1 #include "audio_mixer.h"
5 #include <bmusb/bmusb.h>
10 #include <immintrin.h>
18 using namespace bmusb;
20 using namespace std::placeholders;
24 // TODO: If these prove to be a bottleneck, they can be SSSE3-optimized
25 // (usually including multiple channels at a time).
27 void convert_fixed16_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
28 const uint8_t *src, size_t in_channel, size_t in_num_channels,
31 assert(in_channel < in_num_channels);
32 assert(out_channel < out_num_channels);
33 src += in_channel * 2;
36 for (size_t i = 0; i < num_samples; ++i) {
37 int16_t s = le16toh(*(int16_t *)src);
38 *dst = s * (1.0f / 32768.0f);
40 src += 2 * in_num_channels;
41 dst += out_num_channels;
45 void convert_fixed24_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
46 const uint8_t *src, size_t in_channel, size_t in_num_channels,
49 assert(in_channel < in_num_channels);
50 assert(out_channel < out_num_channels);
51 src += in_channel * 3;
54 for (size_t i = 0; i < num_samples; ++i) {
58 uint32_t s = s1 | (s1 << 8) | (s2 << 16) | (s3 << 24);
59 *dst = int(s) * (1.0f / 2147483648.0f);
61 src += 3 * in_num_channels;
62 dst += out_num_channels;
66 void convert_fixed32_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
67 const uint8_t *src, size_t in_channel, size_t in_num_channels,
70 assert(in_channel < in_num_channels);
71 assert(out_channel < out_num_channels);
72 src += in_channel * 4;
75 for (size_t i = 0; i < num_samples; ++i) {
76 int32_t s = le32toh(*(int32_t *)src);
77 *dst = s * (1.0f / 2147483648.0f);
79 src += 4 * in_num_channels;
80 dst += out_num_channels;
84 float find_peak_plain(const float *samples, size_t num_samples) __attribute__((unused));
86 float find_peak_plain(const float *samples, size_t num_samples)
88 float m = fabs(samples[0]);
89 for (size_t i = 1; i < num_samples; ++i) {
90 m = max(m, fabs(samples[i]));
96 static inline float horizontal_max(__m128 m)
98 __m128 tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 0, 3, 2));
99 m = _mm_max_ps(m, tmp);
100 tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 3, 0, 1));
101 m = _mm_max_ps(m, tmp);
102 return _mm_cvtss_f32(m);
105 float find_peak(const float *samples, size_t num_samples)
107 const __m128 abs_mask = _mm_castsi128_ps(_mm_set1_epi32(0x7fffffffu));
108 __m128 m = _mm_setzero_ps();
109 for (size_t i = 0; i < (num_samples & ~3); i += 4) {
110 __m128 x = _mm_loadu_ps(samples + i);
111 x = _mm_and_ps(x, abs_mask);
112 m = _mm_max_ps(m, x);
114 float result = horizontal_max(m);
116 for (size_t i = (num_samples & ~3); i < num_samples; ++i) {
117 result = max(result, fabs(samples[i]));
121 // Self-test. We should be bit-exact the same.
122 float reference_result = find_peak_plain(samples, num_samples);
123 if (result != reference_result) {
124 fprintf(stderr, "Error: Peak is %f [%f %f %f %f]; should be %f.\n",
126 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(0, 0, 0, 0))),
127 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 1, 1, 1))),
128 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 2, 2, 2))),
129 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(3, 3, 3, 3))),
137 float find_peak(const float *samples, size_t num_samples)
139 return find_peak_plain(samples, num_samples);
143 void deinterleave_samples(const vector<float> &in, vector<float> *out_l, vector<float> *out_r)
145 size_t num_samples = in.size() / 2;
146 out_l->resize(num_samples);
147 out_r->resize(num_samples);
149 const float *inptr = in.data();
150 float *lptr = &(*out_l)[0];
151 float *rptr = &(*out_r)[0];
152 for (size_t i = 0; i < num_samples; ++i) {
160 AudioMixer::AudioMixer(unsigned num_cards)
161 : num_cards(num_cards),
162 limiter(OUTPUT_FREQUENCY),
163 correlation(OUTPUT_FREQUENCY)
165 global_audio_mixer = this;
167 for (unsigned bus_index = 0; bus_index < MAX_BUSES; ++bus_index) {
168 locut[bus_index].init(FILTER_HPF, 2);
169 locut_enabled[bus_index] = global_flags.locut_enabled;
170 eq[bus_index][EQ_BAND_BASS].init(FILTER_LOW_SHELF, 1);
171 // Note: EQ_BAND_MID isn't used (see comments in apply_eq()).
172 eq[bus_index][EQ_BAND_TREBLE].init(FILTER_HIGH_SHELF, 1);
174 gain_staging_db[bus_index] = global_flags.initial_gain_staging_db;
175 compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
176 compressor_threshold_dbfs[bus_index] = ref_level_dbfs - 12.0f; // -12 dB.
177 compressor_enabled[bus_index] = global_flags.compressor_enabled;
178 level_compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
179 level_compressor_enabled[bus_index] = global_flags.gain_staging_auto;
181 set_limiter_enabled(global_flags.limiter_enabled);
182 set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
184 // Generate a very simple, default input mapping.
185 InputMapping::Bus input;
187 input.device.type = InputSourceType::CAPTURE_CARD;
188 input.device.index = 0;
189 input.source_channel[0] = 0;
190 input.source_channel[1] = 1;
192 InputMapping new_input_mapping;
193 new_input_mapping.buses.push_back(input);
194 set_input_mapping(new_input_mapping);
198 r128.init(2, OUTPUT_FREQUENCY);
201 // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
202 // and there's a limit to how important the peak meter is.
203 peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
206 void AudioMixer::reset_resampler(DeviceSpec device_spec)
208 lock_guard<timed_mutex> lock(audio_mutex);
209 reset_resampler_mutex_held(device_spec);
212 void AudioMixer::reset_resampler_mutex_held(DeviceSpec device_spec)
214 AudioDevice *device = find_audio_device(device_spec);
216 if (device->interesting_channels.empty()) {
217 device->resampling_queue.reset();
219 // TODO: ResamplingQueue should probably take the full device spec.
220 // (It's only used for console output, though.)
221 device->resampling_queue.reset(new ResamplingQueue(device_spec.index, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size()));
223 device->next_local_pts = 0;
226 bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length)
228 AudioDevice *device = find_audio_device(device_spec);
230 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
231 if (!lock.try_lock_for(chrono::milliseconds(10))) {
234 if (device->resampling_queue == nullptr) {
235 // No buses use this device; throw it away.
239 unsigned num_channels = device->interesting_channels.size();
240 assert(num_channels > 0);
242 // Convert the audio to fp32.
243 unique_ptr<float[]> audio(new float[num_samples * num_channels]);
244 unsigned channel_index = 0;
245 for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) {
246 switch (audio_format.bits_per_sample) {
248 assert(num_samples == 0);
251 convert_fixed16_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
254 convert_fixed24_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
257 convert_fixed32_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
260 fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
266 int64_t local_pts = device->next_local_pts;
267 device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.get(), num_samples);
268 device->next_local_pts = local_pts + frame_length;
272 bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length)
274 AudioDevice *device = find_audio_device(device_spec);
276 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
277 if (!lock.try_lock_for(chrono::milliseconds(10))) {
280 if (device->resampling_queue == nullptr) {
281 // No buses use this device; throw it away.
285 unsigned num_channels = device->interesting_channels.size();
286 assert(num_channels > 0);
288 vector<float> silence(samples_per_frame * num_channels, 0.0f);
289 for (unsigned i = 0; i < num_frames; ++i) {
290 device->resampling_queue->add_input_samples(device->next_local_pts / double(TIMEBASE), silence.data(), samples_per_frame);
291 // Note that if the format changed in the meantime, we have
292 // no way of detecting that; we just have to assume the frame length
293 // is always the same.
294 device->next_local_pts += frame_length;
299 bool AudioMixer::silence_card(DeviceSpec device_spec, bool silence)
301 AudioDevice *device = find_audio_device(device_spec);
303 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
304 if (!lock.try_lock_for(chrono::milliseconds(10))) {
308 if (device->silenced && !silence) {
309 reset_resampler_mutex_held(device_spec);
311 device->silenced = silence;
315 AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
317 switch (device.type) {
318 case InputSourceType::CAPTURE_CARD:
319 return &video_cards[device.index];
320 case InputSourceType::ALSA_INPUT:
321 return &alsa_inputs[device.index];
322 case InputSourceType::SILENCE:
329 // Get a pointer to the given channel from the given device.
330 // The channel must be picked out earlier and resampled.
331 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)
333 static float zero = 0.0f;
334 if (source_channel == -1 || device_spec.type == InputSourceType::SILENCE) {
339 AudioDevice *device = find_audio_device(device_spec);
340 assert(device->interesting_channels.count(source_channel) != 0);
341 unsigned channel_index = 0;
342 for (int channel : device->interesting_channels) {
343 if (channel == source_channel) break;
346 assert(channel_index < device->interesting_channels.size());
347 const auto it = samples_card.find(device_spec);
348 assert(it != samples_card.end());
349 *srcptr = &(it->second)[channel_index];
350 *stride = device->interesting_channels.size();
353 // TODO: Can be SSSE3-optimized if need be.
354 void AudioMixer::fill_audio_bus(const map<DeviceSpec, vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output)
356 if (bus.device.type == InputSourceType::SILENCE) {
357 memset(output, 0, num_samples * sizeof(*output));
359 assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
360 bus.device.type == InputSourceType::ALSA_INPUT);
361 const float *lsrc, *rsrc;
362 unsigned lstride, rstride;
363 float *dptr = output;
364 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[0], &lsrc, &lstride);
365 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[1], &rsrc, &rstride);
366 for (unsigned i = 0; i < num_samples; ++i) {
375 vector<DeviceSpec> AudioMixer::get_active_devices() const
377 vector<DeviceSpec> ret;
378 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
379 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
380 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
381 ret.push_back(device_spec);
384 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
385 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
386 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
387 ret.push_back(device_spec);
393 vector<float> AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
395 map<DeviceSpec, vector<float>> samples_card;
396 vector<float> samples_bus;
398 lock_guard<timed_mutex> lock(audio_mutex);
400 // Pick out all the interesting channels from all the cards.
401 for (const DeviceSpec &device_spec : get_active_devices()) {
402 AudioDevice *device = find_audio_device(device_spec);
403 samples_card[device_spec].resize(num_samples * device->interesting_channels.size());
404 if (device->silenced) {
405 memset(&samples_card[device_spec][0], 0, samples_card[device_spec].size() * sizeof(float));
407 device->resampling_queue->get_output_samples(
409 &samples_card[device_spec][0],
411 rate_adjustment_policy);
415 vector<float> samples_out, left, right;
416 samples_out.resize(num_samples * 2);
417 samples_bus.resize(num_samples * 2);
418 for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
419 fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, &samples_bus[0]);
420 apply_eq(bus_index, &samples_bus);
423 lock_guard<mutex> lock(compressor_mutex);
425 // Apply a level compressor to get the general level right.
426 // Basically, if it's over about -40 dBFS, we squeeze it down to that level
427 // (or more precisely, near it, since we don't use infinite ratio),
428 // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
429 // entirely arbitrary, but from practical tests with speech, it seems to
430 // put ut around -23 LUFS, so it's a reasonable starting point for later use.
431 if (level_compressor_enabled[bus_index]) {
432 float threshold = 0.01f; // -40 dBFS.
434 float attack_time = 0.5f;
435 float release_time = 20.0f;
436 float makeup_gain = from_db(ref_level_dbfs - (-40.0f)); // +26 dB.
437 level_compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
438 gain_staging_db[bus_index] = to_db(level_compressor[bus_index]->get_attenuation() * makeup_gain);
440 // Just apply the gain we already had.
441 float g = from_db(gain_staging_db[bus_index]);
442 for (size_t i = 0; i < samples_bus.size(); ++i) {
448 printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
449 level_compressor.get_level(), to_db(level_compressor.get_level()),
450 level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()),
451 to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
454 // The real compressor.
455 if (compressor_enabled[bus_index]) {
456 float threshold = from_db(compressor_threshold_dbfs[bus_index]);
458 float attack_time = 0.005f;
459 float release_time = 0.040f;
460 float makeup_gain = 2.0f; // +6 dB.
461 compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
462 // compressor_att = compressor.get_attenuation();
466 add_bus_to_master(bus_index, samples_bus, &samples_out);
467 deinterleave_samples(samples_bus, &left, &right);
468 measure_bus_levels(bus_index, left, right);
472 lock_guard<mutex> lock(compressor_mutex);
474 // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
475 // Note that since ratio is not infinite, we could go slightly higher than this.
476 if (limiter_enabled) {
477 float threshold = from_db(limiter_threshold_dbfs);
479 float attack_time = 0.0f; // Instant.
480 float release_time = 0.020f;
481 float makeup_gain = 1.0f; // 0 dB.
482 limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
483 // limiter_att = limiter.get_attenuation();
486 // printf("limiter=%+5.1f compressor=%+5.1f\n", to_db(limiter_att), to_db(compressor_att));
489 // At this point, we are most likely close to +0 LU (at least if the
490 // faders sum to 0 dB and the compressors are on), but all of our
491 // measurements have been on raw sample values, not R128 values.
492 // So we have a final makeup gain to get us to +0 LU; the gain
493 // adjustments required should be relatively small, and also, the
494 // offset shouldn't change much (only if the type of audio changes
495 // significantly). Thus, we shoot for updating this value basically
496 // “whenever we process buffers”, since the R128 calculation isn't exactly
497 // something we get out per-sample.
499 // Note that there's a feedback loop here, so we choose a very slow filter
500 // (half-time of 30 seconds).
501 double target_loudness_factor, alpha;
502 double loudness_lu = r128.loudness_M() - ref_level_lufs;
503 double current_makeup_lu = to_db(final_makeup_gain);
504 target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
506 // If we're outside +/- 5 LU uncorrected, we don't count it as
507 // a normal signal (probably silence) and don't change the
508 // correction factor; just apply what we already have.
509 if (fabs(loudness_lu - current_makeup_lu) >= 5.0 || !final_makeup_gain_auto) {
512 // Formula adapted from
513 // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
514 const double half_time_s = 30.0;
515 const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
516 alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
520 lock_guard<mutex> lock(compressor_mutex);
521 double m = final_makeup_gain;
522 for (size_t i = 0; i < samples_out.size(); i += 2) {
523 samples_out[i + 0] *= m;
524 samples_out[i + 1] *= m;
525 m += (target_loudness_factor - m) * alpha;
527 final_makeup_gain = m;
530 update_meters(samples_out);
535 void AudioMixer::apply_eq(unsigned bus_index, vector<float> *samples_bus)
537 constexpr float bass_freq_hz = 200.0f;
538 constexpr float treble_freq_hz = 4700.0f;
540 // Cut away everything under 120 Hz (or whatever the cutoff is);
541 // we don't need it for voice, and it will reduce headroom
542 // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
543 // should be dampened.)
544 if (locut_enabled[bus_index]) {
545 locut[bus_index].render(samples_bus->data(), samples_bus->size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
548 // Apply the rest of the EQ. Since we only have a simple three-band EQ,
549 // we can implement it with two shelf filters. We use a simple gain to
550 // set the mid-level filter, and then offset the low and high bands
551 // from that if we need to. (We could perhaps have folded the gain into
552 // the next part, but it's so cheap that the trouble isn't worth it.)
553 if (fabs(eq_level_db[bus_index][EQ_BAND_MID]) > 0.01f) {
554 float g = from_db(eq_level_db[bus_index][EQ_BAND_MID]);
555 for (size_t i = 0; i < samples_bus->size(); ++i) {
556 (*samples_bus)[i] *= g;
560 float bass_adj_db = eq_level_db[bus_index][EQ_BAND_BASS] - eq_level_db[bus_index][EQ_BAND_MID];
561 if (fabs(bass_adj_db) > 0.01f) {
562 eq[bus_index][EQ_BAND_BASS].render(samples_bus->data(), samples_bus->size() / 2,
563 bass_freq_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f, bass_adj_db / 40.0f);
566 float treble_adj_db = eq_level_db[bus_index][EQ_BAND_TREBLE] - eq_level_db[bus_index][EQ_BAND_MID];
567 if (fabs(treble_adj_db) > 0.01f) {
568 eq[bus_index][EQ_BAND_TREBLE].render(samples_bus->data(), samples_bus->size() / 2,
569 treble_freq_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f, treble_adj_db / 40.0f);
573 void AudioMixer::add_bus_to_master(unsigned bus_index, const vector<float> &samples_bus, vector<float> *samples_out)
575 assert(samples_bus.size() == samples_out->size());
576 assert(samples_bus.size() % 2 == 0);
577 unsigned num_samples = samples_bus.size() / 2;
578 if (fabs(fader_volume_db[bus_index] - last_fader_volume_db[bus_index]) > 1e-3) {
579 // The volume has changed; do a fade over the course of this frame.
580 // (We might have some numerical issues here, but it seems to sound OK.)
581 // For the purpose of fading here, the silence floor is set to -90 dB
582 // (the fader only goes to -84).
583 float old_volume = from_db(max<float>(last_fader_volume_db[bus_index], -90.0f));
584 float volume = from_db(max<float>(fader_volume_db[bus_index], -90.0f));
586 float volume_inc = pow(volume / old_volume, 1.0 / num_samples);
588 if (bus_index == 0) {
589 for (unsigned i = 0; i < num_samples; ++i) {
590 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
591 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
592 volume *= volume_inc;
595 for (unsigned i = 0; i < num_samples; ++i) {
596 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
597 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
598 volume *= volume_inc;
602 float volume = from_db(fader_volume_db[bus_index]);
603 if (bus_index == 0) {
604 for (unsigned i = 0; i < num_samples; ++i) {
605 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
606 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
609 for (unsigned i = 0; i < num_samples; ++i) {
610 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
611 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
616 last_fader_volume_db[bus_index] = fader_volume_db[bus_index];
619 void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right)
621 assert(left.size() == right.size());
622 const float volume = from_db(fader_volume_db[bus_index]);
623 const float peak_levels[2] = {
624 find_peak(left.data(), left.size()) * volume,
625 find_peak(right.data(), right.size()) * volume
627 for (unsigned channel = 0; channel < 2; ++channel) {
628 // Compute the current value, including hold and falloff.
629 // The constants are borrowed from zita-mu1 by Fons Adriaensen.
630 static constexpr float hold_sec = 0.5f;
631 static constexpr float falloff_db_sec = 15.0f; // dB/sec falloff after hold.
633 PeakHistory &history = peak_history[bus_index][channel];
634 history.historic_peak = max(history.historic_peak, peak_levels[channel]);
635 if (history.age_seconds < hold_sec) {
636 current_peak = history.last_peak;
638 current_peak = history.last_peak * from_db(-falloff_db_sec * (history.age_seconds - hold_sec));
641 // See if we have a new peak to replace the old (possibly falling) one.
642 if (peak_levels[channel] > current_peak) {
643 history.last_peak = peak_levels[channel];
644 history.age_seconds = 0.0f; // Not 100% correct, but more than good enough given our frame sizes.
645 current_peak = peak_levels[channel];
647 history.age_seconds += float(left.size()) / OUTPUT_FREQUENCY;
649 history.current_level = peak_levels[channel];
650 history.current_peak = current_peak;
654 void AudioMixer::update_meters(const vector<float> &samples)
656 // Upsample 4x to find interpolated peak.
657 peak_resampler.inp_data = const_cast<float *>(samples.data());
658 peak_resampler.inp_count = samples.size() / 2;
660 vector<float> interpolated_samples;
661 interpolated_samples.resize(samples.size());
663 lock_guard<mutex> lock(audio_measure_mutex);
665 while (peak_resampler.inp_count > 0) { // About four iterations.
666 peak_resampler.out_data = &interpolated_samples[0];
667 peak_resampler.out_count = interpolated_samples.size() / 2;
668 peak_resampler.process();
669 size_t out_stereo_samples = interpolated_samples.size() / 2 - peak_resampler.out_count;
670 peak = max<float>(peak, find_peak(interpolated_samples.data(), out_stereo_samples * 2));
671 peak_resampler.out_data = nullptr;
675 // Find R128 levels and L/R correlation.
676 vector<float> left, right;
677 deinterleave_samples(samples, &left, &right);
678 float *ptrs[] = { left.data(), right.data() };
680 lock_guard<mutex> lock(audio_measure_mutex);
681 r128.process(left.size(), ptrs);
682 correlation.process_samples(samples);
685 send_audio_level_callback();
688 void AudioMixer::reset_meters()
690 lock_guard<mutex> lock(audio_measure_mutex);
691 peak_resampler.reset();
698 void AudioMixer::send_audio_level_callback()
700 if (audio_level_callback == nullptr) {
704 lock_guard<mutex> lock(audio_measure_mutex);
705 double loudness_s = r128.loudness_S();
706 double loudness_i = r128.integrated();
707 double loudness_range_low = r128.range_min();
708 double loudness_range_high = r128.range_max();
710 vector<BusLevel> bus_levels;
711 bus_levels.resize(input_mapping.buses.size());
713 lock_guard<mutex> lock(compressor_mutex);
714 for (unsigned bus_index = 0; bus_index < bus_levels.size(); ++bus_index) {
715 bus_levels[bus_index].current_level_dbfs[0] = to_db(peak_history[bus_index][0].current_level);
716 bus_levels[bus_index].current_level_dbfs[1] = to_db(peak_history[bus_index][1].current_level);
717 bus_levels[bus_index].peak_level_dbfs[0] = to_db(peak_history[bus_index][0].current_peak);
718 bus_levels[bus_index].peak_level_dbfs[1] = to_db(peak_history[bus_index][1].current_peak);
719 bus_levels[bus_index].historic_peak_dbfs = to_db(
720 max(peak_history[bus_index][0].historic_peak,
721 peak_history[bus_index][1].historic_peak));
722 bus_levels[bus_index].gain_staging_db = gain_staging_db[bus_index];
723 if (compressor_enabled[bus_index]) {
724 bus_levels[bus_index].compressor_attenuation_db = -to_db(compressor[bus_index]->get_attenuation());
726 bus_levels[bus_index].compressor_attenuation_db = 0.0;
731 audio_level_callback(loudness_s, to_db(peak), bus_levels,
732 loudness_i, loudness_range_low, loudness_range_high,
733 to_db(final_makeup_gain),
734 correlation.get_correlation());
737 map<DeviceSpec, DeviceInfo> AudioMixer::get_devices()
739 lock_guard<timed_mutex> lock(audio_mutex);
741 map<DeviceSpec, DeviceInfo> devices;
742 for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
743 const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index };
744 const AudioDevice *device = &video_cards[card_index];
746 info.name = device->name;
747 info.num_channels = 8;
748 devices.insert(make_pair(spec, info));
750 vector<ALSAPool::Device> available_alsa_devices = alsa_pool.get_devices();
751 for (unsigned card_index = 0; card_index < available_alsa_devices.size(); ++card_index) {
752 const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index };
753 const ALSAPool::Device &device = available_alsa_devices[card_index];
755 info.name = device.name + " (" + device.info + ")";
756 info.num_channels = device.num_channels;
757 devices.insert(make_pair(spec, info));
762 void AudioMixer::set_name(DeviceSpec device_spec, const string &name)
764 AudioDevice *device = find_audio_device(device_spec);
766 lock_guard<timed_mutex> lock(audio_mutex);
770 void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
772 lock_guard<timed_mutex> lock(audio_mutex);
774 map<DeviceSpec, set<unsigned>> interesting_channels;
775 for (const InputMapping::Bus &bus : new_input_mapping.buses) {
776 if (bus.device.type == InputSourceType::CAPTURE_CARD ||
777 bus.device.type == InputSourceType::ALSA_INPUT) {
778 for (unsigned channel = 0; channel < 2; ++channel) {
779 if (bus.source_channel[channel] != -1) {
780 interesting_channels[bus.device].insert(bus.source_channel[channel]);
786 // Reset resamplers for all cards that don't have the exact same state as before.
787 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
788 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
789 AudioDevice *device = find_audio_device(device_spec);
790 if (device->interesting_channels != interesting_channels[device_spec]) {
791 device->interesting_channels = interesting_channels[device_spec];
792 reset_resampler_mutex_held(device_spec);
795 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
796 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
797 AudioDevice *device = find_audio_device(device_spec);
798 if (interesting_channels[device_spec].empty()) {
799 alsa_pool.release_device(card_index);
801 alsa_pool.hold_device(card_index);
803 if (device->interesting_channels != interesting_channels[device_spec]) {
804 device->interesting_channels = interesting_channels[device_spec];
805 alsa_pool.reset_device(device_spec.index);
806 reset_resampler_mutex_held(device_spec);
810 input_mapping = new_input_mapping;
813 InputMapping AudioMixer::get_input_mapping() const
815 lock_guard<timed_mutex> lock(audio_mutex);
816 return input_mapping;
819 void AudioMixer::reset_peak(unsigned bus_index)
821 lock_guard<timed_mutex> lock(audio_mutex);
822 for (unsigned channel = 0; channel < 2; ++channel) {
823 PeakHistory &history = peak_history[bus_index][channel];
824 history.current_level = 0.0f;
825 history.historic_peak = 0.0f;
826 history.current_peak = 0.0f;
827 history.last_peak = 0.0f;
828 history.age_seconds = 0.0f;
832 AudioMixer *global_audio_mixer = nullptr;