GNOME Bugzilla – Bug 751720
qmlplayer2 example shows incorrect stride when using videotestsrc through an rtsp server
Last modified: 2018-05-04 12:12:46 UTC
Created attachment 306397 [details] qmlplayer2 output when using videotestsrc through an rtsp server Using gstreamer 1.2.0 git master HEAD, When using an RTSP server, the stride looks fine for: gst-launch-1.0 playbin uri=192.168.232.62:8554/test but incorrect for: ./qmlplayer2 192.168.232.62:8554/test See attached image of qmlplayer2 output. I see the same incorrect stride for my own H264 live source. A handful of non-live H.264 videos seem to work fine. The included http big-buck-bunny example in qmlplayer2 also works fine. For the RTSP server, I am using the gst-rtsp-server/examples/test-netclock.c example with the launch line: gst_rtsp_media_factory_set_launch (factory, "(videotestsrc pattern=19 is-live=true horizontal-speed=1 ! video/x-raw, framerate=(fraction)60/1, height=(int)1000, width=(int)1000 ! x264enc ! rtph264pay name=pay0 pt=96 )"); full server code: #include <gst/gst.h> #include <gst/net/gstnettimeprovider.h> #include <gst/rtsp-server/rtsp-server.h> GstClock *global_clock; #define TEST_TYPE_RTSP_MEDIA_FACTORY (test_rtsp_media_factory_get_type ()) #define TEST_TYPE_RTSP_MEDIA (test_rtsp_media_get_type ()) GType test_rtsp_media_factory_get_type (void); GType test_rtsp_media_get_type (void); static GstRTSPMediaFactory *test_rtsp_media_factory_new (void); static GstElement *create_pipeline (GstRTSPMediaFactory * factory, GstRTSPMedia * media); typedef struct TestRTSPMediaFactoryClass TestRTSPMediaFactoryClass; typedef struct TestRTSPMediaFactory TestRTSPMediaFactory; struct TestRTSPMediaFactoryClass { GstRTSPMediaFactoryClass parent; }; struct TestRTSPMediaFactory { GstRTSPMediaFactory parent; }; typedef struct TestRTSPMediaClass TestRTSPMediaClass; typedef struct TestRTSPMedia TestRTSPMedia; struct TestRTSPMediaClass { GstRTSPMediaClass parent; }; struct TestRTSPMedia { GstRTSPMedia parent; }; GstRTSPMediaFactory * test_rtsp_media_factory_new (void) { GstRTSPMediaFactory *result; result = g_object_new (TEST_TYPE_RTSP_MEDIA_FACTORY, NULL); return result; } G_DEFINE_TYPE (TestRTSPMediaFactory, test_rtsp_media_factory, GST_TYPE_RTSP_MEDIA_FACTORY); static gboolean custom_setup_rtpbin (GstRTSPMedia * media, GstElement * rtpbin); static void test_rtsp_media_factory_class_init (TestRTSPMediaFactoryClass * test_klass) { GstRTSPMediaFactoryClass *mf_klass = (GstRTSPMediaFactoryClass *) (test_klass); mf_klass->create_pipeline = create_pipeline; } static void test_rtsp_media_factory_init (TestRTSPMediaFactory * factory) { } static GstElement * create_pipeline (GstRTSPMediaFactory * factory, GstRTSPMedia * media) { GstElement *pipeline; pipeline = gst_pipeline_new ("media-pipeline"); gst_pipeline_use_clock (GST_PIPELINE (pipeline), global_clock); gst_rtsp_media_take_pipeline (media, GST_PIPELINE_CAST (pipeline)); return pipeline; } G_DEFINE_TYPE (TestRTSPMedia, test_rtsp_media, GST_TYPE_RTSP_MEDIA); static void test_rtsp_media_class_init (TestRTSPMediaClass * test_klass) { GstRTSPMediaClass *klass = (GstRTSPMediaClass *) (test_klass); klass->setup_rtpbin = custom_setup_rtpbin; } static void test_rtsp_media_init (TestRTSPMedia * media) { } static gboolean custom_setup_rtpbin (GstRTSPMedia * media, GstElement * rtpbin) { g_object_set (rtpbin, "ntp-time-source", 3, NULL); return 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); global_clock = gst_system_clock_obtain (); gst_net_time_provider_new (global_clock, "0.0.0.0", 8554); /* 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 = test_rtsp_media_factory_new (); gst_rtsp_media_factory_set_launch (factory, "(videotestsrc pattern=19 is-live=true horizontal-speed=1 ! video/x-raw, framerate=(fraction)60/1, height=(int)1000, width=(int)1000 ! x264enc ! rtph264pay name=pay0 pt=96 )"); // gst_rtsp_media_factory_set_launch (factory, "(christiedesktopcapturenvidia ! h264parse ! rtph264pay name=pay0 )"); //gst_rtsp_media_factory_set_launch (factory, "( filesrc location=/mnt/share/HD-60Hz-H264.mov ! qtdemux ! h264parse ! queue ! rtph264pay name=pay0 )"); // gst_rtsp_media_factory_set_launch (factory, "( filesrc location=/mnt/share/batman.begins.mp4 ! qtdemux ! h264parse ! rtph264pay name=pay0 )"); //gst_rtsp_media_factory_set_launch (factory, "( filesrc location=/mnt/share/test/4K-60Hz-H264.mov ! qtdemux ! h264parse ! tee name=t ! queue ! rtph264pay name=pay0 t. ! queue ! vaapiparse_h264 ! vaapidecode ! vaapisink sync=false )"); //gst_rtsp_media_factory_set_launch (factory, "gst-launch-1.0 udpsrc port=6000 caps=\"application/x-rtp, media=video, clock-rate=90000, encoding-name=H264\" ! rtph264depay ! rtph264pay name=pay0 pt=96"); gst_rtsp_media_factory_set_shared (GST_RTSP_MEDIA_FACTORY (factory), TRUE); gst_rtsp_media_factory_set_media_gtype (GST_RTSP_MEDIA_FACTORY (factory), TEST_TYPE_RTSP_MEDIA); /* make a new address pool */ pool = gst_rtsp_address_pool_new (); gst_rtsp_address_pool_add_range (pool, "224.1.1.1", "224.1.1.10", 5000, 5010, 16); gst_rtsp_media_factory_set_address_pool (factory, pool); /* only allow multicast */ //gst_rtsp_media_factory_set_protocols (factory, // GST_RTSP_LOWER_TRANS_UDP_MCAST); g_object_unref (pool); /* 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; }
sorry, missing rtsp on rtsp://192.168.232.62:8554/test
There's also a new Qt5 QML qmlglsink in gst-plugins-bad now (in git master, will be in version >= 1.5.3 which should be out soon) for what it's worth, you might want to try that instead.
Thank you for reporting this bug. Unfortunately, qt-gstreamer is unmaintained and there is no chance that this problem will get fixed, if it's actually a qt-gstreamer bug. As an alternative, you may want to use qmlglsink (from gst-plugins-good) to integrate GStreamer with QML.