mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-08 18:39:54 +00:00
221 lines
7.7 KiB
C
221 lines
7.7 KiB
C
/* 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"
|
|
|
|
/* print the stats of a source */
|
|
static void
|
|
print_source_stats (GObject * source)
|
|
{
|
|
GstStructure *stats;
|
|
gchar *str;
|
|
|
|
/* get the source stats */
|
|
g_object_get (source, "stats", &stats, NULL);
|
|
|
|
/* simply dump the stats structure */
|
|
str = gst_structure_to_string (stats);
|
|
g_print ("source stats: %s\n", str);
|
|
|
|
gst_structure_free (stats);
|
|
g_free (str);
|
|
}
|
|
|
|
/* this function is called every second and dumps the RTP manager stats */
|
|
static gboolean
|
|
print_stats (GstElement * rtpbin)
|
|
{
|
|
GObject *session;
|
|
GValueArray *arr;
|
|
GValue *val;
|
|
guint i;
|
|
|
|
g_print ("***********************************\n");
|
|
|
|
/* get session 0 */
|
|
g_signal_emit_by_name (rtpbin, "get-internal-session", 0, &session);
|
|
|
|
/* print all the sources in the session, this includes the internal source */
|
|
g_object_get (session, "sources", &arr, NULL);
|
|
|
|
for (i = 0; i < arr->n_values; i++) {
|
|
GObject *source;
|
|
|
|
val = g_value_array_get_nth (arr, i);
|
|
source = g_value_get_object (val);
|
|
|
|
print_source_stats (source);
|
|
}
|
|
g_value_array_free (arr);
|
|
|
|
g_object_unref (session);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* 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;
|
|
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);
|
|
|
|
if (!gst_element_link_many (audiosrc, audioconv, audiores, audioenc,
|
|
audiopay, NULL)) {
|
|
g_error ("Failed to link audiosrc, audioconv, audioresample, "
|
|
"audio encoder and audio payloader");
|
|
}
|
|
|
|
/* 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");
|
|
if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
|
|
g_error ("Failed to link audio payloader to rtpbin");
|
|
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");
|
|
if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
|
|
g_error ("Failed to link rtpbin to rtpsink");
|
|
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");
|
|
if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
|
|
g_error ("Failed to link rtpbin to rtcpsink");
|
|
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");
|
|
if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
|
|
g_error ("Failed to link rtcpsrc to rtpbin");
|
|
gst_object_unref (srcpad);
|
|
|
|
/* set the pipeline to playing */
|
|
g_print ("starting sender pipeline\n");
|
|
gst_element_set_state (pipeline, GST_STATE_PLAYING);
|
|
|
|
/* print stats every second */
|
|
g_timeout_add (1000, (GSourceFunc) print_stats, rtpbin);
|
|
|
|
/* 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;
|
|
}
|