/* 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;

}