X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=mixer.cpp;h=96668c840b2ca855e76e3a86f7e6b5956d0956ba;hb=20706e7944f3edd371504be43ebe76ce24778698;hp=fd33abbb18f35d3569b89f4d673c37e9e7a92365;hpb=6f4f5d4a13df2e4562cb9c145fa5b60be2ae63d0;p=nageru

diff --git a/mixer.cpp b/mixer.cpp
index fd33abb..96668c8 100644
--- a/mixer.cpp
+++ b/mixer.cpp
@@ -62,7 +62,7 @@ void convert_fixed24_to_fp32(float *dst, size_t out_channels, const uint8_t *src
 			uint32_t s2 = *src++;
 			uint32_t s3 = *src++;
 			uint32_t s = s1 | (s1 << 8) | (s2 << 16) | (s3 << 24);
-			dst[i * out_channels + j] = int(s) * (1.0f / 4294967296.0f);
+			dst[i * out_channels + j] = int(s) * (1.0f / 2147483648.0f);
 		}
 		src += 3 * (in_channels - out_channels);
 	}
@@ -75,7 +75,7 @@ void convert_fixed32_to_fp32(float *dst, size_t out_channels, const uint8_t *src
 		for (size_t j = 0; j < out_channels; ++j) {
 			// Note: Assumes little-endian.
 			int32_t s = *(int32_t *)src;
-			dst[i * out_channels + j] = s * (1.0f / 4294967296.0f);
+			dst[i * out_channels + j] = s * (1.0f / 2147483648.0f);
 			src += 4;
 		}
 		src += 4 * (in_channels - out_channels);
@@ -201,7 +201,8 @@ Mixer::Mixer(const QSurfaceFormat &format, unsigned num_cards)
 
 	unsigned num_fake_cards = 0;
 	for ( ; card_index < num_cards; ++card_index, ++num_fake_cards) {
-		configure_card(card_index, new FakeCapture(WIDTH, HEIGHT, FAKE_FPS, OUTPUT_FREQUENCY, card_index), /*is_fake_capture=*/true);
+		FakeCapture *capture = new FakeCapture(WIDTH, HEIGHT, FAKE_FPS, OUTPUT_FREQUENCY, card_index, global_flags.fake_cards_audio);
+		configure_card(card_index, capture, /*is_fake_capture=*/true);
 	}
 
 	if (num_fake_cards > 0) {
@@ -817,7 +818,8 @@ void Mixer::handle_hotplugged_cards()
 		CaptureCard *card = &cards[card_index];
 		if (card->capture->get_disconnected()) {
 			fprintf(stderr, "Card %u went away, replacing with a fake card.\n", card_index);
-			configure_card(card_index, new FakeCapture(WIDTH, HEIGHT, FAKE_FPS, OUTPUT_FREQUENCY, card_index), /*is_fake_capture=*/true);
+			FakeCapture *capture = new FakeCapture(WIDTH, HEIGHT, FAKE_FPS, OUTPUT_FREQUENCY, card_index, global_flags.fake_cards_audio);
+			configure_card(card_index, capture, /*is_fake_capture=*/true);
 			card->queue_length_policy.reset(card_index);
 			card->capture->start_bm_capture();
 		}
@@ -1058,21 +1060,6 @@ void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples)
 
 //	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
@@ -1113,6 +1100,21 @@ void Mixer::process_audio_one_frame(int64_t frame_pts_int, int num_samples)
 		final_makeup_gain = m;
 	}
 
+	// 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;
+	}
+
 	// Find R128 levels and L/R correlation.
 	vector<float> left, right;
 	deinterleave_samples(samples_out, &left, &right);