--- /dev/null
+/* 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);
projects / platform / upstream / gst-plugins-good.git / commitdiff