num_cards(num_cards),
mixer_surface(create_surface(format)),
h264_encoder_surface(create_surface(format)),
+ correlation(OUTPUT_FREQUENCY),
level_compressor(OUTPUT_FREQUENCY),
limiter(OUTPUT_FREQUENCY),
compressor(OUTPUT_FREQUENCY)
decode_video_format(video_format, &width, &height, &second_field_start, &extra_lines_top, &extra_lines_bottom,
&frame_rate_nom, &frame_rate_den, &interlaced); // Ignore return value for now.
- int64_t frame_length = TIMEBASE * frame_rate_den / frame_rate_nom;
+ int64_t frame_length = int64_t(TIMEBASE * frame_rate_den) / frame_rate_nom;
size_t num_samples = (audio_frame.len >= audio_offset) ? (audio_frame.len - audio_offset) / 8 / 3 : 0;
if (num_samples > OUTPUT_FREQUENCY / 10) {
}
// Resample the audio as needed, including from previously dropped frames.
+ assert(num_cards > 0);
for (unsigned frame_num = 0; frame_num < card_copy[0].dropped_frames + 1; ++frame_num) {
{
// Signal to the audio thread to process this frame.
audio_level_callback(loudness_s, 20.0 * log10(peak),
loudness_i, loudness_range_low, loudness_range_high,
- gain_staging_db, 20.0 * log10(final_makeup_gain));
+ gain_staging_db, 20.0 * log10(final_makeup_gain),
+ correlation.get_correlation());
}
for (unsigned card_index = 1; card_index < num_cards; ++card_index) {
// 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);
+ 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
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
final_makeup_gain = m;
}
- // Find R128 levels.
+ // 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.
peak = 0.0f;
r128.reset();
r128.integr_start();
+ correlation.reset();
}
Mixer::OutputChannel::~OutputChannel()