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Add LEQ(m) when analysing audio (#1382).
[dcpomatic.git]
/
src
/
lib
/
ffmpeg_decoder.cc
diff --git
a/src/lib/ffmpeg_decoder.cc
b/src/lib/ffmpeg_decoder.cc
index 350362478e92dfb7522e9f8ffdd7d666ddd4d656..cfaf0361b9ab79da14f4f4ae841a5bd8a243cf9d 100644
(file)
--- a/
src/lib/ffmpeg_decoder.cc
+++ b/
src/lib/ffmpeg_decoder.cc
@@
-210,6
+210,9
@@
FFmpegDecoder::deinterleave_audio (shared_ptr<FFmpegAudioStream> stream) const
0, stream->stream(_format_context)->codec->channels, _frame->nb_samples, audio_sample_format (stream), 1
);
0, stream->stream(_format_context)->codec->channels, _frame->nb_samples, audio_sample_format (stream), 1
);
+ /* XXX: can't we just use _frame->nb_samples directly here? */
+ /* XXX: can't we use swr_convert() to do the format conversion? */
+
/* Deinterleave and convert to float */
/* total_samples and frames will be rounded down here, so if there are stray samples at the end
/* Deinterleave and convert to float */
/* total_samples and frames will be rounded down here, so if there are stray samples at the end