<html><head></head><body><div style="color:#000; background-color:#fff; font-family:lucida console, sans-serif;font-size:13px"><br><div id="yui_3_16_0_ym19_1_1509048077181_10584"><span></span></div> <div class="qtdSeparateBR" id="yui_3_16_0_ym19_1_1509048077181_10899"><br><br></div><div class="yahoo_quoted" id="yui_3_16_0_ym19_1_1509048077181_10611" style="display: block;"> <div style="font-family: lucida console, sans-serif; font-size: 13px;" id="yui_3_16_0_ym19_1_1509048077181_10610"> <div style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 16px;" id="yui_3_16_0_ym19_1_1509048077181_10609"> <div dir="ltr" id="yui_3_16_0_ym19_1_1509048077181_10897"><font id="yui_3_16_0_ym19_1_1509048077181_10898" face="Arial" size="2"> Le Jeudi 26 octobre 2017 20h35, Stefan Sauer <ensonic@hora-obscura.de> a écrit :<br></font></div> <br><br> <div class="y_msg_container" id="yui_3_16_0_ym19_1_1509048077181_10608"><div dir="ltr" id="yui_3_16_0_ym19_1_1509048077181_10607">On 10/25/2017 01:43 PM, avinashgst wrote:<br clear="none">> Hi,<br clear="none">><br clear="none">> I have the pipeline in which need to mix 2 audio at run time.<br clear="none">><br clear="none">> when single file src1:<br clear="none">> filesrc ! decodebin ! audioresample ! audioconvert ! (sink_0) adder !<br clear="none">> audioconvert ! autoaudiosink<br clear="none">><br clear="none">> During runtime adding one more file src2:<br clear="none">> filesrc ! decodebin ! audioresample ! audioconvert ! ---><br clear="none">> <br clear="none">> (sink_0) (sink_01) adder ! audioconvert ! autoaudiosink<br clear="none">> filesrc ! decodebin ! audioresample ! audioconvert !----><br clear="none">><br clear="none">> This is working fine. <br clear="none">><br clear="none">> To pause src1:<br clear="none">> unlinking audioconvert src and sink_0 by gst_pad_unlink and called<br clear="none">> gst_element_release_request_pad for sink_0.<br clear="none">> and setting filesrc, decodebin, audioresample, audioconvert to<br clear="none">> GST_STATE_PAUSE<br clear="none">> Src2 is continue to play and its working as expected.<br clear="none">><br clear="none">> Now after some time wants to resume the src1 while src2 still playing.<br clear="none">> filesrc ! decodebin ! audioresample ! audioconvert ! ---><br clear="none">> <br clear="none">> (sink_01) (sink_02) adder ! audioconvert ! autoaudiosink<br clear="none">> filesrc ! decodebin ! audioresample ! audioconvert !----><br clear="none">><br clear="none">> linked audioconvert src and sink_2 by gst_pad_link and setting<br clear="none">> filesrc, decodebin, audioresample, audioconvert to GST_STATE_PLAYING<br clear="none">><br clear="none">> In that case Src2 also stopped playing and getting underflow error.<br clear="none">> From gstreamer log its seems that src1 is not pushing any data after setting<br clear="none">> elements to GST_STATE_PLAYING<br clear="none">><br clear="none">> Any suggestion how to achieve pause/resume operation with adder element. <br clear="none">><br clear="none">> Regards,<br clear="none">> Avinash<br clear="none">><br clear="none">If you pause/unpause branches you might want to try liveadder. Also you<br clear="none">don't describe how you re-link src1. Are you using pad-probes? We have<br clear="none">some examples under gst-plugins-base/tests/examples/dynamic/<br clear="none"><br clear="none">Stefan<div class="yqt3626365209" id="yqtfd81617"><br clear="none">_______________________________________________<br clear="none">gstreamer-devel mailing list<br clear="none"><a shape="rect" ymailto="mailto:gstreamer-devel@lists.freedesktop.org" href="mailto:gstreamer-devel@lists.freedesktop.org" id="yui_3_16_0_ym19_1_1509048077181_10942">gstreamer-devel@lists.freedesktop.org</a><br clear="none"><a shape="rect" href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" target="_blank" id="yui_3_16_0_ym19_1_1509048077181_10913">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a><br clear="none"></div></div><br><div dir="ltr" id="yui_3_16_0_ym19_1_1509048077181_10921">You can try to set provide-clock to false on your audio sink.<br id="yui_3_16_0_ym19_1_1509048077181_10922"></div><div id="yui_3_16_0_ym19_1_1509048077181_10923"><br id="yui_3_16_0_ym19_1_1509048077181_10924"></div><div dir="ltr" id="yui_3_16_0_ym19_1_1509048077181_10925">Which version of GStreamer are you using and on which OS ?</div></div> </div> </div> </div></div></body></html>