#include "audio_mixer.h"
#include <assert.h>
-#include <endian.h>
#include <bmusb/bmusb.h>
-#include <stdio.h>
#include <endian.h>
-#include <cmath>
-#include <limits>
-#ifdef __SSE__
+#include <math.h>
+#ifdef __SSE2__
#include <immintrin.h>
#endif
+#include <stdbool.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <algorithm>
+#include <chrono>
+#include <cmath>
+#include <cstddef>
+#include <limits>
+#include <utility>
#include "db.h"
#include "flags.h"
-#include "mixer.h"
#include "state.pb.h"
#include "timebase.h"
{
BusSettings settings;
settings.fader_volume_db = 0.0f;
+ settings.muted = false;
settings.locut_enabled = global_flags.locut_enabled;
for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
settings.eq_level_db[band_index] = 0.0f;
lock_guard<timed_mutex> lock(audio_mutex);
BusSettings settings;
settings.fader_volume_db = fader_volume_db[bus_index];
+ settings.muted = mute[bus_index];
settings.locut_enabled = locut_enabled[bus_index];
for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
settings.eq_level_db[band_index] = eq_level_db[bus_index][band_index];
{
lock_guard<timed_mutex> lock(audio_mutex);
fader_volume_db[bus_index] = settings.fader_volume_db;
+ mute[bus_index] = settings.muted;
locut_enabled[bus_index] = settings.locut_enabled;
for (unsigned band_index = 0; band_index < NUM_EQ_BANDS; ++band_index) {
eq_level_db[bus_index][band_index] = settings.eq_level_db[band_index];
}
gain_staging_db[bus_index] = settings.gain_staging_db;
+ last_gain_staging_db[bus_index] = gain_staging_db[bus_index];
level_compressor_enabled[bus_index] = settings.level_compressor_enabled;
compressor_threshold_dbfs[bus_index] = settings.compressor_threshold_dbfs;
compressor_enabled[bus_index] = settings.compressor_enabled;
void AudioMixer::fill_audio_bus(const map<DeviceSpec, vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output)
{
if (bus.device.type == InputSourceType::SILENCE) {
- memset(output, 0, num_samples * sizeof(*output));
+ memset(output, 0, num_samples * 2 * sizeof(*output));
} else {
assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
bus.device.type == InputSourceType::ALSA_INPUT);
return ret;
}
+namespace {
+
+void apply_gain(float db, float last_db, vector<float> *samples)
+{
+ if (fabs(db - last_db) < 1e-3) {
+ // Constant over this frame.
+ const float gain = from_db(db);
+ for (size_t i = 0; i < samples->size(); ++i) {
+ (*samples)[i] *= gain;
+ }
+ } else {
+ // We need to do a fade.
+ unsigned num_samples = samples->size() / 2;
+ float gain = from_db(last_db);
+ const float gain_inc = pow(from_db(db - last_db), 1.0 / num_samples);
+ for (size_t i = 0; i < num_samples; ++i) {
+ (*samples)[i * 2 + 0] *= gain;
+ (*samples)[i * 2 + 1] *= gain;
+ gain *= gain_inc;
+ }
+ }
+}
+
+} // namespace
+
vector<float> AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
{
map<DeviceSpec, vector<float>> samples_card;
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[bus_index]);
- for (size_t i = 0; i < samples_bus.size(); ++i) {
- samples_bus[i] *= g;
- }
+ float db = gain_staging_db[bus_index];
+ float last_db = last_gain_staging_db[bus_index];
+ apply_gain(db, last_db, &samples_bus);
}
+ last_gain_staging_db[bus_index] = gain_staging_db[bus_index];
#if 0
printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
// (half-time of 30 seconds).
double target_loudness_factor, alpha;
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);
- // If we're outside +/- 5 LU uncorrected, we don't count it as
+ // If we're outside +/- 5 LU (after correction), we don't count it as
// a normal signal (probably silence) and don't change the
// correction factor; just apply what we already have.
- if (fabs(loudness_lu - current_makeup_lu) >= 5.0 || !final_makeup_gain_auto) {
+ if (fabs(loudness_lu) >= 5.0 || !final_makeup_gain_auto) {
alpha = 0.0;
} else {
// Formula adapted from
assert(samples_bus->size() % 2 == 0);
const unsigned num_samples = samples_bus->size() / 2;
- if (fabs(mid_db - last_mid_db) < 1e-3) {
- // Constant over this frame.
- const float gain = from_db(mid_db);
- for (size_t i = 0; i < samples_bus->size(); ++i) {
- (*samples_bus)[i] *= gain;
- }
- } else {
- // We need to do a fade.
- float gain = from_db(last_mid_db);
- const float gain_inc = pow(from_db(mid_db - last_mid_db), 1.0 / num_samples);
- for (size_t i = 0; i < num_samples; ++i) {
- (*samples_bus)[i * 2 + 0] *= gain;
- (*samples_bus)[i * 2 + 1] *= gain;
- gain *= gain_inc;
- }
- }
+ apply_gain(mid_db, last_mid_db, samples_bus);
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);
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);
assert(samples_bus.size() == samples_out->size());
assert(samples_bus.size() % 2 == 0);
unsigned num_samples = samples_bus.size() / 2;
- if (fabs(fader_volume_db[bus_index] - last_fader_volume_db[bus_index]) > 1e-3) {
+ const float new_volume_db = mute[bus_index] ? -90.0f : fader_volume_db[bus_index].load();
+ if (fabs(new_volume_db - last_fader_volume_db[bus_index]) > 1e-3) {
// The volume has changed; do a fade over the course of this frame.
// (We might have some numerical issues here, but it seems to sound OK.)
// For the purpose of fading here, the silence floor is set to -90 dB
// (the fader only goes to -84).
float old_volume = from_db(max<float>(last_fader_volume_db[bus_index], -90.0f));
- float volume = from_db(max<float>(fader_volume_db[bus_index], -90.0f));
+ float volume = from_db(max<float>(new_volume_db, -90.0f));
float volume_inc = pow(volume / old_volume, 1.0 / num_samples);
volume = old_volume;
volume *= volume_inc;
}
}
- } else {
- float volume = from_db(fader_volume_db[bus_index]);
+ } else if (new_volume_db > -90.0f) {
+ float volume = from_db(new_volume_db);
if (bus_index == 0) {
for (unsigned i = 0; i < num_samples; ++i) {
(*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
}
}
- last_fader_volume_db[bus_index] = fader_volume_db[bus_index];
+ last_fader_volume_db[bus_index] = new_volume_db;
}
void AudioMixer::measure_bus_levels(unsigned bus_index, const vector<float> &left, const vector<float> &right)
{
assert(left.size() == right.size());
- const float volume = from_db(fader_volume_db[bus_index]);
+ const float volume = mute[bus_index] ? 0.0f : from_db(fader_volume_db[bus_index]);
const float peak_levels[2] = {
find_peak(left.data(), left.size()) * volume,
find_peak(right.data(), right.size()) * volume
return input_mapping;
}
+unsigned AudioMixer::num_buses() const
+{
+ lock_guard<timed_mutex> lock(audio_mutex);
+ return input_mapping.buses.size();
+}
+
void AudioMixer::reset_peak(unsigned bus_index)
{
lock_guard<timed_mutex> lock(audio_mutex);
projects / nageru / blobdiff