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 <rtsp://127.0.0.1:8554/test\n>\n <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