<html> <head> <meta content="text/html; charset=windows-1252" http-equiv="Content-Type"> </head> <body bgcolor="#FFFFFF" text="#000000"> <div class="moz-cite-prefix">On 09/15/2014 04:08 PM, Douglas Amorim Ferreira wrote: </div> <blockquote cite="mid:CAGJZr=VbhmNTXVCcZg_ewUpSxA1t-EbkufrHyLoDYFsOtXZh0Q@mail.gmail.com" type="cite"> <div dir="ltr">Hello, I need to mix two different live audio streams into one. Each of these audio streams is from a rtsp-server. To listen just one of these audio streams, I used this pipeline: gst-launch-1.0 rtspsrc location=<a class="moz-txt-link-freetext" href="rtsp://ip_host:port0/audio0">rtsp://ip_host:port0/audio0</a> latency=2000 ! application/x-rtp, "media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA" ! rtppcmadepay ! alawdec ! autoaudiosink And this pipeline worked fine. Thus, I created a pipeline to mix differents audio streams: gst-launch-1.0 rtspsrc location=<a class="moz-txt-link-freetext" href="rtsp://ip_host:port0/audio0">rtsp://ip_host:port0/audio0</a> latency=2000 ! application/x-rtp, "media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA" ! rtppcmadepay ! alawdec ! audioresample ! liveadder name=m_pAdder ! interleave name=m_pInterleave ! autoaudiosink rtspsrc location=<a class="moz-txt-link-freetext" href="rtsp://ip_host:port1/audio1">rtsp://ip_host:port1/audio1</a> latency=2000 ! application/x-rtp, "media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA" ! rtppcmadepay ! alawdec ! audioresample ! m_pAdder. And also was good.. Now I wanna have freedom to play any of these audio in any channel (left and right), with the possibility to reconfigure in real time. Ok, I think the first step is convert a single mono audio stream in a stereo one. And I do it: gst-launch-1.0 interleave name=m_pInterleave ! autoaudiosink rtspsrc location=<a class="moz-txt-link-freetext" href="rtsp://ip_host:port0/audio0">rtsp://ip_host:port0/audio0</a> latency=2000 ! application/x-rtp, "media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA" ! rtppcmadepay ! alawdec ! audioconvert ! "audio/x-raw, channels=2" ! deinterleave name=m_pAudio0 m_pAudio0.src_0 ! queue ! m_pInterleave.sink_0 m_pAudio0.src_1 ! queue ! m_pInterleave.sink_1 Works! Now, I just mix two stereo streams (mix left channel0 with left channel1 and right with right). The pipeline is: gst-launch-1.0 interleave name=m_pInterleave ! autoaudiosink rtspsrc location=<a class="moz-txt-link-freetext" href="rtsp://ip_host:port0/audio0">rtsp://ip_host:port0/audio0</a> latency=2000 ! application/x-rtp, "media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA" ! rtppcmadepay ! alawdec ! audioconvert ! "audio/x-raw,channels=2" ! deinterleave name=m_pAudio0 m_pAudio0.src_0 ! queue ! liveadder name=m_pAdderEsq ! m_pInterleave.sink_0 rtspsrc location=<a class="moz-txt-link-freetext" href="rtsp://ip_host:port1/audio1">rtsp://ip_host:port1/audio1</a> latency=2000 ! application/x-rtp, "media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA" ! rtppcmadepay ! alawdec ! audioconvert ! "audio/x-raw,channels=2" ! deinterleave name=m_pAudio1 m_pAudio1.src_0 ! queue ! m_pAdderEsq. m_pAudio0.src_1 ! queue ! liveadder name=m_pAdderDir ! m_pInterleave.sink_1 m_pAudio1.src_1 ! queue ! m_pAdderDir. And... I've got EOS !!!! Anybody know what is the problem? What is the mistake I made? </div> </blockquote> The pipeline looks fine. Does it fail if you replace the rtspsrc and rtp depayloading + decoding with audiotestsrcs? It should be easy to track what is pushing EOS from a GST_DEBUG=6 log. Search for "creating new event 0x[0-9a-f]* eos" and look at the lines around it to find the elements that are creating the EOS event. It might be happening for both your rtsp sources. You could also try using audiomixer instead of liveadder. <blockquote cite="mid:CAGJZr=VbhmNTXVCcZg_ewUpSxA1t-EbkufrHyLoDYFsOtXZh0Q@mail.gmail.com" type="cite"> <div dir="ltr"> Thanks in advance! -- <div dir="ltr">Douglas Amorim Ferreira</div> </div> <fieldset class="mimeAttachmentHeader"></fieldset> <pre wrap="">_______________________________________________ gstreamer-devel mailing list <a class="moz-txt-link-abbreviated" href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a> <a class="moz-txt-link-freetext" href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> </pre> </blockquote> <pre class="moz-signature" cols="72">-- Thiago Sousa Santos Senior Multimedia Engineer, Open Source Group Samsung Research America - Silicon Valley</pre> </body> </html>