| /* 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, *mad, *audioconvert, *dccpclientsrc; |
| |
| /* 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"); |
| dccpclientsrc = gst_element_factory_make ("dccpclientsrc", "client-source"); |
| mad = gst_element_factory_make ("mad", "mad"); |
| audioconvert = gst_element_factory_make ("audioconvert", "audioconvert"); |
| alsasink = gst_element_factory_make ("alsasink", "alsa-sink"); |
| |
| if (!pipeline || !alsasink || !mad || !audioconvert || !dccpclientsrc) { |
| g_print ("One element could not be created\n"); |
| return -1; |
| } |
| |
| 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, mad, audioconvert, |
| alsasink, NULL); |
| |
| gst_element_link_many (dccpclientsrc, mad, audioconvert, 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; |
| } |