mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-24 01:00:37 +00:00
Code Issues 1 Projects Releases Wiki Activity

tests/check: add rtprtx::test_push_forward_seq

Browse source 
add simple unit test that manually push buffers
in rtprtxsend connected to rtprtxreceive.
Drops some buffers and make sure they are retransmisted.
This commit is contained in:
Julien Isorce 2013-11-01 16:17:51 +00:00 committed by Wim Taymans
parent 5a1aa75961
commit 2e4ce28443
3 changed files with 271 additions and 0 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

4
tests/check/Makefile.am
View file

@ -149,6 +149,7 @@ check_PROGRAMS = \
elements/rtpbin_buffer_list \ elements/rtpbin_buffer_list \
elements/rtpjitterbuffer \ elements/rtpjitterbuffer \
elements/rtpmux \ elements/rtpmux \
elements/rtprtx \
elements/shapewipe \ elements/shapewipe \
elements/spectrum \ elements/spectrum \
elements/udpsink \ elements/udpsink \
@ -324,6 +325,9 @@ elements_videofilter_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstvideo-$(GST_API_VERSI
elements_rtpjitterbuffer_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS) elements_rtpjitterbuffer_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
elements_rtpjitterbuffer_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD) elements_rtpjitterbuffer_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)
elements_rtprtx_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
elements_rtprtx_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)
elements_rtpsession_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS) elements_rtpsession_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
elements_rtpsession_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD) elements_rtpsession_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)

1
tests/check/elements/.gitignore vendored
View file

@ -53,6 +53,7 @@ rtpcollision
rtpjitterbuffer rtpjitterbuffer
rtpsession rtpsession
rtpmux rtpmux
rtprtx
shapewipe shapewipe
souphttpsrc souphttpsrc
spectrum spectrum

266
tests/check/elements/rtprtx.c Normal file
View file

@ -0,0 +1,266 @@
/* 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/check/gstconsistencychecker.h>
#include <gst/check/gsttestclock.h>
#include <gst/rtp/gstrtpbuffer.h>
/* For ease of programming we use globals to keep refs for our floating
* src and sink pads we create; otherwise we always have to do get_pad,
* get_peer, and then remove references in every test function */
static GstPad *srcpad, *sinkpad;
/* we also have a list of src buffers */
static GList *inbuffers = NULL;
#define RTP_CAPS_STRING \
"application/x-rtp, " \
"media = (string)audio, " \
"payload = (int) 0, " \
"clock-rate = (int) 8000, " \
"encoding-name = (string)PCMU"
#define RTP_FRAME_SIZE 20
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp")
);
static void
setup_rtprtx (GstElement * rtprtxsend, GstElement * rtprtxreceive,
gint num_buffers)
{
GstClock *clock;
GstBuffer *buffer;
GstPad *sendsrcpad;
GstPad *receivesinkpad;
gboolean ret = FALSE;
/* a 20 sample audio block (2,5 ms) generated with
* gst-launch audiotestsrc wave=silence blocksize=40 num-buffers=3 !
* "audio/x-raw,channels=1,rate=8000" ! mulawenc ! rtppcmupay !
* fakesink dump=1
*/
guint8 in[] = { /* first 4 bytes are rtp-header, next 4 bytes are timestamp */
0x80, 0x80, 0x1c, 0x24, 0x46, 0xcd, 0xb7, 0x11, 0x3c, 0x3a, 0x7c, 0x5b,
0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff
};
GstClockTime ts = G_GUINT64_CONSTANT (0);
GstClockTime tso = gst_util_uint64_scale (RTP_FRAME_SIZE, GST_SECOND, 8000);
gint i;
/* we need a clock here */
clock = gst_system_clock_obtain ();
gst_element_set_clock (rtprtxsend, clock);
gst_object_unref (clock);
srcpad = gst_check_setup_src_pad (rtprtxsend, &srctemplate);
sendsrcpad = gst_element_get_static_pad (rtprtxsend, "src");
ret = gst_pad_set_active (srcpad, TRUE);
fail_if (ret == FALSE);
sinkpad = gst_check_setup_sink_pad (rtprtxreceive, &sinktemplate);
receivesinkpad = gst_element_get_static_pad (rtprtxreceive, "sink");
ret = gst_pad_set_active (sinkpad, TRUE);
fail_if (ret == FALSE);
fail_if (gst_pad_link (sendsrcpad, receivesinkpad) != GST_PAD_LINK_OK);
ret = gst_pad_set_active (sendsrcpad, TRUE);
fail_if (ret == FALSE);
ret = gst_pad_set_active (receivesinkpad, TRUE);
fail_if (ret == FALSE);
gst_object_unref (sendsrcpad);
gst_object_unref (receivesinkpad);
for (i = 0; i < num_buffers; i++) {
buffer = gst_buffer_new_and_alloc (sizeof (in));
gst_buffer_fill (buffer, 0, in, sizeof (in));
GST_BUFFER_DTS (buffer) = ts;
GST_BUFFER_PTS (buffer) = ts;
GST_BUFFER_DURATION (buffer) = tso;
GST_DEBUG ("created buffer: %p", buffer);
/*if (!i)
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT); */
inbuffers = g_list_append (inbuffers, buffer);
/* hackish way to update the rtp header */
in[1] = 0x00;
in[3]++; /* seqnumber */
in[7] += RTP_FRAME_SIZE; /* inc. timestamp with framesize */
ts += tso;
}
}
static GstStateChangeReturn
start_rtprtx (GstElement * element)
{
GstStateChangeReturn ret;
GstClockTime now;
GstClock *clock;
clock = gst_element_get_clock (element);
if (clock) {
now = gst_clock_get_time (clock);
gst_object_unref (clock);
gst_element_set_base_time (element, now);
}
ret = gst_element_set_state (element, GST_STATE_PLAYING);
ck_assert_int_ne (ret, GST_STATE_CHANGE_FAILURE);
ret = gst_element_get_state (element, NULL, NULL, GST_CLOCK_TIME_NONE);
ck_assert_int_ne (ret, GST_STATE_CHANGE_FAILURE);
return ret;
}
static void
cleanup_rtprtx (GstElement * rtprtxsend, GstElement * rtprtxreceive)
{
GST_DEBUG ("cleanup_rtprtx");
g_list_free (inbuffers);
inbuffers = NULL;
gst_pad_set_active (srcpad, FALSE);
gst_check_teardown_src_pad (rtprtxsend);
gst_check_teardown_element (rtprtxsend);
gst_pad_set_active (sinkpad, FALSE);
gst_check_teardown_sink_pad (rtprtxreceive);
gst_check_teardown_element (rtprtxreceive);
}
static void
check_rtprtx_results (GstElement * rtprtxsend, GstElement * rtprtxreceive,
gint num_buffers)
{
guint nbrtxrequests = 0;
guint nbrtxpackets = 0;
g_object_get (G_OBJECT (rtprtxsend), "num-rtx-requests", &nbrtxrequests,
NULL);
fail_unless_equals_int (nbrtxrequests, 3);
g_object_get (G_OBJECT (rtprtxsend), "num-rtx-packets", &nbrtxpackets, NULL);
fail_unless_equals_int (nbrtxpackets, 3);
g_object_get (G_OBJECT (rtprtxreceive), "num-rtx-requests", &nbrtxrequests,
NULL);
fail_unless_equals_int (nbrtxrequests, 3);
g_object_get (G_OBJECT (rtprtxreceive), "num-rtx-packets", &nbrtxpackets,
NULL);
fail_unless_equals_int (nbrtxpackets, 3);
g_object_get (G_OBJECT (rtprtxreceive), "num-rtx-assoc-packets",
&nbrtxpackets, NULL);
fail_unless_equals_int (nbrtxpackets, 3);
}
GST_START_TEST (test_push_forward_seq)
{
GstElement *rtprtxsend;
GstElement *rtprtxreceive;
const guint num_buffers = 4;
GList *node;
gint i = 0;
GstCaps *caps = NULL;
rtprtxsend = gst_check_setup_element ("rtprtxsend");
rtprtxreceive = gst_check_setup_element ("rtprtxreceive");
setup_rtprtx (rtprtxsend, rtprtxreceive, num_buffers);
GST_DEBUG ("setup_rtprtx");
fail_unless (start_rtprtx (rtprtxsend)
== GST_STATE_CHANGE_SUCCESS, "could not set to playing");
fail_unless (start_rtprtx (rtprtxreceive)
== GST_STATE_CHANGE_SUCCESS, "could not set to playing");
caps = gst_caps_from_string (RTP_CAPS_STRING);
gst_check_setup_events (srcpad, rtprtxsend, caps, GST_FORMAT_TIME);
gst_caps_unref (caps);
g_object_set (rtprtxsend, "rtx-payload-type", 97, NULL);
g_object_set (rtprtxreceive, "rtx-payload-types", "97", NULL);
/* push buffers: 0,1,2, */
for (node = inbuffers; node; node = g_list_next (node)) {
GstEvent *event = NULL;
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
GstBuffer *buffer = (GstBuffer *) node->data;
fail_unless (gst_pad_push (srcpad, buffer) == GST_FLOW_OK);
if (i < 3) {
gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
gst_structure_new ("GstRTPRetransmissionRequest",
"seqnum", G_TYPE_UINT, (guint) gst_rtp_buffer_get_seq (&rtp),
"ssrc", G_TYPE_UINT, (guint) gst_rtp_buffer_get_ssrc (&rtp),
"payload-type", G_TYPE_UINT,
(guint) gst_rtp_buffer_get_payload_type (&rtp), NULL));
fail_unless (gst_pad_push_event (sinkpad, event));
gst_rtp_buffer_unmap (&rtp);
}
gst_buffer_unref (buffer);
++i;
}
/* check the buffer list */
check_rtprtx_results (rtprtxsend, rtprtxreceive, num_buffers);
/* cleanup */
cleanup_rtprtx (rtprtxsend, rtprtxreceive);
}
GST_END_TEST;
static Suite *
rtprtx_suite (void)
{
Suite *s = suite_create ("rtprtx");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_push_forward_seq);
return s;
}
GST_CHECK_MAIN (rtprtx);