tests/examples/rtp/client-PCMA.c - imx-gst-plugins-good - Git at Google

 /* 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., 51 Franklin St, Fifth Floor,
  * Boston, MA 02110-1301, 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"

 /* print the stats of a source */
 static void
 print_source_stats (GObject * source)
 {
   GstStructure *stats;
   gchar *str;

   g_return_if_fail (source != NULL);

   /* 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);
 }

 /* will be called when rtpbin signals on-ssrc-active. It means that an RTCP
  * packet was received from another source. */
 static void
 on_ssrc_active_cb (GstElement * rtpbin, guint sessid, guint ssrc,
     GstElement * depay)
 {
   GObject *session, *osrc;

   g_print ("got RTCP from session %u, SSRC %u\n", sessid, ssrc);

   /* get the right session */
   g_signal_emit_by_name (rtpbin, "get-internal-session", sessid, &session);

 #if 0
   /* FIXME: This is broken in rtpbin */
   /* get the internal source (the SSRC allocated to us, the receiver */
   g_object_get (session, "internal-source", &isrc, NULL);
   print_source_stats (isrc);
   g_object_unref (isrc);
 #endif

   /* get the remote source that sent us RTCP */
   g_signal_emit_by_name (session, "get-source-by-ssrc", ssrc, &osrc);
   print_source_stats (osrc);
   g_object_unref (osrc);
   g_object_unref (session);
 }

 /* 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-1.0 -v rtpbin name=rtpbin                                                \
  *      udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_0              \
  *        rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! autoaudiosink \
  *      udpsrc port=5003 ! rtpbin.recv_rtcp_sink_0                              \
  *        rtpbin.send_rtcp_src_0 ! udpsink port=5007 host=$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 ("rtpbin", "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);

   /* give some stats when we receive RTCP */
   g_signal_connect (rtpbin, "on-ssrc-active", G_CALLBACK (on_ssrc_active_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;
 }
	/* 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., 51 Franklin St, Fifth Floor,
	* Boston, MA 02110-1301, 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"

	/* print the stats of a source */
	static void
	print_source_stats (GObject * source)
	{
	GstStructure *stats;
	gchar *str;

	g_return_if_fail (source != NULL);

	/* 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);
	}

	/* will be called when rtpbin signals on-ssrc-active. It means that an RTCP
	* packet was received from another source. */
	static void
	on_ssrc_active_cb (GstElement * rtpbin, guint sessid, guint ssrc,
	GstElement * depay)
	{
	GObject session, osrc;

	g_print ("got RTCP from session %u, SSRC %u\n", sessid, ssrc);

	/* get the right session */
	g_signal_emit_by_name (rtpbin, "get-internal-session", sessid, &session);

	#if 0
	/* FIXME: This is broken in rtpbin */
	/* get the internal source (the SSRC allocated to us, the receiver */
	g_object_get (session, "internal-source", &isrc, NULL);
	print_source_stats (isrc);
	g_object_unref (isrc);
	#endif

	/* get the remote source that sent us RTCP */
	g_signal_emit_by_name (session, "get-source-by-ssrc", ssrc, &osrc);
	print_source_stats (osrc);
	g_object_unref (osrc);
	g_object_unref (session);
	}

	/* 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-1.0 -v rtpbin name=rtpbin \
	* udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_0 \
	* rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! autoaudiosink \
	* udpsrc port=5003 ! rtpbin.recv_rtcp_sink_0 \
	* rtpbin.send_rtcp_src_0 ! udpsink port=5007 host=$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 ("rtpbin", "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);

	/* give some stats when we receive RTCP */
	g_signal_connect (rtpbin, "on-ssrc-active", G_CALLBACK (on_ssrc_active_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;
	}