1 #include "audio_mixer.h"
5 #include <bmusb/bmusb.h>
10 #include <immintrin.h>
17 using namespace bmusb;
19 using namespace std::placeholders;
23 // TODO: If these prove to be a bottleneck, they can be SSSE3-optimized
24 // (usually including multiple channels at a time).
26 void convert_fixed16_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
27 const uint8_t *src, size_t in_channel, size_t in_num_channels,
30 assert(in_channel < in_num_channels);
31 assert(out_channel < out_num_channels);
32 src += in_channel * 2;
35 for (size_t i = 0; i < num_samples; ++i) {
36 int16_t s = le16toh(*(int16_t *)src);
37 *dst = s * (1.0f / 32768.0f);
39 src += 2 * in_num_channels;
40 dst += out_num_channels;
44 void convert_fixed24_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
45 const uint8_t *src, size_t in_channel, size_t in_num_channels,
48 assert(in_channel < in_num_channels);
49 assert(out_channel < out_num_channels);
50 src += in_channel * 3;
53 for (size_t i = 0; i < num_samples; ++i) {
57 uint32_t s = s1 | (s1 << 8) | (s2 << 16) | (s3 << 24);
58 *dst = int(s) * (1.0f / 2147483648.0f);
60 src += 3 * in_num_channels;
61 dst += out_num_channels;
65 void convert_fixed32_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
66 const uint8_t *src, size_t in_channel, size_t in_num_channels,
69 assert(in_channel < in_num_channels);
70 assert(out_channel < out_num_channels);
71 src += in_channel * 4;
74 for (size_t i = 0; i < num_samples; ++i) {
75 int32_t s = le32toh(*(int32_t *)src);
76 *dst = s * (1.0f / 2147483648.0f);
78 src += 4 * in_num_channels;
79 dst += out_num_channels;
83 float find_peak_plain(const float *samples, size_t num_samples) __attribute__((unused));
85 float find_peak_plain(const float *samples, size_t num_samples)
87 float m = fabs(samples[0]);
88 for (size_t i = 1; i < num_samples; ++i) {
89 m = max(m, fabs(samples[i]));
95 static inline float horizontal_max(__m128 m)
97 __m128 tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 0, 3, 2));
98 m = _mm_max_ps(m, tmp);
99 tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 3, 0, 1));
100 m = _mm_max_ps(m, tmp);
101 return _mm_cvtss_f32(m);
104 float find_peak(const float *samples, size_t num_samples)
106 const __m128 abs_mask = _mm_castsi128_ps(_mm_set1_epi32(0x7fffffffu));
107 __m128 m = _mm_setzero_ps();
108 for (size_t i = 0; i < (num_samples & ~3); i += 4) {
109 __m128 x = _mm_loadu_ps(samples + i);
110 x = _mm_and_ps(x, abs_mask);
111 m = _mm_max_ps(m, x);
113 float result = horizontal_max(m);
115 for (size_t i = (num_samples & ~3); i < num_samples; ++i) {
116 result = max(result, fabs(samples[i]));
120 // Self-test. We should be bit-exact the same.
121 float reference_result = find_peak_plain(samples, num_samples);
122 if (result != reference_result) {
123 fprintf(stderr, "Error: Peak is %f [%f %f %f %f]; should be %f.\n",
125 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(0, 0, 0, 0))),
126 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 1, 1, 1))),
127 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 2, 2, 2))),
128 _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(3, 3, 3, 3))),
136 float find_peak(const float *samples, size_t num_samples)
138 return find_peak_plain(samples, num_samples);
142 void deinterleave_samples(const vector<float> &in, vector<float> *out_l, vector<float> *out_r)
144 size_t num_samples = in.size() / 2;
145 out_l->resize(num_samples);
146 out_r->resize(num_samples);
148 const float *inptr = in.data();
149 float *lptr = &(*out_l)[0];
150 float *rptr = &(*out_r)[0];
151 for (size_t i = 0; i < num_samples; ++i) {
159 AudioMixer::AudioMixer(unsigned num_cards)
160 : num_cards(num_cards),
161 limiter(OUTPUT_FREQUENCY),
162 correlation(OUTPUT_FREQUENCY)
164 for (unsigned bus_index = 0; bus_index < MAX_BUSES; ++bus_index) {
165 locut[bus_index].init(FILTER_HPF, 2);
166 locut_enabled[bus_index] = global_flags.locut_enabled;
167 gain_staging_db[bus_index] = global_flags.initial_gain_staging_db;
168 compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
169 compressor_threshold_dbfs[bus_index] = ref_level_dbfs - 12.0f; // -12 dB.
170 compressor_enabled[bus_index] = global_flags.compressor_enabled;
171 level_compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
172 level_compressor_enabled[bus_index] = global_flags.gain_staging_auto;
174 set_limiter_enabled(global_flags.limiter_enabled);
175 set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
177 // Generate a very simple, default input mapping.
178 InputMapping::Bus input;
180 input.device.type = InputSourceType::CAPTURE_CARD;
181 input.device.index = 0;
182 input.source_channel[0] = 0;
183 input.source_channel[1] = 1;
185 InputMapping new_input_mapping;
186 new_input_mapping.buses.push_back(input);
187 set_input_mapping(new_input_mapping);
189 // Look for ALSA cards.
190 available_alsa_cards = ALSAInput::enumerate_devices();
192 r128.init(2, OUTPUT_FREQUENCY);
195 // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
196 // and there's a limit to how important the peak meter is.
197 peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
200 AudioMixer::~AudioMixer()
202 for (unsigned card_index = 0; card_index < available_alsa_cards.size(); ++card_index) {
203 const AudioDevice &device = alsa_inputs[card_index];
204 if (device.alsa_device != nullptr) {
205 device.alsa_device->stop_capture_thread();
211 void AudioMixer::reset_resampler(DeviceSpec device_spec)
213 lock_guard<timed_mutex> lock(audio_mutex);
214 reset_resampler_mutex_held(device_spec);
217 void AudioMixer::reset_resampler_mutex_held(DeviceSpec device_spec)
219 AudioDevice *device = find_audio_device(device_spec);
221 if (device->interesting_channels.empty()) {
222 device->resampling_queue.reset();
224 // TODO: ResamplingQueue should probably take the full device spec.
225 // (It's only used for console output, though.)
226 device->resampling_queue.reset(new ResamplingQueue(device_spec.index, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size()));
228 device->next_local_pts = 0;
231 void AudioMixer::reset_alsa_mutex_held(DeviceSpec device_spec)
233 assert(device_spec.type == InputSourceType::ALSA_INPUT);
234 unsigned card_index = device_spec.index;
235 AudioDevice *device = find_audio_device(device_spec);
237 if (device->alsa_device != nullptr) {
238 device->alsa_device->stop_capture_thread();
240 if (device->interesting_channels.empty()) {
241 device->alsa_device.reset();
243 const ALSAInput::Device &alsa_dev = available_alsa_cards[card_index];
244 device->alsa_device.reset(new ALSAInput(alsa_dev.address.c_str(), OUTPUT_FREQUENCY, alsa_dev.num_channels, bind(&AudioMixer::add_audio, this, device_spec, _1, _2, _3, _4)));
245 device->capture_frequency = device->alsa_device->get_sample_rate();
246 device->alsa_device->start_capture_thread();
250 bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length)
252 AudioDevice *device = find_audio_device(device_spec);
254 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
255 if (!lock.try_lock_for(chrono::milliseconds(10))) {
258 if (device->resampling_queue == nullptr) {
259 // No buses use this device; throw it away.
263 unsigned num_channels = device->interesting_channels.size();
264 assert(num_channels > 0);
266 // Convert the audio to fp32.
268 audio.resize(num_samples * num_channels);
269 unsigned channel_index = 0;
270 for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) {
271 switch (audio_format.bits_per_sample) {
273 assert(num_samples == 0);
276 convert_fixed16_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
279 convert_fixed24_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
282 convert_fixed32_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
285 fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
291 int64_t local_pts = device->next_local_pts;
292 device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.data(), num_samples);
293 device->next_local_pts = local_pts + frame_length;
297 bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length)
299 AudioDevice *device = find_audio_device(device_spec);
301 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
302 if (!lock.try_lock_for(chrono::milliseconds(10))) {
305 if (device->resampling_queue == nullptr) {
306 // No buses use this device; throw it away.
310 unsigned num_channels = device->interesting_channels.size();
311 assert(num_channels > 0);
313 vector<float> silence(samples_per_frame * num_channels, 0.0f);
314 for (unsigned i = 0; i < num_frames; ++i) {
315 device->resampling_queue->add_input_samples(device->next_local_pts / double(TIMEBASE), silence.data(), samples_per_frame);
316 // Note that if the format changed in the meantime, we have
317 // no way of detecting that; we just have to assume the frame length
318 // is always the same.
319 device->next_local_pts += frame_length;
324 AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
326 switch (device.type) {
327 case InputSourceType::CAPTURE_CARD:
328 return &video_cards[device.index];
329 case InputSourceType::ALSA_INPUT:
330 return &alsa_inputs[device.index];
331 case InputSourceType::SILENCE:
338 // Get a pointer to the given channel from the given device.
339 // The channel must be picked out earlier and resampled.
340 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)
342 static float zero = 0.0f;
343 if (source_channel == -1 || device_spec.type == InputSourceType::SILENCE) {
348 AudioDevice *device = find_audio_device(device_spec);
349 assert(device->interesting_channels.count(source_channel) != 0);
350 unsigned channel_index = 0;
351 for (int channel : device->interesting_channels) {
352 if (channel == source_channel) break;
355 assert(channel_index < device->interesting_channels.size());
356 const auto it = samples_card.find(device_spec);
357 assert(it != samples_card.end());
358 *srcptr = &(it->second)[channel_index];
359 *stride = device->interesting_channels.size();
362 // TODO: Can be SSSE3-optimized if need be.
363 void AudioMixer::fill_audio_bus(const map<DeviceSpec, vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output)
365 if (bus.device.type == InputSourceType::SILENCE) {
366 memset(output, 0, num_samples * sizeof(*output));
368 assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
369 bus.device.type == InputSourceType::ALSA_INPUT);
370 const float *lsrc, *rsrc;
371 unsigned lstride, rstride;
372 float *dptr = output;
373 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[0], &lsrc, &lstride);
374 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[1], &rsrc, &rstride);
375 for (unsigned i = 0; i < num_samples; ++i) {
384 vector<float> AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
386 map<DeviceSpec, vector<float>> samples_card;
387 vector<float> samples_bus;
389 lock_guard<timed_mutex> lock(audio_mutex);
391 // Pick out all the interesting channels from all the cards.
392 // TODO: If the card has been hotswapped, the number of channels
393 // might have changed; if so, we need to do some sort of remapping
395 for (const auto &spec_and_info : get_devices_mutex_held()) {
396 const DeviceSpec &device_spec = spec_and_info.first;
397 AudioDevice *device = find_audio_device(device_spec);
398 if (!device->interesting_channels.empty()) {
399 samples_card[device_spec].resize(num_samples * device->interesting_channels.size());
400 device->resampling_queue->get_output_samples(
402 &samples_card[device_spec][0],
404 rate_adjustment_policy);
408 vector<float> samples_out, left, right;
409 samples_out.resize(num_samples * 2);
410 samples_bus.resize(num_samples * 2);
411 for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
412 fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, &samples_bus[0]);
414 // Cut away everything under 120 Hz (or whatever the cutoff is);
415 // we don't need it for voice, and it will reduce headroom
416 // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
417 // should be dampened.)
418 if (locut_enabled[bus_index]) {
419 locut[bus_index].render(samples_bus.data(), samples_bus.size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
423 lock_guard<mutex> lock(compressor_mutex);
425 // Apply a level compressor to get the general level right.
426 // Basically, if it's over about -40 dBFS, we squeeze it down to that level
427 // (or more precisely, near it, since we don't use infinite ratio),
428 // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
429 // entirely arbitrary, but from practical tests with speech, it seems to
430 // put ut around -23 LUFS, so it's a reasonable starting point for later use.
431 if (level_compressor_enabled[bus_index]) {
432 float threshold = 0.01f; // -40 dBFS.
434 float attack_time = 0.5f;
435 float release_time = 20.0f;
436 float makeup_gain = from_db(ref_level_dbfs - (-40.0f)); // +26 dB.
437 level_compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
438 gain_staging_db[bus_index] = to_db(level_compressor[bus_index]->get_attenuation() * makeup_gain);
440 // Just apply the gain we already had.
441 float g = from_db(gain_staging_db[bus_index]);
442 for (size_t i = 0; i < samples_bus.size(); ++i) {
448 printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
449 level_compressor.get_level(), to_db(level_compressor.get_level()),
450 level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()),
451 to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
454 // The real compressor.
455 if (compressor_enabled[bus_index]) {
456 float threshold = from_db(compressor_threshold_dbfs[bus_index]);
458 float attack_time = 0.005f;
459 float release_time = 0.040f;
460 float makeup_gain = 2.0f; // +6 dB.
461 compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
462 // compressor_att = compressor.get_attenuation();
466 // TODO: We should measure post-fader.
467 deinterleave_samples(samples_bus, &left, &right);
468 measure_bus_levels(bus_index, left, right);
470 float volume = from_db(fader_volume_db[bus_index]);
471 if (bus_index == 0) {
472 for (unsigned i = 0; i < num_samples * 2; ++i) {
473 samples_out[i] = samples_bus[i] * volume;
476 for (unsigned i = 0; i < num_samples * 2; ++i) {
477 samples_out[i] += samples_bus[i] * volume;
483 lock_guard<mutex> lock(compressor_mutex);
485 // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
486 // Note that since ratio is not infinite, we could go slightly higher than this.
487 if (limiter_enabled) {
488 float threshold = from_db(limiter_threshold_dbfs);
490 float attack_time = 0.0f; // Instant.
491 float release_time = 0.020f;
492 float makeup_gain = 1.0f; // 0 dB.
493 limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
494 // limiter_att = limiter.get_attenuation();
497 // printf("limiter=%+5.1f compressor=%+5.1f\n", to_db(limiter_att), to_db(compressor_att));
500 // At this point, we are most likely close to +0 LU (at least if the
501 // faders sum to 0 dB and the compressors are on), but all of our
502 // measurements have been on raw sample values, not R128 values.
503 // So we have a final makeup gain to get us to +0 LU; the gain
504 // adjustments required should be relatively small, and also, the
505 // offset shouldn't change much (only if the type of audio changes
506 // significantly). Thus, we shoot for updating this value basically
507 // “whenever we process buffers”, since the R128 calculation isn't exactly
508 // something we get out per-sample.
510 // Note that there's a feedback loop here, so we choose a very slow filter
511 // (half-time of 30 seconds).
512 double target_loudness_factor, alpha;
513 double loudness_lu = r128.loudness_M() - ref_level_lufs;
514 double current_makeup_lu = to_db(final_makeup_gain);
515 target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
517 // If we're outside +/- 5 LU uncorrected, we don't count it as
518 // a normal signal (probably silence) and don't change the
519 // correction factor; just apply what we already have.
520 if (fabs(loudness_lu - current_makeup_lu) >= 5.0 || !final_makeup_gain_auto) {
523 // Formula adapted from
524 // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
525 const double half_time_s = 30.0;
526 const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
527 alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
531 lock_guard<mutex> lock(compressor_mutex);
532 double m = final_makeup_gain;
533 for (size_t i = 0; i < samples_out.size(); i += 2) {
534 samples_out[i + 0] *= m;
535 samples_out[i + 1] *= m;
536 m += (target_loudness_factor - m) * alpha;
538 final_makeup_gain = m;
541 update_meters(samples_out);
546 void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right)
548 const float *ptrs[] = { left.data(), right.data() };
550 lock_guard<mutex> lock(audio_measure_mutex);
551 bus_r128[bus_index]->process(left.size(), const_cast<float **>(ptrs));
555 void AudioMixer::update_meters(const vector<float> &samples)
557 // Upsample 4x to find interpolated peak.
558 peak_resampler.inp_data = const_cast<float *>(samples.data());
559 peak_resampler.inp_count = samples.size() / 2;
561 vector<float> interpolated_samples;
562 interpolated_samples.resize(samples.size());
564 lock_guard<mutex> lock(audio_measure_mutex);
566 while (peak_resampler.inp_count > 0) { // About four iterations.
567 peak_resampler.out_data = &interpolated_samples[0];
568 peak_resampler.out_count = interpolated_samples.size() / 2;
569 peak_resampler.process();
570 size_t out_stereo_samples = interpolated_samples.size() / 2 - peak_resampler.out_count;
571 peak = max<float>(peak, find_peak(interpolated_samples.data(), out_stereo_samples * 2));
572 peak_resampler.out_data = nullptr;
576 // Find R128 levels and L/R correlation.
577 vector<float> left, right;
578 deinterleave_samples(samples, &left, &right);
579 float *ptrs[] = { left.data(), right.data() };
581 lock_guard<mutex> lock(audio_measure_mutex);
582 r128.process(left.size(), ptrs);
583 correlation.process_samples(samples);
586 send_audio_level_callback();
589 void AudioMixer::reset_meters()
591 lock_guard<mutex> lock(audio_measure_mutex);
592 peak_resampler.reset();
599 void AudioMixer::send_audio_level_callback()
601 if (audio_level_callback == nullptr) {
605 lock_guard<mutex> lock(audio_measure_mutex);
606 double loudness_s = r128.loudness_S();
607 double loudness_i = r128.integrated();
608 double loudness_range_low = r128.range_min();
609 double loudness_range_high = r128.range_max();
611 vector<BusLevel> bus_levels;
612 bus_levels.resize(input_mapping.buses.size());
614 lock_guard<mutex> lock(compressor_mutex);
615 for (unsigned bus_index = 0; bus_index < bus_r128.size(); ++bus_index) {
616 bus_levels[bus_index].loudness_lufs = bus_r128[bus_index]->loudness_S();
617 bus_levels[bus_index].gain_staging_db = gain_staging_db[bus_index];
618 if (compressor_enabled[bus_index]) {
619 bus_levels[bus_index].compressor_attenuation_db = -to_db(compressor[bus_index]->get_attenuation());
621 bus_levels[bus_index].compressor_attenuation_db = 0.0;
626 audio_level_callback(loudness_s, to_db(peak), bus_levels,
627 loudness_i, loudness_range_low, loudness_range_high,
628 to_db(final_makeup_gain),
629 correlation.get_correlation());
632 map<DeviceSpec, DeviceInfo> AudioMixer::get_devices() const
634 lock_guard<timed_mutex> lock(audio_mutex);
635 return get_devices_mutex_held();
638 map<DeviceSpec, DeviceInfo> AudioMixer::get_devices_mutex_held() const
640 map<DeviceSpec, DeviceInfo> devices;
641 for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
642 const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index };
643 const AudioDevice *device = &video_cards[card_index];
645 info.name = device->name;
646 info.num_channels = 8; // FIXME: This is wrong for fake cards.
647 devices.insert(make_pair(spec, info));
649 for (unsigned card_index = 0; card_index < available_alsa_cards.size(); ++card_index) {
650 const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index };
651 const ALSAInput::Device &device = available_alsa_cards[card_index];
653 info.name = device.name + " (" + device.info + ")";
654 info.num_channels = device.num_channels;
655 devices.insert(make_pair(spec, info));
660 void AudioMixer::set_name(DeviceSpec device_spec, const string &name)
662 AudioDevice *device = find_audio_device(device_spec);
664 lock_guard<timed_mutex> lock(audio_mutex);
668 void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
670 lock_guard<timed_mutex> lock(audio_mutex);
672 map<DeviceSpec, set<unsigned>> interesting_channels;
673 for (const InputMapping::Bus &bus : new_input_mapping.buses) {
674 if (bus.device.type == InputSourceType::CAPTURE_CARD ||
675 bus.device.type == InputSourceType::ALSA_INPUT) {
676 for (unsigned channel = 0; channel < 2; ++channel) {
677 if (bus.source_channel[channel] != -1) {
678 interesting_channels[bus.device].insert(bus.source_channel[channel]);
684 // Reset resamplers for all cards that don't have the exact same state as before.
685 for (const auto &spec_and_info : get_devices_mutex_held()) {
686 const DeviceSpec &device_spec = spec_and_info.first;
687 AudioDevice *device = find_audio_device(device_spec);
688 if (device->interesting_channels != interesting_channels[device_spec]) {
689 device->interesting_channels = interesting_channels[device_spec];
690 if (device_spec.type == InputSourceType::ALSA_INPUT) {
691 reset_alsa_mutex_held(device_spec);
693 reset_resampler_mutex_held(device_spec);
698 lock_guard<mutex> lock(audio_measure_mutex);
699 bus_r128.resize(new_input_mapping.buses.size());
700 for (unsigned bus_index = 0; bus_index < bus_r128.size(); ++bus_index) {
701 if (bus_r128[bus_index] == nullptr) {
702 bus_r128[bus_index].reset(new Ebu_r128_proc);
704 bus_r128[bus_index]->init(2, OUTPUT_FREQUENCY);
708 input_mapping = new_input_mapping;
711 InputMapping AudioMixer::get_input_mapping() const
713 lock_guard<timed_mutex> lock(audio_mutex);
714 return input_mapping;