RTSP appsrc sample application fails

[Anuj Pahuja]

I've faced the same issue before. Any suggestions would be appreciated.

Thanks,
Anuj

On Nov 25, 2016 9:08 PM, "Tejas Ravichandran" <venkatakrishnan164 at gmail.com>
wrote:

> Hi sorry for reposting .
> I am using the rtsp test-appsrc application and modified it to work for
> I420 format images with rtpvrawpay instead of rtph264pay .
>
> however for certain resolutions the pipeline fails
> like for e.g for 2100 x 576 it fails whereas works for 2104 x 576 but
> fails for 2106 x 576 and so on... i am not able to understand the pattern .
> This is the appsrc modified code for rtsp:
>
> #include <gst/gst.h>
>
> #include <gst/rtsp-server/rtsp-server.h>
>
> typedef struct
> {
>   gboolean white;
>   GstClockTime timestamp;
> } MyContext;
>
> /* called when we need to give data to appsrc */
> static void
> need_data (GstElement * appsrc, guint unused, MyContext * ctx)
> {
>   GstBuffer *buffer;
>   guint size;
>   GstFlowReturn ret;
>
>   unsigned int sizeInt = (2100*576*3) >> 1;
>
>   size = sizeInt;//2100 * 576 * 1.5;
>
>   buffer = gst_buffer_new_allocate (NULL, size, NULL);
>
>   /* this makes the image black/white */
>   gst_buffer_memset (buffer, 0, ctx->white ? 0xff : 0x0, size);
>
>   ctx->white = !ctx->white;
>
>   /* increment the timestamp every 1/2 second */
>   GST_BUFFER_PTS (buffer) = ctx->timestamp;
>   GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale_int (1, GST_SECOND,
> 2);
>   ctx->timestamp += GST_BUFFER_DURATION (buffer);
>
>   g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret);
> }
>
> /* called when a new media pipeline is constructed. We can query the
>  * pipeline and configure our appsrc */
> static void
> media_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media,
>     gpointer user_data)
> {
>   GstElement *element, *appsrc;
>   MyContext *ctx;
>
>   /* get the element used for providing the streams of the media */
>   element = gst_rtsp_media_get_element (media);
>
>   /* get our appsrc, we named it 'mysrc' with the name property */
>   appsrc = gst_bin_get_by_name_recurse_up (GST_BIN (element), "mysrc");
>
>   /* this instructs appsrc that we will be dealing with timed buffer */
>   gst_util_set_object_arg (G_OBJECT (appsrc), "format", "time");
>   /* configure the caps of the video */
>   g_object_set (G_OBJECT (appsrc), "caps",
>       gst_caps_new_simple ("video/x-raw",
>           "format", G_TYPE_STRING, "I420",
>           "width", G_TYPE_INT, 2100,
>           "height", G_TYPE_INT, 576,
>           "framerate", GST_TYPE_FRACTION, 0, 1, NULL), NULL);
>
>   ctx = g_new0 (MyContext, 1);
>   ctx->white = FALSE;
>   ctx->timestamp = 0;
>   /* make sure ther datais freed when the media is gone */
>   g_object_set_data_full (G_OBJECT (media), "my-extra-data", ctx,
>       (GDestroyNotify) g_free);
>
>   /* install the callback that will be called when a buffer is needed */
>   g_signal_connect (appsrc, "need-data", (GCallback) need_data, ctx);
>   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);
>
>   /* create a server instance */
>   server = gst_rtsp_server_new ();
> gst_rtsp_server_set_address(server,"10.0.0.2");
>   /* 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 = gst_rtsp_media_factory_new ();
>   gst_rtsp_media_factory_set_launch (factory,
>       "( appsrc name=mysrc ! rtpvrawpay  name=pay0 pt=96 )");
>
>   /* notify when our media is ready, This is called whenever someone asks
> for
>    * the media and a new pipeline with our appsrc is created */
>   g_signal_connect (factory, "media-configure", (GCallback)
> media_configure,
>       NULL);
>
>   /* attach the test factory to the /test url */
>   gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
>
>   /* don't need the ref to the mounts 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
> <http://127.0.0.1:8554/test%5Cn>");
>   g_main_loop_run (loop);
>
>   return 0;
> }
> This pipeline fails with a segmentation fault when the client side is run
> as follows:
> gst-launch-1.0 rtspsrc latency=50 location=rtsp://10.0.0.2:8555/testing !
> rtpvrawdepay ! videoconvert ! ximagesink -v
> with "could not read from resource" message on the client side.
>
> However the same code with 2104 x 576 works fine .
>
> NOTE:
> Also when i change I420 to NV12 and add a videoconvert before rtpvrawpay
> every resolution works well i.e by making the following change
>
>
> "( appsrc name=mysrc ! videconvert ! rtpvrawpay  name=pay0 pt=96 )");
>
> gst_caps_new_simple ("video/x-raw",
>           "format", G_TYPE_STRING, "NV12",
>           "width", G_TYPE_INT, 2100,
>           "height", G_TYPE_INT, 576,
>
> Any help would be greatly appreciated.
>
> _______________________________________________
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> gstreamer-devel at lists.freedesktop.org
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
>
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