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tests/examples/rtp/: Add two C examples of using gstrtpbin as a sender and a receiver.

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Original commit message from CVS:
* tests/examples/rtp/.cvsignore:
* tests/examples/rtp/Makefile.am:
* tests/examples/rtp/client-PCMA.c: (pad_added_cb), (main):
* tests/examples/rtp/server-alsasrc-PCMA.c: (main):
Add two C examples of using gstrtpbin as a sender and a receiver.
This commit is contained in:
Wim Taymans 2009-01-02 15:20:48 +00:00
parent f2c94d1468
commit 9f46e70477
5 changed files with 379 additions and 0 deletions
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8
ChangeLog
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@ -1,3 +1,11 @@
2009-01-02 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/examples/rtp/.cvsignore:
* tests/examples/rtp/Makefile.am:
* tests/examples/rtp/client-PCMA.c: (pad_added_cb), (main):
* tests/examples/rtp/server-alsasrc-PCMA.c: (main):
Add two C examples of using gstrtpbin as a sender and a receiver.
2008-12-31 Jan Schmidt <jan.schmidt@sun.com>
* ChangeLog:

2
tests/examples/rtp/.gitignore vendored Normal file
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client-PCMA
server-alsasrc-PCMA

10
tests/examples/rtp/Makefile.am
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noinst_PROGRAMS = server-alsasrc-PCMA client-PCMA
server_alsasrc_PCMA_SOURCES = server-alsasrc-PCMA.c
server_alsasrc_PCMA_CFLAGS = $(GST_CFLAGS)
server_alsasrc_PCMA_LDADD = $(GST_LIBS) $(LIBM)
client_PCMA_SOURCES = client-PCMA.c
client_PCMA_CFLAGS = $(GST_CFLAGS)
client_PCMA_LDADD = $(GST_LIBS) $(LIBM)
noinst_SCRIPTS=client-H263p-AMR.sh \
client-H263p-PCMA.sh \
client-H264-PCMA.sh \

191
tests/examples/rtp/client-PCMA.c Executable file
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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 receiver
*
* receives alaw encoded RTP audio on port 5002, RTCP is received on port 5003.
* the receiver RTCP reports are sent to port 5007
*
* .-------. .----------. .---------. .-------. .--------.
* RTP |udpsrc | | rtpbin | |pcmadepay| |alawdec| |alsasink|
* port=5002 | src->recv_rtp recv_rtp->sink src->sink src->sink |
* '-------' | | '---------' '-------' '--------'
* | |
* | | .-------.
* | | |udpsink| RTCP
* | send_rtcp->sink | port=5007
* .-------. | | '-------' sync=false
* RTCP |udpsrc | | | async=false
* port=5003 | src->recv_rtcp |
* '-------' '----------'
*/
/* the caps of the sender RTP stream. This is usually negotiated out of band with
* SDP or RTSP. */
#define AUDIO_CAPS "application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA"
#define AUDIO_DEPAY "rtppcmadepay"
#define AUDIO_DEC "alawdec"
#define AUDIO_SINK "autoaudiosink"
/* the destination machine to send RTCP to. This is the address of the sender and
* is used to send back the RTCP reports of this receiver. If the data is sent
* from another machine, change this address. */
#define DEST_HOST "127.0.0.1"
/* will be called when rtpbin has validated a payload that we can depayload */
static void
pad_added_cb (GstElement * rtpbin, GstPad * new_pad, GstElement * depay)
{
GstPad *sinkpad;
GstPadLinkReturn lres;
g_print ("new payload on pad: %s\n", GST_PAD_NAME (new_pad));
sinkpad = gst_element_get_static_pad (depay, "sink");
g_assert (sinkpad);
lres = gst_pad_link (new_pad, sinkpad);
g_assert (lres == GST_PAD_LINK_OK);
gst_object_unref (sinkpad);
}
/* build a pipeline equivalent to:
*
* gst-launch -v gstrtpbin name=rtpbin \
* udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_0 \
* rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! alsasink \
* udpsrc port=5003 ! rtpbin.recv_rtcp_sink_0 \
* rtpbin.send_rtcp_src_0 ! udpsink port=5007 host=$DEST sync=false async=false
*/
int
main (int argc, char *argv[])
{
GstElement *rtpbin, *rtpsrc, *rtcpsrc, *rtcpsink;
GstElement *audiodepay, *audiodec, *audiores, *audioconv, *audiosink;
GstElement *pipeline;
GMainLoop *loop;
GstCaps *caps;
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 udp src and source we will use for RTP and RTCP */
rtpsrc = gst_element_factory_make ("udpsrc", "rtpsrc");
g_assert (rtpsrc);
g_object_set (rtpsrc, "port", 5002, NULL);
/* we need to set caps on the udpsrc for the RTP data */
caps = gst_caps_from_string (AUDIO_CAPS);
g_object_set (rtpsrc, "caps", caps, NULL);
gst_caps_unref (caps);
rtcpsrc = gst_element_factory_make ("udpsrc", "rtcpsrc");
g_assert (rtcpsrc);
g_object_set (rtcpsrc, "port", 5003, NULL);
rtcpsink = gst_element_factory_make ("udpsink", "rtcpsink");
g_assert (rtcpsink);
g_object_set (rtcpsink, "port", 5007, "host", DEST_HOST, NULL);
/* no need for synchronisation or preroll on the RTCP sink */
g_object_set (rtcpsink, "async", FALSE, "sync", FALSE, NULL);
gst_bin_add_many (GST_BIN (pipeline), rtpsrc, rtcpsrc, rtcpsink, NULL);
/* the depayloading and decoding */
audiodepay = gst_element_factory_make (AUDIO_DEPAY, "audiodepay");
g_assert (audiodepay);
audiodec = gst_element_factory_make (AUDIO_DEC, "audiodec");
g_assert (audiodec);
/* the audio playback and format conversion */
audioconv = gst_element_factory_make ("audioconvert", "audioconv");
g_assert (audioconv);
audiores = gst_element_factory_make ("audioresample", "audiores");
g_assert (audiores);
audiosink = gst_element_factory_make (AUDIO_SINK, "audiosink");
g_assert (audiosink);
/* add depayloading and playback to the pipeline and link */
gst_bin_add_many (GST_BIN (pipeline), audiodepay, audiodec, audioconv,
audiores, audiosink, NULL);
res = gst_element_link_many (audiodepay, audiodec, audioconv, audiores,
audiosink, 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);
/* now link all to the rtpbin, start by getting an RTP sinkpad for session 0 */
srcpad = gst_element_get_static_pad (rtpsrc, "src");
sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_0");
lres = gst_pad_link (srcpad, sinkpad);
g_assert (lres == GST_PAD_LINK_OK);
gst_object_unref (srcpad);
/* get an RTCP sinkpad in session 0 */
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);
gst_object_unref (sinkpad);
/* get an RTCP srcpad for sending RTCP back to the sender */
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);
/* the RTP pad that we have to connect to the depayloader will be created
* dynamically so we connect to the pad-added signal, pass the depayloader as
* user_data so that we can link to it. */
g_signal_connect (rtpbin, "pad-added", G_CALLBACK (pad_added_cb), audiodepay);
/* set the pipeline to playing */
g_print ("starting receiver 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 receiver pipeline\n");
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
return 0;
}

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tests/examples/rtp/server-alsasrc-PCMA.c Executable file
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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;
}