tests/examples/rtp/server-alsasrc-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.
  */

 /* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
  * with newer GLib versions (>= 2.31.0) */
 #define GLIB_DISABLE_DEPRECATION_WARNINGS

 #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-1.0 -v rtpbin 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 ("rtpbin", "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_seconds (1, (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;
 }
	/* 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.
	*/

	/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
	* with newer GLib versions (>= 2.31.0) */
	#define GLIB_DISABLE_DEPRECATION_WARNINGS

	#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-1.0 -v rtpbin 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 ("rtpbin", "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_seconds (1, (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;
	}