1 files changed, 380 insertions, 0 deletions

diff --git a/subprojects/gst-plugins-good/tests/examples/rtp/client-rtpaux.c b/subprojects/gst-plugins-good/tests/examples/rtp/client-rtpaux.c
new file mode 100644
index 0000000000..c7aa781d1d
--- /dev/null
+++ b/subprojects/gst-plugins-good/tests/examples/rtp/client-rtpaux.c
@@ -0,0 +1,380 @@
+/* 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;
+  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);
+
+  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);
+
+  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) {
+    caps_str = gst_caps_to_string (data->caps);
+    g_print ("Returning %s\n", caps_str);
+    g_free (caps_str);
+    return gst_caps_ref (data->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;
+
+  GST_INFO ("creating AUX receiver");
+  bin = gst_bin_new (NULL);
+  rtx = gst_element_factory_make ("rtprtxreceive", NULL);
+  pt_map = gst_structure_new ("application/x-rtp-pt-map",
+      "8", G_TYPE_UINT, 98, "96", G_TYPE_UINT, 99, NULL);
+  g_object_set (rtx, "payload-type-map", pt_map, NULL);
+  gst_structure_free (pt_map);
+  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;
+}