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
                );
 
+       /* 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