rtsp server appsrc application not working

[tejas95]

I modified the test-appsrc application in gst-rtsp-server to stream out 2100 x 576 I420 image with rtpvrawpay as follows,

#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;

  size = 1000 * 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, 1000,
          "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://<a href="http://127.0.0.1:8554/test\n">127.0.0.1:8554/test\n");
  g_main_loop_run (loop);

  return 0;
}

with client  side as:gst-launch-1.0 rtspsrc latency=50 location=rtsp://10.0.0.2:8554/test ! queue ! rtpjitterbuffer ! rtpvrawdepay !  videoconvert ! ximagesink -v

however when i increase the resolution of the image to 1500 x 576 or 2100 x 576 it fails giving a segmentation fault. and an "unable to read from resource" error on the client side.

stuck in this problem for a lot of time. Any help would be greatly appreciated.

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