problem with synchronization between multiple audio clients

[Henrik Kjær Nielsen]

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



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