--- /dev/null
+/* GStreamer
+ *
+ * Copyright (C) 2013 Collabora Ltd.
+ * @author Julien Isorce <julien.isorce@collabora.co.uk>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/check/gstcheck.h>
+#include <gst/net/gstnetaddressmeta.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/rtp/gstrtcpbuffer.h>
+
+static GMainLoop *main_loop;
+static GstPad *srcpad;
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtcp")
+ );
+
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtcp")
+ );
+
+static void
+message_received (GstBus * bus, GstMessage * message, GstPipeline * bin)
+{
+ GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
+ GST_MESSAGE_SRC (message), message);
+
+ switch (message->type) {
+ case GST_MESSAGE_EOS:
+ g_main_loop_quit (main_loop);
+ break;
+ case GST_MESSAGE_WARNING:{
+ GError *gerror;
+ gchar *debug;
+
+ gst_message_parse_warning (message, &gerror, &debug);
+ gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
+ g_error_free (gerror);
+ g_free (debug);
+ break;
+ }
+ case GST_MESSAGE_ERROR:{
+ GError *gerror;
+ gchar *debug;
+
+ gst_message_parse_error (message, &gerror, &debug);
+ gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
+ g_error_free (gerror);
+ g_free (debug);
+ g_main_loop_quit (main_loop);
+ break;
+ }
+ default:
+ break;
+ }
+}
+
+static GstBuffer *
+create_rtcp_app (guint32 ssrc)
+{
+ GInetAddress *inet_addr_0 = g_inet_address_new_from_string ("192.168.1.1");
+ guint16 port = 5678;
+ GSocketAddress *socket_addr_0 = g_inet_socket_address_new (inet_addr_0, port);
+ GstBuffer *old_rtcp_buffer = gst_rtcp_buffer_new (1400);
+ GstBuffer *rtcp_buffer = NULL;
+ GstRTCPPacket *rtcp_packet = NULL;
+ GstRTCPBuffer rtcp = GST_RTCP_BUFFER_INIT;
+
+ /* make sure to have a writable buffer before
+ * changing anything
+ */
+ rtcp_buffer = gst_buffer_make_writable (old_rtcp_buffer);
+ if (old_rtcp_buffer != rtcp_buffer) {
+ gst_buffer_unref (old_rtcp_buffer);
+ }
+ gst_buffer_add_net_address_meta (rtcp_buffer, socket_addr_0);
+
+ /* need to begin with rr */
+ gst_rtcp_buffer_map (rtcp_buffer, GST_MAP_READWRITE, &rtcp);
+ rtcp_packet = g_slice_new0 (GstRTCPPacket);
+ gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_RR, rtcp_packet);
+ gst_rtcp_packet_rr_set_ssrc (rtcp_packet, ssrc);
+
+ /* useful to make the rtcp buffer valid */
+ rtcp_packet = g_slice_new0 (GstRTCPPacket);
+ gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_APP, rtcp_packet);
+ gst_rtcp_buffer_unmap (&rtcp);
+
+ return rtcp_buffer;
+}
+
+static guint ssrc_before;
+static guint ssrc_after;
+
+static GstPadProbeReturn
+rtpsession_sinkpad_probe (GstPad * pad, GstPadProbeInfo * info,
+ gpointer user_data)
+{
+ GstPadProbeReturn ret = GST_PAD_PROBE_OK;
+
+ if (info->type == (GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH)) {
+ GstBuffer *buffer = GST_BUFFER (info->data);
+ GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
+
+ static gint i = 0;
+
+ /* retrieve current ssrc */
+ gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
+ if (i < 3)
+ ssrc_before = gst_rtp_buffer_get_ssrc (&rtp);
+ else
+ ssrc_after = gst_rtp_buffer_get_ssrc (&rtp);
+ gst_rtp_buffer_unmap (&rtp);
+
+ /* feint a collision on recv_rtcp_sink pad of gstrtpsession
+ * (note that after being marked as collied the rtpsession ignores
+ * all non bye packets)
+ */
+ if (i == 2) {
+ GstBuffer *rtcp_buffer = create_rtcp_app (ssrc_before);
+
+ /* push collied packet on recv_rtcp_sink */
+ gst_pad_push (srcpad, rtcp_buffer);
+ }
+
+ ++i;
+ }
+
+ return ret;
+}
+
+static GstFlowReturn
+fake_udp_sink_chain_func (GstPad * pad, GstObject * parent, GstBuffer * buffer)
+{
+ return GST_FLOW_OK;
+}
+
+/* This test build the pipeline audiotestsrc ! speexenc ! rtpspeexpay ! \
+ * rtpsession ! fakesink
+ * It manually pushs buffer into rtpsession with same ssrc but different
+ * ip so that collision can be detected
+ * The test checks that the payloader change their ssrc
+ */
+GST_START_TEST (test_master_ssrc_collision)
+{
+ GstElement *bin, *src, *encoder, *rtppayloader, *rtpsession, *sink;
+ GstBus *bus = NULL;
+ gboolean res = FALSE;
+ GstSegment segment;
+ GstPad *sinkpad = NULL;
+ GstPad *rtcp_sinkpad = NULL;
+ GstPad *fake_udp_sinkpad = NULL;
+ GstPad *rtcp_srcpad = NULL;
+ GstStateChangeReturn state_res = GST_STATE_CHANGE_FAILURE;
+
+ GST_INFO ("preparing test");
+
+ /* build pipeline */
+ bin = gst_pipeline_new ("pipeline");
+ bus = gst_element_get_bus (bin);
+ gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
+
+ src = gst_element_factory_make ("audiotestsrc", "src");
+ g_object_set (src, "num-buffers", 5, NULL);
+ encoder = gst_element_factory_make ("speexenc", NULL);
+ rtppayloader = gst_element_factory_make ("rtpspeexpay", NULL);
+ g_object_set (rtppayloader, "pt", 96, NULL);
+ rtpsession = gst_element_factory_make ("rtpsession", NULL);
+ sink = gst_element_factory_make ("fakesink", "sink");
+ gst_bin_add_many (GST_BIN (bin), src, encoder, rtppayloader,
+ rtpsession, sink, NULL);
+
+ /* link elements */
+ res = gst_element_link (src, encoder);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link (encoder, rtppayloader);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link_pads_full (rtppayloader, "src",
+ rtpsession, "send_rtp_sink", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+ res = gst_element_link_pads_full (rtpsession, "send_rtp_src",
+ sink, "sink", GST_PAD_LINK_CHECK_NOTHING);
+ fail_unless (res == TRUE, NULL);
+
+ /* add probe on rtpsession sink pad to induce collision */
+ sinkpad = gst_element_get_static_pad (rtpsession, "send_rtp_sink");
+ gst_pad_add_probe (sinkpad,
+ (GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH),
+ (GstPadProbeCallback) rtpsession_sinkpad_probe, NULL, NULL);
+ gst_object_unref (sinkpad);
+
+ /* setup rtcp link */
+ srcpad = gst_pad_new_from_static_template (&srctemplate, "src");
+ rtcp_sinkpad = gst_element_get_request_pad (rtpsession, "recv_rtcp_sink");
+ fail_unless (gst_pad_link (srcpad, rtcp_sinkpad) == GST_PAD_LINK_OK, NULL);
+ gst_object_unref (rtcp_sinkpad);
+ res = gst_pad_set_active (srcpad, TRUE);
+ fail_if (res == FALSE);
+ res =
+ gst_pad_push_event (srcpad,
+ gst_event_new_stream_start ("my_rtcp_stream_id"));
+ fail_if (res == FALSE);
+ res = gst_pad_push_event (srcpad, gst_event_new_segment (&segment));
+ fail_if (res == FALSE);
+
+ fake_udp_sinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink");
+ gst_pad_set_chain_function (fake_udp_sinkpad, fake_udp_sink_chain_func);
+ rtcp_srcpad = gst_element_get_request_pad (rtpsession, "send_rtcp_src");
+ fail_unless (gst_pad_link (rtcp_srcpad, fake_udp_sinkpad) == GST_PAD_LINK_OK,
+ NULL);
+ gst_object_unref (rtcp_srcpad);
+ res = gst_pad_set_active (fake_udp_sinkpad, TRUE);
+ fail_if (res == FALSE);
+
+ /* connect messages */
+ main_loop = g_main_loop_new (NULL, FALSE);
+ g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
+ g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
+ g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
+
+ state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
+ ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+ GST_INFO ("running main loop");
+ g_main_loop_run (main_loop);
+
+ state_res = gst_element_set_state (bin, GST_STATE_NULL);
+ ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
+
+ /* cleanup */
+ gst_object_unref (srcpad);
+ gst_object_unref (fake_udp_sinkpad);
+ g_main_loop_unref (main_loop);
+ gst_bus_remove_signal_watch (bus);
+ gst_object_unref (bus);
+ gst_object_unref (bin);
+
+ /* check results */
+ fail_if (ssrc_before == ssrc_after);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtpcollision_suite (void)
+{
+ Suite *s = suite_create ("rtpcollision");
+ TCase *tc_chain = tcase_create ("general");
+
+ tcase_set_timeout (tc_chain, 10);
+
+ suite_add_tcase (s, tc_chain);
+
+ tcase_add_test (tc_chain, test_master_ssrc_collision);
+
+ return s;
+}
+
+GST_CHECK_MAIN (rtpcollision);
projects / platform / upstream / gstreamer.git / commitdiff