Fix an endianness issue.

[nageru] / mixer.cpp

diff --git a/mixer.cpp b/mixer.cpp
index fd33abbb18f35d3569b89f4d673c37e9e7a92365..05b7c51dcb78110f5bb94dcf910dbd3feeecef81 100644 (file)
--- a/mixer.cpp
+++ b/mixer.cpp
@@ -3,6 +3,7 @@
 #include "mixer.h"
 
 #include <assert.h>
+#include <endian.h>
 #include <epoxy/egl.h>
 #include <movit/effect_chain.h>
 #include <movit/effect_util.h>
@@ -62,7 +63,7 @@ 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);
        }
@@ -73,9 +74,8 @@ void convert_fixed32_to_fp32(float *dst, size_t out_channels, const uint8_t *src
        assert(in_channels >= out_channels);
        for (size_t i = 0; i < num_samples; ++i) {
                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);
+                       int32_t s = le32toh(*(int32_t *)src);
+                       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);