[gst-devel] Ideal pipe for streaming live with minimal delay

[Arnout Vandecappelle]

 You should first find out where the latency is.  Is it in the encoder, in 
the network, in the rtp jitter buffer, or in the transcoder?

 Regards,
 Arnout

On Thursday 05 August 2010 16:30:18, 0rr3 wrote:
> Hi all!
> We're trying to receive, process and transmit a live network video stream
> from an ip camera.
> The idea is to receive the h264 video from the camera with rtspsrc, then
> decode and encode it into mjpg and finally pass it forward to a web
> browser.
> 
> This all works great except that a increasing delay of the live stream
> ruins everything. Therefore we have experimented with different
> elements, leaks and caps to minimize the delay and now the pipe looks
> like this:
> 
> rtspsrc location=rtsp://****@****:554/h264/media.amp protocols=0x00000004
> latency=200 ! rtph264depay ! ffdec_h264 ! queue leaky=2
> max-size-buffers=10 ! jpegenc idct-method=1 quality=100 ! multipartmux
> boundary=*** ! queue name=exit leaky=2 max-size-buffers=10 ! fakesink
> 
> The ip camera stream is received over a mobile 3g network and therefore
> we are forced to use TCP instead of UDP due to heavy packet loss.
> 
> Our question for you is if anyone knows how to optimize the pipe and its
> caps to handle a live stream with minimal delay?

-- 
Arnout Vandecappelle                               arnout at mind be
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