         ast_timer_set_rate(timer, (1000 / softmix_data->internal_mixing_interval));
 892                         update_all_rates = 1; /* if the interval changes, the rates must be adjusted as well just to be notified new interval.*/
 893                 }
 894         }
 895
 896         res = 0;
 897
 898 softmix_cleanup:
 899         softmix_translate_helper_destroy(&trans_helper);
 900         softmix_mixing_array_destroy(&mixing_array);
 901         if (softmix_data) {
 902                 ao2_ref(softmix_data, -1);
 903         }
 904         return res;
 905 }
 906
 907 static struct ast_bridge_technology softmix_bridge = {
 908         .name = "softmix",
 909         .capabilities = AST_BRIDGE_CAPABILITY_MULTIMIX | AST_BRIDGE_CAPABILITY_THREAD | AST_BRIDGE_CAPABILITY_MULTITHREADED | AST_BRIDGE_CAPABILITY_OPTIMIZE | AST_BRIDGE_CAPABILITY_VIDEO,
 910         .preference = AST_BRIDGE_PREFERENCE_LOW,
 911         .create = softmix_bridge_create,
 912         .destroy = softmix_bridge_destroy,
 913         .join = softmix_bridge_join,
 914         .leave = softmix_bridge_leave,
 915         .write = softmix_bridge_write,
 916         .thread = softmix_bridge_thread,
 917         .poke = softmix_bridge_poke,
 918 };
 919
 920 static int unload_module(void)
 921 {
 922         ast_format_cap_destroy(softmix_bridge.format_capabilities);
 923         return ast_bridge_technology_unregister(&softmix_bridge);
 924 }
 925
 926 static int load_module(void)
 927 {
 928         struct ast_format tmp;
 929         if (!(softmix_bridge.format_capabilities = ast_format_cap_alloc())) {
 930                 return AST_MODULE_LOAD_DECLINE;
 931         }
 932         ast_format_cap_add(softmix_bridge.format_capabilities, ast_format_set(&tmp, AST_FORMAT_SLINEAR, 0));
 933         return ast_bridge_technology_register(&softmix_bridge);
 934 }
 935
 936 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Multi-party software based channel mixing");