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
  10 #include "db.h"
  11 #include "flags.h"
  12 #include "timebase.h"
  13
  14 using namespace bmusb;
  15 using namespace std;
  16 using namespace std::placeholders;
  17
  18 namespace {
  19
  20 // TODO: If these prove to be a bottleneck, they can be SSSE3-optimized
  21 // (usually including multiple channels at a time).
  22
  23 void convert_fixed16_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
  24                              const uint8_t *src, size_t in_channel, size_t in_num_channels,
  25                              size_t num_samples)
  26 {
  27         assert(in_channel < in_num_channels);
  28         assert(out_channel < out_num_channels);
  29         src += in_channel * 2;
  30         dst += out_channel;
  31
  32         for (size_t i = 0; i < num_samples; ++i) {
  33                 int16_t s = le16toh(*(int16_t *)src);
  34                 *dst = s * (1.0f / 32768.0f);
  35
  36                 src += 2 * in_num_channels;
  37                 dst += out_num_channels;
  38         }
  39 }
  40
  41 void convert_fixed24_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
  42                              const uint8_t *src, size_t in_channel, size_t in_num_channels,
  43                              size_t num_samples)
  44 {
  45         assert(in_channel < in_num_channels);
  46         assert(out_channel < out_num_channels);
  47         src += in_channel * 3;
  48         dst += out_channel;
  49
  50         for (size_t i = 0; i < num_samples; ++i) {
  51                 uint32_t s1 = src[0];
  52                 uint32_t s2 = src[1];
  53                 uint32_t s3 = src[2];
  54                 uint32_t s = s1 | (s1 << 8) | (s2 << 16) | (s3 << 24);
  55                 *dst = int(s) * (1.0f / 2147483648.0f);
  56
  57                 src += 3 * in_num_channels;
  58                 dst += out_num_channels;
  59         }
  60 }
  61
  62 void convert_fixed32_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
  63                              const uint8_t *src, size_t in_channel, size_t in_num_channels,
  64                              size_t num_samples)
  65 {
  66         assert(in_channel < in_num_channels);
  67         assert(out_channel < out_num_channels);
  68         src += in_channel * 4;
  69         dst += out_channel;
  70
  71         for (size_t i = 0; i < num_samples; ++i) {
  72                 int32_t s = le32toh(*(int32_t *)src);
  73                 *dst = s * (1.0f / 2147483648.0f);
  74
  75                 src += 4 * in_num_channels;
  76                 dst += out_num_channels;
  77         }
  78 }
  79
  80 float find_peak(const float *samples, size_t num_samples)
  81 {
  82         float m = fabs(samples[0]);
  83         for (size_t i = 1; i < num_samples; ++i) {
  84                 m = max(m, fabs(samples[i]));
  85         }
  86         return m;
  87 }
  88
  89 void deinterleave_samples(const vector<float> &in, vector<float> *out_l, vector<float> *out_r)
  90 {
  91         size_t num_samples = in.size() / 2;
  92         out_l->resize(num_samples);
  93         out_r->resize(num_samples);
  94
  95         const float *inptr = in.data();
  96         float *lptr = &(*out_l)[0];
  97         float *rptr = &(*out_r)[0];
  98         for (size_t i = 0; i < num_samples; ++i) {
  99                 *lptr++ = *inptr++;
 100                 *rptr++ = *inptr++;
 101         }
 102 }
 103
 104 }  // namespace
 105
 106 AudioMixer::AudioMixer(unsigned num_cards)
 107         : num_cards(num_cards),
 108           level_compressor(OUTPUT_FREQUENCY),
 109           limiter(OUTPUT_FREQUENCY),
 110           compressor(OUTPUT_FREQUENCY),
 111           correlation(OUTPUT_FREQUENCY)
 112 {
 113         locut.init(FILTER_HPF, 2);
 114
 115         set_locut_enabled(global_flags.locut_enabled);
 116         set_gain_staging_db(global_flags.initial_gain_staging_db);
 117         set_gain_staging_auto(global_flags.gain_staging_auto);
 118         set_compressor_enabled(global_flags.compressor_enabled);
 119         set_limiter_enabled(global_flags.limiter_enabled);
 120         set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
 121
 122         // Generate a very simple, default input mapping.
 123         InputMapping::Bus input;
 124         input.name = "Main";
 125         input.device.type = InputSourceType::CAPTURE_CARD;
 126         input.device.index = 0;
 127         input.source_channel[0] = 0;
 128         input.source_channel[1] = 1;
 129
 130         InputMapping new_input_mapping;
 131         new_input_mapping.buses.push_back(input);
 132         set_input_mapping(new_input_mapping);
 133
 134         // Look for ALSA cards.
 135         available_alsa_cards = ALSAInput::enumerate_devices();
 136
 137         r128.init(2, OUTPUT_FREQUENCY);
 138         r128.integr_start();
 139
 140         // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
 141         // and there's a limit to how important the peak meter is.
 142         peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
 143 }
 144
 145 AudioMixer::~AudioMixer()
 146 {
 147         for (unsigned card_index = 0; card_index < available_alsa_cards.size(); ++card_index) {
 148                 const AudioDevice &device = alsa_inputs[card_index];
 149                 if (device.alsa_device != nullptr) {
 150                         device.alsa_device->stop_capture_thread();
 151                 }
 152         }
 153 }
 154
 155
 156 void AudioMixer::reset_resampler(DeviceSpec device_spec)
 157 {
 158         lock_guard<timed_mutex> lock(audio_mutex);
 159         reset_resampler_mutex_held(device_spec);
 160 }
 161
 162 void AudioMixer::reset_resampler_mutex_held(DeviceSpec device_spec)
 163 {
 164         AudioDevice *device = find_audio_device(device_spec);
 165
 166         if (device->interesting_channels.empty()) {
 167                 device->resampling_queue.reset();
 168         } else {
 169                 // TODO: ResamplingQueue should probably take the full device spec.
 170                 // (It's only used for console output, though.)
 171                 device->resampling_queue.reset(new ResamplingQueue(device_spec.index, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size()));
 172         }
 173         device->next_local_pts = 0;
 174 }
 175
 176 void AudioMixer::reset_alsa_mutex_held(DeviceSpec device_spec)
 177 {
 178         assert(device_spec.type == InputSourceType::ALSA_INPUT);
 179         unsigned card_index = device_spec.index;
 180         AudioDevice *device = find_audio_device(device_spec);
 181
 182         if (device->alsa_device != nullptr) {
 183                 device->alsa_device->stop_capture_thread();
 184         }
 185         if (device->interesting_channels.empty()) {
 186                 device->alsa_device.reset();
 187         } else {
 188                 device->alsa_device.reset(new ALSAInput(available_alsa_cards[card_index].address.c_str(), OUTPUT_FREQUENCY, 2, bind(&AudioMixer::add_audio, this, device_spec, _1, _2, _3, _4)));
 189                 device->capture_frequency = device->alsa_device->get_sample_rate();
 190                 device->alsa_device->start_capture_thread();
 191         }
 192 }
 193
 194 bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length)
 195 {
 196         AudioDevice *device = find_audio_device(device_spec);
 197
 198         unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
 199         if (!lock.try_lock_for(chrono::milliseconds(10))) {
 200                 return false;
 201         }
 202         if (device->resampling_queue == nullptr) {
 203                 // No buses use this device; throw it away.
 204                 return true;
 205         }
 206
 207         unsigned num_channels = device->interesting_channels.size();
 208         assert(num_channels > 0);
 209
 210         // Convert the audio to fp32.
 211         vector<float> audio;
 212         audio.resize(num_samples * num_channels);
 213         unsigned channel_index = 0;
 214         for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) {
 215                 switch (audio_format.bits_per_sample) {
 216                 case 0:
 217                         assert(num_samples == 0);
 218                         break;
 219                 case 16:
 220                         convert_fixed16_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
 221                         break;
 222                 case 24:
 223                         convert_fixed24_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
 224                         break;
 225                 case 32:
 226                         convert_fixed32_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
 227                         break;
 228                 default:
 229                         fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
 230                         assert(false);
 231                 }
 232         }
 233
 234         // Now add it.
 235         int64_t local_pts = device->next_local_pts;
 236         device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.data(), num_samples);
 237         device->next_local_pts = local_pts + frame_length;
 238         return true;
 239 }
 240
 241 bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length)
 242 {
 243         AudioDevice *device = find_audio_device(device_spec);
 244
 245         unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
 246         if (!lock.try_lock_for(chrono::milliseconds(10))) {
 247                 return false;
 248         }
 249         if (device->resampling_queue == nullptr) {
 250                 // No buses use this device; throw it away.
 251                 return true;
 252         }
 253
 254         unsigned num_channels = device->interesting_channels.size();
 255         assert(num_channels > 0);
 256
 257         vector<float> silence(samples_per_frame * num_channels, 0.0f);
 258         for (unsigned i = 0; i < num_frames; ++i) {
 259                 device->resampling_queue->add_input_samples(device->next_local_pts / double(TIMEBASE), silence.data(), samples_per_frame);
 260                 // Note that if the format changed in the meantime, we have
 261                 // no way of detecting that; we just have to assume the frame length
 262                 // is always the same.
 263                 device->next_local_pts += frame_length;
 264         }
 265         return true;
 266 }
 267
 268 AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
 269 {
 270         switch (device.type) {
 271         case InputSourceType::CAPTURE_CARD:
 272                 return &video_cards[device.index];
 273         case InputSourceType::ALSA_INPUT:
 274                 return &alsa_inputs[device.index];
 275         case InputSourceType::SILENCE:
 276         default:
 277                 assert(false);
 278         }
 279         return nullptr;
 280 }
 281
 282 // Get a pointer to the given channel from the given device.
 283 // The channel must be picked out earlier and resampled.
 284 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)
 285 {
 286         static float zero = 0.0f;
 287         if (source_channel == -1 || device_spec.type == InputSourceType::SILENCE) {
 288                 *srcptr = &zero;
 289                 *stride = 0;
 290                 return;
 291         }
 292         AudioDevice *device = find_audio_device(device_spec);
 293         assert(device->interesting_channels.count(source_channel) != 0);
 294         unsigned channel_index = 0;
 295         for (int channel : device->interesting_channels) {
 296                 if (channel == source_channel) break;
 297                 ++channel_index;
 298         }
 299         assert(channel_index < device->interesting_channels.size());
 300         const auto it = samples_card.find(device_spec);
 301         assert(it != samples_card.end());
 302         *srcptr = &(it->second)[channel_index];
 303         *stride = device->interesting_channels.size();
 304 }
 305
 306 // TODO: Can be SSSE3-optimized if need be.
 307 void AudioMixer::fill_audio_bus(const map<DeviceSpec, vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output)
 308 {
 309         if (bus.device.type == InputSourceType::SILENCE) {
 310                 memset(output, 0, num_samples * sizeof(*output));
 311         } else {
 312                 assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
 313                        bus.device.type == InputSourceType::ALSA_INPUT);
 314                 const float *lsrc, *rsrc;
 315                 unsigned lstride, rstride;
 316                 float *dptr = output;
 317                 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[0], &lsrc, &lstride);
 318                 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[1], &rsrc, &rstride);
 319                 for (unsigned i = 0; i < num_samples; ++i) {
 320                         *dptr++ = *lsrc;
 321                         *dptr++ = *rsrc;
 322                         lsrc += lstride;
 323                         rsrc += rstride;
 324                 }
 325         }
 326 }
 327
 328 vector<float> AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
 329 {
 330         map<DeviceSpec, vector<float>> samples_card;
 331         vector<float> samples_bus;
 332
 333         lock_guard<timed_mutex> lock(audio_mutex);
 334
 335         // Pick out all the interesting channels from all the cards.
 336         // TODO: If the card has been hotswapped, the number of channels
 337         // might have changed; if so, we need to do some sort of remapping
 338         // to silence.
 339         for (const auto &spec_and_info : get_devices_mutex_held()) {
 340                 const DeviceSpec &device_spec = spec_and_info.first;
 341                 AudioDevice *device = find_audio_device(device_spec);
 342                 if (!device->interesting_channels.empty()) {
 343                         samples_card[device_spec].resize(num_samples * device->interesting_channels.size());
 344                         device->resampling_queue->get_output_samples(
 345                                 pts,
 346                                 &samples_card[device_spec][0],
 347                                 num_samples,
 348                                 rate_adjustment_policy);
 349                 }
 350         }
 351
 352         // TODO: Move lo-cut etc. into each bus.
 353         vector<float> samples_out;
 354         samples_out.resize(num_samples * 2);
 355         samples_bus.resize(num_samples * 2);
 356         for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
 357                 fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, &samples_bus[0]);
 358
 359                 float volume = from_db(fader_volume_db[bus_index]);
 360                 if (bus_index == 0) {
 361                         for (unsigned i = 0; i < num_samples * 2; ++i) {
 362                                 samples_out[i] = samples_bus[i] * volume;
 363                         }
 364                 } else {
 365                         for (unsigned i = 0; i < num_samples * 2; ++i) {
 366                                 samples_out[i] += samples_bus[i] * volume;
 367                         }
 368                 }
 369         }
 370
 371         // Cut away everything under 120 Hz (or whatever the cutoff is);
 372         // we don't need it for voice, and it will reduce headroom
 373         // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
 374         // should be dampened.)
 375         if (locut_enabled) {
 376                 locut.render(samples_out.data(), samples_out.size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
 377         }
 378
 379         {
 380                 lock_guard<mutex> lock(compressor_mutex);
 381
 382                 // Apply a level compressor to get the general level right.
 383                 // Basically, if it's over about -40 dBFS, we squeeze it down to that level
 384                 // (or more precisely, near it, since we don't use infinite ratio),
 385                 // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
 386                 // entirely arbitrary, but from practical tests with speech, it seems to
 387                 // put ut around -23 LUFS, so it's a reasonable starting point for later use.
 388                 {
 389                         if (level_compressor_enabled) {
 390                                 float threshold = 0.01f;   // -40 dBFS.
 391                                 float ratio = 20.0f;
 392                                 float attack_time = 0.5f;
 393                                 float release_time = 20.0f;
 394                                 float makeup_gain = from_db(ref_level_dbfs - (-40.0f));  // +26 dB.
 395                                 level_compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
 396                                 gain_staging_db = to_db(level_compressor.get_attenuation() * makeup_gain);
 397                         } else {
 398                                 // Just apply the gain we already had.
 399                                 float g = from_db(gain_staging_db);
 400                                 for (size_t i = 0; i < samples_out.size(); ++i) {
 401                                         samples_out[i] *= g;
 402                                 }
 403                         }
 404                 }
 405
 406         #if 0
 407                 printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
 408                         level_compressor.get_level(), to_db(level_compressor.get_level()),
 409                         level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()),
 410                         to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
 411         #endif
 412
 413         //      float limiter_att, compressor_att;
 414
 415                 // The real compressor.
 416                 if (compressor_enabled) {
 417                         float threshold = from_db(compressor_threshold_dbfs);
 418                         float ratio = 20.0f;
 419                         float attack_time = 0.005f;
 420                         float release_time = 0.040f;
 421                         float makeup_gain = 2.0f;  // +6 dB.
 422                         compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
 423         //              compressor_att = compressor.get_attenuation();
 424                 }
 425
 426                 // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
 427                 // Note that since ratio is not infinite, we could go slightly higher than this.
 428                 if (limiter_enabled) {
 429                         float threshold = from_db(limiter_threshold_dbfs);
 430                         float ratio = 30.0f;
 431                         float attack_time = 0.0f;  // Instant.
 432                         float release_time = 0.020f;
 433                         float makeup_gain = 1.0f;  // 0 dB.
 434                         limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
 435         //              limiter_att = limiter.get_attenuation();
 436                 }
 437
 438         //      printf("limiter=%+5.1f  compressor=%+5.1f\n", to_db(limiter_att), to_db(compressor_att));
 439         }
 440
 441         // At this point, we are most likely close to +0 LU, but all of our
 442         // measurements have been on raw sample values, not R128 values.
 443         // So we have a final makeup gain to get us to +0 LU; the gain
 444         // adjustments required should be relatively small, and also, the
 445         // offset shouldn't change much (only if the type of audio changes
 446         // significantly). Thus, we shoot for updating this value basically
 447         // “whenever we process buffers”, since the R128 calculation isn't exactly
 448         // something we get out per-sample.
 449         //
 450         // Note that there's a feedback loop here, so we choose a very slow filter
 451         // (half-time of 30 seconds).
 452         double target_loudness_factor, alpha;
 453         double loudness_lu = r128.loudness_M() - ref_level_lufs;
 454         double current_makeup_lu = to_db(final_makeup_gain);
 455         target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
 456
 457         // If we're outside +/- 5 LU uncorrected, we don't count it as
 458         // a normal signal (probably silence) and don't change the
 459         // correction factor; just apply what we already have.
 460         if (fabs(loudness_lu - current_makeup_lu) >= 5.0 || !final_makeup_gain_auto) {
 461                 alpha = 0.0;
 462         } else {
 463                 // Formula adapted from
 464                 // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
 465                 const double half_time_s = 30.0;
 466                 const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
 467                 alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
 468         }
 469
 470         {
 471                 lock_guard<mutex> lock(compressor_mutex);
 472                 double m = final_makeup_gain;
 473                 for (size_t i = 0; i < samples_out.size(); i += 2) {
 474                         samples_out[i + 0] *= m;
 475                         samples_out[i + 1] *= m;
 476                         m += (target_loudness_factor - m) * alpha;
 477                 }
 478                 final_makeup_gain = m;
 479         }
 480
 481         update_meters(samples_out);
 482
 483         return samples_out;
 484 }
 485
 486 void AudioMixer::update_meters(const vector<float> &samples)
 487 {
 488         // Upsample 4x to find interpolated peak.
 489         peak_resampler.inp_data = const_cast<float *>(samples.data());
 490         peak_resampler.inp_count = samples.size() / 2;
 491
 492         vector<float> interpolated_samples;
 493         interpolated_samples.resize(samples.size());
 494         {
 495                 unique_lock<mutex> lock(audio_measure_mutex);
 496
 497                 while (peak_resampler.inp_count > 0) {  // About four iterations.
 498                         peak_resampler.out_data = &interpolated_samples[0];
 499                         peak_resampler.out_count = interpolated_samples.size() / 2;
 500                         peak_resampler.process();
 501                         size_t out_stereo_samples = interpolated_samples.size() / 2 - peak_resampler.out_count;
 502                         peak = max<float>(peak, find_peak(interpolated_samples.data(), out_stereo_samples * 2));
 503                         peak_resampler.out_data = nullptr;
 504                 }
 505         }
 506
 507         // Find R128 levels and L/R correlation.
 508         vector<float> left, right;
 509         deinterleave_samples(samples, &left, &right);
 510         float *ptrs[] = { left.data(), right.data() };
 511         {
 512                 unique_lock<mutex> lock(audio_measure_mutex);
 513                 r128.process(left.size(), ptrs);
 514                 correlation.process_samples(samples);
 515         }
 516
 517         send_audio_level_callback();
 518 }
 519
 520 void AudioMixer::reset_meters()
 521 {
 522         unique_lock<mutex> lock(audio_measure_mutex);
 523         peak_resampler.reset();
 524         peak = 0.0f;
 525         r128.reset();
 526         r128.integr_start();
 527         correlation.reset();
 528 }
 529
 530 void AudioMixer::send_audio_level_callback()
 531 {
 532         if (audio_level_callback == nullptr) {
 533                 return;
 534         }
 535
 536         unique_lock<mutex> lock(audio_measure_mutex);
 537         double loudness_s = r128.loudness_S();
 538         double loudness_i = r128.integrated();
 539         double loudness_range_low = r128.range_min();
 540         double loudness_range_high = r128.range_max();
 541
 542         audio_level_callback(loudness_s, to_db(peak),
 543                 loudness_i, loudness_range_low, loudness_range_high,
 544                 gain_staging_db,
 545                 to_db(final_makeup_gain),
 546                 correlation.get_correlation());
 547 }
 548
 549 map<DeviceSpec, DeviceInfo> AudioMixer::get_devices() const
 550 {
 551         lock_guard<timed_mutex> lock(audio_mutex);
 552         return get_devices_mutex_held();
 553 }
 554
 555 map<DeviceSpec, DeviceInfo> AudioMixer::get_devices_mutex_held() const
 556 {
 557         map<DeviceSpec, DeviceInfo> devices;
 558         for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
 559                 const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index };
 560                 const AudioDevice *device = &video_cards[card_index];
 561                 DeviceInfo info;
 562                 info.name = device->name;
 563                 info.num_channels = 8;  // FIXME: This is wrong for fake cards.
 564                 devices.insert(make_pair(spec, info));
 565         }
 566         for (unsigned card_index = 0; card_index < available_alsa_cards.size(); ++card_index) {
 567                 const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index };
 568                 const ALSAInput::Device &device = available_alsa_cards[card_index];
 569                 DeviceInfo info;
 570                 info.name = device.name + " (" + device.info + ")";
 571                 info.num_channels = device.num_channels;
 572                 devices.insert(make_pair(spec, info));
 573         }
 574         return devices;
 575 }
 576
 577 void AudioMixer::set_name(DeviceSpec device_spec, const string &name)
 578 {
 579         AudioDevice *device = find_audio_device(device_spec);
 580
 581         lock_guard<timed_mutex> lock(audio_mutex);
 582         device->name = name;
 583 }
 584
 585 void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
 586 {
 587         lock_guard<timed_mutex> lock(audio_mutex);
 588
 589         map<DeviceSpec, set<unsigned>> interesting_channels;
 590         for (const InputMapping::Bus &bus : new_input_mapping.buses) {
 591                 if (bus.device.type == InputSourceType::CAPTURE_CARD ||
 592                     bus.device.type == InputSourceType::ALSA_INPUT) {
 593                         for (unsigned channel = 0; channel < 2; ++channel) {
 594                                 if (bus.source_channel[channel] != -1) {
 595                                         interesting_channels[bus.device].insert(bus.source_channel[channel]);
 596                                 }
 597                         }
 598                 }
 599         }
 600
 601         // Reset resamplers for all cards that don't have the exact same state as before.
 602         for (const auto &spec_and_info : get_devices_mutex_held()) {
 603                 const DeviceSpec &device_spec = spec_and_info.first;
 604                 AudioDevice *device = find_audio_device(device_spec);
 605                 if (device->interesting_channels != interesting_channels[device_spec]) {
 606                         device->interesting_channels = interesting_channels[device_spec];
 607                         if (device_spec.type == InputSourceType::ALSA_INPUT) {
 608                                 reset_alsa_mutex_held(device_spec);
 609                         }
 610                         reset_resampler_mutex_held(device_spec);
 611                 }
 612         }
 613
 614         input_mapping = new_input_mapping;
 615 }
 616
 617 InputMapping AudioMixer::get_input_mapping() const
 618 {
 619         lock_guard<timed_mutex> lock(audio_mutex);
 620         return input_mapping;
 621 }