Fix merge.

[dcpomatic.git] / src / lib / ffmpeg_decoder.cc

diff --git a/src/lib/ffmpeg_decoder.cc b/src/lib/ffmpeg_decoder.cc
index c4ca00fa6e3ef930c16613e3ad0d914ff01f6afa..60bb3271e19e902fc125712fa1f3b5c1cafc5782 100644 (file)
--- a/src/lib/ffmpeg_decoder.cc
+++ b/src/lib/ffmpeg_decoder.cc
@@ -369,7 +369,7 @@ FFmpegDecoder::deinterleave_audio (uint8_t* data, int size)
        switch (audio_sample_format()) {
        case AV_SAMPLE_FMT_S16:
        {
-               int16_t* p = (int16_t *) data;
+               int16_t* p = reinterpret_cast<int16_t *> (data);
                int sample = 0;
                int channel = 0;
                for (int i = 0; i < total_samples; ++i) {
@@ -384,13 +384,24 @@ FFmpegDecoder::deinterleave_audio (uint8_t* data, int size)
        }
        break;
 
+       case AV_SAMPLE_FMT_S16P:
+       {
+               int16_t* p = reinterpret_cast<int16_t *> (data);
+               for (int i = 0; i < _film->audio_channels(); ++i) {
+                       for (int j = 0; j < frames; ++j) {
+                               audio->data(i)[j] = static_cast<float>(*p++) / (1 << 15);
+                       }
+               }
+       }
+       break;
+       
        case AV_SAMPLE_FMT_S32:
        {
-               int32_t* p = (int32_t *) data;
+               int32_t* p = reinterpret_cast<int32_t *> (data);
                int sample = 0;
                int channel = 0;
                for (int i = 0; i < total_samples; ++i) {
-                       audio->data(channel)[sample] = float(*p++) / (1 << 31);
+                       audio->data(channel)[sample] = static_cast<float>(*p++) / (1 << 31);
 
                        ++channel;
                        if (channel == _film->audio_channels()) {
@@ -399,7 +410,25 @@ FFmpegDecoder::deinterleave_audio (uint8_t* data, int size)
                        }
                }
        }
+       break;
 
+       case AV_SAMPLE_FMT_FLT:
+       {
+               float* p = reinterpret_cast<float*> (data);
+               int sample = 0;
+               int channel = 0;
+               for (int i = 0; i < total_samples; ++i) {
+                       audio->data(channel)[sample] = *p++;
+
+                       ++channel;
+                       if (channel == _film->audio_channels()) {
+                               channel = 0;
+                               ++sample;
+                       }
+               }
+       }
+       break;
+               
        case AV_SAMPLE_FMT_FLTP:
        {
                float* p = reinterpret_cast<float*> (data);