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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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121 lines
3.5 KiB
C
121 lines
3.5 KiB
C
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/* GStreamer
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* Copyright (C) <2007> Leandro Melo de Sales <leandroal@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <gst/gst.h>
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static gboolean
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bus_call (GstBus * bus, GstMessage * msg, gpointer data)
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{
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GMainLoop *loop = (GMainLoop *) data;
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switch (GST_MESSAGE_TYPE (msg)) {
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case GST_MESSAGE_EOS:
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g_print ("End-of-stream\n");
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g_main_loop_quit (loop);
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break;
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case GST_MESSAGE_ERROR:{
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gchar *debug;
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GError *err;
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gst_message_parse_error (msg, &err, &debug);
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g_free (debug);
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g_print ("Error: %s\n", err->message);
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g_error_free (err);
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g_main_loop_quit (loop);
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break;
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}
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default:
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break;
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}
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return TRUE;
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}
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int
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main (int argc, char *argv[])
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{
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GMainLoop *loop;
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GstBus *bus;
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GstElement *pipeline, *alsasink, *rtpspeexdepay, *speexdec, *dccpclientsrc;
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GstCaps *caps;
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/* initialize GStreamer */
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gst_init (&argc, &argv);
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loop = g_main_loop_new (NULL, FALSE);
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/* check input arguments */
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if (argc != 3) {
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g_print ("%s\n", "see usage: serverHost serverPort");
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return -1;
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}
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/* create elements */
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pipeline = gst_pipeline_new ("audio-sender");
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alsasink = gst_element_factory_make ("alsasink", "alsa-sink");
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rtpspeexdepay = gst_element_factory_make ("rtpspeexdepay", "rtpspeexdepay");
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speexdec = gst_element_factory_make ("speexdec", "speexdec");
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dccpclientsrc = gst_element_factory_make ("dccpclientsrc", "client-source");
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if (!pipeline || !alsasink || !rtpspeexdepay || !speexdec || !dccpclientsrc) {
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g_print ("One element could not be created\n");
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return -1;
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}
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caps =
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gst_caps_from_string
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("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");
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g_object_set (G_OBJECT (dccpclientsrc), "caps", caps, NULL);
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gst_object_unref (caps);
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g_object_set (G_OBJECT (dccpclientsrc), "host", argv[1], NULL);
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g_object_set (G_OBJECT (dccpclientsrc), "port", atoi (argv[2]), NULL);
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bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
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gst_bus_add_watch (bus, bus_call, loop);
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gst_object_unref (bus);
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/* put all elements in a bin */
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gst_bin_add_many (GST_BIN (pipeline), dccpclientsrc, rtpspeexdepay, speexdec,
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alsasink, NULL);
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gst_element_link_many (dccpclientsrc, rtpspeexdepay, speexdec, alsasink,
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NULL);
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/* Now set to playing and iterate. */
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g_print ("Setting to PLAYING\n");
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gst_element_set_state (pipeline, GST_STATE_PLAYING);
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g_print ("Running\n");
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g_main_loop_run (loop);
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/* clean up nicely */
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g_print ("Returned, stopping playback\n");
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gst_element_set_state (pipeline, GST_STATE_NULL);
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g_print ("Deleting pipeline\n");
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gst_object_unref (GST_OBJECT (pipeline));
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return 0;
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}
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