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

   1 #include "audio_mixer.h"
   2
   3 #include <assert.h>
   4 #include <endian.h>
   5 #include <bmusb/bmusb.h>
   6 #include <stdio.h>
   7 #include <endian.h>
   8 #include <cmath>
   9 #ifdef __SSE__
  10 #include <immintrin.h>
  11 #endif
  12
  13 #include "db.h"
  14 #include "flags.h"
  15 #include "timebase.h"
  16
  17 using namespace bmusb;
  18 using namespace std;
  19 using namespace std::placeholders;
  20
  21 namespace {
  22
  23 // TODO: If these prove to be a bottleneck, they can be SSSE3-optimized
  24 // (usually including multiple channels at a time).
  25
  26 void convert_fixed16_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
  27                              const uint8_t *src, size_t in_channel, size_t in_num_channels,
  28                              size_t num_samples)
  29 {
  30         assert(in_channel < in_num_channels);
  31         assert(out_channel < out_num_channels);
  32         src += in_channel * 2;
  33         dst += out_channel;
  34
  35         for (size_t i = 0; i < num_samples; ++i) {
  36                 int16_t s = le16toh(*(int16_t *)src);
  37                 *dst = s * (1.0f / 32768.0f);
  38
  39                 src += 2 * in_num_channels;
  40                 dst += out_num_channels;
  41         }
  42 }
  43
  44 void convert_fixed24_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
  45                              const uint8_t *src, size_t in_channel, size_t in_num_channels,
  46                              size_t num_samples)
  47 {
  48         assert(in_channel < in_num_channels);
  49         assert(out_channel < out_num_channels);
  50         src += in_channel * 3;
  51         dst += out_channel;
  52
  53         for (size_t i = 0; i < num_samples; ++i) {
  54                 uint32_t s1 = src[0];
  55                 uint32_t s2 = src[1];
  56                 uint32_t s3 = src[2];
  57                 uint32_t s = s1 | (s1 << 8) | (s2 << 16) | (s3 << 24);
  58                 *dst = int(s) * (1.0f / 2147483648.0f);
  59
  60                 src += 3 * in_num_channels;
  61                 dst += out_num_channels;
  62         }
  63 }
  64
  65 void convert_fixed32_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
  66                              const uint8_t *src, size_t in_channel, size_t in_num_channels,
  67                              size_t num_samples)
  68 {
  69         assert(in_channel < in_num_channels);
  70         assert(out_channel < out_num_channels);
  71         src += in_channel * 4;
  72         dst += out_channel;
  73
  74         for (size_t i = 0; i < num_samples; ++i) {
  75                 int32_t s = le32toh(*(int32_t *)src);
  76                 *dst = s * (1.0f / 2147483648.0f);
  77
  78                 src += 4 * in_num_channels;
  79                 dst += out_num_channels;
  80         }
  81 }
  82
  83 float find_peak_plain(const float *samples, size_t num_samples) __attribute__((unused));
  84
  85 float find_peak_plain(const float *samples, size_t num_samples)
  86 {
  87         float m = fabs(samples[0]);
  88         for (size_t i = 1; i < num_samples; ++i) {
  89                 m = max(m, fabs(samples[i]));
  90         }
  91         return m;
  92 }
  93
  94 #ifdef __SSE__
  95 static inline float horizontal_max(__m128 m)
  96 {
  97         __m128 tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 0, 3, 2));
  98         m = _mm_max_ps(m, tmp);
  99         tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 3, 0, 1));
 100         m = _mm_max_ps(m, tmp);
 101         return _mm_cvtss_f32(m);
 102 }
 103
 104 float find_peak(const float *samples, size_t num_samples)
 105 {
 106         const __m128 abs_mask = _mm_castsi128_ps(_mm_set1_epi32(0x7fffffffu));
 107         __m128 m = _mm_setzero_ps();
 108         for (size_t i = 0; i < (num_samples & ~3); i += 4) {
 109                 __m128 x = _mm_loadu_ps(samples + i);
 110                 x = _mm_and_ps(x, abs_mask);
 111                 m = _mm_max_ps(m, x);
 112         }
 113         float result = horizontal_max(m);
 114
 115         for (size_t i = (num_samples & ~3); i < num_samples; ++i) {
 116                 result = max(result, fabs(samples[i]));
 117         }
 118
 119 #if 0
 120         // Self-test. We should be bit-exact the same.
 121         float reference_result = find_peak_plain(samples, num_samples);
 122         if (result != reference_result) {
 123                 fprintf(stderr, "Error: Peak is %f [%f %f %f %f]; should be %f.\n",
 124                         result,
 125                         _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(0, 0, 0, 0))),
 126                         _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 1, 1, 1))),
 127                         _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 2, 2, 2))),
 128                         _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(3, 3, 3, 3))),
 129                         reference_result);
 130                 abort();
 131         }
 132 #endif
 133         return result;
 134 }
 135 #else
 136 float find_peak(const float *samples, size_t num_samples)
 137 {
 138         return find_peak_plain(samples, num_samples);
 139 }
 140 #endif
 141
 142 void deinterleave_samples(const vector<float> &in, vector<float> *out_l, vector<float> *out_r)
 143 {
 144         size_t num_samples = in.size() / 2;
 145         out_l->resize(num_samples);
 146         out_r->resize(num_samples);
 147
 148         const float *inptr = in.data();
 149         float *lptr = &(*out_l)[0];
 150         float *rptr = &(*out_r)[0];
 151         for (size_t i = 0; i < num_samples; ++i) {
 152                 *lptr++ = *inptr++;
 153                 *rptr++ = *inptr++;
 154         }
 155 }
 156
 157 }  // namespace
 158
 159 AudioMixer::AudioMixer(unsigned num_cards)
 160         : num_cards(num_cards),
 161           limiter(OUTPUT_FREQUENCY),
 162           correlation(OUTPUT_FREQUENCY)
 163 {
 164         for (unsigned bus_index = 0; bus_index < MAX_BUSES; ++bus_index) {
 165                 locut[bus_index].init(FILTER_HPF, 2);
 166                 locut_enabled[bus_index] = global_flags.locut_enabled;
 167                 eq[bus_index][EQ_BAND_BASS].init(FILTER_LOW_SHELF, 1);
 168                 // Note: EQ_BAND_MID isn't used (see comments in apply_eq()).
 169                 eq[bus_index][EQ_BAND_TREBLE].init(FILTER_HIGH_SHELF, 1);
 170
 171                 gain_staging_db[bus_index] = global_flags.initial_gain_staging_db;
 172                 compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
 173                 compressor_threshold_dbfs[bus_index] = ref_level_dbfs - 12.0f;  // -12 dB.
 174                 compressor_enabled[bus_index] = global_flags.compressor_enabled;
 175                 level_compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
 176                 level_compressor_enabled[bus_index] = global_flags.gain_staging_auto;
 177         }
 178         set_limiter_enabled(global_flags.limiter_enabled);
 179         set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
 180
 181         // Generate a very simple, default input mapping.
 182         InputMapping::Bus input;
 183         input.name = "Main";
 184         input.device.type = InputSourceType::CAPTURE_CARD;
 185         input.device.index = 0;
 186         input.source_channel[0] = 0;
 187         input.source_channel[1] = 1;
 188
 189         InputMapping new_input_mapping;
 190         new_input_mapping.buses.push_back(input);
 191         set_input_mapping(new_input_mapping);
 192
 193         // Look for ALSA cards.
 194         available_alsa_cards = ALSAInput::enumerate_devices();
 195
 196         r128.init(2, OUTPUT_FREQUENCY);
 197         r128.integr_start();
 198
 199         // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
 200         // and there's a limit to how important the peak meter is.
 201         peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
 202 }
 203
 204 AudioMixer::~AudioMixer()
 205 {
 206         for (unsigned card_index = 0; card_index < available_alsa_cards.size(); ++card_index) {
 207                 const AudioDevice &device = alsa_inputs[card_index];
 208                 if (device.alsa_device != nullptr) {
 209                         device.alsa_device->stop_capture_thread();
 210                 }
 211         }
 212 }
 213
 214
 215 void AudioMixer::reset_resampler(DeviceSpec device_spec)
 216 {
 217         lock_guard<timed_mutex> lock(audio_mutex);
 218         reset_resampler_mutex_held(device_spec);
 219 }
 220
 221 void AudioMixer::reset_resampler_mutex_held(DeviceSpec device_spec)
 222 {
 223         AudioDevice *device = find_audio_device(device_spec);
 224
 225         if (device->interesting_channels.empty()) {
 226                 device->resampling_queue.reset();
 227         } else {
 228                 // TODO: ResamplingQueue should probably take the full device spec.
 229                 // (It's only used for console output, though.)
 230                 device->resampling_queue.reset(new ResamplingQueue(device_spec.index, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size()));
 231         }
 232         device->next_local_pts = 0;
 233 }
 234
 235 void AudioMixer::reset_alsa_mutex_held(DeviceSpec device_spec)
 236 {
 237         assert(device_spec.type == InputSourceType::ALSA_INPUT);
 238         unsigned card_index = device_spec.index;
 239         AudioDevice *device = find_audio_device(device_spec);
 240
 241         if (device->alsa_device != nullptr) {
 242                 device->alsa_device->stop_capture_thread();
 243         }
 244         if (device->interesting_channels.empty()) {
 245                 device->alsa_device.reset();
 246         } else {
 247                 const ALSAInput::Device &alsa_dev = available_alsa_cards[card_index];
 248                 device->alsa_device.reset(new ALSAInput(alsa_dev.address.c_str(), OUTPUT_FREQUENCY, alsa_dev.num_channels, bind(&AudioMixer::add_audio, this, device_spec, _1, _2, _3, _4)));
 249                 device->capture_frequency = device->alsa_device->get_sample_rate();
 250                 device->alsa_device->start_capture_thread();
 251         }
 252 }
 253
 254 bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length)
 255 {
 256         AudioDevice *device = find_audio_device(device_spec);
 257
 258         unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
 259         if (!lock.try_lock_for(chrono::milliseconds(10))) {
 260                 return false;
 261         }
 262         if (device->resampling_queue == nullptr) {
 263                 // No buses use this device; throw it away.
 264                 return true;
 265         }
 266
 267         unsigned num_channels = device->interesting_channels.size();
 268         assert(num_channels > 0);
 269
 270         // Convert the audio to fp32.
 271         unique_ptr<float[]> audio(new float[num_samples * num_channels]);
 272         unsigned channel_index = 0;
 273         for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) {
 274                 switch (audio_format.bits_per_sample) {
 275                 case 0:
 276                         assert(num_samples == 0);
 277                         break;
 278                 case 16:
 279                         convert_fixed16_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
 280                         break;
 281                 case 24:
 282                         convert_fixed24_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
 283                         break;
 284                 case 32:
 285                         convert_fixed32_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
 286                         break;
 287                 default:
 288                         fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
 289                         assert(false);
 290                 }
 291         }
 292
 293         // Now add it.
 294         int64_t local_pts = device->next_local_pts;
 295         device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.get(), num_samples);
 296         device->next_local_pts = local_pts + frame_length;
 297         return true;
 298 }
 299
 300 bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length)
 301 {
 302         AudioDevice *device = find_audio_device(device_spec);
 303
 304         unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
 305         if (!lock.try_lock_for(chrono::milliseconds(10))) {
 306                 return false;
 307         }
 308         if (device->resampling_queue == nullptr) {
 309                 // No buses use this device; throw it away.
 310                 return true;
 311         }
 312
 313         unsigned num_channels = device->interesting_channels.size();
 314         assert(num_channels > 0);
 315
 316         vector<float> silence(samples_per_frame * num_channels, 0.0f);
 317         for (unsigned i = 0; i < num_frames; ++i) {
 318                 device->resampling_queue->add_input_samples(device->next_local_pts / double(TIMEBASE), silence.data(), samples_per_frame);
 319                 // Note that if the format changed in the meantime, we have
 320                 // no way of detecting that; we just have to assume the frame length
 321                 // is always the same.
 322                 device->next_local_pts += frame_length;
 323         }
 324         return true;
 325 }
 326
 327 AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
 328 {
 329         switch (device.type) {
 330         case InputSourceType::CAPTURE_CARD:
 331                 return &video_cards[device.index];
 332         case InputSourceType::ALSA_INPUT:
 333                 return &alsa_inputs[device.index];
 334         case InputSourceType::SILENCE:
 335         default:
 336                 assert(false);
 337         }
 338         return nullptr;
 339 }
 340
 341 // Get a pointer to the given channel from the given device.
 342 // The channel must be picked out earlier and resampled.
 343 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)
 344 {
 345         static float zero = 0.0f;
 346         if (source_channel == -1 || device_spec.type == InputSourceType::SILENCE) {
 347                 *srcptr = &zero;
 348                 *stride = 0;
 349                 return;
 350         }
 351         AudioDevice *device = find_audio_device(device_spec);
 352         assert(device->interesting_channels.count(source_channel) != 0);
 353         unsigned channel_index = 0;
 354         for (int channel : device->interesting_channels) {
 355                 if (channel == source_channel) break;
 356                 ++channel_index;
 357         }
 358         assert(channel_index < device->interesting_channels.size());
 359         const auto it = samples_card.find(device_spec);
 360         assert(it != samples_card.end());
 361         *srcptr = &(it->second)[channel_index];
 362         *stride = device->interesting_channels.size();
 363 }
 364
 365 // TODO: Can be SSSE3-optimized if need be.
 366 void AudioMixer::fill_audio_bus(const map<DeviceSpec, vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output)
 367 {
 368         if (bus.device.type == InputSourceType::SILENCE) {
 369                 memset(output, 0, num_samples * sizeof(*output));
 370         } else {
 371                 assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
 372                        bus.device.type == InputSourceType::ALSA_INPUT);
 373                 const float *lsrc, *rsrc;
 374                 unsigned lstride, rstride;
 375                 float *dptr = output;
 376                 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[0], &lsrc, &lstride);
 377                 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[1], &rsrc, &rstride);
 378                 for (unsigned i = 0; i < num_samples; ++i) {
 379                         *dptr++ = *lsrc;
 380                         *dptr++ = *rsrc;
 381                         lsrc += lstride;
 382                         rsrc += rstride;
 383                 }
 384         }
 385 }
 386
 387 vector<float> AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
 388 {
 389         map<DeviceSpec, vector<float>> samples_card;
 390         vector<float> samples_bus;
 391
 392         lock_guard<timed_mutex> lock(audio_mutex);
 393
 394         // Pick out all the interesting channels from all the cards.
 395         // TODO: If the card has been hotswapped, the number of channels
 396         // might have changed; if so, we need to do some sort of remapping
 397         // to silence.
 398         for (const auto &spec_and_info : get_devices_mutex_held()) {
 399                 const DeviceSpec &device_spec = spec_and_info.first;
 400                 AudioDevice *device = find_audio_device(device_spec);
 401                 if (!device->interesting_channels.empty()) {
 402                         samples_card[device_spec].resize(num_samples * device->interesting_channels.size());
 403                         device->resampling_queue->get_output_samples(
 404                                 pts,
 405                                 &samples_card[device_spec][0],
 406                                 num_samples,
 407                                 rate_adjustment_policy);
 408                 }
 409         }
 410
 411         vector<float> samples_out, left, right;
 412         samples_out.resize(num_samples * 2);
 413         samples_bus.resize(num_samples * 2);
 414         for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
 415                 fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, &samples_bus[0]);
 416                 apply_eq(bus_index, &samples_bus);
 417
 418                 {
 419                         lock_guard<mutex> lock(compressor_mutex);
 420
 421                         // Apply a level compressor to get the general level right.
 422                         // Basically, if it's over about -40 dBFS, we squeeze it down to that level
 423                         // (or more precisely, near it, since we don't use infinite ratio),
 424                         // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
 425                         // entirely arbitrary, but from practical tests with speech, it seems to
 426                         // put ut around -23 LUFS, so it's a reasonable starting point for later use.
 427                         if (level_compressor_enabled[bus_index]) {
 428                                 float threshold = 0.01f;   // -40 dBFS.
 429                                 float ratio = 20.0f;
 430                                 float attack_time = 0.5f;
 431                                 float release_time = 20.0f;
 432                                 float makeup_gain = from_db(ref_level_dbfs - (-40.0f));  // +26 dB.
 433                                 level_compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
 434                                 gain_staging_db[bus_index] = to_db(level_compressor[bus_index]->get_attenuation() * makeup_gain);
 435                         } else {
 436                                 // Just apply the gain we already had.
 437                                 float g = from_db(gain_staging_db[bus_index]);
 438                                 for (size_t i = 0; i < samples_bus.size(); ++i) {
 439                                         samples_bus[i] *= g;
 440                                 }
 441                         }
 442
 443 #if 0
 444                         printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
 445                                 level_compressor.get_level(), to_db(level_compressor.get_level()),
 446                                 level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()),
 447                                 to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
 448 #endif
 449
 450                         // The real compressor.
 451                         if (compressor_enabled[bus_index]) {
 452                                 float threshold = from_db(compressor_threshold_dbfs[bus_index]);
 453                                 float ratio = 20.0f;
 454                                 float attack_time = 0.005f;
 455                                 float release_time = 0.040f;
 456                                 float makeup_gain = 2.0f;  // +6 dB.
 457                                 compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
 458                 //              compressor_att = compressor.get_attenuation();
 459                         }
 460                 }
 461
 462                 add_bus_to_master(bus_index, samples_bus, &samples_out);
 463                 deinterleave_samples(samples_bus, &left, &right);
 464                 measure_bus_levels(bus_index, left, right);
 465         }
 466
 467         {
 468                 lock_guard<mutex> lock(compressor_mutex);
 469
 470                 // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
 471                 // Note that since ratio is not infinite, we could go slightly higher than this.
 472                 if (limiter_enabled) {
 473                         float threshold = from_db(limiter_threshold_dbfs);
 474                         float ratio = 30.0f;
 475                         float attack_time = 0.0f;  // Instant.
 476                         float release_time = 0.020f;
 477                         float makeup_gain = 1.0f;  // 0 dB.
 478                         limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
 479         //              limiter_att = limiter.get_attenuation();
 480                 }
 481
 482         //      printf("limiter=%+5.1f  compressor=%+5.1f\n", to_db(limiter_att), to_db(compressor_att));
 483         }
 484
 485         // At this point, we are most likely close to +0 LU (at least if the
 486         // faders sum to 0 dB and the compressors are on), but all of our
 487         // measurements have been on raw sample values, not R128 values.
 488         // So we have a final makeup gain to get us to +0 LU; the gain
 489         // adjustments required should be relatively small, and also, the
 490         // offset shouldn't change much (only if the type of audio changes
 491         // significantly). Thus, we shoot for updating this value basically
 492         // “whenever we process buffers”, since the R128 calculation isn't exactly
 493         // something we get out per-sample.
 494         //
 495         // Note that there's a feedback loop here, so we choose a very slow filter
 496         // (half-time of 30 seconds).
 497         double target_loudness_factor, alpha;
 498         double loudness_lu = r128.loudness_M() - ref_level_lufs;
 499         double current_makeup_lu = to_db(final_makeup_gain);
 500         target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
 501
 502         // If we're outside +/- 5 LU uncorrected, we don't count it as
 503         // a normal signal (probably silence) and don't change the
 504         // correction factor; just apply what we already have.
 505         if (fabs(loudness_lu - current_makeup_lu) >= 5.0 || !final_makeup_gain_auto) {
 506                 alpha = 0.0;
 507         } else {
 508                 // Formula adapted from
 509                 // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
 510                 const double half_time_s = 30.0;
 511                 const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
 512                 alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
 513         }
 514
 515         {
 516                 lock_guard<mutex> lock(compressor_mutex);
 517                 double m = final_makeup_gain;
 518                 for (size_t i = 0; i < samples_out.size(); i += 2) {
 519                         samples_out[i + 0] *= m;
 520                         samples_out[i + 1] *= m;
 521                         m += (target_loudness_factor - m) * alpha;
 522                 }
 523                 final_makeup_gain = m;
 524         }
 525
 526         update_meters(samples_out);
 527
 528         return samples_out;
 529 }
 530
 531 void AudioMixer::apply_eq(unsigned bus_index, vector<float> *samples_bus)
 532 {
 533         constexpr float bass_freq_hz = 200.0f;
 534         constexpr float treble_freq_hz = 4700.0f;
 535
 536         // Cut away everything under 120 Hz (or whatever the cutoff is);
 537         // we don't need it for voice, and it will reduce headroom
 538         // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
 539         // should be dampened.)
 540         if (locut_enabled[bus_index]) {
 541                 locut[bus_index].render(samples_bus->data(), samples_bus->size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
 542         }
 543
 544         // Apply the rest of the EQ. Since we only have a simple three-band EQ,
 545         // we can implement it with two shelf filters. We use a simple gain to
 546         // set the mid-level filter, and then offset the low and high bands
 547         // from that if we need to. (We could perhaps have folded the gain into
 548         // the next part, but it's so cheap that the trouble isn't worth it.)
 549         if (fabs(eq_level_db[bus_index][EQ_BAND_MID]) > 0.01f) {
 550                 float g = from_db(eq_level_db[bus_index][EQ_BAND_MID]);
 551                 for (size_t i = 0; i < samples_bus->size(); ++i) {
 552                         (*samples_bus)[i] *= g;
 553                 }
 554         }
 555
 556         float bass_adj_db = eq_level_db[bus_index][EQ_BAND_BASS] - eq_level_db[bus_index][EQ_BAND_MID];
 557         if (fabs(bass_adj_db) > 0.01f) {
 558                 eq[bus_index][EQ_BAND_BASS].render(samples_bus->data(), samples_bus->size() / 2,
 559                         bass_freq_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f, bass_adj_db / 40.0f);
 560         }
 561
 562         float treble_adj_db = eq_level_db[bus_index][EQ_BAND_TREBLE] - eq_level_db[bus_index][EQ_BAND_MID];
 563         if (fabs(treble_adj_db) > 0.01f) {
 564                 eq[bus_index][EQ_BAND_TREBLE].render(samples_bus->data(), samples_bus->size() / 2,
 565                         treble_freq_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f, treble_adj_db / 40.0f);
 566         }
 567 }
 568
 569 void AudioMixer::add_bus_to_master(unsigned bus_index, const vector<float> &samples_bus, vector<float> *samples_out)
 570 {
 571         assert(samples_bus.size() == samples_out->size());
 572         assert(samples_bus.size() % 2 == 0);
 573         unsigned num_samples = samples_bus.size() / 2;
 574         if (fabs(fader_volume_db[bus_index] - last_fader_volume_db[bus_index]) > 1e-3) {
 575                 // The volume has changed; do a fade over the course of this frame.
 576                 // (We might have some numerical issues here, but it seems to sound OK.)
 577                 // For the purpose of fading here, the silence floor is set to -90 dB
 578                 // (the fader only goes to -84).
 579                 float old_volume = from_db(max<float>(last_fader_volume_db[bus_index], -90.0f));
 580                 float volume = from_db(max<float>(fader_volume_db[bus_index], -90.0f));
 581
 582                 float volume_inc = pow(volume / old_volume, 1.0 / num_samples);
 583                 volume = old_volume;
 584                 if (bus_index == 0) {
 585                         for (unsigned i = 0; i < num_samples; ++i) {
 586                                 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
 587                                 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
 588                                 volume *= volume_inc;
 589                         }
 590                 } else {
 591                         for (unsigned i = 0; i < num_samples; ++i) {
 592                                 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
 593                                 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
 594                                 volume *= volume_inc;
 595                         }
 596                 }
 597         } else {
 598                 float volume = from_db(fader_volume_db[bus_index]);
 599                 if (bus_index == 0) {
 600                         for (unsigned i = 0; i < num_samples; ++i) {
 601                                 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
 602                                 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
 603                         }
 604                 } else {
 605                         for (unsigned i = 0; i < num_samples; ++i) {
 606                                 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
 607                                 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
 608                         }
 609                 }
 610         }
 611
 612         last_fader_volume_db[bus_index] = fader_volume_db[bus_index];
 613 }
 614
 615 void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right)
 616 {
 617         assert(left.size() == right.size());
 618         const float volume = from_db(fader_volume_db[bus_index]);
 619         const float peak_levels[2] = {
 620                 find_peak(left.data(), left.size()) * volume,
 621                 find_peak(right.data(), right.size()) * volume
 622         };
 623         for (unsigned channel = 0; channel < 2; ++channel) {
 624                 // Compute the current value, including hold and falloff.
 625                 // The constants are borrowed from zita-mu1 by Fons Adriaensen.
 626                 static constexpr float hold_sec = 0.5f;
 627                 static constexpr float falloff_db_sec = 15.0f;  // dB/sec falloff after hold.
 628                 float current_peak;
 629                 PeakHistory &history = peak_history[bus_index][channel];
 630                 history.historic_peak = max(history.historic_peak, peak_levels[channel]);
 631                 if (history.age_seconds < hold_sec) {
 632                         current_peak = history.last_peak;
 633                 } else {
 634                         current_peak = history.last_peak * from_db(-falloff_db_sec * (history.age_seconds - hold_sec));
 635                 }
 636
 637                 // See if we have a new peak to replace the old (possibly falling) one.
 638                 if (peak_levels[channel] > current_peak) {
 639                         history.last_peak = peak_levels[channel];
 640                         history.age_seconds = 0.0f;  // Not 100% correct, but more than good enough given our frame sizes.
 641                         current_peak = peak_levels[channel];
 642                 } else {
 643                         history.age_seconds += float(left.size()) / OUTPUT_FREQUENCY;
 644                 }
 645                 history.current_level = peak_levels[channel];
 646                 history.current_peak = current_peak;
 647         }
 648 }
 649
 650 void AudioMixer::update_meters(const vector<float> &samples)
 651 {
 652         // Upsample 4x to find interpolated peak.
 653         peak_resampler.inp_data = const_cast<float *>(samples.data());
 654         peak_resampler.inp_count = samples.size() / 2;
 655
 656         vector<float> interpolated_samples;
 657         interpolated_samples.resize(samples.size());
 658         {
 659                 lock_guard<mutex> lock(audio_measure_mutex);
 660
 661                 while (peak_resampler.inp_count > 0) {  // About four iterations.
 662                         peak_resampler.out_data = &interpolated_samples[0];
 663                         peak_resampler.out_count = interpolated_samples.size() / 2;
 664                         peak_resampler.process();
 665                         size_t out_stereo_samples = interpolated_samples.size() / 2 - peak_resampler.out_count;
 666                         peak = max<float>(peak, find_peak(interpolated_samples.data(), out_stereo_samples * 2));
 667                         peak_resampler.out_data = nullptr;
 668                 }
 669         }
 670
 671         // Find R128 levels and L/R correlation.
 672         vector<float> left, right;
 673         deinterleave_samples(samples, &left, &right);
 674         float *ptrs[] = { left.data(), right.data() };
 675         {
 676                 lock_guard<mutex> lock(audio_measure_mutex);
 677                 r128.process(left.size(), ptrs);
 678                 correlation.process_samples(samples);
 679         }
 680
 681         send_audio_level_callback();
 682 }
 683
 684 void AudioMixer::reset_meters()
 685 {
 686         lock_guard<mutex> lock(audio_measure_mutex);
 687         peak_resampler.reset();
 688         peak = 0.0f;
 689         r128.reset();
 690         r128.integr_start();
 691         correlation.reset();
 692 }
 693
 694 void AudioMixer::send_audio_level_callback()
 695 {
 696         if (audio_level_callback == nullptr) {
 697                 return;
 698         }
 699
 700         lock_guard<mutex> lock(audio_measure_mutex);
 701         double loudness_s = r128.loudness_S();
 702         double loudness_i = r128.integrated();
 703         double loudness_range_low = r128.range_min();
 704         double loudness_range_high = r128.range_max();
 705
 706         vector<BusLevel> bus_levels;
 707         bus_levels.resize(input_mapping.buses.size());
 708         {
 709                 lock_guard<mutex> lock(compressor_mutex);
 710                 for (unsigned bus_index = 0; bus_index < bus_levels.size(); ++bus_index) {
 711                         bus_levels[bus_index].current_level_dbfs[0] = to_db(peak_history[bus_index][0].current_level);
 712                         bus_levels[bus_index].current_level_dbfs[1] = to_db(peak_history[bus_index][1].current_level);
 713                         bus_levels[bus_index].peak_level_dbfs[0] = to_db(peak_history[bus_index][0].current_peak);
 714                         bus_levels[bus_index].peak_level_dbfs[1] = to_db(peak_history[bus_index][1].current_peak);
 715                         bus_levels[bus_index].historic_peak_dbfs = to_db(
 716                                 max(peak_history[bus_index][0].historic_peak,
 717                                     peak_history[bus_index][1].historic_peak));
 718                         bus_levels[bus_index].gain_staging_db = gain_staging_db[bus_index];
 719                         if (compressor_enabled[bus_index]) {
 720                                 bus_levels[bus_index].compressor_attenuation_db = -to_db(compressor[bus_index]->get_attenuation());
 721                         } else {
 722                                 bus_levels[bus_index].compressor_attenuation_db = 0.0;
 723                         }
 724                 }
 725         }
 726
 727         audio_level_callback(loudness_s, to_db(peak), bus_levels,
 728                 loudness_i, loudness_range_low, loudness_range_high,
 729                 to_db(final_makeup_gain),
 730                 correlation.get_correlation());
 731 }
 732
 733 map<DeviceSpec, DeviceInfo> AudioMixer::get_devices() const
 734 {
 735         lock_guard<timed_mutex> lock(audio_mutex);
 736         return get_devices_mutex_held();
 737 }
 738
 739 map<DeviceSpec, DeviceInfo> AudioMixer::get_devices_mutex_held() const
 740 {
 741         map<DeviceSpec, DeviceInfo> devices;
 742         for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
 743                 const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index };
 744                 const AudioDevice *device = &video_cards[card_index];
 745                 DeviceInfo info;
 746                 info.name = device->name;
 747                 info.num_channels = 8;  // FIXME: This is wrong for fake cards.
 748                 devices.insert(make_pair(spec, info));
 749         }
 750         for (unsigned card_index = 0; card_index < available_alsa_cards.size(); ++card_index) {
 751                 const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index };
 752                 const ALSAInput::Device &device = available_alsa_cards[card_index];
 753                 DeviceInfo info;
 754                 info.name = device.name + " (" + device.info + ")";
 755                 info.num_channels = device.num_channels;
 756                 devices.insert(make_pair(spec, info));
 757         }
 758         return devices;
 759 }
 760
 761 void AudioMixer::set_name(DeviceSpec device_spec, const string &name)
 762 {
 763         AudioDevice *device = find_audio_device(device_spec);
 764
 765         lock_guard<timed_mutex> lock(audio_mutex);
 766         device->name = name;
 767 }
 768
 769 void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
 770 {
 771         lock_guard<timed_mutex> lock(audio_mutex);
 772
 773         map<DeviceSpec, set<unsigned>> interesting_channels;
 774         for (const InputMapping::Bus &bus : new_input_mapping.buses) {
 775                 if (bus.device.type == InputSourceType::CAPTURE_CARD ||
 776                     bus.device.type == InputSourceType::ALSA_INPUT) {
 777                         for (unsigned channel = 0; channel < 2; ++channel) {
 778                                 if (bus.source_channel[channel] != -1) {
 779                                         interesting_channels[bus.device].insert(bus.source_channel[channel]);
 780                                 }
 781                         }
 782                 }
 783         }
 784
 785         // Reset resamplers for all cards that don't have the exact same state as before.
 786         for (const auto &spec_and_info : get_devices_mutex_held()) {
 787                 const DeviceSpec &device_spec = spec_and_info.first;
 788                 AudioDevice *device = find_audio_device(device_spec);
 789                 if (device->interesting_channels != interesting_channels[device_spec]) {
 790                         device->interesting_channels = interesting_channels[device_spec];
 791                         if (device_spec.type == InputSourceType::ALSA_INPUT) {
 792                                 reset_alsa_mutex_held(device_spec);
 793                         }
 794                         reset_resampler_mutex_held(device_spec);
 795                 }
 796         }
 797
 798         input_mapping = new_input_mapping;
 799 }
 800
 801 InputMapping AudioMixer::get_input_mapping() const
 802 {
 803         lock_guard<timed_mutex> lock(audio_mutex);
 804         return input_mapping;
 805 }
 806
 807 void AudioMixer::reset_peak(unsigned bus_index)
 808 {
 809         lock_guard<timed_mutex> lock(audio_mutex);
 810         for (unsigned channel = 0; channel < 2; ++channel) {
 811                 PeakHistory &history = peak_history[bus_index][channel];
 812                 history.current_level = 0.0f;
 813                 history.historic_peak = 0.0f;
 814                 history.current_peak = 0.0f;
 815                 history.last_peak = 0.0f;
 816                 history.age_seconds = 0.0f;
 817         }
 818 }