rtmpsink

Mike Chong mike.chong at hotmail.com
Mon Nov 26 17:36:30 PST 2012


The following pipeline was working for me:gst-launch videotestsrc ! queue ! x264enc ! flvmux name=muxer ! rtmpsink location='rtmp://<streaming_url>' audiotestsrc ! queue ! audioconvert ! muxer.
However, a similar pipeline was not working within our application.
We discovered that the issue was we were using a tee, and splitting the pipeline to x264enc, and another bin that included an ffmpegcolourspace, and an autovideosink (something I had overlooked...).  The issue we were encountering was described here.
We found that we could resolve the issue bya) making the queue in the other output bin (before the ffmpegcolourspace and autovideosink) leaky (as described in the above link) orb) setting tune=zerolatency.  I'm not sure which of the magics the zerolatency tuning uses fixes the problem, but to say something wrong in the hopes of being corrected, I would guess that one of the zerolatency settings stop x264enc from taking its time and prevent buffers from being filled up (please correct me, I'm curious).
Cheers,
Mike
> Date: Mon, 26 Nov 2012 02:33:23 -0800
> From: orione1974 at email.it
> To: gstreamer-devel at lists.freedesktop.org
> Subject: Re: rtmpsink
> 
> Hi Mike.
> I also work with a similar situation in rtmp streaming functionality.
> Can you post your pipeline? For example 
> 
> gst-launch-0.10 ksvideosrc ! ffmpegcolorspace ! timeoverlay ! x264enc
> tune=zerolatency ! mux. audiotestsrc ! faac ! audio/mpeg, mpegversion=4,
> profile=lc !  mux.  flvmux name=mux ! rtmpsink
> location="rtmp://192.168.0.233/live/gstreamer_test"
> 
> 
> 
> Mike Chong wrote
> > Hey so I've been working on freeseer to add rtmp streaming functionality,
> > but I've hit a snag.
> > I've been able to set up, with gst-launch and a short python script, a
> > pipeline that works (ie, streams rtmp content to a server).  However, for
> > the rtmp plugin, I create an output bin that uses rtmpsink.  While there
> > doesn't appear to be anything wrong with the bin that I create (it works
> > in the short prototype script), when it's put in the pipeline, the
> > pipeline won't play.  It seems to get as far as creating the rtmp object,
> > and then stops.
> > Now, very likely this is a bug in my code, but I'm wondering, what are
> > some good angles to approach this?  Running the app with GST_DEBUG=bin*:6,
> > it looks like my bin receives an async-start message from rtmpsink when
> > I'm setting the state to STATE_PAUSED, but never receives a async-done
> > message.
> > Thanks. 		 	   		  
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> 
> > gstreamer-devel at .freedesktop
> 
> > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
> 
> 
> 
> 
> 
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