[gst-devel] Randomness-> appsrc -> Speakers

[RadicalMcGraw]

Hello, I'm trying to integrate gstreamer into my application with
appsrc/appsink, so this my first step in that direction. The goal of this
code is to just input random noise into appsource to be played on the
speakers.  The code is the following, followed by the runtime errors.

--------------------------------------------------------------------------
#include <stdio.h>
#include <stdlib.h>
#include <time.h>
#include <string.h>
#include <gst/gst.h>
#include <gst/app/gstappsrc.h>
#include <gst/app/gstappbuffer.h>
#include <pthread.h>
/*gcc -Wall $(pkg-config --cflags --libs gstreamer-0.10) -pthread
-lgstapp-0.10 src.c -o src*/
const gchar *audio_caps ="audio/x-raw-int,rate=11000,channels=1";
typedef struct
{
  GMainLoop *loop;
  GstElement *pipeline;
  GstElement *source;
  guint source_id;
  guint num_frame;
} AppData;

int main (int argc, char *argv[])
{
	AppData *app = NULL;
	gchar *string = NULL;
	GstBus *bus = NULL;
	GstElement *appsrc = NULL;
	srand ( time(NULL) );
	gst_init (&argc, &argv);
	app = g_new0 (AppData, 1);
	app->loop = g_main_loop_new (NULL, FALSE);
	string = g_strdup_printf ("appsrc -v name=source caps=\"%s\" ! audioconvert
! audioresample ! osssink",audio_caps);
	app->pipeline = gst_parse_launch (string, NULL);
	g_free (string);

	if (app->pipeline == NULL) {
		g_print ("Bad pipeline\n");
		return -1;
	}
	appsrc = gst_bin_get_by_name (GST_BIN (app->pipeline), "source");
	app->source = appsrc;

	/* add watch for messages */
	bus = gst_element_get_bus (app->pipeline);
	gst_object_unref (bus);

	GstBuffer *buffer;
	buffer=gst_buffer_new_and_alloc(5000);
	int i;
	for (i=0; i<5000;i++)
		buffer->data[i]=rand()%10000;
	gst_app_src_push_buffer(GST_APP_SRC (app->source),buffer);
	gst_app_src_end_of_stream (GST_APP_SRC (app->source));


	/* go to playing and wait in a mainloop */
	gst_element_set_state (app->pipeline, GST_STATE_PLAYING);
	g_main_loop_run(app->loop);
	gst_element_set_state (app->pipeline, GST_STATE_NULL);

	/* Cleaning up */
	gst_object_unref (app->source);
	gst_object_unref (app->pipeline);
	g_main_loop_unref (app->loop);
	g_free (app);

	return 0;
} 

-------------------------------------------------------------------------------------------
wildcat at ubuntu:~/Desktop$ gcc -Wall -pthread -lgstapp-0.10 $(pkg-config
--cflags --libs  gstreamer-0.10) srcdemo.c -o srcdemo;./srcdemo

** (srcdemo:2845): CRITICAL **: gst_app_src_push_buffer_full: assertion
`GST_IS_APP_SRC (appsrc)' failed

** (srcdemo:2845): CRITICAL **: gst_app_src_end_of_stream: assertion
`GST_IS_APP_SRC (appsrc)' failed

-------------------------------------------------------------------------------------------
These messages only come up when I use the verbose option, and it still
continues to run with no sound.
Are there any glaring problems? (Also, are my audio caps sufficient?)
As far as the buffer_full issues, the error remains if I create a buffer of
size 50 or 50,000.

Also, if my code needs ripped apart, my application needs to run alongside
the gstreamer (in its own thread) and needs to input/output data at its own
pace; so I can't use gstreamer's callback functions to input data when
gstreamer needs it, but instead would prefer to have appsink/source block
itself while waiting for data.

Thanks so much, and any general pointers about my program/goals will be
appreciated too.


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