1 #include "audio_mixer.h"
5 #include <bmusb/bmusb.h>
10 #include <immintrin.h>
19 using namespace bmusb;
21 using namespace std::placeholders;
25 // TODO: If these prove to be a bottleneck, they can be SSSE3-optimized
26 // (usually including multiple channels at a time).
28 void convert_fixed16_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
29 const uint8_t *src, size_t in_channel, size_t in_num_channels,
32 assert(in_channel < in_num_channels);
33 assert(out_channel < out_num_channels);
34 src += in_channel * 2;
37 for (size_t i = 0; i < num_samples; ++i) {
38 int16_t s = le16toh(*(int16_t *)src);
39 *dst = s * (1.0f / 32768.0f);
41 src += 2 * in_num_channels;
42 dst += out_num_channels;
46 void convert_fixed24_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
47 const uint8_t *src, size_t in_channel, size_t in_num_channels,
50 assert(in_channel < in_num_channels);
51 assert(out_channel < out_num_channels);
52 src += in_channel * 3;
55 for (size_t i = 0; i < num_samples; ++i) {
59 uint32_t s = s1 | (s1 << 8) | (s2 << 16) | (s3 << 24);
60 *dst = int(s) * (1.0f / 2147483648.0f);
62 src += 3 * in_num_channels;
63 dst += out_num_channels;
67 void convert_fixed32_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
68 const uint8_t *src, size_t in_channel, size_t in_num_channels,
71 assert(in_channel < in_num_channels);
72 assert(out_channel < out_num_channels);
73 src += in_channel * 4;
76 for (size_t i = 0; i < num_samples; ++i) {
77 int32_t s = le32toh(*(int32_t *)src);
78 *dst = s * (1.0f / 2147483648.0f);
80 src += 4 * in_num_channels;
81 dst += out_num_channels;
85 float find_peak_plain(const float *samples, size_t num_samples) __attribute__((unused));
87 float find_peak_plain(const float *samples, size_t num_samples)
89 float m = fabs(samples[0]);
90 for (size_t i = 1; i < num_samples; ++i) {
91 m = max(m, fabs(samples[i]));
97 static inline float horizontal_max(__m128 m)
99 __m128 tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 0, 3, 2));
100 m = _mm_max_ps(m, tmp);
101 tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 3, 0, 1));
102 m = _mm_max_ps(m, tmp);
103 return _mm_cvtss_f32(m);
106 float find_peak(const float *samples, size_t num_samples)
108 const __m128 abs_mask = _mm_castsi128_ps(_mm_set1_epi32(0x7fffffffu));
109 __m128 m = _mm_setzero_ps();
110 for (size_t i = 0; i < (num_samples & ~3); i += 4) {
111 __m128 x = _mm_loadu_ps(samples + i);
112 x = _mm_and_ps(x, abs_mask);
113 m = _mm_max_ps(m, x);
115 float result = horizontal_max(m);
117 for (size_t i = (num_samples & ~3); i < num_samples; ++i) {
118 result = max(result, fabs(samples[i]));
122 // Self-test. We should be bit-exact the same.
123 float reference_result = find_peak_plain(samples, num_samples);
124 if (result != reference_result) {
125 fprintf(stderr, "Error: Peak is %f [%f %f %f %f]; should be %f.\n",
127 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(0, 0, 0, 0))),
128 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 1, 1, 1))),
129 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 2, 2, 2))),
130 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(3, 3, 3, 3))),
138 float find_peak(const float *samples, size_t num_samples)
140 return find_peak_plain(samples, num_samples);
144 void deinterleave_samples(const vector<float> &in, vector<float> *out_l, vector<float> *out_r)
146 size_t num_samples = in.size() / 2;
147 out_l->resize(num_samples);
148 out_r->resize(num_samples);
150 const float *inptr = in.data();
151 float *lptr = &(*out_l)[0];
152 float *rptr = &(*out_r)[0];
153 for (size_t i = 0; i < num_samples; ++i) {
161 AudioMixer::AudioMixer(unsigned num_cards)
162 : num_cards(num_cards),
163 limiter(OUTPUT_FREQUENCY),
164 correlation(OUTPUT_FREQUENCY)
166 global_audio_mixer = this;
168 for (unsigned bus_index = 0; bus_index < MAX_BUSES; ++bus_index) {
169 locut[bus_index].init(FILTER_HPF, 2);
170 locut_enabled[bus_index] = global_flags.locut_enabled;
171 eq[bus_index][EQ_BAND_BASS].init(FILTER_LOW_SHELF, 1);
172 // Note: EQ_BAND_MID isn't used (see comments in apply_eq()).
173 eq[bus_index][EQ_BAND_TREBLE].init(FILTER_HIGH_SHELF, 1);
175 gain_staging_db[bus_index] = global_flags.initial_gain_staging_db;
176 compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
177 compressor_threshold_dbfs[bus_index] = ref_level_dbfs - 12.0f; // -12 dB.
178 compressor_enabled[bus_index] = global_flags.compressor_enabled;
179 level_compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
180 level_compressor_enabled[bus_index] = global_flags.gain_staging_auto;
182 set_limiter_enabled(global_flags.limiter_enabled);
183 set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
186 InputMapping new_input_mapping;
187 if (!global_flags.input_mapping_filename.empty()) {
188 if (!load_input_mapping_from_file(get_devices(),
189 global_flags.input_mapping_filename,
190 &new_input_mapping)) {
191 fprintf(stderr, "Failed to load input mapping from '%s', exiting.\n",
192 global_flags.input_mapping_filename.c_str());
196 // Generate a very simple, default input mapping.
197 InputMapping::Bus input;
199 input.device.type = InputSourceType::CAPTURE_CARD;
200 input.device.index = 0;
201 input.source_channel[0] = 0;
202 input.source_channel[1] = 1;
204 new_input_mapping.buses.push_back(input);
206 set_input_mapping(new_input_mapping);
208 r128.init(2, OUTPUT_FREQUENCY);
211 // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
212 // and there's a limit to how important the peak meter is.
213 peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
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(device_spec.index, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size()));
233 device->next_local_pts = 0;
236 bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length)
238 AudioDevice *device = find_audio_device(device_spec);
240 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
241 if (!lock.try_lock_for(chrono::milliseconds(10))) {
244 if (device->resampling_queue == nullptr) {
245 // No buses use this device; throw it away.
249 unsigned num_channels = device->interesting_channels.size();
250 assert(num_channels > 0);
252 // Convert the audio to fp32.
253 unique_ptr<float[]> audio(new float[num_samples * num_channels]);
254 unsigned channel_index = 0;
255 for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) {
256 switch (audio_format.bits_per_sample) {
258 assert(num_samples == 0);
261 convert_fixed16_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
264 convert_fixed24_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
267 convert_fixed32_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
270 fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
276 int64_t local_pts = device->next_local_pts;
277 device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.get(), num_samples);
278 device->next_local_pts = local_pts + frame_length;
282 bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length)
284 AudioDevice *device = find_audio_device(device_spec);
286 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
287 if (!lock.try_lock_for(chrono::milliseconds(10))) {
290 if (device->resampling_queue == nullptr) {
291 // No buses use this device; throw it away.
295 unsigned num_channels = device->interesting_channels.size();
296 assert(num_channels > 0);
298 vector<float> silence(samples_per_frame * num_channels, 0.0f);
299 for (unsigned i = 0; i < num_frames; ++i) {
300 device->resampling_queue->add_input_samples(device->next_local_pts / double(TIMEBASE), silence.data(), samples_per_frame);
301 // Note that if the format changed in the meantime, we have
302 // no way of detecting that; we just have to assume the frame length
303 // is always the same.
304 device->next_local_pts += frame_length;
309 bool AudioMixer::silence_card(DeviceSpec device_spec, bool silence)
311 AudioDevice *device = find_audio_device(device_spec);
313 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
314 if (!lock.try_lock_for(chrono::milliseconds(10))) {
318 if (device->silenced && !silence) {
319 reset_resampler_mutex_held(device_spec);
321 device->silenced = silence;
325 AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
327 switch (device.type) {
328 case InputSourceType::CAPTURE_CARD:
329 return &video_cards[device.index];
330 case InputSourceType::ALSA_INPUT:
331 return &alsa_inputs[device.index];
332 case InputSourceType::SILENCE:
339 // Get a pointer to the given channel from the given device.
340 // The channel must be picked out earlier and resampled.
341 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)
343 static float zero = 0.0f;
344 if (source_channel == -1 || device_spec.type == InputSourceType::SILENCE) {
349 AudioDevice *device = find_audio_device(device_spec);
350 assert(device->interesting_channels.count(source_channel) != 0);
351 unsigned channel_index = 0;
352 for (int channel : device->interesting_channels) {
353 if (channel == source_channel) break;
356 assert(channel_index < device->interesting_channels.size());
357 const auto it = samples_card.find(device_spec);
358 assert(it != samples_card.end());
359 *srcptr = &(it->second)[channel_index];
360 *stride = device->interesting_channels.size();
363 // TODO: Can be SSSE3-optimized if need be.
364 void AudioMixer::fill_audio_bus(const map<DeviceSpec, vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output)
366 if (bus.device.type == InputSourceType::SILENCE) {
367 memset(output, 0, num_samples * sizeof(*output));
369 assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
370 bus.device.type == InputSourceType::ALSA_INPUT);
371 const float *lsrc, *rsrc;
372 unsigned lstride, rstride;
373 float *dptr = output;
374 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[0], &lsrc, &lstride);
375 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[1], &rsrc, &rstride);
376 for (unsigned i = 0; i < num_samples; ++i) {
385 vector<DeviceSpec> AudioMixer::get_active_devices() const
387 vector<DeviceSpec> ret;
388 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
389 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
390 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
391 ret.push_back(device_spec);
394 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
395 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
396 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
397 ret.push_back(device_spec);
403 vector<float> AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
405 map<DeviceSpec, vector<float>> samples_card;
406 vector<float> samples_bus;
408 lock_guard<timed_mutex> lock(audio_mutex);
410 // Pick out all the interesting channels from all the cards.
411 for (const DeviceSpec &device_spec : get_active_devices()) {
412 AudioDevice *device = find_audio_device(device_spec);
413 samples_card[device_spec].resize(num_samples * device->interesting_channels.size());
414 if (device->silenced) {
415 memset(&samples_card[device_spec][0], 0, samples_card[device_spec].size() * sizeof(float));
417 device->resampling_queue->get_output_samples(
419 &samples_card[device_spec][0],
421 rate_adjustment_policy);
425 vector<float> samples_out, left, right;
426 samples_out.resize(num_samples * 2);
427 samples_bus.resize(num_samples * 2);
428 for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
429 fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, &samples_bus[0]);
430 apply_eq(bus_index, &samples_bus);
433 lock_guard<mutex> lock(compressor_mutex);
435 // Apply a level compressor to get the general level right.
436 // Basically, if it's over about -40 dBFS, we squeeze it down to that level
437 // (or more precisely, near it, since we don't use infinite ratio),
438 // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
439 // entirely arbitrary, but from practical tests with speech, it seems to
440 // put ut around -23 LUFS, so it's a reasonable starting point for later use.
441 if (level_compressor_enabled[bus_index]) {
442 float threshold = 0.01f; // -40 dBFS.
444 float attack_time = 0.5f;
445 float release_time = 20.0f;
446 float makeup_gain = from_db(ref_level_dbfs - (-40.0f)); // +26 dB.
447 level_compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
448 gain_staging_db[bus_index] = to_db(level_compressor[bus_index]->get_attenuation() * makeup_gain);
450 // Just apply the gain we already had.
451 float g = from_db(gain_staging_db[bus_index]);
452 for (size_t i = 0; i < samples_bus.size(); ++i) {
458 printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
459 level_compressor.get_level(), to_db(level_compressor.get_level()),
460 level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()),
461 to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
464 // The real compressor.
465 if (compressor_enabled[bus_index]) {
466 float threshold = from_db(compressor_threshold_dbfs[bus_index]);
468 float attack_time = 0.005f;
469 float release_time = 0.040f;
470 float makeup_gain = 2.0f; // +6 dB.
471 compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
472 // compressor_att = compressor.get_attenuation();
476 add_bus_to_master(bus_index, samples_bus, &samples_out);
477 deinterleave_samples(samples_bus, &left, &right);
478 measure_bus_levels(bus_index, left, right);
482 lock_guard<mutex> lock(compressor_mutex);
484 // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
485 // Note that since ratio is not infinite, we could go slightly higher than this.
486 if (limiter_enabled) {
487 float threshold = from_db(limiter_threshold_dbfs);
489 float attack_time = 0.0f; // Instant.
490 float release_time = 0.020f;
491 float makeup_gain = 1.0f; // 0 dB.
492 limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
493 // limiter_att = limiter.get_attenuation();
496 // printf("limiter=%+5.1f compressor=%+5.1f\n", to_db(limiter_att), to_db(compressor_att));
499 // At this point, we are most likely close to +0 LU (at least if the
500 // faders sum to 0 dB and the compressors are on), but all of our
501 // measurements have been on raw sample values, not R128 values.
502 // So we have a final makeup gain to get us to +0 LU; the gain
503 // adjustments required should be relatively small, and also, the
504 // offset shouldn't change much (only if the type of audio changes
505 // significantly). Thus, we shoot for updating this value basically
506 // “whenever we process buffers”, since the R128 calculation isn't exactly
507 // something we get out per-sample.
509 // Note that there's a feedback loop here, so we choose a very slow filter
510 // (half-time of 30 seconds).
511 double target_loudness_factor, alpha;
512 double loudness_lu = r128.loudness_M() - ref_level_lufs;
513 double current_makeup_lu = to_db(final_makeup_gain);
514 target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
516 // If we're outside +/- 5 LU uncorrected, we don't count it as
517 // a normal signal (probably silence) and don't change the
518 // correction factor; just apply what we already have.
519 if (fabs(loudness_lu - current_makeup_lu) >= 5.0 || !final_makeup_gain_auto) {
522 // Formula adapted from
523 // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
524 const double half_time_s = 30.0;
525 const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
526 alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
530 lock_guard<mutex> lock(compressor_mutex);
531 double m = final_makeup_gain;
532 for (size_t i = 0; i < samples_out.size(); i += 2) {
533 samples_out[i + 0] *= m;
534 samples_out[i + 1] *= m;
535 m += (target_loudness_factor - m) * alpha;
537 final_makeup_gain = m;
540 update_meters(samples_out);
545 void AudioMixer::apply_eq(unsigned bus_index, vector<float> *samples_bus)
547 constexpr float bass_freq_hz = 200.0f;
548 constexpr float treble_freq_hz = 4700.0f;
550 // Cut away everything under 120 Hz (or whatever the cutoff is);
551 // we don't need it for voice, and it will reduce headroom
552 // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
553 // should be dampened.)
554 if (locut_enabled[bus_index]) {
555 locut[bus_index].render(samples_bus->data(), samples_bus->size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
558 // Apply the rest of the EQ. Since we only have a simple three-band EQ,
559 // we can implement it with two shelf filters. We use a simple gain to
560 // set the mid-level filter, and then offset the low and high bands
561 // from that if we need to. (We could perhaps have folded the gain into
562 // the next part, but it's so cheap that the trouble isn't worth it.)
563 if (fabs(eq_level_db[bus_index][EQ_BAND_MID]) > 0.01f) {
564 float g = from_db(eq_level_db[bus_index][EQ_BAND_MID]);
565 for (size_t i = 0; i < samples_bus->size(); ++i) {
566 (*samples_bus)[i] *= g;
570 float bass_adj_db = eq_level_db[bus_index][EQ_BAND_BASS] - eq_level_db[bus_index][EQ_BAND_MID];
571 if (fabs(bass_adj_db) > 0.01f) {
572 eq[bus_index][EQ_BAND_BASS].render(samples_bus->data(), samples_bus->size() / 2,
573 bass_freq_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f, bass_adj_db / 40.0f);
576 float treble_adj_db = eq_level_db[bus_index][EQ_BAND_TREBLE] - eq_level_db[bus_index][EQ_BAND_MID];
577 if (fabs(treble_adj_db) > 0.01f) {
578 eq[bus_index][EQ_BAND_TREBLE].render(samples_bus->data(), samples_bus->size() / 2,
579 treble_freq_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f, treble_adj_db / 40.0f);
583 void AudioMixer::add_bus_to_master(unsigned bus_index, const vector<float> &samples_bus, vector<float> *samples_out)
585 assert(samples_bus.size() == samples_out->size());
586 assert(samples_bus.size() % 2 == 0);
587 unsigned num_samples = samples_bus.size() / 2;
588 if (fabs(fader_volume_db[bus_index] - last_fader_volume_db[bus_index]) > 1e-3) {
589 // The volume has changed; do a fade over the course of this frame.
590 // (We might have some numerical issues here, but it seems to sound OK.)
591 // For the purpose of fading here, the silence floor is set to -90 dB
592 // (the fader only goes to -84).
593 float old_volume = from_db(max<float>(last_fader_volume_db[bus_index], -90.0f));
594 float volume = from_db(max<float>(fader_volume_db[bus_index], -90.0f));
596 float volume_inc = pow(volume / old_volume, 1.0 / num_samples);
598 if (bus_index == 0) {
599 for (unsigned i = 0; i < num_samples; ++i) {
600 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
601 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
602 volume *= volume_inc;
605 for (unsigned i = 0; i < num_samples; ++i) {
606 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
607 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
608 volume *= volume_inc;
612 float volume = from_db(fader_volume_db[bus_index]);
613 if (bus_index == 0) {
614 for (unsigned i = 0; i < num_samples; ++i) {
615 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
616 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
619 for (unsigned i = 0; i < num_samples; ++i) {
620 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
621 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
626 last_fader_volume_db[bus_index] = fader_volume_db[bus_index];
629 void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right)
631 assert(left.size() == right.size());
632 const float volume = from_db(fader_volume_db[bus_index]);
633 const float peak_levels[2] = {
634 find_peak(left.data(), left.size()) * volume,
635 find_peak(right.data(), right.size()) * volume
637 for (unsigned channel = 0; channel < 2; ++channel) {
638 // Compute the current value, including hold and falloff.
639 // The constants are borrowed from zita-mu1 by Fons Adriaensen.
640 static constexpr float hold_sec = 0.5f;
641 static constexpr float falloff_db_sec = 15.0f; // dB/sec falloff after hold.
643 PeakHistory &history = peak_history[bus_index][channel];
644 history.historic_peak = max(history.historic_peak, peak_levels[channel]);
645 if (history.age_seconds < hold_sec) {
646 current_peak = history.last_peak;
648 current_peak = history.last_peak * from_db(-falloff_db_sec * (history.age_seconds - hold_sec));
651 // See if we have a new peak to replace the old (possibly falling) one.
652 if (peak_levels[channel] > current_peak) {
653 history.last_peak = peak_levels[channel];
654 history.age_seconds = 0.0f; // Not 100% correct, but more than good enough given our frame sizes.
655 current_peak = peak_levels[channel];
657 history.age_seconds += float(left.size()) / OUTPUT_FREQUENCY;
659 history.current_level = peak_levels[channel];
660 history.current_peak = current_peak;
664 void AudioMixer::update_meters(const vector<float> &samples)
666 // Upsample 4x to find interpolated peak.
667 peak_resampler.inp_data = const_cast<float *>(samples.data());
668 peak_resampler.inp_count = samples.size() / 2;
670 vector<float> interpolated_samples;
671 interpolated_samples.resize(samples.size());
673 lock_guard<mutex> lock(audio_measure_mutex);
675 while (peak_resampler.inp_count > 0) { // About four iterations.
676 peak_resampler.out_data = &interpolated_samples[0];
677 peak_resampler.out_count = interpolated_samples.size() / 2;
678 peak_resampler.process();
679 size_t out_stereo_samples = interpolated_samples.size() / 2 - peak_resampler.out_count;
680 peak = max<float>(peak, find_peak(interpolated_samples.data(), out_stereo_samples * 2));
681 peak_resampler.out_data = nullptr;
685 // Find R128 levels and L/R correlation.
686 vector<float> left, right;
687 deinterleave_samples(samples, &left, &right);
688 float *ptrs[] = { left.data(), right.data() };
690 lock_guard<mutex> lock(audio_measure_mutex);
691 r128.process(left.size(), ptrs);
692 correlation.process_samples(samples);
695 send_audio_level_callback();
698 void AudioMixer::reset_meters()
700 lock_guard<mutex> lock(audio_measure_mutex);
701 peak_resampler.reset();
708 void AudioMixer::send_audio_level_callback()
710 if (audio_level_callback == nullptr) {
714 lock_guard<mutex> lock(audio_measure_mutex);
715 double loudness_s = r128.loudness_S();
716 double loudness_i = r128.integrated();
717 double loudness_range_low = r128.range_min();
718 double loudness_range_high = r128.range_max();
720 vector<BusLevel> bus_levels;
721 bus_levels.resize(input_mapping.buses.size());
723 lock_guard<mutex> lock(compressor_mutex);
724 for (unsigned bus_index = 0; bus_index < bus_levels.size(); ++bus_index) {
725 bus_levels[bus_index].current_level_dbfs[0] = to_db(peak_history[bus_index][0].current_level);
726 bus_levels[bus_index].current_level_dbfs[1] = to_db(peak_history[bus_index][1].current_level);
727 bus_levels[bus_index].peak_level_dbfs[0] = to_db(peak_history[bus_index][0].current_peak);
728 bus_levels[bus_index].peak_level_dbfs[1] = to_db(peak_history[bus_index][1].current_peak);
729 bus_levels[bus_index].historic_peak_dbfs = to_db(
730 max(peak_history[bus_index][0].historic_peak,
731 peak_history[bus_index][1].historic_peak));
732 bus_levels[bus_index].gain_staging_db = gain_staging_db[bus_index];
733 if (compressor_enabled[bus_index]) {
734 bus_levels[bus_index].compressor_attenuation_db = -to_db(compressor[bus_index]->get_attenuation());
736 bus_levels[bus_index].compressor_attenuation_db = 0.0;
741 audio_level_callback(loudness_s, to_db(peak), bus_levels,
742 loudness_i, loudness_range_low, loudness_range_high,
743 to_db(final_makeup_gain),
744 correlation.get_correlation());
747 map<DeviceSpec, DeviceInfo> AudioMixer::get_devices()
749 lock_guard<timed_mutex> lock(audio_mutex);
751 map<DeviceSpec, DeviceInfo> devices;
752 for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
753 const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index };
754 const AudioDevice *device = &video_cards[card_index];
756 info.display_name = device->display_name;
757 info.num_channels = 8;
758 devices.insert(make_pair(spec, info));
760 vector<ALSAPool::Device> available_alsa_devices = alsa_pool.get_devices();
761 for (unsigned card_index = 0; card_index < available_alsa_devices.size(); ++card_index) {
762 const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index };
763 const ALSAPool::Device &device = available_alsa_devices[card_index];
765 info.display_name = device.display_name();
766 info.num_channels = device.num_channels;
767 info.alsa_name = device.name;
768 info.alsa_info = device.info;
769 info.alsa_address = device.address;
770 devices.insert(make_pair(spec, info));
775 void AudioMixer::set_display_name(DeviceSpec device_spec, const string &name)
777 AudioDevice *device = find_audio_device(device_spec);
779 lock_guard<timed_mutex> lock(audio_mutex);
780 device->display_name = name;
783 void AudioMixer::serialize_device(DeviceSpec device_spec, DeviceSpecProto *device_spec_proto)
785 lock_guard<timed_mutex> lock(audio_mutex);
786 switch (device_spec.type) {
787 case InputSourceType::SILENCE:
788 device_spec_proto->set_type(DeviceSpecProto::SILENCE);
790 case InputSourceType::CAPTURE_CARD:
791 device_spec_proto->set_type(DeviceSpecProto::CAPTURE_CARD);
792 device_spec_proto->set_index(device_spec.index);
793 device_spec_proto->set_display_name(video_cards[device_spec.index].display_name);
795 case InputSourceType::ALSA_INPUT:
796 alsa_pool.serialize_device(device_spec.index, device_spec_proto);
801 void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
803 lock_guard<timed_mutex> lock(audio_mutex);
805 map<DeviceSpec, set<unsigned>> interesting_channels;
806 for (const InputMapping::Bus &bus : new_input_mapping.buses) {
807 if (bus.device.type == InputSourceType::CAPTURE_CARD ||
808 bus.device.type == InputSourceType::ALSA_INPUT) {
809 for (unsigned channel = 0; channel < 2; ++channel) {
810 if (bus.source_channel[channel] != -1) {
811 interesting_channels[bus.device].insert(bus.source_channel[channel]);
817 // Reset resamplers for all cards that don't have the exact same state as before.
818 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
819 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
820 AudioDevice *device = find_audio_device(device_spec);
821 if (device->interesting_channels != interesting_channels[device_spec]) {
822 device->interesting_channels = interesting_channels[device_spec];
823 reset_resampler_mutex_held(device_spec);
826 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
827 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
828 AudioDevice *device = find_audio_device(device_spec);
829 if (interesting_channels[device_spec].empty()) {
830 alsa_pool.release_device(card_index);
832 alsa_pool.hold_device(card_index);
834 if (device->interesting_channels != interesting_channels[device_spec]) {
835 device->interesting_channels = interesting_channels[device_spec];
836 alsa_pool.reset_device(device_spec.index);
837 reset_resampler_mutex_held(device_spec);
841 input_mapping = new_input_mapping;
844 InputMapping AudioMixer::get_input_mapping() const
846 lock_guard<timed_mutex> lock(audio_mutex);
847 return input_mapping;
850 void AudioMixer::reset_peak(unsigned bus_index)
852 lock_guard<timed_mutex> lock(audio_mutex);
853 for (unsigned channel = 0; channel < 2; ++channel) {
854 PeakHistory &history = peak_history[bus_index][channel];
855 history.current_level = 0.0f;
856 history.historic_peak = 0.0f;
857 history.current_peak = 0.0f;
858 history.last_peak = 0.0f;
859 history.age_seconds = 0.0f;
863 AudioMixer *global_audio_mixer = nullptr;