tests/check: add rtprtx::test_push_forward_seq

add simple unit test that manually push buffers
in rtprtxsend connected to rtprtxreceive.
Drops some buffers and make sure they are retransmisted.

tests/check/Makefile.am

@@ -149,6 +149,7 @@ check_PROGRAMS = \
 	elements/rtpbin_buffer_list \
 	elements/rtpjitterbuffer \
 	elements/rtpmux \
+	elements/rtprtx \
 	elements/shapewipe \
 	elements/spectrum \
 	elements/udpsink \
@@ -324,6 +325,9 @@ elements_videofilter_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstvideo-$(GST_API_VERSI
 elements_rtpjitterbuffer_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
 elements_rtpjitterbuffer_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)
 
+elements_rtprtx_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
+elements_rtprtx_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)
+
 elements_rtpsession_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
 elements_rtpsession_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)
 

tests/check/elements/.gitignore

@@ -53,6 +53,7 @@ rtpcollision
 rtpjitterbuffer
 rtpsession
 rtpmux
+rtprtx
 shapewipe
 souphttpsrc
 spectrum

tests/check/elements/rtprtx.c

@@ -0,0 +1,266 @@
+/* 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>
+
+/* For ease of programming we use globals to keep refs for our floating
+ * src and sink pads we create; otherwise we always have to do get_pad,
+ * get_peer, and then remove references in every test function */
+static GstPad *srcpad, *sinkpad;
+/* we also have a list of src buffers */
+static GList *inbuffers = NULL;
+
+#define RTP_CAPS_STRING    \
+    "application/x-rtp, "               \
+    "media = (string)audio, "           \
+    "payload = (int) 0, "               \
+    "clock-rate = (int) 8000, "         \
+    "encoding-name = (string)PCMU"
+
+#define RTP_FRAME_SIZE 20
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp")
+    );
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp")
+    );
+
+static void
+setup_rtprtx (GstElement * rtprtxsend, GstElement * rtprtxreceive,
+    gint num_buffers)
+{
+  GstClock *clock;
+  GstBuffer *buffer;
+  GstPad *sendsrcpad;
+  GstPad *receivesinkpad;
+  gboolean ret = FALSE;
+
+  /* a 20 sample audio block (2,5 ms) generated with
+   * gst-launch audiotestsrc wave=silence blocksize=40 num-buffers=3 !
+   *    "audio/x-raw,channels=1,rate=8000" ! mulawenc ! rtppcmupay !
+   *     fakesink dump=1
+   */
+  guint8 in[] = {               /* first 4 bytes are rtp-header, next 4 bytes are timestamp */
+    0x80, 0x80, 0x1c, 0x24, 0x46, 0xcd, 0xb7, 0x11, 0x3c, 0x3a, 0x7c, 0x5b,
+    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
+    0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff
+  };
+  GstClockTime ts = G_GUINT64_CONSTANT (0);
+  GstClockTime tso = gst_util_uint64_scale (RTP_FRAME_SIZE, GST_SECOND, 8000);
+  gint i;
+
+  /* we need a clock here */
+  clock = gst_system_clock_obtain ();
+  gst_element_set_clock (rtprtxsend, clock);
+  gst_object_unref (clock);
+
+  srcpad = gst_check_setup_src_pad (rtprtxsend, &srctemplate);
+  sendsrcpad = gst_element_get_static_pad (rtprtxsend, "src");
+  ret = gst_pad_set_active (srcpad, TRUE);
+  fail_if (ret == FALSE);
+
+  sinkpad = gst_check_setup_sink_pad (rtprtxreceive, &sinktemplate);
+  receivesinkpad = gst_element_get_static_pad (rtprtxreceive, "sink");
+  ret = gst_pad_set_active (sinkpad, TRUE);
+  fail_if (ret == FALSE);
+
+  fail_if (gst_pad_link (sendsrcpad, receivesinkpad) != GST_PAD_LINK_OK);
+
+  ret = gst_pad_set_active (sendsrcpad, TRUE);
+  fail_if (ret == FALSE);
+  ret = gst_pad_set_active (receivesinkpad, TRUE);
+  fail_if (ret == FALSE);
+
+  gst_object_unref (sendsrcpad);
+  gst_object_unref (receivesinkpad);
+
+  for (i = 0; i < num_buffers; i++) {
+    buffer = gst_buffer_new_and_alloc (sizeof (in));
+    gst_buffer_fill (buffer, 0, in, sizeof (in));
+    GST_BUFFER_DTS (buffer) = ts;
+    GST_BUFFER_PTS (buffer) = ts;
+    GST_BUFFER_DURATION (buffer) = tso;
+    GST_DEBUG ("created buffer: %p", buffer);
+
+    /*if (!i)
+       GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT); */
+
+    inbuffers = g_list_append (inbuffers, buffer);
+
+    /* hackish way to update the rtp header */
+    in[1] = 0x00;
+    in[3]++;                    /* seqnumber */
+    in[7] += RTP_FRAME_SIZE;    /* inc. timestamp with framesize */
+    ts += tso;
+  }
+}
+
+static GstStateChangeReturn
+start_rtprtx (GstElement * element)
+{
+  GstStateChangeReturn ret;
+  GstClockTime now;
+  GstClock *clock;
+
+  clock = gst_element_get_clock (element);
+  if (clock) {
+    now = gst_clock_get_time (clock);
+    gst_object_unref (clock);
+    gst_element_set_base_time (element, now);
+  }
+
+  ret = gst_element_set_state (element, GST_STATE_PLAYING);
+  ck_assert_int_ne (ret, GST_STATE_CHANGE_FAILURE);
+
+  ret = gst_element_get_state (element, NULL, NULL, GST_CLOCK_TIME_NONE);
+  ck_assert_int_ne (ret, GST_STATE_CHANGE_FAILURE);
+
+  return ret;
+}
+
+static void
+cleanup_rtprtx (GstElement * rtprtxsend, GstElement * rtprtxreceive)
+{
+  GST_DEBUG ("cleanup_rtprtx");
+
+  g_list_free (inbuffers);
+  inbuffers = NULL;
+
+  gst_pad_set_active (srcpad, FALSE);
+  gst_check_teardown_src_pad (rtprtxsend);
+  gst_check_teardown_element (rtprtxsend);
+
+  gst_pad_set_active (sinkpad, FALSE);
+  gst_check_teardown_sink_pad (rtprtxreceive);
+  gst_check_teardown_element (rtprtxreceive);
+}
+
+static void
+check_rtprtx_results (GstElement * rtprtxsend, GstElement * rtprtxreceive,
+    gint num_buffers)
+{
+  guint nbrtxrequests = 0;
+  guint nbrtxpackets = 0;
+
+  g_object_get (G_OBJECT (rtprtxsend), "num-rtx-requests", &nbrtxrequests,
+      NULL);
+  fail_unless_equals_int (nbrtxrequests, 3);
+
+  g_object_get (G_OBJECT (rtprtxsend), "num-rtx-packets", &nbrtxpackets, NULL);
+  fail_unless_equals_int (nbrtxpackets, 3);
+
+  g_object_get (G_OBJECT (rtprtxreceive), "num-rtx-requests", &nbrtxrequests,
+      NULL);
+  fail_unless_equals_int (nbrtxrequests, 3);
+
+  g_object_get (G_OBJECT (rtprtxreceive), "num-rtx-packets", &nbrtxpackets,
+      NULL);
+  fail_unless_equals_int (nbrtxpackets, 3);
+
+  g_object_get (G_OBJECT (rtprtxreceive), "num-rtx-assoc-packets",
+      &nbrtxpackets, NULL);
+  fail_unless_equals_int (nbrtxpackets, 3);
+}
+
+
+GST_START_TEST (test_push_forward_seq)
+{
+  GstElement *rtprtxsend;
+  GstElement *rtprtxreceive;
+  const guint num_buffers = 4;
+  GList *node;
+  gint i = 0;
+  GstCaps *caps = NULL;
+
+  rtprtxsend = gst_check_setup_element ("rtprtxsend");
+  rtprtxreceive = gst_check_setup_element ("rtprtxreceive");
+  setup_rtprtx (rtprtxsend, rtprtxreceive, num_buffers);
+  GST_DEBUG ("setup_rtprtx");
+
+  fail_unless (start_rtprtx (rtprtxsend)
+      == GST_STATE_CHANGE_SUCCESS, "could not set to playing");
+
+  fail_unless (start_rtprtx (rtprtxreceive)
+      == GST_STATE_CHANGE_SUCCESS, "could not set to playing");
+
+  caps = gst_caps_from_string (RTP_CAPS_STRING);
+  gst_check_setup_events (srcpad, rtprtxsend, caps, GST_FORMAT_TIME);
+  gst_caps_unref (caps);
+
+  g_object_set (rtprtxsend, "rtx-payload-type", 97, NULL);
+  g_object_set (rtprtxreceive, "rtx-payload-types", "97", NULL);
+
+  /* push buffers: 0,1,2, */
+  for (node = inbuffers; node; node = g_list_next (node)) {
+    GstEvent *event = NULL;
+    GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
+    GstBuffer *buffer = (GstBuffer *) node->data;
+    fail_unless (gst_pad_push (srcpad, buffer) == GST_FLOW_OK);
+
+    if (i < 3) {
+      gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
+
+      event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
+          gst_structure_new ("GstRTPRetransmissionRequest",
+              "seqnum", G_TYPE_UINT, (guint) gst_rtp_buffer_get_seq (&rtp),
+              "ssrc", G_TYPE_UINT, (guint) gst_rtp_buffer_get_ssrc (&rtp),
+              "payload-type", G_TYPE_UINT,
+              (guint) gst_rtp_buffer_get_payload_type (&rtp), NULL));
+
+      fail_unless (gst_pad_push_event (sinkpad, event));
+      gst_rtp_buffer_unmap (&rtp);
+    }
+    gst_buffer_unref (buffer);
+    ++i;
+  }
+
+  /* check the buffer list */
+  check_rtprtx_results (rtprtxsend, rtprtxreceive, num_buffers);
+
+  /* cleanup */
+  cleanup_rtprtx (rtprtxsend, rtprtxreceive);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtprtx_suite (void)
+{
+  Suite *s = suite_create ("rtprtx");
+  TCase *tc_chain = tcase_create ("general");
+
+  suite_add_tcase (s, tc_chain);
+
+  tcase_add_test (tc_chain, test_push_forward_seq);
+
+  return s;
+}
+
+GST_CHECK_MAIN (rtprtx);