1 #include "audio_mixer.h"
5 #include <bmusb/bmusb.h>
14 using namespace bmusb;
16 using namespace std::placeholders;
20 // TODO: If these prove to be a bottleneck, they can be SSSE3-optimized
21 // (usually including multiple channels at a time).
23 void convert_fixed16_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
24 const uint8_t *src, size_t in_channel, size_t in_num_channels,
27 assert(in_channel < in_num_channels);
28 assert(out_channel < out_num_channels);
29 src += in_channel * 2;
32 for (size_t i = 0; i < num_samples; ++i) {
33 int16_t s = le16toh(*(int16_t *)src);
34 *dst = s * (1.0f / 32768.0f);
36 src += 2 * in_num_channels;
37 dst += out_num_channels;
41 void convert_fixed24_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
42 const uint8_t *src, size_t in_channel, size_t in_num_channels,
45 assert(in_channel < in_num_channels);
46 assert(out_channel < out_num_channels);
47 src += in_channel * 3;
50 for (size_t i = 0; i < num_samples; ++i) {
54 uint32_t s = s1 | (s1 << 8) | (s2 << 16) | (s3 << 24);
55 *dst = int(s) * (1.0f / 2147483648.0f);
57 src += 3 * in_num_channels;
58 dst += out_num_channels;
62 void convert_fixed32_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
63 const uint8_t *src, size_t in_channel, size_t in_num_channels,
66 assert(in_channel < in_num_channels);
67 assert(out_channel < out_num_channels);
68 src += in_channel * 4;
71 for (size_t i = 0; i < num_samples; ++i) {
72 int32_t s = le32toh(*(int32_t *)src);
73 *dst = s * (1.0f / 2147483648.0f);
75 src += 4 * in_num_channels;
76 dst += out_num_channels;
82 AudioMixer::AudioMixer(unsigned num_cards)
83 : num_cards(num_cards),
84 level_compressor(OUTPUT_FREQUENCY),
85 limiter(OUTPUT_FREQUENCY),
86 compressor(OUTPUT_FREQUENCY)
88 locut.init(FILTER_HPF, 2);
90 set_locut_enabled(global_flags.locut_enabled);
91 set_gain_staging_db(global_flags.initial_gain_staging_db);
92 set_gain_staging_auto(global_flags.gain_staging_auto);
93 set_compressor_enabled(global_flags.compressor_enabled);
94 set_limiter_enabled(global_flags.limiter_enabled);
95 set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
97 // Generate a very simple, default input mapping.
98 InputMapping::Bus input;
100 input.device.type = InputSourceType::CAPTURE_CARD;
101 input.device.index = 0;
102 input.source_channel[0] = 0;
103 input.source_channel[1] = 1;
105 InputMapping new_input_mapping;
106 new_input_mapping.buses.push_back(input);
107 set_input_mapping(new_input_mapping);
109 // Look for ALSA cards.
110 available_alsa_cards = ALSAInput::enumerate_devices();
113 AudioMixer::~AudioMixer()
115 for (unsigned card_index = 0; card_index < available_alsa_cards.size(); ++card_index) {
116 const AudioDevice &device = alsa_inputs[card_index];
117 if (device.alsa_device != nullptr) {
118 device.alsa_device->stop_capture_thread();
124 void AudioMixer::reset_resampler(DeviceSpec device_spec)
126 lock_guard<timed_mutex> lock(audio_mutex);
127 reset_resampler_mutex_held(device_spec);
130 void AudioMixer::reset_resampler_mutex_held(DeviceSpec device_spec)
132 AudioDevice *device = find_audio_device(device_spec);
134 if (device->interesting_channels.empty()) {
135 device->resampling_queue.reset();
137 // TODO: ResamplingQueue should probably take the full device spec.
138 // (It's only used for console output, though.)
139 device->resampling_queue.reset(new ResamplingQueue(device_spec.index, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size()));
141 device->next_local_pts = 0;
144 void AudioMixer::reset_alsa_mutex_held(DeviceSpec device_spec)
146 assert(device_spec.type == InputSourceType::ALSA_INPUT);
147 unsigned card_index = device_spec.index;
148 AudioDevice *device = find_audio_device(device_spec);
150 if (device->alsa_device != nullptr) {
151 device->alsa_device->stop_capture_thread();
153 if (device->interesting_channels.empty()) {
154 device->alsa_device.reset();
156 device->alsa_device.reset(new ALSAInput(available_alsa_cards[card_index].address.c_str(), OUTPUT_FREQUENCY, 2, bind(&AudioMixer::add_audio, this, device_spec, _1, _2, _3, _4)));
157 device->capture_frequency = device->alsa_device->get_sample_rate();
158 device->alsa_device->start_capture_thread();
162 bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length)
164 AudioDevice *device = find_audio_device(device_spec);
166 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
167 if (!lock.try_lock_for(chrono::milliseconds(10))) {
170 if (device->resampling_queue == nullptr) {
171 // No buses use this device; throw it away.
175 unsigned num_channels = device->interesting_channels.size();
176 assert(num_channels > 0);
178 // Convert the audio to fp32.
180 audio.resize(num_samples * num_channels);
181 unsigned channel_index = 0;
182 for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) {
183 switch (audio_format.bits_per_sample) {
185 assert(num_samples == 0);
188 convert_fixed16_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
191 convert_fixed24_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
194 convert_fixed32_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
197 fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
203 int64_t local_pts = device->next_local_pts;
204 device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.data(), num_samples);
205 device->next_local_pts = local_pts + frame_length;
209 bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length)
211 AudioDevice *device = find_audio_device(device_spec);
213 unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
214 if (!lock.try_lock_for(chrono::milliseconds(10))) {
217 if (device->resampling_queue == nullptr) {
218 // No buses use this device; throw it away.
222 unsigned num_channels = device->interesting_channels.size();
223 assert(num_channels > 0);
225 vector<float> silence(samples_per_frame * num_channels, 0.0f);
226 for (unsigned i = 0; i < num_frames; ++i) {
227 device->resampling_queue->add_input_samples(device->next_local_pts / double(TIMEBASE), silence.data(), samples_per_frame);
228 // Note that if the format changed in the meantime, we have
229 // no way of detecting that; we just have to assume the frame length
230 // is always the same.
231 device->next_local_pts += frame_length;
236 AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
238 switch (device.type) {
239 case InputSourceType::CAPTURE_CARD:
240 return &video_cards[device.index];
241 case InputSourceType::ALSA_INPUT:
242 return &alsa_inputs[device.index];
243 case InputSourceType::SILENCE:
250 // Get a pointer to the given channel from the given device.
251 // The channel must be picked out earlier and resampled.
252 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)
254 static float zero = 0.0f;
255 if (source_channel == -1 || device_spec.type == InputSourceType::SILENCE) {
260 AudioDevice *device = find_audio_device(device_spec);
261 assert(device->interesting_channels.count(source_channel) != 0);
262 unsigned channel_index = 0;
263 for (int channel : device->interesting_channels) {
264 if (channel == source_channel) break;
267 assert(channel_index < device->interesting_channels.size());
268 const auto it = samples_card.find(device_spec);
269 assert(it != samples_card.end());
270 *srcptr = &(it->second)[channel_index];
271 *stride = device->interesting_channels.size();
274 // TODO: Can be SSSE3-optimized if need be.
275 void AudioMixer::fill_audio_bus(const map<DeviceSpec, vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output)
277 if (bus.device.type == InputSourceType::SILENCE) {
278 memset(output, 0, num_samples * sizeof(*output));
280 assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
281 bus.device.type == InputSourceType::ALSA_INPUT);
282 const float *lsrc, *rsrc;
283 unsigned lstride, rstride;
284 float *dptr = output;
285 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[0], &lsrc, &lstride);
286 find_sample_src_from_device(samples_card, bus.device, bus.source_channel[1], &rsrc, &rstride);
287 for (unsigned i = 0; i < num_samples; ++i) {
296 vector<float> AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
298 map<DeviceSpec, vector<float>> samples_card;
299 vector<float> samples_bus;
301 lock_guard<timed_mutex> lock(audio_mutex);
303 // Pick out all the interesting channels from all the cards.
304 // TODO: If the card has been hotswapped, the number of channels
305 // might have changed; if so, we need to do some sort of remapping
307 for (const auto &spec_and_info : get_devices_mutex_held()) {
308 const DeviceSpec &device_spec = spec_and_info.first;
309 AudioDevice *device = find_audio_device(device_spec);
310 if (!device->interesting_channels.empty()) {
311 samples_card[device_spec].resize(num_samples * device->interesting_channels.size());
312 device->resampling_queue->get_output_samples(
314 &samples_card[device_spec][0],
316 rate_adjustment_policy);
320 // TODO: Move lo-cut etc. into each bus.
321 vector<float> samples_out;
322 samples_out.resize(num_samples * 2);
323 samples_bus.resize(num_samples * 2);
324 for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
325 fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, &samples_bus[0]);
327 float volume = from_db(fader_volume_db[bus_index]);
328 if (bus_index == 0) {
329 for (unsigned i = 0; i < num_samples * 2; ++i) {
330 samples_out[i] = samples_bus[i] * volume;
333 for (unsigned i = 0; i < num_samples * 2; ++i) {
334 samples_out[i] += samples_bus[i] * volume;
339 // Cut away everything under 120 Hz (or whatever the cutoff is);
340 // we don't need it for voice, and it will reduce headroom
341 // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
342 // should be dampened.)
344 locut.render(samples_out.data(), samples_out.size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
348 lock_guard<mutex> lock(compressor_mutex);
350 // Apply a level compressor to get the general level right.
351 // Basically, if it's over about -40 dBFS, we squeeze it down to that level
352 // (or more precisely, near it, since we don't use infinite ratio),
353 // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
354 // entirely arbitrary, but from practical tests with speech, it seems to
355 // put ut around -23 LUFS, so it's a reasonable starting point for later use.
357 if (level_compressor_enabled) {
358 float threshold = 0.01f; // -40 dBFS.
360 float attack_time = 0.5f;
361 float release_time = 20.0f;
362 float makeup_gain = from_db(ref_level_dbfs - (-40.0f)); // +26 dB.
363 level_compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
364 gain_staging_db = to_db(level_compressor.get_attenuation() * makeup_gain);
366 // Just apply the gain we already had.
367 float g = from_db(gain_staging_db);
368 for (size_t i = 0; i < samples_out.size(); ++i) {
375 printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
376 level_compressor.get_level(), to_db(level_compressor.get_level()),
377 level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()),
378 to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
381 // float limiter_att, compressor_att;
383 // The real compressor.
384 if (compressor_enabled) {
385 float threshold = from_db(compressor_threshold_dbfs);
387 float attack_time = 0.005f;
388 float release_time = 0.040f;
389 float makeup_gain = 2.0f; // +6 dB.
390 compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
391 // compressor_att = compressor.get_attenuation();
394 // Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
395 // Note that since ratio is not infinite, we could go slightly higher than this.
396 if (limiter_enabled) {
397 float threshold = from_db(limiter_threshold_dbfs);
399 float attack_time = 0.0f; // Instant.
400 float release_time = 0.020f;
401 float makeup_gain = 1.0f; // 0 dB.
402 limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
403 // limiter_att = limiter.get_attenuation();
406 // printf("limiter=%+5.1f compressor=%+5.1f\n", to_db(limiter_att), to_db(compressor_att));
409 // At this point, we are most likely close to +0 LU, but all of our
410 // measurements have been on raw sample values, not R128 values.
411 // So we have a final makeup gain to get us to +0 LU; the gain
412 // adjustments required should be relatively small, and also, the
413 // offset shouldn't change much (only if the type of audio changes
414 // significantly). Thus, we shoot for updating this value basically
415 // “whenever we process buffers”, since the R128 calculation isn't exactly
416 // something we get out per-sample.
418 // Note that there's a feedback loop here, so we choose a very slow filter
419 // (half-time of 30 seconds).
420 double target_loudness_factor, alpha;
421 double loudness_lu = loudness_lufs - ref_level_lufs;
422 double current_makeup_lu = to_db(final_makeup_gain);
423 target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
425 // If we're outside +/- 5 LU uncorrected, we don't count it as
426 // a normal signal (probably silence) and don't change the
427 // correction factor; just apply what we already have.
428 if (fabs(loudness_lu - current_makeup_lu) >= 5.0 || !final_makeup_gain_auto) {
431 // Formula adapted from
432 // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
433 const double half_time_s = 30.0;
434 const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
435 alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
439 lock_guard<mutex> lock(compressor_mutex);
440 double m = final_makeup_gain;
441 for (size_t i = 0; i < samples_out.size(); i += 2) {
442 samples_out[i + 0] *= m;
443 samples_out[i + 1] *= m;
444 m += (target_loudness_factor - m) * alpha;
446 final_makeup_gain = m;
452 map<DeviceSpec, DeviceInfo> AudioMixer::get_devices() const
454 lock_guard<timed_mutex> lock(audio_mutex);
455 return get_devices_mutex_held();
458 map<DeviceSpec, DeviceInfo> AudioMixer::get_devices_mutex_held() const
460 map<DeviceSpec, DeviceInfo> devices;
461 for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
462 const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index };
463 const AudioDevice *device = &video_cards[card_index];
465 info.name = device->name;
466 info.num_channels = 8; // FIXME: This is wrong for fake cards.
467 devices.insert(make_pair(spec, info));
469 for (unsigned card_index = 0; card_index < available_alsa_cards.size(); ++card_index) {
470 const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index };
471 const ALSAInput::Device &device = available_alsa_cards[card_index];
473 info.name = device.name + " (" + device.info + ")";
474 info.num_channels = device.num_channels;
475 devices.insert(make_pair(spec, info));
480 void AudioMixer::set_name(DeviceSpec device_spec, const string &name)
482 AudioDevice *device = find_audio_device(device_spec);
484 lock_guard<timed_mutex> lock(audio_mutex);
488 void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
490 lock_guard<timed_mutex> lock(audio_mutex);
492 map<DeviceSpec, set<unsigned>> interesting_channels;
493 for (const InputMapping::Bus &bus : new_input_mapping.buses) {
494 if (bus.device.type == InputSourceType::CAPTURE_CARD ||
495 bus.device.type == InputSourceType::ALSA_INPUT) {
496 for (unsigned channel = 0; channel < 2; ++channel) {
497 if (bus.source_channel[channel] != -1) {
498 interesting_channels[bus.device].insert(bus.source_channel[channel]);
504 // Reset resamplers for all cards that don't have the exact same state as before.
505 for (const auto &spec_and_info : get_devices_mutex_held()) {
506 const DeviceSpec &device_spec = spec_and_info.first;
507 AudioDevice *device = find_audio_device(device_spec);
508 if (device->interesting_channels != interesting_channels[device_spec]) {
509 device->interesting_channels = interesting_channels[device_spec];
510 if (device_spec.type == InputSourceType::ALSA_INPUT) {
511 reset_alsa_mutex_held(device_spec);
513 reset_resampler_mutex_held(device_spec);
517 input_mapping = new_input_mapping;
520 InputMapping AudioMixer::get_input_mapping() const
522 lock_guard<timed_mutex> lock(audio_mutex);
523 return input_mapping;