1 #include "audio_mixer.h"
4 #include <bmusb/bmusb.h>
26 using namespace bmusb;
28 using namespace std::chrono;
29 using namespace std::placeholders;
33 // TODO: If these prove to be a bottleneck, they can be SSSE3-optimized
34 // (usually including multiple channels at a time).
36 void convert_fixed16_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
37 const uint8_t *src, size_t in_channel, size_t in_num_channels,
40 assert(in_channel < in_num_channels);
41 assert(out_channel < out_num_channels);
42 src += in_channel * 2;
45 for (size_t i = 0; i < num_samples; ++i) {
46 int16_t s = le16toh(*(int16_t *)src);
47 *dst = s * (1.0f / 32768.0f);
49 src += 2 * in_num_channels;
50 dst += out_num_channels;
54 void convert_fixed24_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
55 const uint8_t *src, size_t in_channel, size_t in_num_channels,
58 assert(in_channel < in_num_channels);
59 assert(out_channel < out_num_channels);
60 src += in_channel * 3;
63 for (size_t i = 0; i < num_samples; ++i) {
67 uint32_t s = s1 | (s1 << 8) | (s2 << 16) | (s3 << 24);
68 *dst = int(s) * (1.0f / 2147483648.0f);
70 src += 3 * in_num_channels;
71 dst += out_num_channels;
75 void convert_fixed32_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
76 const uint8_t *src, size_t in_channel, size_t in_num_channels,
79 assert(in_channel < in_num_channels);
80 assert(out_channel < out_num_channels);
81 src += in_channel * 4;
84 for (size_t i = 0; i < num_samples; ++i) {
85 int32_t s = le32toh(*(int32_t *)src);
86 *dst = s * (1.0f / 2147483648.0f);
88 src += 4 * in_num_channels;
89 dst += out_num_channels;
93 float find_peak_plain(const float *samples, size_t num_samples) __attribute__((unused));
95 float find_peak_plain(const float *samples, size_t num_samples)
97 float m = fabs(samples[0]);
98 for (size_t i = 1; i < num_samples; ++i) {
99 m = max(m, fabs(samples[i]));
105 static inline float horizontal_max(__m128 m)
107 __m128 tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 0, 3, 2));
108 m = _mm_max_ps(m, tmp);
109 tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 3, 0, 1));
110 m = _mm_max_ps(m, tmp);
111 return _mm_cvtss_f32(m);
114 float find_peak(const float *samples, size_t num_samples)
116 const __m128 abs_mask = _mm_castsi128_ps(_mm_set1_epi32(0x7fffffffu));
117 __m128 m = _mm_setzero_ps();
118 for (size_t i = 0; i < (num_samples & ~3); i += 4) {
119 __m128 x = _mm_loadu_ps(samples + i);
120 x = _mm_and_ps(x, abs_mask);
121 m = _mm_max_ps(m, x);
123 float result = horizontal_max(m);
125 for (size_t i = (num_samples & ~3); i < num_samples; ++i) {
126 result = max(result, fabs(samples[i]));
130 // Self-test. We should be bit-exact the same.
131 float reference_result = find_peak_plain(samples, num_samples);
132 if (result != reference_result) {
133 fprintf(stderr, "Error: Peak is %f [%f %f %f %f]; should be %f.\n",
135 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(0, 0, 0, 0))),
136 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 1, 1, 1))),
137 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 2, 2, 2))),
138 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(3, 3, 3, 3))),
146 float find_peak(const float *samples, size_t num_samples)
148 return find_peak_plain(samples, num_samples);
152 void deinterleave_samples(const vector<float> &in, vector<float> *out_l, vector<float> *out_r)
154 size_t num_samples = in.size() / 2;
155 out_l->resize(num_samples);
156 out_r->resize(num_samples);
158 const float *inptr = in.data();
159 float *lptr = &(*out_l)[0];
160 float *rptr = &(*out_r)[0];
161 for (size_t i = 0; i < num_samples; ++i) {
169 AudioMixer::AudioMixer(unsigned num_cards)
170 : num_cards(num_cards),
171 limiter(OUTPUT_FREQUENCY),
172 correlation(OUTPUT_FREQUENCY)
174 for (unsigned bus_index = 0; bus_index < MAX_BUSES; ++bus_index) {
175 locut[bus_index].init(FILTER_HPF, 2);
176 eq[bus_index][EQ_BAND_BASS].init(FILTER_LOW_SHELF, 1);
177 // Note: EQ_BAND_MID isn't used (see comments in apply_eq()).
178 eq[bus_index][EQ_BAND_TREBLE].init(FILTER_HIGH_SHELF, 1);
179 compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
180 level_compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
182 set_bus_settings(bus_index, get_default_bus_settings());
184 set_limiter_enabled(global_flags.limiter_enabled);
185 set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
187 if (!global_flags.input_mapping_filename.empty()) {
188 current_mapping_mode = MappingMode::MULTICHANNEL;
189 InputMapping new_input_mapping;
190 if (!load_input_mapping_from_file(get_devices(),
191 global_flags.input_mapping_filename,
192 &new_input_mapping)) {
193 fprintf(stderr, "Failed to load input mapping from '%s', exiting.\n",
194 global_flags.input_mapping_filename.c_str());
197 set_input_mapping(new_input_mapping);
199 set_simple_input(/*card_index=*/0);
200 if (global_flags.multichannel_mapping_mode) {
201 current_mapping_mode = MappingMode::MULTICHANNEL;
205 r128.init(2, OUTPUT_FREQUENCY);
208 // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
209 // and there's a limit to how important the peak meter is.
210 peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
212 global_audio_mixer = this;
216 void AudioMixer::reset_resampler(DeviceSpec device_spec)
218 lock_guard<timed_mutex> lock(audio_mutex);
219 reset_resampler_mutex_held(device_spec);
222 void AudioMixer::reset_resampler_mutex_held(DeviceSpec device_spec)
224 AudioDevice *device = find_audio_device(device_spec);
226 if (device->interesting_channels.empty()) {
227 device->resampling_queue.reset();
229 // TODO: ResamplingQueue should probably take the full device spec.
230 // (It's only used for console output, though.)
231 device->resampling_queue.reset(new ResamplingQueue(
232 device_spec.index, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size(),
233 global_flags.audio_queue_length_ms * 0.001));
237 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)
239 AudioDevice *device = find_audio_device(device_spec);
241 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
242 if (!lock.try_lock_for(chrono::milliseconds(10))) {
245 if (device->resampling_queue == nullptr) {
246 // No buses use this device; throw it away.
250 unsigned num_channels = device->interesting_channels.size();
251 assert(num_channels > 0);
253 // Convert the audio to fp32.
254 unique_ptr<float[]> audio(new float[num_samples * num_channels]);
255 unsigned channel_index = 0;
256 for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) {
257 switch (audio_format.bits_per_sample) {
259 assert(num_samples == 0);
262 convert_fixed16_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
265 convert_fixed24_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
268 convert_fixed32_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
271 fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
277 device->resampling_queue->add_input_samples(frame_time, audio.get(), num_samples, ResamplingQueue::ADJUST_RATE);
281 bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length)
283 AudioDevice *device = find_audio_device(device_spec);
285 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
286 if (!lock.try_lock_for(chrono::milliseconds(10))) {
289 if (device->resampling_queue == nullptr) {
290 // No buses use this device; throw it away.
294 unsigned num_channels = device->interesting_channels.size();
295 assert(num_channels > 0);
297 vector<float> silence(samples_per_frame * num_channels, 0.0f);
298 for (unsigned i = 0; i < num_frames; ++i) {
299 device->resampling_queue->add_input_samples(steady_clock::now(), silence.data(), samples_per_frame, ResamplingQueue::DO_NOT_ADJUST_RATE);
304 bool AudioMixer::silence_card(DeviceSpec device_spec, bool silence)
306 AudioDevice *device = find_audio_device(device_spec);
308 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
309 if (!lock.try_lock_for(chrono::milliseconds(10))) {
313 if (device->silenced && !silence) {
314 reset_resampler_mutex_held(device_spec);
316 device->silenced = silence;
320 AudioMixer::BusSettings AudioMixer::get_default_bus_settings()
322 BusSettings settings;
323 settings.fader_volume_db = 0.0f;
324 settings.muted = false;
325 settings.locut_enabled = global_flags.locut_enabled;
326 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
327 settings.eq_level_db[band_index] = 0.0f;
329 settings.gain_staging_db = global_flags.initial_gain_staging_db;
330 settings.level_compressor_enabled = global_flags.gain_staging_auto;
331 settings.compressor_threshold_dbfs = ref_level_dbfs - 12.0f; // -12 dB.
332 settings.compressor_enabled = global_flags.compressor_enabled;
336 AudioMixer::BusSettings AudioMixer::get_bus_settings(unsigned bus_index) const
338 lock_guard<timed_mutex> lock(audio_mutex);
339 BusSettings settings;
340 settings.fader_volume_db = fader_volume_db[bus_index];
341 settings.muted = mute[bus_index];
342 settings.locut_enabled = locut_enabled[bus_index];
343 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
344 settings.eq_level_db[band_index] = eq_level_db[bus_index][band_index];
346 settings.gain_staging_db = gain_staging_db[bus_index];
347 settings.level_compressor_enabled = level_compressor_enabled[bus_index];
348 settings.compressor_threshold_dbfs = compressor_threshold_dbfs[bus_index];
349 settings.compressor_enabled = compressor_enabled[bus_index];
353 void AudioMixer::set_bus_settings(unsigned bus_index, const AudioMixer::BusSettings &settings)
355 lock_guard<timed_mutex> lock(audio_mutex);
356 fader_volume_db[bus_index] = settings.fader_volume_db;
357 mute[bus_index] = settings.muted;
358 locut_enabled[bus_index] = settings.locut_enabled;
359 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
360 eq_level_db[bus_index][band_index] = settings.eq_level_db[band_index];
362 gain_staging_db[bus_index] = settings.gain_staging_db;
363 last_gain_staging_db[bus_index] = gain_staging_db[bus_index];
364 level_compressor_enabled[bus_index] = settings.level_compressor_enabled;
365 compressor_threshold_dbfs[bus_index] = settings.compressor_threshold_dbfs;
366 compressor_enabled[bus_index] = settings.compressor_enabled;
369 AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
371 switch (device.type) {
372 case InputSourceType::CAPTURE_CARD:
373 return &video_cards[device.index];
374 case InputSourceType::ALSA_INPUT:
375 return &alsa_inputs[device.index];
376 case InputSourceType::SILENCE:
383 // Get a pointer to the given channel from the given device.
384 // The channel must be picked out earlier and resampled.
385 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)
387 static float zero = 0.0f;
388 if (source_channel == -1 || device_spec.type == InputSourceType::SILENCE) {
393 AudioDevice *device = find_audio_device(device_spec);
394 assert(device->interesting_channels.count(source_channel) != 0);
395 unsigned channel_index = 0;
396 for (int channel : device->interesting_channels) {
397 if (channel == source_channel) break;
400 assert(channel_index < device->interesting_channels.size());
401 const auto it = samples_card.find(device_spec);
402 assert(it != samples_card.end());
403 *srcptr = &(it->second)[channel_index];
404 *stride = device->interesting_channels.size();
407 // TODO: Can be SSSE3-optimized if need be.
408 void AudioMixer::fill_audio_bus(const map<DeviceSpec, vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output)
410 if (bus.device.type == InputSourceType::SILENCE) {
411 memset(output, 0, num_samples * 2 * sizeof(*output));
413 assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
414 bus.device.type == InputSourceType::ALSA_INPUT);
415 const float *lsrc, *rsrc;
416 unsigned lstride, rstride;
417 float *dptr = output;
418 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[0], &lsrc, &lstride);
419 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[1], &rsrc, &rstride);
420 for (unsigned i = 0; i < num_samples; ++i) {
429 vector<DeviceSpec> AudioMixer::get_active_devices() const
431 vector<DeviceSpec> ret;
432 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
433 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
434 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
435 ret.push_back(device_spec);
438 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
439 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
440 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
441 ret.push_back(device_spec);
449 void apply_gain(float db, float last_db, vector<float> *samples)
451 if (fabs(db - last_db) < 1e-3) {
452 // Constant over this frame.
453 const float gain = from_db(db);
454 for (size_t i = 0; i < samples->size(); ++i) {
455 (*samples)[i] *= gain;
458 // We need to do a fade.
459 unsigned num_samples = samples->size() / 2;
460 float gain = from_db(last_db);
461 const float gain_inc = pow(from_db(db - last_db), 1.0 / num_samples);
462 for (size_t i = 0; i < num_samples; ++i) {
463 (*samples)[i * 2 + 0] *= gain;
464 (*samples)[i * 2 + 1] *= gain;
472 vector<float> AudioMixer::get_output(steady_clock::time_point ts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
474 map<DeviceSpec, vector<float>> samples_card;
475 vector<float> samples_bus;
477 lock_guard<timed_mutex> lock(audio_mutex);
479 // Pick out all the interesting channels from all the cards.
480 for (const DeviceSpec &device_spec : get_active_devices()) {
481 AudioDevice *device = find_audio_device(device_spec);
482 samples_card[device_spec].resize(num_samples * device->interesting_channels.size());
483 if (device->silenced) {
484 memset(&samples_card[device_spec][0], 0, samples_card[device_spec].size() * sizeof(float));
486 device->resampling_queue->get_output_samples(
488 &samples_card[device_spec][0],
490 rate_adjustment_policy);
494 vector<float> samples_out, left, right;
495 samples_out.resize(num_samples * 2);
496 samples_bus.resize(num_samples * 2);
497 for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
498 fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, &samples_bus[0]);
499 apply_eq(bus_index, &samples_bus);
502 lock_guard<mutex> lock(compressor_mutex);
504 // Apply a level compressor to get the general level right.
505 // Basically, if it's over about -40 dBFS, we squeeze it down to that level
506 // (or more precisely, near it, since we don't use infinite ratio),
507 // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
508 // entirely arbitrary, but from practical tests with speech, it seems to
509 // put ut around -23 LUFS, so it's a reasonable starting point for later use.
510 if (level_compressor_enabled[bus_index]) {
511 float threshold = 0.01f; // -40 dBFS.
513 float attack_time = 0.5f;
514 float release_time = 20.0f;
515 float makeup_gain = from_db(ref_level_dbfs - (-40.0f)); // +26 dB.
516 level_compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
517 gain_staging_db[bus_index] = to_db(level_compressor[bus_index]->get_attenuation() * makeup_gain);
519 // Just apply the gain we already had.
520 float db = gain_staging_db[bus_index];
521 float last_db = last_gain_staging_db[bus_index];
522 apply_gain(db, last_db, &samples_bus);
524 last_gain_staging_db[bus_index] = gain_staging_db[bus_index];
527 printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
528 level_compressor.get_level(), to_db(level_compressor.get_level()),
529 level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()),
530 to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
533 // The real compressor.
534 if (compressor_enabled[bus_index]) {
535 float threshold = from_db(compressor_threshold_dbfs[bus_index]);
537 float attack_time = 0.005f;
538 float release_time = 0.040f;
539 float makeup_gain = 2.0f; // +6 dB.
540 compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
541 // compressor_att = compressor.get_attenuation();
545 add_bus_to_master(bus_index, samples_bus, &samples_out);
546 deinterleave_samples(samples_bus, &left, &right);
547 measure_bus_levels(bus_index, left, right);
551 lock_guard<mutex> lock(compressor_mutex);
553 // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
554 // Note that since ratio is not infinite, we could go slightly higher than this.
555 if (limiter_enabled) {
556 float threshold = from_db(limiter_threshold_dbfs);
558 float attack_time = 0.0f; // Instant.
559 float release_time = 0.020f;
560 float makeup_gain = 1.0f; // 0 dB.
561 limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
562 // limiter_att = limiter.get_attenuation();
565 // printf("limiter=%+5.1f compressor=%+5.1f\n", to_db(limiter_att), to_db(compressor_att));
568 // At this point, we are most likely close to +0 LU (at least if the
569 // faders sum to 0 dB and the compressors are on), but all of our
570 // measurements have been on raw sample values, not R128 values.
571 // So we have a final makeup gain to get us to +0 LU; the gain
572 // adjustments required should be relatively small, and also, the
573 // offset shouldn't change much (only if the type of audio changes
574 // significantly). Thus, we shoot for updating this value basically
575 // “whenever we process buffers”, since the R128 calculation isn't exactly
576 // something we get out per-sample.
578 // Note that there's a feedback loop here, so we choose a very slow filter
579 // (half-time of 30 seconds).
580 double target_loudness_factor, alpha;
581 double loudness_lu = r128.loudness_M() - ref_level_lufs;
582 target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
584 // If we're outside +/- 5 LU (after correction), we don't count it as
585 // a normal signal (probably silence) and don't change the
586 // correction factor; just apply what we already have.
587 if (fabs(loudness_lu) >= 5.0 || !final_makeup_gain_auto) {
590 // Formula adapted from
591 // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
592 const double half_time_s = 30.0;
593 const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
594 alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
598 lock_guard<mutex> lock(compressor_mutex);
599 double m = final_makeup_gain;
600 for (size_t i = 0; i < samples_out.size(); i += 2) {
601 samples_out[i + 0] *= m;
602 samples_out[i + 1] *= m;
603 m += (target_loudness_factor - m) * alpha;
605 final_makeup_gain = m;
608 update_meters(samples_out);
615 void apply_filter_fade(StereoFilter *filter, float *data, unsigned num_samples, float cutoff_hz, float db, float last_db)
617 // A granularity of 32 samples is an okay tradeoff between speed and
618 // smoothness; recalculating the filters is pretty expensive, so it's
619 // good that we don't do this all the time.
620 static constexpr unsigned filter_granularity_samples = 32;
622 const float cutoff_linear = cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY;
623 if (fabs(db - last_db) < 1e-3) {
624 // Constant over this frame.
625 if (fabs(db) > 0.01f) {
626 filter->render(data, num_samples, cutoff_linear, 0.5f, db / 40.0f);
629 // We need to do a fade. (Rounding up avoids division by zero.)
630 unsigned num_blocks = (num_samples + filter_granularity_samples - 1) / filter_granularity_samples;
631 const float inc_db_norm = (db - last_db) / 40.0f / num_blocks;
632 float db_norm = db / 40.0f;
633 for (size_t i = 0; i < num_samples; i += filter_granularity_samples) {
634 size_t samples_this_block = std::min<size_t>(num_samples - i, filter_granularity_samples);
635 filter->render(data + i * 2, samples_this_block, cutoff_linear, 0.5f, db_norm);
636 db_norm += inc_db_norm;
643 void AudioMixer::apply_eq(unsigned bus_index, vector<float> *samples_bus)
645 constexpr float bass_freq_hz = 200.0f;
646 constexpr float treble_freq_hz = 4700.0f;
648 // Cut away everything under 120 Hz (or whatever the cutoff is);
649 // we don't need it for voice, and it will reduce headroom
650 // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
651 // should be dampened.)
652 if (locut_enabled[bus_index]) {
653 locut[bus_index].render(samples_bus->data(), samples_bus->size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
656 // Apply the rest of the EQ. Since we only have a simple three-band EQ,
657 // we can implement it with two shelf filters. We use a simple gain to
658 // set the mid-level filter, and then offset the low and high bands
659 // from that if we need to. (We could perhaps have folded the gain into
660 // the next part, but it's so cheap that the trouble isn't worth it.)
662 // If any part of the EQ has changed appreciably since last frame,
663 // we fade smoothly during the course of this frame.
664 const float bass_db = eq_level_db[bus_index][EQ_BAND_BASS];
665 const float mid_db = eq_level_db[bus_index][EQ_BAND_MID];
666 const float treble_db = eq_level_db[bus_index][EQ_BAND_TREBLE];
668 const float last_bass_db = last_eq_level_db[bus_index][EQ_BAND_BASS];
669 const float last_mid_db = last_eq_level_db[bus_index][EQ_BAND_MID];
670 const float last_treble_db = last_eq_level_db[bus_index][EQ_BAND_TREBLE];
672 assert(samples_bus->size() % 2 == 0);
673 const unsigned num_samples = samples_bus->size() / 2;
675 apply_gain(mid_db, last_mid_db, samples_bus);
677 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);
678 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);
680 last_eq_level_db[bus_index][EQ_BAND_BASS] = bass_db;
681 last_eq_level_db[bus_index][EQ_BAND_MID] = mid_db;
682 last_eq_level_db[bus_index][EQ_BAND_TREBLE] = treble_db;
685 void AudioMixer::add_bus_to_master(unsigned bus_index, const vector<float> &samples_bus, vector<float> *samples_out)
687 assert(samples_bus.size() == samples_out->size());
688 assert(samples_bus.size() % 2 == 0);
689 unsigned num_samples = samples_bus.size() / 2;
690 const float new_volume_db = mute[bus_index] ? -90.0f : fader_volume_db[bus_index].load();
691 if (fabs(new_volume_db - last_fader_volume_db[bus_index]) > 1e-3) {
692 // The volume has changed; do a fade over the course of this frame.
693 // (We might have some numerical issues here, but it seems to sound OK.)
694 // For the purpose of fading here, the silence floor is set to -90 dB
695 // (the fader only goes to -84).
696 float old_volume = from_db(max<float>(last_fader_volume_db[bus_index], -90.0f));
697 float volume = from_db(max<float>(new_volume_db, -90.0f));
699 float volume_inc = pow(volume / old_volume, 1.0 / num_samples);
701 if (bus_index == 0) {
702 for (unsigned i = 0; i < num_samples; ++i) {
703 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
704 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
705 volume *= volume_inc;
708 for (unsigned i = 0; i < num_samples; ++i) {
709 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
710 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
711 volume *= volume_inc;
714 } else if (new_volume_db > -90.0f) {
715 float volume = from_db(new_volume_db);
716 if (bus_index == 0) {
717 for (unsigned i = 0; i < num_samples; ++i) {
718 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
719 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
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;
729 last_fader_volume_db[bus_index] = new_volume_db;
732 void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right)
734 assert(left.size() == right.size());
735 const float volume = mute[bus_index] ? 0.0f : from_db(fader_volume_db[bus_index]);
736 const float peak_levels[2] = {
737 find_peak(left.data(), left.size()) * volume,
738 find_peak(right.data(), right.size()) * volume
740 for (unsigned channel = 0; channel < 2; ++channel) {
741 // Compute the current value, including hold and falloff.
742 // The constants are borrowed from zita-mu1 by Fons Adriaensen.
743 static constexpr float hold_sec = 0.5f;
744 static constexpr float falloff_db_sec = 15.0f; // dB/sec falloff after hold.
746 PeakHistory &history = peak_history[bus_index][channel];
747 history.historic_peak = max(history.historic_peak, peak_levels[channel]);
748 if (history.age_seconds < hold_sec) {
749 current_peak = history.last_peak;
751 current_peak = history.last_peak * from_db(-falloff_db_sec * (history.age_seconds - hold_sec));
754 // See if we have a new peak to replace the old (possibly falling) one.
755 if (peak_levels[channel] > current_peak) {
756 history.last_peak = peak_levels[channel];
757 history.age_seconds = 0.0f; // Not 100% correct, but more than good enough given our frame sizes.
758 current_peak = peak_levels[channel];
760 history.age_seconds += float(left.size()) / OUTPUT_FREQUENCY;
762 history.current_level = peak_levels[channel];
763 history.current_peak = current_peak;
767 void AudioMixer::update_meters(const vector<float> &samples)
769 // Upsample 4x to find interpolated peak.
770 peak_resampler.inp_data = const_cast<float *>(samples.data());
771 peak_resampler.inp_count = samples.size() / 2;
773 vector<float> interpolated_samples;
774 interpolated_samples.resize(samples.size());
776 lock_guard<mutex> lock(audio_measure_mutex);
778 while (peak_resampler.inp_count > 0) { // About four iterations.
779 peak_resampler.out_data = &interpolated_samples[0];
780 peak_resampler.out_count = interpolated_samples.size() / 2;
781 peak_resampler.process();
782 size_t out_stereo_samples = interpolated_samples.size() / 2 - peak_resampler.out_count;
783 peak = max<float>(peak, find_peak(interpolated_samples.data(), out_stereo_samples * 2));
784 peak_resampler.out_data = nullptr;
788 // Find R128 levels and L/R correlation.
789 vector<float> left, right;
790 deinterleave_samples(samples, &left, &right);
791 float *ptrs[] = { left.data(), right.data() };
793 lock_guard<mutex> lock(audio_measure_mutex);
794 r128.process(left.size(), ptrs);
795 correlation.process_samples(samples);
798 send_audio_level_callback();
801 void AudioMixer::reset_meters()
803 lock_guard<mutex> lock(audio_measure_mutex);
804 peak_resampler.reset();
811 void AudioMixer::send_audio_level_callback()
813 if (audio_level_callback == nullptr) {
817 lock_guard<mutex> lock(audio_measure_mutex);
818 double loudness_s = r128.loudness_S();
819 double loudness_i = r128.integrated();
820 double loudness_range_low = r128.range_min();
821 double loudness_range_high = r128.range_max();
823 vector<BusLevel> bus_levels;
824 bus_levels.resize(input_mapping.buses.size());
826 lock_guard<mutex> lock(compressor_mutex);
827 for (unsigned bus_index = 0; bus_index < bus_levels.size(); ++bus_index) {
828 bus_levels[bus_index].current_level_dbfs[0] = to_db(peak_history[bus_index][0].current_level);
829 bus_levels[bus_index].current_level_dbfs[1] = to_db(peak_history[bus_index][1].current_level);
830 bus_levels[bus_index].peak_level_dbfs[0] = to_db(peak_history[bus_index][0].current_peak);
831 bus_levels[bus_index].peak_level_dbfs[1] = to_db(peak_history[bus_index][1].current_peak);
832 bus_levels[bus_index].historic_peak_dbfs = to_db(
833 max(peak_history[bus_index][0].historic_peak,
834 peak_history[bus_index][1].historic_peak));
835 bus_levels[bus_index].gain_staging_db = gain_staging_db[bus_index];
836 if (compressor_enabled[bus_index]) {
837 bus_levels[bus_index].compressor_attenuation_db = -to_db(compressor[bus_index]->get_attenuation());
839 bus_levels[bus_index].compressor_attenuation_db = 0.0;
844 audio_level_callback(loudness_s, to_db(peak), bus_levels,
845 loudness_i, loudness_range_low, loudness_range_high,
846 to_db(final_makeup_gain),
847 correlation.get_correlation());
850 map<DeviceSpec, DeviceInfo> AudioMixer::get_devices()
852 lock_guard<timed_mutex> lock(audio_mutex);
854 map<DeviceSpec, DeviceInfo> devices;
855 for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
856 const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index };
857 const AudioDevice *device = &video_cards[card_index];
859 info.display_name = device->display_name;
860 info.num_channels = 8;
861 devices.insert(make_pair(spec, info));
863 vector<ALSAPool::Device> available_alsa_devices = alsa_pool.get_devices();
864 for (unsigned card_index = 0; card_index < available_alsa_devices.size(); ++card_index) {
865 const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index };
866 const ALSAPool::Device &device = available_alsa_devices[card_index];
868 info.display_name = device.display_name();
869 info.num_channels = device.num_channels;
870 info.alsa_name = device.name;
871 info.alsa_info = device.info;
872 info.alsa_address = device.address;
873 devices.insert(make_pair(spec, info));
878 void AudioMixer::set_display_name(DeviceSpec device_spec, const string &name)
880 AudioDevice *device = find_audio_device(device_spec);
882 lock_guard<timed_mutex> lock(audio_mutex);
883 device->display_name = name;
886 void AudioMixer::serialize_device(DeviceSpec device_spec, DeviceSpecProto *device_spec_proto)
888 lock_guard<timed_mutex> lock(audio_mutex);
889 switch (device_spec.type) {
890 case InputSourceType::SILENCE:
891 device_spec_proto->set_type(DeviceSpecProto::SILENCE);
893 case InputSourceType::CAPTURE_CARD:
894 device_spec_proto->set_type(DeviceSpecProto::CAPTURE_CARD);
895 device_spec_proto->set_index(device_spec.index);
896 device_spec_proto->set_display_name(video_cards[device_spec.index].display_name);
898 case InputSourceType::ALSA_INPUT:
899 alsa_pool.serialize_device(device_spec.index, device_spec_proto);
904 void AudioMixer::set_simple_input(unsigned card_index)
906 InputMapping new_input_mapping;
907 InputMapping::Bus input;
909 input.device.type = InputSourceType::CAPTURE_CARD;
910 input.device.index = card_index;
911 input.source_channel[0] = 0;
912 input.source_channel[1] = 1;
914 new_input_mapping.buses.push_back(input);
916 lock_guard<timed_mutex> lock(audio_mutex);
917 current_mapping_mode = MappingMode::SIMPLE;
918 set_input_mapping_lock_held(new_input_mapping);
919 fader_volume_db[0] = 0.0f;
922 unsigned AudioMixer::get_simple_input() const
924 lock_guard<timed_mutex> lock(audio_mutex);
925 if (input_mapping.buses.size() == 1 &&
926 input_mapping.buses[0].device.type == InputSourceType::CAPTURE_CARD &&
927 input_mapping.buses[0].source_channel[0] == 0 &&
928 input_mapping.buses[0].source_channel[1] == 1) {
929 return input_mapping.buses[0].device.index;
931 return numeric_limits<unsigned>::max();
935 void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
937 lock_guard<timed_mutex> lock(audio_mutex);
938 set_input_mapping_lock_held(new_input_mapping);
939 current_mapping_mode = MappingMode::MULTICHANNEL;
942 AudioMixer::MappingMode AudioMixer::get_mapping_mode() const
944 lock_guard<timed_mutex> lock(audio_mutex);
945 return current_mapping_mode;
948 void AudioMixer::set_input_mapping_lock_held(const InputMapping &new_input_mapping)
950 map<DeviceSpec, set<unsigned>> interesting_channels;
951 for (const InputMapping::Bus &bus : new_input_mapping.buses) {
952 if (bus.device.type == InputSourceType::CAPTURE_CARD ||
953 bus.device.type == InputSourceType::ALSA_INPUT) {
954 for (unsigned channel = 0; channel < 2; ++channel) {
955 if (bus.source_channel[channel] != -1) {
956 interesting_channels[bus.device].insert(bus.source_channel[channel]);
962 // Reset resamplers for all cards that don't have the exact same state as before.
963 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
964 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
965 AudioDevice *device = find_audio_device(device_spec);
966 if (device->interesting_channels != interesting_channels[device_spec]) {
967 device->interesting_channels = interesting_channels[device_spec];
968 reset_resampler_mutex_held(device_spec);
971 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
972 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
973 AudioDevice *device = find_audio_device(device_spec);
974 if (interesting_channels[device_spec].empty()) {
975 alsa_pool.release_device(card_index);
977 alsa_pool.hold_device(card_index);
979 if (device->interesting_channels != interesting_channels[device_spec]) {
980 device->interesting_channels = interesting_channels[device_spec];
981 alsa_pool.reset_device(device_spec.index);
982 reset_resampler_mutex_held(device_spec);
986 input_mapping = new_input_mapping;
989 InputMapping AudioMixer::get_input_mapping() const
991 lock_guard<timed_mutex> lock(audio_mutex);
992 return input_mapping;
995 unsigned AudioMixer::num_buses() const
997 lock_guard<timed_mutex> lock(audio_mutex);
998 return input_mapping.buses.size();
1001 void AudioMixer::reset_peak(unsigned bus_index)
1003 lock_guard<timed_mutex> lock(audio_mutex);
1004 for (unsigned channel = 0; channel < 2; ++channel) {
1005 PeakHistory &history = peak_history[bus_index][channel];
1006 history.current_level = 0.0f;
1007 history.historic_peak = 0.0f;
1008 history.current_peak = 0.0f;
1009 history.last_peak = 0.0f;
1010 history.age_seconds = 0.0f;
1014 AudioMixer *global_audio_mixer = nullptr;