[gst-devel] alsasrc ! alawenc ! rtppcmapay ! udpsink - can receive, record then playback, but not playback live.

Viraj Karandikar virajk at gmail.com
Mon Oct 5 20:20:18 CEST 2009

When you play from filesrc, the alsasink will make sure that the pipeline
runs realtime.
So actually no hacking is required. Did you face any issues without hacking

Reason for sporadic bursts could be loss of packets while pipeline is busy
in playing out.
Try putting a queue before alsasink. That will decouple the udp receive and
alsasink playout and should avoid packets loss. But I could work without any

- Viraj

On Mon, Oct 5, 2009 at 8:38 PM, john oliver <johnwpoliver at hotmail.com>wrote:

> ------------------------------
> Date: Mon, 5 Oct 2009 19:56:50 +0530
> From: virajk at gmail.com
> To: gstreamer-devel at lists.sourceforge.net
> Subject: Re: [gst-devel] alsasrc ! alawenc ! rtppcmapay ! udpsink - can
> receive, record then playback, but not playback live.
> Hi,
> I dont think there is any bug in gstreamer.
> I have successfully run "audio capture - UDP send" on one board and "UDP
> receive - audio playout" on other board.
> But we didnt specify ssrc, clock-base, seqnum-base etc.. parameters for
> udpsrc.
> Try this once (your command with minimal udpsrc parameters)
> gst-launch-0.10 -v udpsrc  port=1234 caps="application/x-rtp,
> media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA"  !
> rtppcmadepay  ! audio/x-alaw, rate=8000, channels=1  ! alawdec  ! alsasink
> Thanks Viraj, however I am still only hearing sporadic bursts with that
> pipeline at the client end. I should have also mentioned that when I
> successfully streamed to vlc, on the server end I was time-stamping from a
> filesrc, but I had hacked gstfilesrc.c to sleep between frames for the
> sample length equivelent to my frame size.
> John
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- Viraj
Reality is merely an illusion, albeit a very persistent one.
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