RTSP appsrc sample application fails

Tejas Ravichandran venkatakrishnan164 at gmail.com
Fri Nov 25 15:38:46 UTC 2016


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");
  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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