More trivial bridge code cleanup.
[asterisk/asterisk.git] / 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;
288
289         AST_LIST_TRAVERSE(&trans_helper->entries, entry, entry) {
290                 if (entry->out_frame) {
291                         ast_frfree(entry->out_frame);
292                         entry->out_frame = NULL;
293                 }
294                 entry->num_times_requested = 0;
295         }
296 }
297
298 static void softmix_bridge_data_destroy(void *obj)
299 {
300         struct softmix_bridge_data *softmix_data = obj;
301
302         if (softmix_data->timer) {
303                 ast_timer_close(softmix_data->timer);
304                 softmix_data->timer = NULL;
305         }
306 }
307
308 /*! \brief Function called when a bridge is created */
309 static int softmix_bridge_create(struct ast_bridge *bridge)
310 {
311         struct softmix_bridge_data *softmix_data;
312
313         if (!(softmix_data = ao2_alloc(sizeof(*softmix_data), softmix_bridge_data_destroy))) {
314                 return -1;
315         }
316         if (!(softmix_data->timer = ast_timer_open())) {
317                 ao2_ref(softmix_data, -1);
318                 return -1;
319         }
320
321         /* start at 8khz, let it grow from there */
322         softmix_data->internal_rate = 8000;
323         softmix_data->internal_mixing_interval = DEFAULT_SOFTMIX_INTERVAL;
324
325         bridge->bridge_pvt = softmix_data;
326         return 0;
327 }
328
329 /*! \brief Function called when a bridge is destroyed */
330 static int softmix_bridge_destroy(struct ast_bridge *bridge)
331 {
332         struct softmix_bridge_data *softmix_data;
333
334         softmix_data = bridge->bridge_pvt;
335         if (!softmix_data) {
336                 return -1;
337         }
338         ao2_ref(softmix_data, -1);
339         bridge->bridge_pvt = NULL;
340         return 0;
341 }
342
343 static void set_softmix_bridge_data(int rate, int interval, struct ast_bridge_channel *bridge_channel, int reset)
344 {
345         struct softmix_channel *sc = bridge_channel->bridge_pvt;
346         unsigned int channel_read_rate = ast_format_rate(ast_channel_rawreadformat(bridge_channel->chan));
347
348         ast_mutex_lock(&sc->lock);
349         if (reset) {
350                 ast_slinfactory_destroy(&sc->factory);
351                 ast_dsp_free(sc->dsp);
352         }
353         /* Setup read/write frame parameters */
354         sc->write_frame.frametype = AST_FRAME_VOICE;
355         ast_format_set(&sc->write_frame.subclass.format, ast_format_slin_by_rate(rate), 0);
356         sc->write_frame.data.ptr = sc->final_buf;
357         sc->write_frame.datalen = SOFTMIX_DATALEN(rate, interval);
358         sc->write_frame.samples = SOFTMIX_SAMPLES(rate, interval);
359
360         sc->read_frame.frametype = AST_FRAME_VOICE;
361         ast_format_set(&sc->read_frame.subclass.format, ast_format_slin_by_rate(channel_read_rate), 0);
362         sc->read_frame.data.ptr = sc->our_buf;
363         sc->read_frame.datalen = SOFTMIX_DATALEN(channel_read_rate, interval);
364         sc->read_frame.samples = SOFTMIX_SAMPLES(channel_read_rate, interval);
365
366         /* Setup smoother */
367         ast_slinfactory_init_with_format(&sc->factory, &sc->write_frame.subclass.format);
368
369         /* set new read and write formats on channel. */
370         ast_set_read_format(bridge_channel->chan, &sc->read_frame.subclass.format);
371         ast_set_write_format(bridge_channel->chan, &sc->write_frame.subclass.format);
372
373         /* set up new DSP.  This is on the read side only right before the read frame enters the smoother.  */
374         sc->dsp = ast_dsp_new_with_rate(channel_read_rate);
375         /* we want to aggressively detect silence to avoid feedback */
376         if (bridge_channel->tech_args.talking_threshold) {
377                 ast_dsp_set_threshold(sc->dsp, bridge_channel->tech_args.talking_threshold);
378         } else {
379                 ast_dsp_set_threshold(sc->dsp, DEFAULT_SOFTMIX_TALKING_THRESHOLD);
380         }
381
382         ast_mutex_unlock(&sc->lock);
383 }
384
385 /*! \brief Function called when a channel is joined into the bridge */
386 static int softmix_bridge_join(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
387 {
388         struct softmix_channel *sc;
389         struct softmix_bridge_data *softmix_data = bridge->bridge_pvt;
390
391         /* Create a new softmix_channel structure and allocate various things on it */
392         if (!(sc = ast_calloc(1, sizeof(*sc)))) {
393                 return -1;
394         }
395
396         /* Can't forget the lock */
397         ast_mutex_init(&sc->lock);
398
399         /* Can't forget to record our pvt structure within the bridged channel structure */
400         bridge_channel->bridge_pvt = sc;
401
402         set_softmix_bridge_data(softmix_data->internal_rate,
403                 softmix_data->internal_mixing_interval ? softmix_data->internal_mixing_interval : DEFAULT_SOFTMIX_INTERVAL,
404                 bridge_channel, 0);
405
406         return 0;
407 }
408
409 /*! \brief Function called when a channel leaves the bridge */
410 static int softmix_bridge_leave(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
411 {
412         struct softmix_channel *sc = bridge_channel->bridge_pvt;
413
414         if (!(bridge_channel->bridge_pvt)) {
415                 return 0;
416         }
417         bridge_channel->bridge_pvt = NULL;
418
419         /* Drop mutex lock */
420         ast_mutex_destroy(&sc->lock);
421
422         /* Drop the factory */
423         ast_slinfactory_destroy(&sc->factory);
424
425         /* Drop the DSP */
426         ast_dsp_free(sc->dsp);
427
428         /* Eep! drop ourselves */
429         ast_free(sc);
430
431         return 0;
432 }
433
434 /*!
435  * \internal
436  * \brief If the bridging core passes DTMF to us, then they want it to be distributed out to all memebers. Do that here.
437  */
438 static void softmix_pass_dtmf(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame)
439 {
440         struct ast_bridge_channel *tmp;
441         AST_LIST_TRAVERSE(&bridge->channels, tmp, entry) {
442                 if (tmp == bridge_channel) {
443                         continue;
444                 }
445                 ast_write(tmp->chan, frame);
446         }
447 }
448
449 static void softmix_pass_video_top_priority(struct ast_bridge *bridge, struct ast_frame *frame)
450 {
451         struct ast_bridge_channel *tmp;
452         AST_LIST_TRAVERSE(&bridge->channels, tmp, entry) {
453                 if (tmp->suspended) {
454                         continue;
455                 }
456                 if (ast_bridge_is_video_src(bridge, tmp->chan) == 1) {
457                         ast_write(tmp->chan, frame);
458                         break;
459                 }
460         }
461 }
462
463 static void softmix_pass_video_all(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame, int echo)
464 {
465         struct ast_bridge_channel *tmp;
466         AST_LIST_TRAVERSE(&bridge->channels, tmp, entry) {
467                 if (tmp->suspended) {
468                         continue;
469                 }
470                 if ((tmp->chan == bridge_channel->chan) && !echo) {
471                         continue;
472                 }
473                 ast_write(tmp->chan, frame);
474         }
475 }
476
477 /*! \brief Function called when a channel writes a frame into the bridge */
478 static enum ast_bridge_write_result softmix_bridge_write(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame)
479 {
480         struct softmix_channel *sc = bridge_channel->bridge_pvt;
481         struct softmix_bridge_data *softmix_data = bridge->bridge_pvt;
482         int totalsilence = 0;
483         int cur_energy = 0;
484         int silence_threshold = bridge_channel->tech_args.silence_threshold ?
485                 bridge_channel->tech_args.silence_threshold :
486                 DEFAULT_SOFTMIX_SILENCE_THRESHOLD;
487         char update_talking = -1;  /* if this is set to 0 or 1, tell the bridge that the channel has started or stopped talking. */
488         int res = AST_BRIDGE_WRITE_SUCCESS;
489
490         /* Only accept audio frames, all others are unsupported */
491         if (frame->frametype == AST_FRAME_DTMF_END || frame->frametype == AST_FRAME_DTMF_BEGIN) {
492                 softmix_pass_dtmf(bridge, bridge_channel, frame);
493                 goto bridge_write_cleanup;
494         } else if (frame->frametype != AST_FRAME_VOICE && frame->frametype != AST_FRAME_VIDEO) {
495                 res = AST_BRIDGE_WRITE_UNSUPPORTED;
496                 goto bridge_write_cleanup;
497         } else if (frame->datalen == 0) {
498                 goto bridge_write_cleanup;
499         }
500
501         /* Determine if this video frame should be distributed or not */
502         if (frame->frametype == AST_FRAME_VIDEO) {
503                 int num_src = ast_bridge_number_video_src(bridge);
504                 int video_src_priority = ast_bridge_is_video_src(bridge, bridge_channel->chan);
505
506                 switch (bridge->video_mode.mode) {
507                 case AST_BRIDGE_VIDEO_MODE_NONE:
508                         break;
509                 case AST_BRIDGE_VIDEO_MODE_SINGLE_SRC:
510                         if (video_src_priority == 1) {
511                                 softmix_pass_video_all(bridge, bridge_channel, frame, 1);
512                         }
513                         break;
514                 case AST_BRIDGE_VIDEO_MODE_TALKER_SRC:
515                         ast_mutex_lock(&sc->lock);
516                         ast_bridge_update_talker_src_video_mode(bridge, bridge_channel->chan, sc->video_talker.energy_average, ast_format_get_video_mark(&frame->subclass.format));
517                         ast_mutex_unlock(&sc->lock);
518                         if (video_src_priority == 1) {
519                                 int echo = num_src > 1 ? 0 : 1;
520                                 softmix_pass_video_all(bridge, bridge_channel, frame, echo);
521                         } else if (video_src_priority == 2) {
522                                 softmix_pass_video_top_priority(bridge, frame);
523                         }
524                         break;
525                 }
526                 goto bridge_write_cleanup;
527         }
528
529         /* If we made it here, we are going to write the frame into the conference */
530         ast_mutex_lock(&sc->lock);
531         ast_dsp_silence_with_energy(sc->dsp, frame, &totalsilence, &cur_energy);
532
533         if (bridge->video_mode.mode == AST_BRIDGE_VIDEO_MODE_TALKER_SRC) {
534                 int cur_slot = sc->video_talker.energy_history_cur_slot;
535                 sc->video_talker.energy_accum -= sc->video_talker.energy_history[cur_slot];
536                 sc->video_talker.energy_accum += cur_energy;
537                 sc->video_talker.energy_history[cur_slot] = cur_energy;
538                 sc->video_talker.energy_average = sc->video_talker.energy_accum / DEFAULT_ENERGY_HISTORY_LEN;
539                 sc->video_talker.energy_history_cur_slot++;
540                 if (sc->video_talker.energy_history_cur_slot == DEFAULT_ENERGY_HISTORY_LEN) {
541                         sc->video_talker.energy_history_cur_slot = 0; /* wrap around */
542                 }
543         }
544
545         if (totalsilence < silence_threshold) {
546                 if (!sc->talking) {
547                         update_talking = 1;
548                 }
549                 sc->talking = 1; /* tell the write process we have audio to be mixed out */
550         } else {
551                 if (sc->talking) {
552                         update_talking = 0;
553                 }
554                 sc->talking = 0;
555         }
556
557         /* Before adding audio in, make sure we haven't fallen behind. If audio has fallen
558          * behind 4 times the amount of samples mixed on every iteration of the mixer, Re-sync
559          * the audio by flushing the buffer before adding new audio in. */
560         if (ast_slinfactory_available(&sc->factory) > (4 * SOFTMIX_SAMPLES(softmix_data->internal_rate, softmix_data->internal_mixing_interval))) {
561                 ast_slinfactory_flush(&sc->factory);
562         }
563
564         /* If a frame was provided add it to the smoother, unless drop silence is enabled and this frame
565          * is not determined to be talking. */
566         if (!(bridge_channel->tech_args.drop_silence && !sc->talking) &&
567                 (frame->frametype == AST_FRAME_VOICE && ast_format_is_slinear(&frame->subclass.format))) {
568                 ast_slinfactory_feed(&sc->factory, frame);
569         }
570
571         /* If a frame is ready to be written out, do so */
572         if (sc->have_frame) {
573                 ast_write(bridge_channel->chan, &sc->write_frame);
574                 sc->have_frame = 0;
575         }
576
577         /* Alllll done */
578         ast_mutex_unlock(&sc->lock);
579
580         if (update_talking != -1) {
581                 ast_bridge_notify_talking(bridge, bridge_channel, update_talking);
582         }
583
584         return res;
585
586 bridge_write_cleanup:
587         /* Even though the frame is not being written into the conference because it is not audio,
588          * we should use this opportunity to check to see if a frame is ready to be written out from
589          * the conference to the channel. */
590         ast_mutex_lock(&sc->lock);
591         if (sc->have_frame) {
592                 ast_write(bridge_channel->chan, &sc->write_frame);
593                 sc->have_frame = 0;
594         }
595         ast_mutex_unlock(&sc->lock);
596
597         return res;
598 }
599
600 /*! \brief Function called when the channel's thread is poked */
601 static int softmix_bridge_poke(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
602 {
603         struct softmix_channel *sc = bridge_channel->bridge_pvt;
604
605         ast_mutex_lock(&sc->lock);
606
607         if (sc->have_frame) {
608                 ast_write(bridge_channel->chan, &sc->write_frame);
609                 sc->have_frame = 0;
610         }
611
612         ast_mutex_unlock(&sc->lock);
613
614         return 0;
615 }
616
617 static void gather_softmix_stats(struct softmix_stats *stats,
618         const struct softmix_bridge_data *softmix_data,
619         struct ast_bridge_channel *bridge_channel)
620 {
621         int channel_native_rate;
622         int i;
623         /* Gather stats about channel sample rates. */
624         channel_native_rate = MAX(ast_format_rate(ast_channel_rawwriteformat(bridge_channel->chan)),
625                 ast_format_rate(ast_channel_rawreadformat(bridge_channel->chan)));
626
627         if (channel_native_rate > stats->highest_supported_rate) {
628                 stats->highest_supported_rate = channel_native_rate;
629         }
630         if (channel_native_rate > softmix_data->internal_rate) {
631                 for (i = 0; i < ARRAY_LEN(stats->sample_rates); i++) {
632                         if (stats->sample_rates[i] == channel_native_rate) {
633                                 stats->num_channels[i]++;
634                                 break;
635                         } else if (!stats->sample_rates[i]) {
636                                 stats->sample_rates[i] = channel_native_rate;
637                                 stats->num_channels[i]++;
638                                 break;
639                         }
640                 }
641                 stats->num_above_internal_rate++;
642         } else if (channel_native_rate == softmix_data->internal_rate) {
643                 stats->num_at_internal_rate++;
644         }
645 }
646 /*!
647  * \internal
648  * \brief Analyse mixing statistics and change bridges internal rate
649  * if necessary.
650  *
651  * \retval 0, no changes to internal rate 
652  * \ratval 1, internal rate was changed, update all the channels on the next mixing iteration.
653  */
654 static unsigned int analyse_softmix_stats(struct softmix_stats *stats, struct softmix_bridge_data *softmix_data)
655 {
656         int i;
657         /* Re-adjust the internal bridge sample rate if
658          * 1. The bridge's internal sample rate is locked in at a sample
659          *    rate other than the current sample rate being used.
660          * 2. two or more channels support a higher sample rate
661          * 3. no channels support the current sample rate or a higher rate
662          */
663         if (stats->locked_rate) {
664                 /* if the rate is locked by the bridge, only update it if it differs
665                  * from the current rate we are using. */
666                 if (softmix_data->internal_rate != stats->locked_rate) {
667                         softmix_data->internal_rate = stats->locked_rate;
668                         ast_debug(1, " Bridge is locked in at sample rate %d\n", softmix_data->internal_rate);
669                         return 1;
670                 }
671         } else if (stats->num_above_internal_rate >= 2) {
672                 /* the highest rate is just used as a starting point */
673                 unsigned int best_rate = stats->highest_supported_rate;
674                 int best_index = -1;
675
676                 for (i = 0; i < ARRAY_LEN(stats->num_channels); i++) {
677                         if (stats->num_channels[i]) {
678                                 break;
679                         }
680                         /* best_rate starts out being the first sample rate
681                          * greater than the internal sample rate that 2 or
682                          * more channels support. */
683                         if (stats->num_channels[i] >= 2 && (best_index == -1)) {
684                                 best_rate = stats->sample_rates[i];
685                                 best_index = i;
686                         /* If it has been detected that multiple rates above
687                          * the internal rate are present, compare those rates
688                          * to each other and pick the highest one two or more
689                          * channels support. */
690                         } else if (((best_index != -1) &&
691                                 (stats->num_channels[i] >= 2) &&
692                                 (stats->sample_rates[best_index] < stats->sample_rates[i]))) {
693                                 best_rate = stats->sample_rates[i];
694                                 best_index = i;
695                         /* It is possible that multiple channels exist with native sample
696                          * rates above the internal sample rate, but none of those channels
697                          * have the same rate in common.  In this case, the lowest sample
698                          * rate among those channels is picked. Over time as additional
699                          * statistic runs are made the internal sample rate number will
700                          * adjust to the most optimal sample rate, but it may take multiple
701                          * iterations. */
702                         } else if (best_index == -1) {
703                                 best_rate = MIN(best_rate, stats->sample_rates[i]);
704                         }
705                 }
706
707                 ast_debug(1, " Bridge changed from %d To %d\n", softmix_data->internal_rate, best_rate);
708                 softmix_data->internal_rate = best_rate;
709                 return 1;
710         } else if (!stats->num_at_internal_rate && !stats->num_above_internal_rate) {
711                 /* In this case, the highest supported rate is actually lower than the internal rate */
712                 softmix_data->internal_rate = stats->highest_supported_rate;
713                 ast_debug(1, " Bridge changed from %d to %d\n", softmix_data->internal_rate, stats->highest_supported_rate);
714                 return 1;
715         }
716         return 0;
717 }
718
719 static int softmix_mixing_array_init(struct softmix_mixing_array *mixing_array, unsigned int starting_num_entries)
720 {
721         memset(mixing_array, 0, sizeof(*mixing_array));
722         mixing_array->max_num_entries = starting_num_entries;
723         if (!(mixing_array->buffers = ast_calloc(mixing_array->max_num_entries, sizeof(int16_t *)))) {
724                 ast_log(LOG_NOTICE, "Failed to allocate softmix mixing structure.\n");
725                 return -1;
726         }
727         return 0;
728 }
729
730 static void softmix_mixing_array_destroy(struct softmix_mixing_array *mixing_array)
731 {
732         ast_free(mixing_array->buffers);
733 }
734
735 static int softmix_mixing_array_grow(struct softmix_mixing_array *mixing_array, unsigned int num_entries)
736 {
737         int16_t **tmp;
738         /* give it some room to grow since memory is cheap but allocations can be expensive */
739         mixing_array->max_num_entries = num_entries;
740         if (!(tmp = ast_realloc(mixing_array->buffers, (mixing_array->max_num_entries * sizeof(int16_t *))))) {
741                 ast_log(LOG_NOTICE, "Failed to re-allocate softmix mixing structure.\n");
742                 return -1;
743         }
744         mixing_array->buffers = tmp;
745         return 0;
746 }
747
748 /*! \brief Function which acts as the mixing thread */
749 static int softmix_bridge_thread(struct ast_bridge *bridge)
750 {
751         struct softmix_stats stats = { { 0 }, };
752         struct softmix_mixing_array mixing_array;
753         struct softmix_bridge_data *softmix_data;
754         struct ast_timer *timer;
755         struct softmix_translate_helper trans_helper;
756         int16_t buf[MAX_DATALEN];
757         unsigned int stat_iteration_counter = 0; /* counts down, gather stats at zero and reset. */
758         int timingfd;
759         int update_all_rates = 0; /* set this when the internal sample rate has changed */
760         int i, x;
761         int res = -1;
762
763         softmix_data = bridge->bridge_pvt;
764         if (!softmix_data) {
765                 goto softmix_cleanup;
766         }
767
768         ao2_ref(softmix_data, 1);
769         timer = softmix_data->timer;
770         timingfd = ast_timer_fd(timer);
771         softmix_translate_helper_init(&trans_helper, softmix_data->internal_rate);
772         ast_timer_set_rate(timer, (1000 / softmix_data->internal_mixing_interval));
773
774         /* Give the mixing array room to grow, memory is cheap but allocations are expensive. */
775         if (softmix_mixing_array_init(&mixing_array, bridge->num + 10)) {
776                 goto softmix_cleanup;
777         }
778
779         while (!bridge->stop && !bridge->refresh && bridge->array_num) {
780                 struct ast_bridge_channel *bridge_channel;
781                 int timeout = -1;
782                 enum ast_format_id cur_slin_id = ast_format_slin_by_rate(softmix_data->internal_rate);
783                 unsigned int softmix_samples = SOFTMIX_SAMPLES(softmix_data->internal_rate, softmix_data->internal_mixing_interval);
784                 unsigned int softmix_datalen = SOFTMIX_DATALEN(softmix_data->internal_rate, softmix_data->internal_mixing_interval);
785
786                 if (softmix_datalen > MAX_DATALEN) {
787                         /* This should NEVER happen, but if it does we need to know about it. Almost
788                          * all the memcpys used during this process depend on this assumption.  Rather
789                          * than checking this over and over again through out the code, this single
790                          * verification is done on each iteration. */
791                         ast_log(LOG_WARNING, "Conference mixing error, requested mixing length greater than mixing buffer.\n");
792                         goto softmix_cleanup;
793                 }
794
795                 /* Grow the mixing array buffer as participants are added. */
796                 if (mixing_array.max_num_entries < bridge->num
797                         && softmix_mixing_array_grow(&mixing_array, bridge->num + 5)) {
798                         goto softmix_cleanup;
799                 }
800
801                 /* init the number of buffers stored in the mixing array to 0.
802                  * As buffers are added for mixing, this number is incremented. */
803                 mixing_array.used_entries = 0;
804
805                 /* These variables help determine if a rate change is required */
806                 if (!stat_iteration_counter) {
807                         memset(&stats, 0, sizeof(stats));
808                         stats.locked_rate = bridge->internal_sample_rate;
809                 }
810
811                 /* If the sample rate has changed, update the translator helper */
812                 if (update_all_rates) {
813                         softmix_translate_helper_change_rate(&trans_helper, softmix_data->internal_rate);
814                 }
815
816                 /* Go through pulling audio from each factory that has it available */
817                 AST_LIST_TRAVERSE(&bridge->channels, bridge_channel, entry) {
818                         struct softmix_channel *sc = bridge_channel->bridge_pvt;
819
820                         /* Update the sample rate to match the bridge's native sample rate if necessary. */
821                         if (update_all_rates) {
822                                 set_softmix_bridge_data(softmix_data->internal_rate, softmix_data->internal_mixing_interval, bridge_channel, 1);
823                         }
824
825                         /* If stat_iteration_counter is 0, then collect statistics during this mixing interation */
826                         if (!stat_iteration_counter) {
827                                 gather_softmix_stats(&stats, softmix_data, bridge_channel);
828                         }
829
830                         /* if the channel is suspended, don't check for audio, but still gather stats */
831                         if (bridge_channel->suspended) {
832                                 continue;
833                         }
834
835                         /* Try to get audio from the factory if available */
836                         ast_mutex_lock(&sc->lock);
837                         if ((mixing_array.buffers[mixing_array.used_entries] = softmix_process_read_audio(sc, softmix_samples))) {
838                                 mixing_array.used_entries++;
839                         }
840                         ast_mutex_unlock(&sc->lock);
841                 }
842
843                 /* mix it like crazy */
844                 memset(buf, 0, softmix_datalen);
845                 for (i = 0; i < mixing_array.used_entries; i++) {
846                         for (x = 0; x < softmix_samples; x++) {
847                                 ast_slinear_saturated_add(buf + x, mixing_array.buffers[i] + x);
848                         }
849                 }
850
851                 /* Next step go through removing the channel's own audio and creating a good frame... */
852                 AST_LIST_TRAVERSE(&bridge->channels, bridge_channel, entry) {
853                         struct softmix_channel *sc = bridge_channel->bridge_pvt;
854
855                         if (bridge_channel->suspended) {
856                                 continue;
857                         }
858
859                         ast_mutex_lock(&sc->lock);
860
861                         /* Make SLINEAR write frame from local buffer */
862                         if (sc->write_frame.subclass.format.id != cur_slin_id) {
863                                 ast_format_set(&sc->write_frame.subclass.format, cur_slin_id, 0);
864                         }
865                         sc->write_frame.datalen = softmix_datalen;
866                         sc->write_frame.samples = softmix_samples;
867                         memcpy(sc->final_buf, buf, softmix_datalen);
868
869                         /* process the softmix channel's new write audio */
870                         softmix_process_write_audio(&trans_helper, ast_channel_rawwriteformat(bridge_channel->chan), sc);
871
872                         /* The frame is now ready for use... */
873                         sc->have_frame = 1;
874
875                         ast_mutex_unlock(&sc->lock);
876
877                         /* Poke bridged channel thread just in case */
878                         pthread_kill(bridge_channel->thread, SIGURG);
879                 }
880
881                 update_all_rates = 0;
882                 if (!stat_iteration_counter) {
883                         update_all_rates = analyse_softmix_stats(&stats, softmix_data);
884                         stat_iteration_counter = SOFTMIX_STAT_INTERVAL;
885                 }
886                 stat_iteration_counter--;
887
888                 ao2_unlock(bridge);
889                 /* cleanup any translation frame data from the previous mixing iteration. */
890                 softmix_translate_helper_cleanup(&trans_helper);
891                 /* Wait for the timing source to tell us to wake up and get things done */
892                 ast_waitfor_n_fd(&timingfd, 1, &timeout, NULL);
893                 if (ast_timer_ack(timer, 1) < 0) {
894                         ast_log(LOG_ERROR, "Failed to acknowledge timer in softmix bridge.\n");
895                         ao2_lock(bridge);
896                         goto softmix_cleanup;
897                 }
898                 ao2_lock(bridge);
899
900                 /* make sure to detect mixing interval changes if they occur. */
901                 if (bridge->internal_mixing_interval && (bridge->internal_mixing_interval != softmix_data->internal_mixing_interval)) {
902                         softmix_data->internal_mixing_interval = bridge->internal_mixing_interval;
903                         ast_timer_set_rate(timer, (1000 / softmix_data->internal_mixing_interval));
904                         update_all_rates = 1; /* if the interval changes, the rates must be adjusted as well just to be notified new interval.*/
905                 }
906         }
907
908         res = 0;
909
910 softmix_cleanup:
911         softmix_translate_helper_destroy(&trans_helper);
912         softmix_mixing_array_destroy(&mixing_array);
913         if (softmix_data) {
914                 ao2_ref(softmix_data, -1);
915         }
916         return res;
917 }
918
919 static struct ast_bridge_technology softmix_bridge = {
920         .name = "softmix",
921         .capabilities = AST_BRIDGE_CAPABILITY_MULTIMIX | AST_BRIDGE_CAPABILITY_THREAD | AST_BRIDGE_CAPABILITY_MULTITHREADED | AST_BRIDGE_CAPABILITY_OPTIMIZE | AST_BRIDGE_CAPABILITY_VIDEO,
922         .preference = AST_BRIDGE_PREFERENCE_LOW,
923         .create = softmix_bridge_create,
924         .destroy = softmix_bridge_destroy,
925         .join = softmix_bridge_join,
926         .leave = softmix_bridge_leave,
927         .write = softmix_bridge_write,
928         .thread = softmix_bridge_thread,
929         .poke = softmix_bridge_poke,
930 };
931
932 static int unload_module(void)
933 {
934         ast_format_cap_destroy(softmix_bridge.format_capabilities);
935         return ast_bridge_technology_unregister(&softmix_bridge);
936 }
937
938 static int load_module(void)
939 {
940         struct ast_format tmp;
941         if (!(softmix_bridge.format_capabilities = ast_format_cap_alloc())) {
942                 return AST_MODULE_LOAD_DECLINE;
943         }
944         ast_format_cap_add(softmix_bridge.format_capabilities, ast_format_set(&tmp, AST_FORMAT_SLINEAR, 0));
945         return ast_bridge_technology_register(&softmix_bridge);
946 }
947
948 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Multi-party software based channel mixing");