<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"> <head> <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> <meta name="Generator" content="Microsoft Word 15 (filtered medium)"> <style></style> </head> <body lang="EN-CA" link="blue" vlink="purple" style="word-wrap:break-word"> <div class="WordSection1"> Same behaviour with both main and constrained-baseline<o:p></o:p> <o:p> </o:p> But since the rtsp server does not emit the “need-data” signal more than once, I would suppose the issue is coming from here, and not the encoding<o:p></o:p> (but I don’t know at all how all this work im new, so it may be completely wrong)<o:p></o:p> <o:p> </o:p> <div style="border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0cm 0cm 0cm"> From: Nicolas Dufresne <nicolas@ndufresne.ca> Sent: Thursday, September 21, 2023 8:33 PM To: Discussion of the development of and with GStreamer <gstreamer-devel@lists.freedesktop.org> Cc: LEVY David <David.Levy@ino.ca>; Michael Gruner <michael.gruner@ridgerun.com> Subject: Re: Issue with rtsp server source h264 encoded<o:p></o:p> </div> <o:p> </o:p> <div> 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).<o:p></o:p> <div> <o:p> </o:p> </div> <div> Hopefully this will improve your situation.<o:p></o:p> </div> </div> <o:p> </o:p> <div> <div> Le jeu. 21 sept. 2023, 16 h 30, LEVY David via gstreamer-devel <<a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a>> a écrit :<o:p></o:p> </div> <blockquote style="border:none;border-left:solid #CCCCCC 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm"> <div> <div> Same behaviour.<o:p></o:p> <o:p></o:p> I just noted that “need-data” is only called once with ffplay, and twice with gst-launch (probably once udp and once tcp)<o:p></o:p> <o:p></o:p> <div> <div style="border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0cm 0cm 0cm"> From: Michael Gruner <<a href="mailto:michael.gruner@ridgerun.com" target="_blank">michael.gruner@ridgerun.com</a>> Sent: Thursday, September 21, 2023 2:45 PM To: Discussion of the development of and with GStreamer <<a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a>> Cc: LEVY David <<a href="mailto:David.Levy@ino.ca" target="_blank">David.Levy@ino.ca</a>> Subject: Re: Issue with rtsp server source h264 encoded<o:p></o:p> </div> </div> <o:p></o:p> Hi<o:p></o:p> <div> <o:p></o:p> </div> <div> Try sending periodic I frames, "x264enc key-int-max=10" or, programmatically:<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> <blockquote style="margin-top:5.0pt;margin-bottom:5.0pt"> <div> g_object_set( lEncoder, "key-int-max", 10, NULL ); // 4 = zerolatency<o:p></o:p> </div> </blockquote> <div> <div> <o:p></o:p> </div> </div> <div> <o:p></o:p> </div> <div> 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.<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> Michael.<o:p></o:p> </div> <div> <o:p> </o:p> <blockquote style="margin-top:5.0pt;margin-bottom:5.0pt"> <div> On 21 Sep 2023, at 09:59, LEVY David via gstreamer-devel <<a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a>> wrote:<o:p></o:p> </div> <o:p></o:p> <div> <div> Hello, i'm new to gstreamer / rtsp so, sorry if it's a basic mistake.<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> Im trying to set up a pipeline that end with an appsink that push data into a rtsp server starting with an appsrc.<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> It would be:<o:p></o:p> </div> <div> videotestsrc -> x264enc -> appsink (push data to src as soon as received)<o:p></o:p> </div> <div> and<o:p></o:p> </div> <div> appsrc -> rtph264pay<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> I can’t find any example similar, but I found this one: <a href="https://github.com/GStreamer/gst-rtsp-server/blob/master/examples/test-appsrc.c" target="_blank">https://github.com/GStreamer/gst-rtsp-server/blob/master/examples/test-appsrc.c</a><o:p></o:p> </div> <div> <o:p></o:p> </div> <div> Which does: appsrc-> videoconvert -> x264enc -> rtph264pay (and uses “need-data”) <o:p></o:p> </div> <div> <o:p></o:p> </div> <div> I have successfully tweaked it to videotestsrc -> appsink<o:p></o:p> </div> <div> appsrc-> videoconvert -> x264enc -> rtph264pay (and uses “need-data”)<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> My next step would be<o:p></o:p> </div> <div> Videotestsrc -> x264enc -> appsink <o:p></o:p> </div> <div> appsrc-> rtph264pay (and uses “need-data”)<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> but when I try to read the stream from either <o:p></o:p> </div> <div> 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<o:p></o:p> </div> <div> or ffplay rtsp://127.0.0.1:8554/test<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> I don’t have any playback: Setting pipeline to PAUSED ...<o:p></o:p> </div> <div> Pipeline is live and does not need PREROLL ...<o:p></o:p> </div> <div> Got context from element 'autovideosink0': gst.gl.GLDisplay=context, gst.gl.GLDisplay=(GstGLDisplay)"\(GstGLDisplayWayland\)\ gldisplaywayland0";<o:p></o:p> </div> <div> Progress: (open) Opening Stream<o:p></o:p> </div> <div> Pipeline is PREROLLED ...<o:p></o:p> </div> <div> Prerolled, waiting for progress to finish...<o:p></o:p> </div> <div> Progress: (connect) Connecting to rtsp://127.0.0.1:8554/test<o:p></o:p> </div> <div> Progress: (open) Retrieving server options<o:p></o:p> </div> <div> Progress: (open) Retrieving media info<o:p></o:p> </div> <div> Progress: (request) SETUP stream 0<o:p></o:p> </div> <div> Progress: (open) Opened Stream<o:p></o:p> </div> <div> Setting pipeline to PLAYING ...<o:p></o:p> </div> <div> New clock: GstSystemClock<o:p></o:p> </div> <div> Progress: (request) Sending PLAY request<o:p></o:p> </div> <div> Redistribute latency...<o:p></o:p> </div> <div> Progress: (request) Sending PLAY request<o:p></o:p> </div> <div> Redistribute latency...<o:p></o:p> </div> <div> Progress: (request) Sent PLAY request<o:p></o:p> </div> <div> Redistribute latency...<o:p></o:p> </div> <div> Redistribute latency...<o:p></o:p> </div> <div> 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.<o:p></o:p> </div> <div> WARNING: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Could not read from resource.<o:p></o:p> </div> <div> Additional debug info:<o:p></o:p> </div> <div> ../gst/rtsp/gstrtspsrc.c(5964): gst_rtspsrc_reconnect (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:<o:p></o:p> </div> <div> Could not receive any UDP packets for 5.0000 seconds, maybe your firewall is blocking it. Retrying using a tcp connection.<o:p></o:p> </div> <div> Redistribute latency...<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> And it is stuck here I’m using the code below: <o:p></o:p> </div> <div> #include <gst/gst.h><o:p></o:p> </div> <div> <o:p></o:p> </div> <div> #include <gst/app/gstappsink.h><o:p></o:p> </div> <div> #include <gst/app/gstappsrc.h><o:p></o:p> </div> <div> #include <gst/rtsp-server/rtsp-server.h><o:p></o:p> </div> <div> <o:p></o:p> </div> <div> GstElement *gSink = nullptr;<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> static void InitPipeline()<o:p></o:p> </div> <div> {<o:p></o:p> </div> <div> GstElement *lPipeline = gst_pipeline_new( "x264-pipeline" );<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> GstElement *lSource = gst_element_factory_make( "videotestsrc", "source" );<o:p></o:p> </div> <div> GstElement *lEncoder = gst_element_factory_make( "x264enc", "x264-encoder" );<o:p></o:p> </div> <div> GstElement *lSink = gst_element_factory_make( "appsink", "appsink" );<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> gst_bin_add_many( GST_BIN( lPipeline ), lSource, lEncoder, lSink, NULL );<o:p></o:p> </div> <div> // gst_bin_add_many( GST_BIN( lPipeline ), lSource, lSink, NULL );<o:p></o:p> </div> <div> g_object_set( lEncoder, "tune", 4, NULL ); // 4 = zerolatency<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> GstCaps *lCaps =<o:p></o:p> </div> <div> gst_caps_new_simple( "video/x-raw", "width", G_TYPE_INT, 384, "height", G_TYPE_INT, 288, "framerate", GST_TYPE_FRACTION, 10, 1, NULL );<o:p></o:p> </div> <div> if( !gst_element_link_filtered( lSource, lEncoder, lCaps ) )<o:p></o:p> </div> <div> // if( !gst_element_link_filtered( lSource, lSink, lCaps ) )<o:p></o:p> </div> <div> {<o:p></o:p> </div> <div> gst_caps_unref( lCaps );<o:p></o:p> </div> <div> throw;<o:p></o:p> </div> <div> }<o:p></o:p> </div> <div> gst_caps_unref( lCaps );<o:p></o:p> </div> <div> gst_element_link( lEncoder, lSink );<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> GstStateChangeReturn ret = gst_element_set_state( lPipeline, GST_STATE_PLAYING );<o:p></o:p> </div> <div> if( ret == GST_STATE_CHANGE_FAILURE )<o:p></o:p> </div> <div> {<o:p></o:p> </div> <div> throw;<o:p></o:p> </div> <div> }<o:p></o:p> </div> <div> gSink = lSink;<o:p></o:p> </div> <div> }<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> static void need_data( GstElement *appsrc, guint unused, void *aUnused )<o:p></o:p> </div> <div> {<o:p></o:p> </div> <div> GstSample *lSample = gst_app_sink_pull_sample( GST_APP_SINK_CAST( gSink ) );<o:p></o:p> </div> <div> GstFlowReturn lRet = gst_app_src_push_sample( GST_APP_SRC( appsrc ), lSample );<o:p></o:p> </div> <div> gst_sample_unref( lSample );<o:p></o:p> </div> <div> return;<o:p></o:p> </div> <div> }<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> static void media_configure( GstRTSPMediaFactory *factory, GstRTSPMedia *media, gpointer user_data )<o:p></o:p> </div> <div> {<o:p></o:p> </div> <div> GstElement *element, *appsrc;<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> element = gst_rtsp_media_get_element( media );<o:p></o:p> </div> <div> appsrc = gst_bin_get_by_name_recurse_up( GST_BIN( element ), "mysrc" );<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> gst_util_set_object_arg( G_OBJECT( appsrc ), "format", "time" );<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> if( GstPad *pad = gst_element_get_static_pad( gSink, "sink" ) )<o:p></o:p> </div> <div> {<o:p></o:p> </div> <div> g_print( "Got pad\n" );<o:p></o:p> </div> <div> GstCaps *caps = gst_pad_get_current_caps( pad );<o:p></o:p> </div> <div> if( caps )<o:p></o:p> </div> <div> {<o:p></o:p> </div> <div> auto *caps_str = gst_caps_to_string( caps );<o:p></o:p> </div> <div> g_print( "Got caps %s\n", caps_str );<o:p></o:p> </div> <div> g_object_set( G_OBJECT( appsrc ), "caps", caps, NULL );<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> gst_caps_unref( caps );<o:p></o:p> </div> <div> }<o:p></o:p> </div> <div> }<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> g_signal_connect( appsrc, "need-data", (GCallback)need_data, NULL );<o:p></o:p> </div> <div> gst_object_unref( appsrc );<o:p></o:p> </div> <div> gst_object_unref( element );<o:p></o:p> </div> <div> }<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> int main( int argc, char *argv[] )<o:p></o:p> </div> <div> {<o:p></o:p> </div> <div> GMainLoop *loop;<o:p></o:p> </div> <div> GstRTSPServer *server;<o:p></o:p> </div> <div> GstRTSPMountPoints *mounts;<o:p></o:p> </div> <div> GstRTSPMediaFactory *factory;<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> gst_init( &argc, &argv );<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> loop = g_main_loop_new( NULL, FALSE );<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> InitPipeline();<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> server = gst_rtsp_server_new();<o:p></o:p> </div> <div> mounts = gst_rtsp_server_get_mount_points( server );<o:p></o:p> </div> <div> factory = gst_rtsp_media_factory_new();<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> // gst_rtsp_media_factory_set_launch( factory, "( appsrc name=mysrc ! videoconvert ! x264enc ! rtph264pay name=pay0 pt=96 )" );<o:p></o:p> </div> <div> gst_rtsp_media_factory_set_launch( factory, "( appsrc name=mysrc ! rtph264pay name=pay0 pt=96 )" );<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> g_signal_connect( factory, "media-configure", (GCallback)media_configure, NULL );<o:p></o:p> </div> <div> gst_rtsp_mount_points_add_factory( mounts, "/test", factory );<o:p></o:p> </div> <div> g_object_unref( mounts );<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> gst_rtsp_server_attach( server, NULL );<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> g_print( "stream ready at rtsp://127.0.0.1:8554/test\n" );<o:p></o:p> </div> <div> g_main_loop_run( loop );<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> return 0;<o:p></o:p> </div> <div> }<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> <o:p></o:p> </div> <div> Is it a good idea to convert before the appsink?<o:p></o:p> </div> <div> <o:p></o:p> </div> <div> Thanks for your time<o:p></o:p> </div> </div> </blockquote> </div> <o:p></o:p> </div> </div> </div> </blockquote> </div> </div> </body> </html>