}
}
+float find_peak(const float *samples, size_t num_samples)
+{
+ float m = fabs(samples[0]);
+ for (size_t i = 1; i < num_samples; ++i) {
+ m = max(m, fabs(samples[i]));
+ }
+ return m;
+}
+
+void deinterleave_samples(const vector<float> &in, vector<float> *out_l, vector<float> *out_r)
+{
+ size_t num_samples = in.size() / 2;
+ out_l->resize(num_samples);
+ out_r->resize(num_samples);
+
+ const float *inptr = in.data();
+ float *lptr = &(*out_l)[0];
+ float *rptr = &(*out_r)[0];
+ for (size_t i = 0; i < num_samples; ++i) {
+ *lptr++ = *inptr++;
+ *rptr++ = *inptr++;
+ }
+}
+
} // namespace
AudioMixer::AudioMixer(unsigned num_cards)
: num_cards(num_cards),
level_compressor(OUTPUT_FREQUENCY),
limiter(OUTPUT_FREQUENCY),
- compressor(OUTPUT_FREQUENCY)
+ compressor(OUTPUT_FREQUENCY),
+ correlation(OUTPUT_FREQUENCY)
{
locut.init(FILTER_HPF, 2);
// Look for ALSA cards.
available_alsa_cards = ALSAInput::enumerate_devices();
+
+ r128.init(2, OUTPUT_FREQUENCY);
+ r128.integr_start();
+
+ // hlen=16 is pretty low quality, but we use quite a bit of CPU otherwise,
+ // and there's a limit to how important the peak meter is.
+ peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
}
AudioMixer::~AudioMixer()
// Note that there's a feedback loop here, so we choose a very slow filter
// (half-time of 30 seconds).
double target_loudness_factor, alpha;
- double loudness_lu = loudness_lufs - ref_level_lufs;
+ double loudness_lu = r128.loudness_M() - ref_level_lufs;
double current_makeup_lu = to_db(final_makeup_gain);
target_loudness_factor = final_makeup_gain * from_db(-loudness_lu);
final_makeup_gain = m;
}
+ update_meters(samples_out);
+
return samples_out;
}
+void AudioMixer::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());
+ {
+ unique_lock<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() };
+ {
+ unique_lock<mutex> lock(audio_measure_mutex);
+ r128.process(left.size(), ptrs);
+ correlation.process_samples(samples);
+ }
+
+ send_audio_level_callback();
+}
+
+void AudioMixer::reset_meters()
+{
+ unique_lock<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;
+ }
+
+ unique_lock<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();
+
+ audio_level_callback(loudness_s, to_db(peak),
+ loudness_i, loudness_range_low, loudness_range_high,
+ gain_staging_db,
+ to_db(final_makeup_gain),
+ correlation.get_correlation());
+}
+
map<DeviceSpec, DeviceInfo> AudioMixer::get_devices() const
{
lock_guard<timed_mutex> lock(audio_mutex);
// all together into one final audio signal.
//
// All operations on AudioMixer (except destruction) are thread-safe.
-//
-// TODO: There might be more audio stuff that should be moved here
-// from Mixer.
#include <math.h>
#include <stdint.h>
#include <mutex>
#include <set>
#include <vector>
+#include <zita-resampler/resampler.h>
#include "alsa_input.h"
#include "bmusb/bmusb.h"
+#include "correlation_measurer.h"
#include "db.h"
#include "defs.h"
+#include "ebu_r128_proc.h"
#include "filter.h"
#include "resampling_queue.h"
#include "stereocompressor.h"
AudioMixer(unsigned num_cards);
~AudioMixer();
void reset_resampler(DeviceSpec device_spec);
+ void reset_meters();
// Add audio (or silence) to the given device's queue. Can return false if
// the lock wasn't successfully taken; if so, you should simply try again.
std::vector<float> get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy);
- // See comments inside get_output().
- void set_current_loudness(double level_lufs) { loudness_lufs = level_lufs; }
-
void set_fader_volume(unsigned bus_index, float level_db) { fader_volume_db[bus_index] = level_db; }
std::map<DeviceSpec, DeviceInfo> get_devices() const;
void set_name(DeviceSpec device_spec, const std::string &name);
return final_makeup_gain_auto;
}
+ typedef std::function<void(float level_lufs, float peak_db,
+ float global_level_lufs, float range_low_lufs, float range_high_lufs,
+ float gain_staging_db, float final_makeup_gain_db,
+ float correlation)> audio_level_callback_t;
+ void set_audio_level_callback(audio_level_callback_t callback)
+ {
+ audio_level_callback = callback;
+ }
+
private:
struct AudioDevice {
std::unique_ptr<ResamplingQueue> resampling_queue;
void reset_resampler_mutex_held(DeviceSpec device_spec);
void reset_alsa_mutex_held(DeviceSpec device_spec);
std::map<DeviceSpec, DeviceInfo> get_devices_mutex_held() const;
+ void update_meters(const std::vector<float> &samples);
+ void send_audio_level_callback();
unsigned num_cards;
static constexpr float ref_level_dbfs = -14.0f; // Chosen so that we end up around 0 LU in practice.
static constexpr float ref_level_lufs = -23.0f; // 0 LU, more or less by definition.
- std::atomic<float> loudness_lufs{ref_level_lufs};
-
StereoCompressor limiter;
std::atomic<float> limiter_threshold_dbfs{ref_level_dbfs + 4.0f}; // 4 dB.
std::atomic<bool> limiter_enabled{true};
InputMapping input_mapping; // Under audio_mutex.
std::atomic<float> fader_volume_db[MAX_BUSES] {{ 0.0f }};
+
+ audio_level_callback_t audio_level_callback = nullptr;
+ mutable std::mutex audio_measure_mutex;
+ Ebu_r128_proc r128; // Under audio_measure_mutex.
+ CorrelationMeasurer correlation; // Under audio_measure_mutex.
+ Resampler peak_resampler; // Under audio_measure_mutex.
+ std::atomic<float> peak{0.0f};
};
#endif // !defined(_AUDIO_MIXER_H)
global_mixer->get_audio_mixer()->set_compressor_enabled(state == Qt::Checked);
});
connect(ui->reset_meters_button, &QPushButton::clicked, this, &MainWindow::reset_meters_button_clicked);
- mixer->set_audio_level_callback(bind(&MainWindow::audio_level_callback, this, _1, _2, _3, _4, _5, _6, _7, _8));
+ mixer->get_audio_mixer()->set_audio_level_callback(bind(&MainWindow::audio_level_callback, this, _1, _2, _3, _4, _5, _6, _7, _8));
struct sigaction act;
memset(&act, 0, sizeof(act));
void MainWindow::reset_meters_button_clicked()
{
- global_mixer->reset_meters();
+ global_mixer->get_audio_mixer()->reset_meters();
ui->peak_display->setText(QString::fromStdString(format_db(-HUGE_VAL, DB_WITH_SIGN | DB_BARE)));
ui->peak_display->setStyleSheet("");
}
num_cards(num_cards),
mixer_surface(create_surface(format)),
h264_encoder_surface(create_surface(format)),
- audio_mixer(num_cards),
- correlation(OUTPUT_FREQUENCY)
+ audio_mixer(num_cards)
{
CHECK(init_movit(MOVIT_SHADER_DIR, MOVIT_DEBUG_OFF));
check_error();
cbcr_position_attribute_index = glGetAttribLocation(cbcr_program_num, "position");
cbcr_texcoord_attribute_index = glGetAttribLocation(cbcr_program_num, "texcoord");
- 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);
-
if (global_flags.enable_alsa_output) {
alsa.reset(new ALSAOutput(OUTPUT_FREQUENCY, /*num_channels=*/2));
}
}
}
-float find_peak(const float *samples, size_t num_samples)
-{
- float m = fabs(samples[0]);
- for (size_t i = 1; i < num_samples; ++i) {
- m = max(m, fabs(samples[i]));
- }
- return m;
-}
-
-void deinterleave_samples(const vector<float> &in, vector<float> *out_l, vector<float> *out_r)
-{
- size_t num_samples = in.size() / 2;
- out_l->resize(num_samples);
- out_r->resize(num_samples);
-
- const float *inptr = in.data();
- float *lptr = &(*out_l)[0];
- float *rptr = &(*out_r)[0];
- for (size_t i = 0; i < num_samples; ++i) {
- *lptr++ = *inptr++;
- *rptr++ = *inptr++;
- }
-}
-
} // namespace
void Mixer::bm_frame(unsigned card_index, uint16_t timecode,
get_one_frame_from_each_card(master_card_index, new_frames, has_new_frame, num_samples);
schedule_audio_resampling_tasks(new_frames[master_card_index].dropped_frames, num_samples[master_card_index], new_frames[master_card_index].length);
stats_dropped_frames += new_frames[master_card_index].dropped_frames;
- send_audio_level_callback();
handle_hotplugged_cards();
}
}
-void Mixer::send_audio_level_callback()
-{
- if (audio_level_callback == nullptr) {
- return;
- }
-
- unique_lock<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();
-
- audio_level_callback(loudness_s, to_db(peak),
- loudness_i, loudness_range_low, loudness_range_high,
- audio_mixer.get_gain_staging_db(),
- audio_mixer.get_final_makeup_gain_db(),
- correlation.get_correlation());
-}
-
void Mixer::audio_thread_func()
{
while (!should_quit) {
ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy =
task.adjust_rate ? ResamplingQueue::ADJUST_RATE : ResamplingQueue::DO_NOT_ADJUST_RATE;
- process_audio_one_frame(task.pts_int, task.num_samples, rate_adjustment_policy);
- }
-}
-
-void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy)
-{
- vector<float> samples_out = audio_mixer.get_output(double(frame_pts_int) / TIMEBASE, num_samples, rate_adjustment_policy);
+ vector<float> samples_out = audio_mixer.get_output(double(task.pts_int) / TIMEBASE, task.num_samples, rate_adjustment_policy);
- // Upsample 4x to find interpolated peak.
- peak_resampler.inp_data = samples_out.data();
- peak_resampler.inp_count = samples_out.size() / 2;
-
- vector<float> interpolated_samples_out;
- interpolated_samples_out.resize(samples_out.size());
- {
- unique_lock<mutex> lock(audio_measure_mutex);
-
- while (peak_resampler.inp_count > 0) { // About four iterations.
- peak_resampler.out_data = &interpolated_samples_out[0];
- peak_resampler.out_count = interpolated_samples_out.size() / 2;
- peak_resampler.process();
- size_t out_stereo_samples = interpolated_samples_out.size() / 2 - peak_resampler.out_count;
- peak = max<float>(peak, find_peak(interpolated_samples_out.data(), out_stereo_samples * 2));
- peak_resampler.out_data = nullptr;
+ // Send the samples to the sound card, then add them to the output.
+ if (alsa) {
+ alsa->write(samples_out);
}
+ video_encoder->add_audio(task.pts_int, move(samples_out));
}
-
- // Find R128 levels and L/R correlation.
- vector<float> left, right;
- deinterleave_samples(samples_out, &left, &right);
- float *ptrs[] = { left.data(), right.data() };
- {
- unique_lock<mutex> lock(audio_measure_mutex);
- r128.process(left.size(), ptrs);
- audio_mixer.set_current_loudness(r128.loudness_M());
- correlation.process_samples(samples_out);
- }
-
- // Send the samples to the sound card.
- if (alsa) {
- alsa->write(samples_out);
- }
-
- // And finally add them to the output.
- video_encoder->add_audio(frame_pts_int, move(samples_out));
}
void Mixer::subsample_chroma(GLuint src_tex, GLuint dst_tex)
theme->channel_clicked(preview_num);
}
-void Mixer::reset_meters()
-{
- unique_lock<mutex> lock(audio_measure_mutex);
- peak_resampler.reset();
- peak = 0.0f;
- r128.reset();
- r128.integr_start();
- correlation.reset();
-}
-
void Mixer::start_mode_scanning(unsigned card_index)
{
assert(card_index < num_cards);
#include <movit/flat_input.h>
#include <stdbool.h>
#include <stdint.h>
-#include <zita-resampler/resampler.h>
#include <atomic>
#include <chrono>
#include "alsa_output.h"
#include "audio_mixer.h"
#include "bmusb/bmusb.h"
-#include "correlation_measurer.h"
#include "defs.h"
-#include "ebu_r128_proc.h"
#include "httpd.h"
#include "input_state.h"
#include "pbo_frame_allocator.h"
output_channel[output].set_color_updated_callback(callback);
}
- typedef std::function<void(float level_lufs, float peak_db,
- float global_level_lufs, float range_low_lufs, float range_high_lufs,
- float gain_staging_db, float final_makeup_gain_db,
- float correlation)> audio_level_callback_t;
- void set_audio_level_callback(audio_level_callback_t callback)
- {
- audio_level_callback = callback;
- }
-
std::vector<std::string> get_transition_names()
{
return theme->get_transition_names(pts());
should_cut = true;
}
- void reset_meters();
-
unsigned get_num_cards() const { return num_cards; }
std::string get_card_description(unsigned card_index) const {
void handle_hotplugged_cards();
void schedule_audio_resampling_tasks(unsigned dropped_frames, int num_samples_per_frame, int length_per_frame);
void render_one_frame(int64_t duration);
- void send_audio_level_callback();
void audio_thread_func();
- void process_audio_one_frame(int64_t frame_pts_int, int num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy);
void subsample_chroma(GLuint src_tex, GLuint dst_dst);
void release_display_frame(DisplayFrame *frame);
double pts() { return double(pts_int) / TIMEBASE; }
std::atomic<bool> should_quit{false};
std::atomic<bool> should_cut{false};
- audio_level_callback_t audio_level_callback = nullptr;
- mutable std::mutex audio_measure_mutex;
- Ebu_r128_proc r128; // Under audio_measure_mutex.
- CorrelationMeasurer correlation; // Under audio_measure_mutex.
- Resampler peak_resampler; // Under audio_measure_mutex.
- std::atomic<float> peak{0.0f};
-
std::unique_ptr<ALSAOutput> alsa;
struct AudioTask {
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