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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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ee7e80d615
Some endpoints (like Tandberg E20) can send BYE packet containing our internal SSRC. I this case we would detect SSRC collision and get rid of the source at some point. But because we are still sending packets with that SSRC the source will be recreated immediately. This brand new internal source will not have some variables incorrectly set in its state. For example 'seqnum-base` and `clock-rate` values will be -1. The fix is not to act on BYE RTCP if it contains internal or unknown SSRC. https://bugzilla.gnome.org/show_bug.cgi?id=762219
827 lines
27 KiB
C
827 lines
27 KiB
C
/* GStreamer
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*
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* unit test for gstrtpsession
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*
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* Copyright (C) <2009> Wim Taymans <wim.taymans@gmail.com>
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* Copyright (C) 2013 Collabora Ltd.
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#define GLIB_DISABLE_DEPRECATION_WARNINGS
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#include <gst/check/gstharness.h>
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#include <gst/check/gstcheck.h>
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#include <gst/check/gsttestclock.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/rtp/gstrtcpbuffer.h>
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#include <gst/net/gstnetaddressmeta.h>
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static const guint payload_size = 160;
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static const guint clock_rate = 8000;
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static const guint payload_type = 0;
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typedef struct
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{
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GstElement *session;
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GstPad *src, *rtcp_sink, *rtpsrc;
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GstClock *clock;
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GAsyncQueue *rtcp_queue;
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} TestData;
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static GstCaps *
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generate_caps (void)
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{
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return gst_caps_new_simple ("application/x-rtp",
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"clock-rate", G_TYPE_INT, clock_rate,
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"payload-type", G_TYPE_INT, payload_type, NULL);
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}
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static GstBuffer *
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generate_test_buffer (GstClockTime gst_ts,
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gboolean marker_bit, guint seq_num, guint32 rtp_ts, guint ssrc)
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{
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GstBuffer *buf;
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guint8 *payload;
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guint i;
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GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
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buf = gst_rtp_buffer_new_allocate (payload_size, 0, 0);
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GST_BUFFER_DTS (buf) = gst_ts;
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GST_BUFFER_PTS (buf) = gst_ts;
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gst_rtp_buffer_map (buf, GST_MAP_READWRITE, &rtp);
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gst_rtp_buffer_set_payload_type (&rtp, payload_type);
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gst_rtp_buffer_set_marker (&rtp, marker_bit);
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gst_rtp_buffer_set_seq (&rtp, seq_num);
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gst_rtp_buffer_set_timestamp (&rtp, rtp_ts);
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gst_rtp_buffer_set_ssrc (&rtp, ssrc);
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payload = gst_rtp_buffer_get_payload (&rtp);
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for (i = 0; i < payload_size; i++)
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payload[i] = 0xff;
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gst_rtp_buffer_unmap (&rtp);
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return buf;
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}
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static GstFlowReturn
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test_sink_pad_chain_cb (GstPad * pad, GstObject * parent, GstBuffer * buffer)
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{
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TestData *data = gst_pad_get_element_private (pad);
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g_async_queue_push (data->rtcp_queue, buffer);
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GST_DEBUG ("chained");
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return GST_FLOW_OK;
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}
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static GstCaps *
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pt_map_requested (GstElement * elemen, guint pt, gpointer data)
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{
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return generate_caps ();
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}
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static void
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destroy_testharness (TestData * data)
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{
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g_assert_cmpint (gst_element_set_state (data->session, GST_STATE_NULL),
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==, GST_STATE_CHANGE_SUCCESS);
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gst_object_unref (data->session);
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data->session = NULL;
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gst_object_unref (data->src);
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data->src = NULL;
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gst_object_unref (data->rtcp_sink);
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data->rtcp_sink = NULL;
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gst_object_unref (data->rtpsrc);
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data->rtpsrc = NULL;
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gst_object_unref (data->clock);
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data->clock = NULL;
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g_async_queue_unref (data->rtcp_queue);
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data->rtcp_queue = NULL;
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}
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static void
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setup_testharness (TestData * data, gboolean session_as_sender)
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{
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GstPad *rtp_sink_pad, *rtcp_src_pad, *rtp_src_pad;
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GstSegment seg;
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GstMiniObject *obj;
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GstCaps *caps;
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data->clock = gst_test_clock_new ();
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GST_DEBUG ("Setting default system clock to test clock");
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gst_system_clock_set_default (data->clock);
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g_assert (data->clock);
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gst_test_clock_set_time (GST_TEST_CLOCK (data->clock), 0);
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data->session = gst_element_factory_make ("rtpsession", NULL);
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g_signal_connect (data->session, "request-pt-map",
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(GCallback) pt_map_requested, data);
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g_assert (data->session);
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gst_element_set_clock (data->session, data->clock);
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g_assert_cmpint (gst_element_set_state (data->session,
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GST_STATE_PLAYING), !=, GST_STATE_CHANGE_FAILURE);
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data->rtcp_queue =
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g_async_queue_new_full ((GDestroyNotify) gst_mini_object_unref);
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/* link in the test source-pad */
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data->src = gst_pad_new ("src", GST_PAD_SRC);
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g_assert (data->src);
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rtp_sink_pad = gst_element_get_request_pad (data->session,
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session_as_sender ? "send_rtp_sink" : "recv_rtp_sink");
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g_assert (rtp_sink_pad);
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g_assert_cmpint (gst_pad_link (data->src, rtp_sink_pad), ==, GST_PAD_LINK_OK);
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gst_object_unref (rtp_sink_pad);
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data->rtpsrc = gst_pad_new ("sink", GST_PAD_SINK);
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g_assert (data->rtpsrc);
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rtp_src_pad = gst_element_get_static_pad (data->session,
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session_as_sender ? "send_rtp_src" : "recv_rtp_src");
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g_assert (rtp_src_pad);
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g_assert_cmpint (gst_pad_link (rtp_src_pad, data->rtpsrc), ==,
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GST_PAD_LINK_OK);
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gst_object_unref (rtp_src_pad);
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/* link in the test sink-pad */
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data->rtcp_sink = gst_pad_new ("sink", GST_PAD_SINK);
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g_assert (data->rtcp_sink);
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gst_pad_set_element_private (data->rtcp_sink, data);
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caps = generate_caps ();
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gst_pad_set_caps (data->rtcp_sink, caps);
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gst_pad_set_chain_function (data->rtcp_sink, test_sink_pad_chain_cb);
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rtcp_src_pad = gst_element_get_request_pad (data->session, "send_rtcp_src");
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g_assert (rtcp_src_pad);
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g_assert_cmpint (gst_pad_link (rtcp_src_pad, data->rtcp_sink), ==,
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GST_PAD_LINK_OK);
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gst_object_unref (rtcp_src_pad);
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g_assert (gst_pad_set_active (data->src, TRUE));
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g_assert (gst_pad_set_active (data->rtcp_sink, TRUE));
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gst_segment_init (&seg, GST_FORMAT_TIME);
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gst_pad_push_event (data->src, gst_event_new_stream_start ("stream0"));
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gst_pad_set_caps (data->src, caps);
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gst_pad_push_event (data->src, gst_event_new_segment (&seg));
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gst_caps_unref (caps);
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while ((obj = g_async_queue_try_pop (data->rtcp_queue)))
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gst_mini_object_unref (obj);
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}
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GST_START_TEST (test_multiple_ssrc_rr)
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{
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TestData data;
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GstFlowReturn res;
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GstClockID id, tid;
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GstBuffer *in_buf, *out_buf;
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GstRTCPBuffer rtcp = GST_RTCP_BUFFER_INIT;
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GstRTCPPacket rtcp_packet;
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int i;
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guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
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gint32 packetslost;
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guint8 fractionlost;
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setup_testharness (&data, FALSE);
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gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), 10 * GST_MSECOND);
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for (i = 0; i < 5; i++) {
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GST_DEBUG ("Push %i", i);
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in_buf =
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generate_test_buffer (i * 20 * GST_MSECOND, FALSE, i, i * 20,
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0x01BADBAD);
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res = gst_pad_push (data.src, in_buf);
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fail_unless (res == GST_FLOW_OK || res == GST_FLOW_FLUSHING);
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gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data.clock), &id);
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tid = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (data.clock));
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gst_clock_id_unref (id);
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if (tid)
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gst_clock_id_unref (tid);
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in_buf =
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generate_test_buffer (i * 20 * GST_MSECOND, FALSE, i, i * 20,
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0xDEADBEEF);
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res = gst_pad_push (data.src, in_buf);
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fail_unless (res == GST_FLOW_OK || res == GST_FLOW_FLUSHING);
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gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data.clock), &id);
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tid = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (data.clock));
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GST_DEBUG ("pushed %i", i);
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gst_test_clock_set_time (GST_TEST_CLOCK (data.clock),
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gst_clock_id_get_time (id));
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gst_clock_id_unref (id);
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if (tid)
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gst_clock_id_unref (tid);
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}
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out_buf = g_async_queue_try_pop (data.rtcp_queue);
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if (out_buf)
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gst_buffer_unref (out_buf);
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gst_test_clock_set_time (GST_TEST_CLOCK (data.clock),
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gst_clock_id_get_time (id) + (5 * GST_SECOND));
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gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data.clock), &id);
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tid = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (data.clock));
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gst_clock_id_unref (id);
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gst_clock_id_unref (tid);
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out_buf = g_async_queue_pop (data.rtcp_queue);
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g_assert (out_buf != NULL);
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g_assert (gst_rtcp_buffer_validate (out_buf));
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gst_rtcp_buffer_map (out_buf, GST_MAP_READ, &rtcp);
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g_assert (gst_rtcp_buffer_get_first_packet (&rtcp, &rtcp_packet));
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g_assert (gst_rtcp_packet_get_type (&rtcp_packet) == GST_RTCP_TYPE_RR);
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g_assert_cmpint (gst_rtcp_packet_get_rb_count (&rtcp_packet), ==, 2);
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gst_rtcp_packet_get_rb (&rtcp_packet, 0, &ssrc, &fractionlost, &packetslost,
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&exthighestseq, &jitter, &lsr, &dlsr);
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g_assert_cmpint (ssrc, ==, 0x01BADBAD);
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gst_rtcp_packet_get_rb (&rtcp_packet, 1, &ssrc, &fractionlost, &packetslost,
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&exthighestseq, &jitter, &lsr, &dlsr);
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g_assert_cmpint (ssrc, ==, 0xDEADBEEF);
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gst_rtcp_buffer_unmap (&rtcp);
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gst_buffer_unref (out_buf);
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destroy_testharness (&data);
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}
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GST_END_TEST;
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/* This verifies that rtpsession will correctly place RBs round-robin
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* across multiple SRs when there are too many senders that their RBs
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* do not fit in one SR */
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GST_START_TEST (test_multiple_senders_roundrobin_rbs)
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{
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TestData data;
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GstFlowReturn res;
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GstClockID id, tid;
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GstBuffer *buf;
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GstRTCPBuffer rtcp = GST_RTCP_BUFFER_INIT;
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GstRTCPPacket rtcp_packet;
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GstClockTime time;
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gint queue_length;
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gint i, j, k;
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guint32 ssrc;
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GHashTable *sr_ssrcs, *rb_ssrcs, *tmp_set;
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setup_testharness (&data, TRUE);
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/* only the RTCP thread waits on the clock */
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gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data.clock), &id);
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for (i = 0; i < 2; i++) { /* cycles between SR reports */
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for (j = 0; j < 5; j++) { /* packets per ssrc */
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gint seq = (i * 5) + j;
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GST_DEBUG ("Push %i", seq);
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gst_test_clock_advance_time (GST_TEST_CLOCK (data.clock),
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200 * GST_MSECOND);
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for (k = 0; k < 35; k++) { /* number of ssrcs */
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buf =
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generate_test_buffer (seq * 200 * GST_MSECOND, FALSE, seq,
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seq * 200, 10000 + k);
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res = gst_pad_push (data.src, buf);
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fail_unless (res == GST_FLOW_OK || res == GST_FLOW_FLUSHING);
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}
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GST_DEBUG ("pushed %i", seq);
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}
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queue_length = g_async_queue_length (data.rtcp_queue);
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do {
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/* crank the RTCP pad thread */
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time = gst_clock_id_get_time (id);
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GST_DEBUG ("Advancing time to %" GST_TIME_FORMAT, GST_TIME_ARGS (time));
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gst_test_clock_set_time (GST_TEST_CLOCK (data.clock), time);
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tid = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (data.clock));
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fail_unless_equals_pointer (tid, id);
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gst_clock_id_unref (id);
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gst_clock_id_unref (tid);
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/* wait for the RTCP pad thread to output its data
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* and start waiting on the next timeout */
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gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data.clock),
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&id);
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/* and retry as long as there are no new RTCP packets out,
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* because the RTCP thread may randomly decide to reschedule
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* the RTCP timeout for later */
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} while (g_async_queue_length (data.rtcp_queue) == queue_length);
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GST_DEBUG ("RTCP timeout processed");
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}
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gst_clock_id_unref (id);
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sr_ssrcs = g_hash_table_new (g_direct_hash, g_direct_equal);
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rb_ssrcs = g_hash_table_new_full (g_direct_hash, g_direct_equal, NULL,
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(GDestroyNotify) g_hash_table_unref);
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/* verify the rtcp packets */
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for (i = 0; i < 2 * 35; i++) {
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guint expected_rb_count = (i < 35) ? GST_RTCP_MAX_RB_COUNT :
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(35 - GST_RTCP_MAX_RB_COUNT - 1);
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GST_DEBUG ("pop %d", i);
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buf = g_async_queue_pop (data.rtcp_queue);
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g_assert (buf != NULL);
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g_assert (gst_rtcp_buffer_validate (buf));
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gst_rtcp_buffer_map (buf, GST_MAP_READ, &rtcp);
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g_assert (gst_rtcp_buffer_get_first_packet (&rtcp, &rtcp_packet));
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g_assert_cmpint (gst_rtcp_packet_get_type (&rtcp_packet), ==,
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GST_RTCP_TYPE_SR);
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gst_rtcp_packet_sr_get_sender_info (&rtcp_packet, &ssrc, NULL, NULL, NULL,
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NULL);
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g_assert_cmpint (ssrc, >=, 10000);
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g_assert_cmpint (ssrc, <=, 10035);
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g_hash_table_add (sr_ssrcs, GUINT_TO_POINTER (ssrc));
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/* inspect the RBs */
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g_assert_cmpint (gst_rtcp_packet_get_rb_count (&rtcp_packet), ==,
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expected_rb_count);
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if (i < 35) {
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tmp_set = g_hash_table_new (g_direct_hash, g_direct_equal);
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g_hash_table_insert (rb_ssrcs, GUINT_TO_POINTER (ssrc), tmp_set);
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} else {
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tmp_set = g_hash_table_lookup (rb_ssrcs, GUINT_TO_POINTER (ssrc));
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g_assert (tmp_set);
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}
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for (j = 0; j < expected_rb_count; j++) {
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gst_rtcp_packet_get_rb (&rtcp_packet, j, &ssrc, NULL, NULL,
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NULL, NULL, NULL, NULL);
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g_assert_cmpint (ssrc, >=, 10000);
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g_assert_cmpint (ssrc, <=, 10035);
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g_hash_table_add (tmp_set, GUINT_TO_POINTER (ssrc));
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}
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gst_rtcp_buffer_unmap (&rtcp);
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gst_buffer_unref (buf);
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/* cycle done, verify all ssrcs have issued SR reports */
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if ((i + 1) == 35 || (i + 1) == (2 * 35)) {
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g_assert_cmpint (g_hash_table_size (sr_ssrcs), ==, 35);
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g_hash_table_remove_all (sr_ssrcs);
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}
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}
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/* now verify all other ssrcs have been reported on each ssrc's SR */
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g_assert_cmpint (g_hash_table_size (rb_ssrcs), ==, 35);
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for (i = 10000; i < 10035; i++) {
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tmp_set = g_hash_table_lookup (rb_ssrcs, GUINT_TO_POINTER (i));
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g_assert (tmp_set);
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/* SR contains RBs for each other ssrc except the ssrc of the SR */
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g_assert_cmpint (g_hash_table_size (tmp_set), ==, 34);
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g_assert (!g_hash_table_contains (tmp_set, GUINT_TO_POINTER (i)));
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}
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g_hash_table_unref (sr_ssrcs);
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g_hash_table_unref (rb_ssrcs);
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destroy_testharness (&data);
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}
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GST_END_TEST;
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static void
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crank_rtcp_thread (TestData * data, GstClockTime * time, GstClockID * id)
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{
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gint queue_length;
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GstClockID *tid;
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|
|
queue_length = g_async_queue_length (data->rtcp_queue);
|
|
do {
|
|
*time = gst_clock_id_get_time (*id);
|
|
GST_DEBUG ("Advancing time to %" GST_TIME_FORMAT, GST_TIME_ARGS (*time));
|
|
if (*time > gst_clock_get_time (data->clock))
|
|
gst_test_clock_set_time (GST_TEST_CLOCK (data->clock), *time);
|
|
tid = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (data->clock));
|
|
fail_unless_equals_pointer (tid, *id);
|
|
|
|
gst_clock_id_unref (tid);
|
|
gst_clock_id_unref (*id);
|
|
*id = NULL;
|
|
|
|
/* wait for the RTCP pad thread to output its data
|
|
* and start waiting on the next timeout */
|
|
gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data->clock), id);
|
|
|
|
/* and retry as long as there are no new RTCP packets out,
|
|
* because the RTCP thread may randomly decide to reschedule
|
|
* the RTCP timeout for later */
|
|
} while (g_async_queue_length (data->rtcp_queue) == queue_length);
|
|
}
|
|
|
|
GST_START_TEST (test_internal_sources_timeout)
|
|
{
|
|
TestData data;
|
|
GstClockID id;
|
|
GstClockTime time;
|
|
GObject *internal_session;
|
|
guint internal_ssrc;
|
|
guint32 ssrc;
|
|
GstBuffer *buf;
|
|
GstRTCPBuffer rtcp = GST_RTCP_BUFFER_INIT;
|
|
GstRTCPPacket rtcp_packet;
|
|
GstFlowReturn res;
|
|
gint i, j;
|
|
GstCaps *caps;
|
|
|
|
setup_testharness (&data, TRUE);
|
|
g_object_get (data.session, "internal-session", &internal_session, NULL);
|
|
g_object_set (internal_session, "internal-ssrc", 0xDEADBEEF, NULL);
|
|
|
|
/* only the RTCP thread waits on the clock */
|
|
gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data.clock), &id);
|
|
|
|
/* crank the RTCP pad thread until it creates a RR for its internal-ssrc
|
|
* source, since we have not pushed any RTP packets and it doesn't have
|
|
* any other source available */
|
|
crank_rtcp_thread (&data, &time, &id);
|
|
|
|
g_object_get (internal_session, "internal-ssrc", &internal_ssrc, NULL);
|
|
g_assert_cmpint (internal_ssrc, ==, 0xDEADBEEF);
|
|
|
|
/* verify that rtpsession has sent RR for an internally-created
|
|
* RTPSource that is using the internal-ssrc */
|
|
buf = g_async_queue_pop (data.rtcp_queue);
|
|
g_assert (buf != NULL);
|
|
g_assert (gst_rtcp_buffer_validate (buf));
|
|
gst_rtcp_buffer_map (buf, GST_MAP_READ, &rtcp);
|
|
g_assert (gst_rtcp_buffer_get_first_packet (&rtcp, &rtcp_packet));
|
|
g_assert_cmpint (gst_rtcp_packet_get_type (&rtcp_packet), ==,
|
|
GST_RTCP_TYPE_RR);
|
|
ssrc = gst_rtcp_packet_rr_get_ssrc (&rtcp_packet);
|
|
g_assert_cmpint (ssrc, ==, internal_ssrc);
|
|
gst_rtcp_buffer_unmap (&rtcp);
|
|
gst_buffer_unref (buf);
|
|
|
|
/* ok, now let's push some RTP packets */
|
|
caps =
|
|
gst_caps_new_simple ("application/x-rtp", "ssrc", G_TYPE_UINT, 0x01BADBAD,
|
|
NULL);
|
|
gst_pad_set_caps (data.src, caps);
|
|
gst_caps_unref (caps);
|
|
|
|
for (i = 1; i < 4; i++) {
|
|
gst_test_clock_advance_time (GST_TEST_CLOCK (data.clock),
|
|
200 * GST_MSECOND);
|
|
buf =
|
|
generate_test_buffer (time + i * 200 * GST_MSECOND, FALSE, i, i * 200,
|
|
0x01BADBAD);
|
|
res = gst_pad_push (data.src, buf);
|
|
fail_unless (res == GST_FLOW_OK || res == GST_FLOW_FLUSHING);
|
|
}
|
|
|
|
/* internal ssrc must have changed already */
|
|
g_object_get (internal_session, "internal-ssrc", &internal_ssrc, NULL);
|
|
g_assert_cmpint (ssrc, !=, internal_ssrc);
|
|
g_assert_cmpint (internal_ssrc, ==, 0x01BADBAD);
|
|
|
|
/* wait for SR */
|
|
crank_rtcp_thread (&data, &time, &id);
|
|
|
|
/* verify SR and RR */
|
|
j = 0;
|
|
for (i = 0; i < 2; i++) {
|
|
buf = g_async_queue_pop (data.rtcp_queue);
|
|
g_assert (buf != NULL);
|
|
g_assert (gst_rtcp_buffer_validate (buf));
|
|
gst_rtcp_buffer_map (buf, GST_MAP_READ, &rtcp);
|
|
g_assert (gst_rtcp_buffer_get_first_packet (&rtcp, &rtcp_packet));
|
|
if (gst_rtcp_packet_get_type (&rtcp_packet) == GST_RTCP_TYPE_SR) {
|
|
gst_rtcp_packet_sr_get_sender_info (&rtcp_packet, &ssrc, NULL, NULL, NULL,
|
|
NULL);
|
|
g_assert_cmpint (ssrc, ==, internal_ssrc);
|
|
g_assert_cmpint (ssrc, ==, 0x01BADBAD);
|
|
j |= 0x1;
|
|
} else if (gst_rtcp_packet_get_type (&rtcp_packet) == GST_RTCP_TYPE_RR) {
|
|
ssrc = gst_rtcp_packet_rr_get_ssrc (&rtcp_packet);
|
|
g_assert_cmpint (ssrc, !=, internal_ssrc);
|
|
g_assert_cmpint (ssrc, ==, 0xDEADBEEF);
|
|
j |= 0x2;
|
|
}
|
|
gst_rtcp_buffer_unmap (&rtcp);
|
|
gst_buffer_unref (buf);
|
|
}
|
|
g_assert_cmpint (j, ==, 0x3); /* verify we got both SR and RR */
|
|
|
|
/* go 30 seconds in the future and observe both sources timing out:
|
|
* 0xDEADBEEF -> BYE, 0x01BADBAD -> becomes receiver only */
|
|
gst_test_clock_advance_time (GST_TEST_CLOCK (data.clock), 30 * GST_SECOND);
|
|
crank_rtcp_thread (&data, &time, &id);
|
|
|
|
/* verify BYE and RR */
|
|
j = 0;
|
|
for (i = 0; i < 2; i++) {
|
|
buf = g_async_queue_pop (data.rtcp_queue);
|
|
g_assert (buf != NULL);
|
|
g_assert (gst_rtcp_buffer_validate (buf));
|
|
gst_rtcp_buffer_map (buf, GST_MAP_READ, &rtcp);
|
|
|
|
g_assert (gst_rtcp_buffer_get_first_packet (&rtcp, &rtcp_packet));
|
|
g_assert_cmpint (gst_rtcp_packet_get_type (&rtcp_packet), ==,
|
|
GST_RTCP_TYPE_RR);
|
|
ssrc = gst_rtcp_packet_rr_get_ssrc (&rtcp_packet);
|
|
if (ssrc == 0x01BADBAD) {
|
|
j |= 0x1;
|
|
g_assert_cmpint (ssrc, ==, internal_ssrc);
|
|
/* 2 => RR, SDES. There is no BYE here */
|
|
g_assert_cmpint (gst_rtcp_buffer_get_packet_count (&rtcp), ==, 2);
|
|
} else if (ssrc == 0xDEADBEEF) {
|
|
j |= 0x2;
|
|
g_assert_cmpint (ssrc, !=, internal_ssrc);
|
|
/* 3 => RR, SDES, BYE */
|
|
g_assert_cmpint (gst_rtcp_buffer_get_packet_count (&rtcp), ==, 3);
|
|
g_assert (gst_rtcp_packet_move_to_next (&rtcp_packet));
|
|
g_assert (gst_rtcp_packet_move_to_next (&rtcp_packet));
|
|
g_assert_cmpint (gst_rtcp_packet_get_type (&rtcp_packet), ==,
|
|
GST_RTCP_TYPE_BYE);
|
|
}
|
|
|
|
gst_rtcp_buffer_unmap (&rtcp);
|
|
gst_buffer_unref (buf);
|
|
}
|
|
g_assert_cmpint (j, ==, 0x3); /* verify we got both BYE and RR */
|
|
gst_clock_id_unref (id);
|
|
|
|
g_object_unref (internal_session);
|
|
destroy_testharness (&data);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
typedef struct
|
|
{
|
|
guint8 subtype;
|
|
guint32 ssrc;
|
|
gchar *name;
|
|
GstBuffer *data;
|
|
} RTCPAppResult;
|
|
|
|
static void
|
|
on_app_rtcp_cb (GObject * session, guint subtype, guint ssrc,
|
|
const gchar * name, GstBuffer * data, RTCPAppResult * result)
|
|
{
|
|
result->subtype = subtype;
|
|
result->ssrc = ssrc;
|
|
result->name = g_strdup (name);
|
|
result->data = data ? gst_buffer_ref (data) : NULL;
|
|
}
|
|
|
|
GST_START_TEST (test_receive_rtcp_app_packet)
|
|
{
|
|
GstHarness *h;
|
|
GstBuffer *buffer;
|
|
GstRTCPBuffer rtcp = GST_RTCP_BUFFER_INIT;
|
|
GstRTCPPacket packet;
|
|
RTCPAppResult result = { 0 };
|
|
GstElement *internal_session;
|
|
guint8 data[] = { 0x11, 0x22, 0x33, 0x44 };
|
|
|
|
h = gst_harness_new_with_padnames ("rtpsession", "recv_rtcp_sink", NULL);
|
|
g_object_get (h->element, "internal-session", &internal_session, NULL);
|
|
|
|
g_signal_connect (internal_session, "on-app-rtcp",
|
|
G_CALLBACK (on_app_rtcp_cb), &result);
|
|
|
|
/* Push APP buffer with no data */
|
|
buffer = gst_rtcp_buffer_new (1000);
|
|
fail_unless (gst_rtcp_buffer_map (buffer, GST_MAP_READWRITE, &rtcp));
|
|
fail_unless (gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_APP, &packet));
|
|
gst_rtcp_packet_app_set_subtype (&packet, 21);
|
|
gst_rtcp_packet_app_set_ssrc (&packet, 0x11111111);
|
|
gst_rtcp_packet_app_set_name (&packet, "Test");
|
|
gst_rtcp_buffer_unmap (&rtcp);
|
|
|
|
gst_harness_set_src_caps_str (h, "application/x-rtcp");
|
|
fail_unless_equals_int (gst_harness_push (h, buffer), GST_FLOW_OK);
|
|
|
|
fail_unless_equals_int (result.subtype, 21);
|
|
fail_unless_equals_int (result.ssrc, 0x11111111);
|
|
fail_unless_equals_string (result.name, "Test");
|
|
fail_unless_equals_pointer (result.data, NULL);
|
|
|
|
g_free (result.name);
|
|
|
|
/* Push APP buffer with data */
|
|
memset (&result, 0, sizeof (result));
|
|
buffer = gst_rtcp_buffer_new (1000);
|
|
fail_unless (gst_rtcp_buffer_map (buffer, GST_MAP_READWRITE, &rtcp));
|
|
fail_unless (gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_APP, &packet));
|
|
gst_rtcp_packet_app_set_subtype (&packet, 22);
|
|
gst_rtcp_packet_app_set_ssrc (&packet, 0x22222222);
|
|
gst_rtcp_packet_app_set_name (&packet, "Test");
|
|
gst_rtcp_packet_app_set_data_length (&packet, sizeof (data) / 4);
|
|
memcpy (gst_rtcp_packet_app_get_data (&packet), data, sizeof (data));
|
|
gst_rtcp_buffer_unmap (&rtcp);
|
|
|
|
fail_unless_equals_int (gst_harness_push (h, buffer), GST_FLOW_OK);
|
|
|
|
fail_unless_equals_int (result.subtype, 22);
|
|
fail_unless_equals_int (result.ssrc, 0x22222222);
|
|
fail_unless_equals_string (result.name, "Test");
|
|
fail_unless (gst_buffer_memcmp (result.data, 0, data, sizeof (data)) == 0);
|
|
|
|
g_free (result.name);
|
|
gst_buffer_unref (result.data);
|
|
|
|
gst_object_unref (internal_session);
|
|
gst_harness_teardown (h);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static void
|
|
stats_test_cb (GObject * object, GParamSpec * spec, gpointer data)
|
|
{
|
|
guint num_sources = 0;
|
|
gboolean *cb_called = data;
|
|
g_assert (*cb_called == FALSE);
|
|
*cb_called = TRUE;
|
|
|
|
/* We should be able to get a rtpsession property
|
|
without introducing the deadlock */
|
|
g_object_get (object, "num-sources", &num_sources, NULL);
|
|
}
|
|
|
|
GST_START_TEST (test_dont_lock_on_stats)
|
|
{
|
|
GstHarness * h_rtcp;
|
|
GstHarness * h_send;
|
|
GstClock * clock = gst_test_clock_new ();
|
|
GstTestClock * testclock = GST_TEST_CLOCK (clock);
|
|
gboolean cb_called = FALSE;
|
|
|
|
/* use testclock as the systemclock to capture the rtcp thread waits */
|
|
gst_system_clock_set_default (GST_CLOCK (testclock));
|
|
|
|
h_rtcp = gst_harness_new_with_padnames (
|
|
"rtpsession", "recv_rtcp_sink", "send_rtcp_src");
|
|
h_send = gst_harness_new_with_element (
|
|
h_rtcp->element, "send_rtp_sink", "send_rtp_src");
|
|
|
|
/* connect to the stats-reporting */
|
|
g_signal_connect (h_rtcp->element, "notify::stats",
|
|
G_CALLBACK (stats_test_cb), &cb_called);
|
|
|
|
/* "crank" and check the stats */
|
|
g_assert (gst_test_clock_crank (testclock));
|
|
gst_buffer_unref (gst_harness_pull (h_rtcp));
|
|
fail_unless (cb_called);
|
|
|
|
gst_harness_teardown (h_send);
|
|
gst_harness_teardown (h_rtcp);
|
|
gst_object_unref (clock);
|
|
}
|
|
GST_END_TEST;
|
|
|
|
static void
|
|
suspicious_bye_cb (GObject * object, GParamSpec * spec, gpointer data)
|
|
{
|
|
GValueArray *stats_arr;
|
|
GstStructure *stats, *internal_stats;
|
|
gboolean *cb_called = data;
|
|
gboolean internal = FALSE, sent_bye = TRUE;
|
|
guint ssrc = 0;
|
|
guint i;
|
|
|
|
g_assert (*cb_called == FALSE);
|
|
*cb_called = TRUE;
|
|
|
|
g_object_get (object, "stats", &stats, NULL);
|
|
stats_arr =
|
|
g_value_get_boxed (gst_structure_get_value (stats, "source-stats"));
|
|
g_assert (stats_arr != NULL);
|
|
fail_unless (stats_arr->n_values >= 1);
|
|
|
|
for (i = 0; i < stats_arr->n_values; i++) {
|
|
internal_stats = g_value_get_boxed (g_value_array_get_nth (stats_arr, i));
|
|
g_assert (internal_stats != NULL);
|
|
|
|
gst_structure_get (internal_stats,
|
|
"ssrc", G_TYPE_UINT, &ssrc,
|
|
"internal", G_TYPE_BOOLEAN, &internal,
|
|
"received-bye", G_TYPE_BOOLEAN, &sent_bye, NULL);
|
|
|
|
if (ssrc == 0xDEADBEEF) {
|
|
fail_unless (internal);
|
|
fail_unless (!sent_bye);
|
|
break;
|
|
}
|
|
}
|
|
fail_unless_equals_int (ssrc, 0xDEADBEEF);
|
|
|
|
gst_structure_free (stats);
|
|
}
|
|
|
|
static GstBuffer *
|
|
create_bye_rtcp (guint32 ssrc)
|
|
{
|
|
GstRTCPPacket packet;
|
|
GstRTCPBuffer rtcp = GST_RTCP_BUFFER_INIT;
|
|
GSocketAddress *saddr;
|
|
GstBuffer *buffer = gst_rtcp_buffer_new (1000);
|
|
|
|
fail_unless (gst_rtcp_buffer_map (buffer, GST_MAP_READWRITE, &rtcp));
|
|
fail_unless (gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_BYE, &packet));
|
|
gst_rtcp_packet_bye_add_ssrc (&packet, ssrc);
|
|
gst_rtcp_buffer_unmap (&rtcp);
|
|
|
|
/* Need to add meta to trigger collision detection */
|
|
saddr = g_inet_socket_address_new_from_string ("127.0.0.1", 3490);
|
|
gst_buffer_add_net_address_meta (buffer, saddr);
|
|
g_object_unref (saddr);
|
|
return buffer;
|
|
}
|
|
|
|
GST_START_TEST (test_ignore_suspicious_bye)
|
|
{
|
|
GstHarness *h_rtcp = NULL;
|
|
GstHarness *h_send = NULL;
|
|
gboolean cb_called = FALSE;
|
|
GstTestClock *testclock = GST_TEST_CLOCK (gst_test_clock_new ());
|
|
|
|
/* use testclock as the systemclock to capture the rtcp thread waits */
|
|
gst_system_clock_set_default (GST_CLOCK (testclock));
|
|
|
|
h_rtcp =
|
|
gst_harness_new_with_padnames ("rtpsession", "recv_rtcp_sink",
|
|
"send_rtcp_src");
|
|
h_send =
|
|
gst_harness_new_with_element (h_rtcp->element, "send_rtp_sink",
|
|
"send_rtp_src");
|
|
|
|
/* connect to the stats-reporting */
|
|
g_signal_connect (h_rtcp->element, "notify::stats",
|
|
G_CALLBACK (suspicious_bye_cb), &cb_called);
|
|
|
|
/* Push RTP buffer making our internal SSRC=0xDEADBEEF */
|
|
gst_harness_set_src_caps_str (h_send,
|
|
"application/x-rtp,ssrc=(uint)0xDEADBEEF,"
|
|
"clock-rate=90000,seqnum-offset=(uint)12345");
|
|
gst_harness_push (h_send,
|
|
generate_test_buffer (0, FALSE, 12345, 0, 0xDEADBEEF));
|
|
|
|
/* Push BYE RTCP with internal SSRC (0xDEADBEEF) */
|
|
gst_harness_set_src_caps_str (h_rtcp, "application/x-rtcp");
|
|
gst_harness_push (h_rtcp, create_bye_rtcp (0xDEADBEEF));
|
|
|
|
/* "crank" and check the stats */
|
|
g_assert (gst_test_clock_crank (testclock));
|
|
gst_buffer_unref (gst_harness_pull (h_rtcp));
|
|
fail_unless (cb_called);
|
|
|
|
gst_harness_teardown (h_send);
|
|
gst_harness_teardown (h_rtcp);
|
|
gst_object_unref (testclock);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static Suite *
|
|
rtpsession_suite (void)
|
|
{
|
|
Suite *s = suite_create ("rtpsession");
|
|
TCase *tc_chain = tcase_create ("general");
|
|
|
|
suite_add_tcase (s, tc_chain);
|
|
tcase_add_test (tc_chain, test_multiple_ssrc_rr);
|
|
tcase_add_test (tc_chain, test_multiple_senders_roundrobin_rbs);
|
|
tcase_add_test (tc_chain, test_internal_sources_timeout);
|
|
tcase_add_test (tc_chain, test_receive_rtcp_app_packet);
|
|
tcase_add_test (tc_chain, test_dont_lock_on_stats);
|
|
tcase_add_test (tc_chain, test_ignore_suspicious_bye);
|
|
|
|
return s;
|
|
}
|
|
|
|
GST_CHECK_MAIN (rtpsession);
|