<html> <head> <style></style></head> <body class='hmmessage'><div dir='ltr'>The following pipeline was working for me:<div><span style="color: rgb(46, 47, 45); font-family: Arial, Helvetica, sans-serif; font-size: 12px; line-height: 18px; background-color: rgb(255, 255, 255);">gst-launch videotestsrc ! queue ! x264enc ! flvmux name=muxer ! rtmpsink location='rtmp://<streaming_url>' audiotestsrc ! queue ! audioconvert ! muxer.</span></div><div><span style="color: rgb(46, 47, 45); font-family: Arial, Helvetica, sans-serif; font-size: 12px; line-height: 18px; background-color: rgb(255, 255, 255);"><br></span></div><div><span style="font-size: 12pt;">However, a similar pipeline was not working within our application.</span></div><div><br></div><div>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 <a href="http://www.oz9aec.net/index.php/gstreamer/410-x264enc-problem-in-gstreamer-video-switcher-solved">here</a>.</div><div><br></div><div>We found that we could resolve the issue by</div><div>a) making the queue in the other output bin (before the ffmpegcolourspace and autovideosink) leaky (as described in the above link) or</div><div>b) 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).</div><div><br></div><div>Cheers,</div><div><br></div><div>Mike</div><div><br><div><div id="SkyDrivePlaceholder"></div>> Date: Mon, 26 Nov 2012 02:33:23 -0800<br>> From: orione1974@email.it<br>> To: gstreamer-devel@lists.freedesktop.org<br>> Subject: Re: rtmpsink<br>> <br>> Hi Mike.<br>> I also work with a similar situation in rtmp streaming functionality.<br>> Can you post your pipeline? For example <br>> <br>> gst-launch-0.10 ksvideosrc ! ffmpegcolorspace ! timeoverlay ! x264enc<br>> tune=zerolatency ! mux. audiotestsrc ! faac ! audio/mpeg, mpegversion=4,<br>> profile=lc ! mux. flvmux name=mux ! rtmpsink<br>> location="rtmp://192.168.0.233/live/gstreamer_test"<br>> <br>> <br>> <br>> Mike Chong wrote<br>> > Hey so I've been working on freeseer to add rtmp streaming functionality,<br>> > but I've hit a snag.<br>> > I've been able to set up, with gst-launch and a short python script, a<br>> > pipeline that works (ie, streams rtmp content to a server). However, for<br>> > the rtmp plugin, I create an output bin that uses rtmpsink. While there<br>> > doesn't appear to be anything wrong with the bin that I create (it works<br>> > in the short prototype script), when it's put in the pipeline, the<br>> > pipeline won't play. It seems to get as far as creating the rtmp object,<br>> > and then stops.<br>> > Now, very likely this is a bug in my code, but I'm wondering, what are<br>> > some good angles to approach this? Running the app with GST_DEBUG=bin*:6,<br>> > it looks like my bin receives an async-start message from rtmpsink when<br>> > I'm setting the state to STATE_PAUSED, but never receives a async-done<br>> > message.<br>> > Thanks. <br>> > _______________________________________________<br>> > gstreamer-devel mailing list<br>> <br>> > gstreamer-devel@.freedesktop<br>> <br>> > http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel<br>> <br>> <br>> <br>> <br>> <br>> --<br>> View this message in context: http://gstreamer-devel.966125.n4.nabble.com/rtmpsink-tp4657056p4657062.html<br>> Sent from the GStreamer-devel mailing list archive at Nabble.com.<br>> _______________________________________________<br>> gstreamer-devel mailing list<br>> gstreamer-devel@lists.freedesktop.org<br>> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel<br></div></div> </div></body> </html>