RE: Issue with rtsp server source h264 encoded

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]<mailto:[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