hrev50639 adds 1 changeset to branch 'master'
old head: 905ee3fb8fde036bd8233479ca7b82df069bf62f
new head: 7af63063bb1d7f6c510144805d43bb89fdf61a82
overview:
http://cgit.haiku-os.org/haiku/log/?qt=range&q=7af63063bb1d+%5E905ee3fb8fde
----------------------------------------------------------------------------
7af63063bb1d: media/ffmpeg: skips libswr, and interleaves samples for planar
audio.
* workaround for #12720.
[ Jérôme Duval <jerome.duval@xxxxxxxxx> ]
----------------------------------------------------------------------------
Revision: hrev50639
Commit: 7af63063bb1d7f6c510144805d43bb89fdf61a82
URL: http://cgit.haiku-os.org/haiku/commit/?id=7af63063bb1d
Author: Jérôme Duval <jerome.duval@xxxxxxxxx>
Date: Mon Oct 31 08:07:53 2016 UTC
Ticket: https://dev.haiku-os.org/ticket/12720
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1 file changed, 16 insertions(+)
src/add-ons/media/plugins/ffmpeg/AVCodecDecoder.cpp | 16 ++++++++++++++++
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diff --git a/src/add-ons/media/plugins/ffmpeg/AVCodecDecoder.cpp
b/src/add-ons/media/plugins/ffmpeg/AVCodecDecoder.cpp
index 27ec813..2b219f9 100644
--- a/src/add-ons/media/plugins/ffmpeg/AVCodecDecoder.cpp
+++ b/src/add-ons/media/plugins/ffmpeg/AVCodecDecoder.cpp
@@ -928,6 +928,7 @@
AVCodecDecoder::_MoveAudioFramesToRawDecodedAudioAndUpdateStartTimes()
// "planar" audio (each channel separated instead of interleaved
samples).
// In that case, we use swresample to convert the data
if (av_sample_fmt_is_planar(fContext->sample_fmt)) {
+#if 0
const uint8_t* ptr[8];
for (int i = 0; i < 8; i++) {
if (fDecodedDataBuffer->data[i] == NULL)
@@ -950,6 +951,21 @@
AVCodecDecoder::_MoveAudioFramesToRawDecodedAudioAndUpdateStartTimes()
if (frames < 0)
debugger("resampling failed");
+#else
+ // interleave planar audio with same format
+ uintptr_t out = (uintptr_t)fRawDecodedAudio->data[0];
+ int32 offset = fDecodedDataBufferOffset;
+ for (int i = 0; i < frames; i++) {
+ for (int j = 0; j < fContext->channels; j++) {
+ memcpy((void*)out, fDecodedDataBuffer->data[j]
+ + offset, fInputFrameSize);
+ out += fInputFrameSize;
+ }
+ offset += fInputFrameSize;
+ }
+ outFrames = frames;
+ inFrames = frames;
+#endif
} else {
memcpy(fRawDecodedAudio->data[0], fDecodedDataBuffer->data[0]
+ fDecodedDataBufferOffset, frames *
fOutputFrameSize);
