gstreamer/tests/icles/dccp/mp3Speex/DCCPClientPlaySpeexMP3.c
2012-11-04 00:09:59 +00:00

120 lines
3.5 KiB
C

/* GStreamer
* Copyright (C) <2007> Leandro Melo de Sales <leandroal@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#include <stdio.h>
#include <stdlib.h>
#include <gst/gst.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 *err;
gst_message_parse_error (msg, &err, &debug);
g_free (debug);
g_print ("Error: %s\n", err->message);
g_error_free (err);
g_main_loop_quit (loop);
break;
}
default:
break;
}
return TRUE;
}
int
main (int argc, char *argv[])
{
GMainLoop *loop;
GstBus *bus;
GstElement *pipeline, *alsasink, *rtpspeexdepay, *speexdec, *dccpclientsrc;
GstCaps *caps;
/* initialize GStreamer */
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
/* check input arguments */
if (argc != 3) {
g_print ("%s\n", "see usage: serverHost serverPort");
return -1;
}
/* create elements */
pipeline = gst_pipeline_new ("audio-sender");
alsasink = gst_element_factory_make ("alsasink", "alsa-sink");
rtpspeexdepay = gst_element_factory_make ("rtpspeexdepay", "rtpspeexdepay");
speexdec = gst_element_factory_make ("speexdec", "speexdec");
dccpclientsrc = gst_element_factory_make ("dccpclientsrc", "client-source");
if (!pipeline || !alsasink || !rtpspeexdepay || !speexdec || !dccpclientsrc) {
g_print ("One element could not be created\n");
return -1;
}
caps =
gst_caps_from_string
("application/x-rtp, media=(string)audio, payload=(int)110, clock-rate=(int)44100, encoding-name=(string)SPEEX, ssrc=(guint)152981653, clock-base=(guint)1553719649, seqnum-base=(guint)3680, encoding-params=(string)1");
g_object_set (G_OBJECT (dccpclientsrc), "caps", caps, NULL);
gst_object_unref (caps);
g_object_set (G_OBJECT (dccpclientsrc), "host", argv[1], NULL);
g_object_set (G_OBJECT (dccpclientsrc), "port", atoi (argv[2]), NULL);
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_add_watch (bus, bus_call, loop);
gst_object_unref (bus);
/* put all elements in a bin */
gst_bin_add_many (GST_BIN (pipeline), dccpclientsrc, rtpspeexdepay, speexdec,
alsasink, NULL);
gst_element_link_many (dccpclientsrc, rtpspeexdepay, speexdec, alsasink,
NULL);
/* Now set to playing and iterate. */
g_print ("Setting to PLAYING\n");
gst_element_set_state (pipeline, GST_STATE_PLAYING);
g_print ("Running\n");
g_main_loop_run (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;
}