scenic

/* 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>

#include <gst/app/gstappsink.h>

#include "shared-video.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   |     |         |   |       |   |        |

 *  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. */

/* 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"

typedef struct _App App;

struct _App

{

    GstElement *pipeline;

    ScenicSharedVideo::Writer *writer;

    std::string socketName;

};

App app;

/* 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 gstrtpbin 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,

                    gpointer user_data/*GstElement * depay*/)

 {

   GObject *session, *isrc, *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);

   /* get the internal source (the SSRC allocated to us, the receiver */

   g_object_get (session, "internal-source", &isrc, NULL);

   print_source_stats (isrc);

   /* get the remote source that sent us RTCP */

   g_signal_emit_by_name (session, "get-source-by-ssrc", ssrc, &osrc);

   print_source_stats (osrc);

 }

 static void

 pad_removed_cb (GstElement * rtpbin, GstPad * new_pad, gpointer user_data/*GstElement * depay*/)

 {

     g_print ("pad removed: %s\n", GST_PAD_NAME (new_pad));

 }

 /* will be called when rtpbin has validated a payload that we can depayload */

 static void

 pad_added_cb (GstElement * rtpbin, GstPad * new_pad, gpointer user_data/*GstElement * depay*/)

 {

   GstPad *sinkpad;

   GstPadLinkReturn lres;

   App *app = (App *)user_data;

   g_print ("new payload on pad: %s\n", GST_PAD_NAME (new_pad));

  /* the depayloading and decoding */

   GstElement *gstdepay = gst_element_factory_make ("rtpgstdepay" , NULL);

   g_assert (gstdepay);

   gst_element_set_state (gstdepay,GST_STATE_PLAYING);

   /* add depayloading and playback to the pipeline and link */

   gst_bin_add (GST_BIN (app->pipeline), gstdepay);

    sinkpad = gst_element_get_static_pad (gstdepay, "sink");

    g_assert (sinkpad);

    lres = gst_pad_link (new_pad, sinkpad);

    g_assert (lres == GST_PAD_LINK_OK);

    gst_object_unref (sinkpad);

    app->writer = new ScenicSharedVideo::Writer (app->pipeline,gstdepay,app->socketName);

 }

void

leave(int sig) {

    gst_element_set_state (app.pipeline, GST_STATE_NULL);

    gst_object_unref (GST_OBJECT (app.pipeline));

    exit(sig);

}

/* build a pipeline equivalent to:

 *

 * gst-launch -v gstrtpbin name=rtpbin                                                \

 *      udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_0              \

 *        rtpbin. ! rtpgstdepay ! appsink \

 *      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, *rtcpsrc, *rtcpsink;

    GstElement *rtpsrc;

    GMainLoop *loop;

    GstCaps *caps;

    GstPadLinkReturn lres;

    GstPad *srcpad, *sinkpad;

     /* Check input arguments */

    if (argc != 2) {

        g_printerr ("Usage: %s <socket-path>\n", argv[0]);

        return -1;

    }

    app.socketName.append (argv[1]);

    (void) signal(SIGINT,leave);

  /* always init first */

  gst_init (&argc, &argv);

  /* the pipeline to hold everything */

  app.pipeline = gst_pipeline_new (NULL);

  g_assert (app.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 inside caps is obtained with "echo application/nicolas | base64", removing the "0=" at the end

     caps = gst_caps_from_string ("application/x-rtp,media=(string)application,payload=(int)96,clock-rate=(int)90000,encoding-name=(string)X-GST,caps=(string)YXBwbGljYXRpb24vc2NlbmljZGF0YV8K");

     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 (app.pipeline), rtpsrc, 

                       rtcpsrc, rtcpsink, 

                       NULL);

    /* the rtpbin element */

    rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");

    g_assert (rtpbin);

    gst_bin_add (GST_BIN (app.pipeline), rtpbin);

    g_print ("rtp sinkpad\n");

    /* now link all to the rtpbin, start by getting an RTP sinkpad for session 0 */

     srcpad = gst_element_get_static_pad (rtpsrc, "src");

     g_assert (srcpad);

     sinkpad = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_0");

     g_assert (sinkpad);

     lres = gst_pad_link (srcpad, sinkpad);

     g_print ("%d \n",lres);

     g_assert (lres == GST_PAD_LINK_OK);

     gst_object_unref (srcpad);

    g_print ("rtcp sinkpad\n");

    /* 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);

    g_print ("rtcp srcpad\n");

    /* 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), &app);

    g_signal_connect (rtpbin, "pad-removed", G_CALLBACK (pad_removed_cb), &app);

    /* give some stats when we receive RTCP */

    g_signal_connect (rtpbin, "on-ssrc-active", G_CALLBACK (on_ssrc_active_cb),

                           NULL);

  /* set the pipeline to playing */

  g_print ("starting receiver pipeline\n");

  gst_element_set_state (app.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 (app.pipeline, GST_STATE_NULL);

  gst_object_unref (app.pipeline);

  return 0;

}