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 for (unsigned bus_index = 0; bus_index < MAX_BUSES; ++bus_index) {
166 locut[bus_index].init(FILTER_HPF, 2);
167 locut_enabled[bus_index] = global_flags.locut_enabled;
168 eq[bus_index][EQ_BAND_BASS].init(FILTER_LOW_SHELF, 1);
169 // Note: EQ_BAND_MID isn't used (see comments in apply_eq()).
170 eq[bus_index][EQ_BAND_TREBLE].init(FILTER_HIGH_SHELF, 1);
172 gain_staging_db[bus_index] = global_flags.initial_gain_staging_db;
173 compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
174 compressor_threshold_dbfs[bus_index] = ref_level_dbfs - 12.0f; // -12 dB.
175 compressor_enabled[bus_index] = global_flags.compressor_enabled;
176 level_compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
177 level_compressor_enabled[bus_index] = global_flags.gain_staging_auto;
179 set_limiter_enabled(global_flags.limiter_enabled);
180 set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
182 // Generate a very simple, default input mapping.
183 InputMapping::Bus input;
185 input.device.type = InputSourceType::CAPTURE_CARD;
186 input.device.index = 0;
187 input.source_channel[0] = 0;
188 input.source_channel[1] = 1;
190 InputMapping new_input_mapping;
191 new_input_mapping.buses.push_back(input);
192 set_input_mapping(new_input_mapping);
196 r128.init(2, OUTPUT_FREQUENCY);
199 // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
200 // and there's a limit to how important the peak meter is.
201 peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
204 void AudioMixer::reset_resampler(DeviceSpec device_spec)
206 lock_guard<timed_mutex> lock(audio_mutex);
207 reset_resampler_mutex_held(device_spec);
210 void AudioMixer::reset_resampler_mutex_held(DeviceSpec device_spec)
212 AudioDevice *device = find_audio_device(device_spec);
214 if (device->interesting_channels.empty()) {
215 device->resampling_queue.reset();
217 // TODO: ResamplingQueue should probably take the full device spec.
218 // (It's only used for console output, though.)
219 device->resampling_queue.reset(new ResamplingQueue(device_spec.index, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size()));
221 device->next_local_pts = 0;
224 bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length)
226 AudioDevice *device = find_audio_device(device_spec);
228 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
229 if (!lock.try_lock_for(chrono::milliseconds(10))) {
232 if (device->resampling_queue == nullptr) {
233 // No buses use this device; throw it away.
237 unsigned num_channels = device->interesting_channels.size();
238 assert(num_channels > 0);
240 // Convert the audio to fp32.
241 unique_ptr<float[]> audio(new float[num_samples * num_channels]);
242 unsigned channel_index = 0;
243 for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) {
244 switch (audio_format.bits_per_sample) {
246 assert(num_samples == 0);
249 convert_fixed16_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
252 convert_fixed24_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
255 convert_fixed32_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
258 fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
264 int64_t local_pts = device->next_local_pts;
265 device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.get(), num_samples);
266 device->next_local_pts = local_pts + frame_length;
270 bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length)
272 AudioDevice *device = find_audio_device(device_spec);
274 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
275 if (!lock.try_lock_for(chrono::milliseconds(10))) {
278 if (device->resampling_queue == nullptr) {
279 // No buses use this device; throw it away.
283 unsigned num_channels = device->interesting_channels.size();
284 assert(num_channels > 0);
286 vector<float> silence(samples_per_frame * num_channels, 0.0f);
287 for (unsigned i = 0; i < num_frames; ++i) {
288 device->resampling_queue->add_input_samples(device->next_local_pts / double(TIMEBASE), silence.data(), samples_per_frame);
289 // Note that if the format changed in the meantime, we have
290 // no way of detecting that; we just have to assume the frame length
291 // is always the same.
292 device->next_local_pts += frame_length;
297 bool AudioMixer::silence_card(DeviceSpec device_spec, bool silence)
299 AudioDevice *device = find_audio_device(device_spec);
301 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
302 if (!lock.try_lock_for(chrono::milliseconds(10))) {
306 if (device->silenced && !silence) {
307 reset_resampler_mutex_held(device_spec);
309 device->silenced = silence;
313 AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
315 switch (device.type) {
316 case InputSourceType::CAPTURE_CARD:
317 return &video_cards[device.index];
318 case InputSourceType::ALSA_INPUT:
319 return &alsa_inputs[device.index];
320 case InputSourceType::SILENCE:
327 // Get a pointer to the given channel from the given device.
328 // The channel must be picked out earlier and resampled.
329 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)
331 static float zero = 0.0f;
332 if (source_channel == -1 || device_spec.type == InputSourceType::SILENCE) {
337 AudioDevice *device = find_audio_device(device_spec);
338 assert(device->interesting_channels.count(source_channel) != 0);
339 unsigned channel_index = 0;
340 for (int channel : device->interesting_channels) {
341 if (channel == source_channel) break;
344 assert(channel_index < device->interesting_channels.size());
345 const auto it = samples_card.find(device_spec);
346 assert(it != samples_card.end());
347 *srcptr = &(it->second)[channel_index];
348 *stride = device->interesting_channels.size();
351 // TODO: Can be SSSE3-optimized if need be.
352 void AudioMixer::fill_audio_bus(const map<DeviceSpec, vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output)
354 if (bus.device.type == InputSourceType::SILENCE) {
355 memset(output, 0, num_samples * sizeof(*output));
357 assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
358 bus.device.type == InputSourceType::ALSA_INPUT);
359 const float *lsrc, *rsrc;
360 unsigned lstride, rstride;
361 float *dptr = output;
362 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[0], &lsrc, &lstride);
363 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[1], &rsrc, &rstride);
364 for (unsigned i = 0; i < num_samples; ++i) {
373 vector<DeviceSpec> AudioMixer::get_active_devices() const
375 vector<DeviceSpec> ret;
376 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
377 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
378 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
379 ret.push_back(device_spec);
382 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
383 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
384 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
385 ret.push_back(device_spec);
391 vector<float> AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
393 map<DeviceSpec, vector<float>> samples_card;
394 vector<float> samples_bus;
396 lock_guard<timed_mutex> lock(audio_mutex);
398 // Pick out all the interesting channels from all the cards.
399 for (const DeviceSpec &device_spec : get_active_devices()) {
400 AudioDevice *device = find_audio_device(device_spec);
401 samples_card[device_spec].resize(num_samples * device->interesting_channels.size());
402 if (device->silenced) {
403 memset(&samples_card[device_spec][0], 0, samples_card[device_spec].size() * sizeof(float));
405 device->resampling_queue->get_output_samples(
407 &samples_card[device_spec][0],
409 rate_adjustment_policy);
413 vector<float> samples_out, left, right;
414 samples_out.resize(num_samples * 2);
415 samples_bus.resize(num_samples * 2);
416 for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
417 fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, &samples_bus[0]);
418 apply_eq(bus_index, &samples_bus);
421 lock_guard<mutex> lock(compressor_mutex);
423 // Apply a level compressor to get the general level right.
424 // Basically, if it's over about -40 dBFS, we squeeze it down to that level
425 // (or more precisely, near it, since we don't use infinite ratio),
426 // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
427 // entirely arbitrary, but from practical tests with speech, it seems to
428 // put ut around -23 LUFS, so it's a reasonable starting point for later use.
429 if (level_compressor_enabled[bus_index]) {
430 float threshold = 0.01f; // -40 dBFS.
432 float attack_time = 0.5f;
433 float release_time = 20.0f;
434 float makeup_gain = from_db(ref_level_dbfs - (-40.0f)); // +26 dB.
435 level_compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
436 gain_staging_db[bus_index] = to_db(level_compressor[bus_index]->get_attenuation() * makeup_gain);
438 // Just apply the gain we already had.
439 float g = from_db(gain_staging_db[bus_index]);
440 for (size_t i = 0; i < samples_bus.size(); ++i) {
446 printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
447 level_compressor.get_level(), to_db(level_compressor.get_level()),
448 level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()),
449 to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
452 // The real compressor.
453 if (compressor_enabled[bus_index]) {
454 float threshold = from_db(compressor_threshold_dbfs[bus_index]);
456 float attack_time = 0.005f;
457 float release_time = 0.040f;
458 float makeup_gain = 2.0f; // +6 dB.
459 compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
460 // compressor_att = compressor.get_attenuation();
464 add_bus_to_master(bus_index, samples_bus, &samples_out);
465 deinterleave_samples(samples_bus, &left, &right);
466 measure_bus_levels(bus_index, left, right);
470 lock_guard<mutex> lock(compressor_mutex);
472 // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
473 // Note that since ratio is not infinite, we could go slightly higher than this.
474 if (limiter_enabled) {
475 float threshold = from_db(limiter_threshold_dbfs);
477 float attack_time = 0.0f; // Instant.
478 float release_time = 0.020f;
479 float makeup_gain = 1.0f; // 0 dB.
480 limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
481 // limiter_att = limiter.get_attenuation();
484 // printf("limiter=%+5.1f compressor=%+5.1f\n", to_db(limiter_att), to_db(compressor_att));
487 // At this point, we are most likely close to +0 LU (at least if the
488 // faders sum to 0 dB and the compressors are on), but all of our
489 // measurements have been on raw sample values, not R128 values.
490 // So we have a final makeup gain to get us to +0 LU; the gain
491 // adjustments required should be relatively small, and also, the
492 // offset shouldn't change much (only if the type of audio changes
493 // significantly). Thus, we shoot for updating this value basically
494 // “whenever we process buffers”, since the R128 calculation isn't exactly
495 // something we get out per-sample.
497 // Note that there's a feedback loop here, so we choose a very slow filter
498 // (half-time of 30 seconds).
499 double target_loudness_factor, alpha;
500 double loudness_lu = r128.loudness_M() - ref_level_lufs;
501 double current_makeup_lu = to_db(final_makeup_gain);
502 target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
504 // If we're outside +/- 5 LU uncorrected, we don't count it as
505 // a normal signal (probably silence) and don't change the
506 // correction factor; just apply what we already have.
507 if (fabs(loudness_lu - current_makeup_lu) >= 5.0 || !final_makeup_gain_auto) {
510 // Formula adapted from
511 // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
512 const double half_time_s = 30.0;
513 const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
514 alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
518 lock_guard<mutex> lock(compressor_mutex);
519 double m = final_makeup_gain;
520 for (size_t i = 0; i < samples_out.size(); i += 2) {
521 samples_out[i + 0] *= m;
522 samples_out[i + 1] *= m;
523 m += (target_loudness_factor - m) * alpha;
525 final_makeup_gain = m;
528 update_meters(samples_out);
533 void AudioMixer::apply_eq(unsigned bus_index, vector<float> *samples_bus)
535 constexpr float bass_freq_hz = 200.0f;
536 constexpr float treble_freq_hz = 4700.0f;
538 // Cut away everything under 120 Hz (or whatever the cutoff is);
539 // we don't need it for voice, and it will reduce headroom
540 // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
541 // should be dampened.)
542 if (locut_enabled[bus_index]) {
543 locut[bus_index].render(samples_bus->data(), samples_bus->size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
546 // Apply the rest of the EQ. Since we only have a simple three-band EQ,
547 // we can implement it with two shelf filters. We use a simple gain to
548 // set the mid-level filter, and then offset the low and high bands
549 // from that if we need to. (We could perhaps have folded the gain into
550 // the next part, but it's so cheap that the trouble isn't worth it.)
551 if (fabs(eq_level_db[bus_index][EQ_BAND_MID]) > 0.01f) {
552 float g = from_db(eq_level_db[bus_index][EQ_BAND_MID]);
553 for (size_t i = 0; i < samples_bus->size(); ++i) {
554 (*samples_bus)[i] *= g;
558 float bass_adj_db = eq_level_db[bus_index][EQ_BAND_BASS] - eq_level_db[bus_index][EQ_BAND_MID];
559 if (fabs(bass_adj_db) > 0.01f) {
560 eq[bus_index][EQ_BAND_BASS].render(samples_bus->data(), samples_bus->size() / 2,
561 bass_freq_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f, bass_adj_db / 40.0f);
564 float treble_adj_db = eq_level_db[bus_index][EQ_BAND_TREBLE] - eq_level_db[bus_index][EQ_BAND_MID];
565 if (fabs(treble_adj_db) > 0.01f) {
566 eq[bus_index][EQ_BAND_TREBLE].render(samples_bus->data(), samples_bus->size() / 2,
567 treble_freq_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f, treble_adj_db / 40.0f);
571 void AudioMixer::add_bus_to_master(unsigned bus_index, const vector<float> &samples_bus, vector<float> *samples_out)
573 assert(samples_bus.size() == samples_out->size());
574 assert(samples_bus.size() % 2 == 0);
575 unsigned num_samples = samples_bus.size() / 2;
576 if (fabs(fader_volume_db[bus_index] - last_fader_volume_db[bus_index]) > 1e-3) {
577 // The volume has changed; do a fade over the course of this frame.
578 // (We might have some numerical issues here, but it seems to sound OK.)
579 // For the purpose of fading here, the silence floor is set to -90 dB
580 // (the fader only goes to -84).
581 float old_volume = from_db(max<float>(last_fader_volume_db[bus_index], -90.0f));
582 float volume = from_db(max<float>(fader_volume_db[bus_index], -90.0f));
584 float volume_inc = pow(volume / old_volume, 1.0 / num_samples);
586 if (bus_index == 0) {
587 for (unsigned i = 0; i < num_samples; ++i) {
588 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
589 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
590 volume *= volume_inc;
593 for (unsigned i = 0; i < num_samples; ++i) {
594 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
595 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
596 volume *= volume_inc;
600 float volume = from_db(fader_volume_db[bus_index]);
601 if (bus_index == 0) {
602 for (unsigned i = 0; i < num_samples; ++i) {
603 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
604 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
607 for (unsigned i = 0; i < num_samples; ++i) {
608 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
609 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
614 last_fader_volume_db[bus_index] = fader_volume_db[bus_index];
617 void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right)
619 assert(left.size() == right.size());
620 const float volume = from_db(fader_volume_db[bus_index]);
621 const float peak_levels[2] = {
622 find_peak(left.data(), left.size()) * volume,
623 find_peak(right.data(), right.size()) * volume
625 for (unsigned channel = 0; channel < 2; ++channel) {
626 // Compute the current value, including hold and falloff.
627 // The constants are borrowed from zita-mu1 by Fons Adriaensen.
628 static constexpr float hold_sec = 0.5f;
629 static constexpr float falloff_db_sec = 15.0f; // dB/sec falloff after hold.
631 PeakHistory &history = peak_history[bus_index][channel];
632 history.historic_peak = max(history.historic_peak, peak_levels[channel]);
633 if (history.age_seconds < hold_sec) {
634 current_peak = history.last_peak;
636 current_peak = history.last_peak * from_db(-falloff_db_sec * (history.age_seconds - hold_sec));
639 // See if we have a new peak to replace the old (possibly falling) one.
640 if (peak_levels[channel] > current_peak) {
641 history.last_peak = peak_levels[channel];
642 history.age_seconds = 0.0f; // Not 100% correct, but more than good enough given our frame sizes.
643 current_peak = peak_levels[channel];
645 history.age_seconds += float(left.size()) / OUTPUT_FREQUENCY;
647 history.current_level = peak_levels[channel];
648 history.current_peak = current_peak;
652 void AudioMixer::update_meters(const vector<float> &samples)
654 // Upsample 4x to find interpolated peak.
655 peak_resampler.inp_data = const_cast<float *>(samples.data());
656 peak_resampler.inp_count = samples.size() / 2;
658 vector<float> interpolated_samples;
659 interpolated_samples.resize(samples.size());
661 lock_guard<mutex> lock(audio_measure_mutex);
663 while (peak_resampler.inp_count > 0) { // About four iterations.
664 peak_resampler.out_data = &interpolated_samples[0];
665 peak_resampler.out_count = interpolated_samples.size() / 2;
666 peak_resampler.process();
667 size_t out_stereo_samples = interpolated_samples.size() / 2 - peak_resampler.out_count;
668 peak = max<float>(peak, find_peak(interpolated_samples.data(), out_stereo_samples * 2));
669 peak_resampler.out_data = nullptr;
673 // Find R128 levels and L/R correlation.
674 vector<float> left, right;
675 deinterleave_samples(samples, &left, &right);
676 float *ptrs[] = { left.data(), right.data() };
678 lock_guard<mutex> lock(audio_measure_mutex);
679 r128.process(left.size(), ptrs);
680 correlation.process_samples(samples);
683 send_audio_level_callback();
686 void AudioMixer::reset_meters()
688 lock_guard<mutex> lock(audio_measure_mutex);
689 peak_resampler.reset();
696 void AudioMixer::send_audio_level_callback()
698 if (audio_level_callback == nullptr) {
702 lock_guard<mutex> lock(audio_measure_mutex);
703 double loudness_s = r128.loudness_S();
704 double loudness_i = r128.integrated();
705 double loudness_range_low = r128.range_min();
706 double loudness_range_high = r128.range_max();
708 vector<BusLevel> bus_levels;
709 bus_levels.resize(input_mapping.buses.size());
711 lock_guard<mutex> lock(compressor_mutex);
712 for (unsigned bus_index = 0; bus_index < bus_levels.size(); ++bus_index) {
713 bus_levels[bus_index].current_level_dbfs[0] = to_db(peak_history[bus_index][0].current_level);
714 bus_levels[bus_index].current_level_dbfs[1] = to_db(peak_history[bus_index][1].current_level);
715 bus_levels[bus_index].peak_level_dbfs[0] = to_db(peak_history[bus_index][0].current_peak);
716 bus_levels[bus_index].peak_level_dbfs[1] = to_db(peak_history[bus_index][1].current_peak);
717 bus_levels[bus_index].historic_peak_dbfs = to_db(
718 max(peak_history[bus_index][0].historic_peak,
719 peak_history[bus_index][1].historic_peak));
720 bus_levels[bus_index].gain_staging_db = gain_staging_db[bus_index];
721 if (compressor_enabled[bus_index]) {
722 bus_levels[bus_index].compressor_attenuation_db = -to_db(compressor[bus_index]->get_attenuation());
724 bus_levels[bus_index].compressor_attenuation_db = 0.0;
729 audio_level_callback(loudness_s, to_db(peak), bus_levels,
730 loudness_i, loudness_range_low, loudness_range_high,
731 to_db(final_makeup_gain),
732 correlation.get_correlation());
735 map<DeviceSpec, DeviceInfo> AudioMixer::get_devices()
737 lock_guard<timed_mutex> lock(audio_mutex);
739 map<DeviceSpec, DeviceInfo> devices;
740 for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
741 const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index };
742 const AudioDevice *device = &video_cards[card_index];
744 info.name = device->name;
745 info.num_channels = 8; // FIXME: This is wrong for fake cards.
746 devices.insert(make_pair(spec, info));
748 vector<ALSAPool::Device> available_alsa_devices = alsa_pool.get_devices();
749 for (unsigned card_index = 0; card_index < available_alsa_devices.size(); ++card_index) {
750 const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index };
751 const ALSAPool::Device &device = available_alsa_devices[card_index];
753 info.name = device.name + " (" + device.info + ")";
754 info.num_channels = device.num_channels;
755 devices.insert(make_pair(spec, info));
760 void AudioMixer::set_name(DeviceSpec device_spec, const string &name)
762 AudioDevice *device = find_audio_device(device_spec);
764 lock_guard<timed_mutex> lock(audio_mutex);
768 void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
770 lock_guard<timed_mutex> lock(audio_mutex);
772 map<DeviceSpec, set<unsigned>> interesting_channels;
773 for (const InputMapping::Bus &bus : new_input_mapping.buses) {
774 if (bus.device.type == InputSourceType::CAPTURE_CARD ||
775 bus.device.type == InputSourceType::ALSA_INPUT) {
776 for (unsigned channel = 0; channel < 2; ++channel) {
777 if (bus.source_channel[channel] != -1) {
778 interesting_channels[bus.device].insert(bus.source_channel[channel]);
784 // Reset resamplers for all cards that don't have the exact same state as before.
785 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
786 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
787 AudioDevice *device = find_audio_device(device_spec);
788 if (device->interesting_channels != interesting_channels[device_spec]) {
789 device->interesting_channels = interesting_channels[device_spec];
790 reset_resampler_mutex_held(device_spec);
793 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
794 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
795 AudioDevice *device = find_audio_device(device_spec);
796 if (interesting_channels[device_spec].empty()) {
797 alsa_pool.release_device(card_index);
799 alsa_pool.hold_device(card_index);
801 if (device->interesting_channels != interesting_channels[device_spec]) {
802 device->interesting_channels = interesting_channels[device_spec];
803 alsa_pool.reset_device(device_spec.index);
804 reset_resampler_mutex_held(device_spec);
808 input_mapping = new_input_mapping;
811 InputMapping AudioMixer::get_input_mapping() const
813 lock_guard<timed_mutex> lock(audio_mutex);
814 return input_mapping;
817 void AudioMixer::reset_peak(unsigned bus_index)
819 lock_guard<timed_mutex> lock(audio_mutex);
820 for (unsigned channel = 0; channel < 2; ++channel) {
821 PeakHistory &history = peak_history[bus_index][channel];
822 history.current_level = 0.0f;
823 history.historic_peak = 0.0f;
824 history.current_peak = 0.0f;
825 history.last_peak = 0.0f;
826 history.age_seconds = 0.0f;
830 AudioMixer *global_audio_mixer = nullptr;