Mercurial > mplayer.hg
view libmpdemux/demux_rtp.h @ 24900:9079c9745ff9
A/V sync: take audio filter buffers into account
Substract the delay caused by filter buffering when calculating
currently playing audio position. This matters for af_scaletempo which
buffers significant and varying amounts of data. For other current
filters the effect is normally insignificant.
Instead of the old time-based filter delay field (which was ignored)
this version stores the per-filter delay in units of bytes input read
without corresponding output. This allows the current scaletempo
behavior where other filters before and after it can see the same
nominal samplerate even though the real duration of the data varies;
in this case the other filters can not know the delay they're causing
in terms of real time.
author | uau |
---|---|
date | Thu, 01 Nov 2007 06:52:50 +0000 |
parents | 3f0d00abc073 |
children | 3baf6a2283da |
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#ifndef DEMUX_RTP_H #define DEMUX_RTP_H #include <stdlib.h> #include <stdio.h> #ifndef STREAM_H #include "stream/stream.h" #endif #ifndef DEMUXER_H #include "demuxer.h" #endif // Open a RTP demuxer (which was initiated either from a SDP file, // or from a RTSP URL): demuxer_t* demux_open_rtp(demuxer_t* demuxer); // Test whether a RTP demuxer is for a MPEG stream: int demux_is_mpeg_rtp_stream(demuxer_t* demuxer); // Test whether a RTP demuxer contains combined (multiplexed) // audio+video (and so needs to be demuxed by higher-level code): int demux_is_multiplexed_rtp_stream(demuxer_t* demuxer); // Read from a RTP demuxer: int demux_rtp_fill_buffer(demuxer_t *demux, demux_stream_t* ds); // Close a RTP demuxer void demux_close_rtp(demuxer_t* demuxer); #endif