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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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check: add rtpsession test
Add a basic rtpsession test to ensure that RR blocks are generated when multiple SSRC senders are active. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711270
This commit is contained in:
Wim Taymans
parent
49d52a64d6
commit
3623ebf01e
3 changed files with 284 additions and 2 deletions
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@ -144,6 +144,7 @@ check_PROGRAMS = \
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elements/rgvolume \
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elements/rtp-payloading \
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elements/rtpbin \
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elements/rtpsession \
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elements/rtpbin_buffer_list \
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elements/rtpjitterbuffer \
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elements/rtpmux \
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@ -293,8 +294,8 @@ elements_qtmux_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstpbutils-@GST_API_VERSION@ \
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elements_rtpbin_buffer_list_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) \
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$(WARNING_CFLAGS) $(ERROR_CFLAGS) $(GST_CHECK_CFLAGS) $(AM_CFLAGS)
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elements_rtpbin_buffer_list_LDADD = $(GST_PLUGINS_BASE_LIBS) \
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-lgstrtp-@GST_API_VERSION@ \
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$(GST_BASE_LIBS) $(GST_LIBS) $(GST_CHECK_LIBS)
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-lgstrtp-$(GST_API_VERSION) \
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$(GST_BASE_LIBS) $(GST_LIBS) $(GST_CHECK_LIBS) $(LDADD)
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elements_rtpbin_buffer_list_SOURCES = elements/rtpbin_buffer_list.c
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elements_rtpmux_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
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@ -322,6 +323,9 @@ elements_videofilter_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstvideo-$(GST_API_VERSI
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elements_rtpjitterbuffer_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
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elements_rtpjitterbuffer_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)
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elements_rtpsession_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
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elements_rtpsession_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)
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# FIXME: configure should check for gdk-pixbuf not gtk
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# only need video.h header, not the lib
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elements_gdkpixbufsink_CFLAGS = \
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1
tests/check/elements/.gitignore
vendored
1
tests/check/elements/.gitignore
vendored
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@ -50,6 +50,7 @@ rtp-payloading
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rtpbin
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rtpbin_buffer_list
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rtpjitterbuffer
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rtpsession
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rtpmux
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shapewipe
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souphttpsrc
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277
tests/check/elements/rtpsession.c
Normal file
277
tests/check/elements/rtpsession.c
Normal file
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@ -0,0 +1,277 @@
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/* 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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*
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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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#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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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)
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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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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, "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, "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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gst_pad_set_caps (data->rtcp_sink, generate_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, generate_caps ());
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gst_pad_push_event (data->src, gst_event_new_segment (&seg));
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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;
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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);
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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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gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (data.clock));
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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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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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}
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gst_test_clock_set_time (GST_TEST_CLOCK (data.clock),
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gst_clock_id_get_time (id) + (2 * GST_SECOND));
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gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (data.clock), &id);
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gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (data.clock));
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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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destroy_testharness (&data);
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}
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GST_END_TEST;
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static Suite *
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gstrtpsession_suite (void)
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{
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Suite *s = suite_create ("rtpsession");
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TCase *tc_chain = tcase_create ("general");
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suite_add_tcase (s, tc_chain);
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tcase_add_test (tc_chain, test_multiple_ssrc_rr);
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return s;
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}
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int
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main (int argc, char **argv)
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{
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int nf;
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Suite *s = gstrtpsession_suite ();
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SRunner *sr = srunner_create (s);
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gst_check_init (&argc, &argv);
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srunner_run_all (sr, CK_NORMAL);
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nf = srunner_ntests_failed (sr);
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srunner_free (sr);
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return nf;
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}
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