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tests/check: add rtpcollision::test_master_ssrc_collision unit test
It checks the payloader changes its ssrc when collision happens
This commit is contained in:
Julien Isorce
Wim Taymans
parent
c78a115154
commit
7b001e35ed
3 changed files with 284 additions and 0 deletions
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@ -142,6 +142,7 @@ check_PROGRAMS = \
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elements/rganalysis \
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elements/rglimiter \
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elements/rgvolume \
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elements/rtpcollision \
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elements/rtp-payloading \
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elements/rtpbin \
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elements/rtpsession \
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@ -326,6 +327,9 @@ elements_rtpjitterbuffer_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VER
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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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elements_rtpcollision_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
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elements_rtpcollision_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstnet-$(GST_API_VERSION) -lgstrtp-$(GST_API_VERSION) $(GIO_LIBS) $(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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@ -49,6 +49,7 @@ rgvolume
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rtp-payloading
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rtpbin
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rtpbin_buffer_list
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rtpcollision
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rtpjitterbuffer
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rtpsession
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rtpmux
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279
tests/check/elements/rtpcollision.c
Normal file
279
tests/check/elements/rtpcollision.c
Normal file
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@ -0,0 +1,279 @@
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/* GStreamer
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*
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* Copyright (C) 2013 Collabora Ltd.
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* @author Julien Isorce <julien.isorce@collabora.co.uk>
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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/net/gstnetaddressmeta.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/rtp/gstrtcpbuffer.h>
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static GMainLoop *main_loop;
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static GstPad *srcpad;
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static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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static void
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message_received (GstBus * bus, GstMessage * message, GstPipeline * bin)
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{
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GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
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GST_MESSAGE_SRC (message), message);
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switch (message->type) {
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case GST_MESSAGE_EOS:
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g_main_loop_quit (main_loop);
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break;
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case GST_MESSAGE_WARNING:{
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GError *gerror;
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gchar *debug;
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gst_message_parse_warning (message, &gerror, &debug);
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gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
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g_error_free (gerror);
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g_free (debug);
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break;
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}
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case GST_MESSAGE_ERROR:{
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GError *gerror;
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gchar *debug;
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gst_message_parse_error (message, &gerror, &debug);
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gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
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g_error_free (gerror);
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g_free (debug);
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g_main_loop_quit (main_loop);
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break;
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}
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default:
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break;
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}
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}
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static GstBuffer *
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create_rtcp_app (guint32 ssrc)
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{
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GInetAddress *inet_addr_0 = g_inet_address_new_from_string ("192.168.1.1");
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guint16 port = 5678;
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GSocketAddress *socket_addr_0 = g_inet_socket_address_new (inet_addr_0, port);
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GstBuffer *old_rtcp_buffer = gst_rtcp_buffer_new (1400);
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GstBuffer *rtcp_buffer = NULL;
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GstRTCPPacket *rtcp_packet = NULL;
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GstRTCPBuffer rtcp = GST_RTCP_BUFFER_INIT;
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/* make sure to have a writable buffer before
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* changing anything
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*/
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rtcp_buffer = gst_buffer_make_writable (old_rtcp_buffer);
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if (old_rtcp_buffer != rtcp_buffer) {
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gst_buffer_unref (old_rtcp_buffer);
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}
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gst_buffer_add_net_address_meta (rtcp_buffer, socket_addr_0);
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/* need to begin with rr */
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gst_rtcp_buffer_map (rtcp_buffer, GST_MAP_READWRITE, &rtcp);
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rtcp_packet = g_slice_new0 (GstRTCPPacket);
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gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_RR, rtcp_packet);
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gst_rtcp_packet_rr_set_ssrc (rtcp_packet, ssrc);
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/* useful to make the rtcp buffer valid */
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rtcp_packet = g_slice_new0 (GstRTCPPacket);
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gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_APP, rtcp_packet);
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gst_rtcp_buffer_unmap (&rtcp);
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return rtcp_buffer;
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}
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static guint ssrc_before;
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static guint ssrc_after;
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static GstPadProbeReturn
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rtpsession_sinkpad_probe (GstPad * pad, GstPadProbeInfo * info,
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gpointer user_data)
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{
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GstPadProbeReturn ret = GST_PAD_PROBE_OK;
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if (info->type == (GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH)) {
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GstBuffer *buffer = GST_BUFFER (info->data);
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GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
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static gint i = 0;
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/* retrieve current ssrc */
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gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
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if (i < 3)
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ssrc_before = gst_rtp_buffer_get_ssrc (&rtp);
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else
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ssrc_after = gst_rtp_buffer_get_ssrc (&rtp);
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gst_rtp_buffer_unmap (&rtp);
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/* feint a collision on recv_rtcp_sink pad of gstrtpsession
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* (note that after being marked as collied the rtpsession ignores
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* all non bye packets)
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*/
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if (i == 2) {
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GstBuffer *rtcp_buffer = create_rtcp_app (ssrc_before);
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/* push collied packet on recv_rtcp_sink */
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gst_pad_push (srcpad, rtcp_buffer);
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}
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++i;
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}
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return ret;
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}
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static GstFlowReturn
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fake_udp_sink_chain_func (GstPad * pad, GstObject * parent, GstBuffer * buffer)
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{
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return GST_FLOW_OK;
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}
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/* This test build the pipeline audiotestsrc ! speexenc ! rtpspeexpay ! \
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* rtpsession ! fakesink
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* It manually pushs buffer into rtpsession with same ssrc but different
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* ip so that collision can be detected
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* The test checks that the payloader change their ssrc
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*/
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GST_START_TEST (test_master_ssrc_collision)
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{
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GstElement *bin, *src, *encoder, *rtppayloader, *rtpsession, *sink;
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GstBus *bus = NULL;
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gboolean res = FALSE;
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GstSegment segment;
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GstPad *sinkpad = NULL;
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GstPad *rtcp_sinkpad = NULL;
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GstPad *fake_udp_sinkpad = NULL;
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GstPad *rtcp_srcpad = NULL;
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GstStateChangeReturn state_res = GST_STATE_CHANGE_FAILURE;
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GST_INFO ("preparing test");
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/* build pipeline */
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bin = gst_pipeline_new ("pipeline");
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bus = gst_element_get_bus (bin);
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gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
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src = gst_element_factory_make ("audiotestsrc", "src");
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g_object_set (src, "num-buffers", 5, NULL);
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encoder = gst_element_factory_make ("speexenc", NULL);
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rtppayloader = gst_element_factory_make ("rtpspeexpay", NULL);
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g_object_set (rtppayloader, "pt", 96, NULL);
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rtpsession = gst_element_factory_make ("rtpsession", NULL);
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sink = gst_element_factory_make ("fakesink", "sink");
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gst_bin_add_many (GST_BIN (bin), src, encoder, rtppayloader,
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rtpsession, sink, NULL);
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/* link elements */
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res = gst_element_link (src, encoder);
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fail_unless (res == TRUE, NULL);
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res = gst_element_link (encoder, rtppayloader);
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fail_unless (res == TRUE, NULL);
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res = gst_element_link_pads_full (rtppayloader, "src",
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rtpsession, "send_rtp_sink", GST_PAD_LINK_CHECK_NOTHING);
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fail_unless (res == TRUE, NULL);
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res = gst_element_link_pads_full (rtpsession, "send_rtp_src",
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sink, "sink", GST_PAD_LINK_CHECK_NOTHING);
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fail_unless (res == TRUE, NULL);
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/* add probe on rtpsession sink pad to induce collision */
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sinkpad = gst_element_get_static_pad (rtpsession, "send_rtp_sink");
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gst_pad_add_probe (sinkpad,
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(GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH),
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(GstPadProbeCallback) rtpsession_sinkpad_probe, NULL, NULL);
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gst_object_unref (sinkpad);
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/* setup rtcp link */
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srcpad = gst_pad_new_from_static_template (&srctemplate, "src");
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rtcp_sinkpad = gst_element_get_request_pad (rtpsession, "recv_rtcp_sink");
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fail_unless (gst_pad_link (srcpad, rtcp_sinkpad) == GST_PAD_LINK_OK, NULL);
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gst_object_unref (rtcp_sinkpad);
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res = gst_pad_set_active (srcpad, TRUE);
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fail_if (res == FALSE);
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res =
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gst_pad_push_event (srcpad,
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gst_event_new_stream_start ("my_rtcp_stream_id"));
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fail_if (res == FALSE);
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res = gst_pad_push_event (srcpad, gst_event_new_segment (&segment));
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fail_if (res == FALSE);
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fake_udp_sinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink");
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gst_pad_set_chain_function (fake_udp_sinkpad, fake_udp_sink_chain_func);
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rtcp_srcpad = gst_element_get_request_pad (rtpsession, "send_rtcp_src");
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fail_unless (gst_pad_link (rtcp_srcpad, fake_udp_sinkpad) == GST_PAD_LINK_OK,
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NULL);
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gst_object_unref (rtcp_srcpad);
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res = gst_pad_set_active (fake_udp_sinkpad, TRUE);
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fail_if (res == FALSE);
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/* connect messages */
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main_loop = g_main_loop_new (NULL, FALSE);
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g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
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g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
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g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
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state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
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ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
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GST_INFO ("running main loop");
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g_main_loop_run (main_loop);
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state_res = gst_element_set_state (bin, GST_STATE_NULL);
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ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
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/* cleanup */
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gst_object_unref (srcpad);
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gst_object_unref (fake_udp_sinkpad);
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g_main_loop_unref (main_loop);
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gst_bus_remove_signal_watch (bus);
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gst_object_unref (bus);
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gst_object_unref (bin);
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/* check results */
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fail_if (ssrc_before == ssrc_after);
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}
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GST_END_TEST;
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static Suite *
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rtpcollision_suite (void)
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{
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Suite *s = suite_create ("rtpcollision");
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TCase *tc_chain = tcase_create ("general");
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tcase_set_timeout (tc_chain, 10);
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suite_add_tcase (s, tc_chain);
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tcase_add_test (tc_chain, test_master_ssrc_collision);
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return s;
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}
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GST_CHECK_MAIN (rtpcollision);
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