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| author | Michael Niedermayer <michaelni@gmx.at> | 2004-05-29 02:06:32 +0000 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2004-05-29 02:06:32 +0000 |
| commit | e928649b0bf6c8c7b87eb09d5e393a70387b10e9 (patch) | |
| tree | 3f065adcb0e5c14127d01162d7cc8a0d934c073f /libavformat/audio.c | |
| parent | a7b2871cd1401ce7be59b153eed3f25565b0bb23 (diff) | |
pass AVPacket into av_write_frame()
fixes the random dts/pts during encoding
asf preroll fix
no more initial zero frames for b frame encoding
mpeg-es dts during demuxing fixed
.ffm timestamp scale fixed, ffm is still broken though
Originally committed as revision 3168 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/audio.c')
| -rw-r--r-- | libavformat/audio.c | 5 |
1 files changed, 3 insertions, 2 deletions
diff --git a/libavformat/audio.c b/libavformat/audio.c index 00f705c372..e3e5888e9b 100644 --- a/libavformat/audio.c +++ b/libavformat/audio.c @@ -164,11 +164,12 @@ static int audio_write_header(AVFormatContext *s1) } } -static int audio_write_packet(AVFormatContext *s1, int stream_index, - const uint8_t *buf, int size, int64_t pts) +static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt) { AudioData *s = s1->priv_data; int len, ret; + int size= pkt->size; + uint8_t *buf= pkt->data; while (size > 0) { len = AUDIO_BLOCK_SIZE - s->buffer_ptr; |