+void Mixer::audio_thread_func()
+{
+ while (!should_quit) {
+ AudioTask task;
+
+ {
+ unique_lock<mutex> lock(audio_mutex);
+ audio_task_queue_changed.wait(lock, [this]{ return !audio_task_queue.empty(); });
+ task = audio_task_queue.front();
+ audio_task_queue.pop();
+ }
+
+ process_audio_one_frame(task.pts_int, task.num_samples);
+ }
+}
+
+void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples)
+{
+ vector<float> samples_card;
+ vector<float> samples_out;
+ for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
+ samples_card.resize(num_samples * 2);
+ {
+ unique_lock<mutex> lock(cards[card_index].audio_mutex);
+ if (!cards[card_index].resampling_queue->get_output_samples(double(frame_pts_int) / TIMEBASE, &samples_card[0], num_samples)) {
+ printf("Card %d reported previous underrun.\n", card_index);
+ }
+ }
+ // TODO: Allow using audio from the other card(s) as well.
+ if (card_index == 0) {
+ samples_out = move(samples_card);
+ }
+ }
+
+ // 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);
+ }
+
+ // 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.
+ {
+ unique_lock<mutex> lock(compressor_mutex);
+ if (level_compressor_enabled) {
+ float threshold = 0.01f; // -40 dBFS.
+ float ratio = 20.0f;
+ float attack_time = 0.5f;
+ float release_time = 20.0f;
+ float makeup_gain = pow(10.0f, (ref_level_dbfs - (-40.0f)) / 20.0f); // +26 dB.
+ level_compressor.process(samples_out.data(), samples_out.size() / 2, threshold, ratio, attack_time, release_time, makeup_gain);
+ gain_staging_db = 20.0 * log10(level_compressor.get_attenuation() * makeup_gain);
+ } else {
+ // Just apply the gain we already had.
+ float g = pow(10.0f, gain_staging_db / 20.0f);
+ for (size_t i = 0; i < samples_out.size(); ++i) {
+ samples_out[i] *= g;
+ }
+ }
+ }
+
+#if 0
+ printf("level=%f (%+5.2f dBFS) attenuation=%f (%+5.2f dB) end_result=%+5.2f dB\n",
+ level_compressor.get_level(), 20.0 * log10(level_compressor.get_level()),
+ level_compressor.get_attenuation(), 20.0 * log10(level_compressor.get_attenuation()),
+ 20.0 * log10(level_compressor.get_level() * level_compressor.get_attenuation() * makeup_gain));
+#endif
+
+// float limiter_att, compressor_att;
+
+ // 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;
+ 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();
+ }
+
+ // 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));
+
+ // 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));
+ peak_resampler.out_data = nullptr;
+ }
+
+ // At this point, we are most likely close to +0 LU, 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 100 seconds).
+ double target_loudness_factor, alpha;
+ {
+ unique_lock<mutex> lock(compressor_mutex);
+ double loudness_lu = r128.loudness_M() - ref_level_lufs;
+ double current_makeup_lu = 20.0f * log10(final_makeup_gain);
+ target_loudness_factor = pow(10.0f, -loudness_lu / 20.0f);
+
+ // If we're outside +/- 5 LU uncorrected, 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) {
+ 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 = 100.0;
+ 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);
+ }
+
+ 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;
+ }
+
+ // 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(compressor_mutex);
+ r128.process(left.size(), ptrs);
+ 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.
+ h264_encoder->add_audio(frame_pts_int, move(samples_out));
+}
+