<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><pre style="padding: 0px; margin-top: 0px; margin-bottom: 0px; font-size: 13px; background-color: rgb(255, 255, 255); position: static; z-index: auto; "><code>Thanks Sebastian,</code></pre><pre style="padding: 0px; margin-top: 0px; margin-bottom: 0px; font-size: 13px; background-color: rgb(255, 255, 255); position: static; z-index: auto; ">Last question,</pre><pre style="padding: 0px; margin-top: 0px; margin-bottom: 0px; font-size: 13px; background-color: rgb(255, 255, 255); position: static; z-index: auto; ">in this example <a href="http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh">http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh</a> </pre><pre style="padding: 0px; margin-top: 0px; margin-bottom: 0px; font-size: 13px; background-color: rgb(255, 255, 255); position: static; z-index: auto; ">I see: </pre><pre style="padding: 0px; margin-top: 0px; margin-bottom: 0px; font-size: 13px; background-color: rgb(255, 255, 255); position: static; z-index: auto; "><code># .-------. .-------. .-------. .----------. .-------.
# |v4lssrc| |h264enc| |h264pay| | rtpbin | |udpsink| RTP
# | src->sink src->sink src->send_rtp send_rtp->sink | port=5000
# '-------' '-------' '-------' | | '-------'
# | |
# | | .-------.
# | | |udpsink| RTCP
# | send_rtcp->sink | port=5001
# .-------. | | '-------' sync=false
# RTCP |udpsrc | | | async=false
# port=5005 | src->recv_rtcp |
# '-------' | |
# | |
# .--------. .-------. .-------. | | .-------.
# |audiosrc| |alawenc| |pcmapay| | rtpbin | |udpsink| RTP
# | src->sink src->sink src->send_rtp send_rtp->sink | port=5002
# '--------' '-------' '-------' | | '-------'
# | |
# | | .-------.
# | | |udpsink| RTCP
# | send_rtcp->sink | port=5003
# .-------. | | '-------' sync=false
# RTCP |udpsrc | | | async=false
# port=5007 | src->recv_rtcp |
# '-------' '----------'</code></pre><div><br></div><div>Which sources should I use for iOS instead of <span style="background-color: rgb(255, 255, 255); font-size: 13px; ">audiosrc, v4lssrc</span>?</div><div>I mean, I suppose video4linux is not suitable for iOS platform :-)</div><div><br></div><div>Thanks a lot</div><div>Elio</div><div><br></div><div><br></div><div><div>On Oct 20, 2013, at 5:10 PM, Sebastian Dröge <<a href="mailto:sebastian@centricular.com">sebastian@centricular.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">On So, 2013-10-20 at 14:49 +0200, elio francesconi wrote:<br><blockquote type="cite">I try to explain better my purpose.<br>I' m evaluating to use GStreamer to open in/out audio/video streams.<br>My purpose is to:<br>● capture audio and send as a rtp stream<br>● capture video and send as a rtp stream<br>● receive a stream rtp and play audio<br>● receive a stream rtp and play video<br><br>Do you think gstreamer 1.2 for ios can handle this?<br></blockquote><br>Yes<br><br><blockquote type="cite">In case can you provide me some hints?<br></blockquote><br>Take a look at the shell scripts in gst-plugins-good/tests/examples/rtp.<br>You'll have to convert that to C and do it in your application, but<br>that's exactly how it would work.<br><br>Alternatively you could also do RTSP. RTSP client support is directly<br>included, for server support you need to get gst-rtsp-server. The latter<br>would be required to build yourself (which you could do by cerbero for<br>example, or just manually).<br>_______________________________________________<br>gstreamer-devel mailing list<br><a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a><br>http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel<br></blockquote></div><br></body></html>