Re: Issue with rtsp server source h264 encoded

videotestsrc ! x264enc tune=zerolayency tends to negotiate yuv444+high
profile. That can cause introp issues. Ide suggest a caps filter after the
encoder with caps video/x-h264,profile=main (or constrained-baseline,
though that will not compress as much).

Hopefully this will improve your situation.

Le jeu. 21 sept. 2023, 16 h 30, LEVY David via gstreamer-devel <
[email protected]> a écrit :

> Same behaviour.
>
>
>
> I just noted that “need-data” is only called once with ffplay, and twice
> with gst-launch (probably once udp and once tcp)
>
>
>
> *From:* Michael Gruner <[email protected]>
> *Sent:* Thursday, September 21, 2023 2:45 PM
> *To:* Discussion of the development of and with GStreamer <
> [email protected]>
> *Cc:* LEVY David <[email protected]>
> *Subject:* Re: Issue with rtsp server source h264 encoded
>
>
>
> Hi
>
>
>
> Try sending periodic I frames, "x264enc key-int-max=10" or,
> programmatically:
>
>
>
>     g_object_set( lEncoder, "key-int-max", 10, NULL ); // 4 = zerolatency
>
>
>
>
>
> 10 is just a made up number to send a reference frame every second (given
> that your framerate is 10/1). You may space them according to your needs.
>
>
>
> Michael.
>
>
>
> On 21 Sep 2023, at 09:59, LEVY David via gstreamer-devel <
> [email protected]> wrote:
>
>
>
> Hello,
> i'm new to gstreamer / rtsp so, sorry if it's a basic mistake.
>
>
>
> Im trying to set up a pipeline that end with an appsink that push data
> into a rtsp server starting with an appsrc.
>
>
>
> It would be:
>
> videotestsrc -> x264enc -> appsink (push data to src as soon as received)
>
> and
>
> appsrc -> rtph264pay
>
>
>
> I can’t find any example similar, but I found this one:
>
> https://github.com/GStreamer/gst-rtsp-server/blob/master/examples/test-appsrc.c
>
>
>
> Which does:
> appsrc-> videoconvert -> x264enc -> rtph264pay (and uses “need-data”)
>
>
>
> I have successfully tweaked it to
> videotestsrc -> appsink
>
> appsrc-> videoconvert -> x264enc -> rtph264pay (and uses “need-data”)
>
>
>
> My next step would be
>
> Videotestsrc -> x264enc -> appsink
>
> appsrc-> rtph264pay (and uses “need-data”)
>
>
>
> but when I try to read the stream from either
>
>     gst-launch-1.0 --gst-debug=rtspsrc:2 rtspsrc location=
> rtsp://127.0.0.1:8554/test ! queue ! rtph264depay ! h264parse !
> avdec_h264 ! videoconvert ! videoscale ! autovideosink
>
> or ffplay rtsp://127.0.0.1:8554/test
>
>
>
> I don’t have any playback:
>
> Setting pipeline to PAUSED ...
>
> Pipeline is live and does not need PREROLL ...
>
> Got context from element 'autovideosink0': gst.gl.GLDisplay=context,
> gst.gl.GLDisplay=(GstGLDisplay)"\(GstGLDisplayWayland\)\ gldisplaywayland0";
>
> Progress: (open) Opening Stream
>
> Pipeline is PREROLLED ...
>
> Prerolled, waiting for progress to finish...
>
> Progress: (connect) Connecting to rtsp://127.0.0.1:8554/test
>
> Progress: (open) Retrieving server options
>
> Progress: (open) Retrieving media info
>
> Progress: (request) SETUP stream 0
>
> Progress: (open) Opened Stream
>
> Setting pipeline to PLAYING ...
>
> New clock: GstSystemClock
>
> Progress: (request) Sending PLAY request
>
> Redistribute latency...
>
> Progress: (request) Sending PLAY request
>
> Redistribute latency...
>
> Progress: (request) Sent PLAY request
>
> Redistribute latency...
>
> Redistribute latency...
>
> 0:00:05.945770874 15996 0x55c83ec97f60 WARN                 rtspsrc
> gstrtspsrc.c:5964:gst_rtspsrc_reconnect:<rtspsrc0> warning: Could not
> receive any UDP packets for 5.0000 seconds, maybe your firewall is blocking
> it. Retrying using a tcp connection.
>
> WARNING: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Could
> not read from resource.
>
> Additional debug info:
>
> ../gst/rtsp/gstrtspsrc.c(5964): gst_rtspsrc_reconnect ():
> /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:
>
> Could not receive any UDP packets for 5.0000 seconds, maybe your firewall
> is blocking it. Retrying using a tcp connection.
>
> Redistribute latency...
>
>
>
> And it is stuck here
>
> I’m using the code below:
>
>
>
> #include <gst/gst.h>
>
>
>
> #include <gst/app/gstappsink.h>
>
> #include <gst/app/gstappsrc.h>
>
> #include <gst/rtsp-server/rtsp-server.h>
>
>
>
> GstElement *gSink = nullptr;
>
>
>
> static void InitPipeline()
>
> {
>
>     GstElement *lPipeline = gst_pipeline_new( "x264-pipeline" );
>
>
>
>     GstElement *lSource  = gst_element_factory_make( "videotestsrc",
> "source" );
>
>     GstElement *lEncoder = gst_element_factory_make( "x264enc",
> "x264-encoder" );
>
>     GstElement *lSink    = gst_element_factory_make( "appsink", "appsink"
> );
>
>
>
>     gst_bin_add_many( GST_BIN( lPipeline ), lSource, lEncoder, lSink, NULL
>  );
>
>     // gst_bin_add_many( GST_BIN( lPipeline ), lSource, lSink, NULL );
>
>     g_object_set( lEncoder, "tune", 4, NULL ); // 4 = zerolatency
>
>
>
>     GstCaps *lCaps =
>
>         gst_caps_new_simple( "video/x-raw", "width", G_TYPE_INT, 384,
> "height", G_TYPE_INT, 288, "framerate", GST_TYPE_FRACTION, 10, 1, NULL );
>
>     if( !gst_element_link_filtered( lSource, lEncoder, lCaps ) )
>
>     // if( !gst_element_link_filtered( lSource, lSink, lCaps ) )
>
>     {
>
>         gst_caps_unref( lCaps );
>
>         throw;
>
>     }
>
>     gst_caps_unref( lCaps );
>
>     gst_element_link( lEncoder, lSink );
>
>
>
>     GstStateChangeReturn ret = gst_element_set_state( lPipeline,
> GST_STATE_PLAYING );
>
>     if( ret == GST_STATE_CHANGE_FAILURE )
>
>     {
>
>         throw;
>
>     }
>
>     gSink = lSink;
>
> }
>
>
>
> static void need_data( GstElement *appsrc, guint unused, void *aUnused )
>
> {
>
>     GstSample *lSample = gst_app_sink_pull_sample( GST_APP_SINK_CAST(
> gSink ) );
>
>     GstFlowReturn lRet = gst_app_src_push_sample( GST_APP_SRC( appsrc ),
> lSample );
>
>     gst_sample_unref( lSample );
>
>     return;
>
> }
>
>
>
> static void media_configure( GstRTSPMediaFactory *factory, GstRTSPMedia *
> media, gpointer user_data )
>
> {
>
>     GstElement *element, *appsrc;
>
>
>
>     element = gst_rtsp_media_get_element( media );
>
>     appsrc = gst_bin_get_by_name_recurse_up( GST_BIN( element ), "mysrc"
> );
>
>
>
>     gst_util_set_object_arg( G_OBJECT( appsrc ), "format", "time" );
>
>
>
>     if( GstPad *pad = gst_element_get_static_pad( gSink, "sink" ) )
>
>     {
>
>         g_print( "Got pad\n" );
>
>         GstCaps *caps = gst_pad_get_current_caps( pad );
>
>         if( caps )
>
>         {
>
>             auto *caps_str = gst_caps_to_string( caps );
>
>             g_print( "Got caps %s\n", caps_str );
>
>             g_object_set( G_OBJECT( appsrc ), "caps", caps, NULL );
>
>
>
>             gst_caps_unref( caps );
>
>         }
>
>     }
>
>
>
>     g_signal_connect( appsrc, "need-data", (GCallback)need_data, NULL );
>
>     gst_object_unref( appsrc );
>
>     gst_object_unref( element );
>
> }
>
>
>
> int main( int argc, char *argv[] )
>
> {
>
>     GMainLoop *loop;
>
>     GstRTSPServer *server;
>
>     GstRTSPMountPoints *mounts;
>
>     GstRTSPMediaFactory *factory;
>
>
>
>     gst_init( &argc, &argv );
>
>
>
>     loop = g_main_loop_new( NULL, FALSE );
>
>
>
>     InitPipeline();
>
>
>
>     server = gst_rtsp_server_new();
>
>     mounts = gst_rtsp_server_get_mount_points( server );
>
>     factory = gst_rtsp_media_factory_new();
>
>
>
>     // gst_rtsp_media_factory_set_launch( factory, "( appsrc name=mysrc !
> videoconvert ! x264enc ! rtph264pay name=pay0 pt=96 )" );
>
>     gst_rtsp_media_factory_set_launch( factory, "( appsrc name=mysrc !
> rtph264pay name=pay0 pt=96 )" );
>
>
>
>     g_signal_connect( factory, "media-configure", (GCallback)
> media_configure, NULL );
>
>     gst_rtsp_mount_points_add_factory( mounts, "/test", factory );
>
>     g_object_unref( mounts );
>
>
>
>     gst_rtsp_server_attach( server, NULL );
>
>
>
>     g_print( "stream ready at rtsp://127.0.0.1:8554/test\n" );
>
>     g_main_loop_run( loop );
>
>
>
>     return 0;
>
> }
>
>
>
>
>
> Is it a good idea to convert before the appsink?
>
>
>
> Thanks for your time
>
>
>