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tests/check: add rtpaux::test_simple_rtpbin_aux

Browse source 
It shows how to use "set-aux-receive" and "set-aux-send"
properties of rtpbin to set rtprtxsend and rtprtxreceive

Build 2 pipelines, one for rtpbin as a sender and one for
rtobin as a receive. Then transmit an audio stream.

It also drops some packets to activate restransmission and
check they are actually retransmited.
This commit is contained in:
Julien Isorce 2013-11-01 16:57:15 +00:00 committed by Wim Taymans
parent 68149d14e1
commit 5f360f3b13
3 changed files with 412 additions and 0 deletions
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4
tests/check/Makefile.am
View file

@ -142,6 +142,7 @@ check_PROGRAMS = \
elements/rganalysis \
elements/rglimiter \
elements/rgvolume \
elements/rtpaux \
elements/rtpcollision \
elements/rtp-payloading \
elements/rtpbin \
@ -334,6 +335,9 @@ elements_rtpsession_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VERSION)
elements_rtpcollision_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
elements_rtpcollision_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstnet-$(GST_API_VERSION) -lgstrtp-$(GST_API_VERSION) $(GIO_LIBS) $(LDADD)
elements_rtpaux_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
elements_rtpaux_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-$(GST_API_VERSION) $(LDADD)
# FIXME: configure should check for gdk-pixbuf not gtk
# only need video.h header, not the lib
elements_gdkpixbufsink_CFLAGS = \

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

@ -47,6 +47,7 @@ rganalysis
rglimiter
rgvolume
rtp-payloading
rtpaux
rtpbin
rtpbin_buffer_list
rtpcollision

407
tests/check/elements/rtpaux.c Normal file
View file

@ -0,0 +1,407 @@
/* 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>
static GMainLoop *main_loop;
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;
}
}
typedef struct
{
guint count;
guint nb_packets;
guint drop_every_n_packets;
} RTXSendData;
static GstPadProbeReturn
rtprtxsend_srcpad_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);
RTXSendData *rtxdata = (RTXSendData *) user_data;
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
guint payload_type = 0;
gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
payload_type = gst_rtp_buffer_get_payload_type (&rtp);
/* main stream packets */
if (payload_type == 96) {
/* count packets of the main stream */
++rtxdata->nb_packets;
/* drop some packets */
if (rtxdata->count < rtxdata->drop_every_n_packets) {
++rtxdata->count;
} else {
/* drop a packet every 'rtxdata->count' packets */
rtxdata->count = 1;
ret = GST_PAD_PROBE_DROP;
}
} else {
/* retransmission packets */
}
gst_rtp_buffer_unmap (&rtp);
}
return ret;
}
static void
on_rtpbinreceive_pad_added (GstElement * element, GstPad * newPad,
gpointer data)
{
GstElement *rtpdepayloader = GST_ELEMENT (data);
gchar *padName = gst_pad_get_name (newPad);
if (g_str_has_prefix (padName, "recv_rtp_src_")) {
GstPad *sinkpad = gst_element_get_static_pad (rtpdepayloader, "sink");
gst_pad_link (newPad, sinkpad);
gst_object_unref (sinkpad);
}
g_free (padName);
}
static gboolean
on_timeout (gpointer data)
{
GstEvent *eos = gst_event_new_eos ();
if (!gst_element_send_event (GST_ELEMENT (data), eos)) {
GST_ERROR ("failed to send end of stream event");
gst_event_unref (eos);
}
return FALSE;
}
static GstElement *
request_aux_receive (GstElement * rtpbin, guint sessid, GstElement * receive)
{
GstElement *bin;
GstPad *pad;
GST_INFO ("creating AUX receiver");
bin = gst_bin_new (NULL);
gst_bin_add (GST_BIN (bin), receive);
pad = gst_element_get_static_pad (receive, "src");
gst_element_add_pad (bin, gst_ghost_pad_new ("src_0", pad));
gst_object_unref (pad);
pad = gst_element_get_static_pad (receive, "sink");
gst_element_add_pad (bin, gst_ghost_pad_new ("sink_0", pad));
gst_object_unref (pad);
return bin;
}
static GstElement *
request_aux_send (GstElement * rtpbin, guint sessid, GstElement * send)
{
GstElement *bin;
GstPad *pad;
GST_INFO ("creating AUX sender");
bin = gst_bin_new (NULL);
gst_bin_add (GST_BIN (bin), send);
pad = gst_element_get_static_pad (send, "src");
gst_element_add_pad (bin, gst_ghost_pad_new ("src_0", pad));
gst_object_unref (pad);
pad = gst_element_get_static_pad (send, "sink");
gst_element_add_pad (bin, gst_ghost_pad_new ("sink_0", pad));
gst_object_unref (pad);
return bin;
}
GST_START_TEST (test_simple_rtpbin_aux)
{
GstElement *binsend, *rtpbinsend, *src, *encoder, *rtppayloader,
*rtprtxsend, *sendrtp_udpsink, *sendrtcp_udpsink, *sendrtcp_udpsrc;
GstElement *binreceive, *rtpbinreceive, *recvrtp_udpsrc, *recvrtcp_udpsrc,
*recvrtcp_udpsink, *rtprtxreceive, *rtpdepayloader, *decoder, *converter,
*sink;
GstBus *bussend;
GstBus *busreceive;
gboolean res;
GstCaps *rtpcaps = NULL;
GstStateChangeReturn state_res = GST_STATE_CHANGE_FAILURE;
GstPad *srcpad = NULL;
guint nb_rtx_send_packets = 0;
guint nb_rtx_recv_packets = 0;
RTXSendData send_rtxdata;
send_rtxdata.count = 1;
send_rtxdata.nb_packets = 0;
send_rtxdata.drop_every_n_packets = 50;
GST_INFO ("preparing test");
/* build pipeline */
binsend = gst_pipeline_new ("pipeline_send");
bussend = gst_element_get_bus (binsend);
gst_bus_add_signal_watch_full (bussend, G_PRIORITY_HIGH);
binreceive = gst_pipeline_new ("pipeline_receive");
busreceive = gst_element_get_bus (binreceive);
gst_bus_add_signal_watch_full (busreceive, G_PRIORITY_HIGH);
rtpbinsend = gst_element_factory_make ("rtpbin", "rtpbinsend");
g_object_set (rtpbinsend, "latency", 200, "do-retransmission", TRUE, NULL);
src = gst_element_factory_make ("audiotestsrc", "src");
encoder = gst_element_factory_make ("speexenc", "encoder");
rtppayloader = gst_element_factory_make ("rtpspeexpay", "rtppayloader");
rtprtxsend = gst_element_factory_make ("rtprtxsend", "rtprtxsend");
sendrtp_udpsink = gst_element_factory_make ("udpsink", "sendrtp_udpsink");
g_object_set (sendrtp_udpsink, "host", "127.0.0.1", NULL);
g_object_set (sendrtp_udpsink, "port", 5006, NULL);
sendrtcp_udpsink = gst_element_factory_make ("udpsink", "sendrtcp_udpsink");
g_object_set (sendrtcp_udpsink, "host", "127.0.0.1", NULL);
g_object_set (sendrtcp_udpsink, "port", 5007, NULL);
g_object_set (sendrtcp_udpsink, "sync", FALSE, NULL);
g_object_set (sendrtcp_udpsink, "async", FALSE, NULL);
sendrtcp_udpsrc = gst_element_factory_make ("udpsrc", "sendrtcp_udpsrc");
g_object_set (sendrtcp_udpsrc, "port", 5009, NULL);
rtpbinreceive = gst_element_factory_make ("rtpbin", "rtpbinreceive");
g_object_set (rtpbinreceive, "latency", 200, "do-retransmission", TRUE, NULL);
recvrtp_udpsrc = gst_element_factory_make ("udpsrc", "recvrtp_udpsrc");
g_object_set (recvrtp_udpsrc, "port", 5006, NULL);
rtpcaps =
gst_caps_from_string
("application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)SPEEX,encoding-params=(string)1,octet-align=(string)1");
g_object_set (recvrtp_udpsrc, "caps", rtpcaps, NULL);
gst_caps_unref (rtpcaps);
recvrtcp_udpsrc = gst_element_factory_make ("udpsrc", "recvrtcp_udpsrc");
g_object_set (recvrtcp_udpsrc, "port", 5007, NULL);
recvrtcp_udpsink = gst_element_factory_make ("udpsink", "recvrtcp_udpsink");
g_object_set (recvrtcp_udpsink, "host", "127.0.0.1", NULL);
g_object_set (recvrtcp_udpsink, "port", 5009, NULL);
g_object_set (recvrtcp_udpsink, "sync", FALSE, NULL);
g_object_set (recvrtcp_udpsink, "async", FALSE, NULL);
rtprtxreceive = gst_element_factory_make ("rtprtxreceive", "rtprtxreceive");
rtpdepayloader = gst_element_factory_make ("rtpspeexdepay", "rtpdepayloader");
decoder = gst_element_factory_make ("speexdec", "decoder");
converter = gst_element_factory_make ("identity", "converter");
sink = gst_element_factory_make ("alsasink", "sink");
gst_bin_add_many (GST_BIN (binsend), rtpbinsend, src, encoder, rtppayloader,
sendrtp_udpsink, sendrtcp_udpsink, sendrtcp_udpsrc, NULL);
gst_bin_add_many (GST_BIN (binreceive), rtpbinreceive,
recvrtp_udpsrc, recvrtcp_udpsrc, recvrtcp_udpsink,
rtpdepayloader, decoder, converter, sink, NULL);
g_signal_connect (rtpbinreceive, "pad-added",
G_CALLBACK (on_rtpbinreceive_pad_added), rtpdepayloader);
g_object_set (rtppayloader, "pt", 96, NULL);
g_object_set (rtppayloader, "seqnum-offset", 1, NULL);
g_object_set (rtprtxsend, "rtx-payload-type", 99, NULL);
g_object_set (rtprtxreceive, "rtx-payload-types", "99:111:125", NULL);
/* set rtp aux receive */
g_signal_connect (rtpbinreceive, "request-aux-receiver", (GCallback)
request_aux_receive, rtprtxreceive);
/* set rtp aux send */
g_signal_connect (rtpbinsend, "request-aux-sender", (GCallback)
request_aux_send, rtprtxsend);
/* gst-launch-1.0 rtpbin name=rtpbin audiotestsrc ! amrnbenc ! rtpamrpay ! \
* rtpbin.send_rtp_sink_1 rtpbin.send_rtp_src_1 ! udpsink host="127.0.0.1" \
* port=5002 rtpbin.send_rtcp_src_1 ! udpsink host="127.0.0.1" port=5003 \
* sync=false async=false udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
*/
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", rtpbinsend,
"send_rtp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
fail_unless (res == TRUE, NULL);
res =
gst_element_link_pads_full (rtpbinsend, "send_rtp_src_0", sendrtp_udpsink,
"sink", GST_PAD_LINK_CHECK_NOTHING);
fail_unless (res == TRUE, NULL);
res =
gst_element_link_pads_full (rtpbinsend, "send_rtcp_src_0",
sendrtcp_udpsink, "sink", GST_PAD_LINK_CHECK_NOTHING);
fail_unless (res == TRUE, NULL);
res =
gst_element_link_pads_full (sendrtcp_udpsrc, "src", rtpbinsend,
"recv_rtcp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
fail_unless (res == TRUE, NULL);
srcpad = gst_element_get_static_pad (rtpbinsend, "send_rtp_src_0");
gst_pad_add_probe (srcpad,
(GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH),
(GstPadProbeCallback) rtprtxsend_srcpad_probe, &send_rtxdata, NULL);
gst_object_unref (srcpad);
/* gst-launch-1.0 rtpbin name=rtpbin udpsrc caps="application/x-rtp,media=(string)audio, \
* clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,o
* ctet-align=(string)1" port=5002 ! rtpbin.recv_rtp_sink_1 rtpbin. ! rtpamrdepay ! \
* amrnbdec ! alsasink udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
* rtpbin.send_rtcp_src_1 ! udpsink host="127.0.0.1" port=5007 sync=false async=false
*/
res =
gst_element_link_pads_full (recvrtp_udpsrc, "src", rtpbinreceive,
"recv_rtp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
fail_unless (res == TRUE, NULL);
res =
gst_element_link_pads_full (rtpdepayloader, "src", decoder, "sink",
GST_PAD_LINK_CHECK_NOTHING);
fail_unless (res == TRUE, NULL);
res = gst_element_link (decoder, converter);
fail_unless (res == TRUE, NULL);
res =
gst_element_link_pads_full (converter, "src", sink, "sink",
GST_PAD_LINK_CHECK_NOTHING);
fail_unless (res == TRUE, NULL);
res =
gst_element_link_pads_full (recvrtcp_udpsrc, "src", rtpbinreceive,
"recv_rtcp_sink_0", GST_PAD_LINK_CHECK_NOTHING);
fail_unless (res == TRUE, NULL);
res =
gst_element_link_pads_full (rtpbinreceive, "send_rtcp_src_0",
recvrtcp_udpsink, "sink", GST_PAD_LINK_CHECK_NOTHING);
fail_unless (res == TRUE, NULL);
main_loop = g_main_loop_new (NULL, FALSE);
g_signal_connect (bussend, "message::error", (GCallback) message_received,
binsend);
g_signal_connect (bussend, "message::warning", (GCallback) message_received,
binsend);
g_signal_connect (bussend, "message::eos", (GCallback) message_received,
binsend);
g_signal_connect (busreceive, "message::error", (GCallback) message_received,
binreceive);
g_signal_connect (busreceive, "message::warning",
(GCallback) message_received, binreceive);
g_signal_connect (busreceive, "message::eos", (GCallback) message_received,
binreceive);
state_res = gst_element_set_state (binreceive, GST_STATE_PLAYING);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
state_res = gst_element_set_state (binsend, GST_STATE_PLAYING);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
g_timeout_add (5000, on_timeout, binsend);
g_timeout_add (5000, on_timeout, binreceive);
GST_INFO ("enter mainloop");
g_main_loop_run (main_loop);
g_main_loop_run (main_loop);
GST_INFO ("exit mainloop");
/* check that FB NACK is working */
g_object_get (G_OBJECT (rtprtxsend), "num-rtx-requests", &nb_rtx_send_packets,
NULL);
g_object_get (G_OBJECT (rtprtxreceive), "num-rtx-requests",
&nb_rtx_recv_packets, NULL);
state_res = gst_element_set_state (binsend, GST_STATE_NULL);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
state_res = gst_element_set_state (binreceive, GST_STATE_NULL);
ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
GST_INFO ("nb_rtx_send_packets %d", nb_rtx_send_packets);
GST_INFO ("nb_rtx_recv_packets %d", nb_rtx_recv_packets);
fail_if (nb_rtx_send_packets < 1);
fail_if (nb_rtx_recv_packets < 1);
/* cleanup */
g_main_loop_unref (main_loop);
gst_bus_remove_signal_watch (bussend);
gst_object_unref (bussend);
gst_object_unref (binsend);
gst_bus_remove_signal_watch (busreceive);
gst_object_unref (busreceive);
gst_object_unref (binreceive);
}
GST_END_TEST;
static Suite *
rtpaux_suite (void)
{
Suite *s = suite_create ("rtpaux");
TCase *tc_chain = tcase_create ("general");
tcase_set_timeout (tc_chain, 10000);
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_simple_rtpbin_aux);
return s;
}
GST_CHECK_MAIN (rtpaux);