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| author | Michael Niedermayer <michaelni@gmx.at> | 2012-03-04 02:03:25 +0100 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2012-03-04 04:26:04 +0100 |
| commit | 15c6be8c7da7b94b5e131396e9a82ab6fe4a6b64 (patch) | |
| tree | 84db7f4851faba26561f846b4f112ef64d01b3ad /libavformat/utils.c | |
| parent | f972193a15026a99eb2b08e7913a03f2123663da (diff) | |
| parent | b7beabab4b78cc253d06c0a33f15b8ff79866e85 (diff) | |
| download | ffmpeg-15c6be8c7da7b94b5e131396e9a82ab6fe4a6b64.tar.gz | |
Merge remote-tracking branch 'qatar/master'
* qatar/master:
tiertexseq: set correct block_align for audio
tiertexseq: set audio stream start time to 0
voc/avs: Do not change the sample rate mid-stream.
segafilm: use the sample rate as the time base for audio streams
ea: fix audio pts
psx-str: fix audio pts
vqf: set packet duration
tta demuxer: set packet duration
mpegaudio_parser: do not ignore information from the first parsed frame
mpegaudio_parser: be less picky about the start position
thp: set audio packet durations
avcodec: add a Vorbis parser to get packet duration
vorbisdec: read the previous window flag for long windows
lavc: free the output packet when encoding failed or produced no output.
lavc: preserve avpkt->destruct in ff_alloc_packet().
lavc: clarify the meaning of AVCodecContext.frame_number.
mpegts: Pad the packet buffer in handle_packet().
mpegts: Do not call read_sl_header() when no bytes remain in the buffer.
Conflicts:
libavcodec/mpegaudio_parser.c
libavcodec/version.h
libavformat/mpegts.c
tests/ref/fate/pva-demux
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavformat/utils.c')
| -rw-r--r-- | libavformat/utils.c | 6 |
1 files changed, 1 insertions, 5 deletions
diff --git a/libavformat/utils.c b/libavformat/utils.c index 580b9fea3e..7f6bfaa88a 100644 --- a/libavformat/utils.c +++ b/libavformat/utils.c @@ -759,9 +759,6 @@ static int get_audio_frame_size(AVCodecContext *enc, int size) { int frame_size; - if(enc->codec_id == CODEC_ID_VORBIS) - return -1; - if (enc->frame_size <= 1) { int bits_per_sample = av_get_bits_per_sample(enc->codec_id); @@ -2105,8 +2102,7 @@ static int has_codec_parameters(AVCodecContext *avctx) case AVMEDIA_TYPE_AUDIO: val = avctx->sample_rate && avctx->channels && avctx->sample_fmt != AV_SAMPLE_FMT_NONE; if (!avctx->frame_size && - (avctx->codec_id == CODEC_ID_VORBIS || - avctx->codec_id == CODEC_ID_AAC || + (avctx->codec_id == CODEC_ID_AAC || avctx->codec_id == CODEC_ID_MP1 || avctx->codec_id == CODEC_ID_MP2 || avctx->codec_id == CODEC_ID_MP3 || |