}
}
+float find_peak(const float *samples, size_t num_samples)
+{
+ float m = fabs(samples[0]);
+ for (size_t i = 1; i < num_samples; ++i) {
+ m = max(m, fabs(samples[i]));
+ }
+ return m;
+}
+
+void deinterleave_samples(const vector<float> &in, vector<float> *out_l, vector<float> *out_r)
+{
+ size_t num_samples = in.size() / 2;
+ out_l->resize(num_samples);
+ out_r->resize(num_samples);
+
+ const float *inptr = in.data();
+ float *lptr = &(*out_l)[0];
+ float *rptr = &(*out_r)[0];
+ for (size_t i = 0; i < num_samples; ++i) {
+ *lptr++ = *inptr++;
+ *rptr++ = *inptr++;
+ }
+}
+
} // namespace
AudioMixer::AudioMixer(unsigned num_cards)
: num_cards(num_cards),
level_compressor(OUTPUT_FREQUENCY),
limiter(OUTPUT_FREQUENCY),
- compressor(OUTPUT_FREQUENCY)
+ compressor(OUTPUT_FREQUENCY),
+ correlation(OUTPUT_FREQUENCY)
{
locut.init(FILTER_HPF, 2);
// Look for ALSA cards.
available_alsa_cards = ALSAInput::enumerate_devices();
+
+ r128.init(2, OUTPUT_FREQUENCY);
+ r128.integr_start();
+
+ // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
+ // and there's a limit to how important the peak meter is.
+ peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
}
AudioMixer::~AudioMixer()
void AudioMixer::reset_resampler(DeviceSpec device_spec)
{
- lock_guard<mutex> lock(audio_mutex);
+ lock_guard<timed_mutex> lock(audio_mutex);
reset_resampler_mutex_held(device_spec);
}
if (device->interesting_channels.empty()) {
device->alsa_device.reset();
} else {
- 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)));
+ const ALSAInput::Device &alsa_dev = available_alsa_cards[card_index];
+ 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)));
device->capture_frequency = device->alsa_device->get_sample_rate();
device->alsa_device->start_capture_thread();
}
}
-void AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length)
+bool AudioMixer::add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, AudioFormat audio_format, int64_t frame_length)
{
AudioDevice *device = find_audio_device(device_spec);
- lock_guard<mutex> lock(audio_mutex);
+ unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
+ if (!lock.try_lock_for(chrono::milliseconds(10))) {
+ return false;
+ }
if (device->resampling_queue == nullptr) {
// No buses use this device; throw it away.
- return;
+ return true;
}
unsigned num_channels = device->interesting_channels.size();
int64_t local_pts = device->next_local_pts;
device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.data(), num_samples);
device->next_local_pts = local_pts + frame_length;
+ return true;
}
-void AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length)
+bool AudioMixer::add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length)
{
AudioDevice *device = find_audio_device(device_spec);
- lock_guard<mutex> lock(audio_mutex);
+ unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
+ if (!lock.try_lock_for(chrono::milliseconds(10))) {
+ return false;
+ }
if (device->resampling_queue == nullptr) {
// No buses use this device; throw it away.
- return;
+ return true;
}
unsigned num_channels = device->interesting_channels.size();
// is always the same.
device->next_local_pts += frame_length;
}
+ return true;
}
AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
map<DeviceSpec, vector<float>> samples_card;
vector<float> samples_bus;
- lock_guard<mutex> lock(audio_mutex);
+ lock_guard<timed_mutex> lock(audio_mutex);
// Pick out all the interesting channels from all the cards.
// TODO: If the card has been hotswapped, the number of channels
}
// TODO: Move lo-cut etc. into each bus.
- vector<float> samples_out;
+ vector<float> samples_out, left, right;
samples_out.resize(num_samples * 2);
samples_bus.resize(num_samples * 2);
for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, &samples_bus[0]);
+ // TODO: We should measure post-fader.
+ deinterleave_samples(samples_bus, &left, &right);
+ measure_bus_levels(bus_index, left, right);
+
float volume = from_db(fader_volume_db[bus_index]);
if (bus_index == 0) {
for (unsigned i = 0; i < num_samples * 2; ++i) {
// Note that there's a feedback loop here, so we choose a very slow filter
// (half-time of 30 seconds).
double target_loudness_factor, alpha;
- double loudness_lu = loudness_lufs - ref_level_lufs;
+ double loudness_lu = r128.loudness_M() - ref_level_lufs;
double current_makeup_lu = to_db(final_makeup_gain);
target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
final_makeup_gain = m;
}
+ update_meters(samples_out);
+
return samples_out;
}
+void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right)
+{
+ const float *ptrs[] = { left.data(), right.data() };
+ {
+ lock_guard<mutex> lock(audio_measure_mutex);
+ bus_r128[bus_index]->process(left.size(), const_cast<float **>(ptrs));
+ }
+}
+
+void AudioMixer::update_meters(const vector<float> &samples)
+{
+ // Upsample 4x to find interpolated peak.
+ peak_resampler.inp_data = const_cast<float *>(samples.data());
+ peak_resampler.inp_count = samples.size() / 2;
+
+ vector<float> interpolated_samples;
+ interpolated_samples.resize(samples.size());
+ {
+ lock_guard<mutex> lock(audio_measure_mutex);
+
+ while (peak_resampler.inp_count > 0) { // About four iterations.
+ peak_resampler.out_data = &interpolated_samples[0];
+ peak_resampler.out_count = interpolated_samples.size() / 2;
+ peak_resampler.process();
+ size_t out_stereo_samples = interpolated_samples.size() / 2 - peak_resampler.out_count;
+ peak = max<float>(peak, find_peak(interpolated_samples.data(), out_stereo_samples * 2));
+ peak_resampler.out_data = nullptr;
+ }
+ }
+
+ // Find R128 levels and L/R correlation.
+ vector<float> left, right;
+ deinterleave_samples(samples, &left, &right);
+ float *ptrs[] = { left.data(), right.data() };
+ {
+ lock_guard<mutex> lock(audio_measure_mutex);
+ r128.process(left.size(), ptrs);
+ correlation.process_samples(samples);
+ }
+
+ send_audio_level_callback();
+}
+
+void AudioMixer::reset_meters()
+{
+ lock_guard<mutex> lock(audio_measure_mutex);
+ peak_resampler.reset();
+ peak = 0.0f;
+ r128.reset();
+ r128.integr_start();
+ correlation.reset();
+}
+
+void AudioMixer::send_audio_level_callback()
+{
+ if (audio_level_callback == nullptr) {
+ return;
+ }
+
+ lock_guard<mutex> lock(audio_measure_mutex);
+ double loudness_s = r128.loudness_S();
+ double loudness_i = r128.integrated();
+ double loudness_range_low = r128.range_min();
+ double loudness_range_high = r128.range_max();
+
+ vector<float> bus_loudness;
+ bus_loudness.resize(input_mapping.buses.size());
+ for (unsigned bus_index = 0; bus_index < bus_r128.size(); ++bus_index) {
+ bus_loudness[bus_index] = bus_r128[bus_index]->loudness_S();
+ }
+
+ audio_level_callback(loudness_s, to_db(peak), bus_loudness,
+ loudness_i, loudness_range_low, loudness_range_high,
+ gain_staging_db,
+ to_db(final_makeup_gain),
+ correlation.get_correlation());
+}
+
map<DeviceSpec, DeviceInfo> AudioMixer::get_devices() const
{
- lock_guard<mutex> lock(audio_mutex);
+ lock_guard<timed_mutex> lock(audio_mutex);
return get_devices_mutex_held();
}
{
AudioDevice *device = find_audio_device(device_spec);
- lock_guard<mutex> lock(audio_mutex);
+ lock_guard<timed_mutex> lock(audio_mutex);
device->name = name;
}
void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
{
- lock_guard<mutex> lock(audio_mutex);
+ lock_guard<timed_mutex> lock(audio_mutex);
map<DeviceSpec, set<unsigned>> interesting_channels;
for (const InputMapping::Bus &bus : new_input_mapping.buses) {
}
}
+ {
+ lock_guard<mutex> lock(audio_measure_mutex);
+ bus_r128.resize(new_input_mapping.buses.size());
+ for (unsigned bus_index = 0; bus_index < bus_r128.size(); ++bus_index) {
+ if (bus_r128[bus_index] == nullptr) {
+ bus_r128[bus_index].reset(new Ebu_r128_proc);
+ }
+ bus_r128[bus_index]->init(2, OUTPUT_FREQUENCY);
+ }
+ }
+
input_mapping = new_input_mapping;
}
InputMapping AudioMixer::get_input_mapping() const
{
- lock_guard<mutex> lock(audio_mutex);
+ lock_guard<timed_mutex> lock(audio_mutex);
return input_mapping;
}
projects / nageru / blobdiff