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author | Michael Niedermayer <michaelni@gmx.at> | 2004-05-29 02:06:32 +0000 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2004-05-29 02:06:32 +0000 |
commit | e928649b0bf6c8c7b87eb09d5e393a70387b10e9 (patch) | |
tree | 3f065adcb0e5c14127d01162d7cc8a0d934c073f /libavformat/rtp.c | |
parent | a7b2871cd1401ce7be59b153eed3f25565b0bb23 (diff) | |
download | ffmpeg-e928649b0bf6c8c7b87eb09d5e393a70387b10e9.tar.gz |
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/rtp.c')
-rw-r--r-- | libavformat/rtp.c | 9 |
1 files changed, 5 insertions, 4 deletions
diff --git a/libavformat/rtp.c b/libavformat/rtp.c index 5d4884d8f7..31e4a21dc1 100644 --- a/libavformat/rtp.c +++ b/libavformat/rtp.c @@ -669,16 +669,17 @@ static void rtp_send_mpegts_raw(AVFormatContext *s1, } /* write an RTP packet. 'buf1' must contain a single specific frame. */ -static int rtp_write_packet(AVFormatContext *s1, int stream_index, - const uint8_t *buf1, int size, int64_t pts) +static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) { RTPDemuxContext *s = s1->priv_data; AVStream *st = s1->streams[0]; int rtcp_bytes; int64_t ntp_time; + int size= pkt->size; + uint8_t *buf1= pkt->data; #ifdef DEBUG - printf("%d: write len=%d\n", stream_index, size); + printf("%d: write len=%d\n", pkt->stream_index, size); #endif /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ @@ -687,7 +688,7 @@ static int rtp_write_packet(AVFormatContext *s1, int stream_index, if (s->first_packet || rtcp_bytes >= 28) { /* compute NTP time */ /* XXX: 90 kHz timestamp hardcoded */ - ntp_time = (pts << 28) / 5625; + ntp_time = (pkt->pts << 28) / 5625; rtcp_send_sr(s1, ntp_time); s->last_octet_count = s->octet_count; s->first_packet = 0; |