RTSP pipeline implemented via C code not working? - c

Question

My Scenario is as follows :-
I have set up a RTSP server at IP 192.168.1.24 at port 554.I use the following gst-launch command on client side to receive packets and everything works fine.
gst-launch rtspsrc location = rtsp://admin:admin123#192.168.1.24:554/axis-media/media.amp ! fakesink
But when I implement the same thing via C code it gives me error.My C code is as follows:-
#include <gst.h>
#include <glib.h>
static gboolean bus-call (GstBus *bus, GstMessage *msg, gpointer data)
{
GMainLoop *loop = (GMainLoop *) data;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_EOS:
g_print ("End of stream\n");
g_main_loop_quit (loop);
break;
case GST_MESSAGE_ERROR: {
gchar *debug;
GError *error;
gst_message_parse_error (msg, &error, &debug);
g_free (debug);
g_printerr ("Error: %s\n", error->message);
g_error_free (error);
g_main_loop_quit (loop);
break;
}
default:
break;
}
return true;
}
int main (int argc, char *argv[])
{
GMainLoop *loop;
GstElement *pipeline, *source, *sink;
GstBus *bus;
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
if (argc != 2) {
return -1;
}
pipeline = gst_pipeline_new ("network-player");
source = gst_element_factory_make ("rtspsrc","file-source");
sink = gst_element_factory_make ("fakesink","fake");
if (!pipeline || !source || !sink) {
g_printerr ("One element could not be created. Exiting.\n");
return -1;
}
g_object_set (G_OBJECT (source), "location", argv[1], NULL);
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_add_watch (bus, bus_call, loop);
gst_object_unref (bus);
gst_bin_add_many (GST_BIN (pipeline),source, sink, NULL);
gst_element_link_many (source, sink, NULL);
/* Set the pipeline to "playing" state*/
gst_element_set_state (pipeline, GST_STATE_PLAYING);
/* Iterate */
g_print ("Running...\n");
g_main_loop_run (loop);
/* Out of the main loop, clean up nicely */
g_print ("Returned, stopping playback\n");
gst_element_set_state (pipeline, GST_STATE_NULL);
g_print ("Deleting pipeline\n");
gst_object_unref (GST_OBJECT (pipeline));
return 0;
}
I am able to compile the code without any error.
But when I run the binary generated with the following format:-
user#user:~ ./helloworld rtsp://admin:admin123#192.168.1.24:554/axis-media/media.amp
I get the following error:-
Now playing: rtsp://root:nlss123#192.168.1.24:554/axis-media/media.amp
Running...
**Error: Internal data flow error**.
Returned, stopping playback
Deleting pipeline
Can anyone suggest we there is Internal Data flow error ?

Answer 1

i also had the same problem.
You should link source to to sink with "pad-added" signal.
In brief:
typedef struct myDataTag {
GstElement *pipeline;
GstElement *rtspsrc;
GstElement *depayloader;
GstElement *decoder;
*sink;
} myData_t;
myData_t appData;
appData->pipeline = gst_pipeline_new ("videoclient");
appData->rtspsrc = gst_element_factory_make ("rtspsrc", "rtspsrc");
g_object_set (G_OBJECT (appData->rtspsrc), "location", "rtsp://192.168.1.10:554/myStreamPath", NULL);
appData->depayloader = gst_element_factory_make ("rtph264depay","depayloader");
appData->decoder = gst_element_factory_make ("h264dec", "decoder");
appData->sink = gst_element_factory_make ("autovideosink", "sink");
//then add all elements together
gst_bin_add_many (GST_BIN (appData->pipeline), appData->rtspsrc, appData->depayloader, appData->decoder, appData->sink, NULL);
//link everythink after source
gst_element_link_many (appData->depayloader, appData->decoder, appData->sink, NULL);
/*
* Connect to the pad-added signal for the rtpbin. This allows us to link
* the dynamic RTP source pad to the depayloader when it is created.
*/
g_signal_connect (appData->rtspsrc, "pad-added", G_CALLBACK (pad_added_handler), &appData);
/* Set the pipeline to "playing" state*/
gst_element_set_state (appData->pipeline, GST_STATE_PLAYING);
/* pad added handler */
static void pad_added_handler (GstElement *src, GstPad *new_pad, myData_t *pThis) {
GstPad *sink_pad = gst_element_get_static_pad (pThis->depayloader, "sink");
GstPadLinkReturn ret;
GstCaps *new_pad_caps = NULL;
GstStructure *new_pad_struct = NULL;
const gchar *new_pad_type = NULL;
g_print ("Received new pad '%s' from '%s':\n", GST_PAD_NAME (new_pad), GST_ELEMENT_NAME (src));
/* Check the new pad's name */
if (!g_str_has_prefix (GST_PAD_NAME (new_pad), "recv_rtp_src_")) {
g_print (" It is not the right pad. Need recv_rtp_src_. Ignoring.\n");
goto exit;
}
/* If our converter is already linked, we have nothing to do here */
if (gst_pad_is_linked (sink_pad)) {
g_print (" Sink pad from %s already linked. Ignoring.\n", GST_ELEMENT_NAME (src));
goto exit;
}
/* Check the new pad's type */
new_pad_caps = gst_pad_get_caps (new_pad);
new_pad_struct = gst_caps_get_structure (new_pad_caps, 0);
new_pad_type = gst_structure_get_name (new_pad_struct);
/* Attempt the link */
ret = gst_pad_link (new_pad, sink_pad);
if (GST_PAD_LINK_FAILED (ret)) {
g_print (" Type is '%s' but link failed.\n", new_pad_type);
} else {
g_print (" Link succeeded (type '%s').\n", new_pad_type);
}
exit:
/* Unreference the new pad's caps, if we got them */
if (new_pad_caps != NULL)
gst_caps_unref (new_pad_caps);
/* Unreference the sink pad */
gst_object_unref (sink_pad);
}
Hope that this will help someone..:)

Answer 2

you can get verbose error logs by running the apps by --gst-debug=*rtsp*:5 e.g.
./yourApplication --gst-debug=*rtsp*:5