1 #include "audio_mixer.h"
4 #include <bmusb/bmusb.h>
26 using namespace bmusb;
28 using namespace std::placeholders;
32 // TODO: If these prove to be a bottleneck, they can be SSSE3-optimized
33 // (usually including multiple channels at a time).
35 void convert_fixed16_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
36 const uint8_t *src, size_t in_channel, size_t in_num_channels,
39 assert(in_channel < in_num_channels);
40 assert(out_channel < out_num_channels);
41 src += in_channel * 2;
44 for (size_t i = 0; i < num_samples; ++i) {
45 int16_t s = le16toh(*(int16_t *)src);
46 *dst = s * (1.0f / 32768.0f);
48 src += 2 * in_num_channels;
49 dst += out_num_channels;
53 void convert_fixed24_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
54 const uint8_t *src, size_t in_channel, size_t in_num_channels,
57 assert(in_channel < in_num_channels);
58 assert(out_channel < out_num_channels);
59 src += in_channel * 3;
62 for (size_t i = 0; i < num_samples; ++i) {
66 uint32_t s = s1 | (s1 << 8) | (s2 << 16) | (s3 << 24);
67 *dst = int(s) * (1.0f / 2147483648.0f);
69 src += 3 * in_num_channels;
70 dst += out_num_channels;
74 void convert_fixed32_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
75 const uint8_t *src, size_t in_channel, size_t in_num_channels,
78 assert(in_channel < in_num_channels);
79 assert(out_channel < out_num_channels);
80 src += in_channel * 4;
83 for (size_t i = 0; i < num_samples; ++i) {
84 int32_t s = le32toh(*(int32_t *)src);
85 *dst = s * (1.0f / 2147483648.0f);
87 src += 4 * in_num_channels;
88 dst += out_num_channels;
92 float find_peak_plain(const float *samples, size_t num_samples) __attribute__((unused));
94 float find_peak_plain(const float *samples, size_t num_samples)
96 float m = fabs(samples[0]);
97 for (size_t i = 1; i < num_samples; ++i) {
98 m = max(m, fabs(samples[i]));
104 static inline float horizontal_max(__m128 m)
106 __m128 tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 0, 3, 2));
107 m = _mm_max_ps(m, tmp);
108 tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 3, 0, 1));
109 m = _mm_max_ps(m, tmp);
110 return _mm_cvtss_f32(m);
113 float find_peak(const float *samples, size_t num_samples)
115 const __m128 abs_mask = _mm_castsi128_ps(_mm_set1_epi32(0x7fffffffu));
116 __m128 m = _mm_setzero_ps();
117 for (size_t i = 0; i < (num_samples & ~3); i += 4) {
118 __m128 x = _mm_loadu_ps(samples + i);
119 x = _mm_and_ps(x, abs_mask);
120 m = _mm_max_ps(m, x);
122 float result = horizontal_max(m);
124 for (size_t i = (num_samples & ~3); i < num_samples; ++i) {
125 result = max(result, fabs(samples[i]));
129 // Self-test. We should be bit-exact the same.
130 float reference_result = find_peak_plain(samples, num_samples);
131 if (result != reference_result) {
132 fprintf(stderr, "Error: Peak is %f [%f %f %f %f]; should be %f.\n",
134 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(0, 0, 0, 0))),
135 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 1, 1, 1))),
136 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 2, 2, 2))),
137 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(3, 3, 3, 3))),
145 float find_peak(const float *samples, size_t num_samples)
147 return find_peak_plain(samples, num_samples);
151 void deinterleave_samples(const vector<float> &in, vector<float> *out_l, vector<float> *out_r)
153 size_t num_samples = in.size() / 2;
154 out_l->resize(num_samples);
155 out_r->resize(num_samples);
157 const float *inptr = in.data();
158 float *lptr = &(*out_l)[0];
159 float *rptr = &(*out_r)[0];
160 for (size_t i = 0; i < num_samples; ++i) {
168 AudioMixer::AudioMixer(unsigned num_cards)
169 : num_cards(num_cards),
170 limiter(OUTPUT_FREQUENCY),
171 correlation(OUTPUT_FREQUENCY)
173 global_audio_mixer = this;
175 for (unsigned bus_index = 0; bus_index < MAX_BUSES; ++bus_index) {
176 locut[bus_index].init(FILTER_HPF, 2);
177 eq[bus_index][EQ_BAND_BASS].init(FILTER_LOW_SHELF, 1);
178 // Note: EQ_BAND_MID isn't used (see comments in apply_eq()).
179 eq[bus_index][EQ_BAND_TREBLE].init(FILTER_HIGH_SHELF, 1);
180 compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
181 level_compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
183 set_bus_settings(bus_index, get_default_bus_settings());
185 set_limiter_enabled(global_flags.limiter_enabled);
186 set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
189 if (!global_flags.input_mapping_filename.empty()) {
190 current_mapping_mode = MappingMode::MULTICHANNEL;
191 InputMapping new_input_mapping;
192 if (!load_input_mapping_from_file(get_devices(),
193 global_flags.input_mapping_filename,
194 &new_input_mapping)) {
195 fprintf(stderr, "Failed to load input mapping from '%s', exiting.\n",
196 global_flags.input_mapping_filename.c_str());
199 set_input_mapping(new_input_mapping);
201 set_simple_input(/*card_index=*/0);
202 if (global_flags.multichannel_mapping_mode) {
203 current_mapping_mode = MappingMode::MULTICHANNEL;
207 r128.init(2, OUTPUT_FREQUENCY);
210 // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
211 // and there's a limit to how important the peak meter is.
212 peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
215 void AudioMixer::reset_resampler(DeviceSpec device_spec)
217 lock_guard<timed_mutex> lock(audio_mutex);
218 reset_resampler_mutex_held(device_spec);
221 void AudioMixer::reset_resampler_mutex_held(DeviceSpec device_spec)
223 AudioDevice *device = find_audio_device(device_spec);
225 if (device->interesting_channels.empty()) {
226 device->resampling_queue.reset();
228 // TODO: ResamplingQueue should probably take the full device spec.
229 // (It's only used for console output, though.)
230 device->resampling_queue.reset(new ResamplingQueue(device_spec.index, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size()));
232 device->next_local_pts = 0;
235 bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length)
237 AudioDevice *device = find_audio_device(device_spec);
239 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
240 if (!lock.try_lock_for(chrono::milliseconds(10))) {
243 if (device->resampling_queue == nullptr) {
244 // No buses use this device; throw it away.
248 unsigned num_channels = device->interesting_channels.size();
249 assert(num_channels > 0);
251 // Convert the audio to fp32.
252 unique_ptr<float[]> audio(new float[num_samples * num_channels]);
253 unsigned channel_index = 0;
254 for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) {
255 switch (audio_format.bits_per_sample) {
257 assert(num_samples == 0);
260 convert_fixed16_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
263 convert_fixed24_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
266 convert_fixed32_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
269 fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
275 int64_t local_pts = device->next_local_pts;
276 device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.get(), num_samples);
277 device->next_local_pts = local_pts + frame_length;
281 bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length)
283 AudioDevice *device = find_audio_device(device_spec);
285 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
286 if (!lock.try_lock_for(chrono::milliseconds(10))) {
289 if (device->resampling_queue == nullptr) {
290 // No buses use this device; throw it away.
294 unsigned num_channels = device->interesting_channels.size();
295 assert(num_channels > 0);
297 vector<float> silence(samples_per_frame * num_channels, 0.0f);
298 for (unsigned i = 0; i < num_frames; ++i) {
299 device->resampling_queue->add_input_samples(device->next_local_pts / double(TIMEBASE), silence.data(), samples_per_frame);
300 // Note that if the format changed in the meantime, we have
301 // no way of detecting that; we just have to assume the frame length
302 // is always the same.
303 device->next_local_pts += frame_length;
308 bool AudioMixer::silence_card(DeviceSpec device_spec, bool silence)
310 AudioDevice *device = find_audio_device(device_spec);
312 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
313 if (!lock.try_lock_for(chrono::milliseconds(10))) {
317 if (device->silenced && !silence) {
318 reset_resampler_mutex_held(device_spec);
320 device->silenced = silence;
324 AudioMixer::BusSettings AudioMixer::get_default_bus_settings()
326 BusSettings settings;
327 settings.fader_volume_db = 0.0f;
328 settings.muted = false;
329 settings.locut_enabled = global_flags.locut_enabled;
330 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
331 settings.eq_level_db[band_index] = 0.0f;
333 settings.gain_staging_db = global_flags.initial_gain_staging_db;
334 settings.level_compressor_enabled = global_flags.gain_staging_auto;
335 settings.compressor_threshold_dbfs = ref_level_dbfs - 12.0f; // -12 dB.
336 settings.compressor_enabled = global_flags.compressor_enabled;
340 AudioMixer::BusSettings AudioMixer::get_bus_settings(unsigned bus_index) const
342 lock_guard<timed_mutex> lock(audio_mutex);
343 BusSettings settings;
344 settings.fader_volume_db = fader_volume_db[bus_index];
345 settings.muted = mute[bus_index];
346 settings.locut_enabled = locut_enabled[bus_index];
347 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
348 settings.eq_level_db[band_index] = eq_level_db[bus_index][band_index];
350 settings.gain_staging_db = gain_staging_db[bus_index];
351 settings.level_compressor_enabled = level_compressor_enabled[bus_index];
352 settings.compressor_threshold_dbfs = compressor_threshold_dbfs[bus_index];
353 settings.compressor_enabled = compressor_enabled[bus_index];
357 void AudioMixer::set_bus_settings(unsigned bus_index, const AudioMixer::BusSettings &settings)
359 lock_guard<timed_mutex> lock(audio_mutex);
360 fader_volume_db[bus_index] = settings.fader_volume_db;
361 mute[bus_index] = settings.muted;
362 locut_enabled[bus_index] = settings.locut_enabled;
363 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
364 eq_level_db[bus_index][band_index] = settings.eq_level_db[band_index];
366 gain_staging_db[bus_index] = settings.gain_staging_db;
367 last_gain_staging_db[bus_index] = gain_staging_db[bus_index];
368 level_compressor_enabled[bus_index] = settings.level_compressor_enabled;
369 compressor_threshold_dbfs[bus_index] = settings.compressor_threshold_dbfs;
370 compressor_enabled[bus_index] = settings.compressor_enabled;
373 AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
375 switch (device.type) {
376 case InputSourceType::CAPTURE_CARD:
377 return &video_cards[device.index];
378 case InputSourceType::ALSA_INPUT:
379 return &alsa_inputs[device.index];
380 case InputSourceType::SILENCE:
387 // Get a pointer to the given channel from the given device.
388 // The channel must be picked out earlier and resampled.
389 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)
391 static float zero = 0.0f;
392 if (source_channel == -1 || device_spec.type == InputSourceType::SILENCE) {
397 AudioDevice *device = find_audio_device(device_spec);
398 assert(device->interesting_channels.count(source_channel) != 0);
399 unsigned channel_index = 0;
400 for (int channel : device->interesting_channels) {
401 if (channel == source_channel) break;
404 assert(channel_index < device->interesting_channels.size());
405 const auto it = samples_card.find(device_spec);
406 assert(it != samples_card.end());
407 *srcptr = &(it->second)[channel_index];
408 *stride = device->interesting_channels.size();
411 // TODO: Can be SSSE3-optimized if need be.
412 void AudioMixer::fill_audio_bus(const map<DeviceSpec, vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output)
414 if (bus.device.type == InputSourceType::SILENCE) {
415 memset(output, 0, num_samples * 2 * sizeof(*output));
417 assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
418 bus.device.type == InputSourceType::ALSA_INPUT);
419 const float *lsrc, *rsrc;
420 unsigned lstride, rstride;
421 float *dptr = output;
422 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[0], &lsrc, &lstride);
423 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[1], &rsrc, &rstride);
424 for (unsigned i = 0; i < num_samples; ++i) {
433 vector<DeviceSpec> AudioMixer::get_active_devices() const
435 vector<DeviceSpec> ret;
436 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
437 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
438 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
439 ret.push_back(device_spec);
442 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
443 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
444 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
445 ret.push_back(device_spec);
453 void apply_gain(float db, float last_db, vector<float> *samples)
455 if (fabs(db - last_db) < 1e-3) {
456 // Constant over this frame.
457 const float gain = from_db(db);
458 for (size_t i = 0; i < samples->size(); ++i) {
459 (*samples)[i] *= gain;
462 // We need to do a fade.
463 unsigned num_samples = samples->size() / 2;
464 float gain = from_db(last_db);
465 const float gain_inc = pow(from_db(db - last_db), 1.0 / num_samples);
466 for (size_t i = 0; i < num_samples; ++i) {
467 (*samples)[i * 2 + 0] *= gain;
468 (*samples)[i * 2 + 1] *= gain;
476 vector<float> AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
478 map<DeviceSpec, vector<float>> samples_card;
479 vector<float> samples_bus;
481 lock_guard<timed_mutex> lock(audio_mutex);
483 // Pick out all the interesting channels from all the cards.
484 for (const DeviceSpec &device_spec : get_active_devices()) {
485 AudioDevice *device = find_audio_device(device_spec);
486 samples_card[device_spec].resize(num_samples * device->interesting_channels.size());
487 if (device->silenced) {
488 memset(&samples_card[device_spec][0], 0, samples_card[device_spec].size() * sizeof(float));
490 device->resampling_queue->get_output_samples(
492 &samples_card[device_spec][0],
494 rate_adjustment_policy);
498 vector<float> samples_out, left, right;
499 samples_out.resize(num_samples * 2);
500 samples_bus.resize(num_samples * 2);
501 for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
502 fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, &samples_bus[0]);
503 apply_eq(bus_index, &samples_bus);
506 lock_guard<mutex> lock(compressor_mutex);
508 // Apply a level compressor to get the general level right.
509 // Basically, if it's over about -40 dBFS, we squeeze it down to that level
510 // (or more precisely, near it, since we don't use infinite ratio),
511 // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
512 // entirely arbitrary, but from practical tests with speech, it seems to
513 // put ut around -23 LUFS, so it's a reasonable starting point for later use.
514 if (level_compressor_enabled[bus_index]) {
515 float threshold = 0.01f; // -40 dBFS.
517 float attack_time = 0.5f;
518 float release_time = 20.0f;
519 float makeup_gain = from_db(ref_level_dbfs - (-40.0f)); // +26 dB.
520 level_compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
521 gain_staging_db[bus_index] = to_db(level_compressor[bus_index]->get_attenuation() * makeup_gain);
523 // Just apply the gain we already had.
524 float db = gain_staging_db[bus_index];
525 float last_db = last_gain_staging_db[bus_index];
526 apply_gain(db, last_db, &samples_bus);
528 last_gain_staging_db[bus_index] = gain_staging_db[bus_index];
531 printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
532 level_compressor.get_level(), to_db(level_compressor.get_level()),
533 level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()),
534 to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
537 // The real compressor.
538 if (compressor_enabled[bus_index]) {
539 float threshold = from_db(compressor_threshold_dbfs[bus_index]);
541 float attack_time = 0.005f;
542 float release_time = 0.040f;
543 float makeup_gain = 2.0f; // +6 dB.
544 compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
545 // compressor_att = compressor.get_attenuation();
549 add_bus_to_master(bus_index, samples_bus, &samples_out);
550 deinterleave_samples(samples_bus, &left, &right);
551 measure_bus_levels(bus_index, left, right);
555 lock_guard<mutex> lock(compressor_mutex);
557 // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
558 // Note that since ratio is not infinite, we could go slightly higher than this.
559 if (limiter_enabled) {
560 float threshold = from_db(limiter_threshold_dbfs);
562 float attack_time = 0.0f; // Instant.
563 float release_time = 0.020f;
564 float makeup_gain = 1.0f; // 0 dB.
565 limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
566 // limiter_att = limiter.get_attenuation();
569 // printf("limiter=%+5.1f compressor=%+5.1f\n", to_db(limiter_att), to_db(compressor_att));
572 // At this point, we are most likely close to +0 LU (at least if the
573 // faders sum to 0 dB and the compressors are on), but all of our
574 // measurements have been on raw sample values, not R128 values.
575 // So we have a final makeup gain to get us to +0 LU; the gain
576 // adjustments required should be relatively small, and also, the
577 // offset shouldn't change much (only if the type of audio changes
578 // significantly). Thus, we shoot for updating this value basically
579 // “whenever we process buffers”, since the R128 calculation isn't exactly
580 // something we get out per-sample.
582 // Note that there's a feedback loop here, so we choose a very slow filter
583 // (half-time of 30 seconds).
584 double target_loudness_factor, alpha;
585 double loudness_lu = r128.loudness_M() - ref_level_lufs;
586 double current_makeup_lu = to_db(final_makeup_gain);
587 target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
589 // If we're outside +/- 5 LU uncorrected, we don't count it as
590 // a normal signal (probably silence) and don't change the
591 // correction factor; just apply what we already have.
592 if (fabs(loudness_lu - current_makeup_lu) >= 5.0 || !final_makeup_gain_auto) {
595 // Formula adapted from
596 // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
597 const double half_time_s = 30.0;
598 const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
599 alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
603 lock_guard<mutex> lock(compressor_mutex);
604 double m = final_makeup_gain;
605 for (size_t i = 0; i < samples_out.size(); i += 2) {
606 samples_out[i + 0] *= m;
607 samples_out[i + 1] *= m;
608 m += (target_loudness_factor - m) * alpha;
610 final_makeup_gain = m;
613 update_meters(samples_out);
620 void apply_filter_fade(StereoFilter *filter, float *data, unsigned num_samples, float cutoff_hz, float db, float last_db)
622 // A granularity of 32 samples is an okay tradeoff between speed and
623 // smoothness; recalculating the filters is pretty expensive, so it's
624 // good that we don't do this all the time.
625 static constexpr unsigned filter_granularity_samples = 32;
627 const float cutoff_linear = cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY;
628 if (fabs(db - last_db) < 1e-3) {
629 // Constant over this frame.
630 if (fabs(db) > 0.01f) {
631 filter->render(data, num_samples, cutoff_linear, 0.5f, db / 40.0f);
634 // We need to do a fade. (Rounding up avoids division by zero.)
635 unsigned num_blocks = (num_samples + filter_granularity_samples - 1) / filter_granularity_samples;
636 const float inc_db_norm = (db - last_db) / 40.0f / num_blocks;
637 float db_norm = db / 40.0f;
638 for (size_t i = 0; i < num_samples; i += filter_granularity_samples) {
639 size_t samples_this_block = std::min<size_t>(num_samples - i, filter_granularity_samples);
640 filter->render(data + i * 2, samples_this_block, cutoff_linear, 0.5f, db_norm);
641 db_norm += inc_db_norm;
648 void AudioMixer::apply_eq(unsigned bus_index, vector<float> *samples_bus)
650 constexpr float bass_freq_hz = 200.0f;
651 constexpr float treble_freq_hz = 4700.0f;
653 // Cut away everything under 120 Hz (or whatever the cutoff is);
654 // we don't need it for voice, and it will reduce headroom
655 // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
656 // should be dampened.)
657 if (locut_enabled[bus_index]) {
658 locut[bus_index].render(samples_bus->data(), samples_bus->size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
661 // Apply the rest of the EQ. Since we only have a simple three-band EQ,
662 // we can implement it with two shelf filters. We use a simple gain to
663 // set the mid-level filter, and then offset the low and high bands
664 // from that if we need to. (We could perhaps have folded the gain into
665 // the next part, but it's so cheap that the trouble isn't worth it.)
667 // If any part of the EQ has changed appreciably since last frame,
668 // we fade smoothly during the course of this frame.
669 const float bass_db = eq_level_db[bus_index][EQ_BAND_BASS];
670 const float mid_db = eq_level_db[bus_index][EQ_BAND_MID];
671 const float treble_db = eq_level_db[bus_index][EQ_BAND_TREBLE];
673 const float last_bass_db = last_eq_level_db[bus_index][EQ_BAND_BASS];
674 const float last_mid_db = last_eq_level_db[bus_index][EQ_BAND_MID];
675 const float last_treble_db = last_eq_level_db[bus_index][EQ_BAND_TREBLE];
677 assert(samples_bus->size() % 2 == 0);
678 const unsigned num_samples = samples_bus->size() / 2;
680 apply_gain(mid_db, last_mid_db, samples_bus);
682 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);
683 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);
685 last_eq_level_db[bus_index][EQ_BAND_BASS] = bass_db;
686 last_eq_level_db[bus_index][EQ_BAND_MID] = mid_db;
687 last_eq_level_db[bus_index][EQ_BAND_TREBLE] = treble_db;
690 void AudioMixer::add_bus_to_master(unsigned bus_index, const vector<float> &samples_bus, vector<float> *samples_out)
692 assert(samples_bus.size() == samples_out->size());
693 assert(samples_bus.size() % 2 == 0);
694 unsigned num_samples = samples_bus.size() / 2;
695 const float new_volume_db = mute[bus_index] ? -90.0f : fader_volume_db[bus_index].load();
696 if (fabs(new_volume_db - last_fader_volume_db[bus_index]) > 1e-3) {
697 // The volume has changed; do a fade over the course of this frame.
698 // (We might have some numerical issues here, but it seems to sound OK.)
699 // For the purpose of fading here, the silence floor is set to -90 dB
700 // (the fader only goes to -84).
701 float old_volume = from_db(max<float>(last_fader_volume_db[bus_index], -90.0f));
702 float volume = from_db(max<float>(new_volume_db, -90.0f));
704 float volume_inc = pow(volume / old_volume, 1.0 / num_samples);
706 if (bus_index == 0) {
707 for (unsigned i = 0; i < num_samples; ++i) {
708 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
709 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
710 volume *= volume_inc;
713 for (unsigned i = 0; i < num_samples; ++i) {
714 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
715 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
716 volume *= volume_inc;
719 } else if (new_volume_db > -90.0f) {
720 float volume = from_db(new_volume_db);
721 if (bus_index == 0) {
722 for (unsigned i = 0; i < num_samples; ++i) {
723 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
724 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
727 for (unsigned i = 0; i < num_samples; ++i) {
728 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
729 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
734 last_fader_volume_db[bus_index] = new_volume_db;
737 void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right)
739 assert(left.size() == right.size());
740 const float volume = mute[bus_index] ? 0.0f : from_db(fader_volume_db[bus_index]);
741 const float peak_levels[2] = {
742 find_peak(left.data(), left.size()) * volume,
743 find_peak(right.data(), right.size()) * volume
745 for (unsigned channel = 0; channel < 2; ++channel) {
746 // Compute the current value, including hold and falloff.
747 // The constants are borrowed from zita-mu1 by Fons Adriaensen.
748 static constexpr float hold_sec = 0.5f;
749 static constexpr float falloff_db_sec = 15.0f; // dB/sec falloff after hold.
751 PeakHistory &history = peak_history[bus_index][channel];
752 history.historic_peak = max(history.historic_peak, peak_levels[channel]);
753 if (history.age_seconds < hold_sec) {
754 current_peak = history.last_peak;
756 current_peak = history.last_peak * from_db(-falloff_db_sec * (history.age_seconds - hold_sec));
759 // See if we have a new peak to replace the old (possibly falling) one.
760 if (peak_levels[channel] > current_peak) {
761 history.last_peak = peak_levels[channel];
762 history.age_seconds = 0.0f; // Not 100% correct, but more than good enough given our frame sizes.
763 current_peak = peak_levels[channel];
765 history.age_seconds += float(left.size()) / OUTPUT_FREQUENCY;
767 history.current_level = peak_levels[channel];
768 history.current_peak = current_peak;
772 void AudioMixer::update_meters(const vector<float> &samples)
774 // Upsample 4x to find interpolated peak.
775 peak_resampler.inp_data = const_cast<float *>(samples.data());
776 peak_resampler.inp_count = samples.size() / 2;
778 vector<float> interpolated_samples;
779 interpolated_samples.resize(samples.size());
781 lock_guard<mutex> lock(audio_measure_mutex);
783 while (peak_resampler.inp_count > 0) { // About four iterations.
784 peak_resampler.out_data = &interpolated_samples[0];
785 peak_resampler.out_count = interpolated_samples.size() / 2;
786 peak_resampler.process();
787 size_t out_stereo_samples = interpolated_samples.size() / 2 - peak_resampler.out_count;
788 peak = max<float>(peak, find_peak(interpolated_samples.data(), out_stereo_samples * 2));
789 peak_resampler.out_data = nullptr;
793 // Find R128 levels and L/R correlation.
794 vector<float> left, right;
795 deinterleave_samples(samples, &left, &right);
796 float *ptrs[] = { left.data(), right.data() };
798 lock_guard<mutex> lock(audio_measure_mutex);
799 r128.process(left.size(), ptrs);
800 correlation.process_samples(samples);
803 send_audio_level_callback();
806 void AudioMixer::reset_meters()
808 lock_guard<mutex> lock(audio_measure_mutex);
809 peak_resampler.reset();
816 void AudioMixer::send_audio_level_callback()
818 if (audio_level_callback == nullptr) {
822 lock_guard<mutex> lock(audio_measure_mutex);
823 double loudness_s = r128.loudness_S();
824 double loudness_i = r128.integrated();
825 double loudness_range_low = r128.range_min();
826 double loudness_range_high = r128.range_max();
828 vector<BusLevel> bus_levels;
829 bus_levels.resize(input_mapping.buses.size());
831 lock_guard<mutex> lock(compressor_mutex);
832 for (unsigned bus_index = 0; bus_index < bus_levels.size(); ++bus_index) {
833 bus_levels[bus_index].current_level_dbfs[0] = to_db(peak_history[bus_index][0].current_level);
834 bus_levels[bus_index].current_level_dbfs[1] = to_db(peak_history[bus_index][1].current_level);
835 bus_levels[bus_index].peak_level_dbfs[0] = to_db(peak_history[bus_index][0].current_peak);
836 bus_levels[bus_index].peak_level_dbfs[1] = to_db(peak_history[bus_index][1].current_peak);
837 bus_levels[bus_index].historic_peak_dbfs = to_db(
838 max(peak_history[bus_index][0].historic_peak,
839 peak_history[bus_index][1].historic_peak));
840 bus_levels[bus_index].gain_staging_db = gain_staging_db[bus_index];
841 if (compressor_enabled[bus_index]) {
842 bus_levels[bus_index].compressor_attenuation_db = -to_db(compressor[bus_index]->get_attenuation());
844 bus_levels[bus_index].compressor_attenuation_db = 0.0;
849 audio_level_callback(loudness_s, to_db(peak), bus_levels,
850 loudness_i, loudness_range_low, loudness_range_high,
851 to_db(final_makeup_gain),
852 correlation.get_correlation());
855 map<DeviceSpec, DeviceInfo> AudioMixer::get_devices()
857 lock_guard<timed_mutex> lock(audio_mutex);
859 map<DeviceSpec, DeviceInfo> devices;
860 for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
861 const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index };
862 const AudioDevice *device = &video_cards[card_index];
864 info.display_name = device->display_name;
865 info.num_channels = 8;
866 devices.insert(make_pair(spec, info));
868 vector<ALSAPool::Device> available_alsa_devices = alsa_pool.get_devices();
869 for (unsigned card_index = 0; card_index < available_alsa_devices.size(); ++card_index) {
870 const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index };
871 const ALSAPool::Device &device = available_alsa_devices[card_index];
873 info.display_name = device.display_name();
874 info.num_channels = device.num_channels;
875 info.alsa_name = device.name;
876 info.alsa_info = device.info;
877 info.alsa_address = device.address;
878 devices.insert(make_pair(spec, info));
883 void AudioMixer::set_display_name(DeviceSpec device_spec, const string &name)
885 AudioDevice *device = find_audio_device(device_spec);
887 lock_guard<timed_mutex> lock(audio_mutex);
888 device->display_name = name;
891 void AudioMixer::serialize_device(DeviceSpec device_spec, DeviceSpecProto *device_spec_proto)
893 lock_guard<timed_mutex> lock(audio_mutex);
894 switch (device_spec.type) {
895 case InputSourceType::SILENCE:
896 device_spec_proto->set_type(DeviceSpecProto::SILENCE);
898 case InputSourceType::CAPTURE_CARD:
899 device_spec_proto->set_type(DeviceSpecProto::CAPTURE_CARD);
900 device_spec_proto->set_index(device_spec.index);
901 device_spec_proto->set_display_name(video_cards[device_spec.index].display_name);
903 case InputSourceType::ALSA_INPUT:
904 alsa_pool.serialize_device(device_spec.index, device_spec_proto);
909 void AudioMixer::set_simple_input(unsigned card_index)
911 InputMapping new_input_mapping;
912 InputMapping::Bus input;
914 input.device.type = InputSourceType::CAPTURE_CARD;
915 input.device.index = card_index;
916 input.source_channel[0] = 0;
917 input.source_channel[1] = 1;
919 new_input_mapping.buses.push_back(input);
921 lock_guard<timed_mutex> lock(audio_mutex);
922 current_mapping_mode = MappingMode::SIMPLE;
923 set_input_mapping_lock_held(new_input_mapping);
924 fader_volume_db[0] = 0.0f;
927 unsigned AudioMixer::get_simple_input() const
929 lock_guard<timed_mutex> lock(audio_mutex);
930 if (input_mapping.buses.size() == 1 &&
931 input_mapping.buses[0].device.type == InputSourceType::CAPTURE_CARD &&
932 input_mapping.buses[0].source_channel[0] == 0 &&
933 input_mapping.buses[0].source_channel[1] == 1) {
934 return input_mapping.buses[0].device.index;
936 return numeric_limits<unsigned>::max();
940 void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
942 lock_guard<timed_mutex> lock(audio_mutex);
943 set_input_mapping_lock_held(new_input_mapping);
944 current_mapping_mode = MappingMode::MULTICHANNEL;
947 AudioMixer::MappingMode AudioMixer::get_mapping_mode() const
949 lock_guard<timed_mutex> lock(audio_mutex);
950 return current_mapping_mode;
953 void AudioMixer::set_input_mapping_lock_held(const InputMapping &new_input_mapping)
955 map<DeviceSpec, set<unsigned>> interesting_channels;
956 for (const InputMapping::Bus &bus : new_input_mapping.buses) {
957 if (bus.device.type == InputSourceType::CAPTURE_CARD ||
958 bus.device.type == InputSourceType::ALSA_INPUT) {
959 for (unsigned channel = 0; channel < 2; ++channel) {
960 if (bus.source_channel[channel] != -1) {
961 interesting_channels[bus.device].insert(bus.source_channel[channel]);
967 // Reset resamplers for all cards that don't have the exact same state as before.
968 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
969 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
970 AudioDevice *device = find_audio_device(device_spec);
971 if (device->interesting_channels != interesting_channels[device_spec]) {
972 device->interesting_channels = interesting_channels[device_spec];
973 reset_resampler_mutex_held(device_spec);
976 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
977 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
978 AudioDevice *device = find_audio_device(device_spec);
979 if (interesting_channels[device_spec].empty()) {
980 alsa_pool.release_device(card_index);
982 alsa_pool.hold_device(card_index);
984 if (device->interesting_channels != interesting_channels[device_spec]) {
985 device->interesting_channels = interesting_channels[device_spec];
986 alsa_pool.reset_device(device_spec.index);
987 reset_resampler_mutex_held(device_spec);
991 input_mapping = new_input_mapping;
994 InputMapping AudioMixer::get_input_mapping() const
996 lock_guard<timed_mutex> lock(audio_mutex);
997 return input_mapping;
1000 unsigned AudioMixer::num_buses() const
1002 lock_guard<timed_mutex> lock(audio_mutex);
1003 return input_mapping.buses.size();
1006 void AudioMixer::reset_peak(unsigned bus_index)
1008 lock_guard<timed_mutex> lock(audio_mutex);
1009 for (unsigned channel = 0; channel < 2; ++channel) {
1010 PeakHistory &history = peak_history[bus_index][channel];
1011 history.current_level = 0.0f;
1012 history.historic_peak = 0.0f;
1013 history.current_peak = 0.0f;
1014 history.last_peak = 0.0f;
1015 history.age_seconds = 0.0f;
1019 AudioMixer *global_audio_mixer = nullptr;