tests/check: add rtpaux::test_simple_rtpbin_aux

It shows how to use "set-aux-receive" and "set-aux-send"
properties of rtpbin to set rtprtxsend and rtprtxreceive

Build 2 pipelines, one for rtpbin as a sender and one for
rtobin as a receive. Then transmit an audio stream.

It also drops some packets to activate restransmission and
check they are actually retransmited.

3 files changed, 412 insertions, 0 deletions

diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index 8e7f43591..9c3e13592 100644
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -142,6 +142,7 @@ check_PROGRAMS = \
 	elements/rganalysis \
 	elements/rglimiter \
 	elements/rgvolume \
+	elements/rtpaux \
 	elements/rtpcollision \
 	elements/rtp-payloading \
 	elements/rtpbin \
@@ -334,6 +335,9 @@ elements_rtpsession_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VERSION)
 elements_rtpcollision_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
 elements_rtpcollision_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstnet-$(GST_API_VERSION) -lgstrtp-$(GST_API_VERSION) $(GIO_LIBS) $(LDADD)
 
+elements_rtpaux_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
+elements_rtpaux_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)
+
 # FIXME: configure should check for gdk-pixbuf not gtk
 # only need video.h header, not the lib
 elements_gdkpixbufsink_CFLAGS = \
diff --git a/tests/check/elements/.gitignore b/tests/check/elements/.gitignore
index e6b0f5618..dd17aac1d 100644
--- a/tests/check/elements/.gitignore
+++ b/tests/check/elements/.gitignore
@@ -47,6 +47,7 @@ rganalysis
 rglimiter
 rgvolume
 rtp-payloading
+rtpaux
 rtpbin
 rtpbin_buffer_list
 rtpcollision
diff --git a/tests/check/elements/rtpaux.c b/tests/check/elements/rtpaux.c
new file mode 100644
index 000000000..a2fb02f0b
--- /dev/null
+++ b/tests/check/elements/rtpaux.c
@@ -0,0 +1,407 @@
+/* GStreamer
+ *
+ * Copyright (C) 2013 Collabora Ltd.
+ *   @author Julien Isorce <julien.isorce@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/check/gstcheck.h>
+#include <gst/check/gstconsistencychecker.h>
+#include <gst/check/gsttestclock.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
+static GMainLoop *main_loop;
+
+static void
+message_received (GstBus * bus, GstMessage * message, GstPipeline * bin)
+{
+  GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
+      GST_MESSAGE_SRC (message), message);
+
+  switch (message->type) {
+    case GST_MESSAGE_EOS:
+      g_main_loop_quit (main_loop);
+      break;
+    case GST_MESSAGE_WARNING:{
+      GError *gerror;
+      gchar *debug;
+
+      gst_message_parse_warning (message, &gerror, &debug);
+      gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
+      g_error_free (gerror);
+      g_free (debug);
+      break;
+    }
+    case GST_MESSAGE_ERROR:{
+      GError *gerror;
+      gchar *debug;
+
+      gst_message_parse_error (message, &gerror, &debug);
+      gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
+      g_error_free (gerror);
+      g_free (debug);
+      g_main_loop_quit (main_loop);
+      break;
+    }
+    default:
+      break;
+  }
+}
+
+typedef struct
+{
+  guint count;
+  guint nb_packets;
+  guint drop_every_n_packets;
+} RTXSendData;
+
+static GstPadProbeReturn
+rtprtxsend_srcpad_probe (GstPad * pad, GstPadProbeInfo * info,
+    gpointer user_data)
+{
+  GstPadProbeReturn ret = GST_PAD_PROBE_OK;
+
+  if (info->type == (GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH)) {
+    GstBuffer *buffer = GST_BUFFER (info->data);
+    RTXSendData *rtxdata = (RTXSendData *) user_data;
+    GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
+    guint payload_type = 0;
+
+    gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
+    payload_type = gst_rtp_buffer_get_payload_type (&rtp);
+
+    /* main stream packets */
+    if (payload_type == 96) {
+      /* count packets of the main stream */
+      ++rtxdata->nb_packets;
+      /* drop some packets */
+      if (rtxdata->count < rtxdata->drop_every_n_packets) {
+        ++rtxdata->count;
+      } else {
+        /* drop a packet every 'rtxdata->count' packets */
+        rtxdata->count = 1;
+        ret = GST_PAD_PROBE_DROP;
+      }
+    } else {
+      /* retransmission packets */
+    }
+
+    gst_rtp_buffer_unmap (&rtp);
+  }
+
+  return ret;
+}
+
+static void
+on_rtpbinreceive_pad_added (GstElement * element, GstPad * newPad,
+    gpointer data)
+{
+  GstElement *rtpdepayloader = GST_ELEMENT (data);
+
+  gchar *padName = gst_pad_get_name (newPad);
+  if (g_str_has_prefix (padName, "recv_rtp_src_")) {
+    GstPad *sinkpad = gst_element_get_static_pad (rtpdepayloader, "sink");
+    gst_pad_link (newPad, sinkpad);
+    gst_object_unref (sinkpad);
+  }
+  g_free (padName);
+}
+
+static gboolean
+on_timeout (gpointer data)
+{
+  GstEvent *eos = gst_event_new_eos ();
+  if (!gst_element_send_event (GST_ELEMENT (data), eos)) {
+    GST_ERROR ("failed to send end of stream event");
+    gst_event_unref (eos);
+  }
+
+  return FALSE;
+}
+
+static GstElement *
+request_aux_receive (GstElement * rtpbin, guint sessid, GstElement * receive)
+{
+  GstElement *bin;
+  GstPad *pad;
+
+  GST_INFO ("creating AUX receiver");
+  bin = gst_bin_new (NULL);
+  gst_bin_add (GST_BIN (bin), receive);
+
+  pad = gst_element_get_static_pad (receive, "src");
+  gst_element_add_pad (bin, gst_ghost_pad_new ("src_0", pad));
+  gst_object_unref (pad);
+  pad = gst_element_get_static_pad (receive, "sink");
+  gst_element_add_pad (bin, gst_ghost_pad_new ("sink_0", pad));
+  gst_object_unref (pad);
+
+  return bin;
+}
+
+static GstElement *
+request_aux_send (GstElement * rtpbin, guint sessid, GstElement * send)
+{
+  GstElement *bin;
+  GstPad *pad;
+
+  GST_INFO ("creating AUX sender");
+  bin = gst_bin_new (NULL);
+  gst_bin_add (GST_BIN (bin), send);
+
+  pad = gst_element_get_static_pad (send, "src");
+  gst_element_add_pad (bin, gst_ghost_pad_new ("src_0", pad));
+  gst_object_unref (pad);
+  pad = gst_element_get_static_pad (send, "sink");
+  gst_element_add_pad (bin, gst_ghost_pad_new ("sink_0", pad));
+  gst_object_unref (pad);
+
+  return bin;
+}
+
+
+GST_START_TEST (test_simple_rtpbin_aux)
+{
+  GstElement *binsend, *rtpbinsend, *src, *encoder, *rtppayloader,
+      *rtprtxsend, *sendrtp_udpsink, *sendrtcp_udpsink, *sendrtcp_udpsrc;
+  GstElement *binreceive, *rtpbinreceive, *recvrtp_udpsrc, *recvrtcp_udpsrc,
+      *recvrtcp_udpsink, *rtprtxreceive, *rtpdepayloader, *decoder, *converter,
+      *sink;
+  GstBus *bussend;
+  GstBus *busreceive;
+  gboolean res;
+  GstCaps *rtpcaps = NULL;
+  GstStateChangeReturn state_res = GST_STATE_CHANGE_FAILURE;
+  GstPad *srcpad = NULL;
+  guint nb_rtx_send_packets = 0;
+  guint nb_rtx_recv_packets = 0;
+  RTXSendData send_rtxdata;
+  send_rtxdata.count = 1;
+  send_rtxdata.nb_packets = 0;
+  send_rtxdata.drop_every_n_packets = 50;
+
+  GST_INFO ("preparing test");
+
+  /* build pipeline */
+  binsend = gst_pipeline_new ("pipeline_send");
+  bussend = gst_element_get_bus (binsend);
+  gst_bus_add_signal_watch_full (bussend, G_PRIORITY_HIGH);
+
+  binreceive = gst_pipeline_new ("pipeline_receive");
+  busreceive = gst_element_get_bus (binreceive);
+  gst_bus_add_signal_watch_full (busreceive, G_PRIORITY_HIGH);
+
+  rtpbinsend = gst_element_factory_make ("rtpbin", "rtpbinsend");
+  g_object_set (rtpbinsend, "latency", 200, "do-retransmission", TRUE, NULL);
+  src = gst_element_factory_make ("audiotestsrc", "src");
+  encoder = gst_element_factory_make ("speexenc", "encoder");
+  rtppayloader = gst_element_factory_make ("rtpspeexpay", "rtppayloader");
+  rtprtxsend = gst_element_factory_make ("rtprtxsend", "rtprtxsend");
+  sendrtp_udpsink = gst_element_factory_make ("udpsink", "sendrtp_udpsink");
+  g_object_set (sendrtp_udpsink, "host", "127.0.0.1", NULL);
+  g_object_set (sendrtp_udpsink, "port", 5006, NULL);
+  sendrtcp_udpsink = gst_element_factory_make ("udpsink", "sendrtcp_udpsink");
+  g_object_set (sendrtcp_udpsink, "host", "127.0.0.1", NULL);
+  g_object_set (sendrtcp_udpsink, "port", 5007, NULL);
+  g_object_set (sendrtcp_udpsink, "sync", FALSE, NULL);
+  g_object_set (sendrtcp_udpsink, "async", FALSE, NULL);
+  sendrtcp_udpsrc = gst_element_factory_make ("udpsrc", "sendrtcp_udpsrc");
+  g_object_set (sendrtcp_udpsrc, "port", 5009, NULL);
+
+  rtpbinreceive = gst_element_factory_make ("rtpbin", "rtpbinreceive");
+  g_object_set (rtpbinreceive, "latency", 200, "do-retransmission", TRUE, NULL);
+  recvrtp_udpsrc = gst_element_factory_make ("udpsrc", "recvrtp_udpsrc");
+  g_object_set (recvrtp_udpsrc, "port", 5006, NULL);
+  rtpcaps =
+      gst_caps_from_string
+      ("application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)SPEEX,encoding-params=(string)1,octet-align=(string)1");
+  g_object_set (recvrtp_udpsrc, "caps", rtpcaps, NULL);
+  gst_caps_unref (rtpcaps);
+  recvrtcp_udpsrc = gst_element_factory_make ("udpsrc", "recvrtcp_udpsrc");
+  g_object_set (recvrtcp_udpsrc, "port", 5007, NULL);
+  recvrtcp_udpsink = gst_element_factory_make ("udpsink", "recvrtcp_udpsink");
+  g_object_set (recvrtcp_udpsink, "host", "127.0.0.1", NULL);
+  g_object_set (recvrtcp_udpsink, "port", 5009, NULL);
+  g_object_set (recvrtcp_udpsink, "sync", FALSE, NULL);
+  g_object_set (recvrtcp_udpsink, "async", FALSE, NULL);
+  rtprtxreceive = gst_element_factory_make ("rtprtxreceive", "rtprtxreceive");
+  rtpdepayloader = gst_element_factory_make ("rtpspeexdepay", "rtpdepayloader");
+  decoder = gst_element_factory_make ("speexdec", "decoder");
+  converter = gst_element_factory_make ("identity", "converter");
+  sink = gst_element_factory_make ("alsasink", "sink");
+
+  gst_bin_add_many (GST_BIN (binsend), rtpbinsend, src, encoder, rtppayloader,
+      sendrtp_udpsink, sendrtcp_udpsink, sendrtcp_udpsrc, NULL);
+
+  gst_bin_add_many (GST_BIN (binreceive), rtpbinreceive,
+      recvrtp_udpsrc, recvrtcp_udpsrc, recvrtcp_udpsink,
+      rtpdepayloader, decoder, converter, sink, NULL);
+
+  g_signal_connect (rtpbinreceive, "pad-added",
+      G_CALLBACK (on_rtpbinreceive_pad_added), rtpdepayloader);
+
+  g_object_set (rtppayloader, "pt", 96, NULL);
+  g_object_set (rtppayloader, "seqnum-offset", 1, NULL);
+  g_object_set (rtprtxsend, "rtx-payload-type", 99, NULL);
+  g_object_set (rtprtxreceive, "rtx-payload-types", "99:111:125", NULL);
+
+  /* set rtp aux receive */
+  g_signal_connect (rtpbinreceive, "request-aux-receiver", (GCallback)
+      request_aux_receive, rtprtxreceive);
+  /* set rtp aux send */
+  g_signal_connect (rtpbinsend, "request-aux-sender", (GCallback)
+      request_aux_send, rtprtxsend);
+
+  /* gst-launch-1.0 rtpbin name=rtpbin audiotestsrc ! amrnbenc ! rtpamrpay ! \
+   * rtpbin.send_rtp_sink_1 rtpbin.send_rtp_src_1 ! udpsink host="127.0.0.1" \
+   * port=5002 rtpbin.send_rtcp_src_1 ! udpsink host="127.0.0.1" port=5003 \
+   * sync=false async=false  udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
+   */
+
+  res = gst_element_link (src, encoder);
+  fail_unless (res == TRUE, NULL);
+  res = gst_element_link (encoder, rtppayloader);
+  fail_unless (res == TRUE, NULL);
+  res =
+      gst_element_link_pads_full (rtppayloader, "src", rtpbinsend,
+      "send_rtp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
+  fail_unless (res == TRUE, NULL);
+  res =
+      gst_element_link_pads_full (rtpbinsend, "send_rtp_src_0", sendrtp_udpsink,
+      "sink", GST_PAD_LINK_CHECK_NOTHING);
+  fail_unless (res == TRUE, NULL);
+  res =
+      gst_element_link_pads_full (rtpbinsend, "send_rtcp_src_0",
+      sendrtcp_udpsink, "sink", GST_PAD_LINK_CHECK_NOTHING);
+  fail_unless (res == TRUE, NULL);
+  res =
+      gst_element_link_pads_full (sendrtcp_udpsrc, "src", rtpbinsend,
+      "recv_rtcp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
+  fail_unless (res == TRUE, NULL);
+
+  srcpad = gst_element_get_static_pad (rtpbinsend, "send_rtp_src_0");
+  gst_pad_add_probe (srcpad,
+      (GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH),
+      (GstPadProbeCallback) rtprtxsend_srcpad_probe, &send_rtxdata, NULL);
+  gst_object_unref (srcpad);
+
+  /* gst-launch-1.0 rtpbin name=rtpbin udpsrc caps="application/x-rtp,media=(string)audio, \
+   * clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,o
+   * ctet-align=(string)1" port=5002 ! rtpbin.recv_rtp_sink_1 rtpbin. ! rtpamrdepay ! \
+   * amrnbdec ! alsasink  udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
+   * rtpbin.send_rtcp_src_1 ! udpsink host="127.0.0.1" port=5007 sync=false async=false
+   */
+
+  res =
+      gst_element_link_pads_full (recvrtp_udpsrc, "src", rtpbinreceive,
+      "recv_rtp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
+  fail_unless (res == TRUE, NULL);
+  res =
+      gst_element_link_pads_full (rtpdepayloader, "src", decoder, "sink",
+      GST_PAD_LINK_CHECK_NOTHING);
+  fail_unless (res == TRUE, NULL);
+  res = gst_element_link (decoder, converter);
+  fail_unless (res == TRUE, NULL);
+  res =
+      gst_element_link_pads_full (converter, "src", sink, "sink",
+      GST_PAD_LINK_CHECK_NOTHING);
+  fail_unless (res == TRUE, NULL);
+  res =
+      gst_element_link_pads_full (recvrtcp_udpsrc, "src", rtpbinreceive,
+      "recv_rtcp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
+  fail_unless (res == TRUE, NULL);
+  res =
+      gst_element_link_pads_full (rtpbinreceive, "send_rtcp_src_0",
+      recvrtcp_udpsink, "sink", GST_PAD_LINK_CHECK_NOTHING);
+  fail_unless (res == TRUE, NULL);
+
+  main_loop = g_main_loop_new (NULL, FALSE);
+  g_signal_connect (bussend, "message::error", (GCallback) message_received,
+      binsend);
+  g_signal_connect (bussend, "message::warning", (GCallback) message_received,
+      binsend);
+  g_signal_connect (bussend, "message::eos", (GCallback) message_received,
+      binsend);
+
+  g_signal_connect (busreceive, "message::error", (GCallback) message_received,
+      binreceive);
+  g_signal_connect (busreceive, "message::warning",
+      (GCallback) message_received, binreceive);
+  g_signal_connect (busreceive, "message::eos", (GCallback) message_received,
+      binreceive);
+
+  state_res = gst_element_set_state (binreceive, GST_STATE_PLAYING);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+  state_res = gst_element_set_state (binsend, GST_STATE_PLAYING);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+  g_timeout_add (5000, on_timeout, binsend);
+  g_timeout_add (5000, on_timeout, binreceive);
+
+  GST_INFO ("enter mainloop");
+  g_main_loop_run (main_loop);
+  g_main_loop_run (main_loop);
+  GST_INFO ("exit mainloop");
+
+  /* check that FB NACK is working */
+  g_object_get (G_OBJECT (rtprtxsend), "num-rtx-requests", &nb_rtx_send_packets,
+      NULL);
+  g_object_get (G_OBJECT (rtprtxreceive), "num-rtx-requests",
+      &nb_rtx_recv_packets, NULL);
+
+  state_res = gst_element_set_state (binsend, GST_STATE_NULL);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+  state_res = gst_element_set_state (binreceive, GST_STATE_NULL);
+  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+  GST_INFO ("nb_rtx_send_packets %d", nb_rtx_send_packets);
+  GST_INFO ("nb_rtx_recv_packets %d", nb_rtx_recv_packets);
+  fail_if (nb_rtx_send_packets < 1);
+  fail_if (nb_rtx_recv_packets < 1);
+
+  /* cleanup */
+  g_main_loop_unref (main_loop);
+
+  gst_bus_remove_signal_watch (bussend);
+  gst_object_unref (bussend);
+  gst_object_unref (binsend);
+
+  gst_bus_remove_signal_watch (busreceive);
+  gst_object_unref (busreceive);
+  gst_object_unref (binreceive);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtpaux_suite (void)
+{
+  Suite *s = suite_create ("rtpaux");
+  TCase *tc_chain = tcase_create ("general");
+
+  tcase_set_timeout (tc_chain, 10000);
+
+  suite_add_tcase (s, tc_chain);
+
+  tcase_add_test (tc_chain, test_simple_rtpbin_aux);
+
+  return s;
+}
+
+GST_CHECK_MAIN (rtpaux);