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

[john oliver]

Hi Viraj,

 

I was able to play the the filesrc and aslasink in the same pipeline - the problem was when they were separated by an rtp link. However I have things going now with a customised file src which seems to send things out at a rate which vlc recognises. I don't need to receive it in gst-inspect anymore so I'm not going to persevere with this line of enquiry. But I appreciate your help on this.

 

Thanks,

John
 


Date: Mon, 5 Oct 2009 23:50:18 +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.

John,
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 filesrc?

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 queue.

- 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

------------------------------------------------------------------------------
Come build with us! The BlackBerry® Developer Conference in SF, CA
is the only developer event you need to attend this year. Jumpstart your
developing skills, take BlackBerry mobile applications to market and stay
ahead of the curve. Join us from November 9-12, 2009. Register now!
http://p.sf.net/sfu/devconf
_______________________________________________
gstreamer-devel mailing list
gstreamer-devel at lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/gstreamer-devel



-- 
- Viraj
Reality is merely an illusion, albeit a very persistent one.
 		 	   		  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freedesktop.org/archives/gstreamer-devel/attachments/20091008/ed48fd11/attachment.htm>