#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 "metrics.h"
#include "state.pb.h"
#include "timebase.h"
using namespace bmusb;
using namespace std;
+using namespace std::chrono;
using namespace std::placeholders;
namespace {
} // namespace
-AudioMixer::AudioMixer(unsigned num_cards)
- : num_cards(num_cards),
+AudioMixer::AudioMixer(unsigned num_capture_cards, unsigned num_ffmpeg_inputs)
+ : num_capture_cards(num_capture_cards),
+ num_ffmpeg_inputs(num_ffmpeg_inputs),
+ ffmpeg_inputs(new AudioDevice[num_ffmpeg_inputs]),
limiter(OUTPUT_FREQUENCY),
correlation(OUTPUT_FREQUENCY)
{
- global_audio_mixer = this;
-
for (unsigned bus_index = 0; bus_index < MAX_BUSES; ++bus_index) {
locut[bus_index].init(FILTER_HPF, 2);
eq[bus_index][EQ_BAND_BASS].init(FILTER_LOW_SHELF, 1);
}
set_limiter_enabled(global_flags.limiter_enabled);
set_final_makeup_gain_auto(global_flags.final_makeup_gain_auto);
+
+ 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);
+
+ global_audio_mixer = this;
alsa_pool.init();
if (!global_flags.input_mapping_filename.empty()) {
+ // Must happen after ALSAPool is initialized, as it needs to know the card list.
current_mapping_mode = MappingMode::MULTICHANNEL;
InputMapping new_input_mapping;
if (!load_input_mapping_from_file(get_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);
+ global_metrics.add("audio_loudness_short_lufs", &metric_audio_loudness_short_lufs, Metrics::TYPE_GAUGE);
+ global_metrics.add("audio_loudness_integrated_lufs", &metric_audio_loudness_integrated_lufs, Metrics::TYPE_GAUGE);
+ global_metrics.add("audio_loudness_range_low_lufs", &metric_audio_loudness_range_low_lufs, Metrics::TYPE_GAUGE);
+ global_metrics.add("audio_loudness_range_high_lufs", &metric_audio_loudness_range_high_lufs, Metrics::TYPE_GAUGE);
+ global_metrics.add("audio_peak_dbfs", &metric_audio_peak_dbfs, Metrics::TYPE_GAUGE);
+ global_metrics.add("audio_final_makeup_gain_db", &metric_audio_final_makeup_gain_db, Metrics::TYPE_GAUGE);
+ global_metrics.add("audio_correlation", &metric_audio_correlation, Metrics::TYPE_GAUGE);
}
void AudioMixer::reset_resampler(DeviceSpec device_spec)
if (device->interesting_channels.empty()) {
device->resampling_queue.reset();
} else {
- // TODO: ResamplingQueue should probably take the full device spec.
- // (It's only used for console output, though.)
- device->resampling_queue.reset(new ResamplingQueue(device_spec.index, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size()));
+ device->resampling_queue.reset(new ResamplingQueue(
+ device_spec, device->capture_frequency, OUTPUT_FREQUENCY, device->interesting_channels.size(),
+ global_flags.audio_queue_length_ms * 0.001));
}
- device->next_local_pts = 0;
}
-bool 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, steady_clock::time_point frame_time)
{
AudioDevice *device = find_audio_device(device_spec);
}
}
+ // If we changed frequency since last frame, we'll need to reset the resampler.
+ if (audio_format.sample_rate != device->capture_frequency) {
+ device->capture_frequency = audio_format.sample_rate;
+ reset_resampler_mutex_held(device_spec);
+ }
+
// Now add it.
- int64_t local_pts = device->next_local_pts;
- device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.get(), num_samples);
- device->next_local_pts = local_pts + frame_length;
+ device->resampling_queue->add_input_samples(frame_time, audio.get(), num_samples, ResamplingQueue::ADJUST_RATE);
return true;
}
vector<float> silence(samples_per_frame * num_channels, 0.0f);
for (unsigned i = 0; i < num_frames; ++i) {
- device->resampling_queue->add_input_samples(device->next_local_pts / double(TIMEBASE), silence.data(), samples_per_frame);
- // Note that if the format changed in the meantime, we have
- // no way of detecting that; we just have to assume the frame length
- // is always the same.
- device->next_local_pts += frame_length;
+ device->resampling_queue->add_input_samples(steady_clock::now(), silence.data(), samples_per_frame, ResamplingQueue::DO_NOT_ADJUST_RATE);
}
return true;
}
{
BusSettings settings;
settings.fader_volume_db = 0.0f;
+ settings.muted = false;
settings.locut_enabled = global_flags.locut_enabled;
+ settings.stereo_width = 1.0f;
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];
+ settings.stereo_width = stereo_width[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;
+ stereo_width[bus_index] = settings.stereo_width;
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];
}
return &video_cards[device.index];
case InputSourceType::ALSA_INPUT:
return &alsa_inputs[device.index];
+ case InputSourceType::FFMPEG_VIDEO_INPUT:
+ return &ffmpeg_inputs[device.index];
case InputSourceType::SILENCE:
default:
assert(false);
}
// TODO: Can be SSSE3-optimized if need be.
-void AudioMixer::fill_audio_bus(const map<DeviceSpec, vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output)
+void AudioMixer::fill_audio_bus(const map<DeviceSpec, vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float stereo_width, 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);
+ bus.device.type == InputSourceType::ALSA_INPUT ||
+ bus.device.type == InputSourceType::FFMPEG_VIDEO_INPUT);
const float *lsrc, *rsrc;
unsigned lstride, rstride;
float *dptr = output;
find_sample_src_from_device(samples_card, bus.device, bus.source_channel[0], &lsrc, &lstride);
find_sample_src_from_device(samples_card, bus.device, bus.source_channel[1], &rsrc, &rstride);
- for (unsigned i = 0; i < num_samples; ++i) {
- *dptr++ = *lsrc;
- *dptr++ = *rsrc;
- lsrc += lstride;
- rsrc += rstride;
+
+ // Apply stereo width settings. Set stereo width w to a 0..1 range instead of
+ // -1..1, since it makes for much easier calculations (so 0.5 = completely mono).
+ // Then, what we want is
+ //
+ // L' = wL + (1-w)R = R + w(L-R)
+ // R' = wR + (1-w)L = L + w(R-L)
+ //
+ // This can be further simplified calculation-wise by defining the weighted
+ // difference signal D = w(R-L), so that:
+ //
+ // L' = R - D
+ // R' = L + D
+ float w = 0.5f * stereo_width + 0.5f;
+ if (bus.source_channel[0] == bus.source_channel[1]) {
+ // Mono anyway, so no need to bother.
+ w = 1.0f;
+ } else if (fabs(w) < 1e-3) {
+ // Perfect inverse.
+ swap(lsrc, rsrc);
+ swap(lstride, rstride);
+ w = 1.0f;
+ }
+ if (fabs(w - 1.0f) < 1e-3) {
+ // No calculations needed for stereo_width = 1.
+ for (unsigned i = 0; i < num_samples; ++i) {
+ *dptr++ = *lsrc;
+ *dptr++ = *rsrc;
+ lsrc += lstride;
+ rsrc += rstride;
+ }
+ } else {
+ // General case.
+ for (unsigned i = 0; i < num_samples; ++i) {
+ float left = *lsrc, right = *rsrc;
+ float diff = w * (right - left);
+ *dptr++ = right - diff;
+ *dptr++ = left + diff;
+ lsrc += lstride;
+ rsrc += rstride;
+ }
}
}
}
ret.push_back(device_spec);
}
}
+ for (unsigned card_index = 0; card_index < num_ffmpeg_inputs; ++card_index) {
+ const DeviceSpec device_spec{InputSourceType::FFMPEG_VIDEO_INPUT, card_index};
+ if (!find_audio_device(device_spec)->interesting_channels.empty()) {
+ ret.push_back(device_spec);
+ }
+ }
return ret;
}
} // namespace
-vector<float> AudioMixer::get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
+vector<float> AudioMixer::get_output(steady_clock::time_point ts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
{
map<DeviceSpec, vector<float>> samples_card;
vector<float> samples_bus;
memset(&samples_card[device_spec][0], 0, samples_card[device_spec].size() * sizeof(float));
} else {
device->resampling_queue->get_output_samples(
- pts,
+ ts,
&samples_card[device_spec][0],
num_samples,
rate_adjustment_policy);
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]);
+ fill_audio_bus(samples_card, input_mapping.buses[bus_index], num_samples, stereo_width[bus_index], &samples_bus[0]);
apply_eq(bus_index, &samples_bus);
{
// (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() == 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
double loudness_range_low = r128.range_min();
double loudness_range_high = r128.range_max();
+ metric_audio_loudness_short_lufs = loudness_s;
+ metric_audio_loudness_integrated_lufs = loudness_i;
+ metric_audio_loudness_range_low_lufs = loudness_range_low;
+ metric_audio_loudness_range_high_lufs = loudness_range_high;
+ metric_audio_peak_dbfs = to_db(peak);
+ metric_audio_final_makeup_gain_db = to_db(final_makeup_gain);
+ metric_audio_correlation = correlation.get_correlation();
+
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_levels.size(); ++bus_index) {
- bus_levels[bus_index].current_level_dbfs[0] = to_db(peak_history[bus_index][0].current_level);
- bus_levels[bus_index].current_level_dbfs[1] = to_db(peak_history[bus_index][1].current_level);
- bus_levels[bus_index].peak_level_dbfs[0] = to_db(peak_history[bus_index][0].current_peak);
- bus_levels[bus_index].peak_level_dbfs[1] = to_db(peak_history[bus_index][1].current_peak);
- bus_levels[bus_index].historic_peak_dbfs = to_db(
+ BusLevel &levels = bus_levels[bus_index];
+ BusMetrics &metrics = bus_metrics[bus_index];
+
+ levels.current_level_dbfs[0] = metrics.current_level_dbfs[0] = to_db(peak_history[bus_index][0].current_level);
+ levels.current_level_dbfs[1] = metrics.current_level_dbfs[1] = to_db(peak_history[bus_index][1].current_level);
+ levels.peak_level_dbfs[0] = metrics.peak_level_dbfs[0] = to_db(peak_history[bus_index][0].current_peak);
+ levels.peak_level_dbfs[1] = metrics.peak_level_dbfs[1] = to_db(peak_history[bus_index][1].current_peak);
+ levels.historic_peak_dbfs = metrics.historic_peak_dbfs = to_db(
max(peak_history[bus_index][0].historic_peak,
peak_history[bus_index][1].historic_peak));
- bus_levels[bus_index].gain_staging_db = gain_staging_db[bus_index];
+ levels.gain_staging_db = metrics.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());
+ levels.compressor_attenuation_db = metrics.compressor_attenuation_db = -to_db(compressor[bus_index]->get_attenuation());
} else {
- bus_levels[bus_index].compressor_attenuation_db = 0.0;
+ levels.compressor_attenuation_db = 0.0;
+ metrics.compressor_attenuation_db = 0.0 / 0.0;
}
}
}
lock_guard<timed_mutex> lock(audio_mutex);
map<DeviceSpec, DeviceInfo> devices;
- for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
+ for (unsigned card_index = 0; card_index < num_capture_cards; ++card_index) {
const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index };
const AudioDevice *device = &video_cards[card_index];
DeviceInfo info;
info.alsa_address = device.address;
devices.insert(make_pair(spec, info));
}
+ for (unsigned card_index = 0; card_index < num_ffmpeg_inputs; ++card_index) {
+ const DeviceSpec spec{ InputSourceType::FFMPEG_VIDEO_INPUT, card_index };
+ const AudioDevice *device = &ffmpeg_inputs[card_index];
+ DeviceInfo info;
+ info.display_name = device->display_name;
+ info.num_channels = 2;
+ devices.insert(make_pair(spec, info));
+ }
return devices;
}
case InputSourceType::ALSA_INPUT:
alsa_pool.serialize_device(device_spec.index, device_spec_proto);
break;
+ case InputSourceType::FFMPEG_VIDEO_INPUT:
+ device_spec_proto->set_type(DeviceSpecProto::FFMPEG_VIDEO_INPUT);
+ device_spec_proto->set_index(device_spec.index);
+ device_spec_proto->set_display_name(ffmpeg_inputs[device_spec.index].display_name);
+ break;
}
}
void AudioMixer::set_simple_input(unsigned card_index)
{
+ assert(card_index < num_capture_cards + num_ffmpeg_inputs);
InputMapping new_input_mapping;
InputMapping::Bus input;
input.name = "Main";
- input.device.type = InputSourceType::CAPTURE_CARD;
- input.device.index = card_index;
+ if (card_index >= num_capture_cards) {
+ input.device = DeviceSpec{InputSourceType::FFMPEG_VIDEO_INPUT, card_index - num_capture_cards};
+ } else {
+ input.device = DeviceSpec{InputSourceType::CAPTURE_CARD, card_index};
+ }
input.source_channel[0] = 0;
input.source_channel[1] = 1;
input_mapping.buses[0].source_channel[0] == 0 &&
input_mapping.buses[0].source_channel[1] == 1) {
return input_mapping.buses[0].device.index;
+ } else if (input_mapping.buses.size() == 1 &&
+ input_mapping.buses[0].device.type == InputSourceType::FFMPEG_VIDEO_INPUT &&
+ input_mapping.buses[0].source_channel[0] == 0 &&
+ input_mapping.buses[0].source_channel[1] == 1) {
+ return input_mapping.buses[0].device.index + num_capture_cards;
} else {
return numeric_limits<unsigned>::max();
}
map<DeviceSpec, set<unsigned>> interesting_channels;
for (const InputMapping::Bus &bus : new_input_mapping.buses) {
if (bus.device.type == InputSourceType::CAPTURE_CARD ||
- bus.device.type == InputSourceType::ALSA_INPUT) {
+ bus.device.type == InputSourceType::ALSA_INPUT ||
+ bus.device.type == InputSourceType::FFMPEG_VIDEO_INPUT) {
for (unsigned channel = 0; channel < 2; ++channel) {
if (bus.source_channel[channel] != -1) {
interesting_channels[bus.device].insert(bus.source_channel[channel]);
}
}
+ } else {
+ assert(bus.device.type == InputSourceType::SILENCE);
+ }
+ }
+
+ // Kill all the old metrics, and set up new ones.
+ for (unsigned bus_index = 0; bus_index < input_mapping.buses.size(); ++bus_index) {
+ BusMetrics &metrics = bus_metrics[bus_index];
+
+ vector<pair<string, string>> labels_left = metrics.labels;
+ labels_left.emplace_back("channel", "left");
+ vector<pair<string, string>> labels_right = metrics.labels;
+ labels_right.emplace_back("channel", "right");
+
+ global_metrics.remove("bus_current_level_dbfs", labels_left);
+ global_metrics.remove("bus_current_level_dbfs", labels_right);
+ global_metrics.remove("bus_peak_level_dbfs", labels_left);
+ global_metrics.remove("bus_peak_level_dbfs", labels_right);
+ global_metrics.remove("bus_historic_peak_dbfs", metrics.labels);
+ global_metrics.remove("bus_gain_staging_db", metrics.labels);
+ global_metrics.remove("bus_compressor_attenuation_db", metrics.labels);
+ }
+ bus_metrics.reset(new BusMetrics[new_input_mapping.buses.size()]);
+ for (unsigned bus_index = 0; bus_index < new_input_mapping.buses.size(); ++bus_index) {
+ const InputMapping::Bus &bus = new_input_mapping.buses[bus_index];
+ BusMetrics &metrics = bus_metrics[bus_index];
+
+ char bus_index_str[16], source_index_str[16], source_channels_str[64];
+ snprintf(bus_index_str, sizeof(bus_index_str), "%u", bus_index);
+ snprintf(source_index_str, sizeof(source_index_str), "%u", bus.device.index);
+ snprintf(source_channels_str, sizeof(source_channels_str), "%d:%d", bus.source_channel[0], bus.source_channel[1]);
+
+ vector<pair<string, string>> labels;
+ metrics.labels.emplace_back("index", bus_index_str);
+ metrics.labels.emplace_back("name", bus.name);
+ if (bus.device.type == InputSourceType::SILENCE) {
+ metrics.labels.emplace_back("source_type", "silence");
+ } else if (bus.device.type == InputSourceType::CAPTURE_CARD) {
+ metrics.labels.emplace_back("source_type", "capture_card");
+ } else if (bus.device.type == InputSourceType::ALSA_INPUT) {
+ metrics.labels.emplace_back("source_type", "alsa_input");
+ } else if (bus.device.type == InputSourceType::FFMPEG_VIDEO_INPUT) {
+ metrics.labels.emplace_back("source_type", "ffmpeg_video_input");
+ } else {
+ assert(false);
}
+ metrics.labels.emplace_back("source_index", source_index_str);
+ metrics.labels.emplace_back("source_channels", source_channels_str);
+
+ vector<pair<string, string>> labels_left = metrics.labels;
+ labels_left.emplace_back("channel", "left");
+ vector<pair<string, string>> labels_right = metrics.labels;
+ labels_right.emplace_back("channel", "right");
+
+ global_metrics.add("bus_current_level_dbfs", labels_left, &metrics.current_level_dbfs[0], Metrics::TYPE_GAUGE);
+ global_metrics.add("bus_current_level_dbfs", labels_right, &metrics.current_level_dbfs[1], Metrics::TYPE_GAUGE);
+ global_metrics.add("bus_peak_level_dbfs", labels_left, &metrics.peak_level_dbfs[0], Metrics::TYPE_GAUGE);
+ global_metrics.add("bus_peak_level_dbfs", labels_right, &metrics.peak_level_dbfs[1], Metrics::TYPE_GAUGE);
+ global_metrics.add("bus_historic_peak_dbfs", metrics.labels, &metrics.historic_peak_dbfs, Metrics::TYPE_GAUGE);
+ global_metrics.add("bus_gain_staging_db", metrics.labels, &metrics.gain_staging_db, Metrics::TYPE_GAUGE);
+ global_metrics.add("bus_compressor_attenuation_db", metrics.labels, &metrics.compressor_attenuation_db, Metrics::TYPE_GAUGE);
}
// Reset resamplers for all cards that don't have the exact same state as before.
reset_resampler_mutex_held(device_spec);
}
}
+ for (unsigned card_index = 0; card_index < num_ffmpeg_inputs; ++card_index) {
+ const DeviceSpec device_spec{InputSourceType::FFMPEG_VIDEO_INPUT, card_index};
+ AudioDevice *device = find_audio_device(device_spec);
+ if (device->interesting_channels != interesting_channels[device_spec]) {
+ device->interesting_channels = interesting_channels[device_spec];
+ reset_resampler_mutex_held(device_spec);
+ }
+ }
input_mapping = new_input_mapping;
}
}
}
+bool AudioMixer::is_mono(unsigned bus_index)
+{
+ lock_guard<timed_mutex> lock(audio_mutex);
+ const InputMapping::Bus &bus = input_mapping.buses[bus_index];
+ if (bus.device.type == InputSourceType::SILENCE) {
+ return true;
+ } else {
+ assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
+ bus.device.type == InputSourceType::ALSA_INPUT ||
+ bus.device.type == InputSourceType::FFMPEG_VIDEO_INPUT);
+ return bus.source_channel[0] == bus.source_channel[1];
+ }
+}
+
AudioMixer *global_audio_mixer = nullptr;
projects / nageru / blobdiff