1 #include "audio_mixer.h"
4 #include <bmusb/bmusb.h>
14 #include <immintrin.h>
28 #include "alsa_pool.h"
29 #include "card_type.h"
34 #include "input_mapping.h"
35 #include "resampling_queue.h"
36 #include "shared/metrics.h"
37 #include "shared/shared_defs.h"
39 #include "stereocompressor.h"
41 using namespace bmusb;
43 using namespace std::chrono;
44 using namespace std::placeholders;
48 // TODO: If these prove to be a bottleneck, they can be SSSE3-optimized
49 // (usually including multiple channels at a time).
51 void convert_fixed16_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
52 const uint8_t *src, size_t in_channel, size_t in_num_channels,
55 assert(in_channel < in_num_channels);
56 assert(out_channel < out_num_channels);
57 src += in_channel * 2;
60 for (size_t i = 0; i < num_samples; ++i) {
61 int16_t s = le16toh(*(int16_t *)src);
62 *dst = s * (1.0f / 32768.0f);
64 src += 2 * in_num_channels;
65 dst += out_num_channels;
69 void convert_fixed16_to_fixed32(int32_t *dst, size_t out_channel, size_t out_num_channels,
70 const uint8_t *src, size_t in_channel, size_t in_num_channels,
73 assert(in_channel < in_num_channels);
74 assert(out_channel < out_num_channels);
75 src += in_channel * 2;
78 for (size_t i = 0; i < num_samples; ++i) {
79 uint32_t s = uint32_t(uint16_t(le16toh(*(int16_t *)src))) << 16;
81 // Keep the sign bit in place, repeat the other 15 bits as far as they go.
82 *dst = s | ((s & 0x7fffffff) >> 15) | ((s & 0x7fffffff) >> 30);
84 src += 2 * in_num_channels;
85 dst += out_num_channels;
89 void convert_fixed24_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
90 const uint8_t *src, size_t in_channel, size_t in_num_channels,
93 assert(in_channel < in_num_channels);
94 assert(out_channel < out_num_channels);
95 src += in_channel * 3;
98 for (size_t i = 0; i < num_samples; ++i) {
100 uint32_t s2 = src[1];
101 uint32_t s3 = src[2];
102 uint32_t s = (s1 << 8) | (s2 << 16) | (s3 << 24); // Note: The bottom eight bits are zero; s3 includes the sign bit.
103 *dst = int(s) * (1.0f / (256.0f * 8388608.0f)); // 256 for signed down-shift by 8, then 2^23 for the actual conversion.
105 src += 3 * in_num_channels;
106 dst += out_num_channels;
110 void convert_fixed24_to_fixed32(int32_t *dst, size_t out_channel, size_t out_num_channels,
111 const uint8_t *src, size_t in_channel, size_t in_num_channels,
114 assert(in_channel < in_num_channels);
115 assert(out_channel < out_num_channels);
116 src += in_channel * 3;
119 for (size_t i = 0; i < num_samples; ++i) {
120 uint32_t s1 = src[0];
121 uint32_t s2 = src[1];
122 uint32_t s3 = src[2];
123 uint32_t s = (s1 << 8) | (s2 << 16) | (s3 << 24);
125 // Keep the sign bit in place, repeat the other 23 bits as far as they go.
126 *dst = s | ((s & 0x7fffffff) >> 23);
128 src += 3 * in_num_channels;
129 dst += out_num_channels;
133 void convert_fixed32_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
134 const uint8_t *src, size_t in_channel, size_t in_num_channels,
137 assert(in_channel < in_num_channels);
138 assert(out_channel < out_num_channels);
139 src += in_channel * 4;
142 for (size_t i = 0; i < num_samples; ++i) {
143 int32_t s = le32toh(*(int32_t *)src);
144 *dst = s * (1.0f / 2147483648.0f);
146 src += 4 * in_num_channels;
147 dst += out_num_channels;
151 // Basically just a reinterleave.
152 void convert_fixed32_to_fixed32(int32_t *dst, size_t out_channel, size_t out_num_channels,
153 const uint8_t *src, size_t in_channel, size_t in_num_channels,
156 assert(in_channel < in_num_channels);
157 assert(out_channel < out_num_channels);
158 src += in_channel * 4;
161 for (size_t i = 0; i < num_samples; ++i) {
162 int32_t s = le32toh(*(int32_t *)src);
165 src += 4 * in_num_channels;
166 dst += out_num_channels;
170 float find_peak_plain(const float *samples, size_t num_samples) __attribute__((unused));
172 float find_peak_plain(const float *samples, size_t num_samples)
174 float m = fabs(samples[0]);
175 for (size_t i = 1; i < num_samples; ++i) {
176 m = max(m, fabs(samples[i]));
182 static inline float horizontal_max(__m128 m)
184 __m128 tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 0, 3, 2));
185 m = _mm_max_ps(m, tmp);
186 tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 3, 0, 1));
187 m = _mm_max_ps(m, tmp);
188 return _mm_cvtss_f32(m);
191 float find_peak(const float *samples, size_t num_samples)
193 const __m128 abs_mask = _mm_castsi128_ps(_mm_set1_epi32(0x7fffffffu));
194 __m128 m = _mm_setzero_ps();
195 for (size_t i = 0; i < (num_samples & ~3); i += 4) {
196 __m128 x = _mm_loadu_ps(samples + i);
197 x = _mm_and_ps(x, abs_mask);
198 m = _mm_max_ps(m, x);
200 float result = horizontal_max(m);
202 for (size_t i = (num_samples & ~3); i < num_samples; ++i) {
203 result = max(result, fabs(samples[i]));
207 // Self-test. We should be bit-exact the same.
208 float reference_result = find_peak_plain(samples, num_samples);
209 if (result != reference_result) {
210 fprintf(stderr, "Error: Peak is %f [%f %f %f %f]; should be %f.\n",
212 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(0, 0, 0, 0))),
213 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 1, 1, 1))),
214 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 2, 2, 2))),
215 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(3, 3, 3, 3))),
223 float find_peak(const float *samples, size_t num_samples)
225 return find_peak_plain(samples, num_samples);
229 void deinterleave_samples(const vector<float> &in, vector<float> *out_l, vector<float> *out_r)
231 size_t num_samples = in.size() / 2;
232 out_l->resize(num_samples);
233 out_r->resize(num_samples);
235 const float *inptr = in.data();
236 float *lptr = &(*out_l)[0];
237 float *rptr = &(*out_r)[0];
238 for (size_t i = 0; i < num_samples; ++i) {
246 AudioMixer::AudioMixer()
247 : limiter(OUTPUT_FREQUENCY),
248 correlation(OUTPUT_FREQUENCY)
250 for (unsigned bus_index = 0; bus_index < MAX_BUSES; ++bus_index) {
251 locut[bus_index].init(FILTER_HPF, 2);
252 eq[bus_index][EQ_BAND_BASS].init(FILTER_LOW_SHELF, 1);
253 // Note: EQ_BAND_MID isn't used (see comments in apply_eq()).
254 eq[bus_index][EQ_BAND_TREBLE].init(FILTER_HIGH_SHELF, 1);
255 compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
256 level_compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
258 set_bus_settings(bus_index, get_default_bus_settings());
260 set_limiter_enabled(global_flags.limiter_enabled);
261 set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
263 r128.init(2, OUTPUT_FREQUENCY);
266 // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
267 // and there's a limit to how important the peak meter is.
268 peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
270 global_audio_mixer = this;
273 if (!global_flags.input_mapping_filename.empty()) {
274 // Must happen after ALSAPool is initialized, as it needs to know the card list.
275 current_mapping_mode = MappingMode::MULTICHANNEL;
276 InputMapping new_input_mapping;
277 if (!load_input_mapping_from_file(get_devices(),
278 global_flags.input_mapping_filename,
279 &new_input_mapping)) {
280 fprintf(stderr, "Failed to load input mapping from '%s', exiting.\n",
281 global_flags.input_mapping_filename.c_str());
284 set_input_mapping(new_input_mapping);
286 set_simple_input(/*card_index=*/0);
287 if (global_flags.multichannel_mapping_mode) {
288 current_mapping_mode = MappingMode::MULTICHANNEL;
292 global_metrics.add("audio_loudness_short_lufs", &metric_audio_loudness_short_lufs, Metrics::TYPE_GAUGE);
293 global_metrics.add("audio_loudness_integrated_lufs", &metric_audio_loudness_integrated_lufs, Metrics::TYPE_GAUGE);
294 global_metrics.add("audio_loudness_range_low_lufs", &metric_audio_loudness_range_low_lufs, Metrics::TYPE_GAUGE);
295 global_metrics.add("audio_loudness_range_high_lufs", &metric_audio_loudness_range_high_lufs, Metrics::TYPE_GAUGE);
296 global_metrics.add("audio_peak_dbfs", &metric_audio_peak_dbfs, Metrics::TYPE_GAUGE);
297 global_metrics.add("audio_final_makeup_gain_db", &metric_audio_final_makeup_gain_db, Metrics::TYPE_GAUGE);
298 global_metrics.add("audio_correlation", &metric_audio_correlation, Metrics::TYPE_GAUGE);
301 void AudioMixer::reset_resampler(DeviceSpec device_spec)
303 lock_guard<timed_mutex> lock(audio_mutex);
304 reset_resampler_mutex_held(device_spec);
307 void AudioMixer::reset_resampler_mutex_held(DeviceSpec device_spec)
309 AudioDevice *device = find_audio_device(device_spec);
311 if (device->interesting_channels.empty()) {
312 device->resampling_queue.reset();
314 device->resampling_queue.reset(new ResamplingQueue(
315 spec_to_string(device_spec), device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size(),
316 global_flags.audio_queue_length_ms * 0.001));
320 bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, steady_clock::time_point frame_time)
322 AudioDevice *device = find_audio_device(device_spec);
324 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
325 if (!lock.try_lock_for(chrono::milliseconds(10))) {
328 if (device->resampling_queue == nullptr) {
329 // No buses use this device; throw it away.
333 unsigned num_channels = device->interesting_channels.size();
334 if (num_channels == 0) {
335 // No buses use this device; throw it away. (Normally, we should not
336 // be here, but probably, we are in the process of changing a mapping,
337 // and the queue just isn't gone yet. In any case, returning is harmless.)
341 // Convert the audio to fp32.
342 unique_ptr<float[]> audio(new float[num_samples * num_channels]);
343 unsigned channel_index = 0;
344 for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) {
345 switch (audio_format.bits_per_sample) {
347 assert(num_samples == 0);
350 convert_fixed16_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
353 convert_fixed24_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
356 convert_fixed32_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
359 fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
364 // If we changed frequency since last frame, we'll need to reset the resampler.
365 if (audio_format.sample_rate != device->capture_frequency) {
366 device->capture_frequency = audio_format.sample_rate;
367 reset_resampler_mutex_held(device_spec);
371 device->resampling_queue->add_input_samples(frame_time, audio.get(), num_samples, ResamplingQueue::ADJUST_RATE);
375 vector<int32_t> convert_audio_to_fixed32(const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, unsigned num_channels)
377 vector<int32_t> audio;
379 if (num_channels > audio_format.num_channels) {
380 audio.resize(num_samples * num_channels, 0);
382 audio.resize(num_samples * num_channels);
384 for (unsigned channel_index = 0; channel_index < num_channels && channel_index < audio_format.num_channels; ++channel_index) {
385 switch (audio_format.bits_per_sample) {
387 assert(num_samples == 0);
390 convert_fixed16_to_fixed32(&audio[0], channel_index, num_channels, data, channel_index, audio_format.num_channels, num_samples);
393 convert_fixed24_to_fixed32(&audio[0], channel_index, num_channels, data, channel_index, audio_format.num_channels, num_samples);
396 convert_fixed32_to_fixed32(&audio[0], channel_index, num_channels, data, channel_index, audio_format.num_channels, num_samples);
399 fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
407 bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames)
409 AudioDevice *device = find_audio_device(device_spec);
411 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
412 if (!lock.try_lock_for(chrono::milliseconds(10))) {
415 if (device->resampling_queue == nullptr) {
416 // No buses use this device; throw it away.
420 unsigned num_channels = device->interesting_channels.size();
421 assert(num_channels > 0);
423 vector<float> silence(samples_per_frame * num_channels, 0.0f);
424 for (unsigned i = 0; i < num_frames; ++i) {
425 device->resampling_queue->add_input_samples(steady_clock::now(), silence.data(), samples_per_frame, ResamplingQueue::DO_NOT_ADJUST_RATE);
430 bool AudioMixer::silence_card(DeviceSpec device_spec, bool silence)
432 AudioDevice *device = find_audio_device(device_spec);
434 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
435 if (!lock.try_lock_for(chrono::milliseconds(10))) {
439 if (device->silenced && !silence) {
440 reset_resampler_mutex_held(device_spec);
442 device->silenced = silence;
446 AudioMixer::BusSettings AudioMixer::get_default_bus_settings()
448 BusSettings settings;
449 settings.fader_volume_db = 0.0f;
450 settings.muted = false;
451 settings.locut_enabled = global_flags.locut_enabled;
452 settings.stereo_width = 1.0f;
453 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
454 settings.eq_level_db[band_index] = 0.0f;
456 settings.gain_staging_db = global_flags.initial_gain_staging_db;
457 settings.level_compressor_enabled = global_flags.gain_staging_auto;
458 settings.compressor_threshold_dbfs = ref_level_dbfs - 12.0f; // -12 dB.
459 settings.compressor_enabled = global_flags.compressor_enabled;
463 AudioMixer::BusSettings AudioMixer::get_bus_settings(unsigned bus_index) const
465 lock_guard<timed_mutex> lock(audio_mutex);
466 BusSettings settings;
467 settings.fader_volume_db = fader_volume_db[bus_index];
468 settings.muted = mute[bus_index];
469 settings.locut_enabled = locut_enabled[bus_index];
470 settings.stereo_width = stereo_width[bus_index];
471 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
472 settings.eq_level_db[band_index] = eq_level_db[bus_index][band_index];
474 settings.gain_staging_db = gain_staging_db[bus_index];
475 settings.level_compressor_enabled = level_compressor_enabled[bus_index];
476 settings.compressor_threshold_dbfs = compressor_threshold_dbfs[bus_index];
477 settings.compressor_enabled = compressor_enabled[bus_index];
481 void AudioMixer::set_bus_settings(unsigned bus_index, const AudioMixer::BusSettings &settings)
483 lock_guard<timed_mutex> lock(audio_mutex);
484 fader_volume_db[bus_index] = settings.fader_volume_db;
485 mute[bus_index] = settings.muted;
486 locut_enabled[bus_index] = settings.locut_enabled;
487 stereo_width[bus_index] = settings.stereo_width;
488 for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
489 eq_level_db[bus_index][band_index] = settings.eq_level_db[band_index];
491 gain_staging_db[bus_index] = settings.gain_staging_db;
492 last_gain_staging_db[bus_index] = gain_staging_db[bus_index];
493 level_compressor_enabled[bus_index] = settings.level_compressor_enabled;
494 compressor_threshold_dbfs[bus_index] = settings.compressor_threshold_dbfs;
495 compressor_enabled[bus_index] = settings.compressor_enabled;
498 AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
500 switch (device.type) {
501 case InputSourceType::CAPTURE_CARD:
502 return &video_cards[device.index];
503 case InputSourceType::ALSA_INPUT:
504 return &alsa_inputs[device.index];
505 case InputSourceType::SILENCE:
512 // Get a pointer to the given channel from the given device.
513 // The channel must be picked out earlier and resampled.
514 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)
516 static float zero = 0.0f;
517 if (source_channel == -1 || device_spec.type == InputSourceType::SILENCE) {
522 AudioDevice *device = find_audio_device(device_spec);
523 assert(device->interesting_channels.count(source_channel) != 0);
524 unsigned channel_index = 0;
525 for (int channel : device->interesting_channels) {
526 if (channel == source_channel) break;
529 assert(channel_index < device->interesting_channels.size());
530 const auto it = samples_card.find(device_spec);
531 assert(it != samples_card.end());
532 *srcptr = &(it->second)[channel_index];
533 *stride = device->interesting_channels.size();
536 // TODO: Can be SSSE3-optimized if need be.
537 void AudioMixer::fill_audio_bus(const map<DeviceSpec, vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float stereo_width, float *output)
539 if (bus.device.type == InputSourceType::SILENCE) {
540 memset(output, 0, num_samples * 2 * sizeof(*output));
542 assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
543 bus.device.type == InputSourceType::ALSA_INPUT);
544 const float *lsrc, *rsrc;
545 unsigned lstride, rstride;
546 float *dptr = output;
547 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[0], &lsrc, &lstride);
548 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[1], &rsrc, &rstride);
550 // Apply stereo width settings. Set stereo width w to a 0..1 range instead of
551 // -1..1, since it makes for much easier calculations (so 0.5 = completely mono).
552 // Then, what we want is
554 // L' = wL + (1-w)R = R + w(L-R)
555 // R' = wR + (1-w)L = L + w(R-L)
557 // This can be further simplified calculation-wise by defining the weighted
558 // difference signal D = w(R-L), so that:
562 float w = 0.5f * stereo_width + 0.5f;
563 if (bus.source_channel[0] == bus.source_channel[1]) {
564 // Mono anyway, so no need to bother.
566 } else if (fabs(w) < 1e-3) {
569 swap(lstride, rstride);
572 if (fabs(w - 1.0f) < 1e-3) {
573 // No calculations needed for stereo_width = 1.
574 for (unsigned i = 0; i < num_samples; ++i) {
582 for (unsigned i = 0; i < num_samples; ++i) {
583 float left = *lsrc, right = *rsrc;
584 float diff = w * (right - left);
585 *dptr++ = right - diff;
586 *dptr++ = left + diff;
594 vector<DeviceSpec> AudioMixer::get_active_devices() const
596 vector<DeviceSpec> ret;
597 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
598 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
599 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
600 ret.push_back(device_spec);
603 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
604 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
605 if (!find_audio_device(device_spec)->interesting_channels.empty()) {
606 ret.push_back(device_spec);
614 void apply_gain(float db, float last_db, vector<float> *samples)
616 if (fabs(db - last_db) < 1e-3) {
617 // Constant over this frame.
618 const float gain = from_db(db);
619 for (size_t i = 0; i < samples->size(); ++i) {
620 (*samples)[i] *= gain;
623 // We need to do a fade.
624 unsigned num_samples = samples->size() / 2;
625 float gain = from_db(last_db);
626 const float gain_inc = pow(from_db(db - last_db), 1.0 / num_samples);
627 for (size_t i = 0; i < num_samples; ++i) {
628 (*samples)[i * 2 + 0] *= gain;
629 (*samples)[i * 2 + 1] *= gain;
637 vector<float> AudioMixer::get_output(steady_clock::time_point ts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
639 map<DeviceSpec, vector<float>> samples_card;
640 vector<float> samples_bus;
642 lock_guard<timed_mutex> lock(audio_mutex);
644 // Pick out all the interesting channels from all the cards.
645 for (const DeviceSpec &device_spec : get_active_devices()) {
646 AudioDevice *device = find_audio_device(device_spec);
647 samples_card[device_spec].resize(num_samples * device->interesting_channels.size());
648 if (device->silenced) {
649 memset(&samples_card[device_spec][0], 0, samples_card[device_spec].size() * sizeof(float));
651 device->resampling_queue->get_output_samples(
653 &samples_card[device_spec][0],
655 rate_adjustment_policy);
659 vector<float> samples_out, left, right;
660 samples_out.resize(num_samples * 2);
661 samples_bus.resize(num_samples * 2);
662 for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
663 fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, stereo_width[bus_index], &samples_bus[0]);
664 apply_eq(bus_index, &samples_bus);
667 lock_guard<mutex> lock(compressor_mutex);
669 // Apply a level compressor to get the general level right.
670 // Basically, if it's over about -40 dBFS, we squeeze it down to that level
671 // (or more precisely, near it, since we don't use infinite ratio),
672 // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
673 // entirely arbitrary, but from practical tests with speech, it seems to
674 // put ut around -23 LUFS, so it's a reasonable starting point for later use.
675 if (level_compressor_enabled[bus_index]) {
676 float threshold = 0.01f; // -40 dBFS.
678 float attack_time = 0.5f;
679 float release_time = 20.0f;
680 float makeup_gain = from_db(ref_level_dbfs - (-40.0f)); // +26 dB.
681 level_compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
682 gain_staging_db[bus_index] = to_db(level_compressor[bus_index]->get_attenuation() * makeup_gain);
684 // Just apply the gain we already had.
685 float db = gain_staging_db[bus_index];
686 float last_db = last_gain_staging_db[bus_index];
687 apply_gain(db, last_db, &samples_bus);
689 last_gain_staging_db[bus_index] = gain_staging_db[bus_index];
692 printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
693 level_compressor.get_level(), to_db(level_compressor.get_level()),
694 level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()),
695 to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
698 // The real compressor.
699 if (compressor_enabled[bus_index]) {
700 float threshold = from_db(compressor_threshold_dbfs[bus_index]);
702 float attack_time = 0.005f;
703 float release_time = 0.040f;
704 float makeup_gain = 2.0f; // +6 dB.
705 compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
706 // compressor_att = compressor.get_attenuation();
710 add_bus_to_master(bus_index, samples_bus, &samples_out);
711 deinterleave_samples(samples_bus, &left, &right);
712 measure_bus_levels(bus_index, left, right);
716 lock_guard<mutex> lock(compressor_mutex);
718 // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
719 // Note that since ratio is not infinite, we could go slightly higher than this.
720 if (limiter_enabled) {
721 float threshold = from_db(limiter_threshold_dbfs);
723 float attack_time = 0.0f; // Instant.
724 float release_time = 0.020f;
725 float makeup_gain = 1.0f; // 0 dB.
726 limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
727 // limiter_att = limiter.get_attenuation();
730 // printf("limiter=%+5.1f compressor=%+5.1f\n", to_db(limiter_att), to_db(compressor_att));
733 // At this point, we are most likely close to +0 LU (at least if the
734 // faders sum to 0 dB and the compressors are on), but all of our
735 // measurements have been on raw sample values, not R128 values.
736 // So we have a final makeup gain to get us to +0 LU; the gain
737 // adjustments required should be relatively small, and also, the
738 // offset shouldn't change much (only if the type of audio changes
739 // significantly). Thus, we shoot for updating this value basically
740 // “whenever we process buffers”, since the R128 calculation isn't exactly
741 // something we get out per-sample.
743 // Note that there's a feedback loop here, so we choose a very slow filter
744 // (half-time of 30 seconds).
745 double target_loudness_factor, alpha;
746 double loudness_lu = r128.loudness_M() - ref_level_lufs;
747 target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
749 // If we're outside +/- 5 LU (after correction), we don't count it as
750 // a normal signal (probably silence) and don't change the
751 // correction factor; just apply what we already have.
752 if (fabs(loudness_lu) >= 5.0 || !final_makeup_gain_auto) {
755 // Formula adapted from
756 // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
757 const double half_time_s = 30.0;
758 const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
759 alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
763 lock_guard<mutex> lock(compressor_mutex);
764 double m = final_makeup_gain;
765 for (size_t i = 0; i < samples_out.size(); i += 2) {
766 samples_out[i + 0] *= m;
767 samples_out[i + 1] *= m;
768 m += (target_loudness_factor - m) * alpha;
770 final_makeup_gain = m;
773 update_meters(samples_out);
780 void apply_filter_fade(StereoFilter *filter, float *data, unsigned num_samples, float cutoff_hz, float db, float last_db)
782 // A granularity of 32 samples is an okay tradeoff between speed and
783 // smoothness; recalculating the filters is pretty expensive, so it's
784 // good that we don't do this all the time.
785 static constexpr unsigned filter_granularity_samples = 32;
787 const float cutoff_linear = cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY;
788 if (fabs(db - last_db) < 1e-3) {
789 // Constant over this frame.
790 if (fabs(db) > 0.01f) {
791 filter->render(data, num_samples, cutoff_linear, 0.5f, db / 40.0f);
794 // We need to do a fade. (Rounding up avoids division by zero.)
795 unsigned num_blocks = (num_samples + filter_granularity_samples - 1) / filter_granularity_samples;
796 const float inc_db_norm = (db - last_db) / 40.0f / num_blocks;
797 float db_norm = db / 40.0f;
798 for (size_t i = 0; i < num_samples; i += filter_granularity_samples) {
799 size_t samples_this_block = std::min<size_t>(num_samples - i, filter_granularity_samples);
800 filter->render(data + i * 2, samples_this_block, cutoff_linear, 0.5f, db_norm);
801 db_norm += inc_db_norm;
808 void AudioMixer::apply_eq(unsigned bus_index, vector<float> *samples_bus)
810 constexpr float bass_freq_hz = 200.0f;
811 constexpr float treble_freq_hz = 4700.0f;
813 // Cut away everything under 120 Hz (or whatever the cutoff is);
814 // we don't need it for voice, and it will reduce headroom
815 // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
816 // should be dampened.)
817 if (locut_enabled[bus_index]) {
818 locut[bus_index].render(samples_bus->data(), samples_bus->size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
821 // Apply the rest of the EQ. Since we only have a simple three-band EQ,
822 // we can implement it with two shelf filters. We use a simple gain to
823 // set the mid-level filter, and then offset the low and high bands
824 // from that if we need to. (We could perhaps have folded the gain into
825 // the next part, but it's so cheap that the trouble isn't worth it.)
827 // If any part of the EQ has changed appreciably since last frame,
828 // we fade smoothly during the course of this frame.
829 const float bass_db = eq_level_db[bus_index][EQ_BAND_BASS];
830 const float mid_db = eq_level_db[bus_index][EQ_BAND_MID];
831 const float treble_db = eq_level_db[bus_index][EQ_BAND_TREBLE];
833 const float last_bass_db = last_eq_level_db[bus_index][EQ_BAND_BASS];
834 const float last_mid_db = last_eq_level_db[bus_index][EQ_BAND_MID];
835 const float last_treble_db = last_eq_level_db[bus_index][EQ_BAND_TREBLE];
837 assert(samples_bus->size() % 2 == 0);
838 const unsigned num_samples = samples_bus->size() / 2;
840 apply_gain(mid_db, last_mid_db, samples_bus);
842 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);
843 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);
845 last_eq_level_db[bus_index][EQ_BAND_BASS] = bass_db;
846 last_eq_level_db[bus_index][EQ_BAND_MID] = mid_db;
847 last_eq_level_db[bus_index][EQ_BAND_TREBLE] = treble_db;
850 void AudioMixer::add_bus_to_master(unsigned bus_index, const vector<float> &samples_bus, vector<float> *samples_out)
852 assert(samples_bus.size() == samples_out->size());
853 assert(samples_bus.size() % 2 == 0);
854 unsigned num_samples = samples_bus.size() / 2;
855 const float new_volume_db = mute[bus_index] ? -90.0f : fader_volume_db[bus_index].load();
856 if (fabs(new_volume_db - last_fader_volume_db[bus_index]) > 1e-3) {
857 // The volume has changed; do a fade over the course of this frame.
858 // (We might have some numerical issues here, but it seems to sound OK.)
859 // For the purpose of fading here, the silence floor is set to -90 dB
860 // (the fader only goes to -84).
861 float old_volume = from_db(max<float>(last_fader_volume_db[bus_index], -90.0f));
862 float volume = from_db(max<float>(new_volume_db, -90.0f));
864 float volume_inc = pow(volume / old_volume, 1.0 / num_samples);
866 if (bus_index == 0) {
867 for (unsigned i = 0; i < num_samples; ++i) {
868 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
869 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
870 volume *= volume_inc;
873 for (unsigned i = 0; i < num_samples; ++i) {
874 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
875 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
876 volume *= volume_inc;
879 } else if (new_volume_db > -90.0f) {
880 float volume = from_db(new_volume_db);
881 if (bus_index == 0) {
882 for (unsigned i = 0; i < num_samples; ++i) {
883 (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
884 (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
887 for (unsigned i = 0; i < num_samples; ++i) {
888 (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
889 (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
894 last_fader_volume_db[bus_index] = new_volume_db;
897 void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right)
899 assert(left.size() == right.size());
900 const float volume = mute[bus_index] ? 0.0f : from_db(fader_volume_db[bus_index]);
901 const float peak_levels[2] = {
902 find_peak(left.data(), left.size()) * volume,
903 find_peak(right.data(), right.size()) * volume
905 for (unsigned channel = 0; channel < 2; ++channel) {
906 // Compute the current value, including hold and falloff.
907 // The constants are borrowed from zita-mu1 by Fons Adriaensen.
908 static constexpr float hold_sec = 0.5f;
909 static constexpr float falloff_db_sec = 15.0f; // dB/sec falloff after hold.
911 PeakHistory &history = peak_history[bus_index][channel];
912 history.historic_peak = max(history.historic_peak, peak_levels[channel]);
913 if (history.age_seconds < hold_sec) {
914 current_peak = history.last_peak;
916 current_peak = history.last_peak * from_db(-falloff_db_sec * (history.age_seconds - hold_sec));
919 // See if we have a new peak to replace the old (possibly falling) one.
920 if (peak_levels[channel] > current_peak) {
921 history.last_peak = peak_levels[channel];
922 history.age_seconds = 0.0f; // Not 100% correct, but more than good enough given our frame sizes.
923 current_peak = peak_levels[channel];
925 history.age_seconds += float(left.size()) / OUTPUT_FREQUENCY;
927 history.current_level = peak_levels[channel];
928 history.current_peak = current_peak;
932 void AudioMixer::update_meters(const vector<float> &samples)
934 // Upsample 4x to find interpolated peak.
935 peak_resampler.inp_data = const_cast<float *>(samples.data());
936 peak_resampler.inp_count = samples.size() / 2;
938 vector<float> interpolated_samples;
939 interpolated_samples.resize(samples.size());
941 lock_guard<mutex> lock(audio_measure_mutex);
943 while (peak_resampler.inp_count > 0) { // About four iterations.
944 peak_resampler.out_data = &interpolated_samples[0];
945 peak_resampler.out_count = interpolated_samples.size() / 2;
946 peak_resampler.process();
947 size_t out_stereo_samples = interpolated_samples.size() / 2 - peak_resampler.out_count;
948 peak = max<float>(peak, find_peak(interpolated_samples.data(), out_stereo_samples * 2));
949 peak_resampler.out_data = nullptr;
953 // Find R128 levels and L/R correlation.
954 vector<float> left, right;
955 deinterleave_samples(samples, &left, &right);
956 float *ptrs[] = { left.data(), right.data() };
958 lock_guard<mutex> lock(audio_measure_mutex);
959 r128.process(left.size(), ptrs);
960 correlation.process_samples(samples);
963 send_audio_level_callback();
966 void AudioMixer::reset_meters()
968 lock_guard<mutex> lock(audio_measure_mutex);
969 peak_resampler.reset();
976 void AudioMixer::send_audio_level_callback()
978 if (audio_level_callback == nullptr) {
982 lock_guard<mutex> lock(audio_measure_mutex);
983 double loudness_s = r128.loudness_S();
984 double loudness_i = r128.integrated();
985 double loudness_range_low = r128.range_min();
986 double loudness_range_high = r128.range_max();
988 metric_audio_loudness_short_lufs = loudness_s;
989 metric_audio_loudness_integrated_lufs = loudness_i;
990 metric_audio_loudness_range_low_lufs = loudness_range_low;
991 metric_audio_loudness_range_high_lufs = loudness_range_high;
992 metric_audio_peak_dbfs = to_db(peak);
993 metric_audio_final_makeup_gain_db = to_db(final_makeup_gain);
994 metric_audio_correlation = correlation.get_correlation();
996 vector<BusLevel> bus_levels;
997 bus_levels.resize(input_mapping.buses.size());
999 lock_guard<mutex> lock(compressor_mutex);
1000 for (unsigned bus_index = 0; bus_index < bus_levels.size(); ++bus_index) {
1001 BusLevel &levels = bus_levels[bus_index];
1002 BusMetrics &metrics = bus_metrics[bus_index];
1004 levels.current_level_dbfs[0] = metrics.current_level_dbfs[0] = to_db(peak_history[bus_index][0].current_level);
1005 levels.current_level_dbfs[1] = metrics.current_level_dbfs[1] = to_db(peak_history[bus_index][1].current_level);
1006 levels.peak_level_dbfs[0] = metrics.peak_level_dbfs[0] = to_db(peak_history[bus_index][0].current_peak);
1007 levels.peak_level_dbfs[1] = metrics.peak_level_dbfs[1] = to_db(peak_history[bus_index][1].current_peak);
1008 levels.historic_peak_dbfs = metrics.historic_peak_dbfs = to_db(
1009 max(peak_history[bus_index][0].historic_peak,
1010 peak_history[bus_index][1].historic_peak));
1011 levels.gain_staging_db = metrics.gain_staging_db = gain_staging_db[bus_index];
1012 if (compressor_enabled[bus_index]) {
1013 levels.compressor_attenuation_db = metrics.compressor_attenuation_db = -to_db(compressor[bus_index]->get_attenuation());
1015 levels.compressor_attenuation_db = 0.0;
1016 metrics.compressor_attenuation_db = 0.0 / 0.0;
1021 audio_level_callback(loudness_s, to_db(peak), bus_levels,
1022 loudness_i, loudness_range_low, loudness_range_high,
1023 to_db(final_makeup_gain),
1024 correlation.get_correlation());
1027 map<DeviceSpec, DeviceInfo> AudioMixer::get_devices()
1029 lock_guard<timed_mutex> lock(audio_mutex);
1031 map<DeviceSpec, DeviceInfo> devices;
1032 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
1033 const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index };
1034 const AudioDevice *device = &video_cards[card_index];
1036 info.display_name = device->display_name;
1037 info.num_channels = device->num_channels;
1038 devices.insert(make_pair(spec, info));
1040 vector<ALSAPool::Device> available_alsa_devices = alsa_pool.get_devices();
1041 for (unsigned card_index = 0; card_index < available_alsa_devices.size(); ++card_index) {
1042 const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index };
1043 const ALSAPool::Device &device = available_alsa_devices[card_index];
1045 info.display_name = device.display_name();
1046 info.num_channels = device.num_channels;
1047 info.alsa_name = device.name;
1048 info.alsa_info = device.info;
1049 info.alsa_address = device.address;
1050 devices.insert(make_pair(spec, info));
1055 void AudioMixer::set_device_parameters(DeviceSpec device_spec, const std::string &display_name, CardType card_type, unsigned num_channels, bool active)
1057 AudioDevice *device = find_audio_device(device_spec);
1059 lock_guard<timed_mutex> lock(audio_mutex);
1060 if (active || device->display_name.empty()) {
1061 device->display_name = display_name;
1063 device->card_type = card_type;
1064 device->active = active;
1067 bool AudioMixer::get_active(DeviceSpec device_spec)
1069 AudioDevice *device = find_audio_device(device_spec);
1071 lock_guard<timed_mutex> lock(audio_mutex);
1072 return device->active;
1075 void AudioMixer::serialize_device(DeviceSpec device_spec, DeviceSpecProto *device_spec_proto)
1077 lock_guard<timed_mutex> lock(audio_mutex);
1078 switch (device_spec.type) {
1079 case InputSourceType::SILENCE:
1080 device_spec_proto->set_type(DeviceSpecProto::SILENCE);
1082 case InputSourceType::CAPTURE_CARD:
1083 device_spec_proto->set_type(DeviceSpecProto::CAPTURE_CARD);
1084 device_spec_proto->set_index(device_spec.index);
1085 device_spec_proto->set_display_name(video_cards[device_spec.index].display_name);
1087 case InputSourceType::ALSA_INPUT:
1088 alsa_pool.serialize_device(device_spec.index, device_spec_proto);
1093 void AudioMixer::set_simple_input(unsigned card_index)
1095 assert(card_index < MAX_VIDEO_CARDS);
1096 InputMapping new_input_mapping;
1097 InputMapping::Bus input;
1098 input.name = "Main";
1099 input.device = DeviceSpec{InputSourceType::CAPTURE_CARD, card_index};
1100 input.source_channel[0] = 0;
1101 input.source_channel[1] = 1;
1103 new_input_mapping.buses.push_back(input);
1105 lock_guard<timed_mutex> lock(audio_mutex);
1106 current_mapping_mode = MappingMode::SIMPLE;
1107 set_input_mapping_lock_held(new_input_mapping);
1108 fader_volume_db[0] = 0.0f;
1111 unsigned AudioMixer::get_simple_input() const
1113 lock_guard<timed_mutex> lock(audio_mutex);
1114 if (input_mapping.buses.size() == 1 &&
1115 input_mapping.buses[0].device.type == InputSourceType::CAPTURE_CARD &&
1116 input_mapping.buses[0].source_channel[0] == 0 &&
1117 input_mapping.buses[0].source_channel[1] == 1) {
1118 return input_mapping.buses[0].device.index;
1120 return numeric_limits<unsigned>::max();
1124 void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
1126 lock_guard<timed_mutex> lock(audio_mutex);
1127 set_input_mapping_lock_held(new_input_mapping);
1128 current_mapping_mode = MappingMode::MULTICHANNEL;
1131 AudioMixer::MappingMode AudioMixer::get_mapping_mode() const
1133 lock_guard<timed_mutex> lock(audio_mutex);
1134 return current_mapping_mode;
1137 void AudioMixer::set_input_mapping_lock_held(const InputMapping &new_input_mapping)
1139 map<DeviceSpec, set<unsigned>> interesting_channels;
1140 for (const InputMapping::Bus &bus : new_input_mapping.buses) {
1141 if (bus.device.type == InputSourceType::CAPTURE_CARD ||
1142 bus.device.type == InputSourceType::ALSA_INPUT) {
1143 for (unsigned channel = 0; channel < 2; ++channel) {
1144 if (bus.source_channel[channel] != -1) {
1145 interesting_channels[bus.device].insert(bus.source_channel[channel]);
1149 assert(bus.device.type == InputSourceType::SILENCE);
1153 // Kill all the old metrics, and set up new ones.
1154 for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
1155 BusMetrics &metrics = bus_metrics[bus_index];
1157 vector<pair<string, string>> labels_left = metrics.labels;
1158 labels_left.emplace_back("channel", "left");
1159 vector<pair<string, string>> labels_right = metrics.labels;
1160 labels_right.emplace_back("channel", "right");
1162 global_metrics.remove("bus_current_level_dbfs", labels_left);
1163 global_metrics.remove("bus_current_level_dbfs", labels_right);
1164 global_metrics.remove("bus_peak_level_dbfs", labels_left);
1165 global_metrics.remove("bus_peak_level_dbfs", labels_right);
1166 global_metrics.remove("bus_historic_peak_dbfs", metrics.labels);
1167 global_metrics.remove("bus_gain_staging_db", metrics.labels);
1168 global_metrics.remove("bus_compressor_attenuation_db", metrics.labels);
1170 bus_metrics.reset(new BusMetrics[new_input_mapping.buses.size()]);
1171 for (unsigned bus_index = 0; bus_index < new_input_mapping.buses.size(); ++bus_index) {
1172 const InputMapping::Bus &bus = new_input_mapping.buses[bus_index];
1173 BusMetrics &metrics = bus_metrics[bus_index];
1175 char bus_index_str[16], source_index_str[16], source_channels_str[64];
1176 snprintf(bus_index_str, sizeof(bus_index_str), "%u", bus_index);
1177 snprintf(source_index_str, sizeof(source_index_str), "%u", bus.device.index);
1178 snprintf(source_channels_str, sizeof(source_channels_str), "%d:%d", bus.source_channel[0], bus.source_channel[1]);
1180 vector<pair<string, string>> labels;
1181 metrics.labels.emplace_back("index", bus_index_str);
1182 metrics.labels.emplace_back("name", bus.name);
1183 if (bus.device.type == InputSourceType::SILENCE) {
1184 metrics.labels.emplace_back("source_type", "silence");
1185 } else if (bus.device.type == InputSourceType::CAPTURE_CARD) {
1186 AudioDevice *device = find_audio_device(bus.device);
1187 if (device->card_type == CardType::FFMPEG_INPUT) {
1188 metrics.labels.emplace_back("source_type", "ffmpeg_video_input");
1190 metrics.labels.emplace_back("source_type", "capture_card");
1192 } else if (bus.device.type == InputSourceType::ALSA_INPUT) {
1193 metrics.labels.emplace_back("source_type", "alsa_input");
1197 metrics.labels.emplace_back("source_index", source_index_str);
1198 metrics.labels.emplace_back("source_channels", source_channels_str);
1200 vector<pair<string, string>> labels_left = metrics.labels;
1201 labels_left.emplace_back("channel", "left");
1202 vector<pair<string, string>> labels_right = metrics.labels;
1203 labels_right.emplace_back("channel", "right");
1205 global_metrics.add("bus_current_level_dbfs", labels_left, &metrics.current_level_dbfs[0], Metrics::TYPE_GAUGE);
1206 global_metrics.add("bus_current_level_dbfs", labels_right, &metrics.current_level_dbfs[1], Metrics::TYPE_GAUGE);
1207 global_metrics.add("bus_peak_level_dbfs", labels_left, &metrics.peak_level_dbfs[0], Metrics::TYPE_GAUGE);
1208 global_metrics.add("bus_peak_level_dbfs", labels_right, &metrics.peak_level_dbfs[1], Metrics::TYPE_GAUGE);
1209 global_metrics.add("bus_historic_peak_dbfs", metrics.labels, &metrics.historic_peak_dbfs, Metrics::TYPE_GAUGE);
1210 global_metrics.add("bus_gain_staging_db", metrics.labels, &metrics.gain_staging_db, Metrics::TYPE_GAUGE);
1211 global_metrics.add("bus_compressor_attenuation_db", metrics.labels, &metrics.compressor_attenuation_db, Metrics::TYPE_GAUGE);
1214 // Reset resamplers for all cards that don't have the exact same state as before.
1215 for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
1216 const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
1217 AudioDevice *device = find_audio_device(device_spec);
1218 if (device->interesting_channels != interesting_channels[device_spec]) {
1219 device->interesting_channels = interesting_channels[device_spec];
1220 reset_resampler_mutex_held(device_spec);
1223 for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
1224 const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
1225 AudioDevice *device = find_audio_device(device_spec);
1226 if (interesting_channels[device_spec].empty()) {
1227 alsa_pool.release_device(card_index);
1229 alsa_pool.hold_device(card_index);
1231 if (device->interesting_channels != interesting_channels[device_spec]) {
1232 device->interesting_channels = interesting_channels[device_spec];
1233 alsa_pool.reset_device(device_spec.index);
1234 reset_resampler_mutex_held(device_spec);
1238 input_mapping = new_input_mapping;
1241 InputMapping AudioMixer::get_input_mapping() const
1243 lock_guard<timed_mutex> lock(audio_mutex);
1244 return input_mapping;
1247 unsigned AudioMixer::num_buses() const
1249 lock_guard<timed_mutex> lock(audio_mutex);
1250 return input_mapping.buses.size();
1253 void AudioMixer::reset_peak(unsigned bus_index)
1255 lock_guard<timed_mutex> lock(audio_mutex);
1256 for (unsigned channel = 0; channel < 2; ++channel) {
1257 PeakHistory &history = peak_history[bus_index][channel];
1258 history.current_level = 0.0f;
1259 history.historic_peak = 0.0f;
1260 history.current_peak = 0.0f;
1261 history.last_peak = 0.0f;
1262 history.age_seconds = 0.0f;
1266 bool AudioMixer::is_mono(unsigned bus_index)
1268 lock_guard<timed_mutex> lock(audio_mutex);
1269 const InputMapping::Bus &bus = input_mapping.buses[bus_index];
1270 if (bus.device.type == InputSourceType::SILENCE) {
1273 assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
1274 bus.device.type == InputSourceType::ALSA_INPUT);
1275 return bus.source_channel[0] == bus.source_channel[1];
1279 // This is perhaps not the most user-friendly output, but it's at least better
1280 // than the raw index. It would be nice to have it identical to
1281 // Mixer::description_for_card for capture cards, though.
1282 string AudioMixer::spec_to_string(DeviceSpec device_spec) const
1286 switch (device_spec.type) {
1287 case InputSourceType::SILENCE:
1289 case InputSourceType::CAPTURE_CARD: {
1290 const AudioDevice *device = find_audio_device(device_spec);
1291 if (device->card_type == CardType::FFMPEG_INPUT) {
1292 snprintf(buf, sizeof(buf), "Virtual capture card %u (%s)", device_spec.index, device->display_name.c_str());
1294 snprintf(buf, sizeof(buf), "Capture card %u (%s)", device_spec.index, device->display_name.c_str());
1298 case InputSourceType::ALSA_INPUT:
1299 snprintf(buf, sizeof(buf), "ALSA input %u", device_spec.index);
1307 AudioMixer *global_audio_mixer = nullptr;