gstreamer/tests/check/elements/rtpcollision.c
2014-04-17 16:39:59 +02:00

462 lines
15 KiB
C

/* 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;
guint16 port = 5678;
GSocketAddress *socket_addr_0;
GstBuffer *rtcp_buffer;
GstRTCPPacket *rtcp_packet = NULL;
GstRTCPBuffer rtcp = GST_RTCP_BUFFER_INIT;
inet_addr_0 = g_inet_address_new_from_string ("192.168.1.1");
socket_addr_0 = g_inet_socket_address_new (inet_addr_0, port);
g_object_unref (inet_addr_0);
rtcp_buffer = gst_rtcp_buffer_new (1400);
gst_buffer_add_net_address_meta (rtcp_buffer, socket_addr_0);
g_object_unref (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);
g_slice_free (GstRTCPPacket, rtcp_packet);
/* 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);
g_slice_free (GstRTCPPacket, rtcp_packet);
gst_rtcp_buffer_unmap (&rtcp);
return rtcp_buffer;
}
static guint nb_ssrc_changes;
static guint ssrc_prev;
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;
GstBuffer *rtcp_buffer = 0;
guint ssrc = 0;
/* retrieve current ssrc */
gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
ssrc = gst_rtp_buffer_get_ssrc (&rtp);
gst_rtp_buffer_unmap (&rtp);
/* if not first buffer, check that our ssrc has changed */
if (ssrc_prev != -1 && ssrc != ssrc_prev)
++nb_ssrc_changes;
/* update prev ssrc */
ssrc_prev = ssrc;
/* feint a collision on recv_rtcp_sink pad of gstrtpsession
* (note that after being marked as collied the rtpsession ignores
* all non bye packets)
*/
rtcp_buffer = create_rtcp_app (ssrc);
/* push collied packet on recv_rtcp_sink */
gst_pad_push (srcpad, rtcp_buffer);
}
return ret;
}
static GstFlowReturn
fake_udp_sink_chain_func (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
gst_buffer_unref (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");
nb_ssrc_changes = 0;
ssrc_prev = -1;
/* 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);
gst_segment_init (&segment, GST_FORMAT_TIME);
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_unless_equals_int (nb_ssrc_changes, 7);
}
GST_END_TEST;
static guint ssrc_before;
static guint ssrc_after;
static guint rtx_ssrc_before;
static guint rtx_ssrc_after;
static GstPadProbeReturn
rtpsession_sinkpad_probe2 (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;
guint payload_type = 0;
static gint i = 0;
/* retrieve current ssrc for retransmission stream only */
gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
payload_type = gst_rtp_buffer_get_payload_type (&rtp);
if (payload_type == 99) {
if (i < 3)
rtx_ssrc_before = gst_rtp_buffer_get_ssrc (&rtp);
else
rtx_ssrc_after = gst_rtp_buffer_get_ssrc (&rtp);
} else {
/* ask to retransmit every packet */
GstEvent *event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
gst_structure_new ("GstRTPRetransmissionRequest",
"seqnum", G_TYPE_UINT, gst_rtp_buffer_get_seq (&rtp),
"ssrc", G_TYPE_UINT, gst_rtp_buffer_get_ssrc (&rtp),
NULL));
gst_pad_push_event (pad, event);
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 (rtx_ssrc_before);
/* push collied packet on recv_rtcp_sink */
gst_pad_push (srcpad, rtcp_buffer);
}
++i;
}
return ret;
}
/* This test build the pipeline audiotestsrc ! speexenc ! rtpspeexpay ! \
* rtprtxsend ! rtpsession ! fakesink
* It manually pushs buffer into rtpsession with same ssrc than rtx stream
* but different ip so that collision can be detected
* The test checks that the rtx elements changes its ssrc whereas
* the payloader keeps its master ssrc
*/
GST_START_TEST (test_rtx_ssrc_collision)
{
GstElement *bin, *src, *encoder, *rtppayloader, *rtprtxsend, *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;
GstStructure *pt_map;
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);
rtprtxsend = gst_element_factory_make ("rtprtxsend", NULL);
pt_map = gst_structure_new ("application/x-rtp-pt-map",
"96", G_TYPE_UINT, 99, NULL);
g_object_set (rtprtxsend, "payload-type-map", pt_map, NULL);
gst_structure_free (pt_map);
rtpsession = gst_element_factory_make ("rtpsession", NULL);
sink = gst_element_factory_make ("fakesink", "sink");
gst_bin_add_many (GST_BIN (bin), src, encoder, rtppayloader, rtprtxsend,
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 (rtppayloader, rtprtxsend);
fail_unless (res == TRUE, NULL);
res = gst_element_link_pads_full (rtprtxsend, "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_probe2, 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);
gst_segment_init (&segment, GST_FORMAT_TIME);
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 (rtx_ssrc_before == rtx_ssrc_after);
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);
tcase_add_test (tc_chain, test_rtx_ssrc_collision);
return s;
}
GST_CHECK_MAIN (rtpcollision);