Commit graph

882 commits

Author SHA1 Message Date
Sebastian Dröge
1a78a7eb22 souphttpsrc: Fix implicit enum conversion compiler warning
error: implicit conversion from enumeration type
'SoupStatus' to different enumeration type 'SoupKnownStatusCode'
2014-02-08 17:43:32 +01:00
Sebastian Dröge
ec1899e456 interleave: Fix unitialized variable compiler warning in test
error: variable 'mask' is used uninitialized
whenever 'if' condition is false [-Werror,-Wsometimes-uninitialized]
2014-02-08 17:41:21 +01:00
Edward Hervey
ceb602073a check: Use fakesink sync=True instead of an audio sink
Ensures the test can run on systems without alsa (or any audio output for
that matter), and will avoid people running build slaves wondering what
the hell was beeping during the night :)
2014-01-29 10:37:53 +01:00
George Kiagiadakis
016e1562a6 tests: rtprtx::test_rtxreceive_data_reconstruction: remove useless code for triggering retransmission
There is no need anymore to push yet another buffer in rtxsend
in order to trigger the previously requested retransmissions
to actually happen.
2014-01-21 15:00:54 +01:00
George Kiagiadakis
184553151d tests: rtprtx::test_rtxreceive_data_reconstruction: fix race condition
Now with rtprtxsend pushing rtx buffers from a different thread,
this is necessary to ensure that the result of the test is deterministic.

This code makes use of GstCheck's global GMutex and GCond that are
being used inside GstCheck's sink pad chain() function in order
to synchronize with it.
2014-01-21 15:00:53 +01:00
George Kiagiadakis
7677aec2fa tests: rtprtx::test_rtxsender_packet_retention: fix race condition
Now with rtprtxsend pushing rtx buffers from a different thread,
this is necessary to ensure that the result of the test is deterministic.

This code makes use of GstCheck's global GMutex and GCond that are
being used inside GstCheck's sink pad chain() function in order
to synchronize with it.
2014-01-21 15:00:53 +01:00
George Kiagiadakis
7011f98d7e tests: rtprtx::test_push_forward_seq: fix race condition
Now with rtprtxsend pushing rtx buffers from a different thread,
this is necessary to ensure that the result of the test is deterministic.

This code makes use of GstCheck's global GMutex and GCond that are
being used inside GstCheck's sink pad chain() function in order
to synchronize with it.
2014-01-21 15:00:53 +01:00
George Kiagiadakis
c702e37091 tests: rtprtx::test_push_forward_seq: fix buffer refcounting 2014-01-21 15:00:53 +01:00
Olivier Crête
8a143dfcbc tests: Remove usage of the system clock from the rtprtx test 2014-01-15 10:13:12 +01:00
Olivier Crête
f0a4f26fa7 tests: Initial segment in rtpcollision test 2014-01-15 10:13:12 +01:00
George Kiagiadakis
a7823bc522 examples/*-rtpaux: specify payload type association for the audio stream, so that rtx works also for audio 2014-01-15 10:13:12 +01:00
Stefan Sauer
d1223ebd10 wavparse: split the test
This way one failure won't shadow the other test and also if one fails we get
better disgnostics through the test-name.
2014-01-06 21:13:37 +01:00
George Kiagiadakis
94e4cd203b test/check: Verify rtprtxsend::ssrc-map property works as expected 2014-01-03 20:48:29 +01:00
George Kiagiadakis
9226091235 rtprtxreceive: modify to use a payload-type map like rtprtxsend 2014-01-03 20:48:29 +01:00
Wim Taymans
130ad1b1fa rtprtxsend: Allow SSRC-multiplexing and multiple payload types in the original stream
Conflicts:
	tests/examples/rtp/server-rtpaux.c
2014-01-03 20:48:29 +01:00
Torrie Fischer
e29b5f8b41 examples: rtp: Add end-to-end rtpbin example with RTX elements
This example demonstrates how to use rtpbin with retransmission (rtx)
elements set in the place of rtpbin's "aux" elements in order to
enable RTP retransmission according to the rules of RFC4588.
2014-01-03 20:48:29 +01:00
Julien Isorce
5f360f3b13 tests/check: add rtpaux::test_simple_rtpbin_aux
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.
2014-01-03 20:48:29 +01:00
Julien Isorce
68149d14e1 tests/check: add rtpcollision::test_rtx_ssrc_collision unit test
check that rtxrtpsend changes its retransmission ssrc when
collision happens
2014-01-03 20:48:28 +01:00
George Kiagiadakis
123bc46b60 tests/check: add rtprtx::test_rtxreceive_data_reconstruction
This unit test verifies that retransmitted rtp packets coming out
of rtprtxreceive are the same as the original ones.
2014-01-03 20:48:28 +01:00
George Kiagiadakis
487fa8c989 rtprtxsend: retransmit packets in the same order as the rtx requests 2014-01-03 20:48:28 +01:00
George Kiagiadakis
3e818e218b tests/check: Add unit test for rtxsend's max_size_time property 2014-01-03 20:48:28 +01:00
George Kiagiadakis
f7277db9e4 tests/check: Add rtprtx::test_rtxsender_packet_retention
This unit test verifies that the rtxsend element correctly maintains
a buffer of already transmitted rtp packets and that it can
re-transmit all of them correctly on demand. It also verifies
that the limit of this buffer (max-size-packets property) is respected.
2014-01-03 20:48:28 +01:00
Julien Isorce
71bdb5e088 tests/check: add rtprtx::test_drop_multiple_sender unit test
Several senders / one receiver

Similar than test_drop_one_sender but with multiple senders
mixed through the funnel element.
It drops some packets and checks that they are retransmited
correctly.
2014-01-03 20:48:28 +01:00
Julien Isorce
2a2fa7ebc0 tests/check: add rtprtx::test_drop_one_sender unit test
Test for one sender / one receiver

Build the pipeline
videotestsrc ! rtpvrawpay ! rtprtxsend ! rtprtxreceive ! fakesink
and drop some buffers between rtprtxsend and rtprtxreceive
Then it checks that every dropped packet has been re-sent.
It also checks that not too much requests has been sent.
2014-01-03 20:48:27 +01:00
Julien Isorce
2e4ce28443 tests/check: add rtprtx::test_push_forward_seq
add simple unit test that manually push buffers
in rtprtxsend connected to rtprtxreceive.
Drops some buffers and make sure they are retransmisted.
2014-01-03 20:48:27 +01:00
Wim Taymans
c83ed4f61e tests: add AUX receiver unit test 2013-12-31 15:08:49 +01:00
Wim Taymans
b91e0096b7 tests: improve rtpbin test 2013-12-31 15:08:49 +01:00
Wim Taymans
3e83e6a33d tests: add AUX sender unit test 2013-12-31 15:08:49 +01:00
Wim Taymans
841f9ad050 tests: add decoder test 2013-12-31 15:08:48 +01:00
Wim Taymans
3f3b2d0886 rtpbin: handle multiple encoder instances
Keep track of elements that are added to multiple sessions and make sure
we only add them to the rtpbin once and that we clean them when no
session refers to them anymore.
2013-12-30 16:28:57 +01:00
Wim Taymans
76e4cbc753 tests: add unit test for encoder element 2013-12-30 15:17:05 +01:00
Wim Taymans
bcd1589a91 tests: fix leak 2013-12-30 15:17:05 +01:00
Sebastian Dröge
29840bfd96 wavpackdec: Send a CAPS event in the unit test 2013-12-30 11:07:03 +01:00
Olivier Crête
dc845c1899 tests: Initialize segment in rtpcollision test 2013-12-13 16:05:41 -05:00
George Kiagiadakis
f9b7f44938 tests/check: add an rtpsession unit test to verify all RBs are included in all SRs, roundrobin
This test checks that when we have multiple internal sender sources
in rtpsession, SRs contain RBs for every other sender source, and that
they are included roundrobin when they exceed ST_RTCP_MAX_RB_COUNT,
which is the max number of RBs that can fit in a SR.
2013-12-12 16:02:56 +01:00
Julien Isorce
d562263852 tests/check: improve rtpcollision::test_master_ssrc_collision to ensure that a collision does not BYE the whole session
Conflicts:
	tests/check/elements/rtpcollision.c
2013-12-12 15:39:40 +01:00
Julien Isorce
7b001e35ed tests/check: add rtpcollision::test_master_ssrc_collision unit test
It checks the payloader changes its ssrc when collision happens
2013-12-12 15:39:39 +01:00
Wim Taymans
eee515cb2c rtpjitterbuffer: serialize events in the buffer
Serialize events into the jitterbuffer by inserting them with a -1
seqnum.
Update unit test to expect events from the streaming thread.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=652986
2013-12-10 11:57:37 +01:00
Wim Taymans
e0a5c07e8d audioparsers: use ACCEPT_INTERSECT flag
The parser can accept input that is not completely specified. Use the
ACCEPT_INTERSECT flag on the sinkpad to tweak the acceptcaps function to
check for intersection only. This allows us to proxy downstream
constraints while still allowing non-subset caps as input.
We can then also remove the appended template caps workaround.
Make a unit-test to check the new feature.

This reverts commit 26040ee38c

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=705024
2013-12-03 22:26:44 +01:00
Nicolas Dufresne
77833b886d videoflip: Add unit test for the 'automatic' method
These new tests send a tag event before seding the buffer. Tested case are an
empty tag list, a tag list with orientation-180 set and an invalid orientation value.

https://bugzilla.gnome.org/show_bug.cgi?id=719497
2013-11-28 11:59:05 -05:00
Wim Taymans
29d9b1e7de check: fix jitterbuffer check
Don't advance the clock to 240ms too early.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710013
2013-11-25 17:39:52 +01:00
Wim Taymans
710d1f3a2a rtpjitterbuffer: improve clear-pt-map handling
Don't reset the expected output seqnum when clearing the pt map because this
could stall the jitterbuffer forever.
Add a unit test for this.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=709800
2013-11-25 15:52:22 +01:00
George Kiagiadakis
387e3b918a rtpjitterbuffer: Fix stats property field names and documentation 2013-11-15 16:23:34 +02:00
Torrie Fischer
22ceb80ba9 rtpjitterbuffer: implement rtx statistics 2013-11-14 09:24:26 +01:00
Wim Taymans
3623ebf01e check: add rtpsession test
Add a basic rtpsession test to ensure that RR blocks are generated when
multiple SSRC senders are active.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711270
2013-11-11 14:28:52 +01:00
Thiago Santos
43602e2d8a tests: souphttpsrc: add explicit cast to silence warning
Silencing this warning:

elements/souphttpsrc.c:533:14: error: comparison between ‘SoupKnownStatusCode’ and ‘enum <anonymous>’ [-Werror=enum-compare]
   if (status != SOUP_STATUS_OK && !send_error_doc)

With gcc 4.8.2 (debian)
2013-10-31 13:22:40 -03:00
Wim Taymans
3c69d65b85 tests: add test for retransmission because of reordering 2013-09-23 14:45:27 +02:00
Wim Taymans
f40d6689f2 tests: remove timeouts from check
Timeouts make the test unreliable and are not needed.
2013-09-23 14:45:26 +02:00
Wim Taymans
a71014518c tests: add test for packet delay and retransmission 2013-09-23 14:45:24 +02:00
Wim Taymans
c959cdc8c1 tests: check both PTS and DTS 2013-09-23 14:45:24 +02:00