Release Nageru 1.0.0, with some documentation updates.

[nageru] / mixer.cpp

diff --git a/mixer.cpp b/mixer.cpp
index f3ff4c385c1d46b4f3b03dc5832a6ba816a8587a..133fc28108a9b50e68e1e7ffa46848313a5e3d65 100644 (file)
--- a/mixer.cpp
+++ b/mixer.cpp
@@ -98,6 +98,7 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards)
          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)
@@ -184,7 +185,7 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards)
 
        // 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);
+       peak_resampler.setup(OUTPUT_FREQUENCY, OUTPUT_FREQUENCY * 4, /*num_channels=*/2, /*hlen=*/16, /*frel=*/1.0);
 
        alsa.reset(new ALSAOutput(OUTPUT_FREQUENCY, /*num_channels=*/2));
 }
@@ -255,7 +256,7 @@ void Mixer::bm_frame(unsigned card_index, uint16_t timecode,
 
        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) {
@@ -509,6 +510,7 @@ void Mixer::thread_func()
                }
 
                // 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.
@@ -534,7 +536,8 @@ void Mixer::thread_func()
 
                        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) {
@@ -689,6 +692,11 @@ void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples)
 {
        vector<float> samples_card;
        vector<float> samples_out;
+
+       // TODO: Allow mixing audio from several sources.
+       unsigned selected_audio_card = theme->map_signal(audio_source_channel);
+       assert(selected_audio_card < num_cards);
+
        for (unsigned card_index = 0; card_index < num_cards; ++card_index) {
                samples_card.resize(num_samples * 2);
                {
@@ -697,8 +705,7 @@ void Mixer::process_audio_one_frame(int64_t frame_pts_int, int 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) {
+               if (card_index == selected_audio_card) {
                        samples_out = move(samples_card);
                }
        }
@@ -707,7 +714,9 @@ void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples)
        // 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
@@ -780,6 +789,7 @@ void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples)
                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
@@ -822,13 +832,14 @@ void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples)
                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.
@@ -933,6 +944,7 @@ void Mixer::reset_meters()
        peak = 0.0f;
        r128.reset();
        r128.integr_start();
+       correlation.reset();
 }
 
 Mixer::OutputChannel::~OutputChannel()