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| author | Michael Niedermayer <michaelni@gmx.at> | 2012-11-02 14:15:28 +0100 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2012-11-02 14:20:33 +0100 |
| commit | 6788350281c418f0f395a8279eee82f7abe7c63b (patch) | |
| tree | 69cd76f699eff929f5b13f76b42eabc7f25f9355 /libavcodec/flac_parser.c | |
| parent | 00aa7fa786e41b5fc8404732453869aa3c14e33a (diff) | |
| parent | 50a65e7a540ce6747f81d6dbf6a602ad35be77ff (diff) | |
| download | ffmpeg-6788350281c418f0f395a8279eee82f7abe7c63b.tar.gz | |
Merge commit '50a65e7a540ce6747f81d6dbf6a602ad35be77ff'
* commit '50a65e7a540ce6747f81d6dbf6a602ad35be77ff': (24 commits)
vmdaudio: set channel layout
twinvq: validate sample rate code
twinvq: set channel layout
twinvq: validate that channels is not <= 0
truespeech: set channel layout
sipr: set channel layout
shorten: validate that the channel count in the header is not <= 0
ra288dec: set channel layout
ra144dec: set channel layout
qdm2: remove unneeded checks for channel count
qdm2: make sure channels is not <= 0 and set channel layout
qcelpdec: set channel layout
nellymoserdec: set channels to 1
libopencore-amr: set channel layout for amr-nb or if not set by the user
libilbc: set channel layout
dpcm: use AVCodecContext.channels instead of keeping a private copy
imc: set channels to 1 instead of validating it
gsmdec: always set channel layout and sample rate at initialization
libgsmdec: always set channel layout and sample rate at initialization
g726dec: do not validate sample rate
...
Conflicts:
libavcodec/dpcm.c
libavcodec/qdm2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/flac_parser.c')
| -rw-r--r-- | libavcodec/flac_parser.c | 1 |
1 files changed, 1 insertions, 0 deletions
diff --git a/libavcodec/flac_parser.c b/libavcodec/flac_parser.c index f38d7aae67..7d3c5c4973 100644 --- a/libavcodec/flac_parser.c +++ b/libavcodec/flac_parser.c @@ -459,6 +459,7 @@ static int get_best_header(FLACParseContext* fpc, const uint8_t **poutbuf, fpc->avctx->sample_rate = header->fi.samplerate; fpc->avctx->channels = header->fi.channels; + ff_flac_set_channel_layout(fpc->avctx); fpc->pc->duration = header->fi.blocksize; *poutbuf = flac_fifo_read_wrap(fpc, header->offset, *poutbuf_size, &fpc->wrap_buf, |