-#define WIDTH 1280
-#define HEIGHT 720
#define EXTRAHEIGHT 30
#undef Success
} // namespace
Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards)
- : httpd("test.ts", WIDTH, HEIGHT),
+ : httpd(LOCAL_DUMP_FILE_NAME, WIDTH, HEIGHT),
num_cards(num_cards),
mixer_surface(create_surface(format)),
h264_encoder_surface(create_surface(format)),
r128.integr_start();
locut.init(FILTER_HPF, 2);
+
+ // 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);
+
+ alsa.reset(new ALSAOutput(OUTPUT_FREQUENCY, /*num_channels=*/2));
}
Mixer::~Mixer()
}
cards[card_index].usb->stop_dequeue_thread();
}
+
+ h264_encoder.reset(nullptr);
}
namespace {
}
}
-float find_peak(const vector<float> &samples)
+float find_peak(const float *samples, size_t num_samples)
{
float m = fabs(samples[0]);
- for (size_t i = 1; i < samples.size(); ++i) {
+ for (size_t i = 1; i < num_samples; ++i) {
m = std::max(m, fabs(samples[i]));
}
return m;
// Resample the audio as needed, including from previously dropped frames.
for (unsigned frame_num = 0; frame_num < card_copy[0].dropped_frames + 1; ++frame_num) {
- process_audio_one_frame();
+ {
+ // Signal to the audio thread to process this frame.
+ unique_lock<mutex> lock(audio_mutex);
+ audio_pts_queue.push(pts_int);
+ audio_pts_queue_changed.notify_one();
+ }
if (frame_num != card_copy[0].dropped_frames) {
// For dropped frames, increase the pts.
++dropped_frames;
resource_pool->clean_context();
}
-void Mixer::process_audio_one_frame()
+void Mixer::audio_thread_func()
+{
+ while (!should_quit) {
+ int64_t frame_pts_int;
+
+ {
+ unique_lock<mutex> lock(audio_mutex);
+ audio_pts_queue_changed.wait(lock, [this]{ return !audio_pts_queue.empty(); });
+ frame_pts_int = audio_pts_queue.front();
+ audio_pts_queue.pop();
+ }
+
+ process_audio_one_frame(frame_pts_int);
+ }
+}
+
+void Mixer::process_audio_one_frame(int64_t frame_pts_int)
{
vector<float> samples_card;
vector<float> samples_out;
samples_card.resize((OUTPUT_FREQUENCY / FPS) * 2);
{
unique_lock<mutex> lock(cards[card_index].audio_mutex);
- if (!cards[card_index].resampling_queue->get_output_samples(pts(), &samples_card[0], OUTPUT_FREQUENCY / FPS)) {
+ if (!cards[card_index].resampling_queue->get_output_samples(double(frame_pts_int) / TIMEBASE, &samples_card[0], OUTPUT_FREQUENCY / FPS)) {
printf("Card %d reported previous underrun.\n", card_index);
}
}
}
}
- // Cut away everything under 150 Hz; 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.)
+ // 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.)
locut.render(samples_out.data(), samples_out.size() / 2, locut_cutoff_hz * 2.0 * M_PI / OUTPUT_FREQUENCY, 0.5f);
// Apply a level compressor to get the general level right.
// float limiter_att, compressor_att;
- // Then a limiter at +0 dB (so, -14 dBFS) to take out the worst peaks only.
- // Note that since ratio is not infinite, we could go slightly higher than this.
- // Probably more tuning is warranted here.
- {
- float threshold = pow(10.0f, (ref_level_dbfs + 0.0f) / 20.0f); // +0 dB.
- float ratio = 30.0f;
- float attack_time = 0.0f; // Instant.
- float release_time = 0.005f;
- 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();
- }
-
- // Finally, the real compressor.
- {
- float threshold = pow(10.0f, (ref_level_dbfs - 12.0f) / 20.0f); // -12 dB.
+ // The real compressor.
+ if (compressor_enabled) {
+ float threshold = pow(10.0f, compressor_threshold_dbfs / 20.0f);
float ratio = 20.0f;
float attack_time = 0.005f;
float release_time = 0.040f;
// compressor_att = compressor.get_attenuation();
}
+ // 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 = pow(10.0f, limiter_threshold_dbfs / 20.0f);
+ 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", 20.0*log10(limiter_att), 20.0*log10(compressor_att));
- // Find peak and R128 levels.
- peak = max<float>(peak, find_peak(samples_out));
+ // 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());
+ 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));
+ }
+
+ // Find R128 levels.
vector<float> left, right;
deinterleave_samples(samples_out, &left, &right);
float *ptrs[] = { left.data(), right.data() };
r128.process(left.size(), ptrs);
- // Actually add the samples to the output.
- h264_encoder->add_audio(pts_int, move(samples_out));
+ // Send the samples to the sound card.
+ if (alsa) {
+ alsa->write(samples_out);
+ }
+
+ // And finally add them to the output.
+ h264_encoder->add_audio(frame_pts_int, move(samples_out));
}
void Mixer::subsample_chroma(GLuint src_tex, GLuint dst_tex)
void Mixer::start()
{
mixer_thread = thread(&Mixer::thread_func, this);
+ audio_thread = thread(&Mixer::audio_thread_func, this);
}
void Mixer::quit()
{
should_quit = true;
mixer_thread.join();
+ audio_thread.join();
}
void Mixer::transition_clicked(int transition_num)
void Mixer::reset_meters()
{
+ peak_resampler.reset();
peak = 0.0f;
r128.reset();
r128.integr_start();
projects / nageru / blobdiff