Commit graph

967 commits

Author SHA1 Message Date
Philippe Normand
b8b5704445 interleave: set output caps layout to interleaved
Set output caps layout independently from input caps layout which can
be either non-interleaved or interleaved.

https://bugzilla.gnome.org/show_bug.cgi?id=733866
2014-07-29 11:49:32 +02:00
Sebastian Rasmussen
c541e56b0c tests: fix pipeline leak in videocrop test
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732976
2014-07-10 22:26:43 +01:00
Sebastian Rasmussen
5cd0261f77 examples: client-rtpaux: Release reference to parent when done
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732976
2014-07-10 22:25:25 +01:00
Tim-Philipp Müller
46f6687bf6 tests: qtmux: suppress glib criticals caused by testing deprecated dts methods 2014-07-04 19:46:41 +01:00
Sebastian Dröge
0e13172837 rtpsession: Fix memory leaks in unit test 2014-06-30 00:00:43 +02:00
Tim-Philipp Müller
4edbd4c368 tests: matroskaparse: fail on errors and disable pull mode test
Actually look for error messages on the bus and fail if there
is one before the EOS message. Disable pull mode test which is
pointless as long as matroskaparse only supports push mode
(pull mode support has not been ported over to 1.0).
2014-06-28 17:40:45 +01:00
Ravi Kiran K N
e4f0133cb1 videobox: Add unit test
https://bugzilla.gnome.org/show_bug.cgi?id=732144
2014-06-26 18:52:17 +02:00
Tim-Philipp Müller
f7aeb57858 tests: add udpsink test to check client add/remove 2014-06-24 10:48:39 +01:00
Tim-Philipp Müller
495dfe3c5b tests: port udpsink tests to 1.0
They all seem a bit pointless though.
2014-06-24 10:48:32 +01:00
Tim-Philipp Müller
c7c72c00b1 rtph264pay: push single buffer directly, no need to wrap it in a bufferlist
No point in a buffer list if we just have one single
buffer to push. Fix up unit test to handle that case
as well.
2014-06-18 14:54:58 +01:00
Olivier Crête
4377dfeadd rtprtx: Reset state on each iteration
Otherwise it didn't wait for the test to finish before checking the results.

https://bugzilla.gnome.org/show_bug.cgi?id=728501
2014-06-03 17:59:32 -04:00
George Kiagiadakis
b19c830a1c tests/check: rtpsession: test internal sources timing out 2014-05-14 16:01:50 +02:00
Wim Taymans
b2e1598e4a rtpjitterbuffer: increment accepted packets after loss
When we detect a lost packet, expect packets with higher
seqnum on the input.

Also update the unit test.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729524
2014-05-09 18:10:32 +02:00
Jason Litzinger
9068e1bb8e Add new test case. 2014-05-09 18:10:32 +02:00
Wim Taymans
5c90f17cf0 shapewipe: no need to activate pads
Activation will happen in the state change
2014-05-09 18:10:31 +02:00
Tim-Philipp Müller
c3bd2bdcf4 tests: fix compilation of souphttpsrc test for libsoup 2.40 for real
https://bugzilla.gnome.org/show_bug.cgi?id=727329
2014-05-07 15:49:39 +01:00
Tim-Philipp Müller
cf94d498e6 tests: fix compilation of souphttpsrc test for libsoup 2.40
SOUP_CHECK_VERSION was only added in 2.41, but we only
depend on 2.40.

https://bugzilla.gnome.org/show_bug.cgi?id=727329
2014-05-07 13:27:10 +01:00
Olivier Crête
985897d8d9 rtpaux/rtprtx: Make tests non-racy
Fix the raciness by iterating on a condition instead of using the gmainloop.
Don't use the EOS as the target, otherwise the retransmission of the last
packets are lost. Also count the retranmissions requests that are dropped.
Check the condition before blocking on the GCond

https://bugzilla.gnome.org/show_bug.cgi?id=728501
2014-05-04 22:37:26 -04:00
Olivier Crête
0742a5a257 rtpmux: Always let upstream chose the ssrc if it wishes 2014-05-04 19:11:03 -04:00
Olivier Crête
2e54d38dd0 rtpsession: Keep local conflicting addresses in the session
As we now replace the local RTPSource on a conflict, it's no longer possible
to keep local conflicts in the RTPSource, so they instead need to be kept
in the RTPSession.

Also fix the rtpcollision test to generate multiple collisions instead of
one by change the address, as otherwise we detected that it was a single one.
2014-05-03 18:30:20 -04:00
Sebastian Dröge
3b5deb2b45 shapewipe: Send initial events after setting the elements to PLAYING
Otherwise we send them too early, and setting the elements to PLAYING
afterwards will drop all the events again.
2014-05-03 11:43:21 +02:00
Sebastian Dröge
2149d5a9bd rtprtx: Don't forget to unmap rtp buffer in the test 2014-04-17 18:07:09 +02:00
Sebastian Dröge
5dba8dfe59 rtprtx: Provide an ssrc in the test
And increase timeout to allow all tests to run in valgrind.
2014-04-17 17:43:12 +02:00
Sebastian Dröge
02d9b5e6f8 rtpsession: Fix memory leaks in test 2014-04-17 17:33:46 +02:00
Sebastian Dröge
02e62c139d rtpjitterbuffer: Fix hundreds of memory leaks in the test 2014-04-17 17:26:36 +02:00
Sebastian Dröge
0c073b2d1d rtpcollision: Fix memory leaks in unit test 2014-04-17 16:39:59 +02:00
Sebastian Dröge
cd4c17031b videomixer: Fix memory leak in unit test 2014-04-16 19:03:47 +02:00
Sebastian Dröge
6c02593386 aacparse: Fix memory leak in the test 2014-04-16 17:35:42 +02:00
Reynaldo H. Verdejo Pinochet
f187d9fc7c tests: souphttpsrc: use SoupKnownStatusCode if needed
From libsoup docs:

Prior to 2.44 SoupStatus was called SoupKnownStatusCode,
but the individual values have always had the names they
have now.

Fixes:
  https://bugzilla.gnome.org/show_bug.cgi?id=727329
2014-04-07 17:48:35 -03:00
Nicolas Dufresne
418a4940a8 v4l2: Use a copy of videodev2.h header
With years the amount of ifdef have grown up and we are not even sure if the
old code path compiles. Each time we need to update the v4l2 framework to add
the new feature, we break compilation on older kernel. With exception of two
controls in the video orientation control, this patch get rid of all ifdef by
including the latest version of videodev2.h inside GStreamer.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723446
2014-03-15 13:54:47 +01:00
Sebastian Rasmussen
1a91ab31d1 tests: Don't build disabled plugins' check tests
https://bugzilla.gnome.org/show_bug.cgi?id=723502
2014-02-26 21:07:57 +01:00
Stefan Sauer
ce683b0031 autodetect: improve the tests
Add fake audio/video sinks. Previously running the test might be flaky due to
the use of real elements (hardware in use), which we don't want to test here.
Add two more tests that check that the fakes are chosen.
2014-02-19 21:07:28 +01:00
Stefan Sauer
02e59756a9 autodetect: fix the disabled test
Use a shared helper for both tests. It turns out that the valgrind variant is
fine (maybe due to picking up pulsesink though).
2014-02-19 11:26:22 +01:00
Stefan Sauer
c40e8b1210 autodetect: remove cruft from the test
Remove the obsolete version check and use the ignore macro for the disabled test.
2014-02-19 11:05:35 +01:00
Wim Taymans
6af234e29e tests: fix typecast to fix compilation 2014-02-14 15:53:55 +01:00
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
Wim Taymans
2e3346eff9 tests: add unit-test for multiple missing packets
Check if multiple missing packets generate retransmission events and that the
retranmission requests are canceled when the missing packet arrives.
2013-09-23 14:45:24 +02:00
Wim Taymans
3b9e2ee5a0 tests: improve debug 2013-09-23 14:45:23 +02:00
Wim Taymans
ee817363c7 tests: add retransmission jitterbuffer test
Store both DTS and PTS on buffers.
Make a queue for srcpad events.
Activate pads after linking so that we don't get RECONFIGURE events.
Add test for retransmission.
2013-09-19 14:32:48 +02:00
Wim Taymans
ee75d4b01e tests: fix comments 2013-09-18 15:08:45 +02:00
Wim Taymans
5c1f2f0045 rtpjitterbuffer: add another test
The test is modified slightly because the late lost packets are only
generated now when a large gap is received.
2013-09-18 14:23:55 +02:00
Wim Taymans
5d5fc03e04 rtpjitterbuffer: round gap duration to multiple of duration
Make sure the gap duration in the lost event is a multiple of the packet
duration.
Enable another test.
2013-09-18 14:12:47 +02:00
Wim Taymans
6e4a051d40 rtpjitterbuffer: keep track of duration
Keep track of the estimated duration of missing packets and use it in the lost
event.
Enable another unit test
2013-09-18 12:29:38 +02:00
Wim Taymans
ac3bb3acf6 rtpjitterbuffer: handle large gaps with one lost event
When we have a large number of missing packets, generate one lost event for all
the packets that have no chance of being pushed out in time.
Fix and activate unit test for large gaps.
2013-09-18 11:59:28 +02:00
Wim Taymans
99675d1161 check: change for videomixer renamed orc file 2013-09-17 15:11:41 +02:00
Wim Taymans
aa9af2c82e examples: we don't need the queue anymore 2013-09-16 15:55:55 +02:00
David Holroyd
a956a6ceb2 rtp: add L24 pay and depayloader
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707734
2013-09-09 15:13:46 +02:00
Tim-Philipp Müller
cc4f985332 tests: rganalysis: rename function for clarity 2013-09-02 22:50:58 +01:00
Christoph Reiter
92dde0760b tests: fix skipped rganalysis tests
In 0.10 elements would post tag messages on the bus
directly, and rganalysis would only post a tag message
when it changed tags. In 1.0, only sinks post tag
messages when they receive the serialised tag event.
This means that we get an additional tag message on
the bus now where we didn't expect one before.

https://bugzilla.gnome.org/show_bug.cgi?id=695090
2013-09-02 20:00:29 +01:00
Wim Taymans
43359b9244 tests: add retransmission example 2013-08-23 12:10:19 +02:00
Thibault Saunier
6c349d6ec3 videomixer: Send EOS if buf_end >= segment.stop
That means the whole segment is already played, and we are sure we
are EOS at that point.

Also handle segment seeks, and do not send EOS in that case.
2013-08-11 19:05:18 +02:00