want help in setting a delay in the stream.
eslam.ahmed at avidbeam.com
Wed Nov 24 07:56:20 UTC 2021
What about using the latency property of rtspsrc?
On Tue, Nov 23, 2021 at 2:30 PM 136_B4_USAMA _TAHSEEN via gstreamer-devel <
gstreamer-devel at lists.freedesktop.org> wrote:
> Hi ,
> I want to delay a rtp stream by some precise amount of time
> I am using a browser client which send h264 encoded (webrtc) rtp stream to
> server then i pass this rtp stream to a gstreamer pipeline and add the
> amount of dealy i want in the stream PTS and DTS and the receive back the
> rtp stream in server and the sends back to the browser-client.
> GST_BUFFER_PTS (buffer)+=5000000000;(nano sec)
> GST_BUFFER_DTS (buffer)+=5000000000;
> browser-sender-client --->media-server---> Gstreamer-C-pipeline ---> media-server ---> browser-receiver-client
> i want the stream to be played at browser-receiver-client by some precise
> amount of delay from the time it was created at sender side.
> but the problem is i am getting some extra delay then 5s
> udp sink blocking the pipeline until the timestamp that i have changed
> extra delay is due to the amount of time stream takes to reach back
> to the browser client.
> and when I set udpsink to sync =false the browser plays the stream without
> any delay even though I have changed pts dts in the pipeline at the server
> on localhost amount of extra delay varies between 5.55 s to 5.158 s
> but when i hosted to aws the amount if delay increased upto 6 to 7 s
> Thank you
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