Commit graph

1102 commits

Author SHA1 Message Date
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
Nirbheek Chauhan
5c152314de Revert "meson: move gstreamer-check-1.0 dependency to tests/check"
This reverts commit 4663269466.

Does not actually work. See:
https://bugzilla.gnome.org/show_bug.cgi?id=773114#c31
2016-10-25 11:47:22 +05:30
Tim-Philipp Müller
6beadb9062 meson: fix build outside of gst-all 2016-10-24 00:34:15 +01:00
Scott D Phillips
4663269466 meson: move gstreamer-check-1.0 dependency to tests/check
https://bugzilla.gnome.org/show_bug.cgi?id=773114
2016-10-21 06:01:10 -03:00
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
Thibault Saunier
887a5911f5 meson: Make use of new environment object and set plugin path to builddir
Workaround source_root being the root directory of all projects in the subproject
case and remove now unneeded getpluginsdir

Bump meson requirement to 0.35
2016-10-11 02:09:04 +02:00
Gaurav Gupta
6542edd909 tests: Fix memory leak in test rtpaux test
https://bugzilla.gnome.org/show_bug.cgi?id=772496
2016-10-06 13:23:28 +03:00
Thibault Saunier
b910ecca68 meson: Setup pre commit hook and fix getpluginsdir for standalone case 2016-09-30 12:57:51 -03:00
Arun Raghavan
153b716490 tests: Fix tagschecking failure due to missing PTS
qtmux now needs the PTS (commit a993883b7), so let's make sure we
produce one with our buffers.

https://bugzilla.gnome.org/show_bug.cgi?id=772228
2016-09-30 13:25:10 +05:30
Tim-Philipp Müller
7f294ad125 meson: tests: fix vp8 availability checks
Those variables are not defined if vp8 was not found.
2016-09-27 12:41:21 +01:00
Thibault Saunier
375f3aab89 meson: Add gst-plugins-base plugins directories to be used by tests 2016-09-26 13:22:29 -03:00
Tim-Philipp Müller
787f47604d meson: add unit tests
Only works properly in an installed setup currently, most
likely won't work with a subprojects setup yet.
2016-09-26 14:31:09 +01:00
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
Sebastian Dröge
63b0e519fa tests/examples: #define GDK_DISABLE_DEPRECATION_WARNINGS
We use gdk_cairo_create() which is deprecated since 3.22.
2016-09-01 10:59:51 +03:00
Josep Torra
d40f007d61 gitignore: ignore qtdemux, rtph261 and rtpvp9 tests 2016-08-26 21:32:07 +02:00
Josep Torra
8f89d2a439 tests: use GST_NET_LIBS instead of hardcoded -lgstnet
Fixes build in OSX when running 'make check' in gst-uninstalled.
2016-08-26 21:22:16 +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
Josep Torra
77585fdade build: silence error about pthread for 'make check' in osx
Fixes "clang: error: argument unused during compilation: '-pthread'"
2016-08-26 21:11:59 +02:00
Sebastian Dröge
bc99a86472 vp9enc: Fix build of unit test by letting it link to libgstvideo 2016-08-26 20:31:10 +03:00
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
Nirbheek Chauhan
b09f478e80 Add support for Meson as alternative/parallel build system
https://github.com/mesonbuild/meson

With contributions from:

Tim-Philipp Müller <tim@centricular.com>
Jussi Pakkanen <jpakkane@gmail.com> (original port)

Highlights of the features provided are:
* Faster builds on Linux (~40-50% faster)
* The ability to build with MSVC on Windows
* Generate Visual Studio project files
* Generate XCode project files
* Much faster builds on Windows (on-par with Linux)
* Seriously fast configure and building on embedded

... and many more. For more details see:

http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.html
http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html

Building with Meson should work on both Linux and Windows, but may
need a few more tweaks on other operating systems.
2016-08-20 11:21:12 +01:00
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
Olivier Crête
a390a6791c rtp example: Fix leak
Also stop fetching the internal source as this
functionality has been broken.
2016-07-11 11:59:21 -04:00
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