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 r128.init(2, OUTPUT_FREQUENCY);
190 // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
191 // and there's a limit to how important the peak meter is.
192 peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
194 global_audio_mixer = this;
197 if (!global_flags.input_mapping_filename.empty()) {
198 // Must happen after ALSAPool is initialized, as it needs to know the card list.
199 current_mapping_mode = MappingMode::MULTICHANNEL;
200 InputMapping new_input_mapping;
201 if (!load_input_mapping_from_file(get_devices(),
202 global_flags.input_mapping_filename,
203 &new_input_mapping)) {
204 fprintf(stderr, "Failed to load input mapping from '%s', exiting.\n",
205 global_flags.input_mapping_filename.c_str());
208 set_input_mapping(new_input_mapping);
210 set_simple_input(/*card_index=*/0);
211 if (global_flags.multichannel_mapping_mode) {
212 current_mapping_mode = MappingMode::MULTICHANNEL;
217 void AudioMixer::reset_resampler(DeviceSpec device_spec)
219 lock_guard<timed_mutex> lock(audio_mutex);
220 reset_resampler_mutex_held(device_spec);
223 void AudioMixer::reset_resampler_mutex_held(DeviceSpec device_spec)
225 AudioDevice *device = find_audio_device(device_spec);
227 if (device->interesting_channels.empty()) {
228 device->resampling_queue.reset();
230 // TODO: ResamplingQueue should probably take the full device spec.
231 // (It's only used for console output, though.)
232 device->resampling_queue.reset(new ResamplingQueue(
233 device_spec.index, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size(),
234 global_flags.audio_queue_length_ms * 0.001));
238 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)
240 AudioDevice *device = find_audio_device(device_spec);
242 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
243 if (!lock.try_lock_for(chrono::milliseconds(10))) {
246 if (device->resampling_queue == nullptr) {
247 // No buses use this device; throw it away.
251 unsigned num_channels = device->interesting_channels.size();
252 assert(num_channels > 0);
254 // Convert the audio to fp32.
255 unique_ptr<float[]> audio(new float[num_samples * num_channels]);
256 unsigned channel_index = 0;
257 for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) {
258 switch (audio_format.bits_per_sample) {
260 assert(num_samples == 0);
263 convert_fixed16_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
266 convert_fixed24_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
269 convert_fixed32_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
272 fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
277 // If we changed frequency since last frame, we'll need to reset the resampler.
278 if (audio_format.sample_rate != device->capture_frequency) {
279 device->capture_frequency = audio_format.sample_rate;
280 reset_resampler_mutex_held(device_spec);
284 device->resampling_queue->add_input_samples(frame_time, audio.get(), num_samples, ResamplingQueue::ADJUST_RATE);
288 bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length)
290 AudioDevice *device = find_audio_device(device_spec);
292 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
293 if (!lock.try_lock_for(chrono::milliseconds(10))) {
296 if (device->resampling_queue == nullptr) {
297 // No buses use this device; throw it away.
301 unsigned num_channels = device->interesting_channels.size();
302 assert(num_channels > 0);
304 vector<float> silence(samples_per_frame * num_channels, 0.0f);
305 for (unsigned i = 0; i < num_frames; ++i) {
306 device->resampling_queue->add_input_samples(steady_clock::now(), silence.data(), samples_per_frame, ResamplingQueue::DO_NOT_ADJUST_RATE);
311 bool AudioMixer::silence_card(DeviceSpec device_spec, bool silence)
313 AudioDevice *device = find_audio_device(device_spec);
315 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
316 if (!lock.try_lock_for(chrono::milliseconds(10))) {
320 if (device->silenced && !silence) {
321 reset_resampler_mutex_held(device_spec);
323 device->silenced = silence;
327 AudioMixer::BusSettings AudioMixer::get_default_bus_settings()
329 BusSettings settings;
330 settings.fader_volume_db = 0.0f;
331 settings.muted = false;
332 settings.locut_enabled = global_flags.locut_enabled;
333 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
334 settings.eq_level_db[band_index] = 0.0f;
336 settings.gain_staging_db = global_flags.initial_gain_staging_db;
337 settings.level_compressor_enabled = global_flags.gain_staging_auto;
338 settings.compressor_threshold_dbfs = ref_level_dbfs - 12.0f; // -12 dB.
339 settings.compressor_enabled = global_flags.compressor_enabled;
343 AudioMixer::BusSettings AudioMixer::get_bus_settings(unsigned bus_index) const
345 lock_guard<timed_mutex> lock(audio_mutex);
346 BusSettings settings;
347 settings.fader_volume_db = fader_volume_db[bus_index];
348 settings.muted = mute[bus_index];
349 settings.locut_enabled = locut_enabled[bus_index];
350 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
351 settings.eq_level_db[band_index] = eq_level_db[bus_index][band_index];
353 settings.gain_staging_db = gain_staging_db[bus_index];
354 settings.level_compressor_enabled = level_compressor_enabled[bus_index];
355 settings.compressor_threshold_dbfs = compressor_threshold_dbfs[bus_index];
356 settings.compressor_enabled = compressor_enabled[bus_index];
360 void AudioMixer::set_bus_settings(unsigned bus_index, const AudioMixer::BusSettings &settings)
362 lock_guard<timed_mutex> lock(audio_mutex);
363 fader_volume_db[bus_index] = settings.fader_volume_db;
364 mute[bus_index] = settings.muted;
365 locut_enabled[bus_index] = settings.locut_enabled;
366 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
367 eq_level_db[bus_index][band_index] = settings.eq_level_db[band_index];
369 gain_staging_db[bus_index] = settings.gain_staging_db;
370 last_gain_staging_db[bus_index] = gain_staging_db[bus_index];
371 level_compressor_enabled[bus_index] = settings.level_compressor_enabled;
372 compressor_threshold_dbfs[bus_index] = settings.compressor_threshold_dbfs;
373 compressor_enabled[bus_index] = settings.compressor_enabled;
376 AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
378 switch (device.type) {
379 case InputSourceType::CAPTURE_CARD:
380 return &video_cards[device.index];
381 case InputSourceType::ALSA_INPUT:
382 return &alsa_inputs[device.index];
383 case InputSourceType::SILENCE:
390 // Get a pointer to the given channel from the given device.
391 // The channel must be picked out earlier and resampled.
392 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)
394 static float zero = 0.0f;
395 if (source_channel == -1 || device_spec.type == InputSourceType::SILENCE) {
400 AudioDevice *device = find_audio_device(device_spec);
401 assert(device->interesting_channels.count(source_channel) != 0);
402 unsigned channel_index = 0;
403 for (int channel : device->interesting_channels) {
404 if (channel == source_channel) break;
407 assert(channel_index < device->interesting_channels.size());
408 const auto it = samples_card.find(device_spec);
409 assert(it != samples_card.end());
410 *srcptr = &(it->second)[channel_index];
411 *stride = device->interesting_channels.size();
414 // TODO: Can be SSSE3-optimized if need be.
415 void AudioMixer::fill_audio_bus(const map<DeviceSpec, vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output)
417 if (bus.device.type == InputSourceType::SILENCE) {
418 memset(output, 0, num_samples * 2 * sizeof(*output));
420 assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
421 bus.device.type == InputSourceType::ALSA_INPUT);
422 const float *lsrc, *rsrc;
423 unsigned lstride, rstride;
424 float *dptr = output;
425 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[0], &lsrc, &lstride);
426 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[1], &rsrc, &rstride);
427 for (unsigned i = 0; i < num_samples; ++i) {
436 vector<DeviceSpec> AudioMixer::get_active_devices() const
438 vector<DeviceSpec> ret;
439 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
440 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
441 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
442 ret.push_back(device_spec);
445 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
446 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
447 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
448 ret.push_back(device_spec);
456 void apply_gain(float db, float last_db, vector<float> *samples)
458 if (fabs(db - last_db) < 1e-3) {
459 // Constant over this frame.
460 const float gain = from_db(db);
461 for (size_t i = 0; i < samples->size(); ++i) {
462 (*samples)[i] *= gain;
465 // We need to do a fade.
466 unsigned num_samples = samples->size() / 2;
467 float gain = from_db(last_db);
468 const float gain_inc = pow(from_db(db - last_db), 1.0 / num_samples);
469 for (size_t i = 0; i < num_samples; ++i) {
470 (*samples)[i * 2 + 0] *= gain;
471 (*samples)[i * 2 + 1] *= gain;
479 vector<float> AudioMixer::get_output(steady_clock::time_point ts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
481 map<DeviceSpec, vector<float>> samples_card;
482 vector<float> samples_bus;
484 lock_guard<timed_mutex> lock(audio_mutex);
486 // Pick out all the interesting channels from all the cards.
487 for (const DeviceSpec &device_spec : get_active_devices()) {
488 AudioDevice *device = find_audio_device(device_spec);
489 samples_card[device_spec].resize(num_samples * device->interesting_channels.size());
490 if (device->silenced) {
491 memset(&samples_card[device_spec][0], 0, samples_card[device_spec].size() * sizeof(float));
493 device->resampling_queue->get_output_samples(
495 &samples_card[device_spec][0],
497 rate_adjustment_policy);
501 vector<float> samples_out, left, right;
502 samples_out.resize(num_samples * 2);
503 samples_bus.resize(num_samples * 2);
504 for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
505 fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, &samples_bus[0]);
506 apply_eq(bus_index, &samples_bus);
509 lock_guard<mutex> lock(compressor_mutex);
511 // Apply a level compressor to get the general level right.
512 // Basically, if it's over about -40 dBFS, we squeeze it down to that level
513 // (or more precisely, near it, since we don't use infinite ratio),
514 // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
515 // entirely arbitrary, but from practical tests with speech, it seems to
516 // put ut around -23 LUFS, so it's a reasonable starting point for later use.
517 if (level_compressor_enabled[bus_index]) {
518 float threshold = 0.01f; // -40 dBFS.
520 float attack_time = 0.5f;
521 float release_time = 20.0f;
522 float makeup_gain = from_db(ref_level_dbfs - (-40.0f)); // +26 dB.
523 level_compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
524 gain_staging_db[bus_index] = to_db(level_compressor[bus_index]->get_attenuation() * makeup_gain);
526 // Just apply the gain we already had.
527 float db = gain_staging_db[bus_index];
528 float last_db = last_gain_staging_db[bus_index];
529 apply_gain(db, last_db, &samples_bus);
531 last_gain_staging_db[bus_index] = gain_staging_db[bus_index];
534 printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
535 level_compressor.get_level(), to_db(level_compressor.get_level()),
536 level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()),
537 to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
540 // The real compressor.
541 if (compressor_enabled[bus_index]) {
542 float threshold = from_db(compressor_threshold_dbfs[bus_index]);
544 float attack_time = 0.005f;
545 float release_time = 0.040f;
546 float makeup_gain = 2.0f; // +6 dB.
547 compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
548 // compressor_att = compressor.get_attenuation();
552 add_bus_to_master(bus_index, samples_bus, &samples_out);
553 deinterleave_samples(samples_bus, &left, &right);
554 measure_bus_levels(bus_index, left, right);
558 lock_guard<mutex> lock(compressor_mutex);
560 // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
561 // Note that since ratio is not infinite, we could go slightly higher than this.
562 if (limiter_enabled) {
563 float threshold = from_db(limiter_threshold_dbfs);
565 float attack_time = 0.0f; // Instant.
566 float release_time = 0.020f;
567 float makeup_gain = 1.0f; // 0 dB.
568 limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
569 // limiter_att = limiter.get_attenuation();
572 // printf("limiter=%+5.1f compressor=%+5.1f\n", to_db(limiter_att), to_db(compressor_att));
575 // At this point, we are most likely close to +0 LU (at least if the
576 // faders sum to 0 dB and the compressors are on), but all of our
577 // measurements have been on raw sample values, not R128 values.
578 // So we have a final makeup gain to get us to +0 LU; the gain
579 // adjustments required should be relatively small, and also, the
580 // offset shouldn't change much (only if the type of audio changes
581 // significantly). Thus, we shoot for updating this value basically
582 // “whenever we process buffers”, since the R128 calculation isn't exactly
583 // something we get out per-sample.
585 // Note that there's a feedback loop here, so we choose a very slow filter
586 // (half-time of 30 seconds).
587 double target_loudness_factor, alpha;
588 double loudness_lu = r128.loudness_M() - ref_level_lufs;
589 target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
591 // If we're outside +/- 5 LU (after correction), we don't count it as
592 // a normal signal (probably silence) and don't change the
593 // correction factor; just apply what we already have.
594 if (fabs(loudness_lu) >= 5.0 || !final_makeup_gain_auto) {
597 // Formula adapted from
598 // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
599 const double half_time_s = 30.0;
600 const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
601 alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
605 lock_guard<mutex> lock(compressor_mutex);
606 double m = final_makeup_gain;
607 for (size_t i = 0; i < samples_out.size(); i += 2) {
608 samples_out[i + 0] *= m;
609 samples_out[i + 1] *= m;
610 m += (target_loudness_factor - m) * alpha;
612 final_makeup_gain = m;
615 update_meters(samples_out);
622 void apply_filter_fade(StereoFilter *filter, float *data, unsigned num_samples, float cutoff_hz, float db, float last_db)
624 // A granularity of 32 samples is an okay tradeoff between speed and
625 // smoothness; recalculating the filters is pretty expensive, so it's
626 // good that we don't do this all the time.
627 static constexpr unsigned filter_granularity_samples = 32;
629 const float cutoff_linear = cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY;
630 if (fabs(db - last_db) < 1e-3) {
631 // Constant over this frame.
632 if (fabs(db) > 0.01f) {
633 filter->render(data, num_samples, cutoff_linear, 0.5f, db / 40.0f);
636 // We need to do a fade. (Rounding up avoids division by zero.)
637 unsigned num_blocks = (num_samples + filter_granularity_samples - 1) / filter_granularity_samples;
638 const float inc_db_norm = (db - last_db) / 40.0f / num_blocks;
639 float db_norm = db / 40.0f;
640 for (size_t i = 0; i < num_samples; i += filter_granularity_samples) {
641 size_t samples_this_block = std::min<size_t>(num_samples - i, filter_granularity_samples);
642 filter->render(data + i * 2, samples_this_block, cutoff_linear, 0.5f, db_norm);
643 db_norm += inc_db_norm;
650 void AudioMixer::apply_eq(unsigned bus_index, vector<float> *samples_bus)
652 constexpr float bass_freq_hz = 200.0f;
653 constexpr float treble_freq_hz = 4700.0f;
655 // Cut away everything under 120 Hz (or whatever the cutoff is);
656 // we don't need it for voice, and it will reduce headroom
657 // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
658 // should be dampened.)
659 if (locut_enabled[bus_index]) {
660 locut[bus_index].render(samples_bus->data(), samples_bus->size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
663 // Apply the rest of the EQ. Since we only have a simple three-band EQ,
664 // we can implement it with two shelf filters. We use a simple gain to
665 // set the mid-level filter, and then offset the low and high bands
666 // from that if we need to. (We could perhaps have folded the gain into
667 // the next part, but it's so cheap that the trouble isn't worth it.)
669 // If any part of the EQ has changed appreciably since last frame,
670 // we fade smoothly during the course of this frame.
671 const float bass_db = eq_level_db[bus_index][EQ_BAND_BASS];
672 const float mid_db = eq_level_db[bus_index][EQ_BAND_MID];
673 const float treble_db = eq_level_db[bus_index][EQ_BAND_TREBLE];
675 const float last_bass_db = last_eq_level_db[bus_index][EQ_BAND_BASS];
676 const float last_mid_db = last_eq_level_db[bus_index][EQ_BAND_MID];
677 const float last_treble_db = last_eq_level_db[bus_index][EQ_BAND_TREBLE];
679 assert(samples_bus->size() % 2 == 0);
680 const unsigned num_samples = samples_bus->size() / 2;
682 apply_gain(mid_db, last_mid_db, samples_bus);
684 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);
685 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);
687 last_eq_level_db[bus_index][EQ_BAND_BASS] = bass_db;
688 last_eq_level_db[bus_index][EQ_BAND_MID] = mid_db;
689 last_eq_level_db[bus_index][EQ_BAND_TREBLE] = treble_db;
692 void AudioMixer::add_bus_to_master(unsigned bus_index, const vector<float> &samples_bus, vector<float> *samples_out)
694 assert(samples_bus.size() == samples_out->size());
695 assert(samples_bus.size() % 2 == 0);
696 unsigned num_samples = samples_bus.size() / 2;
697 const float new_volume_db = mute[bus_index] ? -90.0f : fader_volume_db[bus_index].load();
698 if (fabs(new_volume_db - last_fader_volume_db[bus_index]) > 1e-3) {
699 // The volume has changed; do a fade over the course of this frame.
700 // (We might have some numerical issues here, but it seems to sound OK.)
701 // For the purpose of fading here, the silence floor is set to -90 dB
702 // (the fader only goes to -84).
703 float old_volume = from_db(max<float>(last_fader_volume_db[bus_index], -90.0f));
704 float volume = from_db(max<float>(new_volume_db, -90.0f));
706 float volume_inc = pow(volume / old_volume, 1.0 / num_samples);
708 if (bus_index == 0) {
709 for (unsigned i = 0; i < num_samples; ++i) {
710 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
711 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
712 volume *= volume_inc;
715 for (unsigned i = 0; i < num_samples; ++i) {
716 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
717 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
718 volume *= volume_inc;
721 } else if (new_volume_db > -90.0f) {
722 float volume = from_db(new_volume_db);
723 if (bus_index == 0) {
724 for (unsigned i = 0; i < num_samples; ++i) {
725 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
726 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
729 for (unsigned i = 0; i < num_samples; ++i) {
730 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
731 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
736 last_fader_volume_db[bus_index] = new_volume_db;
739 void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right)
741 assert(left.size() == right.size());
742 const float volume = mute[bus_index] ? 0.0f : from_db(fader_volume_db[bus_index]);
743 const float peak_levels[2] = {
744 find_peak(left.data(), left.size()) * volume,
745 find_peak(right.data(), right.size()) * volume
747 for (unsigned channel = 0; channel < 2; ++channel) {
748 // Compute the current value, including hold and falloff.
749 // The constants are borrowed from zita-mu1 by Fons Adriaensen.
750 static constexpr float hold_sec = 0.5f;
751 static constexpr float falloff_db_sec = 15.0f; // dB/sec falloff after hold.
753 PeakHistory &history = peak_history[bus_index][channel];
754 history.historic_peak = max(history.historic_peak, peak_levels[channel]);
755 if (history.age_seconds < hold_sec) {
756 current_peak = history.last_peak;
758 current_peak = history.last_peak * from_db(-falloff_db_sec * (history.age_seconds - hold_sec));
761 // See if we have a new peak to replace the old (possibly falling) one.
762 if (peak_levels[channel] > current_peak) {
763 history.last_peak = peak_levels[channel];
764 history.age_seconds = 0.0f; // Not 100% correct, but more than good enough given our frame sizes.
765 current_peak = peak_levels[channel];
767 history.age_seconds += float(left.size()) / OUTPUT_FREQUENCY;
769 history.current_level = peak_levels[channel];
770 history.current_peak = current_peak;
774 void AudioMixer::update_meters(const vector<float> &samples)
776 // Upsample 4x to find interpolated peak.
777 peak_resampler.inp_data = const_cast<float *>(samples.data());
778 peak_resampler.inp_count = samples.size() / 2;
780 vector<float> interpolated_samples;
781 interpolated_samples.resize(samples.size());
783 lock_guard<mutex> lock(audio_measure_mutex);
785 while (peak_resampler.inp_count > 0) { // About four iterations.
786 peak_resampler.out_data = &interpolated_samples[0];
787 peak_resampler.out_count = interpolated_samples.size() / 2;
788 peak_resampler.process();
789 size_t out_stereo_samples = interpolated_samples.size() / 2 - peak_resampler.out_count;
790 peak = max<float>(peak, find_peak(interpolated_samples.data(), out_stereo_samples * 2));
791 peak_resampler.out_data = nullptr;
795 // Find R128 levels and L/R correlation.
796 vector<float> left, right;
797 deinterleave_samples(samples, &left, &right);
798 float *ptrs[] = { left.data(), right.data() };
800 lock_guard<mutex> lock(audio_measure_mutex);
801 r128.process(left.size(), ptrs);
802 correlation.process_samples(samples);
805 send_audio_level_callback();
808 void AudioMixer::reset_meters()
810 lock_guard<mutex> lock(audio_measure_mutex);
811 peak_resampler.reset();
818 void AudioMixer::send_audio_level_callback()
820 if (audio_level_callback == nullptr) {
824 lock_guard<mutex> lock(audio_measure_mutex);
825 double loudness_s = r128.loudness_S();
826 double loudness_i = r128.integrated();
827 double loudness_range_low = r128.range_min();
828 double loudness_range_high = r128.range_max();
830 vector<BusLevel> bus_levels;
831 bus_levels.resize(input_mapping.buses.size());
833 lock_guard<mutex> lock(compressor_mutex);
834 for (unsigned bus_index = 0; bus_index < bus_levels.size(); ++bus_index) {
835 bus_levels[bus_index].current_level_dbfs[0] = to_db(peak_history[bus_index][0].current_level);
836 bus_levels[bus_index].current_level_dbfs[1] = to_db(peak_history[bus_index][1].current_level);
837 bus_levels[bus_index].peak_level_dbfs[0] = to_db(peak_history[bus_index][0].current_peak);
838 bus_levels[bus_index].peak_level_dbfs[1] = to_db(peak_history[bus_index][1].current_peak);
839 bus_levels[bus_index].historic_peak_dbfs = to_db(
840 max(peak_history[bus_index][0].historic_peak,
841 peak_history[bus_index][1].historic_peak));
842 bus_levels[bus_index].gain_staging_db = gain_staging_db[bus_index];
843 if (compressor_enabled[bus_index]) {
844 bus_levels[bus_index].compressor_attenuation_db = -to_db(compressor[bus_index]->get_attenuation());
846 bus_levels[bus_index].compressor_attenuation_db = 0.0;
851 audio_level_callback(loudness_s, to_db(peak), bus_levels,
852 loudness_i, loudness_range_low, loudness_range_high,
853 to_db(final_makeup_gain),
854 correlation.get_correlation());
857 map<DeviceSpec, DeviceInfo> AudioMixer::get_devices()
859 lock_guard<timed_mutex> lock(audio_mutex);
861 map<DeviceSpec, DeviceInfo> devices;
862 for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
863 const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index };
864 const AudioDevice *device = &video_cards[card_index];
866 info.display_name = device->display_name;
867 info.num_channels = 8;
868 devices.insert(make_pair(spec, info));
870 vector<ALSAPool::Device> available_alsa_devices = alsa_pool.get_devices();
871 for (unsigned card_index = 0; card_index < available_alsa_devices.size(); ++card_index) {
872 const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index };
873 const ALSAPool::Device &device = available_alsa_devices[card_index];
875 info.display_name = device.display_name();
876 info.num_channels = device.num_channels;
877 info.alsa_name = device.name;
878 info.alsa_info = device.info;
879 info.alsa_address = device.address;
880 devices.insert(make_pair(spec, info));
885 void AudioMixer::set_display_name(DeviceSpec device_spec, const string &name)
887 AudioDevice *device = find_audio_device(device_spec);
889 lock_guard<timed_mutex> lock(audio_mutex);
890 device->display_name = name;
893 void AudioMixer::serialize_device(DeviceSpec device_spec, DeviceSpecProto *device_spec_proto)
895 lock_guard<timed_mutex> lock(audio_mutex);
896 switch (device_spec.type) {
897 case InputSourceType::SILENCE:
898 device_spec_proto->set_type(DeviceSpecProto::SILENCE);
900 case InputSourceType::CAPTURE_CARD:
901 device_spec_proto->set_type(DeviceSpecProto::CAPTURE_CARD);
902 device_spec_proto->set_index(device_spec.index);
903 device_spec_proto->set_display_name(video_cards[device_spec.index].display_name);
905 case InputSourceType::ALSA_INPUT:
906 alsa_pool.serialize_device(device_spec.index, device_spec_proto);
911 void AudioMixer::set_simple_input(unsigned card_index)
913 InputMapping new_input_mapping;
914 InputMapping::Bus input;
916 input.device.type = InputSourceType::CAPTURE_CARD;
917 input.device.index = card_index;
918 input.source_channel[0] = 0;
919 input.source_channel[1] = 1;
921 new_input_mapping.buses.push_back(input);
923 lock_guard<timed_mutex> lock(audio_mutex);
924 current_mapping_mode = MappingMode::SIMPLE;
925 set_input_mapping_lock_held(new_input_mapping);
926 fader_volume_db[0] = 0.0f;
929 unsigned AudioMixer::get_simple_input() const
931 lock_guard<timed_mutex> lock(audio_mutex);
932 if (input_mapping.buses.size() == 1 &&
933 input_mapping.buses[0].device.type == InputSourceType::CAPTURE_CARD &&
934 input_mapping.buses[0].source_channel[0] == 0 &&
935 input_mapping.buses[0].source_channel[1] == 1) {
936 return input_mapping.buses[0].device.index;
938 return numeric_limits<unsigned>::max();
942 void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
944 lock_guard<timed_mutex> lock(audio_mutex);
945 set_input_mapping_lock_held(new_input_mapping);
946 current_mapping_mode = MappingMode::MULTICHANNEL;
949 AudioMixer::MappingMode AudioMixer::get_mapping_mode() const
951 lock_guard<timed_mutex> lock(audio_mutex);
952 return current_mapping_mode;
955 void AudioMixer::set_input_mapping_lock_held(const InputMapping &new_input_mapping)
957 map<DeviceSpec, set<unsigned>> interesting_channels;
958 for (const InputMapping::Bus &bus : new_input_mapping.buses) {
959 if (bus.device.type == InputSourceType::CAPTURE_CARD ||
960 bus.device.type == InputSourceType::ALSA_INPUT) {
961 for (unsigned channel = 0; channel < 2; ++channel) {
962 if (bus.source_channel[channel] != -1) {
963 interesting_channels[bus.device].insert(bus.source_channel[channel]);
969 // Reset resamplers for all cards that don't have the exact same state as before.
970 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
971 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
972 AudioDevice *device = find_audio_device(device_spec);
973 if (device->interesting_channels != interesting_channels[device_spec]) {
974 device->interesting_channels = interesting_channels[device_spec];
975 reset_resampler_mutex_held(device_spec);
978 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
979 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
980 AudioDevice *device = find_audio_device(device_spec);
981 if (interesting_channels[device_spec].empty()) {
982 alsa_pool.release_device(card_index);
984 alsa_pool.hold_device(card_index);
986 if (device->interesting_channels != interesting_channels[device_spec]) {
987 device->interesting_channels = interesting_channels[device_spec];
988 alsa_pool.reset_device(device_spec.index);
989 reset_resampler_mutex_held(device_spec);
993 input_mapping = new_input_mapping;
996 InputMapping AudioMixer::get_input_mapping() const
998 lock_guard<timed_mutex> lock(audio_mutex);
999 return input_mapping;
1002 unsigned AudioMixer::num_buses() const
1004 lock_guard<timed_mutex> lock(audio_mutex);
1005 return input_mapping.buses.size();
1008 void AudioMixer::reset_peak(unsigned bus_index)
1010 lock_guard<timed_mutex> lock(audio_mutex);
1011 for (unsigned channel = 0; channel < 2; ++channel) {
1012 PeakHistory &history = peak_history[bus_index][channel];
1013 history.current_level = 0.0f;
1014 history.historic_peak = 0.0f;
1015 history.current_peak = 0.0f;
1016 history.last_peak = 0.0f;
1017 history.age_seconds = 0.0f;
1021 AudioMixer *global_audio_mixer = nullptr;