}
}
+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)
+ correlation(OUTPUT_FREQUENCY)
{
- locut.init(FILTER_HPF, 2);
-
- set_locut_enabled(global_flags.locut_enabled);
- set_gain_staging_db(global_flags.initial_gain_staging_db);
- set_gain_staging_auto(global_flags.gain_staging_auto);
- set_compressor_enabled(global_flags.compressor_enabled);
+ for (unsigned bus_index = 0; bus_index < MAX_BUSES; ++bus_index) {
+ locut[bus_index].init(FILTER_HPF, 2);
+ locut_enabled[bus_index] = global_flags.locut_enabled;
+ gain_staging_db[bus_index] = global_flags.initial_gain_staging_db;
+ compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
+ compressor_threshold_dbfs[bus_index] = ref_level_dbfs - 12.0f; // -12 dB.
+ compressor_enabled[bus_index] = global_flags.compressor_enabled;
+ level_compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
+ level_compressor_enabled[bus_index] = global_flags.gain_staging_auto;
+ }
set_limiter_enabled(global_flags.limiter_enabled);
set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
// 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()
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();
}
}
}
- // 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]);
- float volume = from_db(fader_volume_db[bus_index]);
- if (bus_index == 0) {
- for (unsigned i = 0; i < num_samples * 2; ++i) {
- samples_out[i] = samples_bus[i] * volume;
- }
- } else {
- for (unsigned i = 0; i < num_samples * 2; ++i) {
- samples_out[i] += samples_bus[i] * volume;
- }
+ // Cut away everything under 120 Hz (or whatever the cutoff is);
+ // we don't need it for voice, and it will reduce headroom
+ // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
+ // should be dampened.)
+ if (locut_enabled[bus_index]) {
+ locut[bus_index].render(samples_bus.data(), samples_bus.size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
}
- }
-
- // Cut away everything under 120 Hz (or whatever the cutoff is);
- // we don't need it for voice, and it will reduce headroom
- // and confuse the compressor. (In particular, any hums at 50 or 60 Hz
- // should be dampened.)
- if (locut_enabled) {
- locut.render(samples_out.data(), samples_out.size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
- }
- {
- lock_guard<mutex> lock(compressor_mutex);
-
- // Apply a level compressor to get the general level right.
- // Basically, if it's over about -40 dBFS, we squeeze it down to that level
- // (or more precisely, near it, since we don't use infinite ratio),
- // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
- // entirely arbitrary, but from practical tests with speech, it seems to
- // put ut around -23 LUFS, so it's a reasonable starting point for later use.
{
- if (level_compressor_enabled) {
+ lock_guard<mutex> lock(compressor_mutex);
+
+ // Apply a level compressor to get the general level right.
+ // Basically, if it's over about -40 dBFS, we squeeze it down to that level
+ // (or more precisely, near it, since we don't use infinite ratio),
+ // then apply a makeup gain to get it to -14 dBFS. -14 dBFS is, of course,
+ // entirely arbitrary, but from practical tests with speech, it seems to
+ // put ut around -23 LUFS, so it's a reasonable starting point for later use.
+ if (level_compressor_enabled[bus_index]) {
float threshold = 0.01f; // -40 dBFS.
float ratio = 20.0f;
float attack_time = 0.5f;
float release_time = 20.0f;
float makeup_gain = from_db(ref_level_dbfs - (-40.0f)); // +26 dB.
- level_compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
- gain_staging_db = to_db(level_compressor.get_attenuation() * makeup_gain);
+ level_compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
+ gain_staging_db[bus_index] = to_db(level_compressor[bus_index]->get_attenuation() * makeup_gain);
} else {
// Just apply the gain we already had.
- float g = from_db(gain_staging_db);
- for (size_t i = 0; i < samples_out.size(); ++i) {
- samples_out[i] *= g;
+ float g = from_db(gain_staging_db[bus_index]);
+ for (size_t i = 0; i < samples_bus.size(); ++i) {
+ samples_bus[i] *= g;
}
}
+
+#if 0
+ printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
+ level_compressor.get_level(), to_db(level_compressor.get_level()),
+ level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()),
+ to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
+#endif
+
+ // The real compressor.
+ if (compressor_enabled[bus_index]) {
+ float threshold = from_db(compressor_threshold_dbfs[bus_index]);
+ float ratio = 20.0f;
+ float attack_time = 0.005f;
+ float release_time = 0.040f;
+ float makeup_gain = 2.0f; // +6 dB.
+ compressor[bus_index]->process(samples_bus.data(), samples_bus.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
+ // compressor_att = compressor.get_attenuation();
+ }
}
- #if 0
- printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
- level_compressor.get_level(), to_db(level_compressor.get_level()),
- level_compressor.get_attenuation(), to_db(level_compressor.get_attenuation()),
- to_db(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
- #endif
-
- // float limiter_att, compressor_att;
-
- // The real compressor.
- if (compressor_enabled) {
- float threshold = from_db(compressor_threshold_dbfs);
- float ratio = 20.0f;
- float attack_time = 0.005f;
- float release_time = 0.040f;
- float makeup_gain = 2.0f; // +6 dB.
- compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
- // compressor_att = compressor.get_attenuation();
+ // 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) {
+ samples_out[i] = samples_bus[i] * volume;
+ }
+ } else {
+ for (unsigned i = 0; i < num_samples * 2; ++i) {
+ samples_out[i] += samples_bus[i] * volume;
+ }
}
+ }
+
+ {
+ lock_guard<mutex> lock(compressor_mutex);
// Finally a limiter at -4 dB (so, -10 dBFS) to take out the worst peaks only.
// Note that since ratio is not infinite, we could go slightly higher than this.
// printf("limiter=%+5.1f compressor=%+5.1f\n", to_db(limiter_att), to_db(compressor_att));
}
- // At this point, we are most likely close to +0 LU, but all of our
+ // At this point, we are most likely close to +0 LU (at least if the
+ // faders sum to 0 dB and the compressors are on), but all of our
// measurements have been on raw sample values, not R128 values.
// So we have a final makeup gain to get us to +0 LU; the gain
// adjustments required should be relatively small, and also, the
// 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<BusLevel> bus_levels;
+ bus_levels.resize(input_mapping.buses.size());
+ {
+ lock_guard<mutex> lock(compressor_mutex);
+ for (unsigned bus_index = 0; bus_index < bus_r128.size(); ++bus_index) {
+ bus_levels[bus_index].loudness_lufs = bus_r128[bus_index]->loudness_S();
+ bus_levels[bus_index].gain_staging_db = gain_staging_db[bus_index];
+ if (compressor_enabled[bus_index]) {
+ bus_levels[bus_index].compressor_attenuation_db = -to_db(compressor[bus_index]->get_attenuation());
+ } else {
+ bus_levels[bus_index].compressor_attenuation_db = 0.0;
+ }
+ }
+ }
+
+ audio_level_callback(loudness_s, to_db(peak), bus_levels,
+ loudness_i, loudness_range_low, loudness_range_high,
+ to_db(final_makeup_gain),
+ correlation.get_correlation());
+}
+
map<DeviceSpec, DeviceInfo> AudioMixer::get_devices() const
{
lock_guard<timed_mutex> lock(audio_mutex);
}
}
+ {
+ 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;
}
projects / nageru / blobdiff