<html> <head> <style></style></head> <body class='hmmessage'><div dir='ltr'>Hi Tim,<div><br></div><div> Thank you very much for your help. I hope I will see the patch in git master soon.</div><div><br></div><div>Currently I'm using a tricky way as workaround as below:</div><div><br></div><div>---------------------------------------------------------------------------------------------------------------</div><div><br></div><div><p style="font-size: 11px; font-family: Menlo; ">gst-launch-<span style="color: #272ad8">1.0</span> -v rtpbin name=rtpbin latency=$LATENCY \</p> <p style="font-size: 11px; font-family: Menlo; "> udpsrc caps=$VIDEO_CAPS port=<span style="color: #272ad8">5000</span> ! rtpbin.recv_rtp_sink_<span style="color: #272ad8">0</span> \</p> <p style="font-size: 11px; font-family: Menlo; "> rtpbin. ! $VIDEO_DEC ! $VIDEO_SINK \</p> <p style="font-size: 11px; font-family: Menlo; "> udpsrc port=<span style="color: #272ad8">5001</span> ! rtpbin.recv_rtcp_sink_<span style="color: #272ad8">0</span> \</p> <p style="font-size: 11px; font-family: Menlo; "> rtpbin.send_rtcp_src_<span style="color: #272ad8">0</span> ! udpsink port=<span style="color: #272ad8">5005</span> host=$DEST sync=false async=false &</p> <p style="font-size: 11px; font-family: Menlo; min-height: 13px; "><br></p> <p style="font-size: 11px; font-family: Menlo; ">gst-launch-<span style="color: #272ad8">0.10</span> udpsrc port=<span style="color: #272ad8">5002</span> caps=<span style="color: #d12f1b">"application/x-rtp"</span> ! queue ! rtppcmudepay ! mulawdec ! audioconvert ! autoaudiosink</p> <p style="font-size: 11px; font-family: Menlo; min-height: 13px; "><br></p> <p style="font-size: 11px; font-family: Menlo; min-height: 13px; "><br></p> <p style="font-size: 11px; font-family: Menlo; ">killall -<span style="color: #272ad8">9</span> gst-launch-<span style="color: #272ad8">1.0</span></p><div><span style="color: #272ad8"><br></span></div><div>_______________________________________________________________________</div><div><br></div><div>Looks like it works well! :) </div><div><br></div><div>Thank you very much for your help again.</div><div><br></div><div>Cheers,</div><div>Haiming Wang</div><div><br><br><div><div id="SkyDrivePlaceholder"></div>> Subject: Re: osx_audio<br>> From: t.i.m@zen.co.uk<br>> To: gstreamer-devel@lists.freedesktop.org<br>> Date: Tue, 9 Apr 2013 09:34:24 +0100<br>> <br>> On Tue, 2013-04-09 at 10:31 +0800, Haiming Wang wrote:<br>> <br>> Hi,<br>> <br>> > My questions is: <br>> > 1. what does the "not ported " mean?<br>> <br>> It means the source code is still for the 0.10 API and doesn't compile<br>> or work with 1.0 yet. Someone needs to port it to the new API.<br>> <br>> > 2. How can I build osxaudio on Mac OSX with the latest code (based on<br>> > streamer-1.0)?<br>> <br>> There is a patch in bugzilla which you can test:<br>> <br>> https://bugzilla.gnome.org/show_bug.cgi?id=697103<br>> <br>> I think it will land in git master soon.<br>> <br>> <br>> > 3. Can I simply copy the binaries of osxaudiosink based on<br>> > gstreamer-0.10 to streamer-1.0 directory to make it work?<br>> > <br>> <br>> No, that won't work.<br>> <br>> Cheers<br>> -Tim<br>> <br>> <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> </div></body> </html>