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/* GStreamer
* Copyright (C) 2009 Wim Taymans <wim.taymans@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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <string.h>
#include <math.h>
#include <gst/gst.h>
/*
* A simple RTP server
* sends the output of alsasrc as alaw encoded RTP on port 5002, RTCP is sent on
* port 5003. The destination is 127.0.0.1.
* the receiver RTCP reports are received on port 5007
*
* .-------. .-------. .-------. .----------. .-------.
* |alsasrc| |alawenc| |pcmapay| | rtpbin | |udpsink| RTP
* | src->sink src->sink src->send_rtp send_rtp->sink | port=5002
* '-------' '-------' '-------' | | '-------'
* | |
* | | .-------.
* | | |udpsink| RTCP
* | send_rtcp->sink | port=5003
* .-------. | | '-------' sync=false
* RTCP |udpsrc | | | async=false
* port=5007 | src->recv_rtcp |
* '-------' '----------'
*/
/* change this to send the RTP data and RTCP to another host */
#define DEST_HOST "127.0.0.1"
/* #define AUDIO_SRC "alsasrc" */
#define AUDIO_SRC "audiotestsrc"
/* the encoder and payloader elements */
#define AUDIO_ENC "alawenc"
#define AUDIO_PAY "rtppcmapay"
/* build a pipeline equivalent to:
*
* gst-launch -v gstrtpbin name=rtpbin \
* $AUDIO_SRC ! audioconvert ! audioresample ! $AUDIO_ENC ! $AUDIO_PAY ! rtpbin.send_rtp_sink_0 \
* rtpbin.send_rtp_src_0 ! udpsink port=5002 host=$DEST \
* rtpbin.send_rtcp_src_0 ! udpsink port=5003 host=$DEST sync=false async=false \
* udpsrc port=5007 ! rtpbin.recv_rtcp_sink_0
*/
int
main (int argc, char *argv[])
{
GstElement *audiosrc, *audioconv, *audiores, *audioenc, *audiopay;
GstElement *rtpbin, *rtpsink, *rtcpsink, *rtcpsrc;
GstElement *pipeline;
GMainLoop *loop;
gboolean res;
GstPadLinkReturn lres;
GstPad *srcpad, *sinkpad;
/* always init first */
gst_init (&argc, &argv);
/* the pipeline to hold everything */
pipeline = gst_pipeline_new (NULL);
g_assert (pipeline);
/* the audio capture and format conversion */
audiosrc = gst_element_factory_make (AUDIO_SRC, "audiosrc");
g_assert (audiosrc);
audioconv = gst_element_factory_make ("audioconvert", "audioconv");
g_assert (audioconv);
audiores = gst_element_factory_make ("audioresample", "audiores");
g_assert (audiores);
/* the encoding and payloading */
audioenc = gst_element_factory_make (AUDIO_ENC, "audioenc");
g_assert (audioenc);
audiopay = gst_element_factory_make (AUDIO_PAY, "audiopay");
g_assert (audiopay);
/* add capture and payloading to the pipeline and link */
gst_bin_add_many (GST_BIN (pipeline), audiosrc, audioconv, audiores,
audioenc, audiopay, NULL);
res = gst_element_link_many (audiosrc, audioconv, audiores, audioenc,
audiopay, NULL);
g_assert (res == TRUE);
/* the rtpbin element */
rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
g_assert (rtpbin);
gst_bin_add (GST_BIN (pipeline), rtpbin);
/* the udp sinks and source we will use for RTP and RTCP */
rtpsink = gst_element_factory_make ("udpsink", "rtpsink");
g_assert (rtpsink);
g_object_set (rtpsink, "port", 5002, "host", DEST_HOST, NULL);
rtcpsink = gst_element_factory_make ("udpsink", "rtcpsink");
g_assert (rtcpsink);
g_object_set (rtcpsink, "port", 5003, "host", DEST_HOST, NULL);
/* no need for synchronisation or preroll on the RTCP sink */
g_object_set (rtcpsink, "async", FALSE, "sync", FALSE, NULL);
rtcpsrc = gst_element_factory_make ("udpsrc", "rtcpsrc");
g_assert (rtcpsrc);
g_object_set (rtcpsrc, "port", 5007, NULL);
gst_bin_add_many (GST_BIN (pipeline), rtpsink, rtcpsink, rtcpsrc, NULL);
/* now link all to the rtpbin, start by getting an RTP sinkpad for session 0 */
sinkpad = gst_element_get_request_pad (rtpbin, "send_rtp_sink_0");
srcpad = gst_element_get_static_pad (audiopay, "src");
lres = gst_pad_link (srcpad, sinkpad);
g_assert (lres == GST_PAD_LINK_OK);
gst_object_unref (srcpad);
/* get the RTP srcpad that was created when we requested the sinkpad above and
* link it to the rtpsink sinkpad*/
srcpad = gst_element_get_static_pad (rtpbin, "send_rtp_src_0");
sinkpad = gst_element_get_static_pad (rtpsink, "sink");
lres = gst_pad_link (srcpad, sinkpad);
g_assert (lres == GST_PAD_LINK_OK);
gst_object_unref (srcpad);
gst_object_unref (sinkpad);
/* get an RTCP srcpad for sending RTCP to the receiver */
srcpad = gst_element_get_request_pad (rtpbin, "send_rtcp_src_0");
sinkpad = gst_element_get_static_pad (rtcpsink, "sink");
lres = gst_pad_link (srcpad, sinkpad);
g_assert (lres == GST_PAD_LINK_OK);
gst_object_unref (sinkpad);
/* we also want to receive RTCP, request an RTCP sinkpad for session 0 and
* link it to the srcpad of the udpsrc for RTCP */
srcpad = gst_element_get_static_pad (rtcpsrc, "src");
sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtcp_sink_0");
lres = gst_pad_link (srcpad, sinkpad);
g_assert (lres == GST_PAD_LINK_OK);
gst_object_unref (srcpad);
/* set the pipeline to playing */
g_print ("starting sender pipeline\n");
gst_element_set_state (pipeline, GST_STATE_PLAYING);
/* we need to run a GLib main loop to get the messages */
loop = g_main_loop_new (NULL, FALSE);
g_main_loop_run (loop);
g_print ("stopping sender pipeline\n");
gst_element_set_state (pipeline, GST_STATE_NULL);
return 0;
}
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