git.sesse.net Git - nageru/blob - audio_mixer.cpp

   1 #include "audio_mixer.h"
   2
   3 #include <assert.h>
   4 #include <bmusb/bmusb.h>
   5 #include <endian.h>
   6 #include <math.h>
   7 #ifdef __SSE2__
   8 #include <immintrin.h>
   9 #endif
  10 #include <stdbool.h>
  11 #include <stdio.h>
  12 #include <stdlib.h>
  13 #include <string.h>
  14 #include <algorithm>
  15 #include <chrono>
  16 #include <cmath>
  17 #include <cstddef>
  18 #include <limits>
  19 #include <utility>
  20
  21 #include "db.h"
  22 #include "flags.h"
  23 #include "metrics.h"
  24 #include "state.pb.h"
  25 #include "timebase.h"
  26
  27 using namespace bmusb;
  28 using namespace std;
  29 using namespace std::chrono;
  30 using namespace std::placeholders;
  31
  32 namespace {
  33
  34 // TODO: If these prove to be a bottleneck, they can be SSSE3-optimized
  35 // (usually including multiple channels at a time).
  36
  37 void convert_fixed16_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
  38                              const uint8_t *src, size_t in_channel, size_t in_num_channels,
  39                              size_t num_samples)
  40 {
  41         assert(in_channel < in_num_channels);
  42         assert(out_channel < out_num_channels);
  43         src += in_channel * 2;
  44         dst += out_channel;
  45
  46         for (size_t i = 0; i < num_samples; ++i) {
  47                 int16_t s = le16toh(*(int16_t *)src);
  48                 *dst = s * (1.0f / 32768.0f);
  49
  50                 src += 2 * in_num_channels;
  51                 dst += out_num_channels;
  52         }
  53 }
  54
  55 void convert_fixed24_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
  56                              const uint8_t *src, size_t in_channel, size_t in_num_channels,
  57                              size_t num_samples)
  58 {
  59         assert(in_channel < in_num_channels);
  60         assert(out_channel < out_num_channels);
  61         src += in_channel * 3;
  62         dst += out_channel;
  63
  64         for (size_t i = 0; i < num_samples; ++i) {
  65                 uint32_t s1 = src[0];
  66                 uint32_t s2 = src[1];
  67                 uint32_t s3 = src[2];
  68                 uint32_t s = s1 | (s1 << 8) | (s2 << 16) | (s3 << 24);
  69                 *dst = int(s) * (1.0f / 2147483648.0f);
  70
  71                 src += 3 * in_num_channels;
  72                 dst += out_num_channels;
  73         }
  74 }
  75
  76 void convert_fixed32_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
  77                              const uint8_t *src, size_t in_channel, size_t in_num_channels,
  78                              size_t num_samples)
  79 {
  80         assert(in_channel < in_num_channels);
  81         assert(out_channel < out_num_channels);
  82         src += in_channel * 4;
  83         dst += out_channel;
  84
  85         for (size_t i = 0; i < num_samples; ++i) {
  86                 int32_t s = le32toh(*(int32_t *)src);
  87                 *dst = s * (1.0f / 2147483648.0f);
  88
  89                 src += 4 * in_num_channels;
  90                 dst += out_num_channels;
  91         }
  92 }
  93
  94 float find_peak_plain(const float *samples, size_t num_samples) __attribute__((unused));
  95
  96 float find_peak_plain(const float *samples, size_t num_samples)
  97 {
  98         float m = fabs(samples[0]);
  99         for (size_t i = 1; i < num_samples; ++i) {
 100                 m = max(m, fabs(samples[i]));
 101         }
 102         return m;
 103 }
 104
 105 #ifdef __SSE__
 106 static inline float horizontal_max(__m128 m)
 107 {
 108         __m128 tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 0, 3, 2));
 109         m = _mm_max_ps(m, tmp);
 110         tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 3, 0, 1));
 111         m = _mm_max_ps(m, tmp);
 112         return _mm_cvtss_f32(m);
 113 }
 114
 115 float find_peak(const float *samples, size_t num_samples)
 116 {
 117         const __m128 abs_mask = _mm_castsi128_ps(_mm_set1_epi32(0x7fffffffu));
 118         __m128 m = _mm_setzero_ps();
 119         for (size_t i = 0; i < (num_samples & ~3); i += 4) {
 120                 __m128 x = _mm_loadu_ps(samples + i);
 121                 x = _mm_and_ps(x, abs_mask);
 122                 m = _mm_max_ps(m, x);
 123         }
 124         float result = horizontal_max(m);
 125
 126         for (size_t i = (num_samples & ~3); i < num_samples; ++i) {
 127                 result = max(result, fabs(samples[i]));
 128         }
 129
 130 #if 0
 131         // Self-test. We should be bit-exact the same.
 132         float reference_result = find_peak_plain(samples, num_samples);
 133         if (result != reference_result) {
 134                 fprintf(stderr, "Error: Peak is %f [%f %f %f %f]; should be %f.\n",
 135                         result,
 136                         _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(0, 0, 0, 0))),
 137                         _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 1, 1, 1))),
 138                         _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 2, 2, 2))),
 139                         _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(3, 3, 3, 3))),
 140                         reference_result);
 141                 abort();
 142         }
 143 #endif
 144         return result;
 145 }
 146 #else
 147 float find_peak(const float *samples, size_t num_samples)
 148 {
 149         return find_peak_plain(samples, num_samples);
 150 }
 151 #endif
 152
 153 void deinterleave_samples(const vector<float> &in, vector<float> *out_l, vector<float> *out_r)
 154 {
 155         size_t num_samples = in.size() / 2;
 156         out_l->resize(num_samples);
 157         out_r->resize(num_samples);
 158
 159         const float *inptr = in.data();
 160         float *lptr = &(*out_l)[0];
 161         float *rptr = &(*out_r)[0];
 162         for (size_t i = 0; i < num_samples; ++i) {
 163                 *lptr++ = *inptr++;
 164                 *rptr++ = *inptr++;
 165         }
 166 }
 167
 168 }  // namespace
 169
 170 AudioMixer::AudioMixer(unsigned num_capture_cards, unsigned num_ffmpeg_inputs)
 171         : num_capture_cards(num_capture_cards),
 172           num_ffmpeg_inputs(num_ffmpeg_inputs),
 173           ffmpeg_inputs(new AudioDevice[num_ffmpeg_inputs]),
 174           limiter(OUTPUT_FREQUENCY),
 175           correlation(OUTPUT_FREQUENCY)
 176 {
 177         for (unsigned bus_index = 0; bus_index < MAX_BUSES; ++bus_index) {
 178                 locut[bus_index].init(FILTER_HPF, 2);
 179                 eq[bus_index][EQ_BAND_BASS].init(FILTER_LOW_SHELF, 1);
 180                 // Note: EQ_BAND_MID isn't used (see comments in apply_eq()).
 181                 eq[bus_index][EQ_BAND_TREBLE].init(FILTER_HIGH_SHELF, 1);
 182                 compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
 183                 level_compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
 184
 185                 set_bus_settings(bus_index, get_default_bus_settings());
 186         }
 187         set_limiter_enabled(global_flags.limiter_enabled);
 188         set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
 189
 190         r128.init(2, OUTPUT_FREQUENCY);
 191         r128.integr_start();
 192
 193         // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
 194         // and there's a limit to how important the peak meter is.
 195         peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
 196
 197         global_audio_mixer = this;
 198         alsa_pool.init();
 199
 200         if (!global_flags.input_mapping_filename.empty()) {
 201                 // Must happen after ALSAPool is initialized, as it needs to know the card list.
 202                 current_mapping_mode = MappingMode::MULTICHANNEL;
 203                 InputMapping new_input_mapping;
 204                 if (!load_input_mapping_from_file(get_devices(),
 205                                                   global_flags.input_mapping_filename,
 206                                                   &new_input_mapping)) {
 207                         fprintf(stderr, "Failed to load input mapping from '%s', exiting.\n",
 208                                 global_flags.input_mapping_filename.c_str());
 209                         exit(1);
 210                 }
 211                 set_input_mapping(new_input_mapping);
 212         } else {
 213                 set_simple_input(/*card_index=*/0);
 214                 if (global_flags.multichannel_mapping_mode) {
 215                         current_mapping_mode = MappingMode::MULTICHANNEL;
 216                 }
 217         }
 218
 219         global_metrics.add("audio_loudness_short_lufs", &metric_audio_loudness_short_lufs, Metrics::TYPE_GAUGE);
 220         global_metrics.add("audio_loudness_integrated_lufs", &metric_audio_loudness_integrated_lufs, Metrics::TYPE_GAUGE);
 221         global_metrics.add("audio_loudness_range_low_lufs", &metric_audio_loudness_range_low_lufs, Metrics::TYPE_GAUGE);
 222         global_metrics.add("audio_loudness_range_high_lufs", &metric_audio_loudness_range_high_lufs, Metrics::TYPE_GAUGE);
 223         global_metrics.add("audio_peak_dbfs", &metric_audio_peak_dbfs, Metrics::TYPE_GAUGE);
 224         global_metrics.add("audio_final_makeup_gain_db", &metric_audio_final_makeup_gain_db, Metrics::TYPE_GAUGE);
 225         global_metrics.add("audio_correlation", &metric_audio_correlation, Metrics::TYPE_GAUGE);
 226 }
 227
 228 void AudioMixer::reset_resampler(DeviceSpec device_spec)
 229 {
 230         lock_guard<timed_mutex> lock(audio_mutex);
 231         reset_resampler_mutex_held(device_spec);
 232 }
 233
 234 void AudioMixer::reset_resampler_mutex_held(DeviceSpec device_spec)
 235 {
 236         AudioDevice *device = find_audio_device(device_spec);
 237
 238         if (device->interesting_channels.empty()) {
 239                 device->resampling_queue.reset();
 240         } else {
 241                 device->resampling_queue.reset(new ResamplingQueue(
 242                         device_spec, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size(),
 243                         global_flags.audio_queue_length_ms * 0.001));
 244         }
 245 }
 246
 247 bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length, steady_clock::time_point frame_time)
 248 {
 249         AudioDevice *device = find_audio_device(device_spec);
 250
 251         unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
 252         if (!lock.try_lock_for(chrono::milliseconds(10))) {
 253                 return false;
 254         }
 255         if (device->resampling_queue == nullptr) {
 256                 // No buses use this device; throw it away.
 257                 return true;
 258         }
 259
 260         unsigned num_channels = device->interesting_channels.size();
 261         assert(num_channels > 0);
 262
 263         // Convert the audio to fp32.
 264         unique_ptr<float[]> audio(new float[num_samples * num_channels]);
 265         unsigned channel_index = 0;
 266         for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) {
 267                 switch (audio_format.bits_per_sample) {
 268                 case 0:
 269                         assert(num_samples == 0);
 270                         break;
 271                 case 16:
 272                         convert_fixed16_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
 273                         break;
 274                 case 24:
 275                         convert_fixed24_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
 276                         break;
 277                 case 32:
 278                         convert_fixed32_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
 279                         break;
 280                 default:
 281                         fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
 282                         assert(false);
 283                 }
 284         }
 285
 286         // If we changed frequency since last frame, we'll need to reset the resampler.
 287         if (audio_format.sample_rate != device->capture_frequency) {
 288                 device->capture_frequency = audio_format.sample_rate;
 289                 reset_resampler_mutex_held(device_spec);
 290         }
 291
 292         // Now add it.
 293         device->resampling_queue->add_input_samples(frame_time, audio.get(), num_samples, ResamplingQueue::ADJUST_RATE);
 294         return true;
 295 }
 296
 297 bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length)
 298 {
 299         AudioDevice *device = find_audio_device(device_spec);
 300
 301         unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
 302         if (!lock.try_lock_for(chrono::milliseconds(10))) {
 303                 return false;
 304         }
 305         if (device->resampling_queue == nullptr) {
 306                 // No buses use this device; throw it away.
 307                 return true;
 308         }
 309
 310         unsigned num_channels = device->interesting_channels.size();
 311         assert(num_channels > 0);
 312
 313         vector<float> silence(samples_per_frame * num_channels, 0.0f);
 314         for (unsigned i = 0; i < num_frames; ++i) {
 315                 device->resampling_queue->add_input_samples(steady_clock::now(), silence.data(), samples_per_frame, ResamplingQueue::DO_NOT_ADJUST_RATE);
 316         }
 317         return true;
 318 }
 319
 320 bool AudioMixer::silence_card(DeviceSpec device_spec, bool silence)
 321 {
 322         AudioDevice *device = find_audio_device(device_spec);
 323
 324         unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
 325         if (!lock.try_lock_for(chrono::milliseconds(10))) {
 326                 return false;
 327         }
 328
 329         if (device->silenced && !silence) {
 330                 reset_resampler_mutex_held(device_spec);
 331         }
 332         device->silenced = silence;
 333         return true;
 334 }
 335
 336 AudioMixer::BusSettings AudioMixer::get_default_bus_settings()
 337 {
 338         BusSettings settings;
 339         settings.fader_volume_db = 0.0f;
 340         settings.muted = false;
 341         settings.locut_enabled = global_flags.locut_enabled;
 342         settings.stereo_width = 1.0f;
 343         for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
 344                 settings.eq_level_db[band_index] = 0.0f;
 345         }
 346         settings.gain_staging_db = global_flags.initial_gain_staging_db;
 347         settings.level_compressor_enabled = global_flags.gain_staging_auto;
 348         settings.compressor_threshold_dbfs = ref_level_dbfs - 12.0f;  // -12 dB.
 349         settings.compressor_enabled = global_flags.compressor_enabled;
 350         return settings;
 351 }
 352
 353 AudioMixer::BusSettings AudioMixer::get_bus_settings(unsigned bus_index) const
 354 {
 355         lock_guard<timed_mutex> lock(audio_mutex);
 356         BusSettings settings;
 357         settings.fader_volume_db = fader_volume_db[bus_index];
 358         settings.muted = mute[bus_index];
 359         settings.locut_enabled = locut_enabled[bus_index];
 360         settings.stereo_width = stereo_width[bus_index];
 361         for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
 362                 settings.eq_level_db[band_index] = eq_level_db[bus_index][band_index];
 363         }
 364         settings.gain_staging_db = gain_staging_db[bus_index];
 365         settings.level_compressor_enabled = level_compressor_enabled[bus_index];
 366         settings.compressor_threshold_dbfs = compressor_threshold_dbfs[bus_index];
 367         settings.compressor_enabled = compressor_enabled[bus_index];
 368         return settings;
 369 }
 370
 371 void AudioMixer::set_bus_settings(unsigned bus_index, const AudioMixer::BusSettings &settings)
 372 {
 373         lock_guard<timed_mutex> lock(audio_mutex);
 374         fader_volume_db[bus_index] = settings.fader_volume_db;
 375         mute[bus_index] = settings.muted;
 376         locut_enabled[bus_index] = settings.locut_enabled;
 377         stereo_width[bus_index] = settings.stereo_width;
 378         for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
 379                 eq_level_db[bus_index][band_index] = settings.eq_level_db[band_index];
 380         }
 381         gain_staging_db[bus_index] = settings.gain_staging_db;
 382         last_gain_staging_db[bus_index] = gain_staging_db[bus_index];
 383         level_compressor_enabled[bus_index] = settings.level_compressor_enabled;
 384         compressor_threshold_dbfs[bus_index] = settings.compressor_threshold_dbfs;
 385         compressor_enabled[bus_index] = settings.compressor_enabled;
 386 }
 387
 388 AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
 389 {
 390         switch (device.type) {
 391         case InputSourceType::CAPTURE_CARD:
 392                 return &video_cards[device.index];
 393         case InputSourceType::ALSA_INPUT:
 394                 return &alsa_inputs[device.index];
 395         case InputSourceType::FFMPEG_VIDEO_INPUT:
 396                 return &ffmpeg_inputs[device.index];
 397         case InputSourceType::SILENCE:
 398         default:
 399                 assert(false);
 400         }
 401         return nullptr;
 402 }
 403
 404 // Get a pointer to the given channel from the given device.
 405 // The channel must be picked out earlier and resampled.
 406 void AudioMixer::find_sample_src_from_device(const map<DeviceSpec, vector<float>> &samples_card, DeviceSpec device_spec, int source_channel, const float **srcptr, unsigned *stride)
 407 {
 408         static float zero = 0.0f;
 409         if (source_channel == -1 || device_spec.type == InputSourceType::SILENCE) {
 410                 *srcptr = &zero;
 411                 *stride = 0;
 412                 return;
 413         }
 414         AudioDevice *device = find_audio_device(device_spec);
 415         assert(device->interesting_channels.count(source_channel) != 0);
 416         unsigned channel_index = 0;
 417         for (int channel : device->interesting_channels) {
 418                 if (channel == source_channel) break;
 419                 ++channel_index;
 420         }
 421         assert(channel_index < device->interesting_channels.size());
 422         const auto it = samples_card.find(device_spec);
 423         assert(it != samples_card.end());
 424         *srcptr = &(it->second)[channel_index];
 425         *stride = device->interesting_channels.size();
 426 }
 427
 428 // TODO: Can be SSSE3-optimized if need be.
 429 void AudioMixer::fill_audio_bus(const map<DeviceSpec, vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float stereo_width, float *output)
 430 {
 431         if (bus.device.type == InputSourceType::SILENCE) {
 432                 memset(output, 0, num_samples * 2 * sizeof(*output));
 433         } else {
 434                 assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
 435                        bus.device.type == InputSourceType::ALSA_INPUT ||
 436                        bus.device.type == InputSourceType::FFMPEG_VIDEO_INPUT);
 437                 const float *lsrc, *rsrc;
 438                 unsigned lstride, rstride;
 439                 float *dptr = output;
 440                 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[0], &lsrc, &lstride);
 441                 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[1], &rsrc, &rstride);
 442
 443                 // Apply stereo width settings. Set stereo width w to a 0..1 range instead of
 444                 // -1..1, since it makes for much easier calculations (so 0.5 = completely mono).
 445                 // Then, what we want is
 446                 //
 447                 //   L' = wL + (1-w)R = R + w(L-R)
 448                 //   R' = wR + (1-w)L = L + w(R-L)
 449                 //
 450                 // This can be further simplified calculation-wise by defining the weighted
 451                 // difference signal D = w(R-L), so that:
 452                 //
 453                 //   L' = R - D
 454                 //   R' = L + D
 455                 float w = 0.5f * stereo_width + 0.5f;
 456                 if (fabs(w) < 1e-3) {
 457                         // Perfect inverse.
 458                         swap(lsrc, rsrc);
 459                         swap(lstride, rstride);
 460                         w = 1.0f;
 461                 }
 462                 if (fabs(w - 1.0f) < 1e-3) {
 463                         // No calculations needed for stereo_width = 1.
 464                         for (unsigned i = 0; i < num_samples; ++i) {
 465                                 *dptr++ = *lsrc;
 466                                 *dptr++ = *rsrc;
 467                                 lsrc += lstride;
 468                                 rsrc += rstride;
 469                         }
 470                 } else {
 471                         // General case.
 472                         for (unsigned i = 0; i < num_samples; ++i) {
 473                                 float left = *lsrc, right = *rsrc;
 474                                 float diff = w * (right - left);
 475                                 *dptr++ = right - diff;
 476                                 *dptr++ = left + diff;
 477                                 lsrc += lstride;
 478                                 rsrc += rstride;
 479                         }
 480                 }
 481         }
 482 }
 483
 484 vector<DeviceSpec> AudioMixer::get_active_devices() const
 485 {
 486         vector<DeviceSpec> ret;
 487         for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
 488                 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
 489                 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
 490                         ret.push_back(device_spec);
 491                 }
 492         }
 493         for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
 494                 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
 495                 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
 496                         ret.push_back(device_spec);
 497                 }
 498         }
 499         for (unsigned card_index = 0; card_index < num_ffmpeg_inputs; ++card_index) {
 500                 const DeviceSpec device_spec{InputSourceType::FFMPEG_VIDEO_INPUT, card_index};
 501                 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
 502                         ret.push_back(device_spec);
 503                 }
 504         }
 505         return ret;
 506 }
 507
 508 namespace {
 509
 510 void apply_gain(float db, float last_db, vector<float> *samples)
 511 {
 512         if (fabs(db - last_db) < 1e-3) {
 513                 // Constant over this frame.
 514                 const float gain = from_db(db);
 515                 for (size_t i = 0; i < samples->size(); ++i) {
 516                         (*samples)[i] *= gain;
 517                 }
 518         } else {
 519                 // We need to do a fade.
 520                 unsigned num_samples = samples->size() / 2;
 521                 float gain = from_db(last_db);
 522                 const float gain_inc = pow(from_db(db - last_db), 1.0 / num_samples);
 523                 for (size_t i = 0; i < num_samples; ++i) {
 524                         (*samples)[i * 2 + 0] *= gain;
 525                         (*samples)[i * 2 + 1] *= gain;
 526                         gain *= gain_inc;
 527                 }
 528         }
 529 }
 530
 531 }  // namespace
 532
 533 vector<float> AudioMixer::get_output(steady_clock::time_point ts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
 534 {
 535         map<DeviceSpec, vector<float>> samples_card;
 536         vector<float> samples_bus;
 537
 538         lock_guard<timed_mutex> lock(audio_mutex);
 539
 540         // Pick out all the interesting channels from all the cards.
 541         for (const DeviceSpec &device_spec : get_active_devices()) {
 542                 AudioDevice *device = find_audio_device(device_spec);
 543                 samples_card[device_spec].resize(num_samples * device->interesting_channels.size());
 544                 if (device->silenced) {
 545                         memset(&samples_card[device_spec][0], 0, samples_card[device_spec].size() * sizeof(float));
 546                 } else {
 547                         device->resampling_queue->get_output_samples(
 548                                 ts,
 549                                 &samples_card[device_spec][0],
 550                                 num_samples,
 551                                 rate_adjustment_policy);
 552                 }
 553         }
 554
 555         vector<float> samples_out, left, right;
 556         samples_out.resize(num_samples * 2);
 557         samples_bus.resize(num_samples * 2);
 558         for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
 559                 fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, stereo_width[bus_index], &samples_bus[0]);
 560                 apply_eq(bus_index, &samples_bus);
 561
 562                 {
 563                         lock_guard<mutex> lock(compressor_mutex);
 564
 565                         // Apply a level compressor to get the general level right.
 566                         // Basically, if it's over about -40 dBFS, we squeeze it down to that level
 567                         // (or more precisely, near it, since we don't use infinite ratio),
 568                         // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
 569                         // entirely arbitrary, but from practical tests with speech, it seems to
 570                         // put ut around -23 LUFS, so it's a reasonable starting point for later use.
 571                         if (level_compressor_enabled[bus_index]) {
 572                                 float threshold = 0.01f;   // -40 dBFS.
 573                                 float ratio = 20.0f;
 574                                 float attack_time = 0.5f;
 575                                 float release_time = 20.0f;
 576                                 float makeup_gain = from_db(ref_level_dbfs - (-40.0f));  // +26 dB.
 577                                 level_compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
 578                                 gain_staging_db[bus_index] = to_db(level_compressor[bus_index]->get_attenuation() * makeup_gain);
 579                         } else {
 580                                 // Just apply the gain we already had.
 581                                 float db = gain_staging_db[bus_index];
 582                                 float last_db = last_gain_staging_db[bus_index];
 583                                 apply_gain(db, last_db, &samples_bus);
 584                         }
 585                         last_gain_staging_db[bus_index] = gain_staging_db[bus_index];
 586
 587 #if 0
 588                         printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
 589                                 level_compressor.get_level(), to_db(level_compressor.get_level()),
 590                                 level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()),
 591                                 to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
 592 #endif
 593
 594                         // The real compressor.
 595                         if (compressor_enabled[bus_index]) {
 596                                 float threshold = from_db(compressor_threshold_dbfs[bus_index]);
 597                                 float ratio = 20.0f;
 598                                 float attack_time = 0.005f;
 599                                 float release_time = 0.040f;
 600                                 float makeup_gain = 2.0f;  // +6 dB.
 601                                 compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
 602                 //              compressor_att = compressor.get_attenuation();
 603                         }
 604                 }
 605
 606                 add_bus_to_master(bus_index, samples_bus, &samples_out);
 607                 deinterleave_samples(samples_bus, &left, &right);
 608                 measure_bus_levels(bus_index, left, right);
 609         }
 610
 611         {
 612                 lock_guard<mutex> lock(compressor_mutex);
 613
 614                 // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
 615                 // Note that since ratio is not infinite, we could go slightly higher than this.
 616                 if (limiter_enabled) {
 617                         float threshold = from_db(limiter_threshold_dbfs);
 618                         float ratio = 30.0f;
 619                         float attack_time = 0.0f;  // Instant.
 620                         float release_time = 0.020f;
 621                         float makeup_gain = 1.0f;  // 0 dB.
 622                         limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
 623         //              limiter_att = limiter.get_attenuation();
 624                 }
 625
 626         //      printf("limiter=%+5.1f  compressor=%+5.1f\n", to_db(limiter_att), to_db(compressor_att));
 627         }
 628
 629         // At this point, we are most likely close to +0 LU (at least if the
 630         // faders sum to 0 dB and the compressors are on), but all of our
 631         // measurements have been on raw sample values, not R128 values.
 632         // So we have a final makeup gain to get us to +0 LU; the gain
 633         // adjustments required should be relatively small, and also, the
 634         // offset shouldn't change much (only if the type of audio changes
 635         // significantly). Thus, we shoot for updating this value basically
 636         // “whenever we process buffers”, since the R128 calculation isn't exactly
 637         // something we get out per-sample.
 638         //
 639         // Note that there's a feedback loop here, so we choose a very slow filter
 640         // (half-time of 30 seconds).
 641         double target_loudness_factor, alpha;
 642         double loudness_lu = r128.loudness_M() - ref_level_lufs;
 643         target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
 644
 645         // If we're outside +/- 5 LU (after correction), we don't count it as
 646         // a normal signal (probably silence) and don't change the
 647         // correction factor; just apply what we already have.
 648         if (fabs(loudness_lu) >= 5.0 || !final_makeup_gain_auto) {
 649                 alpha = 0.0;
 650         } else {
 651                 // Formula adapted from
 652                 // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
 653                 const double half_time_s = 30.0;
 654                 const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
 655                 alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
 656         }
 657
 658         {
 659                 lock_guard<mutex> lock(compressor_mutex);
 660                 double m = final_makeup_gain;
 661                 for (size_t i = 0; i < samples_out.size(); i += 2) {
 662                         samples_out[i + 0] *= m;
 663                         samples_out[i + 1] *= m;
 664                         m += (target_loudness_factor - m) * alpha;
 665                 }
 666                 final_makeup_gain = m;
 667         }
 668
 669         update_meters(samples_out);
 670
 671         return samples_out;
 672 }
 673
 674 namespace {
 675
 676 void apply_filter_fade(StereoFilter *filter, float *data, unsigned num_samples, float cutoff_hz, float db, float last_db)
 677 {
 678         // A granularity of 32 samples is an okay tradeoff between speed and
 679         // smoothness; recalculating the filters is pretty expensive, so it's
 680         // good that we don't do this all the time.
 681         static constexpr unsigned filter_granularity_samples = 32;
 682
 683         const float cutoff_linear = cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY;
 684         if (fabs(db - last_db) < 1e-3) {
 685                 // Constant over this frame.
 686                 if (fabs(db) > 0.01f) {
 687                         filter->render(data, num_samples, cutoff_linear, 0.5f, db / 40.0f);
 688                 }
 689         } else {
 690                 // We need to do a fade. (Rounding up avoids division by zero.)
 691                 unsigned num_blocks = (num_samples + filter_granularity_samples - 1) / filter_granularity_samples;
 692                 const float inc_db_norm = (db - last_db) / 40.0f / num_blocks;
 693                 float db_norm = db / 40.0f;
 694                 for (size_t i = 0; i < num_samples; i += filter_granularity_samples) {
 695                         size_t samples_this_block = std::min<size_t>(num_samples - i, filter_granularity_samples);
 696                         filter->render(data + i * 2, samples_this_block, cutoff_linear, 0.5f, db_norm);
 697                         db_norm += inc_db_norm;
 698                 }
 699         }
 700 }
 701
 702 }  // namespace
 703
 704 void AudioMixer::apply_eq(unsigned bus_index, vector<float> *samples_bus)
 705 {
 706         constexpr float bass_freq_hz = 200.0f;
 707         constexpr float treble_freq_hz = 4700.0f;
 708
 709         // Cut away everything under 120 Hz (or whatever the cutoff is);
 710         // we don't need it for voice, and it will reduce headroom
 711         // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
 712         // should be dampened.)
 713         if (locut_enabled[bus_index]) {
 714                 locut[bus_index].render(samples_bus->data(), samples_bus->size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
 715         }
 716
 717         // Apply the rest of the EQ. Since we only have a simple three-band EQ,
 718         // we can implement it with two shelf filters. We use a simple gain to
 719         // set the mid-level filter, and then offset the low and high bands
 720         // from that if we need to. (We could perhaps have folded the gain into
 721         // the next part, but it's so cheap that the trouble isn't worth it.)
 722         //
 723         // If any part of the EQ has changed appreciably since last frame,
 724         // we fade smoothly during the course of this frame.
 725         const float bass_db = eq_level_db[bus_index][EQ_BAND_BASS];
 726         const float mid_db = eq_level_db[bus_index][EQ_BAND_MID];
 727         const float treble_db = eq_level_db[bus_index][EQ_BAND_TREBLE];
 728
 729         const float last_bass_db = last_eq_level_db[bus_index][EQ_BAND_BASS];
 730         const float last_mid_db = last_eq_level_db[bus_index][EQ_BAND_MID];
 731         const float last_treble_db = last_eq_level_db[bus_index][EQ_BAND_TREBLE];
 732
 733         assert(samples_bus->size() % 2 == 0);
 734         const unsigned num_samples = samples_bus->size() / 2;
 735
 736         apply_gain(mid_db, last_mid_db, samples_bus);
 737
 738         apply_filter_fade(&eq[bus_index][EQ_BAND_BASS], samples_bus->data(), num_samples, bass_freq_hz, bass_db - mid_db, last_bass_db - last_mid_db);
 739         apply_filter_fade(&eq[bus_index][EQ_BAND_TREBLE], samples_bus->data(), num_samples, treble_freq_hz, treble_db - mid_db, last_treble_db - last_mid_db);
 740
 741         last_eq_level_db[bus_index][EQ_BAND_BASS] = bass_db;
 742         last_eq_level_db[bus_index][EQ_BAND_MID] = mid_db;
 743         last_eq_level_db[bus_index][EQ_BAND_TREBLE] = treble_db;
 744 }
 745
 746 void AudioMixer::add_bus_to_master(unsigned bus_index, const vector<float> &samples_bus, vector<float> *samples_out)
 747 {
 748         assert(samples_bus.size() == samples_out->size());
 749         assert(samples_bus.size() % 2 == 0);
 750         unsigned num_samples = samples_bus.size() / 2;
 751         const float new_volume_db = mute[bus_index] ? -90.0f : fader_volume_db[bus_index].load();
 752         if (fabs(new_volume_db - last_fader_volume_db[bus_index]) > 1e-3) {
 753                 // The volume has changed; do a fade over the course of this frame.
 754                 // (We might have some numerical issues here, but it seems to sound OK.)
 755                 // For the purpose of fading here, the silence floor is set to -90 dB
 756                 // (the fader only goes to -84).
 757                 float old_volume = from_db(max<float>(last_fader_volume_db[bus_index], -90.0f));
 758                 float volume = from_db(max<float>(new_volume_db, -90.0f));
 759
 760                 float volume_inc = pow(volume / old_volume, 1.0 / num_samples);
 761                 volume = old_volume;
 762                 if (bus_index == 0) {
 763                         for (unsigned i = 0; i < num_samples; ++i) {
 764                                 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
 765                                 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
 766                                 volume *= volume_inc;
 767                         }
 768                 } else {
 769                         for (unsigned i = 0; i < num_samples; ++i) {
 770                                 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
 771                                 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
 772                                 volume *= volume_inc;
 773                         }
 774                 }
 775         } else if (new_volume_db > -90.0f) {
 776                 float volume = from_db(new_volume_db);
 777                 if (bus_index == 0) {
 778                         for (unsigned i = 0; i < num_samples; ++i) {
 779                                 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
 780                                 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
 781                         }
 782                 } else {
 783                         for (unsigned i = 0; i < num_samples; ++i) {
 784                                 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
 785                                 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
 786                         }
 787                 }
 788         }
 789
 790         last_fader_volume_db[bus_index] = new_volume_db;
 791 }
 792
 793 void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right)
 794 {
 795         assert(left.size() == right.size());
 796         const float volume = mute[bus_index] ? 0.0f : from_db(fader_volume_db[bus_index]);
 797         const float peak_levels[2] = {
 798                 find_peak(left.data(), left.size()) * volume,
 799                 find_peak(right.data(), right.size()) * volume
 800         };
 801         for (unsigned channel = 0; channel < 2; ++channel) {
 802                 // Compute the current value, including hold and falloff.
 803                 // The constants are borrowed from zita-mu1 by Fons Adriaensen.
 804                 static constexpr float hold_sec = 0.5f;
 805                 static constexpr float falloff_db_sec = 15.0f;  // dB/sec falloff after hold.
 806                 float current_peak;
 807                 PeakHistory &history = peak_history[bus_index][channel];
 808                 history.historic_peak = max(history.historic_peak, peak_levels[channel]);
 809                 if (history.age_seconds < hold_sec) {
 810                         current_peak = history.last_peak;
 811                 } else {
 812                         current_peak = history.last_peak * from_db(-falloff_db_sec * (history.age_seconds - hold_sec));
 813                 }
 814
 815                 // See if we have a new peak to replace the old (possibly falling) one.
 816                 if (peak_levels[channel] > current_peak) {
 817                         history.last_peak = peak_levels[channel];
 818                         history.age_seconds = 0.0f;  // Not 100% correct, but more than good enough given our frame sizes.
 819                         current_peak = peak_levels[channel];
 820                 } else {
 821                         history.age_seconds += float(left.size()) / OUTPUT_FREQUENCY;
 822                 }
 823                 history.current_level = peak_levels[channel];
 824                 history.current_peak = current_peak;
 825         }
 826 }
 827
 828 void AudioMixer::update_meters(const vector<float> &samples)
 829 {
 830         // Upsample 4x to find interpolated peak.
 831         peak_resampler.inp_data = const_cast<float *>(samples.data());
 832         peak_resampler.inp_count = samples.size() / 2;
 833
 834         vector<float> interpolated_samples;
 835         interpolated_samples.resize(samples.size());
 836         {
 837                 lock_guard<mutex> lock(audio_measure_mutex);
 838
 839                 while (peak_resampler.inp_count > 0) {  // About four iterations.
 840                         peak_resampler.out_data = &interpolated_samples[0];
 841                         peak_resampler.out_count = interpolated_samples.size() / 2;
 842                         peak_resampler.process();
 843                         size_t out_stereo_samples = interpolated_samples.size() / 2 - peak_resampler.out_count;
 844                         peak = max<float>(peak, find_peak(interpolated_samples.data(), out_stereo_samples * 2));
 845                         peak_resampler.out_data = nullptr;
 846                 }
 847         }
 848
 849         // Find R128 levels and L/R correlation.
 850         vector<float> left, right;
 851         deinterleave_samples(samples, &left, &right);
 852         float *ptrs[] = { left.data(), right.data() };
 853         {
 854                 lock_guard<mutex> lock(audio_measure_mutex);
 855                 r128.process(left.size(), ptrs);
 856                 correlation.process_samples(samples);
 857         }
 858
 859         send_audio_level_callback();
 860 }
 861
 862 void AudioMixer::reset_meters()
 863 {
 864         lock_guard<mutex> lock(audio_measure_mutex);
 865         peak_resampler.reset();
 866         peak = 0.0f;
 867         r128.reset();
 868         r128.integr_start();
 869         correlation.reset();
 870 }
 871
 872 void AudioMixer::send_audio_level_callback()
 873 {
 874         if (audio_level_callback == nullptr) {
 875                 return;
 876         }
 877
 878         lock_guard<mutex> lock(audio_measure_mutex);
 879         double loudness_s = r128.loudness_S();
 880         double loudness_i = r128.integrated();
 881         double loudness_range_low = r128.range_min();
 882         double loudness_range_high = r128.range_max();
 883
 884         metric_audio_loudness_short_lufs = loudness_s;
 885         metric_audio_loudness_integrated_lufs = loudness_i;
 886         metric_audio_loudness_range_low_lufs = loudness_range_low;
 887         metric_audio_loudness_range_high_lufs = loudness_range_high;
 888         metric_audio_peak_dbfs = to_db(peak);
 889         metric_audio_final_makeup_gain_db = to_db(final_makeup_gain);
 890         metric_audio_correlation = correlation.get_correlation();
 891
 892         vector<BusLevel> bus_levels;
 893         bus_levels.resize(input_mapping.buses.size());
 894         {
 895                 lock_guard<mutex> lock(compressor_mutex);
 896                 for (unsigned bus_index = 0; bus_index < bus_levels.size(); ++bus_index) {
 897                         BusLevel &levels = bus_levels[bus_index];
 898                         BusMetrics &metrics = bus_metrics[bus_index];
 899
 900                         levels.current_level_dbfs[0] = metrics.current_level_dbfs[0] = to_db(peak_history[bus_index][0].current_level);
 901                         levels.current_level_dbfs[1] = metrics.current_level_dbfs[1] = to_db(peak_history[bus_index][1].current_level);
 902                         levels.peak_level_dbfs[0] = metrics.peak_level_dbfs[0] = to_db(peak_history[bus_index][0].current_peak);
 903                         levels.peak_level_dbfs[1] = metrics.peak_level_dbfs[1] = to_db(peak_history[bus_index][1].current_peak);
 904                         levels.historic_peak_dbfs = metrics.historic_peak_dbfs = to_db(
 905                                 max(peak_history[bus_index][0].historic_peak,
 906                                     peak_history[bus_index][1].historic_peak));
 907                         levels.gain_staging_db = metrics.gain_staging_db = gain_staging_db[bus_index];
 908                         if (compressor_enabled[bus_index]) {
 909                                 levels.compressor_attenuation_db = metrics.compressor_attenuation_db = -to_db(compressor[bus_index]->get_attenuation());
 910                         } else {
 911                                 levels.compressor_attenuation_db = 0.0;
 912                                 metrics.compressor_attenuation_db = 0.0 / 0.0;
 913                         }
 914                 }
 915         }
 916
 917         audio_level_callback(loudness_s, to_db(peak), bus_levels,
 918                 loudness_i, loudness_range_low, loudness_range_high,
 919                 to_db(final_makeup_gain),
 920                 correlation.get_correlation());
 921 }
 922
 923 map<DeviceSpec, DeviceInfo> AudioMixer::get_devices()
 924 {
 925         lock_guard<timed_mutex> lock(audio_mutex);
 926
 927         map<DeviceSpec, DeviceInfo> devices;
 928         for (unsigned card_index = 0; card_index < num_capture_cards; ++card_index) {
 929                 const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index };
 930                 const AudioDevice *device = &video_cards[card_index];
 931                 DeviceInfo info;
 932                 info.display_name = device->display_name;
 933                 info.num_channels = 8;
 934                 devices.insert(make_pair(spec, info));
 935         }
 936         vector<ALSAPool::Device> available_alsa_devices = alsa_pool.get_devices();
 937         for (unsigned card_index = 0; card_index < available_alsa_devices.size(); ++card_index) {
 938                 const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index };
 939                 const ALSAPool::Device &device = available_alsa_devices[card_index];
 940                 DeviceInfo info;
 941                 info.display_name = device.display_name();
 942                 info.num_channels = device.num_channels;
 943                 info.alsa_name = device.name;
 944                 info.alsa_info = device.info;
 945                 info.alsa_address = device.address;
 946                 devices.insert(make_pair(spec, info));
 947         }
 948         for (unsigned card_index = 0; card_index < num_ffmpeg_inputs; ++card_index) {
 949                 const DeviceSpec spec{ InputSourceType::FFMPEG_VIDEO_INPUT, card_index };
 950                 const AudioDevice *device = &ffmpeg_inputs[card_index];
 951                 DeviceInfo info;
 952                 info.display_name = device->display_name;
 953                 info.num_channels = 2;
 954                 devices.insert(make_pair(spec, info));
 955         }
 956         return devices;
 957 }
 958
 959 void AudioMixer::set_display_name(DeviceSpec device_spec, const string &name)
 960 {
 961         AudioDevice *device = find_audio_device(device_spec);
 962
 963         lock_guard<timed_mutex> lock(audio_mutex);
 964         device->display_name = name;
 965 }
 966
 967 void AudioMixer::serialize_device(DeviceSpec device_spec, DeviceSpecProto *device_spec_proto)
 968 {
 969         lock_guard<timed_mutex> lock(audio_mutex);
 970         switch (device_spec.type) {
 971                 case InputSourceType::SILENCE:
 972                         device_spec_proto->set_type(DeviceSpecProto::SILENCE);
 973                         break;
 974                 case InputSourceType::CAPTURE_CARD:
 975                         device_spec_proto->set_type(DeviceSpecProto::CAPTURE_CARD);
 976                         device_spec_proto->set_index(device_spec.index);
 977                         device_spec_proto->set_display_name(video_cards[device_spec.index].display_name);
 978                         break;
 979                 case InputSourceType::ALSA_INPUT:
 980                         alsa_pool.serialize_device(device_spec.index, device_spec_proto);
 981                         break;
 982                 case InputSourceType::FFMPEG_VIDEO_INPUT:
 983                         device_spec_proto->set_type(DeviceSpecProto::FFMPEG_VIDEO_INPUT);
 984                         device_spec_proto->set_index(device_spec.index);
 985                         device_spec_proto->set_display_name(ffmpeg_inputs[device_spec.index].display_name);
 986                         break;
 987         }
 988 }
 989
 990 void AudioMixer::set_simple_input(unsigned card_index)
 991 {
 992         assert(card_index < num_capture_cards + num_ffmpeg_inputs);
 993         InputMapping new_input_mapping;
 994         InputMapping::Bus input;
 995         input.name = "Main";
 996         if (card_index >= num_capture_cards) {
 997                 input.device = DeviceSpec{InputSourceType::FFMPEG_VIDEO_INPUT, card_index - num_capture_cards};
 998         } else {
 999                 input.device = DeviceSpec{InputSourceType::CAPTURE_CARD, card_index};
1000         }
1001         input.source_channel[0] = 0;
1002         input.source_channel[1] = 1;
1003
1004         new_input_mapping.buses.push_back(input);
1005
1006         lock_guard<timed_mutex> lock(audio_mutex);
1007         current_mapping_mode = MappingMode::SIMPLE;
1008         set_input_mapping_lock_held(new_input_mapping);
1009         fader_volume_db[0] = 0.0f;
1010 }
1011
1012 unsigned AudioMixer::get_simple_input() const
1013 {
1014         lock_guard<timed_mutex> lock(audio_mutex);
1015         if (input_mapping.buses.size() == 1 &&
1016             input_mapping.buses[0].device.type == InputSourceType::CAPTURE_CARD &&
1017             input_mapping.buses[0].source_channel[0] == 0 &&
1018             input_mapping.buses[0].source_channel[1] == 1) {
1019                 return input_mapping.buses[0].device.index;
1020         } else if (input_mapping.buses.size() == 1 &&
1021                    input_mapping.buses[0].device.type == InputSourceType::FFMPEG_VIDEO_INPUT &&
1022                    input_mapping.buses[0].source_channel[0] == 0 &&
1023                    input_mapping.buses[0].source_channel[1] == 1) {
1024                 return input_mapping.buses[0].device.index + num_capture_cards;
1025         } else {
1026                 return numeric_limits<unsigned>::max();
1027         }
1028 }
1029
1030 void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
1031 {
1032         lock_guard<timed_mutex> lock(audio_mutex);
1033         set_input_mapping_lock_held(new_input_mapping);
1034         current_mapping_mode = MappingMode::MULTICHANNEL;
1035 }
1036
1037 AudioMixer::MappingMode AudioMixer::get_mapping_mode() const
1038 {
1039         lock_guard<timed_mutex> lock(audio_mutex);
1040         return current_mapping_mode;
1041 }
1042
1043 void AudioMixer::set_input_mapping_lock_held(const InputMapping &new_input_mapping)
1044 {
1045         map<DeviceSpec, set<unsigned>> interesting_channels;
1046         for (const InputMapping::Bus &bus : new_input_mapping.buses) {
1047                 if (bus.device.type == InputSourceType::CAPTURE_CARD ||
1048                     bus.device.type == InputSourceType::ALSA_INPUT ||
1049                     bus.device.type == InputSourceType::FFMPEG_VIDEO_INPUT) {
1050                         for (unsigned channel = 0; channel < 2; ++channel) {
1051                                 if (bus.source_channel[channel] != -1) {
1052                                         interesting_channels[bus.device].insert(bus.source_channel[channel]);
1053                                 }
1054                         }
1055                 } else {
1056                         assert(bus.device.type == InputSourceType::SILENCE);
1057                 }
1058         }
1059
1060         // Kill all the old metrics, and set up new ones.
1061         for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
1062                 BusMetrics &metrics = bus_metrics[bus_index];
1063
1064                 vector<pair<string, string>> labels_left = metrics.labels;
1065                 labels_left.emplace_back("channel", "left");
1066                 vector<pair<string, string>> labels_right = metrics.labels;
1067                 labels_right.emplace_back("channel", "right");
1068
1069                 global_metrics.remove("bus_current_level_dbfs", labels_left);
1070                 global_metrics.remove("bus_current_level_dbfs", labels_right);
1071                 global_metrics.remove("bus_peak_level_dbfs", labels_left);
1072                 global_metrics.remove("bus_peak_level_dbfs", labels_right);
1073                 global_metrics.remove("bus_historic_peak_dbfs", metrics.labels);
1074                 global_metrics.remove("bus_gain_staging_db", metrics.labels);
1075                 global_metrics.remove("bus_compressor_attenuation_db", metrics.labels);
1076         }
1077         bus_metrics.reset(new BusMetrics[new_input_mapping.buses.size()]);
1078         for (unsigned bus_index = 0; bus_index < new_input_mapping.buses.size(); ++bus_index) {
1079                 const InputMapping::Bus &bus = new_input_mapping.buses[bus_index];
1080                 BusMetrics &metrics = bus_metrics[bus_index];
1081
1082                 char bus_index_str[16], source_index_str[16], source_channels_str[64];
1083                 snprintf(bus_index_str, sizeof(bus_index_str), "%u", bus_index);
1084                 snprintf(source_index_str, sizeof(source_index_str), "%u", bus.device.index);
1085                 snprintf(source_channels_str, sizeof(source_channels_str), "%d:%d", bus.source_channel[0], bus.source_channel[1]);
1086
1087                 vector<pair<string, string>> labels;
1088                 metrics.labels.emplace_back("index", bus_index_str);
1089                 metrics.labels.emplace_back("name", bus.name);
1090                 if (bus.device.type == InputSourceType::SILENCE) {
1091                         metrics.labels.emplace_back("source_type", "silence");
1092                 } else if (bus.device.type == InputSourceType::CAPTURE_CARD) {
1093                         metrics.labels.emplace_back("source_type", "capture_card");
1094                 } else if (bus.device.type == InputSourceType::ALSA_INPUT) {
1095                         metrics.labels.emplace_back("source_type", "alsa_input");
1096                 } else if (bus.device.type == InputSourceType::FFMPEG_VIDEO_INPUT) {
1097                         metrics.labels.emplace_back("source_type", "ffmpeg_video_input");
1098                 } else {
1099                         assert(false);
1100                 }
1101                 metrics.labels.emplace_back("source_index", source_index_str);
1102                 metrics.labels.emplace_back("source_channels", source_channels_str);
1103
1104                 vector<pair<string, string>> labels_left = metrics.labels;
1105                 labels_left.emplace_back("channel", "left");
1106                 vector<pair<string, string>> labels_right = metrics.labels;
1107                 labels_right.emplace_back("channel", "right");
1108
1109                 global_metrics.add("bus_current_level_dbfs", labels_left, &metrics.current_level_dbfs[0], Metrics::TYPE_GAUGE);
1110                 global_metrics.add("bus_current_level_dbfs", labels_right, &metrics.current_level_dbfs[1], Metrics::TYPE_GAUGE);
1111                 global_metrics.add("bus_peak_level_dbfs", labels_left, &metrics.peak_level_dbfs[0], Metrics::TYPE_GAUGE);
1112                 global_metrics.add("bus_peak_level_dbfs", labels_right, &metrics.peak_level_dbfs[1], Metrics::TYPE_GAUGE);
1113                 global_metrics.add("bus_historic_peak_dbfs", metrics.labels, &metrics.historic_peak_dbfs, Metrics::TYPE_GAUGE);
1114                 global_metrics.add("bus_gain_staging_db", metrics.labels, &metrics.gain_staging_db, Metrics::TYPE_GAUGE);
1115                 global_metrics.add("bus_compressor_attenuation_db", metrics.labels, &metrics.compressor_attenuation_db, Metrics::TYPE_GAUGE);
1116         }
1117
1118         // Reset resamplers for all cards that don't have the exact same state as before.
1119         for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
1120                 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
1121                 AudioDevice *device = find_audio_device(device_spec);
1122                 if (device->interesting_channels != interesting_channels[device_spec]) {
1123                         device->interesting_channels = interesting_channels[device_spec];
1124                         reset_resampler_mutex_held(device_spec);
1125                 }
1126         }
1127         for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
1128                 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
1129                 AudioDevice *device = find_audio_device(device_spec);
1130                 if (interesting_channels[device_spec].empty()) {
1131                         alsa_pool.release_device(card_index);
1132                 } else {
1133                         alsa_pool.hold_device(card_index);
1134                 }
1135                 if (device->interesting_channels != interesting_channels[device_spec]) {
1136                         device->interesting_channels = interesting_channels[device_spec];
1137                         alsa_pool.reset_device(device_spec.index);
1138                         reset_resampler_mutex_held(device_spec);
1139                 }
1140         }
1141         for (unsigned card_index = 0; card_index < num_ffmpeg_inputs; ++card_index) {
1142                 const DeviceSpec device_spec{InputSourceType::FFMPEG_VIDEO_INPUT, card_index};
1143                 AudioDevice *device = find_audio_device(device_spec);
1144                 if (device->interesting_channels != interesting_channels[device_spec]) {
1145                         device->interesting_channels = interesting_channels[device_spec];
1146                         reset_resampler_mutex_held(device_spec);
1147                 }
1148         }
1149
1150         input_mapping = new_input_mapping;
1151 }
1152
1153 InputMapping AudioMixer::get_input_mapping() const
1154 {
1155         lock_guard<timed_mutex> lock(audio_mutex);
1156         return input_mapping;
1157 }
1158
1159 unsigned AudioMixer::num_buses() const
1160 {
1161         lock_guard<timed_mutex> lock(audio_mutex);
1162         return input_mapping.buses.size();
1163 }
1164
1165 void AudioMixer::reset_peak(unsigned bus_index)
1166 {
1167         lock_guard<timed_mutex> lock(audio_mutex);
1168         for (unsigned channel = 0; channel < 2; ++channel) {
1169                 PeakHistory &history = peak_history[bus_index][channel];
1170                 history.current_level = 0.0f;
1171                 history.historic_peak = 0.0f;
1172                 history.current_peak = 0.0f;
1173                 history.last_peak = 0.0f;
1174                 history.age_seconds = 0.0f;
1175         }
1176 }
1177
1178 AudioMixer *global_audio_mixer = nullptr;