Transmit Audio + Video - Redistributing Latency

[Nicolas Dufresne]

Le lun. 2 juil. 2018 09:27, mudassar <mudassar87 at hotmail.com> a écrit :

> Hi,
>
> I am streaming video and audio over rtmp to Wowza. Both video and audio are
> captured using appsrc in my application with the following pipeline:
>
> "appsrc name=videosrc format=3 is-live=true do-timestamp=true !
> video/x-raw,
> width=%ld, height=%ld, framerate=20/1, format=ARGB ! queue
> max-size-buffers=1 ! videoconvert  ! vtenc_h264 realtime=true
> max-keyframe-interval=60 bitrate=1000 ! h264parse ! flvmux name=mux
> streamable=true  ! queue ! rtmpsink name=sink sync=true location=%s  appsrc
> name=audiosrc format=3  blocksize=8192 do-timestamp=true is-live=true    !
> queue ! audio/x-raw, channels=1, rate=44100, format=F64LE, width=64,
> depth=64 ! audioconvert ! audiorate ! audioresample ! audio/x-raw,
> channels=1, rate=44100, format=F32LE ! queue ! avenc_aac  ! aacparse !
> mux."
>
> The problem is that audio is always slightly earlier than audio in the live
> stream. I have tried adding more queue in audio pipeline, I have tried to
> set minimum latency=100000000 at videosrc as well as audiosrc with
> different
> combinations ,nothing worked.
>
> What should I do ?
>

It's the timestamp that will disctate synchronization of the audio and
video. If you use do-timestamp, you then need to sync audio/video before
pushing into appsrc.



>
>
>
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