<div dir="ltr">Henrik,<div> </div><div>Have you run and played with Aurena?</div><div><a href="https://github.com/thaytan/aurena">https://github.com/thaytan/aurena</a> </div><div> </div><div>Luis</div></div><div class="gmail_extra"> <div class="gmail_quote">On 30 April 2014 08:22, Henrik Kjær Nielsen <<a href="mailto:henrikkjaernielsen@hotmail.com" target="_blank">henrikkjaernielsen@hotmail.com</a>> wrote: <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"> I am also very interested in using GStreamer to stream synchronized audio (and video) to multiple clients. Now that the new gst-rstp-server v. 1.2.3 has been out for some time, I hope that somebody is willing to share some information on how to prepare a simple synchronization experiment as a proof of concept. I have not been able to find any hints on how to do this except from what has been written in this thread. This is what I did so far: I have created a simple rtsp server based on example code from the source repository: #include <gst/gst.h> #include <gst/rtsp-server/rtsp-server.h> static void media_configure(GstRTSPMediaFactory * factory, GstRTSPMedia * media) { gst_rtsp_media_use_time_provider(media, TRUE); } int main(int argc, char *argv[]) { GMainLoop *loop; GstRTSPServer *server; GstRTSPMountPoints *mounts; GstRTSPMediaFactory *factory; GstRTSPAddressPool *pool; gst_init(&argc, &argv); loop = g_main_loop_new(NULL, FALSE); /* create a server instance */ server = gst_rtsp_server_new(); /* get the mount points for this server, every server has a default object * that be used to map uri mount points to media factories */ mounts = gst_rtsp_server_get_mount_points(server); /* make a media factory for a test stream. The default media factory can use * gst-launch syntax to create pipelines. * any launch line works as long as it contains elements named pay%d. Each * element with pay%d names will be a stream */ factory = gst_rtsp_media_factory_new(); gchar *str; str = g_strdup_printf("( " "filesrc location=%s ! qtdemux name=d " "d. ! queue ! rtph264pay pt=96 name=pay0 " "d. ! queue ! rtpmp4apay pt=97 name=pay1 " ")", argv[1]); gst_rtsp_media_factory_set_launch(factory, str); g_free(str); gst_rtsp_media_factory_set_shared(factory, TRUE); g_signal_connect(factory, "media-configure", (GCallback)media_configure, NULL); /* attach the test factory to the /test url */ gst_rtsp_mount_points_add_factory(mounts, "/test", factory); /* don't need the ref to the mapper anymore */ g_object_unref(mounts); /* attach the server to the default maincontext */ gst_rtsp_server_attach(server, NULL); /* start serving */ g_print("stream ready at rtsp://<a href="http://127.0.0.1:8554/test\n" target="_blank">127.0.0.1:8554/test\n</a>"); g_main_loop_run(loop); return 0; } As suggested earlier in this thread, I call gst_rtsp_media_use_time_provider(media, TRUE). On the clients (I am running multiple client instances on the same computer as the rtsp server instance) I simply run gst-launch-1.0 -v playbin uri="rtsp://<a href="http://127.0.0.1:8554/test" target="_blank">127.0.0.1:8554/test</a>". Well, it does not work, i.e. the audio (and video) is not synchronized. Calling gst_rtsp_media_use_time_provider(media, TRUE) has no effect. The audio is slightly misaligned when running two client instances simultaneously - it sounds a bit like bathroom acoustics. What do I have to do to make it synchronized? My platform is Windows. Regards Henrik -- View this message in context: <a href="http://gstreamer-devel.966125.n4.nabble.com/problem-with-synchronization-between-multiple-audio-clients-tp4663231p4666673.html" target="_blank">http://gstreamer-devel.966125.n4.nabble.com/problem-with-synchronization-between-multiple-audio-clients-tp4663231p4666673.html</a> Sent from the GStreamer-devel mailing list archive at Nabble.com. <div class="HOEnZb"><div class="h5">_______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a> <a href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" target="_blank">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> </div></div></blockquote></div> </div>