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