In my application , I used ffmpeg to publish my camera and microphone 's av
stream as rtmp stream to nginx server.
I use following code to publish stream:
void RtmpLiveEncoder::Run()
{
AVBitStreamFilterContext* aacbsfc = av_bitstream_filter_init("aac_adtstoasc");
start_time = av_gettime();
while(1)
{
do
{
int ret = 0;
AVPacket pkt;
av_init_packet();
ret = av_read_frame(ifmt_ctx,);
if(ret<0 )
{
printf("read video frame failed\n");
break;
}
if(pkt.pts==AV_NOPTS_VALUE)
{
if(_frameduration==0)
{
pkt.dts = pkt.pts=(av_gettime()-start_time)/1000;
}
else
{
pkt.dts = pkt.pts = _lastvideopts;
pkt.duration = _frameduration;
pkt.pos = -1;
_lastvideopts += _frameduration;
}
}
if(av_write_frame(ofmt_ctx,)<0)
{
printf("write video frame failed\n");
}
av_packet_unref();
}while(0);
do
{
if(!_hasaudio)
{
break;
}
if((_lastaudiopts-_lastvideopts)>0)
{
printf("the audio is faster than video, the audio pts is %d, the
video pts is %d\n",_lastaudiopts,_lastvideopts);
break;
}
int ret = 0;
AVPacket audiopacket;
av_init_packet();
ret = av_read_frame(aifmt_ctx,);
if(ret<0)
{
break;
}
AVStream* out_stream = ofmt_ctx->streams[1];
if(av_bitstream_filter_filter(aacbsfc, out_stream->codec, NULL,
, , audiopacket.data, audiopacket.size, 0)<0)
{
printf("remove adts header failed\n");
}
if(av_bitstream_filter_filter(aacbsfc, out_stream->codec, NULL,
>data, >size, audiopacket.buf->data,
audiopacket.buf->size, 0)<0)
{
printf("remove adts header failed\n");
}
audiopacket.stream_index=1;
audiopacket.dts = audiopacket.pts=_lastaudiopts;
audiopacket.duration =
(double)1024/out_stream->codecpar->sample_rate*1000;
_lastaudiopts += audiopacket.duration;
audiopacket.pos = -1;
if(av_write_frame(ofmt_ctx,)<0)
{
printf("write audio failed.\n");
}
av_packet_unref();
}while(0);
}
I used vlc to access the hls stream, but after some minutes, the audio
isn't sync any more.
How could I do?
THANKS IN ADVANCE!___
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