Commit graph

966 commits

Author SHA1 Message Date
Jordan Petridis
515ada7e22
Run gst-indent through the files
This is required before we enabled an indent test in the CI.

https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/33
2018-11-28 05:52:16 +02:00
Linus Svensson
ac94c706da rtpsession: test: Plug memory leak 2018-11-13 12:30:35 +00:00
Havard Graff
65a7d39bd4 flvmux: Test that timestamps are always increasing
Decreasing timestamps break rtmpsink.

With contributions from Olivier Crête.

https://bugzilla.gnome.org/show_bug.cgi?id=796382
2018-11-05 18:17:04 -05:00
Olivier Crête
cc69c876fe rtpsession: Allow changing the SDES at runtime
Make it possible to modify the SDES in a packet at runtime.

https://bugzilla.gnome.org/show_bug.cgi?id=763502
2018-10-28 12:10:36 +00:00
Yeongjin Jeong
301142604e tests: flvmux: Fix pushing invalid audio caps in tests
Previous commit created caps with incorrect aac codec data
that did not match the audio channel.

https://bugzilla.gnome.org/show_bug.cgi?id=797256
2018-10-21 02:44:24 -04:00
Havard Graff
13aa805943 rtpsession: fix up GHashTable-behavior dependent tests
GHashTable iteration order changed in recent GLib,
and tests were relying on that.

https://mail.gnome.org/archives/desktop-devel-list/2018-October/msg00016.html
2018-10-20 12:32:44 +01:00
Havard Graff
53a45b1222 Initial commit of GstRtpFunnel
For funneling together rtp-streams into a single session.
Use-cases include multiplexing and bundle.
2018-10-15 14:20:58 +02:00
Yeongjin Jeong
afa4be4b3b tests: flvdemux: Add new test for channel detect using aac codec-data
https://bugzilla.gnome.org/show_bug.cgi?id=797275
2018-10-12 14:35:37 -04:00
Yeongjin Jeong
7b5f7249e8 tests: flvmux: Add new test for caps change after starting to write headers
https://bugzilla.gnome.org/show_bug.cgi?id=797256
2018-10-11 15:35:24 -04:00
Havard Graff
6c05180dc5 rtpmux: respect downstream "timestamp-offset" in caps.
https://bugzilla.gnome.org/show_bug.cgi?id=795162
2018-10-10 15:39:02 -04:00
Havard Graff
6f37bd8f19 rtpmux: cleanup ssrc-handling code a bit
And add some better logging.

https://bugzilla.gnome.org/show_bug.cgi?id=795162
2018-10-10 15:38:57 -04:00
Havard Graff
7cd36d2914 rtpmux: property should overrule both upstream and downstream
https://bugzilla.gnome.org/show_bug.cgi?id=762213

https://bugzilla.gnome.org/show_bug.cgi?id=795162
2018-10-10 15:35:31 -04:00
Havard Graff
ac6e77acad rtpsession: Don't start the RTCP thread until it's needed
Always wait with starting the RTCP thread until either a RTP or RTCP
packet is sent or received. Special handling is needed to make sure the
RTCP thread is started when requesting an early RTCP packet.

We want to wait with starting the RTCP thread until it's needed in order
to not send RTCP packets for an inactive source.

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-07-12 18:37:33 +02:00
Seungha Yang
aecc17251d tests: qtdemux: Add checking exposed segment event
https://bugzilla.gnome.org/show_bug.cgi?id=796480
2018-06-06 11:19:25 -04:00
Thiago Santos
0de143fa3e tests: qtdemux: Avoid using data beyond array and improve error msg
Makes it easier to debug the failures as well as prevents problems
reading out of bounds data.
2018-05-28 11:25:13 -07:00
Tim-Philipp Müller
48dd93662d tests: rtpstorage: fix potential crashes / test failures on 32-bit
Pass 64 bits to g_object_set() for 64-bit integer properties like
rtpstorage's "size-time" property.

https://bugzilla.gnome.org/show_bug.cgi?id=796429
2018-05-27 20:30:46 +01:00
Vivia Nikolaidou
d11339d616 splitmuxsink: Added new async-finalize mode
This mode is useful for muxers that can take a long time to finalize a
file. Instead of blocking the whole upstream pipeline while the muxer is
doing its stuff, we can unlink it and spawn a new muxer+sink combination
to continue running normally.

This requires us to receive the muxer and sink (if needed) as factories,
optionally accompanied by their respective properties structures. Also
added the muxer-added and sink-added signals, in case custom code has to
be called for them.

https://bugzilla.gnome.org/show_bug.cgi?id=783754
2018-05-24 12:47:24 +03:00
Havard Graff
77f3ce2e45 rtpsession: Add tests for PLI and FIR
https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-15 11:52:45 +01:00
Stian Selnes
457fdf95c4 rtpsession: Drop packet if trying to send from non-internal source
If obtain_internal_source() returns a source that is not internal it
means there exists a non-internal source with the same ssrc. Such an
ssrc collision should be handled by sending a GstRTPCollision event
upstream and choose a new ssrc, but for now we simply drop the packet.
Trying to process the packet further will cause it to be pushed
usptream (!) since the source is not internal (see source_push_rtp()).

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-15 10:34:29 +01:00
Havard Graff
b43ee8f5b1 rtpsession: Try media_ssrc if no src can be found for PLI sender_ssrc
Some RTP stacks out there does not set the sender_ssrc. In order to be
more robust, try to lookup the media_ssrc before dropping the PLI.

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-13 20:41:39 +01:00
Mikhail Fludkov
386ca1d378 rtpsession: Fix on-feedback-rtcp race
If there is an external source which is about to timeout and be removed
from the source hashtable and we receive feedback RTCP packet with the
media ssrc of the source, we unlock the session in
rtp_session_process_feedback before emitting 'on-feedback-rtcp' signal
allowing rtcp timer to kick in and grab the lock. It will get rid of
the source and rtp_session_process_feedback will be left with RTPSource
with ref count 0.

The fix is to grab the ref to the RTPSource object in
rtp_session_process_feedback.

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-13 20:33:56 +01:00
John-Mark Bell
0a2b55ac3c rtpsession: do not emit RBs for internal senders.
These are the sources we send from, so there is no reason to
report receive statistics for them (as we do not receive on them,
and the remote side has no knowledge of them).

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-13 19:16:59 +01:00
Havard Graff
cd8c12f240 tests: rtpsession: fix indentation
https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-13 19:09:29 +01:00
Seungha Yang
3f090be2d1 tests: qtdemux: Add test for stream change
Add test case to verify track-id change and stream change

https://bugzilla.gnome.org/show_bug.cgi?id=684790
2018-05-10 08:09:20 +02:00
Olivier Crête
168fae813b flvmux: Wait for caps from both srcs before writing header
Wait for caps on all pads to start writing data even when source is live.

Includes unit test by Havard Graff that simulates it.

https://bugzilla.gnome.org/show_bug.cgi?id=794722
2018-04-26 15:41:54 -04:00
Mathieu Duponchelle
90f5ae8f45 ulpfecdec: output perfect seqnums
ULP FEC, as defined in RFC 5109, has the protected and protection
packets sharing the same ssrc, and a different payload type, and
implies rewriting the seqnums of the protected stream when encoding
the protection packets. This has the unfortunate drawback of not
being able to tell whether a lost packet was a protection packet.

rtpbasedepayload relies on gaps in the seqnums to set the DISCONT
flag on buffers it outputs. Before that commit, this created two
problems:

* The protection packets don't make it as far as the depayloader,
  which means it will mark buffers as DISCONT every time the previous
  packets were protected

* While we could work around the previous issue by looking at
  the protection packets ignored and dropped in rtpptdemux, we
  would still mark buffers as DISCONT when a FEC packet was lost,
  as we cannot know that it was indeed a FEC packet, even though
  this should have no impact on the decoding of the stream

With this commit, we consider that when using ULPFEC, gaps in
the seqnums are not a reliable indicator of whether buffers should
be marked as DISCONT or not, and thus rewrite the seqnums on
the decoding side as well to form a perfect sequence, this
obviously doesn't prevent the jitterbuffer from doing its job
as the ulpfec decoder is downstream from it.

https://bugzilla.gnome.org/show_bug.cgi?id=794909
2018-04-19 18:17:39 +02:00
Mathieu Duponchelle
9b1aec0f79 flvmux test: refactor looped test.
Looping the test 500 times to only execute the test once every
33 times means we inited and deinited gstreamer 467 times
for no reason at all, which was annoying when running the test
with valgrind.
2018-04-13 23:02:26 +02:00
Mathieu Duponchelle
ec3c49e958 souphttpsrc test: free g_get_current_dir return 2018-04-13 20:35:24 +02:00
Mathieu Duponchelle
cc9fe814d6 rtpulpfec tests: Fix leaks 2018-04-13 17:37:47 +02:00
Sebastian Dröge
ed2ccb1a60 rtp: Fix compilation with non-C99 compilers
By moving variable declarations out of loop headers.
2018-03-20 12:08:28 +02:00
Olivier Crête
96261ce220 flvmux: Duration & unit tests
The muxed buffers will not carry the duration of the
incoming buffers.

https://bugzilla.gnome.org/show_bug.cgi?id=793457
2018-03-01 18:25:02 -05:00
Mathieu Duponchelle
b0dd092ea6 tests: fix redenc tests
The default of the allow-no-red-blocks property was changed in a
previous commit, thus breaking the test assumptions
2018-02-27 16:34:51 +01:00
Tim-Philipp Müller
7f6aa7c344 .gitignore more test binaries 2018-02-22 10:54:02 +00:00
Mikhail Fludkov
d5ad50bd61 rtp: Implement ULPFEC (RFC 5109)
We expose a set of new elements:

* ULPFEC encoder / decoder
* A storage element, which should be placed before jitterbuffers,
  and is used to store packets in order to attempt reconstruction
  after the jitterbuffer has sent PacketLost events
* RED encoder / decoder (RFC 2198), these are necessary to
  use FEC in webrtc, as browsers will propose and expect ulpfec
  packets to be wrapped in red packets

With contributions from:

Mathieu Duponchelle <mathieu@centricular.com>
Sebastian Dröge <sebastian@centricular.com>

https://bugzilla.gnome.org/show_bug.cgi?id=792696
2018-02-21 14:15:22 +01:00
Jan Alexander Steffens (heftig)
54f312644e tests: aacparser: Test that short raw frames don't get concatenated
https://bugzilla.gnome.org/show_bug.cgi?id=792644
2018-01-18 19:09:25 +00:00
Mathieu Duponchelle
03dc22951b rtpbin: fix leak of elements requested by signals
When the signal returns a floating reference, as its return type
is transfer full, we need to sink it ourselves before passing
it to gst_bin_add (which is transfer floating).

This allows us to unref it in bin_remove_element later on, and
thus to also release the reference we now own if the signal
returns a non-floating reference as well.

As we now still hold a reference to the element when removing it,
we also need to lock its state and setting it to NULL before
unreffing it

Also update the request_aux_sender test.

https://bugzilla.gnome.org/show_bug.cgi?id=792543
2018-01-18 15:26:43 +01:00
Sebastian Rasmussen
0d57709d38 tests: udpsink: add check that sets QoS on IPv4/6 sockets
https://bugzilla.gnome.org/show_bug.cgi?id=757449
2017-12-23 12:45:11 +01:00
fengalin
3464aac3c9 matroska: fix memory leaks due to toc related updates
https://bugzilla.gnome.org/show_bug.cgi?id=790686
2017-12-15 16:14:43 +02:00
Sebastian Dröge
c55824e4fa matroskamux: Fix various memory leaks in the unit test
https://bugzilla.gnome.org/show_bug.cgi?id=790686
2017-12-15 16:14:43 +02:00
fengalin
694c07fe63 matroska-mux: migrate test to gst_harness
... following the guide lines from Håvard Graff (see https://gstconf.ubicast.tv/videos/moar-better-tests/).

https://bugzilla.gnome.org/show_bug.cgi?id=790686
2017-12-15 16:14:43 +02:00
fengalin
a6702a76d5 matroska: re-activate and update TOC support
TOC support in mastroskamux has been deactivated for a couple of years. This commit updates it to recent GstToc evolutions and introduces toc unit tests for both matroska-mux and matroska-demux.

There are two UIDs for Chapters in Matroska's specifications:
- The ChapterUID is a mandatory unsigned integer which internally refers to a given chapter. Except for title & language which use dedicated fields, this UID can also be used to add tags to the Chapter. The tags come in a separate section of the container.
- The ChapterStringUID is an optional UTF-8 string which also uniquely refers to a chapter but from an external perspective. It can act as a "WebVTT cue identifier" which "can be used to reference a specific cue, for example from script or CSS".

During muxing, the ChapterUID is generated and checked for unicity, while the ChapterStringUID receives the user defined UID. In order to be able to refer to chapters from the tags section, we maintain an internal Toc tree with the generated ChapterUID.

When demuxing, the ChapterStringUIDs (if available) are assigned to the GstTocEntries UIDs and an internal toc mimicking the toc is used to keep track of the ChapterUIDs and match the tags with the appropriate GstTocEntries.

https://bugzilla.gnome.org/show_bug.cgi?id=790686
2017-12-15 16:14:43 +02:00
Tim-Philipp Müller
c6b686624a tests: ignore rtph264 test binary 2017-12-09 16:15:24 +00:00
George Kiagiadakis
33bddfe321 tests: udpsrc: verify the correct amount of bytes is sent to the socket
https://bugzilla.gnome.org/show_bug.cgi?id=786799
2017-12-09 16:08:49 +00:00
George Kiagiadakis
ea7d2a0257 tests: udpsrc: ensure test won't timeout if the buffers are already received
Sometimes all the buffers are received before the time we lock the
check_mutex, in which case g_cond_wait will wait forever for another
one. Just check if this is the case before waiting.

https://bugzilla.gnome.org/attachment.cgi?id=358397
2017-12-09 16:08:38 +00:00
George Kiagiadakis
45c82ee798 tests: udpsrc: fix test_udpsrc to actually run and fix locking
Previously this would silently be skipped because 1600 != 1400
and there is no assertion on this call.

Also unlock check_mutex after use.

https://bugzilla.gnome.org/show_bug.cgi?id=786799
2017-12-09 16:05:28 +00:00
Haakon Sporsheim
3c0d006c03 rtpsession: Handle zero length feedback packets
https://bugzilla.gnome.org/show_bug.cgi?id=791074
2017-12-02 13:58:34 +00:00
Havard Graff
96d837b301 tests: rtpsession: refactor tests to use GstHarness
This patch simplifies the tests (44% less code) and
makes them much more readable.

The provided SessionHarness also makes it much easier
to write new tests for rtpsession.

https://bugzilla.gnome.org/show_bug.cgi?id=791070
2017-12-02 13:05:01 +00:00
Jan Schmidt
76e458a119 splitmuxsink: Use muxer reserved space properties if present.
If the use-robust-muxing property is set, check if the
assigned muxer has reserved-max-duration and
reserved-duration-remaining properties, and if so set
the configured maximum duration to the reserved-max-duration
property, and monitor the remaining space to start
a new file if the reserved header space is about to run out -
even though it never ought to.
2017-11-25 00:56:11 +11:00
Jan Schmidt
3a813a0dcc splitmux: Fix file switch-on-caps-change.
Switching to a new fragment because the input caps have
changed didn't properly end the previous file. Use the normal
EOS sequence to ensure that happens. Add a test that it works.
2017-11-24 16:56:03 +11:00
Tim-Philipp Müller
bca8ac2cf0 tests: rtp-payloading: add unit test for rtph264pay codec_data
Make sure no trailing zero bytes sneak into our SPS or PPS.

https://bugzilla.gnome.org/show_bug.cgi?id=732758
2017-11-23 09:36:15 +01:00
Tim-Philipp Müller
a9e57f3608 tests: rtph264depay: add test for using downstream memory allocator 2017-11-23 09:36:00 +01:00
Tim-Philipp Müller
5547901a37 mpg123: hook up to build system
https://bugzilla.gnome.org/show_bug.cgi?id=774252
2017-08-20 15:50:22 +01:00
Tim-Philipp Müller
4b1f43ebe3 Moving mpg123 plugin from -ugly 2017-08-20 13:48:48 +01:00
Sebastian Dröge
7e718d6039 Revert "matroskamux: adjust unit test to modified behaviour"
This reverts commit 8fe478c8a7.

We're back to previous behaviour
2017-07-18 10:08:33 +03:00
Olivier Crête
96e71b0286 rtpsession: Send EOS if all internal sources sent bye
The ones which are not internal should not matter, and we should
wait for all sources to have sent their BYEs.

And add unit test

https://bugzilla.gnome.org/show_bug.cgi?id=773218
2017-07-04 21:14:10 -04:00
Jan Alexander Steffens (heftig)
aa8ac28d86 tests: souphttpsrc: Avoid deprecated ssl-ca-file property
SoupSession's ssl-ca-file property is deprecated. Use the recommended
tls-database property.

This is a bit more complex as it requires creating a GTlsFileDatabase
object for an absolute (!) path to the CA certificates file.

https://bugzilla.gnome.org/show_bug.cgi?id=784005
2017-06-29 15:32:30 -04:00
Jan Alexander Steffens (heftig)
9922091f1b tests: souphttpsrc: Avoid deprecated server ssl properties
The ssl-cert-file and ssl-key-file properties are deprecated. Use the
soup_server_set_ssl_cert_file function to load the files.

https://bugzilla.gnome.org/show_bug.cgi?id=784005
2017-06-29 15:32:30 -04:00
Jan Alexander Steffens (heftig)
27a0ea8cf5 tests: souphttpsrc: Make ssl_cert/key_file static
Just a bit of cleanup.

https://bugzilla.gnome.org/show_bug.cgi?id=784005
2017-06-29 15:32:30 -04:00
Tim-Philipp Müller
dd23afb6d4 sys: remove sunaudio plugin
Even though hooked up to the build system, it's clear that no one
has ever built or used this with GStreamer 1.x. It wants to link
against libgstinterfaces, which no longer exists. And uses 0.10-style
raw audio caps. And the last meaningful change was done in 2009.
Let's just remove it.
2017-06-23 20:02:43 +01:00
Tim-Philipp Müller
c35292505b meson: add options to set package name and origin
https://bugzilla.gnome.org/show_bug.cgi?id=782172
2017-05-20 14:53:42 +01:00
Tim-Philipp Müller
4df3669c0c tests: rtp-payloading: add test for rtph264depay avc/byte-stream output
Make sure avc output doesn't contain SPS/PPS inline, but
byte-stream output does.
2017-04-24 17:31:04 +01:00
Edward Hervey
7e9b7658e5 tests: Add vp9enc to gitignore 2017-04-12 11:33:05 +02:00
George Kiagiadakis
21f532f1c6 tests/check/rtprtx: add checks for rtprtxqueue's max-size-{time,packets} properties
https://bugzilla.gnome.org/show_bug.cgi?id=780867
2017-04-11 09:44:33 +03:00
Vincent Penquerc'h
d7212dac2e tests: fix leak in splitmux test
https://bugzilla.gnome.org/show_bug.cgi?id=781025
2017-04-09 11:19:56 +03:00
Jan Schmidt
57939fd98a splitmux test: Use passed first/last timestamps
Don't hard-code the expected timestamp range, use the
values the caller is passing in.
2017-03-14 15:48:08 +11:00
Nicolas Dufresne
27303b5904 tests: Add missing LDADD for libm in tests using math.h
Also, remove the math.h include for the one that just prentend to need
it.
2017-03-08 22:55:09 -05:00
Jan Schmidt
4335c4c160 splitmux: Add unit test for reverse playback
Ensure that reverse playback works and generates the range
of timestamps (0-3s) we expect, in monotonically descending order.
2017-03-04 00:35:32 +11:00
Sebastian Dröge
eefcdc9ee1 rtp-payloading: Add new test for Vorbis renegotiation
Check if encoding, payloading, depayloading and decoding works if the
stream configuration (and thus the headers) change.
2017-02-27 19:25:35 +02:00
George Kiagiadakis
e6bd2a5c18 tests: splitmux: add unit test for content with sparse streams
https://bugzilla.gnome.org/show_bug.cgi?id=761086
2017-02-27 12:58:21 +02:00
Guillaume Desmottes
0f719af307 tests: matroskamux, qtmux: don't add codec_data buffers to template caps
streamheader and codec_data buffers fields are only meant to be
in the negotiated caps, not the template caps.

Fixes false-positive leaks of those buffers detected by the leaks
tracer, as template caps are static, and we decided to not include
code in gstreamer core to handle this unusual case of template caps
having buffers in them.

https://bugzilla.gnome.org/show_bug.cgi?id=768762
2017-02-21 15:47:16 +00:00
Søren Juul
1184429e21 icydemux: reset tags on empty value
Some radio streams uses StreamTitle='' to reset the title after a
track stopped playing, e.g. while the host talks between tracks or
during news segments.
This change forces an empty tag object to be distributed if
StreamTitle or StreamUrl is received with empty value, thus allowing
downstream elements to get notified about this.

https://bugzilla.gnome.org/show_bug.cgi?id=778437
2017-02-14 12:24:13 +02:00
Tim-Philipp Müller
781b5ac781 tests: rtpjitterbuffer: fix compiler warning due to c99-ism
rtpjitterbuffer.c:592:3: error: ‘for’ loop initial declarations are only allowed in C99 mode
2017-01-09 19:04:04 +00:00
Jan Schmidt
f7009eb5d7 splitmuxsink: Add format-location-full signal
Add a new signal for formatting the filename, which receives
a GstSample containing the first buffer from the reference
stream that will be muxed into that file.

Useful for creating filenames that are based on the
running time or other attributes of the buffer.

To make it work, opening of files and setting filenames is
now deferred until there is some data to write to it,
which also requires some changes to how async state changes
and gap events are handled.
2017-01-03 01:34:02 +11:00
Edward Hervey
3a4d4dcd27 check: Remove dead code 2017-01-02 15:06:33 +01:00
Nicola Murino
8fe478c8a7 matroskamux: adjust unit test to modified behaviour
Now matroskamux mark all packets of audio-only streams as keyframes so
in test_block_group after pushing the test audio data 4 buffers are produced
and not more 2. The last buffer is the original data and must match with what
pushed. The remaining ones are matroskamux headers

https://bugzilla.gnome.org/show_bug.cgi?id=754696
2016-12-21 16:58:42 +00:00
Havard Graff
0a81f71df5 tests/jitterbuffer: Major refactoring and cleanups
* Changed PCMU->TEST for common macros
* Changed verify-functions (lost & rtx) into macros.
* Remove option to add marker-bit for test-buffers (not used anywhere)
* Add new push_test_buffer function that makes sure there are correlation
  between dts and the time on the clock. (classic test-mistake)
* Established a generic starting-point for tests with the
  construct_deterministic_initial_state function and use it where
  applicable, which removes lots of "boilerplate" everywhere.
* Add basic lost-event test
* Remove as much "magic constants" as possible.
* Remove 3 tests that no longer are testing anything that others don't,
  and was completely unmaintainable.
* Remove unnecessary use of the testclock
* Verify each test is testing what it actually says it does (and modify
  where it doesn't)

In general, make the tests much smaller, better, more maintainable and
readable.

https://bugzilla.gnome.org/show_bug.cgi?id=774409
2016-12-14 15:00:37 +02:00
Sebastian Dröge
63938ef730 gst: Don't declare variables inside the for loop header
This is a C99 feature.
2016-12-13 22:32:46 +02:00
Philippe Normand
dcd3ce9751 rtpbin: receive bundle support
A new signal named on-bundled-ssrc is provided and can be
used by the application to redirect a stream to a different
GstRtpSession or to keep the RTX stream grouped within the
GstRtpSession of the same media type.

https://bugzilla.gnome.org/show_bug.cgi?id=772740
2016-11-16 08:56:34 +01:00
Havard Graff
1a4393fb4d rtpjitterbuffer: fix timer-reuse bug
When doing rtx, the jitterbuffer will always add an rtx-timer for the next
sequence number.

In the case of the packet corresponding to that sequence number arriving,
that same timer will be reused, and simply moved on to wait for the
following sequence number etc.

Once an rtx-timer expires (after all retries), it will be rescheduled as
a lost-timer instead for the same sequence number.

Now, if this particular sequence-number now arrives (after the timer has
become a lost-timer), the reuse mechanism *should* now set a new
rtx-timer for the next sequence number, but the bug is that it does
not change the timer-type, and hence schedules a lost-timer for that
following sequence number, with the result that you will have a very
early lost-event for a packet that might still arrive, and you will
never be able to send any rtx for this packet.

Found by Erlend Graff - erlend@pexip.com

https://bugzilla.gnome.org/show_bug.cgi?id=773891
2016-11-04 16:56:56 +02:00
Havard Graff
fb9c75db36 rtpjitterbuffer: fix lost-event using dts instead of pts
The lost-event was using a different time-domain (dts) than the outgoing
buffers (pts). Given certain network-conditions these two would become
sufficiently different and the lost-event contained timestamp/duration
that was really wrong. As an example GstAudioDecoder could produce
a stream that jumps back and forth in time after receiving a lost-event.

The previous behavior calculated the pts (based on the rtptime) inside the
rtp_jitter_buffer_insert function, but now this functionality has been
refactored into a new function rtp_jitter_buffer_calculate_pts that is
called much earlier in the _chain function to make pts available to
various calculations that wrongly used dts previously
(like the lost-event).

There are however two calculations where using dts is the right thing to
do: calculating the receive-jitter and the rtx-round-trip-time, where the
arrival time of the buffer from the network is the right metric
(and is what dts in fact is today).

The patch also adds two tests regarding B-frames or the
“rtptime-going-backwards”-scenario, as there were some concerns that this
patch might break this behavior (which the tests shows it does not).
2016-11-04 16:51:20 +02:00
Havard Graff
bea35f97c8 rtpjitterbuffer: fix bug in reschedule_timer
The new timeout is always going to be (timeout + delay), however, the
old behavior compared the current timeout to just (timeout), basically
being (delay) off.

This would happen if rtx-delay == rtx-retry-timeout, with the result that
a second rtx attempt for any buffers would be scheduled immediately instead
of after rtx-delay ms.

Simply calculate (new_timeout = timeout + delay) and then use that instead.

https://bugzilla.gnome.org/show_bug.cgi?id=773905
2016-11-04 16:40:14 +02:00
Tim-Philipp Müller
752dd15c54 tests: wavparse: add test for processing an actual .wav file
https://bugzilla.gnome.org/show_bug.cgi?id=773861
2016-11-03 15:42:29 +02:00
Havard Graff
78ab8cbdcd rtph263ppay: Fix caps leak
Fix leaking caps when downstream has not-fixed caps.

https://bugzilla.gnome.org/show_bug.cgi?id=773515
2016-11-01 20:20:47 +02:00
Tim-Philipp Müller
834339b773 tests: videomixer: disable racy flush_start_flush_stop test
It's been broken for years, and it's unlikely it will ever
be fixed for collectpads/videomixer now that there's compositor
which works fine. So let's disable it, since all it does
is that it creates noise that distracts from other failures.

Also see the corresponding adder bug as it failed in the same way:
 https://bugzilla.gnome.org/show_bug.cgi?id=708891
2016-10-20 22:08:14 +01:00
Jan Alexander Steffens (heftig)
6deab72e10 tests: Fix souphttpsrc tests without CK_FORK=no
It seems that the forked processes all attempt to handle the listening
socket from the server, and only one has to shutdown the socket to break
the server completely.

Create a new server inside each test to avoid this.

https://bugzilla.gnome.org/show_bug.cgi?id=772656
2016-10-20 13:29:07 +03:00
Jan Alexander Steffens (heftig)
22ced681af tests: Fix level test in CK_FORK=no mode
The tests accumulate buffers in GstCheck's buffers list, and the list is
not (consistently) reset between tests. Do that and remove the now
conflicting unrefs for outbuffers.

https://bugzilla.gnome.org/show_bug.cgi?id=772644
2016-10-20 13:23:30 +03:00
Tim-Philipp Müller
e6d188967a tests: fix indentation 2016-09-15 09:53:07 +01:00
Havard Graff
f440b074b1 rtpjitterbuffer: improved rtx-rtt averaging
The basic idea is this:
1. For *larger* rtx-rtt, weigh a new measurement as before
2. For *smaller* rtx-rtt, be a bit more conservative and weigh a bit less
3. For very large measurements, consider them "outliers"
   and count them a lot less

The idea being that reducing the rtx-rtt is much more harmful then
increasing it, since we don't want to be underestimating the rtt of the
network, and when using this number to estimate the latency you need for
you jitterbuffer, you would rather want it to be a bit larger then a bit
smaller, potentially losing rtx-packets. The "outlier-detector" is there
to prevent a single skewed measurement to affect the outcome too much.
On wireless networks, these are surprisingly common.

https://bugzilla.gnome.org/show_bug.cgi?id=769768
2016-09-14 19:37:50 -04:00
Stian Selnes
f8238f0a9f rtpjitterbuffer: Detect whether to assume equidistant spacing when loss
Assuming equidistant packet spacing when that's not true leads to more
loss than necessary in the case of reordering and jitter. Typically this
is true for video where one frame often consists of multiple packets
with the same rtp timestamp. In this case it's better to assume that the
missing packets have the same timestamp as the last received packet, so
that the scheduled lost timer does not time out too early causing the
packets to be considered lost even though they may arrive in time.

https://bugzilla.gnome.org/show_bug.cgi?id=769768
2016-09-14 19:37:50 -04:00
Stian Selnes
2eb7383816 rtpjitterbuffer: Don't request rtx if 'now' is past retry period
There is no need to schedule another EXPECTED timer if we're already
past the retry period. Under normal operation this won't happen, but if
there are more timers than the jitterbuffer is able to process in
real-time, scheduling more timers will just make the situation worse.
Instead, consider this packet as lost and move on. This scenario can
occur with high loss rate, low rtt and high configured latency.

https://bugzilla.gnome.org/show_bug.cgi?id=769768
2016-09-14 19:37:50 -04:00
Stian Selnes
ab49dfd0b2 rtpjitterbuffer: Fix lost duration when gap after lost timer
This patch fixes an issue with the estimated gap duration when there is
a gap immediately after a lost timer has been processed. Previously
there was a discrepancy beteen the gap in seqnum and gap in dts which
would cause wrong calculated duration. The issue would only be seen with
retranmission enabled since when it's disabled lost timers are only
created when a packet is received and the actual gap length and last dts
is known.

https://bugzilla.gnome.org/show_bug.cgi?id=769768
2016-09-14 19:37:50 -04:00
Havard Graff
8087a8a31c rtpjitterbuffer: Improved expected-timer handling when gap > 0
https://bugzilla.gnome.org/show_bug.cgi?id=769768
2016-09-14 19:37:50 -04:00
Stian Selnes
38a7545003 rtpjitterbuffer: Major improvements for RTX stats
Stats should also be collected for unsuccessful packets.

rtx-rtt is very important for determining the necessary configured
latency on the jitterbuffer. It's especially important to be able to
increase the latency when retransmitted packets arrive too late and are
considered lost. This patch includes these late packets in the
calculation of the various rtx stats, making them more correct and
useful.

Also in the case where the original packet arrives after a NACK is sent,
the received RTX packet should update the stats since it provides useful
information about RTT.

The RTT is only updated if and only if all requested retranmissions are
received. That way the RTT is guaranteed to make sense. If not we don't
know which request the packet is a response to and the RTT may be bogus.
A consequence of this patch is that RTT is not updated for a request
when one of the RTX packets for that seqnum is lost, but that since
measured RTT will be more accurate.

The implementation store the RTX information from the timed out timers
and use this when the retransmitted packet arrives. For performance
these timers are stored separately from the "normal" timers in order to
not impact performance (see attached performance test).

https://bugzilla.gnome.org/show_bug.cgi?id=769768
2016-09-14 19:37:50 -04:00
Havard Graff
1b868cc9b1 rtpjitterbuffer: Add and expose more stats and increase testing of it
Add num-pushed and num-lost.
Expose num-late, num-duplicates and avg-jitter.

https://bugzilla.gnome.org/show_bug.cgi?id=769768
2016-09-14 19:37:50 -04:00
Josep Torra
d40f007d61 gitignore: ignore qtdemux, rtph261 and rtpvp9 tests 2016-08-26 21:32:07 +02:00
Josep Torra
ccc7d7e5a3 tests: remove a wrong 'const' specifier
Fixes "error: duplicate 'const' declaration specifier"
2016-08-26 21:14:47 +02:00
Stian Selnes
8bf77e34f2 rtpvp9depay: Support flexible mode 2016-08-26 11:57:15 -04:00
Stian Selnes
195d181828 vp9enc: Fix leak of vpx_image_t 2016-08-26 11:57:15 -04:00
Stian Selnes
5f3b570d53 rtph263pdepay: Don't try to push empty frame
If the result of depayloading is an empty frame, just drop it. This is
likely the result of a buggy payloader.
2016-08-26 11:57:15 -04:00
Stian Selnes
11b7575cff rtph263pdepay: Fix picture header for non-writable payload
Under certain conditions gst_rtp_buffer_get_payload() returns a copy of
the payload. In this case the payload modifications will not affect the
rtp buffer. So instead of modifying the payload buffer directly we
should modify the buffer that actually gets pushed on the adapter.
2016-08-26 11:53:22 -04:00
Stian Selnes
793327cce2 rtph261depay: Fix check of valid payload length
Packets with no H.261 payload should be dropped to avoid invalid
write/reads.
2016-08-26 11:53:22 -04:00
Stian Selnes
64f9d08d3d rtph263pay: Fix double free, invalid reads and leak 2016-08-26 11:53:22 -04:00
Mikhail Fludkov
880f494050 tests/rtprtx: refactor the tests to use gstharness
The functionality of all the tests was kept exactly the same. Some tests
were renamed:
test_push_forward_seq -> test_rtxsend_rtxreceive
test_drop_one_sender -> test_rtxsend_rtxreceive_with_packet_loss
test_drop_multiple_sender -> test_multi_rtxsend_rtxreceive_with_packet_loss

test_rtxreceive_data_reconstruction was testing that retransmitted
buffer produced by rtxsend was correctly transformed to the original
buffer by rtxreceive. Now we are checking for this in all the tests
where both rtxsend & rtxreceive are involved. That's why the test was
removed.
2016-08-25 18:21:10 -04:00
Sebastian Dröge
a1eefe23de rtpjitterbuffer: Fix unit test by disabling adaptive misorder/dropout calculations
Need to set max-misorder-time and max-dropout-time to 0 so the
jitterbuffer does not base them on packet rate calculations.
If it does, out gap is big enough to be considered a new stream and
we wait for a few consecutive packets just to be sure

https://bugzilla.gnome.org/show_bug.cgi?id=751311
2016-08-18 09:58:58 +03:00
Guillaume Desmottes
7da2bac2e3 tests: qtdemux: fix element and pad leak
https://bugzilla.gnome.org/show_bug.cgi?id=768739
2016-07-18 10:54:59 +01:00
Guillaume Desmottes
94232da665 tests: fix bus leaks
gst_bus_add_signal_watch() takes a ref on the bus which should be
released using gst_bus_remove_signal_watch().

https://bugzilla.gnome.org/show_bug.cgi?id=768739
2016-07-18 10:53:19 +01:00
Jonas Holmberg
833c530553 rtph265pay: Accept array_completeness=1
When parsing NAL unit type in codec_data, check the 6bits of
NAL_unit_type only and do not require the array_completeness bit to be
0, since the default and mandatory value of array_completeness is 1 for
hvc1.

https://bugzilla.gnome.org/show_bug.cgi?id=768653
2016-07-11 11:49:41 +03:00
Jonas Holmberg
a06152c40a rtph265pay/depay: Sync against RFC 7798
Handle sprop-vps, sprop-sps and sprop-pps in caps instead of
sprop-parameter-sets.

rtph265pay works with byte-stream and hvc1 formats but not hev1 yet. It
handles profile-id, tier-flag and level-id in caps query.

https://bugzilla.gnome.org/show_bug.cgi?id=753760
2016-07-07 14:59:50 +03:00
Sebastian Dröge
6289280535 qtmux: Use complete AAC caps with codec_data in the tests 2016-07-04 17:45:40 +02:00
Edward Hervey
e3923df800 qtdemux: Handle upstream GAP in push-mode/time segment
This is to handle cases where upstream handles the fragmented streaming in TIME
segments and sends us data with gaps within fragments. This would happen when dealing
with trick-modes.

When upstream (push-based, TIME SEGMENT) wishes to send discontinuous samples,
it must obey the following rules:
* The buffer containing the [moof] must have a valid GST_BUFFER_OFFSET
* The buffers containing the first sample after a gap:
 * MUST start at the beginning of a sample,
 * MUST have the DISCONT flag set,
 * MUST have a valid GST_BUFFER_OFFSET relative to the beginning of the fragment.

https://bugzilla.gnome.org/show_bug.cgi?id=767354
2016-07-01 14:21:04 +02:00
Tim-Philipp Müller
c68b7f944a tests: splitmux: skip tests if theora or ogg plugins are not available
https://bugzilla.gnome.org/show_bug.cgi?id=767861
2016-06-21 17:54:21 +01:00
Guillaume Desmottes
6ae9c23fa3 fix buffer leaks in tests
Need to call gst_check_drop_buffers() to release the buffers exchanged
during the test.

https://bugzilla.gnome.org/show_bug.cgi?id=766561
2016-06-21 10:51:08 +03:00
Guillaume Desmottes
fb41b307a6 interleave: fix message leaks in test
Flush the bus when cleaning up so pending messages are destroyed.

https://bugzilla.gnome.org/show_bug.cgi?id=766561
2016-06-21 10:51:08 +03:00
Guillaume Desmottes
23c8f7128b videomixer: fix event leaks in test
https://bugzilla.gnome.org/show_bug.cgi?id=766561
2016-06-21 10:51:08 +03:00
Guillaume Desmottes
c7621e24ec deinterleave: fix leaks
- Flush the bus so messages aren't leaked
- Fix pad leak

https://bugzilla.gnome.org/show_bug.cgi?id=766561
2016-06-21 10:51:08 +03:00
Guillaume Desmottes
57eb9fec72 tests: rtpbin: fix caps leak
https://bugzilla.gnome.org/show_bug.cgi?id=767156
2016-06-02 14:19:02 +01:00
Guillaume Desmottes
507e99cb96 tests: amrparse: clean up test
- use GST_CHECK_MAIN() to reduce boilerplate
- unref the input caps using a teardown function to prevent leaks

https://bugzilla.gnome.org/show_bug.cgi?id=767156
2016-06-02 14:18:53 +01:00
Mikhail Fludkov
ee7e80d615 rtpsession: don't act on suspicious BYE RTCP
Some endpoints (like Tandberg E20) can send BYE packet containing our
internal SSRC. I this case we would detect SSRC collision and get rid
of the source at some point. But because we are still sending packets
with that SSRC the source will be recreated immediately.
This brand new internal source will not have some variables incorrectly
set in its state. For example 'seqnum-base` and `clock-rate` values will be
-1.
The fix is not to act on BYE RTCP if it contains internal or unknown
SSRC.

https://bugzilla.gnome.org/show_bug.cgi?id=762219
2016-05-20 09:28:39 +03:00
Mikhail Fludkov
fa1c711a2f rtpsession: Add test for locking of the stats signal
Keeping the lock while emitting the stats signal introduces potential
deadlock in those situations when the signal callback wants the access
to rtpsession's properties which also requre the lock.

https://bugzilla.gnome.org/show_bug.cgi?id=762216
2016-05-20 09:26:20 +03:00
Thiago Santos
79e52b9f81 tests: souphttpsrc: replace deprecated API
Avoid using soup_server_run_async and old get_port() APIs,
replace with me soup_server_listen and get the port through the
URIs list returned from the server.
2016-05-14 08:40:12 -03:00
Havard Graff
8f7962e1c3 rtpjitterbuffer: Fix stall when receiving already lost packet
When a packet arrives that has already been considered lost as part of a
large gap the "lost timer" for this will be cancelled. If the remaining
packets of this large gap never arrives, there will be missing entries
in the queue and the loop function will keep waiting for these packets
to arrive and never push another packet, effectively stalling the
pipeline.

The proposed fix conciders parts of a large gap definitely lost (since
they are calculated from latency) and ignores the late arrivals.

In practice the issue is rare since large gaps are scheduled immediately,
and for the stall to happen the late arrival needs to be processed
before this times out.

https://bugzilla.gnome.org/show_bug.cgi?id=765933
2016-05-06 14:32:42 +03:00
George Kiagiadakis
c0dd2029e9 tests: add splitmuxsrc test for new "format-location" signal
https://bugzilla.gnome.org/show_bug.cgi?id=753625
2016-05-05 10:53:23 +01:00
Tim-Philipp Müller
03e2655f70 tests: add unit test for jpeg depayloader packet loss handling
Make sure it always outputs something that looks like a valid
JPEG frame, ie. starts with an SOI marker and ends with an EOI
marker.
2016-04-04 17:42:03 +01:00
Stian Selnes
4c0e509328 rtpsession: Add new signal 'on-app-rtcp'
Similar to the 'on-feedback-rtcp' signal, but emitted for RTCP APP
packets.

https://bugzilla.gnome.org/show_bug.cgi?id=762217
2016-03-30 15:42:01 +03:00
Thiago Santos
d738fa0787 splitmuxsink: only try to create internal sink if it doesn't exist
This allows splitmuxsink to be reused after being put to NULL.

Test included

https://bugzilla.gnome.org/show_bug.cgi?id=762893
2016-03-24 20:10:25 -03:00
Edward Hervey
c4f06420b3 check: Fix indentation 2016-03-24 16:22:31 +01:00
Edward Hervey
f0a27084f9 tests: Remove unused variables 2016-03-24 16:22:28 +01:00
Vivia Nikolaidou
02a932d789 deinterlace: Added unit tests for field=auto
https://bugzilla.gnome.org/show_bug.cgi?id=763869
2016-03-24 14:34:11 +02:00
Havard Graff
bcbb8fc1da flvdemux: don't emit pad-added until caps are ready
In other words, gst_pad_get_current_caps should never return NULL
in a pad-added callback from the demuxer.

Added tests for the two special cases with AAC and H.264 where this
would happen every time.

https://bugzilla.gnome.org/show_bug.cgi?id=763780
2016-03-24 14:33:33 +02:00
Vineeth TM
1071309870 good: use new gst_element_class_add_static_pad_template()
https://bugzilla.gnome.org/show_bug.cgi?id=763076
2016-03-24 14:32:20 +02:00
David Buchmann
2b8b5f2246 flvmux: Test to verify flvmux handles DTS with GST_CLOCK_TIME NONE
https://bugzilla.gnome.org/show_bug.cgi?id=762207
2016-03-24 14:30:21 +02:00
Tim-Philipp Müller
a4d64b5caa rgvolume: make tag list writable before modifying it
Making the event itself writable is not enough, it won't make
the actual taglist in the event writable as well. Instead, just
make a copy of the taglist and then create a new tag event from
that if required, replacing the old one. Before we would
inadvertently modify taglists upstream elements might still
be holding on to. Add unit test for this as well.

https://bugzilla.gnome.org/show_bug.cgi?id=762793
2016-02-28 14:44:39 +00:00
Tim-Philipp Müller
7335d03070 tests: fix indentation 2016-02-19 14:44:11 +00:00
Havard Graff
69436d5a61 tests: rtpjitterbuffer: port testharness to GstHarness and cleanup/improve
Probably found a bug as well, in that there are some timestamps in
there that are looking very wrong. (marked with FIXME)

https://bugzilla.gnome.org/show_bug.cgi?id=762267
2016-02-19 14:44:02 +00:00
Havard Graff
d52765fabb tests: rtpjitterbuffer: test cleanups/improvements
Use fail_unless and friends instead of g_assert
Factor seq-num checking out to separate function
Check more return-values from push and crank and others

https://bugzilla.gnome.org/show_bug.cgi?id=762254
2016-02-19 11:26:45 +00:00
Stian Selnes
fb4c2909ca tests: rtpjitterbuffer: fix leaks in unit test
https://bugzilla.gnome.org/show_bug.cgi?id=762214
2016-02-19 11:07:52 +00:00
Stian Selnes
3eeca9c7d2 rtpjitterbuffer: Add test for big seqnum gap handling
Make sure that the packets queued when detecting a big gap are pushed
after reset (5 consective seqnums) and not dropped.

https://bugzilla.gnome.org/show_bug.cgi?id=762211
2016-02-18 09:39:01 +02:00
Tim-Philipp Müller
52bd182e98 tests: fix mpg123audiodec test for big-endian architectures 2016-02-16 10:40:39 +00:00
Sebastian Dröge
265ecb5171 mpg123audiodec: Fix event handling in unit test 2016-02-16 10:40:39 +00:00
Vineeth TM
03e40efb65 tests: rtpmux: Fix element memory leak
https://bugzilla.gnome.org/show_bug.cgi?id=762057
2016-02-15 09:55:30 +00:00
Sebastian Dröge
5d728b3ce5 deinterlace: Add negotiation unit tests for all 4 modes
These now check the output caps based on the input caps and a following
capsfilter and make sure the caps are exactly as expected.

https://bugzilla.gnome.org/show_bug.cgi?id=760995
https://bugzilla.gnome.org/show_bug.cgi?id=720388
2016-01-27 16:45:29 +01:00
Sebastian Dröge
98fddf090c rganalysis: Fix compiler warnings in the unit test
elements/rganalysis.c:919:66: error: shifting a negative signed value is undefined
      [-Werror,-Wshift-negative-value]
  push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, -1 << 14, 0));
                                                              ~~ ^
elements/rganalysis.c:929:69: error: shifting a negative signed value is undefined
      [-Werror,-Wshift-negative-value]
  push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 0, -1 << 14));
                                                                 ~~ ^
elements/rganalysis.c:939:64: error: shifting a negative signed value is undefined
      [-Werror,-Wshift-negative-value]
  push_buffer (test_buffer_const_int16_mono (8000, 16, 512, -1 << 14));
                                                            ~~ ^
2016-01-08 15:32:47 +02:00
Reynaldo H. Verdejo Pinochet
0b234c9b54 tests: souphttpsrc: grammar fix 2015-12-01 11:27:17 -08:00
Reynaldo H. Verdejo Pinochet
f9b5271694 tests: souphttpsrc: switch shoutcast stream provider
Fixes failing ICY test. Previous provider has
streaming disabled outside UK.

https://bugzilla.gnome.org/show_bug.cgi?id=758114
2015-12-01 11:27:17 -08:00
Tim-Philipp Müller
3026d1094b rtph264pay: change config-interval property type from uint to int
This way we can use -1 as special value, which is nicer than MAXUINT.
This is backwards compatible even with the GValue API, as shown by
a unit test.

https://bugzilla.gnome.org/show_bug.cgi?id=757892
2015-11-27 12:48:09 +00:00
Josep Torra
f8b9360dad tests: rtp-payloading: Test for handling of custom events in rtpgst
Add a simple test that checks proper serialization/deserialization
of custom events with rtpgstpay and rtpgstdepay.
2015-11-17 17:24:28 -08:00
George Kiagiadakis
a4c8bdfb3c tests/check/splitmux: test that the release_pad vfunc of splitmuxsink actually releases pads
https://bugzilla.gnome.org/show_bug.cgi?id=753622
2015-10-28 22:39:44 +11:00
Thiago Santos
cf830a55b1 tests: deinterlace: fix small typo in comment 2015-10-25 10:55:55 -03:00
Havard Graff
240b0ac9f6 flvdemux: output speex vorbiscomment as a GstTagList
This is what speexdec expects.

https://bugzilla.gnome.org/show_bug.cgi?id=755478
2015-10-11 11:12:27 +01:00
Havard Graff
b6f133ba17 flvmux: GST_BUFFER_OFFSETs should be GST_BUFFER_OFFSET_NONE
Or else flvdemux don't understand it

https://bugzilla.gnome.org/show_bug.cgi?id=754435
2015-10-11 11:10:20 +01:00
Havard Graff
cf3a2294da flvmux: use time segment and copy timestamps when streamable
Add a basic test using speex data to verify timestamping.

https://bugzilla.gnome.org/show_bug.cgi?id=754435
2015-10-11 11:09:08 +01:00
Havard Graff
d5e26ab909 gstrtpmux: allow the ssrc-property to decide ssrc on outgoing buffers
By not doing this, the muxer is not effectively a rtpmuxer, rather a
funnel, since it should be a single stream that exists the muxer.

If not specified, take the first ssrc seen on a sinkpad, allowing upstream
to decide ssrc in "passthrough" with only one sinkpad.

Also, let downstream ssrc overrule internal configured one

We hence has the following order for determining the ssrc used by
rtpmux:

0. Suggestion from GstRTPCollision event
1. Downstream caps
2. ssrc-Property
3. (First) upstream caps containing ssrc
4. Randomly generated

https://bugzilla.gnome.org/show_bug.cgi?id=752694
2015-10-02 17:39:06 -04:00
Thiago Santos
5c7b051b90 deinterleave: implement accept-caps
Avoid using default accept-caps handler that will query downstream
and is more expensive. Just check if the caps is compatible with
the template and check if the channels are the same.
2015-09-30 17:35:33 -03:00
Thiago Santos
c0c8d503da tests: deinterleave: also check for caps query results 2015-09-30 12:48:30 -03:00
Tim-Philipp Müller
81a76853cf tests: gdkpixbufoverlay: add minimal unit test
https://bugzilla.gnome.org/show_bug.cgi?id=755773
2015-09-29 11:15:35 +01:00
Olivier Crête
7cc59fcdf6 tests: Fix rtpsession test failure
The time of the first RTCP packet is semi-random, so
sometimes it was produced before enough packets from
the second SSRC were received. First drop queued RTCP
packets, then advance the clock enough to ensure
that at least one new RTCP packet is produced.

https://bugzilla.gnome.org/show_bug.cgi?id=750731
2015-08-31 16:42:30 -04:00
Stefan Sauer
22443b2eed level: improve the test for multi-channel mode
Change the test to verify the read-index for multiple messages per buffer.
See https://bugzilla.gnome.org/show_bug.cgi?id=754144
2015-08-31 13:57:33 +02:00
Tim-Philipp Müller
dd1bd2beb3 tests: souphttpsrc: don't try to connect to dead radio server 2015-08-21 11:52:19 +01:00
Thiago Santos
2b1db23175 tests: aacparse: use caps query instead of accept-caps
The accept-caps query just does a shallow check at the current
element while at this test we want it to also look at downstream.
So use caps query there.

https://bugzilla.gnome.org/show_bug.cgi?id=753623
2015-08-14 13:42:27 -03:00
Thiago Santos
dac431ef3f tests: rtpaux: use a dynamic pt in the test
1) Tests that using dynamic PT instead of the default ones work
2) If we ever decide to change the codec here we don't need to
   worry about change the PT for the default one of the new codec
   in the test

https://bugzilla.gnome.org/show_bug.cgi?id=746445
2015-08-06 01:39:43 -03:00
Thiago Santos
5f9e5bf385 tests: rtpaux: fix test failure
The RTP PT for alaw is 8.
Less than 50 packets are received in the length of this test so it
would never drop a buffer or would drop only the last buffer and
it would fail sometimes when the received wouldn't receive the
retransmission packet in time.

https://bugzilla.gnome.org/show_bug.cgi?id=746445
2015-08-04 18:25:29 -03:00
Tim-Philipp Müller
c1382e97fa tests: add minmal matroskademux test for subtitle output
Some of the subtitle chunks will have embedded
NUL-terminators (last three), some don't (first three),
some will have markup, some won't, some will be valid
UTF-8 (all but last), some won't (last stanza).

https://bugzilla.gnome.org/show_bug.cgi?id=752421
2015-07-21 14:25:12 +01:00
Havard Graff
764bbf99a8 rtpmux: handle different ssrc's on sinkpads
Do this by not putting the ssrc from the src pads in the caps used to
probe other sinkpads, and then  intersecting with it later.

https://bugzilla.gnome.org/show_bug.cgi?id=752491
2015-07-16 16:46:11 -04:00
Thiago Santos
a1bee6eb46 gitignore: ignore rtph263 test 2015-07-09 09:26:09 -03:00
Thiago Santos
241e0c2722 rtpjitterbuffer: fix build error with gcc (Debian 4.9.2-21) 4.9.2
Replace static constants with macros to make gcc happy

  CC       elements/elements_rtpjitterbuffer-rtpjitterbuffer.o
elements/rtpjitterbuffer.c:387:1: error: initializer element is not constant
 static const GstClockTime PCMU_BUF_DURATION = PCMU_BUF_MS * GST_MSECOND;
 ^
elements/rtpjitterbuffer.c:388:1: error: initializer element is not constant
 static const guint PCMU_BUF_SIZE = 64000 * PCMU_BUF_MS / 1000;
 ^
elements/rtpjitterbuffer.c:390:5: error: initializer element is not constant
     PCMU_BUF_CLOCK_RATE * PCMU_BUF_MS / 1000;
2015-07-08 23:49:12 -03:00
Thiago Santos
3edf9e4f58 rtpjitterbuffer: run indent and fix some comments
Fix indent on this file and break some comment lines into two to make
it fit 80 chars per line
2015-07-08 23:49:09 -03:00
Havard Graff
ddd032f56b rtpjitterbuffer: fix gap-time calculation and remove "late"
The amount of time that is completely expired and not worth waiting for,
is the duration of the packets in the gap (gap * duration) - the
latency (size) of the jitterbuffer (priv->latency_ns). This is the duration
that we make a "multi-lost" packet for.

The "late" concept made some sense in 0.10 as it reflected that a buffer
coming in had not been waited for at all, but had a timestamp that was
outside the jitterbuffer to wait for. With the rewrite of the waiting
(timeout) mechanism in 1.0, this no longer makes any sense, and the
variable no longer reflects anything meaningful (num > 0 is useless,
the duration is what matters)

Fixed up the tests that had been slightly modified in 1.0 to allow faulty
behavior to sneak in, and port some of them to use GstHarness.

https://bugzilla.gnome.org/show_bug.cgi?id=738363
2015-07-08 23:18:48 +03:00
Stian Selnes
8a0dbff3f4 rtph263depay: Make sure payload is large enough
Plus new unit test.

https://bugzilla.gnome.org/show_bug.cgi?id=752112
2015-07-08 11:36:55 +01:00
Tim-Philipp Müller
4ed4d0b84c tests: rtp-payloading: add basic unit test for KLV payloading
Also make it so that the mtu is always set if specified, not
only in case of the rather weird bufferlist test code path.
This allows us to easily make the payloader fragment a payload
across multiple output packets by setting a small MTU on it.
2015-07-07 20:11:28 +01:00
Stian Selnes
ef8d630a59 rtp: add H.261 RTP payloader and depayloader
Implementation according to RFC 4587.

Payloader create fragments on MB boundaries in order to match MTU size
the best it can. Some decoders/depayloaders in the wild are very strict
about receiving a continuous bit-stream (e.g. no no-op bits between
frames), so the payloader will shift the compressed bit-stream of a
frame to align with the last significant bit of the previous frame.

Depayloader does not try to be fancy in case of packet loss. It simply
drops all packets for a frame if there is a loss, keeping it simple.

https://bugzilla.gnome.org/show_bug.cgi?id=751886
2015-07-03 11:48:41 +01:00
Sebastian Dröge
3df0cce65d rtpjitterbuffer: If possible, always update the current time before looping over all timers
If we have a clock, update "now" now with the very latest running time we have.
If timers are unscheduled below we otherwise wouldn't update now (it's only updated
when timers expire), and also for the very first loop iteration now would otherwise
always be 0.

Also the time is used for the timeout functions, e.g. to calculate any times
for the next timeouts and we would otherwise pass too old times there.

https://bugzilla.gnome.org/show_bug.cgi?id=751636
2015-07-02 16:45:59 +02:00
Nicolas Dufresne
db63796fd3 qtmux: Correctly test each segments
In presence of gaps, qtdemux will emit multiple segments. The
second segment start should match the CTTS.

https://bugzilla.gnome.org/show_bug.cgi?id=751361
2015-06-23 22:34:36 -04:00
Nicolas Dufresne
89104e35bf qtmux: Test gaps at start of stream
https://bugzilla.gnome.org/show_bug.cgi?id=751242
2015-06-22 17:45:30 -04:00
Thiago Santos
74dcd85de4 tests: qtmux: test for muxing with DTS outside the segment
https://bugzilla.gnome.org/show_bug.cgi?id=740575
2015-06-12 17:18:24 -04:00
Jan Schmidt
c16c381a89 tests: Update mp4 mux test for mdat placeholder change
The mp4 muxer now writes a place-holder mdat as a free
atom followed by a 0-byte mdat that covers the rest of the
file, making it possible to rewrite it as 64-bit, or leave
it as-is if nothing else is written afterward
2015-06-08 14:49:11 +10:00
Sebastian Dröge
b549ebd066 rtpsession: Override the SSRC from the packets' SSRC if none was given via caps or property 2015-06-07 10:33:27 +02:00
Edward Hervey
d524439b35 check: Use GST_CHECK_MAIN () macro everywhere
Makes source code smaller, and ensures we go through common initialization
path (like the one that sets up XML unit test output ...)
2015-06-02 16:27:24 +02:00
Mark Nauwelaerts
692df969ea tests: wavpackparse: fix unit test
See also https://bugzilla.gnome.org/show_bug.cgi?id=738237
2015-05-10 14:22:43 +02:00
Sebastian Dröge
91c8688ed7 rtpjitterbuffer: Fix RTX unit test
The calculations were a bit off everywhere, even before the changes done
recently to the delay for RTX of expected future packets. It only worked by
accident, but now the calculations are all correct again. Hopefully.
2015-04-27 16:37:23 +02:00
Thiago Santos
9f7c659ff0 tests: qtmux: add tests to verify it handles non-0 segments
Both input streams in this test have a segment.start = 10s, so
output should start from 0 anyway.

Another test has both starting at non-0 segments, but the running
time of both streams should still start from 0
2015-04-10 10:05:24 -03:00
Thiago Santos
48c5c0c5b3 tests: qtmux: simple muxing test
Adds a new simple test that verifies that data is properly muxed
and preserved.  PTS, DTS, duration and caps are verified.
2015-04-10 10:05:24 -03:00
Ravi Kiran K N
a833084320 tests: add test suite for alpha
Added test suite for alpha element with test cases
1. alpha
2. chroma keying

https://bugzilla.gnome.org/show_bug.cgi?id=747595
2015-04-10 10:20:03 +01:00
Jan Schmidt
a51073c7de tests: Fix rtprtx test by handling buffer lists
Commit #1018aa made rtprtxsend handle buffer lists, breaking
the test which probes for buffers, but not buffer lists.

Use a utility function to run the probe callback on each buffer
in the list in turn and remove any buffers that are dropped.
2015-04-09 12:58:04 +10:00
Thiago Santos
00e5d90ffc tests: multifile: increment tests to check for multifile messages
Also verify that the multifilesink file messages are being correctly
posted to the bus
2015-04-04 11:45:07 -03:00
Thiago Santos
4bba05339c tests: multifile: handle FIXME for proper checking when test finished
Use a GstBus and wait for EOS to finish the tests instead of
relying on sleeping
2015-04-04 07:58:44 -03:00
Jan Schmidt
3d59b5f814 isomp4: Make non-seekable downstream an error in normal mode
When not in fast-start or fragmented mode, we need to be able
to rewrite the size of the mdat atom, or else the output just
won't be playable - the mdat placeholder with size == 0 will
cover the rest of the file, including any moov atom we write out.

https://bugzilla.gnome.org/show_bug.cgi?id=708808
2015-04-03 23:07:04 +11:00
Sebastian Rasmussen
cf54d4cc67 rtph263pay/-depay: add framesize SDP attribute
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726416
2015-04-02 19:38:21 -04:00
Wim Taymans
13804eab7d check: add jitterbuffer unit test
See https://bugzilla.gnome.org/show_bug.cgi?id=745539
2015-03-06 11:40:53 +01:00
Nicolas Dufresne
96b218d939 splitmux-test: Parse error message
The test had a function to print the error, but was not parsing it.
This was causing warning about dbg_info being used uninitialized. If
the test was testing any errors, this would have crashed.
2015-02-15 21:45:24 -05:00
Jan Schmidt
8ceb58122e splitmux: Add unit test for file splitting
Add a unit test for file splitting, and fix the leaks in the
splitmuxsink it found
2015-02-07 03:58:30 +11:00
Jan Schmidt
aa4c29c5d6 splitmux: Fix memory leaks until the test valgrinds clean 2015-02-07 00:19:36 +11:00
Jan Schmidt
5e2214d309 splitmux: Implement new elements for splitting files at mux level.
Implement 2 new elements - splitmuxsink and splitmuxsrc.

splitmuxsink is a bin which wraps a muxer and takes 1 video stream,
plus audio/subtitle streams, and starts a new file
whenever necessary to avoid overrunning a threshold of either bytes
or time. New files are started at a keyframe, and corresponding audio
and subtitle streams are split at packet boundaries to match
video GOP timestamps.

splitmuxsrc is a corresponding source element which handles
the splitmux:// URL and plays back all component files,
reconstructing the original elementary streams as it goes.
2015-02-06 04:26:59 +11:00
Thiago Santos
59431f663a tests: souphttpsrc: update ssl key/cert pair
Our ones were expired. The new ones were copied from libsoup's
tests files.

Also sets the property to use our own cert to validate the
server, otherwise the default system certs would be used
and it would fail.
2015-02-04 21:37:50 -03:00
Tim-Philipp Müller
d67da4c8ae tests: rtpcollision: use alawenc/dec in these tests instead of Speex
They should always be built, while the speex elements are not.

Need to check for a smaller number of buffers then (7->4) because
speexenc will add 3 header buffers while alawenc will just output
as many buffers as it receives as input.

https://bugzilla.gnome.org/show_bug.cgi?id=742098
2014-12-30 17:19:59 +00:00
Tim-Philipp Müller
d416336a6e tests: rtpaux: use alawenc/dec in these tests instead of Speex
They should always be built, while the speex elements are not.

https://bugzilla.gnome.org/show_bug.cgi?id=742098
2014-12-30 14:58:02 +00:00
Tim-Philipp Müller
da51a99403 tests: don't use deprecated property in level unit test 2014-11-02 16:58:30 +00:00
Sebastian Dröge
543c8772cb aacparse: Fix unit test now that we always have profile/level in the caps 2014-10-27 11:08:40 +01:00
Tim-Philipp Müller
146702e226 tests: fix rgvolume test on big-endian systems 2014-10-25 11:09:57 +01:00
Tim-Philipp Müller
e6f6d9045c tests: fix mulawdec/mulawenc test for big endian systems 2014-10-25 11:09:57 +01:00
Sjoerd Simons
0ee384b251 rtpmux: Don't set PROXY_CAPS flag on the src pad
rtpmux behaves like a funnel in that it forwards whatever upstream is
sending buffers. So setting proxy caps doesn't make sense as the
upstream don't have to have compatible caps, thus resulting in an empty
caps set as a result of a caps query. Instead set fixed caps just
as funnel does.

https://bugzilla.gnome.org/show_bug.cgi?id=738722
2014-10-21 10:52:00 +02:00