<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=rtsp://ip_host:port0/audio0 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=rtsp://ip_host:port0/audio0 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=rtsp://ip_host:port1/audio1 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=rtsp://ip_host:port0/audio0 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=rtsp://ip_host:port0/audio0 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=rtsp://ip_host:port1/audio1 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? Thanks in advance! -- <div dir="ltr">Douglas Amorim Ferreira</div> </div>