+++ /dev/null
-#include "audio_mixer.h"
-
-#include <assert.h>
-#include <bmusb/bmusb.h>
-#include <endian.h>
-#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 "metrics.h"
-#include "state.pb.h"
-#include "timebase.h"
-
-using namespace bmusb;
-using namespace std;
-using namespace std::chrono;
-using namespace std::placeholders;
-
-namespace {
-
-// TODO: If these prove to be a bottleneck, they can be SSSE3-optimized
-// (usually including multiple channels at a time).
-
-void convert_fixed16_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
- const uint8_t *src, size_t in_channel, size_t in_num_channels,
- size_t num_samples)
-{
- assert(in_channel < in_num_channels);
- assert(out_channel < out_num_channels);
- src += in_channel * 2;
- dst += out_channel;
-
- for (size_t i = 0; i < num_samples; ++i) {
- int16_t s = le16toh(*(int16_t *)src);
- *dst = s * (1.0f / 32768.0f);
-
- src += 2 * in_num_channels;
- dst += out_num_channels;
- }
-}
-
-void convert_fixed24_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
- const uint8_t *src, size_t in_channel, size_t in_num_channels,
- size_t num_samples)
-{
- assert(in_channel < in_num_channels);
- assert(out_channel < out_num_channels);
- src += in_channel * 3;
- dst += out_channel;
-
- for (size_t i = 0; i < num_samples; ++i) {
- uint32_t s1 = src[0];
- uint32_t s2 = src[1];
- uint32_t s3 = src[2];
- uint32_t s = s1 | (s1 << 8) | (s2 << 16) | (s3 << 24);
- *dst = int(s) * (1.0f / 2147483648.0f);
-
- src += 3 * in_num_channels;
- dst += out_num_channels;
- }
-}
-
-void convert_fixed32_to_fp32(float *dst, size_t out_channel, size_t out_num_channels,
- const uint8_t *src, size_t in_channel, size_t in_num_channels,
- size_t num_samples)
-{
- assert(in_channel < in_num_channels);
- assert(out_channel < out_num_channels);
- src += in_channel * 4;
- dst += out_channel;
-
- for (size_t i = 0; i < num_samples; ++i) {
- int32_t s = le32toh(*(int32_t *)src);
- *dst = s * (1.0f / 2147483648.0f);
-
- src += 4 * in_num_channels;
- dst += out_num_channels;
- }
-}
-
-float find_peak_plain(const float *samples, size_t num_samples) __attribute__((unused));
-
-float find_peak_plain(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;
-}
-
-#ifdef __SSE__
-static inline float horizontal_max(__m128 m)
-{
- __m128 tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 0, 3, 2));
- m = _mm_max_ps(m, tmp);
- tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 3, 0, 1));
- m = _mm_max_ps(m, tmp);
- return _mm_cvtss_f32(m);
-}
-
-float find_peak(const float *samples, size_t num_samples)
-{
- const __m128 abs_mask = _mm_castsi128_ps(_mm_set1_epi32(0x7fffffffu));
- __m128 m = _mm_setzero_ps();
- for (size_t i = 0; i < (num_samples & ~3); i += 4) {
- __m128 x = _mm_loadu_ps(samples + i);
- x = _mm_and_ps(x, abs_mask);
- m = _mm_max_ps(m, x);
- }
- float result = horizontal_max(m);
-
- for (size_t i = (num_samples & ~3); i < num_samples; ++i) {
- result = max(result, fabs(samples[i]));
- }
-
-#if 0
- // Self-test. We should be bit-exact the same.
- float reference_result = find_peak_plain(samples, num_samples);
- if (result != reference_result) {
- fprintf(stderr, "Error: Peak is %f [%f %f %f %f]; should be %f.\n",
- result,
- _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(0, 0, 0, 0))),
- _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 1, 1, 1))),
- _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 2, 2, 2))),
- _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(3, 3, 3, 3))),
- reference_result);
- abort();
- }
-#endif
- return result;
-}
-#else
-float find_peak(const float *samples, size_t num_samples)
-{
- return find_peak_plain(samples, num_samples);
-}
-#endif
-
-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_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)
-{
- 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);
- // Note: EQ_BAND_MID isn't used (see comments in apply_eq()).
- eq[bus_index][EQ_BAND_TREBLE].init(FILTER_HIGH_SHELF, 1);
- compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
- level_compressor[bus_index].reset(new StereoCompressor(OUTPUT_FREQUENCY));
-
- set_bus_settings(bus_index, get_default_bus_settings());
- }
- 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(),
- global_flags.input_mapping_filename,
- &new_input_mapping)) {
- fprintf(stderr, "Failed to load input mapping from '%s', exiting.\n",
- global_flags.input_mapping_filename.c_str());
- exit(1);
- }
- set_input_mapping(new_input_mapping);
- } else {
- set_simple_input(/*card_index=*/0);
- if (global_flags.multichannel_mapping_mode) {
- current_mapping_mode = MappingMode::MULTICHANNEL;
- }
- }
-
- 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)
-{
- lock_guard<timed_mutex> lock(audio_mutex);
- reset_resampler_mutex_held(device_spec);
-}
-
-void AudioMixer::reset_resampler_mutex_held(DeviceSpec device_spec)
-{
- AudioDevice *device = find_audio_device(device_spec);
-
- if (device->interesting_channels.empty()) {
- device->resampling_queue.reset();
- } else {
- 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));
- }
-}
-
-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);
-
- 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 true;
- }
-
- unsigned num_channels = device->interesting_channels.size();
- assert(num_channels > 0);
-
- // Convert the audio to fp32.
- unique_ptr<float[]> audio(new float[num_samples * num_channels]);
- unsigned channel_index = 0;
- for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) {
- switch (audio_format.bits_per_sample) {
- case 0:
- assert(num_samples == 0);
- break;
- case 16:
- convert_fixed16_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
- break;
- case 24:
- convert_fixed24_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
- break;
- case 32:
- convert_fixed32_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
- break;
- default:
- fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
- assert(false);
- }
- }
-
- // 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.
- device->resampling_queue->add_input_samples(frame_time, audio.get(), num_samples, ResamplingQueue::ADJUST_RATE);
- return true;
-}
-
-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);
-
- 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 true;
- }
-
- unsigned num_channels = device->interesting_channels.size();
- assert(num_channels > 0);
-
- vector<float> silence(samples_per_frame * num_channels, 0.0f);
- for (unsigned i = 0; i < num_frames; ++i) {
- device->resampling_queue->add_input_samples(steady_clock::now(), silence.data(), samples_per_frame, ResamplingQueue::DO_NOT_ADJUST_RATE);
- }
- return true;
-}
-
-bool AudioMixer::silence_card(DeviceSpec device_spec, bool silence)
-{
- AudioDevice *device = find_audio_device(device_spec);
-
- unique_lock<timed_mutex> lock(audio_mutex, defer_lock);
- if (!lock.try_lock_for(chrono::milliseconds(10))) {
- return false;
- }
-
- if (device->silenced && !silence) {
- reset_resampler_mutex_held(device_spec);
- }
- device->silenced = silence;
- return true;
-}
-
-AudioMixer::BusSettings AudioMixer::get_default_bus_settings()
-{
- 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;
- }
- settings.gain_staging_db = global_flags.initial_gain_staging_db;
- settings.level_compressor_enabled = global_flags.gain_staging_auto;
- settings.compressor_threshold_dbfs = ref_level_dbfs - 12.0f; // -12 dB.
- settings.compressor_enabled = global_flags.compressor_enabled;
- return settings;
-}
-
-AudioMixer::BusSettings AudioMixer::get_bus_settings(unsigned bus_index) const
-{
- 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];
- }
- settings.gain_staging_db = gain_staging_db[bus_index];
- settings.level_compressor_enabled = level_compressor_enabled[bus_index];
- settings.compressor_threshold_dbfs = compressor_threshold_dbfs[bus_index];
- settings.compressor_enabled = compressor_enabled[bus_index];
- return settings;
-}
-
-void AudioMixer::set_bus_settings(unsigned bus_index, const AudioMixer::BusSettings &settings)
-{
- 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];
- }
- 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;
-}
-
-AudioMixer::AudioDevice *AudioMixer::find_audio_device(DeviceSpec device)
-{
- switch (device.type) {
- case InputSourceType::CAPTURE_CARD:
- 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);
- }
- return nullptr;
-}
-
-// Get a pointer to the given channel from the given device.
-// The channel must be picked out earlier and resampled.
-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)
-{
- static float zero = 0.0f;
- if (source_channel == -1 || device_spec.type == InputSourceType::SILENCE) {
- *srcptr = &zero;
- *stride = 0;
- return;
- }
- AudioDevice *device = find_audio_device(device_spec);
- assert(device->interesting_channels.count(source_channel) != 0);
- unsigned channel_index = 0;
- for (int channel : device->interesting_channels) {
- if (channel == source_channel) break;
- ++channel_index;
- }
- assert(channel_index < device->interesting_channels.size());
- const auto it = samples_card.find(device_spec);
- assert(it != samples_card.end());
- *srcptr = &(it->second)[channel_index];
- *stride = device->interesting_channels.size();
-}
-
-// 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 stereo_width, float *output)
-{
- if (bus.device.type == InputSourceType::SILENCE) {
- 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::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);
-
- // 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;
- }
- }
- }
-}
-
-vector<DeviceSpec> AudioMixer::get_active_devices() const
-{
- vector<DeviceSpec> ret;
- for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
- const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
- if (!find_audio_device(device_spec)->interesting_channels.empty()) {
- ret.push_back(device_spec);
- }
- }
- for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
- const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
- if (!find_audio_device(device_spec)->interesting_channels.empty()) {
- 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 {
-
-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(steady_clock::time_point ts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
-{
- map<DeviceSpec, vector<float>> samples_card;
- vector<float> samples_bus;
-
- lock_guard<timed_mutex> lock(audio_mutex);
-
- // Pick out all the interesting channels from all the cards.
- for (const DeviceSpec &device_spec : get_active_devices()) {
- AudioDevice *device = find_audio_device(device_spec);
- samples_card[device_spec].resize(num_samples * device->interesting_channels.size());
- if (device->silenced) {
- memset(&samples_card[device_spec][0], 0, samples_card[device_spec].size() * sizeof(float));
- } else {
- device->resampling_queue->get_output_samples(
- ts,
- &samples_card[device_spec][0],
- num_samples,
- rate_adjustment_policy);
- }
- }
-
- 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, stereo_width[bus_index], &samples_bus[0]);
- apply_eq(bus_index, &samples_bus);
-
- {
- 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[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 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",
- 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();
- }
- }
-
- add_bus_to_master(bus_index, samples_bus, &samples_out);
- deinterleave_samples(samples_bus, &left, &right);
- measure_bus_levels(bus_index, left, right);
- }
-
- {
- 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.
- if (limiter_enabled) {
- float threshold = from_db(limiter_threshold_dbfs);
- float ratio = 30.0f;
- float attack_time = 0.0f; // Instant.
- float release_time = 0.020f;
- float makeup_gain = 1.0f; // 0 dB.
- limiter.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
- // limiter_att = limiter.get_attenuation();
- }
-
- // 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 (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
- // offset shouldn't change much (only if the type of audio changes
- // significantly). Thus, we shoot for updating this value basically
- // “whenever we process buffers”, since the R128 calculation isn't exactly
- // something we get out per-sample.
- //
- // 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 = r128.loudness_M() - ref_level_lufs;
- target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
-
- // 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) >= 5.0 || !final_makeup_gain_auto) {
- alpha = 0.0;
- } else {
- // Formula adapted from
- // https://en.wikipedia.org/wiki/Low-pass_filter#Simple_infinite_impulse_response_filter.
- const double half_time_s = 30.0;
- const double fc_mul_2pi_delta_t = 1.0 / (half_time_s * OUTPUT_FREQUENCY);
- alpha = fc_mul_2pi_delta_t / (fc_mul_2pi_delta_t + 1.0);
- }
-
- {
- lock_guard<mutex> lock(compressor_mutex);
- double m = final_makeup_gain;
- for (size_t i = 0; i < samples_out.size(); i += 2) {
- samples_out[i + 0] *= m;
- samples_out[i + 1] *= m;
- m += (target_loudness_factor - m) * alpha;
- }
- final_makeup_gain = m;
- }
-
- update_meters(samples_out);
-
- return samples_out;
-}
-
-namespace {
-
-void apply_filter_fade(StereoFilter *filter, float *data, unsigned num_samples, float cutoff_hz, float db, float last_db)
-{
- // A granularity of 32 samples is an okay tradeoff between speed and
- // smoothness; recalculating the filters is pretty expensive, so it's
- // good that we don't do this all the time.
- static constexpr unsigned filter_granularity_samples = 32;
-
- const float cutoff_linear = cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY;
- if (fabs(db - last_db) < 1e-3) {
- // Constant over this frame.
- if (fabs(db) > 0.01f) {
- filter->render(data, num_samples, cutoff_linear, 0.5f, db / 40.0f);
- }
- } else {
- // We need to do a fade. (Rounding up avoids division by zero.)
- unsigned num_blocks = (num_samples + filter_granularity_samples - 1) / filter_granularity_samples;
- const float inc_db_norm = (db - last_db) / 40.0f / num_blocks;
- float db_norm = db / 40.0f;
- for (size_t i = 0; i < num_samples; i += filter_granularity_samples) {
- size_t samples_this_block = std::min<size_t>(num_samples - i, filter_granularity_samples);
- filter->render(data + i * 2, samples_this_block, cutoff_linear, 0.5f, db_norm);
- db_norm += inc_db_norm;
- }
- }
-}
-
-} // namespace
-
-void AudioMixer::apply_eq(unsigned bus_index, vector<float> *samples_bus)
-{
- constexpr float bass_freq_hz = 200.0f;
- constexpr float treble_freq_hz = 4700.0f;
-
- // 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);
- }
-
- // Apply the rest of the EQ. Since we only have a simple three-band EQ,
- // we can implement it with two shelf filters. We use a simple gain to
- // set the mid-level filter, and then offset the low and high bands
- // from that if we need to. (We could perhaps have folded the gain into
- // the next part, but it's so cheap that the trouble isn't worth it.)
- //
- // If any part of the EQ has changed appreciably since last frame,
- // we fade smoothly during the course of this frame.
- const float bass_db = eq_level_db[bus_index][EQ_BAND_BASS];
- const float mid_db = eq_level_db[bus_index][EQ_BAND_MID];
- const float treble_db = eq_level_db[bus_index][EQ_BAND_TREBLE];
-
- const float last_bass_db = last_eq_level_db[bus_index][EQ_BAND_BASS];
- const float last_mid_db = last_eq_level_db[bus_index][EQ_BAND_MID];
- const float last_treble_db = last_eq_level_db[bus_index][EQ_BAND_TREBLE];
-
- assert(samples_bus->size() % 2 == 0);
- const unsigned num_samples = samples_bus->size() / 2;
-
- 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);
-
- last_eq_level_db[bus_index][EQ_BAND_BASS] = bass_db;
- last_eq_level_db[bus_index][EQ_BAND_MID] = mid_db;
- last_eq_level_db[bus_index][EQ_BAND_TREBLE] = treble_db;
-}
-
-void AudioMixer::add_bus_to_master(unsigned bus_index, const vector<float> &samples_bus, vector<float> *samples_out)
-{
- assert(samples_bus.size() == samples_out->size());
- assert(samples_bus.size() % 2 == 0);
- unsigned num_samples = samples_bus.size() / 2;
- 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>(new_volume_db, -90.0f));
-
- float volume_inc = pow(volume / old_volume, 1.0 / num_samples);
- volume = old_volume;
- if (bus_index == 0) {
- for (unsigned i = 0; i < num_samples; ++i) {
- (*samples_out)[i * 2 + 0] = samples_bus[i * 2 + 0] * volume;
- (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
- volume *= volume_inc;
- }
- } else {
- for (unsigned i = 0; i < num_samples; ++i) {
- (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
- (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
- volume *= volume_inc;
- }
- }
- } 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;
- (*samples_out)[i * 2 + 1] = samples_bus[i * 2 + 1] * volume;
- }
- } else {
- for (unsigned i = 0; i < num_samples; ++i) {
- (*samples_out)[i * 2 + 0] += samples_bus[i * 2 + 0] * volume;
- (*samples_out)[i * 2 + 1] += samples_bus[i * 2 + 1] * volume;
- }
- }
- }
-
- 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 = 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
- };
- for (unsigned channel = 0; channel < 2; ++channel) {
- // Compute the current value, including hold and falloff.
- // The constants are borrowed from zita-mu1 by Fons Adriaensen.
- static constexpr float hold_sec = 0.5f;
- static constexpr float falloff_db_sec = 15.0f; // dB/sec falloff after hold.
- float current_peak;
- PeakHistory &history = peak_history[bus_index][channel];
- history.historic_peak = max(history.historic_peak, peak_levels[channel]);
- if (history.age_seconds < hold_sec) {
- current_peak = history.last_peak;
- } else {
- current_peak = history.last_peak * from_db(-falloff_db_sec * (history.age_seconds - hold_sec));
- }
-
- // See if we have a new peak to replace the old (possibly falling) one.
- if (peak_levels[channel] > current_peak) {
- history.last_peak = peak_levels[channel];
- history.age_seconds = 0.0f; // Not 100% correct, but more than good enough given our frame sizes.
- current_peak = peak_levels[channel];
- } else {
- history.age_seconds += float(left.size()) / OUTPUT_FREQUENCY;
- }
- history.current_level = peak_levels[channel];
- history.current_peak = current_peak;
- }
-}
-
-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();
-
- 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) {
- 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));
- levels.gain_staging_db = metrics.gain_staging_db = gain_staging_db[bus_index];
- if (compressor_enabled[bus_index]) {
- levels.compressor_attenuation_db = metrics.compressor_attenuation_db = -to_db(compressor[bus_index]->get_attenuation());
- } else {
- levels.compressor_attenuation_db = 0.0;
- metrics.compressor_attenuation_db = 0.0 / 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()
-{
- lock_guard<timed_mutex> lock(audio_mutex);
-
- map<DeviceSpec, DeviceInfo> devices;
- 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.display_name = device->display_name;
- info.num_channels = 8;
- devices.insert(make_pair(spec, info));
- }
- vector<ALSAPool::Device> available_alsa_devices = alsa_pool.get_devices();
- for (unsigned card_index = 0; card_index < available_alsa_devices.size(); ++card_index) {
- const DeviceSpec spec{ InputSourceType::ALSA_INPUT, card_index };
- const ALSAPool::Device &device = available_alsa_devices[card_index];
- DeviceInfo info;
- info.display_name = device.display_name();
- info.num_channels = device.num_channels;
- info.alsa_name = device.name;
- info.alsa_info = device.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;
-}
-
-void AudioMixer::set_display_name(DeviceSpec device_spec, const string &name)
-{
- AudioDevice *device = find_audio_device(device_spec);
-
- lock_guard<timed_mutex> lock(audio_mutex);
- device->display_name = name;
-}
-
-void AudioMixer::serialize_device(DeviceSpec device_spec, DeviceSpecProto *device_spec_proto)
-{
- lock_guard<timed_mutex> lock(audio_mutex);
- switch (device_spec.type) {
- case InputSourceType::SILENCE:
- device_spec_proto->set_type(DeviceSpecProto::SILENCE);
- break;
- case InputSourceType::CAPTURE_CARD:
- device_spec_proto->set_type(DeviceSpecProto::CAPTURE_CARD);
- device_spec_proto->set_index(device_spec.index);
- device_spec_proto->set_display_name(video_cards[device_spec.index].display_name);
- break;
- 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";
- 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;
-
- new_input_mapping.buses.push_back(input);
-
- lock_guard<timed_mutex> lock(audio_mutex);
- current_mapping_mode = MappingMode::SIMPLE;
- set_input_mapping_lock_held(new_input_mapping);
- fader_volume_db[0] = 0.0f;
-}
-
-unsigned AudioMixer::get_simple_input() const
-{
- lock_guard<timed_mutex> lock(audio_mutex);
- if (input_mapping.buses.size() == 1 &&
- input_mapping.buses[0].device.type == InputSourceType::CAPTURE_CARD &&
- 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();
- }
-}
-
-void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
-{
- lock_guard<timed_mutex> lock(audio_mutex);
- set_input_mapping_lock_held(new_input_mapping);
- current_mapping_mode = MappingMode::MULTICHANNEL;
-}
-
-AudioMixer::MappingMode AudioMixer::get_mapping_mode() const
-{
- lock_guard<timed_mutex> lock(audio_mutex);
- return current_mapping_mode;
-}
-
-void AudioMixer::set_input_mapping_lock_held(const InputMapping &new_input_mapping)
-{
- 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::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.
- for (unsigned card_index = 0; card_index < MAX_VIDEO_CARDS; ++card_index) {
- const DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, 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);
- }
- }
- for (unsigned card_index = 0; card_index < MAX_ALSA_CARDS; ++card_index) {
- const DeviceSpec device_spec{InputSourceType::ALSA_INPUT, card_index};
- AudioDevice *device = find_audio_device(device_spec);
- if (interesting_channels[device_spec].empty()) {
- alsa_pool.release_device(card_index);
- } else {
- alsa_pool.hold_device(card_index);
- }
- if (device->interesting_channels != interesting_channels[device_spec]) {
- device->interesting_channels = interesting_channels[device_spec];
- alsa_pool.reset_device(device_spec.index);
- 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;
-}
-
-InputMapping AudioMixer::get_input_mapping() const
-{
- lock_guard<timed_mutex> lock(audio_mutex);
- 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);
- for (unsigned channel = 0; channel < 2; ++channel) {
- PeakHistory &history = peak_history[bus_index][channel];
- history.current_level = 0.0f;
- history.historic_peak = 0.0f;
- history.current_peak = 0.0f;
- history.last_peak = 0.0f;
- history.age_seconds = 0.0f;
- }
-}
-
-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