X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=mixer.cpp;h=133fc28108a9b50e68e1e7ffa46848313a5e3d65;hb=67768135b61fe89047836d0443cefcb5cae4b485;hp=f3ff4c385c1d46b4f3b03dc5832a6ba816a8587a;hpb=8fe6a683cb5bc9f04555c8cb9257f33c4d356ded;p=nageru

diff --git a/mixer.cpp b/mixer.cpp
index f3ff4c3..133fc28 100644
--- 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()