1 #include "audio_mixer.h"
4 #include <bmusb/bmusb.h>
23 #include "shared/metrics.h"
25 #include "shared/timebase.h"
27 using namespace bmusb;
29 using namespace std::chrono;
30 using namespace std::placeholders;
34 // TODO: If these prove to be a bottleneck, they can be SSSE3-optimized
35 // (usually including multiple channels at a time).
37 void convert_fixed16_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
38 const uint8_t *src, size_t in_channel, size_t in_num_channels,
41 assert(in_channel < in_num_channels);
42 assert(out_channel < out_num_channels);
43 src += in_channel * 2;
46 for (size_t i = 0; i < num_samples; ++i) {
47 int16_t s = le16toh(*(int16_t *)src);
48 *dst = s * (1.0f / 32768.0f);
50 src += 2 * in_num_channels;
51 dst += out_num_channels;
55 void convert_fixed16_to_fixed32(int32_t *dst, size_t out_channel, size_t out_num_channels,
56 const uint8_t *src, size_t in_channel, size_t in_num_channels,
59 assert(in_channel < in_num_channels);
60 assert(out_channel < out_num_channels);
61 src += in_channel * 2;
64 for (size_t i = 0; i < num_samples; ++i) {
65 uint32_t s = uint32_t(uint16_t(le16toh(*(int16_t *)src))) << 16;
67 // Keep the sign bit in place, repeat the other 15 bits as far as they go.
68 *dst = s | ((s & 0x7fffffff) >> 15) | ((s & 0x7fffffff) >> 30);
70 src += 2 * in_num_channels;
71 dst += out_num_channels;
75 void convert_fixed24_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
76 const uint8_t *src, size_t in_channel, size_t in_num_channels,
79 assert(in_channel < in_num_channels);
80 assert(out_channel < out_num_channels);
81 src += in_channel * 3;
84 for (size_t i = 0; i < num_samples; ++i) {
88 uint32_t s = (s1 << 8) | (s2 << 16) | (s3 << 24); // Note: The bottom eight bits are zero; s3 includes the sign bit.
89 *dst = int(s) * (1.0f / (256.0f * 8388608.0f)); // 256 for signed down-shift by 8, then 2^23 for the actual conversion.
91 src += 3 * in_num_channels;
92 dst += out_num_channels;
96 void convert_fixed24_to_fixed32(int32_t *dst, size_t out_channel, size_t out_num_channels,
97 const uint8_t *src, size_t in_channel, size_t in_num_channels,
100 assert(in_channel < in_num_channels);
101 assert(out_channel < out_num_channels);
102 src += in_channel * 3;
105 for (size_t i = 0; i < num_samples; ++i) {
106 uint32_t s1 = src[0];
107 uint32_t s2 = src[1];
108 uint32_t s3 = src[2];
109 uint32_t s = (s1 << 8) | (s2 << 16) | (s3 << 24);
111 // Keep the sign bit in place, repeat the other 23 bits as far as they go.
112 *dst = s | ((s & 0x7fffffff) >> 23);
114 src += 3 * in_num_channels;
115 dst += out_num_channels;
119 void convert_fixed32_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
120 const uint8_t *src, size_t in_channel, size_t in_num_channels,
123 assert(in_channel < in_num_channels);
124 assert(out_channel < out_num_channels);
125 src += in_channel * 4;
128 for (size_t i = 0; i < num_samples; ++i) {
129 int32_t s = le32toh(*(int32_t *)src);
130 *dst = s * (1.0f / 2147483648.0f);
132 src += 4 * in_num_channels;
133 dst += out_num_channels;
137 // Basically just a reinterleave.
138 void convert_fixed32_to_fixed32(int32_t *dst, size_t out_channel, size_t out_num_channels,
139 const uint8_t *src, size_t in_channel, size_t in_num_channels,
142 assert(in_channel < in_num_channels);
143 assert(out_channel < out_num_channels);
144 src += in_channel * 4;
147 for (size_t i = 0; i < num_samples; ++i) {
148 int32_t s = le32toh(*(int32_t *)src);
151 src += 4 * in_num_channels;
152 dst += out_num_channels;
156 float find_peak_plain(const float *samples, size_t num_samples) __attribute__((unused));
158 float find_peak_plain(const float *samples, size_t num_samples)
160 float m = fabs(samples[0]);
161 for (size_t i = 1; i < num_samples; ++i) {
162 m = max(m, fabs(samples[i]));
168 static inline float horizontal_max(__m128 m)
170 __m128 tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 0, 3, 2));
171 m = _mm_max_ps(m, tmp);
172 tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 3, 0, 1));
173 m = _mm_max_ps(m, tmp);
174 return _mm_cvtss_f32(m);
177 float find_peak(const float *samples, size_t num_samples)
179 const __m128 abs_mask = _mm_castsi128_ps(_mm_set1_epi32(0x7fffffffu));
180 __m128 m = _mm_setzero_ps();
181 for (size_t i = 0; i < (num_samples & ~3); i += 4) {
182 __m128 x = _mm_loadu_ps(samples + i);
183 x = _mm_and_ps(x, abs_mask);
184 m = _mm_max_ps(m, x);
186 float result = horizontal_max(m);
188 for (size_t i = (num_samples & ~3); i < num_samples; ++i) {
189 result = max(result, fabs(samples[i]));
193 // Self-test. We should be bit-exact the same.
194 float reference_result = find_peak_plain(samples, num_samples);
195 if (result != reference_result) {
196 fprintf(stderr, "Error: Peak is %f [%f %f %f %f]; should be %f.\n",
198 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(0, 0, 0, 0))),
199 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 1, 1, 1))),
200 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 2, 2, 2))),
201 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(3, 3, 3, 3))),
209 float find_peak(const float *samples, size_t num_samples)
211 return find_peak_plain(samples, num_samples);
215 void deinterleave_samples(const vector<float> &in, vector<float> *out_l, vector<float> *out_r)
217 size_t num_samples = in.size() / 2;
218 out_l->resize(num_samples);
219 out_r->resize(num_samples);
221 const float *inptr = in.data();
222 float *lptr = &(*out_l)[0];
223 float *rptr = &(*out_r)[0];
224 for (size_t i = 0; i < num_samples; ++i) {
232 AudioMixer::AudioMixer(unsigned num_capture_cards, unsigned num_ffmpeg_inputs)
233 : num_capture_cards(num_capture_cards),
234 num_ffmpeg_inputs(num_ffmpeg_inputs),
235 ffmpeg_inputs(new AudioDevice[num_ffmpeg_inputs]),
236 limiter(OUTPUT_FREQUENCY),
237 correlation(OUTPUT_FREQUENCY)
239 for (unsigned bus_index = 0; bus_index < MAX_BUSES; ++bus_index) {
240 locut[bus_index].init(FILTER_HPF, 2);
241 eq[bus_index][EQ_BAND_BASS].init(FILTER_LOW_SHELF, 1);
242 // Note: EQ_BAND_MID isn't used (see comments in apply_eq()).
243 eq[bus_index][EQ_BAND_TREBLE].init(FILTER_HIGH_SHELF, 1);
244 compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
245 level_compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
247 set_bus_settings(bus_index, get_default_bus_settings());
249 set_limiter_enabled(global_flags.limiter_enabled);
250 set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
252 r128.init(2, OUTPUT_FREQUENCY);
255 // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
256 // and there's a limit to how important the peak meter is.
257 peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
259 global_audio_mixer = this;
262 if (!global_flags.input_mapping_filename.empty()) {
263 // Must happen after ALSAPool is initialized, as it needs to know the card list.
264 current_mapping_mode = MappingMode::MULTICHANNEL;
265 InputMapping new_input_mapping;
266 if (!load_input_mapping_from_file(get_devices(),
267 global_flags.input_mapping_filename,
268 &new_input_mapping)) {
269 fprintf(stderr, "Failed to load input mapping from '%s', exiting.\n",
270 global_flags.input_mapping_filename.c_str());
273 set_input_mapping(new_input_mapping);
275 set_simple_input(/*card_index=*/0);
276 if (global_flags.multichannel_mapping_mode) {
277 current_mapping_mode = MappingMode::MULTICHANNEL;
281 global_metrics.add("audio_loudness_short_lufs", &metric_audio_loudness_short_lufs, Metrics::TYPE_GAUGE);
282 global_metrics.add("audio_loudness_integrated_lufs", &metric_audio_loudness_integrated_lufs, Metrics::TYPE_GAUGE);
283 global_metrics.add("audio_loudness_range_low_lufs", &metric_audio_loudness_range_low_lufs, Metrics::TYPE_GAUGE);
284 global_metrics.add("audio_loudness_range_high_lufs", &metric_audio_loudness_range_high_lufs, Metrics::TYPE_GAUGE);
285 global_metrics.add("audio_peak_dbfs", &metric_audio_peak_dbfs, Metrics::TYPE_GAUGE);
286 global_metrics.add("audio_final_makeup_gain_db", &metric_audio_final_makeup_gain_db, Metrics::TYPE_GAUGE);
287 global_metrics.add("audio_correlation", &metric_audio_correlation, Metrics::TYPE_GAUGE);
290 void AudioMixer::reset_resampler(DeviceSpec device_spec)
292 lock_guard<timed_mutex> lock(audio_mutex);
293 reset_resampler_mutex_held(device_spec);
296 void AudioMixer::reset_resampler_mutex_held(DeviceSpec device_spec)
298 AudioDevice *device = find_audio_device(device_spec);
300 if (device->interesting_channels.empty()) {
301 device->resampling_queue.reset();
303 // Make sure we never get negative delay. Even 1 ms is probably way less than we
304 // could ever hope to actually have; this is just a failsafe.
305 double delay_ms = max(global_flags.audio_queue_length_ms + device->extra_delay_ms, 1.0);
307 device->resampling_queue.reset(new ResamplingQueue(
308 device_spec, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size(),
313 bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, steady_clock::time_point frame_time)
315 AudioDevice *device = find_audio_device(device_spec);
317 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
318 if (!lock.try_lock_for(chrono::milliseconds(10))) {
321 if (device->resampling_queue == nullptr) {
322 // No buses use this device; throw it away.
326 unsigned num_channels = device->interesting_channels.size();
327 assert(num_channels > 0);
329 // Convert the audio to fp32.
330 unique_ptr<float[]> audio(new float[num_samples * num_channels]);
331 unsigned channel_index = 0;
332 for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) {
333 switch (audio_format.bits_per_sample) {
335 assert(num_samples == 0);
338 convert_fixed16_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
341 convert_fixed24_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
344 convert_fixed32_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
347 fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
352 // If we changed frequency since last frame, we'll need to reset the resampler.
353 if (audio_format.sample_rate != device->capture_frequency) {
354 device->capture_frequency = audio_format.sample_rate;
355 reset_resampler_mutex_held(device_spec);
359 device->resampling_queue->add_input_samples(frame_time, audio.get(), num_samples, ResamplingQueue::ADJUST_RATE);
363 vector<int32_t> convert_audio_to_fixed32(const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, unsigned num_channels)
365 vector<int32_t> audio;
367 if (num_channels > audio_format.num_channels) {
368 audio.resize(num_samples * num_channels, 0);
370 audio.resize(num_samples * num_channels);
372 for (unsigned channel_index = 0; channel_index < num_channels && channel_index < audio_format.num_channels; ++channel_index) {
373 switch (audio_format.bits_per_sample) {
375 assert(num_samples == 0);
378 convert_fixed16_to_fixed32(&audio[0], channel_index, num_channels, data, channel_index, audio_format.num_channels, num_samples);
381 convert_fixed24_to_fixed32(&audio[0], channel_index, num_channels, data, channel_index, audio_format.num_channels, num_samples);
384 convert_fixed32_to_fixed32(&audio[0], channel_index, num_channels, data, channel_index, audio_format.num_channels, num_samples);
387 fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
395 bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames)
397 AudioDevice *device = find_audio_device(device_spec);
399 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
400 if (!lock.try_lock_for(chrono::milliseconds(10))) {
403 if (device->resampling_queue == nullptr) {
404 // No buses use this device; throw it away.
408 unsigned num_channels = device->interesting_channels.size();
409 assert(num_channels > 0);
411 vector<float> silence(samples_per_frame * num_channels, 0.0f);
412 for (unsigned i = 0; i < num_frames; ++i) {
413 device->resampling_queue->add_input_samples(steady_clock::now(), silence.data(), samples_per_frame, ResamplingQueue::DO_NOT_ADJUST_RATE);
418 bool AudioMixer::silence_card(DeviceSpec device_spec, bool silence)
420 AudioDevice *device = find_audio_device(device_spec);
422 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
423 if (!lock.try_lock_for(chrono::milliseconds(10))) {
427 if (device->silenced && !silence) {
428 reset_resampler_mutex_held(device_spec);
430 device->silenced = silence;
434 AudioMixer::BusSettings AudioMixer::get_default_bus_settings()
436 BusSettings settings;
437 settings.fader_volume_db = 0.0f;
438 settings.muted = false;
439 settings.locut_enabled = global_flags.locut_enabled;
440 settings.stereo_width = 1.0f;
441 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
442 settings.eq_level_db[band_index] = 0.0f;
444 settings.gain_staging_db = global_flags.initial_gain_staging_db;
445 settings.level_compressor_enabled = global_flags.gain_staging_auto;
446 settings.compressor_threshold_dbfs = ref_level_dbfs - 12.0f; // -12 dB.
447 settings.compressor_enabled = global_flags.compressor_enabled;
451 AudioMixer::BusSettings AudioMixer::get_bus_settings(unsigned bus_index) const
453 lock_guard<timed_mutex> lock(audio_mutex);
454 BusSettings settings;
455 settings.fader_volume_db = fader_volume_db[bus_index];
456 settings.muted = mute[bus_index];
457 settings.locut_enabled = locut_enabled[bus_index];
458 settings.stereo_width = stereo_width[bus_index];
459 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
460 settings.eq_level_db[band_index] = eq_level_db[bus_index][band_index];
462 settings.gain_staging_db = gain_staging_db[bus_index];
463 settings.level_compressor_enabled = level_compressor_enabled[bus_index];
464 settings.compressor_threshold_dbfs = compressor_threshold_dbfs[bus_index];
465 settings.compressor_enabled = compressor_enabled[bus_index];
469 void AudioMixer::set_bus_settings(unsigned bus_index, const AudioMixer::BusSettings &settings)
471 lock_guard<timed_mutex> lock(audio_mutex);
472 fader_volume_db[bus_index] = settings.fader_volume_db;
473 mute[bus_index] = settings.muted;
474 locut_enabled[bus_index] = settings.locut_enabled;
475 stereo_width[bus_index] = settings.stereo_width;
476 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
477 eq_level_db[bus_index][band_index] = settings.eq_level_db[band_index];
479 gain_staging_db[bus_index] = settings.gain_staging_db;
480 last_gain_staging_db[bus_index] = gain_staging_db[bus_index];
481 level_compressor_enabled[bus_index] = settings.level_compressor_enabled;
482 compressor_threshold_dbfs[bus_index] = settings.compressor_threshold_dbfs;
483 compressor_enabled[bus_index] = settings.compressor_enabled;
486 AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
488 switch (device.type) {
489 case InputSourceType::CAPTURE_CARD:
490 return &video_cards[device.index];
491 case InputSourceType::ALSA_INPUT:
492 return &alsa_inputs[device.index];
493 case InputSourceType::FFMPEG_VIDEO_INPUT:
494 return &ffmpeg_inputs[device.index];
495 case InputSourceType::SILENCE:
502 // Get a pointer to the given channel from the given device.
503 // The channel must be picked out earlier and resampled.
504 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)
506 static float zero = 0.0f;
507 if (source_channel == -1 || device_spec.type == InputSourceType::SILENCE) {
512 AudioDevice *device = find_audio_device(device_spec);
513 assert(device->interesting_channels.count(source_channel) != 0);
514 unsigned channel_index = 0;
515 for (int channel : device->interesting_channels) {
516 if (channel == source_channel) break;
519 assert(channel_index < device->interesting_channels.size());
520 const auto it = samples_card.find(device_spec);
521 assert(it != samples_card.end());
522 *srcptr = &(it->second)[channel_index];
523 *stride = device->interesting_channels.size();
526 // TODO: Can be SSSE3-optimized if need be.
527 void AudioMixer::fill_audio_bus(const map<DeviceSpec, vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float stereo_width, float *output)
529 if (bus.device.type == InputSourceType::SILENCE) {
530 memset(output, 0, num_samples * 2 * sizeof(*output));
532 assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
533 bus.device.type == InputSourceType::ALSA_INPUT ||
534 bus.device.type == InputSourceType::FFMPEG_VIDEO_INPUT);
535 const float *lsrc, *rsrc;
536 unsigned lstride, rstride;
537 float *dptr = output;
538 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[0], &lsrc, &lstride);
539 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[1], &rsrc, &rstride);
541 // Apply stereo width settings. Set stereo width w to a 0..1 range instead of
542 // -1..1, since it makes for much easier calculations (so 0.5 = completely mono).
543 // Then, what we want is
545 // L' = wL + (1-w)R = R + w(L-R)
546 // R' = wR + (1-w)L = L + w(R-L)
548 // This can be further simplified calculation-wise by defining the weighted
549 // difference signal D = w(R-L), so that:
553 float w = 0.5f * stereo_width + 0.5f;
554 if (bus.source_channel[0] == bus.source_channel[1]) {
555 // Mono anyway, so no need to bother.
557 } else if (fabs(w) < 1e-3) {
560 swap(lstride, rstride);
563 if (fabs(w - 1.0f) < 1e-3) {
564 // No calculations needed for stereo_width = 1.
565 for (unsigned i = 0; i < num_samples; ++i) {
573 for (unsigned i = 0; i < num_samples; ++i) {
574 float left = *lsrc, right = *rsrc;
575 float diff = w * (right - left);
576 *dptr++ = right - diff;
577 *dptr++ = left + diff;
585 vector<DeviceSpec> AudioMixer::get_active_devices() const
587 vector<DeviceSpec> ret;
588 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
589 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
590 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
591 ret.push_back(device_spec);
594 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
595 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
596 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
597 ret.push_back(device_spec);
600 for (unsigned card_index = 0; card_index < num_ffmpeg_inputs; ++card_index) {
601 const DeviceSpec device_spec{InputSourceType::FFMPEG_VIDEO_INPUT, card_index};
602 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
603 ret.push_back(device_spec);
611 void apply_gain(float db, float last_db, vector<float> *samples)
613 if (fabs(db - last_db) < 1e-3) {
614 // Constant over this frame.
615 const float gain = from_db(db);
616 for (size_t i = 0; i < samples->size(); ++i) {
617 (*samples)[i] *= gain;
620 // We need to do a fade.
621 unsigned num_samples = samples->size() / 2;
622 float gain = from_db(last_db);
623 const float gain_inc = pow(from_db(db - last_db), 1.0 / num_samples);
624 for (size_t i = 0; i < num_samples; ++i) {
625 (*samples)[i * 2 + 0] *= gain;
626 (*samples)[i * 2 + 1] *= gain;
634 vector<float> AudioMixer::get_output(steady_clock::time_point ts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
636 map<DeviceSpec, vector<float>> samples_card;
637 vector<float> samples_bus;
639 lock_guard<timed_mutex> lock(audio_mutex);
641 // Pick out all the interesting channels from all the cards.
642 for (const DeviceSpec &device_spec : get_active_devices()) {
643 AudioDevice *device = find_audio_device(device_spec);
644 samples_card[device_spec].resize(num_samples * device->interesting_channels.size());
645 if (device->silenced) {
646 memset(&samples_card[device_spec][0], 0, samples_card[device_spec].size() * sizeof(float));
648 device->resampling_queue->get_output_samples(
650 &samples_card[device_spec][0],
652 rate_adjustment_policy);
656 vector<float> samples_out, left, right;
657 samples_out.resize(num_samples * 2);
658 samples_bus.resize(num_samples * 2);
659 for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
660 fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, stereo_width[bus_index], &samples_bus[0]);
661 apply_eq(bus_index, &samples_bus);
664 lock_guard<mutex> lock(compressor_mutex);
666 // Apply a level compressor to get the general level right.
667 // Basically, if it's over about -40 dBFS, we squeeze it down to that level
668 // (or more precisely, near it, since we don't use infinite ratio),
669 // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
670 // entirely arbitrary, but from practical tests with speech, it seems to
671 // put ut around -23 LUFS, so it's a reasonable starting point for later use.
672 if (level_compressor_enabled[bus_index]) {
673 float threshold = 0.01f; // -40 dBFS.
675 float attack_time = 0.5f;
676 float release_time = 20.0f;
677 float makeup_gain = from_db(ref_level_dbfs - (-40.0f)); // +26 dB.
678 level_compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
679 gain_staging_db[bus_index] = to_db(level_compressor[bus_index]->get_attenuation() * makeup_gain);
681 // Just apply the gain we already had.
682 float db = gain_staging_db[bus_index];
683 float last_db = last_gain_staging_db[bus_index];
684 apply_gain(db, last_db, &samples_bus);
686 last_gain_staging_db[bus_index] = gain_staging_db[bus_index];
689 printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
690 level_compressor.get_level(), to_db(level_compressor.get_level()),
691 level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()),
692 to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
695 // The real compressor.
696 if (compressor_enabled[bus_index]) {
697 float threshold = from_db(compressor_threshold_dbfs[bus_index]);
699 float attack_time = 0.005f;
700 float release_time = 0.040f;
701 float makeup_gain = 2.0f; // +6 dB.
702 compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
703 // compressor_att = compressor.get_attenuation();
707 add_bus_to_master(bus_index, samples_bus, &samples_out);
708 deinterleave_samples(samples_bus, &left, &right);
709 measure_bus_levels(bus_index, left, right);
713 lock_guard<mutex> lock(compressor_mutex);
715 // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
716 // Note that since ratio is not infinite, we could go slightly higher than this.
717 if (limiter_enabled) {
718 float threshold = from_db(limiter_threshold_dbfs);
720 float attack_time = 0.0f; // Instant.
721 float release_time = 0.020f;
722 float makeup_gain = 1.0f; // 0 dB.
723 limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
724 // limiter_att = limiter.get_attenuation();
727 // printf("limiter=%+5.1f compressor=%+5.1f\n", to_db(limiter_att), to_db(compressor_att));
730 // At this point, we are most likely close to +0 LU (at least if the
731 // faders sum to 0 dB and the compressors are on), but all of our
732 // measurements have been on raw sample values, not R128 values.
733 // So we have a final makeup gain to get us to +0 LU; the gain
734 // adjustments required should be relatively small, and also, the
735 // offset shouldn't change much (only if the type of audio changes
736 // significantly). Thus, we shoot for updating this value basically
737 // “whenever we process buffers”, since the R128 calculation isn't exactly
738 // something we get out per-sample.
740 // Note that there's a feedback loop here, so we choose a very slow filter
741 // (half-time of 30 seconds).
742 double target_loudness_factor, alpha;
743 double loudness_lu = r128.loudness_M() - ref_level_lufs;
744 target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
746 // If we're outside +/- 5 LU (after correction), we don't count it as
747 // a normal signal (probably silence) and don't change the
748 // correction factor; just apply what we already have.
749 if (fabs(loudness_lu) >= 5.0 || !final_makeup_gain_auto) {
752 // Formula adapted from
753 // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
754 const double half_time_s = 30.0;
755 const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
756 alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
760 lock_guard<mutex> lock(compressor_mutex);
761 double m = final_makeup_gain;
762 for (size_t i = 0; i < samples_out.size(); i += 2) {
763 samples_out[i + 0] *= m;
764 samples_out[i + 1] *= m;
765 m += (target_loudness_factor - m) * alpha;
767 final_makeup_gain = m;
770 update_meters(samples_out);
777 void apply_filter_fade(StereoFilter *filter, float *data, unsigned num_samples, float cutoff_hz, float db, float last_db)
779 // A granularity of 32 samples is an okay tradeoff between speed and
780 // smoothness; recalculating the filters is pretty expensive, so it's
781 // good that we don't do this all the time.
782 static constexpr unsigned filter_granularity_samples = 32;
784 const float cutoff_linear = cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY;
785 if (fabs(db - last_db) < 1e-3) {
786 // Constant over this frame.
787 if (fabs(db) > 0.01f) {
788 filter->render(data, num_samples, cutoff_linear, 0.5f, db / 40.0f);
791 // We need to do a fade. (Rounding up avoids division by zero.)
792 unsigned num_blocks = (num_samples + filter_granularity_samples - 1) / filter_granularity_samples;
793 const float inc_db_norm = (db - last_db) / 40.0f / num_blocks;
794 float db_norm = db / 40.0f;
795 for (size_t i = 0; i < num_samples; i += filter_granularity_samples) {
796 size_t samples_this_block = std::min<size_t>(num_samples - i, filter_granularity_samples);
797 filter->render(data + i * 2, samples_this_block, cutoff_linear, 0.5f, db_norm);
798 db_norm += inc_db_norm;
805 void AudioMixer::apply_eq(unsigned bus_index, vector<float> *samples_bus)
807 constexpr float bass_freq_hz = 200.0f;
808 constexpr float treble_freq_hz = 4700.0f;
810 // Cut away everything under 120 Hz (or whatever the cutoff is);
811 // we don't need it for voice, and it will reduce headroom
812 // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
813 // should be dampened.)
814 if (locut_enabled[bus_index]) {
815 locut[bus_index].render(samples_bus->data(), samples_bus->size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
818 // Apply the rest of the EQ. Since we only have a simple three-band EQ,
819 // we can implement it with two shelf filters. We use a simple gain to
820 // set the mid-level filter, and then offset the low and high bands
821 // from that if we need to. (We could perhaps have folded the gain into
822 // the next part, but it's so cheap that the trouble isn't worth it.)
824 // If any part of the EQ has changed appreciably since last frame,
825 // we fade smoothly during the course of this frame.
826 const float bass_db = eq_level_db[bus_index][EQ_BAND_BASS];
827 const float mid_db = eq_level_db[bus_index][EQ_BAND_MID];
828 const float treble_db = eq_level_db[bus_index][EQ_BAND_TREBLE];
830 const float last_bass_db = last_eq_level_db[bus_index][EQ_BAND_BASS];
831 const float last_mid_db = last_eq_level_db[bus_index][EQ_BAND_MID];
832 const float last_treble_db = last_eq_level_db[bus_index][EQ_BAND_TREBLE];
834 assert(samples_bus->size() % 2 == 0);
835 const unsigned num_samples = samples_bus->size() / 2;
837 apply_gain(mid_db, last_mid_db, samples_bus);
839 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);
840 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);
842 last_eq_level_db[bus_index][EQ_BAND_BASS] = bass_db;
843 last_eq_level_db[bus_index][EQ_BAND_MID] = mid_db;
844 last_eq_level_db[bus_index][EQ_BAND_TREBLE] = treble_db;
847 void AudioMixer::add_bus_to_master(unsigned bus_index, const vector<float> &samples_bus, vector<float> *samples_out)
849 assert(samples_bus.size() == samples_out->size());
850 assert(samples_bus.size() % 2 == 0);
851 unsigned num_samples = samples_bus.size() / 2;
852 const float new_volume_db = mute[bus_index] ? -90.0f : fader_volume_db[bus_index].load();
853 if (fabs(new_volume_db - last_fader_volume_db[bus_index]) > 1e-3) {
854 // The volume has changed; do a fade over the course of this frame.
855 // (We might have some numerical issues here, but it seems to sound OK.)
856 // For the purpose of fading here, the silence floor is set to -90 dB
857 // (the fader only goes to -84).
858 float old_volume = from_db(max<float>(last_fader_volume_db[bus_index], -90.0f));
859 float volume = from_db(max<float>(new_volume_db, -90.0f));
861 float volume_inc = pow(volume / old_volume, 1.0 / num_samples);
863 if (bus_index == 0) {
864 for (unsigned i = 0; i < num_samples; ++i) {
865 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
866 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
867 volume *= volume_inc;
870 for (unsigned i = 0; i < num_samples; ++i) {
871 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
872 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
873 volume *= volume_inc;
876 } else if (new_volume_db > -90.0f) {
877 float volume = from_db(new_volume_db);
878 if (bus_index == 0) {
879 for (unsigned i = 0; i < num_samples; ++i) {
880 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
881 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
884 for (unsigned i = 0; i < num_samples; ++i) {
885 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
886 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
891 last_fader_volume_db[bus_index] = new_volume_db;
894 void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right)
896 assert(left.size() == right.size());
897 const float volume = mute[bus_index] ? 0.0f : from_db(fader_volume_db[bus_index]);
898 const float peak_levels[2] = {
899 find_peak(left.data(), left.size()) * volume,
900 find_peak(right.data(), right.size()) * volume
902 for (unsigned channel = 0; channel < 2; ++channel) {
903 // Compute the current value, including hold and falloff.
904 // The constants are borrowed from zita-mu1 by Fons Adriaensen.
905 static constexpr float hold_sec = 0.5f;
906 static constexpr float falloff_db_sec = 15.0f; // dB/sec falloff after hold.
908 PeakHistory &history = peak_history[bus_index][channel];
909 history.historic_peak = max(history.historic_peak, peak_levels[channel]);
910 if (history.age_seconds < hold_sec) {
911 current_peak = history.last_peak;
913 current_peak = history.last_peak * from_db(-falloff_db_sec * (history.age_seconds - hold_sec));
916 // See if we have a new peak to replace the old (possibly falling) one.
917 if (peak_levels[channel] > current_peak) {
918 history.last_peak = peak_levels[channel];
919 history.age_seconds = 0.0f; // Not 100% correct, but more than good enough given our frame sizes.
920 current_peak = peak_levels[channel];
922 history.age_seconds += float(left.size()) / OUTPUT_FREQUENCY;
924 history.current_level = peak_levels[channel];
925 history.current_peak = current_peak;
929 void AudioMixer::update_meters(const vector<float> &samples)
931 // Upsample 4x to find interpolated peak.
932 peak_resampler.inp_data = const_cast<float *>(samples.data());
933 peak_resampler.inp_count = samples.size() / 2;
935 vector<float> interpolated_samples;
936 interpolated_samples.resize(samples.size());
938 lock_guard<mutex> lock(audio_measure_mutex);
940 while (peak_resampler.inp_count > 0) { // About four iterations.
941 peak_resampler.out_data = &interpolated_samples[0];
942 peak_resampler.out_count = interpolated_samples.size() / 2;
943 peak_resampler.process();
944 size_t out_stereo_samples = interpolated_samples.size() / 2 - peak_resampler.out_count;
945 peak = max<float>(peak, find_peak(interpolated_samples.data(), out_stereo_samples * 2));
946 peak_resampler.out_data = nullptr;
950 // Find R128 levels and L/R correlation.
951 vector<float> left, right;
952 deinterleave_samples(samples, &left, &right);
953 float *ptrs[] = { left.data(), right.data() };
955 lock_guard<mutex> lock(audio_measure_mutex);
956 r128.process(left.size(), ptrs);
957 correlation.process_samples(samples);
960 send_audio_level_callback();
963 void AudioMixer::reset_meters()
965 lock_guard<mutex> lock(audio_measure_mutex);
966 peak_resampler.reset();
973 void AudioMixer::send_audio_level_callback()
975 if (audio_level_callback == nullptr) {
979 lock_guard<mutex> lock(audio_measure_mutex);
980 double loudness_s = r128.loudness_S();
981 double loudness_i = r128.integrated();
982 double loudness_range_low = r128.range_min();
983 double loudness_range_high = r128.range_max();
985 metric_audio_loudness_short_lufs = loudness_s;
986 metric_audio_loudness_integrated_lufs = loudness_i;
987 metric_audio_loudness_range_low_lufs = loudness_range_low;
988 metric_audio_loudness_range_high_lufs = loudness_range_high;
989 metric_audio_peak_dbfs = to_db(peak);
990 metric_audio_final_makeup_gain_db = to_db(final_makeup_gain);
991 metric_audio_correlation = correlation.get_correlation();
993 vector<BusLevel> bus_levels;
994 bus_levels.resize(input_mapping.buses.size());
996 lock_guard<mutex> lock(compressor_mutex);
997 for (unsigned bus_index = 0; bus_index < bus_levels.size(); ++bus_index) {
998 BusLevel &levels = bus_levels[bus_index];
999 BusMetrics &metrics = bus_metrics[bus_index];
1001 levels.current_level_dbfs[0] = metrics.current_level_dbfs[0] = to_db(peak_history[bus_index][0].current_level);
1002 levels.current_level_dbfs[1] = metrics.current_level_dbfs[1] = to_db(peak_history[bus_index][1].current_level);
1003 levels.peak_level_dbfs[0] = metrics.peak_level_dbfs[0] = to_db(peak_history[bus_index][0].current_peak);
1004 levels.peak_level_dbfs[1] = metrics.peak_level_dbfs[1] = to_db(peak_history[bus_index][1].current_peak);
1005 levels.historic_peak_dbfs = metrics.historic_peak_dbfs = to_db(
1006 max(peak_history[bus_index][0].historic_peak,
1007 peak_history[bus_index][1].historic_peak));
1008 levels.gain_staging_db = metrics.gain_staging_db = gain_staging_db[bus_index];
1009 if (compressor_enabled[bus_index]) {
1010 levels.compressor_attenuation_db = metrics.compressor_attenuation_db = -to_db(compressor[bus_index]->get_attenuation());
1012 levels.compressor_attenuation_db = 0.0;
1013 metrics.compressor_attenuation_db = 0.0 / 0.0;
1018 audio_level_callback(loudness_s, to_db(peak), bus_levels,
1019 loudness_i, loudness_range_low, loudness_range_high,
1020 to_db(final_makeup_gain),
1021 correlation.get_correlation());
1024 map<DeviceSpec, DeviceInfo> AudioMixer::get_devices()
1026 lock_guard<timed_mutex> lock(audio_mutex);
1028 map<DeviceSpec, DeviceInfo> devices;
1029 for (unsigned card_index = 0; card_index < num_capture_cards; ++card_index) {
1030 const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index };
1031 const AudioDevice *device = &video_cards[card_index];
1033 info.display_name = device->display_name;
1034 info.num_channels = 8;
1035 devices.insert(make_pair(spec, info));
1037 vector<ALSAPool::Device> available_alsa_devices = alsa_pool.get_devices();
1038 for (unsigned card_index = 0; card_index < available_alsa_devices.size(); ++card_index) {
1039 const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index };
1040 const ALSAPool::Device &device = available_alsa_devices[card_index];
1042 info.display_name = device.display_name();
1043 info.num_channels = device.num_channels;
1044 info.alsa_name = device.name;
1045 info.alsa_info = device.info;
1046 info.alsa_address = device.address;
1047 devices.insert(make_pair(spec, info));
1049 for (unsigned card_index = 0; card_index < num_ffmpeg_inputs; ++card_index) {
1050 const DeviceSpec spec{ InputSourceType::FFMPEG_VIDEO_INPUT, card_index };
1051 const AudioDevice *device = &ffmpeg_inputs[card_index];
1053 info.display_name = device->display_name;
1054 info.num_channels = 2;
1055 devices.insert(make_pair(spec, info));
1060 void AudioMixer::set_display_name(DeviceSpec device_spec, const string &name)
1062 AudioDevice *device = find_audio_device(device_spec);
1064 lock_guard<timed_mutex> lock(audio_mutex);
1065 device->display_name = name;
1068 void AudioMixer::serialize_device(DeviceSpec device_spec, DeviceSpecProto *device_spec_proto)
1070 lock_guard<timed_mutex> lock(audio_mutex);
1071 switch (device_spec.type) {
1072 case InputSourceType::SILENCE:
1073 device_spec_proto->set_type(DeviceSpecProto::SILENCE);
1075 case InputSourceType::CAPTURE_CARD:
1076 device_spec_proto->set_type(DeviceSpecProto::CAPTURE_CARD);
1077 device_spec_proto->set_index(device_spec.index);
1078 device_spec_proto->set_display_name(video_cards[device_spec.index].display_name);
1080 case InputSourceType::ALSA_INPUT:
1081 alsa_pool.serialize_device(device_spec.index, device_spec_proto);
1083 case InputSourceType::FFMPEG_VIDEO_INPUT:
1084 device_spec_proto->set_type(DeviceSpecProto::FFMPEG_VIDEO_INPUT);
1085 device_spec_proto->set_index(device_spec.index);
1086 device_spec_proto->set_display_name(ffmpeg_inputs[device_spec.index].display_name);
1091 void AudioMixer::set_simple_input(unsigned card_index)
1093 assert(card_index < num_capture_cards + num_ffmpeg_inputs);
1094 InputMapping new_input_mapping;
1095 InputMapping::Bus input;
1096 input.name = "Main";
1097 if (card_index >= num_capture_cards) {
1098 input.device = DeviceSpec{InputSourceType::FFMPEG_VIDEO_INPUT, card_index - num_capture_cards};
1100 input.device = DeviceSpec{InputSourceType::CAPTURE_CARD, card_index};
1102 input.source_channel[0] = 0;
1103 input.source_channel[1] = 1;
1105 new_input_mapping.buses.push_back(input);
1107 lock_guard<timed_mutex> lock(audio_mutex);
1108 current_mapping_mode = MappingMode::SIMPLE;
1109 set_input_mapping_lock_held(new_input_mapping);
1110 fader_volume_db[0] = 0.0f;
1113 unsigned AudioMixer::get_simple_input() const
1115 lock_guard<timed_mutex> lock(audio_mutex);
1116 if (input_mapping.buses.size() == 1 &&
1117 input_mapping.buses[0].device.type == InputSourceType::CAPTURE_CARD &&
1118 input_mapping.buses[0].source_channel[0] == 0 &&
1119 input_mapping.buses[0].source_channel[1] == 1) {
1120 return input_mapping.buses[0].device.index;
1121 } else if (input_mapping.buses.size() == 1 &&
1122 input_mapping.buses[0].device.type == InputSourceType::FFMPEG_VIDEO_INPUT &&
1123 input_mapping.buses[0].source_channel[0] == 0 &&
1124 input_mapping.buses[0].source_channel[1] == 1) {
1125 return input_mapping.buses[0].device.index + num_capture_cards;
1127 return numeric_limits<unsigned>::max();
1131 void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
1133 lock_guard<timed_mutex> lock(audio_mutex);
1134 set_input_mapping_lock_held(new_input_mapping);
1135 current_mapping_mode = MappingMode::MULTICHANNEL;
1138 AudioMixer::MappingMode AudioMixer::get_mapping_mode() const
1140 lock_guard<timed_mutex> lock(audio_mutex);
1141 return current_mapping_mode;
1144 void AudioMixer::set_input_mapping_lock_held(const InputMapping &new_input_mapping)
1146 map<DeviceSpec, set<unsigned>> interesting_channels;
1147 for (const InputMapping::Bus &bus : new_input_mapping.buses) {
1148 if (bus.device.type == InputSourceType::CAPTURE_CARD ||
1149 bus.device.type == InputSourceType::ALSA_INPUT ||
1150 bus.device.type == InputSourceType::FFMPEG_VIDEO_INPUT) {
1151 for (unsigned channel = 0; channel < 2; ++channel) {
1152 if (bus.source_channel[channel] != -1) {
1153 interesting_channels[bus.device].insert(bus.source_channel[channel]);
1157 assert(bus.device.type == InputSourceType::SILENCE);
1161 // Kill all the old metrics, and set up new ones.
1162 for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
1163 BusMetrics &metrics = bus_metrics[bus_index];
1165 vector<pair<string, string>> labels_left = metrics.labels;
1166 labels_left.emplace_back("channel", "left");
1167 vector<pair<string, string>> labels_right = metrics.labels;
1168 labels_right.emplace_back("channel", "right");
1170 global_metrics.remove("bus_current_level_dbfs", labels_left);
1171 global_metrics.remove("bus_current_level_dbfs", labels_right);
1172 global_metrics.remove("bus_peak_level_dbfs", labels_left);
1173 global_metrics.remove("bus_peak_level_dbfs", labels_right);
1174 global_metrics.remove("bus_historic_peak_dbfs", metrics.labels);
1175 global_metrics.remove("bus_gain_staging_db", metrics.labels);
1176 global_metrics.remove("bus_compressor_attenuation_db", metrics.labels);
1178 bus_metrics.reset(new BusMetrics[new_input_mapping.buses.size()]);
1179 for (unsigned bus_index = 0; bus_index < new_input_mapping.buses.size(); ++bus_index) {
1180 const InputMapping::Bus &bus = new_input_mapping.buses[bus_index];
1181 BusMetrics &metrics = bus_metrics[bus_index];
1183 char bus_index_str[16], source_index_str[16], source_channels_str[64];
1184 snprintf(bus_index_str, sizeof(bus_index_str), "%u", bus_index);
1185 snprintf(source_index_str, sizeof(source_index_str), "%u", bus.device.index);
1186 snprintf(source_channels_str, sizeof(source_channels_str), "%d:%d", bus.source_channel[0], bus.source_channel[1]);
1188 vector<pair<string, string>> labels;
1189 metrics.labels.emplace_back("index", bus_index_str);
1190 metrics.labels.emplace_back("name", bus.name);
1191 if (bus.device.type == InputSourceType::SILENCE) {
1192 metrics.labels.emplace_back("source_type", "silence");
1193 } else if (bus.device.type == InputSourceType::CAPTURE_CARD) {
1194 metrics.labels.emplace_back("source_type", "capture_card");
1195 } else if (bus.device.type == InputSourceType::ALSA_INPUT) {
1196 metrics.labels.emplace_back("source_type", "alsa_input");
1197 } else if (bus.device.type == InputSourceType::FFMPEG_VIDEO_INPUT) {
1198 metrics.labels.emplace_back("source_type", "ffmpeg_video_input");
1202 metrics.labels.emplace_back("source_index", source_index_str);
1203 metrics.labels.emplace_back("source_channels", source_channels_str);
1205 vector<pair<string, string>> labels_left = metrics.labels;
1206 labels_left.emplace_back("channel", "left");
1207 vector<pair<string, string>> labels_right = metrics.labels;
1208 labels_right.emplace_back("channel", "right");
1210 global_metrics.add("bus_current_level_dbfs", labels_left, &metrics.current_level_dbfs[0], Metrics::TYPE_GAUGE);
1211 global_metrics.add("bus_current_level_dbfs", labels_right, &metrics.current_level_dbfs[1], Metrics::TYPE_GAUGE);
1212 global_metrics.add("bus_peak_level_dbfs", labels_left, &metrics.peak_level_dbfs[0], Metrics::TYPE_GAUGE);
1213 global_metrics.add("bus_peak_level_dbfs", labels_right, &metrics.peak_level_dbfs[1], Metrics::TYPE_GAUGE);
1214 global_metrics.add("bus_historic_peak_dbfs", metrics.labels, &metrics.historic_peak_dbfs, Metrics::TYPE_GAUGE);
1215 global_metrics.add("bus_gain_staging_db", metrics.labels, &metrics.gain_staging_db, Metrics::TYPE_GAUGE);
1216 global_metrics.add("bus_compressor_attenuation_db", metrics.labels, &metrics.compressor_attenuation_db, Metrics::TYPE_GAUGE);
1219 // Reset resamplers for all cards that don't have the exact same state as before.
1220 map<DeviceSpec, double> new_extra_delay_ms = new_input_mapping.extra_delay_ms; // Convenience so we can use [].
1221 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
1222 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
1223 AudioDevice *device = find_audio_device(device_spec);
1224 if (device->interesting_channels != interesting_channels[device_spec] ||
1225 device->extra_delay_ms != new_extra_delay_ms[device_spec]) {
1226 device->interesting_channels = interesting_channels[device_spec];
1227 device->extra_delay_ms = new_extra_delay_ms[device_spec];
1228 reset_resampler_mutex_held(device_spec);
1231 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
1232 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
1233 AudioDevice *device = find_audio_device(device_spec);
1234 if (interesting_channels[device_spec].empty()) {
1235 alsa_pool.release_device(card_index);
1237 alsa_pool.hold_device(card_index);
1239 if (device->interesting_channels != interesting_channels[device_spec] ||
1240 device->extra_delay_ms != new_extra_delay_ms[device_spec]) {
1241 device->interesting_channels = interesting_channels[device_spec];
1242 device->extra_delay_ms = new_extra_delay_ms[device_spec];
1243 alsa_pool.reset_device(device_spec.index);
1244 reset_resampler_mutex_held(device_spec);
1247 for (unsigned card_index = 0; card_index < num_ffmpeg_inputs; ++card_index) {
1248 const DeviceSpec device_spec{InputSourceType::FFMPEG_VIDEO_INPUT, card_index};
1249 AudioDevice *device = find_audio_device(device_spec);
1250 if (device->interesting_channels != interesting_channels[device_spec] ||
1251 device->extra_delay_ms != new_extra_delay_ms[device_spec]) {
1252 device->interesting_channels = interesting_channels[device_spec];
1253 device->extra_delay_ms = new_extra_delay_ms[device_spec];
1254 reset_resampler_mutex_held(device_spec);
1258 input_mapping = new_input_mapping;
1261 InputMapping AudioMixer::get_input_mapping() const
1263 lock_guard<timed_mutex> lock(audio_mutex);
1264 return input_mapping;
1267 unsigned AudioMixer::num_buses() const
1269 lock_guard<timed_mutex> lock(audio_mutex);
1270 return input_mapping.buses.size();
1273 void AudioMixer::reset_peak(unsigned bus_index)
1275 lock_guard<timed_mutex> lock(audio_mutex);
1276 for (unsigned channel = 0; channel < 2; ++channel) {
1277 PeakHistory &history = peak_history[bus_index][channel];
1278 history.current_level = 0.0f;
1279 history.historic_peak = 0.0f;
1280 history.current_peak = 0.0f;
1281 history.last_peak = 0.0f;
1282 history.age_seconds = 0.0f;
1286 bool AudioMixer::is_mono(unsigned bus_index)
1288 lock_guard<timed_mutex> lock(audio_mutex);
1289 const InputMapping::Bus &bus = input_mapping.buses[bus_index];
1290 if (bus.device.type == InputSourceType::SILENCE) {
1293 assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
1294 bus.device.type == InputSourceType::ALSA_INPUT ||
1295 bus.device.type == InputSourceType::FFMPEG_VIDEO_INPUT);
1296 return bus.source_channel[0] == bus.source_channel[1];
1300 AudioMixer *global_audio_mixer = nullptr;