#include "audio_mixer.h"
#include <assert.h>
-#include <endian.h>
#include <bmusb/bmusb.h>
-#include <stdio.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 "state.pb.h"
#include "timebase.h"
using namespace bmusb;
using namespace std;
+using namespace std::placeholders;
namespace {
}
}
+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_cards)
: num_cards(num_cards),
- level_compressor(OUTPUT_FREQUENCY),
limiter(OUTPUT_FREQUENCY),
- compressor(OUTPUT_FREQUENCY)
+ correlation(OUTPUT_FREQUENCY)
{
- locut.init(FILTER_HPF, 2);
+ global_audio_mixer = this;
- 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);
+ 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);
+ alsa_pool.init();
+
+ if (!global_flags.input_mapping_filename.empty()) {
+ 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;
+ }
+ }
- // Generate a very simple, default input mapping.
- InputMapping::Bus input;
- input.name = "Main";
- input.device.type = InputSourceType::CAPTURE_CARD;
- input.device.index = 0;
- input.source_channel[0] = 0;
- input.source_channel[1] = 1;
+ r128.init(2, OUTPUT_FREQUENCY);
+ r128.integr_start();
- InputMapping new_input_mapping;
- new_input_mapping.buses.push_back(input);
- set_input_mapping(new_input_mapping);
+ // 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);
}
-void AudioMixer::reset_device(DeviceSpec device_spec)
+void AudioMixer::reset_resampler(DeviceSpec device_spec)
{
- lock_guard<mutex> lock(audio_mutex);
- reset_device_mutex_held(device_spec);
+ lock_guard<timed_mutex> lock(audio_mutex);
+ reset_resampler_mutex_held(device_spec);
}
-void AudioMixer::reset_device_mutex_held(DeviceSpec 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->next_local_pts = 0;
}
-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();
assert(num_channels > 0);
// Convert the audio to fp32.
- vector<float> audio;
- audio.resize(num_samples * num_channels);
+ 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) {
assert(num_samples == 0);
break;
case 16:
- convert_fixed16_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
+ 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[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
+ 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[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
+ 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);
// Now add it.
int64_t local_pts = device->next_local_pts;
- device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.data(), num_samples);
+ device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.get(), 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;
+}
+
+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;
+ 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];
+ 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;
+ 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::SILENCE:
default:
assert(false);
return nullptr;
}
-void AudioMixer::find_sample_src_from_device(const vector<float> *samples_card, DeviceSpec device_spec, int source_channel, const float **srcptr, unsigned *stride)
+// 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) {
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());
- *srcptr = &samples_card[device_spec.index][channel_index];
+ 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 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 *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);
+ assert(bus.device.type == InputSourceType::CAPTURE_CARD ||
+ bus.device.type == InputSourceType::ALSA_INPUT);
const float *lsrc, *rsrc;
unsigned lstride, rstride;
float *dptr = output;
}
}
+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);
+ }
+ }
+ 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)
{
- vector<float> samples_card[MAX_VIDEO_CARDS]; // TODO: Needs room for other kinds of capture cards.
+ 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
- // might have changed; if so, we need to do some sort of remapping
- // to silence.
- for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
- AudioDevice *device = &video_cards[card_index];
- if (!device->interesting_channels.empty()) {
- samples_card[card_index].resize(num_samples * device->interesting_channels.size());
+ 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(
pts,
- &samples_card[card_index][0],
+ &samples_card[device_spec][0],
num_samples,
rate_adjustment_policy);
}
}
- // 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]);
+ apply_eq(bus_index, &samples_bus);
- 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) {
- 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 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();
}
}
- #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();
- }
+ 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.
// 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 current_makeup_lu = to_db(final_makeup_gain);
+ 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 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
final_makeup_gain = m;
}
+ update_meters(samples_out);
+
return samples_out;
}
-map<DeviceSpec, DeviceInfo> AudioMixer::get_devices() const
+namespace {
+
+void apply_filter_fade(StereoFilter *filter, float *data, unsigned num_samples, float cutoff_hz, float db, float last_db)
{
- lock_guard<mutex> lock(audio_mutex);
+ // 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();
+
+ 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(
+ 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];
+ 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()
+{
+ lock_guard<timed_mutex> lock(audio_mutex);
+
map<DeviceSpec, DeviceInfo> devices;
for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
const DeviceSpec spec{ InputSourceType::CAPTURE_CARD, card_index };
const AudioDevice *device = &video_cards[card_index];
DeviceInfo info;
- info.name = device->name;
- info.num_channels = 8; // FIXME: This is wrong for fake cards.
+ 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));
}
return devices;
}
-void AudioMixer::set_name(DeviceSpec device_spec, const string &name)
+void AudioMixer::set_display_name(DeviceSpec device_spec, const string &name)
{
AudioDevice *device = find_audio_device(device_spec);
- lock_guard<mutex> lock(audio_mutex);
- device->name = name;
+ 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;
+ }
+}
+
+void AudioMixer::set_simple_input(unsigned card_index)
+{
+ InputMapping new_input_mapping;
+ InputMapping::Bus input;
+ input.name = "Main";
+ input.device.type = InputSourceType::CAPTURE_CARD;
+ input.device.index = 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 {
+ return numeric_limits<unsigned>::max();
+ }
}
void AudioMixer::set_input_mapping(const InputMapping &new_input_mapping)
{
- lock_guard<mutex> lock(audio_mutex);
+ 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) {
+ if (bus.device.type == InputSourceType::CAPTURE_CARD ||
+ bus.device.type == InputSourceType::ALSA_INPUT) {
for (unsigned channel = 0; channel < 2; ++channel) {
if (bus.source_channel[channel] != -1) {
interesting_channels[bus.device].insert(bus.source_channel[channel]);
}
// Reset resamplers for all cards that don't have the exact same state as before.
- for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
- DeviceSpec device_spec{InputSourceType::CAPTURE_CARD, card_index};
- AudioDevice *device = &video_cards[card_index];
+ 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_device_mutex_held(DeviceSpec{InputSourceType::CAPTURE_CARD, card_index});
+ 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);
}
}
InputMapping AudioMixer::get_input_mapping() const
{
- lock_guard<mutex> lock(audio_mutex);
+ 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;
+ }
+}
+
+AudioMixer *global_audio_mixer = nullptr;
projects / nageru / blobdiff