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 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);
173 compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
174 level_compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
176 set_bus_settings(bus_index, get_default_bus_settings());
178 set_limiter_enabled(global_flags.limiter_enabled);
179 set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
182 InputMapping new_input_mapping;
183 if (!global_flags.input_mapping_filename.empty()) {
184 if (!load_input_mapping_from_file(get_devices(),
185 global_flags.input_mapping_filename,
186 &new_input_mapping)) {
187 fprintf(stderr, "Failed to load input mapping from '%s', exiting.\n",
188 global_flags.input_mapping_filename.c_str());
192 // Generate a very simple, default input mapping.
193 InputMapping::Bus input;
195 input.device.type = InputSourceType::CAPTURE_CARD;
196 input.device.index = 0;
197 input.source_channel[0] = 0;
198 input.source_channel[1] = 1;
200 new_input_mapping.buses.push_back(input);
202 set_input_mapping(new_input_mapping);
204 r128.init(2, OUTPUT_FREQUENCY);
207 // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
208 // and there's a limit to how important the peak meter is.
209 peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
212 void AudioMixer::reset_resampler(DeviceSpec device_spec)
214 lock_guard<timed_mutex> lock(audio_mutex);
215 reset_resampler_mutex_held(device_spec);
218 void AudioMixer::reset_resampler_mutex_held(DeviceSpec device_spec)
220 AudioDevice *device = find_audio_device(device_spec);
222 if (device->interesting_channels.empty()) {
223 device->resampling_queue.reset();
225 // TODO: ResamplingQueue should probably take the full device spec.
226 // (It's only used for console output, though.)
227 device->resampling_queue.reset(new ResamplingQueue(device_spec.index, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size()));
229 device->next_local_pts = 0;
232 bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length)
234 AudioDevice *device = find_audio_device(device_spec);
236 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
237 if (!lock.try_lock_for(chrono::milliseconds(10))) {
240 if (device->resampling_queue == nullptr) {
241 // No buses use this device; throw it away.
245 unsigned num_channels = device->interesting_channels.size();
246 assert(num_channels > 0);
248 // Convert the audio to fp32.
249 unique_ptr<float[]> audio(new float[num_samples * num_channels]);
250 unsigned channel_index = 0;
251 for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) {
252 switch (audio_format.bits_per_sample) {
254 assert(num_samples == 0);
257 convert_fixed16_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
260 convert_fixed24_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
263 convert_fixed32_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
266 fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
272 int64_t local_pts = device->next_local_pts;
273 device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.get(), num_samples);
274 device->next_local_pts = local_pts + frame_length;
278 bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length)
280 AudioDevice *device = find_audio_device(device_spec);
282 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
283 if (!lock.try_lock_for(chrono::milliseconds(10))) {
286 if (device->resampling_queue == nullptr) {
287 // No buses use this device; throw it away.
291 unsigned num_channels = device->interesting_channels.size();
292 assert(num_channels > 0);
294 vector<float> silence(samples_per_frame * num_channels, 0.0f);
295 for (unsigned i = 0; i < num_frames; ++i) {
296 device->resampling_queue->add_input_samples(device->next_local_pts / double(TIMEBASE), silence.data(), samples_per_frame);
297 // Note that if the format changed in the meantime, we have
298 // no way of detecting that; we just have to assume the frame length
299 // is always the same.
300 device->next_local_pts += frame_length;
305 bool AudioMixer::silence_card(DeviceSpec device_spec, bool silence)
307 AudioDevice *device = find_audio_device(device_spec);
309 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
310 if (!lock.try_lock_for(chrono::milliseconds(10))) {
314 if (device->silenced && !silence) {
315 reset_resampler_mutex_held(device_spec);
317 device->silenced = silence;
321 AudioMixer::BusSettings AudioMixer::get_default_bus_settings()
323 BusSettings settings;
324 settings.fader_volume_db = 0.0f;
325 settings.locut_enabled = global_flags.locut_enabled;
326 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
327 settings.eq_level_db[band_index] = 0.0f;
329 settings.gain_staging_db = global_flags.initial_gain_staging_db;
330 settings.level_compressor_enabled = global_flags.gain_staging_auto;
331 settings.compressor_threshold_dbfs = ref_level_dbfs - 12.0f; // -12 dB.
332 settings.compressor_enabled = global_flags.compressor_enabled;
336 AudioMixer::BusSettings AudioMixer::get_bus_settings(unsigned bus_index) const
338 lock_guard<timed_mutex> lock(audio_mutex);
339 BusSettings settings;
340 settings.fader_volume_db = fader_volume_db[bus_index];
341 settings.locut_enabled = locut_enabled[bus_index];
342 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
343 settings.eq_level_db[band_index] = eq_level_db[bus_index][band_index];
345 settings.gain_staging_db = gain_staging_db[bus_index];
346 settings.level_compressor_enabled = level_compressor_enabled[bus_index];
347 settings.compressor_threshold_dbfs = compressor_threshold_dbfs[bus_index];
348 settings.compressor_enabled = compressor_enabled[bus_index];
352 void AudioMixer::set_bus_settings(unsigned bus_index, const AudioMixer::BusSettings &settings)
354 lock_guard<timed_mutex> lock(audio_mutex);
355 fader_volume_db[bus_index] = settings.fader_volume_db;
356 locut_enabled[bus_index] = settings.locut_enabled;
357 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
358 eq_level_db[bus_index][band_index] = settings.eq_level_db[band_index];
360 gain_staging_db[bus_index] = settings.gain_staging_db;
361 level_compressor_enabled[bus_index] = settings.level_compressor_enabled;
362 compressor_threshold_dbfs[bus_index] = settings.compressor_threshold_dbfs;
363 compressor_enabled[bus_index] = settings.compressor_enabled;
366 AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
368 switch (device.type) {
369 case InputSourceType::CAPTURE_CARD:
370 return &video_cards[device.index];
371 case InputSourceType::ALSA_INPUT:
372 return &alsa_inputs[device.index];
373 case InputSourceType::SILENCE:
380 // Get a pointer to the given channel from the given device.
381 // The channel must be picked out earlier and resampled.
382 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)
384 static float zero = 0.0f;
385 if (source_channel == -1 || device_spec.type == InputSourceType::SILENCE) {
390 AudioDevice *device = find_audio_device(device_spec);
391 assert(device->interesting_channels.count(source_channel) != 0);
392 unsigned channel_index = 0;
393 for (int channel : device->interesting_channels) {
394 if (channel == source_channel) break;
397 assert(channel_index < device->interesting_channels.size());
398 const auto it = samples_card.find(device_spec);
399 assert(it != samples_card.end());
400 *srcptr = &(it->second)[channel_index];
401 *stride = device->interesting_channels.size();
404 // TODO: Can be SSSE3-optimized if need be.
405 void AudioMixer::fill_audio_bus(const map<DeviceSpec, vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output)
407 if (bus.device.type == InputSourceType::SILENCE) {
408 memset(output, 0, num_samples * sizeof(*output));
410 assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
411 bus.device.type == InputSourceType::ALSA_INPUT);
412 const float *lsrc, *rsrc;
413 unsigned lstride, rstride;
414 float *dptr = output;
415 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[0], &lsrc, &lstride);
416 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[1], &rsrc, &rstride);
417 for (unsigned i = 0; i < num_samples; ++i) {
426 vector<DeviceSpec> AudioMixer::get_active_devices() const
428 vector<DeviceSpec> ret;
429 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
430 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
431 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
432 ret.push_back(device_spec);
435 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
436 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
437 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
438 ret.push_back(device_spec);
444 vector<float> AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
446 map<DeviceSpec, vector<float>> samples_card;
447 vector<float> samples_bus;
449 lock_guard<timed_mutex> lock(audio_mutex);
451 // Pick out all the interesting channels from all the cards.
452 for (const DeviceSpec &device_spec : get_active_devices()) {
453 AudioDevice *device = find_audio_device(device_spec);
454 samples_card[device_spec].resize(num_samples * device->interesting_channels.size());
455 if (device->silenced) {
456 memset(&samples_card[device_spec][0], 0, samples_card[device_spec].size() * sizeof(float));
458 device->resampling_queue->get_output_samples(
460 &samples_card[device_spec][0],
462 rate_adjustment_policy);
466 vector<float> samples_out, left, right;
467 samples_out.resize(num_samples * 2);
468 samples_bus.resize(num_samples * 2);
469 for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
470 fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, &samples_bus[0]);
471 apply_eq(bus_index, &samples_bus);
474 lock_guard<mutex> lock(compressor_mutex);
476 // Apply a level compressor to get the general level right.
477 // Basically, if it's over about -40 dBFS, we squeeze it down to that level
478 // (or more precisely, near it, since we don't use infinite ratio),
479 // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
480 // entirely arbitrary, but from practical tests with speech, it seems to
481 // put ut around -23 LUFS, so it's a reasonable starting point for later use.
482 if (level_compressor_enabled[bus_index]) {
483 float threshold = 0.01f; // -40 dBFS.
485 float attack_time = 0.5f;
486 float release_time = 20.0f;
487 float makeup_gain = from_db(ref_level_dbfs - (-40.0f)); // +26 dB.
488 level_compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
489 gain_staging_db[bus_index] = to_db(level_compressor[bus_index]->get_attenuation() * makeup_gain);
491 // Just apply the gain we already had.
492 float g = from_db(gain_staging_db[bus_index]);
493 for (size_t i = 0; i < samples_bus.size(); ++i) {
499 printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
500 level_compressor.get_level(), to_db(level_compressor.get_level()),
501 level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()),
502 to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
505 // The real compressor.
506 if (compressor_enabled[bus_index]) {
507 float threshold = from_db(compressor_threshold_dbfs[bus_index]);
509 float attack_time = 0.005f;
510 float release_time = 0.040f;
511 float makeup_gain = 2.0f; // +6 dB.
512 compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
513 // compressor_att = compressor.get_attenuation();
517 add_bus_to_master(bus_index, samples_bus, &samples_out);
518 deinterleave_samples(samples_bus, &left, &right);
519 measure_bus_levels(bus_index, left, right);
523 lock_guard<mutex> lock(compressor_mutex);
525 // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
526 // Note that since ratio is not infinite, we could go slightly higher than this.
527 if (limiter_enabled) {
528 float threshold = from_db(limiter_threshold_dbfs);
530 float attack_time = 0.0f; // Instant.
531 float release_time = 0.020f;
532 float makeup_gain = 1.0f; // 0 dB.
533 limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
534 // limiter_att = limiter.get_attenuation();
537 // printf("limiter=%+5.1f compressor=%+5.1f\n", to_db(limiter_att), to_db(compressor_att));
540 // At this point, we are most likely close to +0 LU (at least if the
541 // faders sum to 0 dB and the compressors are on), but all of our
542 // measurements have been on raw sample values, not R128 values.
543 // So we have a final makeup gain to get us to +0 LU; the gain
544 // adjustments required should be relatively small, and also, the
545 // offset shouldn't change much (only if the type of audio changes
546 // significantly). Thus, we shoot for updating this value basically
547 // “whenever we process buffers”, since the R128 calculation isn't exactly
548 // something we get out per-sample.
550 // Note that there's a feedback loop here, so we choose a very slow filter
551 // (half-time of 30 seconds).
552 double target_loudness_factor, alpha;
553 double loudness_lu = r128.loudness_M() - ref_level_lufs;
554 double current_makeup_lu = to_db(final_makeup_gain);
555 target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
557 // If we're outside +/- 5 LU uncorrected, we don't count it as
558 // a normal signal (probably silence) and don't change the
559 // correction factor; just apply what we already have.
560 if (fabs(loudness_lu - current_makeup_lu) >= 5.0 || !final_makeup_gain_auto) {
563 // Formula adapted from
564 // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
565 const double half_time_s = 30.0;
566 const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
567 alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
571 lock_guard<mutex> lock(compressor_mutex);
572 double m = final_makeup_gain;
573 for (size_t i = 0; i < samples_out.size(); i += 2) {
574 samples_out[i + 0] *= m;
575 samples_out[i + 1] *= m;
576 m += (target_loudness_factor - m) * alpha;
578 final_makeup_gain = m;
581 update_meters(samples_out);
588 void apply_filter_fade(StereoFilter *filter, float *data, unsigned num_samples, float cutoff_hz, float db, float last_db)
590 // A granularity of 32 samples is an okay tradeoff between speed and
591 // smoothness; recalculating the filters is pretty expensive, so it's
592 // good that we don't do this all the time.
593 static constexpr unsigned filter_granularity_samples = 32;
595 const float cutoff_linear = cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY;
596 if (fabs(db - last_db) < 1e-3) {
597 // Constant over this frame.
598 if (fabs(db) > 0.01f) {
599 filter->render(data, num_samples, cutoff_linear, 0.5f, db / 40.0f);
602 // We need to do a fade. (Rounding up avoids division by zero.)
603 unsigned num_blocks = (num_samples + filter_granularity_samples - 1) / filter_granularity_samples;
604 const float inc_db_norm = (db - last_db) / 40.0f / num_blocks;
605 float db_norm = db / 40.0f;
606 for (size_t i = 0; i < num_samples; i += filter_granularity_samples) {
607 size_t samples_this_block = std::min<size_t>(num_samples - i, filter_granularity_samples);
608 filter->render(data + i * 2, samples_this_block, cutoff_linear, 0.5f, db_norm);
609 db_norm += inc_db_norm;
616 void AudioMixer::apply_eq(unsigned bus_index, vector<float> *samples_bus)
618 constexpr float bass_freq_hz = 200.0f;
619 constexpr float treble_freq_hz = 4700.0f;
621 // Cut away everything under 120 Hz (or whatever the cutoff is);
622 // we don't need it for voice, and it will reduce headroom
623 // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
624 // should be dampened.)
625 if (locut_enabled[bus_index]) {
626 locut[bus_index].render(samples_bus->data(), samples_bus->size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
629 // Apply the rest of the EQ. Since we only have a simple three-band EQ,
630 // we can implement it with two shelf filters. We use a simple gain to
631 // set the mid-level filter, and then offset the low and high bands
632 // from that if we need to. (We could perhaps have folded the gain into
633 // the next part, but it's so cheap that the trouble isn't worth it.)
635 // If any part of the EQ has changed appreciably since last frame,
636 // we fade smoothly during the course of this frame.
637 const float bass_db = eq_level_db[bus_index][EQ_BAND_BASS];
638 const float mid_db = eq_level_db[bus_index][EQ_BAND_MID];
639 const float treble_db = eq_level_db[bus_index][EQ_BAND_TREBLE];
641 const float last_bass_db = last_eq_level_db[bus_index][EQ_BAND_BASS];
642 const float last_mid_db = last_eq_level_db[bus_index][EQ_BAND_MID];
643 const float last_treble_db = last_eq_level_db[bus_index][EQ_BAND_TREBLE];
645 assert(samples_bus->size() % 2 == 0);
646 const unsigned num_samples = samples_bus->size() / 2;
648 if (fabs(mid_db - last_mid_db) < 1e-3) {
649 // Constant over this frame.
650 const float gain = from_db(mid_db);
651 for (size_t i = 0; i < samples_bus->size(); ++i) {
652 (*samples_bus)[i] *= gain;
655 // We need to do a fade.
656 float gain = from_db(last_mid_db);
657 const float gain_inc = pow(from_db(mid_db - last_mid_db), 1.0 / num_samples);
658 for (size_t i = 0; i < num_samples; ++i) {
659 (*samples_bus)[i * 2 + 0] *= gain;
660 (*samples_bus)[i * 2 + 1] *= gain;
665 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);
666 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);
668 last_eq_level_db[bus_index][EQ_BAND_BASS] = bass_db;
669 last_eq_level_db[bus_index][EQ_BAND_MID] = mid_db;
670 last_eq_level_db[bus_index][EQ_BAND_TREBLE] = treble_db;
673 void AudioMixer::add_bus_to_master(unsigned bus_index, const vector<float> &samples_bus, vector<float> *samples_out)
675 assert(samples_bus.size() == samples_out->size());
676 assert(samples_bus.size() % 2 == 0);
677 unsigned num_samples = samples_bus.size() / 2;
678 if (fabs(fader_volume_db[bus_index] - last_fader_volume_db[bus_index]) > 1e-3) {
679 // The volume has changed; do a fade over the course of this frame.
680 // (We might have some numerical issues here, but it seems to sound OK.)
681 // For the purpose of fading here, the silence floor is set to -90 dB
682 // (the fader only goes to -84).
683 float old_volume = from_db(max<float>(last_fader_volume_db[bus_index], -90.0f));
684 float volume = from_db(max<float>(fader_volume_db[bus_index], -90.0f));
686 float volume_inc = pow(volume / old_volume, 1.0 / num_samples);
688 if (bus_index == 0) {
689 for (unsigned i = 0; i < num_samples; ++i) {
690 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
691 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
692 volume *= volume_inc;
695 for (unsigned i = 0; i < num_samples; ++i) {
696 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
697 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
698 volume *= volume_inc;
702 float volume = from_db(fader_volume_db[bus_index]);
703 if (bus_index == 0) {
704 for (unsigned i = 0; i < num_samples; ++i) {
705 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
706 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
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;
716 last_fader_volume_db[bus_index] = fader_volume_db[bus_index];
719 void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right)
721 assert(left.size() == right.size());
722 const float volume = from_db(fader_volume_db[bus_index]);
723 const float peak_levels[2] = {
724 find_peak(left.data(), left.size()) * volume,
725 find_peak(right.data(), right.size()) * volume
727 for (unsigned channel = 0; channel < 2; ++channel) {
728 // Compute the current value, including hold and falloff.
729 // The constants are borrowed from zita-mu1 by Fons Adriaensen.
730 static constexpr float hold_sec = 0.5f;
731 static constexpr float falloff_db_sec = 15.0f; // dB/sec falloff after hold.
733 PeakHistory &history = peak_history[bus_index][channel];
734 history.historic_peak = max(history.historic_peak, peak_levels[channel]);
735 if (history.age_seconds < hold_sec) {
736 current_peak = history.last_peak;
738 current_peak = history.last_peak * from_db(-falloff_db_sec * (history.age_seconds - hold_sec));
741 // See if we have a new peak to replace the old (possibly falling) one.
742 if (peak_levels[channel] > current_peak) {
743 history.last_peak = peak_levels[channel];
744 history.age_seconds = 0.0f; // Not 100% correct, but more than good enough given our frame sizes.
745 current_peak = peak_levels[channel];
747 history.age_seconds += float(left.size()) / OUTPUT_FREQUENCY;
749 history.current_level = peak_levels[channel];
750 history.current_peak = current_peak;
754 void AudioMixer::update_meters(const vector<float> &samples)
756 // Upsample 4x to find interpolated peak.
757 peak_resampler.inp_data = const_cast<float *>(samples.data());
758 peak_resampler.inp_count = samples.size() / 2;
760 vector<float> interpolated_samples;
761 interpolated_samples.resize(samples.size());
763 lock_guard<mutex> lock(audio_measure_mutex);
765 while (peak_resampler.inp_count > 0) { // About four iterations.
766 peak_resampler.out_data = &interpolated_samples[0];
767 peak_resampler.out_count = interpolated_samples.size() / 2;
768 peak_resampler.process();
769 size_t out_stereo_samples = interpolated_samples.size() / 2 - peak_resampler.out_count;
770 peak = max<float>(peak, find_peak(interpolated_samples.data(), out_stereo_samples * 2));
771 peak_resampler.out_data = nullptr;
775 // Find R128 levels and L/R correlation.
776 vector<float> left, right;
777 deinterleave_samples(samples, &left, &right);
778 float *ptrs[] = { left.data(), right.data() };
780 lock_guard<mutex> lock(audio_measure_mutex);
781 r128.process(left.size(), ptrs);
782 correlation.process_samples(samples);
785 send_audio_level_callback();
788 void AudioMixer::reset_meters()
790 lock_guard<mutex> lock(audio_measure_mutex);
791 peak_resampler.reset();
798 void AudioMixer::send_audio_level_callback()
800 if (audio_level_callback == nullptr) {
804 lock_guard<mutex> lock(audio_measure_mutex);
805 double loudness_s = r128.loudness_S();
806 double loudness_i = r128.integrated();
807 double loudness_range_low = r128.range_min();
808 double loudness_range_high = r128.range_max();
810 vector<BusLevel> bus_levels;
811 bus_levels.resize(input_mapping.buses.size());
813 lock_guard<mutex> lock(compressor_mutex);
814 for (unsigned bus_index = 0; bus_index < bus_levels.size(); ++bus_index) {
815 bus_levels[bus_index].current_level_dbfs[0] = to_db(peak_history[bus_index][0].current_level);
816 bus_levels[bus_index].current_level_dbfs[1] = to_db(peak_history[bus_index][1].current_level);
817 bus_levels[bus_index].peak_level_dbfs[0] = to_db(peak_history[bus_index][0].current_peak);
818 bus_levels[bus_index].peak_level_dbfs[1] = to_db(peak_history[bus_index][1].current_peak);
819 bus_levels[bus_index].historic_peak_dbfs = to_db(
820 max(peak_history[bus_index][0].historic_peak,
821 peak_history[bus_index][1].historic_peak));
822 bus_levels[bus_index].gain_staging_db = gain_staging_db[bus_index];
823 if (compressor_enabled[bus_index]) {
824 bus_levels[bus_index].compressor_attenuation_db = -to_db(compressor[bus_index]->get_attenuation());
826 bus_levels[bus_index].compressor_attenuation_db = 0.0;
831 audio_level_callback(loudness_s, to_db(peak), bus_levels,
832 loudness_i, loudness_range_low, loudness_range_high,
833 to_db(final_makeup_gain),
834 correlation.get_correlation());
837 map<DeviceSpec, DeviceInfo> AudioMixer::get_devices()
839 lock_guard<timed_mutex> lock(audio_mutex);
841 map<DeviceSpec, DeviceInfo> devices;
842 for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
843 const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index };
844 const AudioDevice *device = &video_cards[card_index];
846 info.display_name = device->display_name;
847 info.num_channels = 8;
848 devices.insert(make_pair(spec, info));
850 vector<ALSAPool::Device> available_alsa_devices = alsa_pool.get_devices();
851 for (unsigned card_index = 0; card_index < available_alsa_devices.size(); ++card_index) {
852 const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index };
853 const ALSAPool::Device &device = available_alsa_devices[card_index];
855 info.display_name = device.display_name();
856 info.num_channels = device.num_channels;
857 info.alsa_name = device.name;
858 info.alsa_info = device.info;
859 info.alsa_address = device.address;
860 devices.insert(make_pair(spec, info));
865 void AudioMixer::set_display_name(DeviceSpec device_spec, const string &name)
867 AudioDevice *device = find_audio_device(device_spec);
869 lock_guard<timed_mutex> lock(audio_mutex);
870 device->display_name = name;
873 void AudioMixer::serialize_device(DeviceSpec device_spec, DeviceSpecProto *device_spec_proto)
875 lock_guard<timed_mutex> lock(audio_mutex);
876 switch (device_spec.type) {
877 case InputSourceType::SILENCE:
878 device_spec_proto->set_type(DeviceSpecProto::SILENCE);
880 case InputSourceType::CAPTURE_CARD:
881 device_spec_proto->set_type(DeviceSpecProto::CAPTURE_CARD);
882 device_spec_proto->set_index(device_spec.index);
883 device_spec_proto->set_display_name(video_cards[device_spec.index].display_name);
885 case InputSourceType::ALSA_INPUT:
886 alsa_pool.serialize_device(device_spec.index, device_spec_proto);
891 void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
893 lock_guard<timed_mutex> lock(audio_mutex);
895 map<DeviceSpec, set<unsigned>> interesting_channels;
896 for (const InputMapping::Bus &bus : new_input_mapping.buses) {
897 if (bus.device.type == InputSourceType::CAPTURE_CARD ||
898 bus.device.type == InputSourceType::ALSA_INPUT) {
899 for (unsigned channel = 0; channel < 2; ++channel) {
900 if (bus.source_channel[channel] != -1) {
901 interesting_channels[bus.device].insert(bus.source_channel[channel]);
907 // Reset resamplers for all cards that don't have the exact same state as before.
908 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
909 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
910 AudioDevice *device = find_audio_device(device_spec);
911 if (device->interesting_channels != interesting_channels[device_spec]) {
912 device->interesting_channels = interesting_channels[device_spec];
913 reset_resampler_mutex_held(device_spec);
916 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
917 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
918 AudioDevice *device = find_audio_device(device_spec);
919 if (interesting_channels[device_spec].empty()) {
920 alsa_pool.release_device(card_index);
922 alsa_pool.hold_device(card_index);
924 if (device->interesting_channels != interesting_channels[device_spec]) {
925 device->interesting_channels = interesting_channels[device_spec];
926 alsa_pool.reset_device(device_spec.index);
927 reset_resampler_mutex_held(device_spec);
931 input_mapping = new_input_mapping;
934 InputMapping AudioMixer::get_input_mapping() const
936 lock_guard<timed_mutex> lock(audio_mutex);
937 return input_mapping;
940 void AudioMixer::reset_peak(unsigned bus_index)
942 lock_guard<timed_mutex> lock(audio_mutex);
943 for (unsigned channel = 0; channel < 2; ++channel) {
944 PeakHistory &history = peak_history[bus_index][channel];
945 history.current_level = 0.0f;
946 history.historic_peak = 0.0f;
947 history.current_peak = 0.0f;
948 history.last_peak = 0.0f;
949 history.age_seconds = 0.0f;
953 AudioMixer *global_audio_mixer = nullptr;