gstreamer/subprojects/gst-plugins-good/tests/examples/rtp/client-rtpaux.c

/* GStreamer
 * Copyright (C) 2013 Collabora Ltd.
 *   @author Torrie Fischer <torrie.fischer@collabora.co.uk>
 *
 * 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 <gst/gst.h>
#include <gst/rtp/rtp.h>
#include <stdlib.h>

/*
 * RTP receiver with RFC4588 retransmission handling enabled
 *
 *  In this example we have two RTP sessions, one for video and one for audio.
 *  Video is received on port 5000, with its RTCP stream received on port 5001
 *  and sent on port 5005. Audio is received on port 5005, with its RTCP stream
 *  received on port 5006 and sent on port 5011.
 *
 *  In both sessions, we set "rtprtxreceive" as the session's "aux" element
 *  in rtpbin, which enables RFC4588 retransmission handling for that session.
 *
 *             .-------.      .----------.        .-----------.   .---------.   .-------------.
 *  RTP        |udpsrc |      | rtpbin   |        |theoradepay|   |theoradec|   |autovideosink|
 *  port=5000  |      src->recv_rtp_0 recv_rtp_0->sink       src->sink     src->sink          |
 *             '-------'      |          |        '-----------'   '---------'   '-------------'
 *                            |          |
 *                            |          |     .-------.
 *                            |          |     |udpsink|  RTCP
 *                            |  send_rtcp_0->sink     | port=5005
 *             .-------.      |          |     '-------' sync=false
 *  RTCP       |udpsrc |      |          |               async=false
 *  port=5001  |     src->recv_rtcp_0    |
 *             '-------'      |          |
 *                            |          |
 *             .-------.      |          |        .---------.   .-------.   .-------------.
 *  RTP        |udpsrc |      |          |        |pcmadepay|   |alawdec|   |autoaudiosink|
 *  port=5006  |      src->recv_rtp_1 recv_rtp_1->sink     src->sink   src->sink          |
 *             '-------'      |          |        '---------'   '-------'   '-------------'
 *                            |          |
 *                            |          |     .-------.
 *                            |          |     |udpsink|  RTCP
 *                            |  send_rtcp_1->sink     | port=5011
 *             .-------.      |          |     '-------' sync=false
 *  RTCP       |udpsrc |      |          |               async=false
 *  port=5007  |     src->recv_rtcp_1    |
 *             '-------'      '----------'
 *
 */

GMainLoop *loop = NULL;

typedef struct _SessionData
{
  int ref;
  GstElement *rtpbin;
  guint sessionNum;
  guint pt, rtxPt;
  GstCaps *caps;
  GstElement *output;
} SessionData;

static SessionData *
session_ref (SessionData * data)
{
  g_atomic_int_inc (&data->ref);
  return data;
}

static void
session_unref (gpointer data)
{
  SessionData *session = (SessionData *) data;
  if (g_atomic_int_dec_and_test (&session->ref)) {
    g_object_unref (session->rtpbin);
    gst_caps_unref (session->caps);
    g_free (session);
  }
}

static SessionData *
session_new (guint sessionNum)
{
  SessionData *ret = g_new0 (SessionData, 1);
  ret->sessionNum = sessionNum;
  return session_ref (ret);
}

static void
setup_ghost_sink (GstElement * sink, GstBin * bin)
{
  GstPad *sinkPad = gst_element_get_static_pad (sink, "sink");
  GstPad *binPad = gst_ghost_pad_new ("sink", sinkPad);
  gst_element_add_pad (GST_ELEMENT (bin), binPad);
}

static SessionData *
make_audio_session (guint sessionNum)
{
  SessionData *ret = session_new (sessionNum);
  GstBin *bin = GST_BIN (gst_bin_new ("audio"));
  GstElement *queue = gst_element_factory_make ("queue", NULL);
  GstElement *sink = gst_element_factory_make ("autoaudiosink", NULL);
  GstElement *audioconvert = gst_element_factory_make ("audioconvert", NULL);
  GstElement *audioresample = gst_element_factory_make ("audioresample", NULL);
  GstElement *depayloader = gst_element_factory_make ("rtppcmadepay", NULL);
  GstElement *decoder = gst_element_factory_make ("alawdec", NULL);

  gst_bin_add_many (bin, queue, depayloader, decoder, audioconvert,
      audioresample, sink, NULL);
  gst_element_link_many (queue, depayloader, decoder, audioconvert,
      audioresample, sink, NULL);

  setup_ghost_sink (queue, bin);

  ret->output = GST_ELEMENT (bin);
  ret->caps = gst_caps_new_simple ("application/x-rtp",
      "media", G_TYPE_STRING, "audio",
      "clock-rate", G_TYPE_INT, 8000,
      "encoding-name", G_TYPE_STRING, "PCMA", NULL);
  ret->pt = 8;
  ret->rtxPt = 98;

  return ret;
}

static SessionData *
make_video_session (guint sessionNum)
{
  SessionData *ret = session_new (sessionNum);
  GstBin *bin = GST_BIN (gst_bin_new ("video"));
  GstElement *queue = gst_element_factory_make ("queue", NULL);
  GstElement *depayloader = gst_element_factory_make ("rtptheoradepay", NULL);
  GstElement *decoder = gst_element_factory_make ("theoradec", NULL);
  GstElement *converter = gst_element_factory_make ("videoconvert", NULL);
  GstElement *sink = gst_element_factory_make ("autovideosink", NULL);

  gst_bin_add_many (bin, depayloader, decoder, converter, queue, sink, NULL);
  gst_element_link_many (queue, depayloader, decoder, converter, sink, NULL);

  setup_ghost_sink (queue, bin);

  ret->output = GST_ELEMENT (bin);
  ret->caps = gst_caps_new_simple ("application/x-rtp",
      "media", G_TYPE_STRING, "video",
      "clock-rate", G_TYPE_INT, 90000,
      "encoding-name", G_TYPE_STRING, "THEORA", NULL);
  ret->pt = 96;
  ret->rtxPt = 99;

  return ret;
}

static GstCaps *
request_pt_map (GstElement * rtpbin, guint session, guint pt,
    gpointer user_data)
{
  SessionData *data = (SessionData *) user_data;
  gchar *caps_str;
  g_print ("Looking for caps for pt %u in session %u, have %u\n", pt, session,
      data->sessionNum);
  if (session == data->sessionNum) {
    GstCaps *caps;

    if (pt == data->pt) {
      caps = gst_caps_ref (data->caps);
    } else if (pt == data->rtxPt) {
      caps = gst_caps_copy (data->caps);
      gst_caps_set_simple (caps, "encoding-name", G_TYPE_STRING, "rtx", NULL);
    } else {
      return NULL;
    }

    caps_str = gst_caps_to_string (caps);
    g_print ("Returning %s\n", caps_str);
    g_free (caps_str);

    return caps;
  }
  return NULL;
}

static void
cb_eos (GstBus * bus, GstMessage * message, gpointer data)
{
  g_print ("Got EOS\n");
  g_main_loop_quit (loop);
}

static void
cb_state (GstBus * bus, GstMessage * message, gpointer data)
{
  GstObject *pipe = GST_OBJECT (data);
  GstState old, new, pending;
  gst_message_parse_state_changed (message, &old, &new, &pending);
  if (message->src == pipe) {
    g_print ("Pipeline %s changed state from %s to %s\n",
        GST_OBJECT_NAME (message->src),
        gst_element_state_get_name (old), gst_element_state_get_name (new));
  }
}

static void
cb_warning (GstBus * bus, GstMessage * message, gpointer data)
{
  GError *error = NULL;
  gst_message_parse_warning (message, &error, NULL);
  g_printerr ("Got warning from %s: %s\n", GST_OBJECT_NAME (message->src),
      error->message);
  g_error_free (error);
}

static void
cb_error (GstBus * bus, GstMessage * message, gpointer data)
{
  GError *error = NULL;
  gst_message_parse_error (message, &error, NULL);
  g_printerr ("Got error from %s: %s\n", GST_OBJECT_NAME (message->src),
      error->message);
  g_error_free (error);
  g_main_loop_quit (loop);
}

static void
handle_new_stream (GstElement * element, GstPad * newPad, gpointer data)
{
  SessionData *session = (SessionData *) data;
  gchar *padName;
  gchar *myPrefix;

  padName = gst_pad_get_name (newPad);
  myPrefix = g_strdup_printf ("recv_rtp_src_%u", session->sessionNum);

  g_print ("New pad: %s, looking for %s_*\n", padName, myPrefix);

  if (g_str_has_prefix (padName, myPrefix)) {
    GstPad *outputSinkPad;
    GstElement *parent;

    parent = GST_ELEMENT (gst_element_get_parent (session->rtpbin));
    gst_bin_add (GST_BIN (parent), session->output);
    gst_element_sync_state_with_parent (session->output);
    gst_object_unref (parent);

    outputSinkPad = gst_element_get_static_pad (session->output, "sink");
    g_assert_cmpint (gst_pad_link (newPad, outputSinkPad), ==, GST_PAD_LINK_OK);
    gst_object_unref (outputSinkPad);

    g_print ("Linked!\n");
  }
  g_free (myPrefix);
  g_free (padName);
}

static GstElement *
request_aux_receiver (GstElement * rtpbin, guint sessid, SessionData * session)
{
  GstElement *rtx, *bin;
  GstPad *pad;
  gchar *name;
  GstStructure *pt_map;
  gchar *media_pt;

  if (sessid != session->sessionNum)
    return NULL;

  GST_INFO ("creating AUX receiver for session %u", sessid);
  bin = gst_bin_new (NULL);
  rtx = gst_element_factory_make ("rtprtxreceive", NULL);

  media_pt = g_strdup_printf ("%d", session->pt);
  pt_map = gst_structure_new ("application/x-rtp-pt-map",
      media_pt, G_TYPE_UINT, session->rtxPt, NULL);
  g_object_set (rtx, "payload-type-map", pt_map, NULL);
  gst_structure_free (pt_map);
  g_free (media_pt);

  gst_bin_add (GST_BIN (bin), rtx);

  pad = gst_element_get_static_pad (rtx, "src");
  name = g_strdup_printf ("src_%u", sessid);
  gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
  g_free (name);
  gst_object_unref (pad);

  pad = gst_element_get_static_pad (rtx, "sink");
  name = g_strdup_printf ("sink_%u", sessid);
  gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
  g_free (name);
  gst_object_unref (pad);

  return bin;
}

static void
join_session (GstElement * pipeline, GstElement * rtpBin, SessionData * session)
{
  GstElement *rtpSrc;
  GstElement *rtcpSrc;
  GstElement *rtcpSink;
  gchar *padName;
  guint basePort;

  g_print ("Joining session %p\n", session);

  session->rtpbin = g_object_ref (rtpBin);

  basePort = 5000 + (session->sessionNum * 6);

  rtpSrc = gst_element_factory_make ("udpsrc", NULL);
  rtcpSrc = gst_element_factory_make ("udpsrc", NULL);
  rtcpSink = gst_element_factory_make ("udpsink", NULL);
  g_object_set (rtpSrc, "port", basePort, "caps", session->caps, NULL);
  g_object_set (rtcpSink, "port", basePort + 5, "host", "127.0.0.1", "sync",
      FALSE, "async", FALSE, NULL);
  g_object_set (rtcpSrc, "port", basePort + 1, NULL);

  g_print ("Connecting to %i/%i/%i\n", basePort, basePort + 1, basePort + 5);

  /* enable RFC4588 retransmission handling by setting rtprtxreceive
   * as the "aux" element of rtpbin */
  g_signal_connect (rtpBin, "request-aux-receiver",
      (GCallback) request_aux_receiver, session);

  gst_bin_add_many (GST_BIN (pipeline), rtpSrc, rtcpSrc, rtcpSink, NULL);

  g_signal_connect_data (rtpBin, "pad-added", G_CALLBACK (handle_new_stream),
      session_ref (session), (GClosureNotify) session_unref, 0);

  g_signal_connect_data (rtpBin, "request-pt-map", G_CALLBACK (request_pt_map),
      session_ref (session), (GClosureNotify) session_unref, 0);

  padName = g_strdup_printf ("recv_rtp_sink_%u", session->sessionNum);
  gst_element_link_pads (rtpSrc, "src", rtpBin, padName);
  g_free (padName);

  padName = g_strdup_printf ("recv_rtcp_sink_%u", session->sessionNum);
  gst_element_link_pads (rtcpSrc, "src", rtpBin, padName);
  g_free (padName);

  padName = g_strdup_printf ("send_rtcp_src_%u", session->sessionNum);
  gst_element_link_pads (rtpBin, padName, rtcpSink, "sink");
  g_free (padName);

  session_unref (session);
}

int
main (int argc, char **argv)
{
  GstPipeline *pipe;
  SessionData *videoSession;
  SessionData *audioSession;
  GstElement *rtpBin;
  GstBus *bus;

  gst_init (&argc, &argv);

  loop = g_main_loop_new (NULL, FALSE);
  pipe = GST_PIPELINE (gst_pipeline_new (NULL));

  bus = gst_element_get_bus (GST_ELEMENT (pipe));
  g_signal_connect (bus, "message::error", G_CALLBACK (cb_error), pipe);
  g_signal_connect (bus, "message::warning", G_CALLBACK (cb_warning), pipe);
  g_signal_connect (bus, "message::state-changed", G_CALLBACK (cb_state), pipe);
  g_signal_connect (bus, "message::eos", G_CALLBACK (cb_eos), NULL);
  gst_bus_add_signal_watch (bus);
  gst_object_unref (bus);

  rtpBin = gst_element_factory_make ("rtpbin", NULL);
  gst_bin_add (GST_BIN (pipe), rtpBin);
  g_object_set (rtpBin, "latency", 200, "do-retransmission", TRUE,
      "rtp-profile", GST_RTP_PROFILE_AVPF, NULL);

  videoSession = make_video_session (0);
  audioSession = make_audio_session (1);

  join_session (GST_ELEMENT (pipe), rtpBin, videoSession);
  join_session (GST_ELEMENT (pipe), rtpBin, audioSession);

  g_print ("starting client pipeline\n");
  gst_element_set_state (GST_ELEMENT (pipe), GST_STATE_PLAYING);

  g_main_loop_run (loop);

  g_print ("stopping client pipeline\n");
  gst_element_set_state (GST_ELEMENT (pipe), GST_STATE_NULL);

  gst_object_unref (pipe);
  g_main_loop_unref (loop);

  return 0;
}