problem with synchronization between multiple audio clients

[Luis de Bethencourt]

Henrik,

Have you run and played with Aurena?
https://github.com/thaytan/aurena

Luis


On 30 April 2014 08:22, Henrik Kjær Nielsen
<henrikkjaernielsen at hotmail.com>wrote:

> 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://127.0.0.1:8554/test\n");
>         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://127.0.0.1:8554/test".
>
> 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:
> http://gstreamer-devel.966125.n4.nabble.com/problem-with-synchronization-between-multiple-audio-clients-tp4663231p4666673.html
> Sent from the GStreamer-devel mailing list archive at Nabble.com.
> _______________________________________________
> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
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>
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