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 for (unsigned bus_index = 0; bus_index < MAX_BUSES; ++bus_index) {
174 locut[bus_index].init(FILTER_HPF, 2);
175 eq[bus_index][EQ_BAND_BASS].init(FILTER_LOW_SHELF, 1);
176 // Note: EQ_BAND_MID isn't used (see comments in apply_eq()).
177 eq[bus_index][EQ_BAND_TREBLE].init(FILTER_HIGH_SHELF, 1);
178 compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
179 level_compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
181 set_bus_settings(bus_index, get_default_bus_settings());
183 set_limiter_enabled(global_flags.limiter_enabled);
184 set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
186 if (!global_flags.input_mapping_filename.empty()) {
187 current_mapping_mode = MappingMode::MULTICHANNEL;
188 InputMapping new_input_mapping;
189 if (!load_input_mapping_from_file(get_devices(),
190 global_flags.input_mapping_filename,
191 &new_input_mapping)) {
192 fprintf(stderr, "Failed to load input mapping from '%s', exiting.\n",
193 global_flags.input_mapping_filename.c_str());
196 set_input_mapping(new_input_mapping);
198 set_simple_input(/*card_index=*/0);
199 if (global_flags.multichannel_mapping_mode) {
200 current_mapping_mode = MappingMode::MULTICHANNEL;
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);
211 global_audio_mixer = this;
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(
231 device_spec.index, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size(),
232 global_flags.audio_queue_length_ms * 0.001));
234 device->next_local_pts = 0;
237 bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length)
239 AudioDevice *device = find_audio_device(device_spec);
241 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
242 if (!lock.try_lock_for(chrono::milliseconds(10))) {
245 if (device->resampling_queue == nullptr) {
246 // No buses use this device; throw it away.
250 unsigned num_channels = device->interesting_channels.size();
251 assert(num_channels > 0);
253 // Convert the audio to fp32.
254 unique_ptr<float[]> audio(new float[num_samples * num_channels]);
255 unsigned channel_index = 0;
256 for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) {
257 switch (audio_format.bits_per_sample) {
259 assert(num_samples == 0);
262 convert_fixed16_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
265 convert_fixed24_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
268 convert_fixed32_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
271 fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
277 int64_t local_pts = device->next_local_pts;
278 device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.get(), num_samples);
279 device->next_local_pts = local_pts + frame_length;
283 bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length)
285 AudioDevice *device = find_audio_device(device_spec);
287 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
288 if (!lock.try_lock_for(chrono::milliseconds(10))) {
291 if (device->resampling_queue == nullptr) {
292 // No buses use this device; throw it away.
296 unsigned num_channels = device->interesting_channels.size();
297 assert(num_channels > 0);
299 vector<float> silence(samples_per_frame * num_channels, 0.0f);
300 for (unsigned i = 0; i < num_frames; ++i) {
301 device->resampling_queue->add_input_samples(device->next_local_pts / double(TIMEBASE), silence.data(), samples_per_frame);
302 // Note that if the format changed in the meantime, we have
303 // no way of detecting that; we just have to assume the frame length
304 // is always the same.
305 device->next_local_pts += frame_length;
310 bool AudioMixer::silence_card(DeviceSpec device_spec, bool silence)
312 AudioDevice *device = find_audio_device(device_spec);
314 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
315 if (!lock.try_lock_for(chrono::milliseconds(10))) {
319 if (device->silenced && !silence) {
320 reset_resampler_mutex_held(device_spec);
322 device->silenced = silence;
326 AudioMixer::BusSettings AudioMixer::get_default_bus_settings()
328 BusSettings settings;
329 settings.fader_volume_db = 0.0f;
330 settings.muted = false;
331 settings.locut_enabled = global_flags.locut_enabled;
332 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
333 settings.eq_level_db[band_index] = 0.0f;
335 settings.gain_staging_db = global_flags.initial_gain_staging_db;
336 settings.level_compressor_enabled = global_flags.gain_staging_auto;
337 settings.compressor_threshold_dbfs = ref_level_dbfs - 12.0f; // -12 dB.
338 settings.compressor_enabled = global_flags.compressor_enabled;
342 AudioMixer::BusSettings AudioMixer::get_bus_settings(unsigned bus_index) const
344 lock_guard<timed_mutex> lock(audio_mutex);
345 BusSettings settings;
346 settings.fader_volume_db = fader_volume_db[bus_index];
347 settings.muted = mute[bus_index];
348 settings.locut_enabled = locut_enabled[bus_index];
349 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
350 settings.eq_level_db[band_index] = eq_level_db[bus_index][band_index];
352 settings.gain_staging_db = gain_staging_db[bus_index];
353 settings.level_compressor_enabled = level_compressor_enabled[bus_index];
354 settings.compressor_threshold_dbfs = compressor_threshold_dbfs[bus_index];
355 settings.compressor_enabled = compressor_enabled[bus_index];
359 void AudioMixer::set_bus_settings(unsigned bus_index, const AudioMixer::BusSettings &settings)
361 lock_guard<timed_mutex> lock(audio_mutex);
362 fader_volume_db[bus_index] = settings.fader_volume_db;
363 mute[bus_index] = settings.muted;
364 locut_enabled[bus_index] = settings.locut_enabled;
365 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
366 eq_level_db[bus_index][band_index] = settings.eq_level_db[band_index];
368 gain_staging_db[bus_index] = settings.gain_staging_db;
369 last_gain_staging_db[bus_index] = gain_staging_db[bus_index];
370 level_compressor_enabled[bus_index] = settings.level_compressor_enabled;
371 compressor_threshold_dbfs[bus_index] = settings.compressor_threshold_dbfs;
372 compressor_enabled[bus_index] = settings.compressor_enabled;
375 AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
377 switch (device.type) {
378 case InputSourceType::CAPTURE_CARD:
379 return &video_cards[device.index];
380 case InputSourceType::ALSA_INPUT:
381 return &alsa_inputs[device.index];
382 case InputSourceType::SILENCE:
389 // Get a pointer to the given channel from the given device.
390 // The channel must be picked out earlier and resampled.
391 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)
393 static float zero = 0.0f;
394 if (source_channel == -1 || device_spec.type == InputSourceType::SILENCE) {
399 AudioDevice *device = find_audio_device(device_spec);
400 assert(device->interesting_channels.count(source_channel) != 0);
401 unsigned channel_index = 0;
402 for (int channel : device->interesting_channels) {
403 if (channel == source_channel) break;
406 assert(channel_index < device->interesting_channels.size());
407 const auto it = samples_card.find(device_spec);
408 assert(it != samples_card.end());
409 *srcptr = &(it->second)[channel_index];
410 *stride = device->interesting_channels.size();
413 // TODO: Can be SSSE3-optimized if need be.
414 void AudioMixer::fill_audio_bus(const map<DeviceSpec, vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output)
416 if (bus.device.type == InputSourceType::SILENCE) {
417 memset(output, 0, num_samples * 2 * sizeof(*output));
419 assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
420 bus.device.type == InputSourceType::ALSA_INPUT);
421 const float *lsrc, *rsrc;
422 unsigned lstride, rstride;
423 float *dptr = output;
424 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[0], &lsrc, &lstride);
425 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[1], &rsrc, &rstride);
426 for (unsigned i = 0; i < num_samples; ++i) {
435 vector<DeviceSpec> AudioMixer::get_active_devices() const
437 vector<DeviceSpec> ret;
438 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
439 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
440 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
441 ret.push_back(device_spec);
444 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
445 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
446 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
447 ret.push_back(device_spec);
455 void apply_gain(float db, float last_db, vector<float> *samples)
457 if (fabs(db - last_db) < 1e-3) {
458 // Constant over this frame.
459 const float gain = from_db(db);
460 for (size_t i = 0; i < samples->size(); ++i) {
461 (*samples)[i] *= gain;
464 // We need to do a fade.
465 unsigned num_samples = samples->size() / 2;
466 float gain = from_db(last_db);
467 const float gain_inc = pow(from_db(db - last_db), 1.0 / num_samples);
468 for (size_t i = 0; i < num_samples; ++i) {
469 (*samples)[i * 2 + 0] *= gain;
470 (*samples)[i * 2 + 1] *= gain;
478 vector<float> AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
480 map<DeviceSpec, vector<float>> samples_card;
481 vector<float> samples_bus;
483 lock_guard<timed_mutex> lock(audio_mutex);
485 // Pick out all the interesting channels from all the cards.
486 for (const DeviceSpec &device_spec : get_active_devices()) {
487 AudioDevice *device = find_audio_device(device_spec);
488 samples_card[device_spec].resize(num_samples * device->interesting_channels.size());
489 if (device->silenced) {
490 memset(&samples_card[device_spec][0], 0, samples_card[device_spec].size() * sizeof(float));
492 device->resampling_queue->get_output_samples(
494 &samples_card[device_spec][0],
496 rate_adjustment_policy);
500 vector<float> samples_out, left, right;
501 samples_out.resize(num_samples * 2);
502 samples_bus.resize(num_samples * 2);
503 for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
504 fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, &samples_bus[0]);
505 apply_eq(bus_index, &samples_bus);
508 lock_guard<mutex> lock(compressor_mutex);
510 // Apply a level compressor to get the general level right.
511 // Basically, if it's over about -40 dBFS, we squeeze it down to that level
512 // (or more precisely, near it, since we don't use infinite ratio),
513 // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
514 // entirely arbitrary, but from practical tests with speech, it seems to
515 // put ut around -23 LUFS, so it's a reasonable starting point for later use.
516 if (level_compressor_enabled[bus_index]) {
517 float threshold = 0.01f; // -40 dBFS.
519 float attack_time = 0.5f;
520 float release_time = 20.0f;
521 float makeup_gain = from_db(ref_level_dbfs - (-40.0f)); // +26 dB.
522 level_compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
523 gain_staging_db[bus_index] = to_db(level_compressor[bus_index]->get_attenuation() * makeup_gain);
525 // Just apply the gain we already had.
526 float db = gain_staging_db[bus_index];
527 float last_db = last_gain_staging_db[bus_index];
528 apply_gain(db, last_db, &samples_bus);
530 last_gain_staging_db[bus_index] = gain_staging_db[bus_index];
533 printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
534 level_compressor.get_level(), to_db(level_compressor.get_level()),
535 level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()),
536 to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
539 // The real compressor.
540 if (compressor_enabled[bus_index]) {
541 float threshold = from_db(compressor_threshold_dbfs[bus_index]);
543 float attack_time = 0.005f;
544 float release_time = 0.040f;
545 float makeup_gain = 2.0f; // +6 dB.
546 compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
547 // compressor_att = compressor.get_attenuation();
551 add_bus_to_master(bus_index, samples_bus, &samples_out);
552 deinterleave_samples(samples_bus, &left, &right);
553 measure_bus_levels(bus_index, left, right);
557 lock_guard<mutex> lock(compressor_mutex);
559 // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
560 // Note that since ratio is not infinite, we could go slightly higher than this.
561 if (limiter_enabled) {
562 float threshold = from_db(limiter_threshold_dbfs);
564 float attack_time = 0.0f; // Instant.
565 float release_time = 0.020f;
566 float makeup_gain = 1.0f; // 0 dB.
567 limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
568 // limiter_att = limiter.get_attenuation();
571 // printf("limiter=%+5.1f compressor=%+5.1f\n", to_db(limiter_att), to_db(compressor_att));
574 // At this point, we are most likely close to +0 LU (at least if the
575 // faders sum to 0 dB and the compressors are on), but all of our
576 // measurements have been on raw sample values, not R128 values.
577 // So we have a final makeup gain to get us to +0 LU; the gain
578 // adjustments required should be relatively small, and also, the
579 // offset shouldn't change much (only if the type of audio changes
580 // significantly). Thus, we shoot for updating this value basically
581 // “whenever we process buffers”, since the R128 calculation isn't exactly
582 // something we get out per-sample.
584 // Note that there's a feedback loop here, so we choose a very slow filter
585 // (half-time of 30 seconds).
586 double target_loudness_factor, alpha;
587 double loudness_lu = r128.loudness_M() - ref_level_lufs;
588 target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
590 // If we're outside +/- 5 LU (after correction), we don't count it as
591 // a normal signal (probably silence) and don't change the
592 // correction factor; just apply what we already have.
593 if (fabs(loudness_lu) >= 5.0 || !final_makeup_gain_auto) {
596 // Formula adapted from
597 // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
598 const double half_time_s = 30.0;
599 const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
600 alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
604 lock_guard<mutex> lock(compressor_mutex);
605 double m = final_makeup_gain;
606 for (size_t i = 0; i < samples_out.size(); i += 2) {
607 samples_out[i + 0] *= m;
608 samples_out[i + 1] *= m;
609 m += (target_loudness_factor - m) * alpha;
611 final_makeup_gain = m;
614 update_meters(samples_out);
621 void apply_filter_fade(StereoFilter *filter, float *data, unsigned num_samples, float cutoff_hz, float db, float last_db)
623 // A granularity of 32 samples is an okay tradeoff between speed and
624 // smoothness; recalculating the filters is pretty expensive, so it's
625 // good that we don't do this all the time.
626 static constexpr unsigned filter_granularity_samples = 32;
628 const float cutoff_linear = cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY;
629 if (fabs(db - last_db) < 1e-3) {
630 // Constant over this frame.
631 if (fabs(db) > 0.01f) {
632 filter->render(data, num_samples, cutoff_linear, 0.5f, db / 40.0f);
635 // We need to do a fade. (Rounding up avoids division by zero.)
636 unsigned num_blocks = (num_samples + filter_granularity_samples - 1) / filter_granularity_samples;
637 const float inc_db_norm = (db - last_db) / 40.0f / num_blocks;
638 float db_norm = db / 40.0f;
639 for (size_t i = 0; i < num_samples; i += filter_granularity_samples) {
640 size_t samples_this_block = std::min<size_t>(num_samples - i, filter_granularity_samples);
641 filter->render(data + i * 2, samples_this_block, cutoff_linear, 0.5f, db_norm);
642 db_norm += inc_db_norm;
649 void AudioMixer::apply_eq(unsigned bus_index, vector<float> *samples_bus)
651 constexpr float bass_freq_hz = 200.0f;
652 constexpr float treble_freq_hz = 4700.0f;
654 // Cut away everything under 120 Hz (or whatever the cutoff is);
655 // we don't need it for voice, and it will reduce headroom
656 // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
657 // should be dampened.)
658 if (locut_enabled[bus_index]) {
659 locut[bus_index].render(samples_bus->data(), samples_bus->size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
662 // Apply the rest of the EQ. Since we only have a simple three-band EQ,
663 // we can implement it with two shelf filters. We use a simple gain to
664 // set the mid-level filter, and then offset the low and high bands
665 // from that if we need to. (We could perhaps have folded the gain into
666 // the next part, but it's so cheap that the trouble isn't worth it.)
668 // If any part of the EQ has changed appreciably since last frame,
669 // we fade smoothly during the course of this frame.
670 const float bass_db = eq_level_db[bus_index][EQ_BAND_BASS];
671 const float mid_db = eq_level_db[bus_index][EQ_BAND_MID];
672 const float treble_db = eq_level_db[bus_index][EQ_BAND_TREBLE];
674 const float last_bass_db = last_eq_level_db[bus_index][EQ_BAND_BASS];
675 const float last_mid_db = last_eq_level_db[bus_index][EQ_BAND_MID];
676 const float last_treble_db = last_eq_level_db[bus_index][EQ_BAND_TREBLE];
678 assert(samples_bus->size() % 2 == 0);
679 const unsigned num_samples = samples_bus->size() / 2;
681 apply_gain(mid_db, last_mid_db, samples_bus);
683 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);
684 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);
686 last_eq_level_db[bus_index][EQ_BAND_BASS] = bass_db;
687 last_eq_level_db[bus_index][EQ_BAND_MID] = mid_db;
688 last_eq_level_db[bus_index][EQ_BAND_TREBLE] = treble_db;
691 void AudioMixer::add_bus_to_master(unsigned bus_index, const vector<float> &samples_bus, vector<float> *samples_out)
693 assert(samples_bus.size() == samples_out->size());
694 assert(samples_bus.size() % 2 == 0);
695 unsigned num_samples = samples_bus.size() / 2;
696 const float new_volume_db = mute[bus_index] ? -90.0f : fader_volume_db[bus_index].load();
697 if (fabs(new_volume_db - last_fader_volume_db[bus_index]) > 1e-3) {
698 // The volume has changed; do a fade over the course of this frame.
699 // (We might have some numerical issues here, but it seems to sound OK.)
700 // For the purpose of fading here, the silence floor is set to -90 dB
701 // (the fader only goes to -84).
702 float old_volume = from_db(max<float>(last_fader_volume_db[bus_index], -90.0f));
703 float volume = from_db(max<float>(new_volume_db, -90.0f));
705 float volume_inc = pow(volume / old_volume, 1.0 / num_samples);
707 if (bus_index == 0) {
708 for (unsigned i = 0; i < num_samples; ++i) {
709 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
710 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
711 volume *= volume_inc;
714 for (unsigned i = 0; i < num_samples; ++i) {
715 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
716 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
717 volume *= volume_inc;
720 } else if (new_volume_db > -90.0f) {
721 float volume = from_db(new_volume_db);
722 if (bus_index == 0) {
723 for (unsigned i = 0; i < num_samples; ++i) {
724 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
725 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
728 for (unsigned i = 0; i < num_samples; ++i) {
729 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
730 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
735 last_fader_volume_db[bus_index] = new_volume_db;
738 void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right)
740 assert(left.size() == right.size());
741 const float volume = mute[bus_index] ? 0.0f : from_db(fader_volume_db[bus_index]);
742 const float peak_levels[2] = {
743 find_peak(left.data(), left.size()) * volume,
744 find_peak(right.data(), right.size()) * volume
746 for (unsigned channel = 0; channel < 2; ++channel) {
747 // Compute the current value, including hold and falloff.
748 // The constants are borrowed from zita-mu1 by Fons Adriaensen.
749 static constexpr float hold_sec = 0.5f;
750 static constexpr float falloff_db_sec = 15.0f; // dB/sec falloff after hold.
752 PeakHistory &history = peak_history[bus_index][channel];
753 history.historic_peak = max(history.historic_peak, peak_levels[channel]);
754 if (history.age_seconds < hold_sec) {
755 current_peak = history.last_peak;
757 current_peak = history.last_peak * from_db(-falloff_db_sec * (history.age_seconds - hold_sec));
760 // See if we have a new peak to replace the old (possibly falling) one.
761 if (peak_levels[channel] > current_peak) {
762 history.last_peak = peak_levels[channel];
763 history.age_seconds = 0.0f; // Not 100% correct, but more than good enough given our frame sizes.
764 current_peak = peak_levels[channel];
766 history.age_seconds += float(left.size()) / OUTPUT_FREQUENCY;
768 history.current_level = peak_levels[channel];
769 history.current_peak = current_peak;
773 void AudioMixer::update_meters(const vector<float> &samples)
775 // Upsample 4x to find interpolated peak.
776 peak_resampler.inp_data = const_cast<float *>(samples.data());
777 peak_resampler.inp_count = samples.size() / 2;
779 vector<float> interpolated_samples;
780 interpolated_samples.resize(samples.size());
782 lock_guard<mutex> lock(audio_measure_mutex);
784 while (peak_resampler.inp_count > 0) { // About four iterations.
785 peak_resampler.out_data = &interpolated_samples[0];
786 peak_resampler.out_count = interpolated_samples.size() / 2;
787 peak_resampler.process();
788 size_t out_stereo_samples = interpolated_samples.size() / 2 - peak_resampler.out_count;
789 peak = max<float>(peak, find_peak(interpolated_samples.data(), out_stereo_samples * 2));
790 peak_resampler.out_data = nullptr;
794 // Find R128 levels and L/R correlation.
795 vector<float> left, right;
796 deinterleave_samples(samples, &left, &right);
797 float *ptrs[] = { left.data(), right.data() };
799 lock_guard<mutex> lock(audio_measure_mutex);
800 r128.process(left.size(), ptrs);
801 correlation.process_samples(samples);
804 send_audio_level_callback();
807 void AudioMixer::reset_meters()
809 lock_guard<mutex> lock(audio_measure_mutex);
810 peak_resampler.reset();
817 void AudioMixer::send_audio_level_callback()
819 if (audio_level_callback == nullptr) {
823 lock_guard<mutex> lock(audio_measure_mutex);
824 double loudness_s = r128.loudness_S();
825 double loudness_i = r128.integrated();
826 double loudness_range_low = r128.range_min();
827 double loudness_range_high = r128.range_max();
829 vector<BusLevel> bus_levels;
830 bus_levels.resize(input_mapping.buses.size());
832 lock_guard<mutex> lock(compressor_mutex);
833 for (unsigned bus_index = 0; bus_index < bus_levels.size(); ++bus_index) {
834 bus_levels[bus_index].current_level_dbfs[0] = to_db(peak_history[bus_index][0].current_level);
835 bus_levels[bus_index].current_level_dbfs[1] = to_db(peak_history[bus_index][1].current_level);
836 bus_levels[bus_index].peak_level_dbfs[0] = to_db(peak_history[bus_index][0].current_peak);
837 bus_levels[bus_index].peak_level_dbfs[1] = to_db(peak_history[bus_index][1].current_peak);
838 bus_levels[bus_index].historic_peak_dbfs = to_db(
839 max(peak_history[bus_index][0].historic_peak,
840 peak_history[bus_index][1].historic_peak));
841 bus_levels[bus_index].gain_staging_db = gain_staging_db[bus_index];
842 if (compressor_enabled[bus_index]) {
843 bus_levels[bus_index].compressor_attenuation_db = -to_db(compressor[bus_index]->get_attenuation());
845 bus_levels[bus_index].compressor_attenuation_db = 0.0;
850 audio_level_callback(loudness_s, to_db(peak), bus_levels,
851 loudness_i, loudness_range_low, loudness_range_high,
852 to_db(final_makeup_gain),
853 correlation.get_correlation());
856 map<DeviceSpec, DeviceInfo> AudioMixer::get_devices()
858 lock_guard<timed_mutex> lock(audio_mutex);
860 map<DeviceSpec, DeviceInfo> devices;
861 for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
862 const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index };
863 const AudioDevice *device = &video_cards[card_index];
865 info.display_name = device->display_name;
866 info.num_channels = 8;
867 devices.insert(make_pair(spec, info));
869 vector<ALSAPool::Device> available_alsa_devices = alsa_pool.get_devices();
870 for (unsigned card_index = 0; card_index < available_alsa_devices.size(); ++card_index) {
871 const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index };
872 const ALSAPool::Device &device = available_alsa_devices[card_index];
874 info.display_name = device.display_name();
875 info.num_channels = device.num_channels;
876 info.alsa_name = device.name;
877 info.alsa_info = device.info;
878 info.alsa_address = device.address;
879 devices.insert(make_pair(spec, info));
884 void AudioMixer::set_display_name(DeviceSpec device_spec, const string &name)
886 AudioDevice *device = find_audio_device(device_spec);
888 lock_guard<timed_mutex> lock(audio_mutex);
889 device->display_name = name;
892 void AudioMixer::serialize_device(DeviceSpec device_spec, DeviceSpecProto *device_spec_proto)
894 lock_guard<timed_mutex> lock(audio_mutex);
895 switch (device_spec.type) {
896 case InputSourceType::SILENCE:
897 device_spec_proto->set_type(DeviceSpecProto::SILENCE);
899 case InputSourceType::CAPTURE_CARD:
900 device_spec_proto->set_type(DeviceSpecProto::CAPTURE_CARD);
901 device_spec_proto->set_index(device_spec.index);
902 device_spec_proto->set_display_name(video_cards[device_spec.index].display_name);
904 case InputSourceType::ALSA_INPUT:
905 alsa_pool.serialize_device(device_spec.index, device_spec_proto);
910 void AudioMixer::set_simple_input(unsigned card_index)
912 InputMapping new_input_mapping;
913 InputMapping::Bus input;
915 input.device.type = InputSourceType::CAPTURE_CARD;
916 input.device.index = card_index;
917 input.source_channel[0] = 0;
918 input.source_channel[1] = 1;
920 new_input_mapping.buses.push_back(input);
922 lock_guard<timed_mutex> lock(audio_mutex);
923 current_mapping_mode = MappingMode::SIMPLE;
924 set_input_mapping_lock_held(new_input_mapping);
925 fader_volume_db[0] = 0.0f;
928 unsigned AudioMixer::get_simple_input() const
930 lock_guard<timed_mutex> lock(audio_mutex);
931 if (input_mapping.buses.size() == 1 &&
932 input_mapping.buses[0].device.type == InputSourceType::CAPTURE_CARD &&
933 input_mapping.buses[0].source_channel[0] == 0 &&
934 input_mapping.buses[0].source_channel[1] == 1) {
935 return input_mapping.buses[0].device.index;
937 return numeric_limits<unsigned>::max();
941 void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
943 lock_guard<timed_mutex> lock(audio_mutex);
944 set_input_mapping_lock_held(new_input_mapping);
945 current_mapping_mode = MappingMode::MULTICHANNEL;
948 AudioMixer::MappingMode AudioMixer::get_mapping_mode() const
950 lock_guard<timed_mutex> lock(audio_mutex);
951 return current_mapping_mode;
954 void AudioMixer::set_input_mapping_lock_held(const InputMapping &new_input_mapping)
956 map<DeviceSpec, set<unsigned>> interesting_channels;
957 for (const InputMapping::Bus &bus : new_input_mapping.buses) {
958 if (bus.device.type == InputSourceType::CAPTURE_CARD ||
959 bus.device.type == InputSourceType::ALSA_INPUT) {
960 for (unsigned channel = 0; channel < 2; ++channel) {
961 if (bus.source_channel[channel] != -1) {
962 interesting_channels[bus.device].insert(bus.source_channel[channel]);
968 // Reset resamplers for all cards that don't have the exact same state as before.
969 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
970 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
971 AudioDevice *device = find_audio_device(device_spec);
972 if (device->interesting_channels != interesting_channels[device_spec]) {
973 device->interesting_channels = interesting_channels[device_spec];
974 reset_resampler_mutex_held(device_spec);
977 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
978 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
979 AudioDevice *device = find_audio_device(device_spec);
980 if (interesting_channels[device_spec].empty()) {
981 alsa_pool.release_device(card_index);
983 alsa_pool.hold_device(card_index);
985 if (device->interesting_channels != interesting_channels[device_spec]) {
986 device->interesting_channels = interesting_channels[device_spec];
987 alsa_pool.reset_device(device_spec.index);
988 reset_resampler_mutex_held(device_spec);
992 input_mapping = new_input_mapping;
995 InputMapping AudioMixer::get_input_mapping() const
997 lock_guard<timed_mutex> lock(audio_mutex);
998 return input_mapping;
1001 unsigned AudioMixer::num_buses() const
1003 lock_guard<timed_mutex> lock(audio_mutex);
1004 return input_mapping.buses.size();
1007 void AudioMixer::reset_peak(unsigned bus_index)
1009 lock_guard<timed_mutex> lock(audio_mutex);
1010 for (unsigned channel = 0; channel < 2; ++channel) {
1011 PeakHistory &history = peak_history[bus_index][channel];
1012 history.current_level = 0.0f;
1013 history.historic_peak = 0.0f;
1014 history.current_peak = 0.0f;
1015 history.last_peak = 0.0f;
1016 history.age_seconds = 0.0f;
1020 AudioMixer *global_audio_mixer = nullptr;