Commit graph

1505 commits

Author SHA1 Message Date
Philippe Normand
c51503fc41 rpicamsrc: add test-color-balance example
This small test will display a live video preview of the rpicam with
the balance controls being updated once a second. The controls to
update can be disabled in the source by setting the CONTROL_* macros
values to 0.
2020-07-10 16:45:02 +01:00
Jan Schmidt
acc7449d28 rpicamsrc: Add dynamic properties example
Python example of adjusting saturation on the fly
2020-07-10 16:44:41 +01:00
Sebastian Dröge
3ad86bdf30 imagefreeze: Add test for new live mode
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/653>
2020-06-29 12:07:14 +03:00
Nirbheek Chauhan
0fcd87e42a meson: Build Qt5 tests with -std=c++11
We already do this for the plugin.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/780#note_548179

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/642>
2020-06-25 15:20:55 +00:00
Havard Graff
cdba5952ed rtpsession: make tests more stable
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/410>
2020-06-20 19:45:33 +00:00
Tim-Philipp Müller
87d4374655 examples: qmlsink: rename qrc file to avoid naming conflicts with older meson versions
Would get "Tried to create target "qt5-qmlsink_qrc", but a
target of that name already exists." with older meson versions.

Work around that by renaming the qrc file.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/633>
2020-06-18 10:58:32 +01:00
Tim-Philipp Müller
d654c6feae tests: don't pull in all -bad plugin, only allow the one we need
Set up our plugin include list for tests in such a way that
we don't pull in *all* plugins from -bad but only the one
used in the splitmuxsink unit test, i.e. the timecode plugin,
so we don't accidentally use other encoders/decoders such as
nvenc/dec for example.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/617>
2020-06-09 15:23:40 +01:00
Guillaume Desmottes
0594d2f981 tests: vp9enc: enforce I420 format
Test was not enforcing a video format on videotestsrc. I420 was picked
as it was the first format in GST_VIDEO_FORMATS_ALL which will no longer
be true (gst-plugins-base!689).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/615>
2020-06-08 17:58:29 +02:00
Mikhail Fludkov
7b390a8bbd vpxenc: Add new bit-per-pixel property to select a better "default" bitrate
As part of this also change the default bitrate value to 0. The default
value was 256000 previously. In reality, if the property was not set the
bitrate value would be scaled according to the resolution which is not
very intuitive behavior. It is better to use 0 for this purpose. Now
together with newly introduced property "bits-per-pixel" 0 means to
assign the bitrate according to resolution/framerate.

The default bitrates are now
 - 1.2Mbps for VP8 720p@30fps
 - 0.8Mbps for VP9 720p@30fps
and scaled accordingly for different resolutions/framerates.

Previously the default bitrate was also not scaled according to the
framerate but only took the resolution into account.

This also fixes the side effect of setting bitrate to 0. Previously
encoder would not produce any data at all.

Addition from Sebastian Dröge <sebastian@centricular.com> to assume
30fps if no framerate is given in the caps instead of not calculating
any bitrate at all.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/611>
2020-06-04 20:14:03 +00:00
Stian Selnes
44e4de43da vpxdec: Check that output width and height != 0
For VP8 it's possible to signal width or height to be 0, but it does
not make sense to do so. For VP9 it's impossible. Hence, we most
likely have a corrupt stream. Trying to negotiate caps downstream with
either width or height as 0 will fail with something like

gst_video_decoder_negotiate_default: assertion 'GST_VIDEO_INFO_WIDTH (&state->info) != 0' failed

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/610>
2020-06-02 23:59:20 +03:00
Tim-Philipp Müller
5f91be7ea0 tests: fix meson test env setup to make sure we use the right gst-plugin-scanner
If core is built as a subproject (e.g. as in gst-build), make sure to use
the gst-plugin-scanner from the built subproject. Without this, gstreamer
might accidentally use the gst-plugin-scanner from the install prefix if
that exists, which in turn might drag in gst library versions we didn't
mean to drag in. Those gst library versions might then be older than
what our current build needs, and might cause our newly-built plugins
to get blacklisted in the test registry because they rely on a symbol
that the wrongly-pulled in gst lib doesn't have.

This should fix running of unit tests in gst-build when invoking
meson test or ninja test from outside the devenv for the case where
there is an older or different-version gst-plugin-scanner installed
in the install prefix.

In case no gst-plugin-scanner is installed in the install prefix, this
will fix "GStreamer-WARNING: External plugin loader failed. This most
likely means that the plugin loader helper binary was not found or
could not be run. You might need to set the GST_PLUGIN_SCANNER
environment variable if your setup is unusual." warnings when running
the unit tests.

In the case where we find GStreamer core via pkg-config we use
a newly-added pkg-config var "pluginscannerdir" to get the right
directory. This has the benefit of working transparently for both
installed and uninstalled pkg-config files/setups.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/603>
2020-05-27 12:42:38 +01:00
Matthew Waters
8a8c8afc86 qtoverlay: add the root item as a property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/595>
2020-05-20 19:37:32 +00:00
Nirbheek Chauhan
d67a658daf meson: Fix gstgl checks for qt and gtk
Also rename from build_ to have_, which is more accurate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/587>
2020-05-12 04:32:01 +05:30
Nirbheek Chauhan
2ecba800bf meson: Revamp qt5qml plugin and example build code
Stricter and simpler. For example, now we properly error out when
gstreamer-gl-1.0 was not found when the qt5 plugin is enabled or when
a C++ compiler is not enabled.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/587>
2020-05-12 04:30:13 +05:30
Seungha Yang
ea1797ccb5 tests: splitmuxsink: Add more timecode based split test
... and split test cases to run tests in parallel
2020-04-20 21:39:54 +09:00
Havard Graff
981d0c02de rtpjitterbuffer: don't use RTX packets in rate-calc and reset-logic
The problem was this:

Due to the highly irregular arrival of RTX-packet the max-misorder variable
could be pushed very low. (-10).

If you then at some point get a big in the sequence-numbers (62 in the
test) you end up sending RTX-requests for some of those packets, and then
if the sender answers those requests, you are going to get a bunch of
RTX-packets arriving. (-13 and then 5 more packets in the test)

Now, if max-misorder is pushed very low at this point, these RTX-packets
will trigger the handle_big_gap_buffer() logic, and because they arriving
so neatly in order, (as they would, since they have been requested like
that), the gst_rtp_jitter_buffer_reset() will be called, and two things
will happen:
1. priv->next_seqnum will be set to the first RTX packet
2. the 5 RTX-packet will be pushed into the chain() function

However, at this point, these RTX-packets are no longer valid, the
jitterbuffer has already pushed lost-events for these, so they will now
be dropped on the floor, and never make it to the waiting loop-function.

And, since we now have a priv->next_seqnum that will never arrive
in the loop-function, the jitterbuffer is now stalled forever, and will
not push out another buffer.

The proposed fixes:
1. Don't use RTX in calculation of the packet-rate.
2. Don't use RTX in large-gap logic, as they are likely to be dropped.
2020-04-16 17:06:31 +02:00
Kristofer Björkström
586fc57e55 rtpjpeg: Use gst_memory_map() instead of gst_buffer_map()
gst_buffer_map () results in memcopying when a GstBuffer contains
more than one GstMemory.
This has quite an impact on performance on systems with limited amount
of resources. With this patch the whole GstBuffer will not be mapped at
once, instead each individual GstMemory will be iterated and mapped
separately.
2020-04-03 17:01:24 +02:00
Havard Graff
d9aaa15a30 rtpopuspay: make depay ! pay work
There is a use-case for a server to re-payload opus going through it.

Problem was that the payloader requires channels in the caps, but
this is not something the depayloader can parse out of the stream, meaning
caps-negotiation would fail.

Removing the requirement of channels in the template-caps fixes this.
2020-04-03 09:04:32 +00:00
Seungha Yang
018218dd73 tests: Split splitmux test case
Since we are adding more and more tests into splitmux,
we need to split it to avoid CI timeout.
2020-04-03 17:08:51 +09:00
Seungha Yang
599066726f splitmuxsink: Don't send too many force key unit event
splitmuxsink should requst keyframe depending on configured
threshold and previously requested time in order to avoid too many
keyframe request.
2020-04-03 15:00:37 +09:00
Havard Graff
9f1062dc05 rtpjitterbuffer: various test-improvements
Mainly generalize all the latest tests that have found various stalls
in the jitterbuffer, so that they only consist of a series of packets
with various seqnum/rtptime/rtx combinations, arriving at a specific time.

This means future tests can be more easily written to prove certain
behavior does not cause stalls.

Also fix the warning on windows:
warning C4244: 'initializing': conversion from 'double' to 'gint', possible loss of data
2020-03-31 04:01:38 +02:00
Jan Schmidt
8ef172d8b4 splitmux: Make the unit test faster
The playback test is considerably faster if it runs with the
appsink set to sync=false
2020-03-26 11:23:24 +00:00
Seungha Yang
d06970c561 tests: splitmux: Add test for timecode based split 2020-03-25 13:22:31 +00:00
Xavier Claessens
6e1758d509 Fix usage of C99
It's 2020, way too early for that, let's stick to C89 for now.
2020-03-23 21:32:04 -04:00
Havard Graff
a710bda1ab rtptimerqueue: remove ->num from the timer
This concept was only used by the "multi"-lost timer, and since that
one is not around any longer, the "num" concept is superfluous.
2020-03-20 13:17:20 +00:00
Havard Graff
f1ff80ced0 rtpjitterbuffer: remove the concept of "already-lost"
This is a concept that only applies when a buffer arrives in the chain
function, and it has already been scheduled as part of a "multi"-lost
timer.

However, "multi"-lost timers are now a thing of the past, making this
whole concept superflous, and this buffer is now simply counted as "late",
having already been pushed out (albeit as a lost-event).
2020-03-20 13:17:20 +00:00
Havard Graff
5dacf366c0 rtpjitterbuffer: immediately insert a lost-event on multiple lost packets
There is a problem with the code today, where a single timer will
be scheduled for a series of lost packets, and then if the first packet
in that series arrives, it will cause a rescheduling of that timer, going
from a "multi"-timer to a single-timer, causing a lot of the packets
in that timer to be unaccounted for, and creating a situation in where
the jitterbuffer will never again push out another packet.

This patch solves the problem by instead of scheduling those lost packets
as another timer, it instead asks to have that lost-event pushed straight
out.

This very much goes with the intent of the code here: These packets are
so desperately late that no cure exists, and we might as well get the
lost-event out of the way and get on with it.

This change has some interesting knock-on effect being presented in
later commits. It completely removes the concept of "already-lost", so
that is why that test has been disabled in this commit, to be
removed later.
2020-03-20 13:17:20 +00:00
Havard Graff
d045b40db9 rtpjitterbuffer: rework large-gap tests
Make sure to set the time the buffer is supposed to arrive at, so
as not to trigger an artificial situation.
2020-03-20 13:17:20 +00:00
Havard Graff
9eaf084d7a test/check: split out rtptimerqueue-tests in a separate file 2020-03-20 13:17:20 +00:00
Seungha Yang
18e09de0a2 splitmuxsink: Decouple keyframe request and the decision for fragmentation
Split the decision for keyframe request and fragmentation in order to
ensure periodic keyframe request.
2020-03-19 10:17:21 +00:00
Matthew Waters
7a25fb5b08 qt: add a qml overlay filter element [part 2]
It takes a qml scene description and renders it using a possible input
stream.

Currently supported on GLX and WGL.

Follow up to (as that MR had an old version of the commit):
- https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/475
- 4778d7166a: qt: add a qml overlay filter element
2020-03-19 17:26:54 +11:00
Matthew Waters
4778d7166a qt: add a qml overlay filter element
It takes a qml scene description and renders it using a possible input
stream.

Currently supported on GLX and WGL.
2020-03-18 11:22:39 +00:00
Matthew Waters
73cd4477af test/qml: add an dynamically adding qmlglsink element
The example shows how to add qmlglsink to an already running pipeline
with pre-existing OpenGL elements.
2020-03-18 11:22:39 +00:00
Tim-Philipp Müller
66296fcae3 tests: rtp-payloading: add minimal vp8/vp9 rtp payloading/depayloading test 2020-03-12 16:55:44 +00:00
Ognyan Tonchev
a78a74bff0 rtph26x: Use gst_memory_map() instead of gst_buffer_map() in avc mode
gst_buffer_map () results in memcopying when a GstBuffer contains
more than one GstMemory and when AVC (length-prefixed) alignment is used.
This has quite an impact on performance on systems with limited amount of
resources. With this patch the whole GstBuffer will not be mapped at once,
instead each individual GstMemory will be iterated and mapped separately.
2020-03-06 10:44:16 +00:00
Havard Graff
026223cde2 rtpjitterbuffer: fix stalling when resetting timers
When calling gst_rtp_jitter_buffer_reset you pass in a seqnum.

This is considered the starting-point for a new stream.

However, the old behavior would unref this buffer, basically lying to
the thread that is pushing out buffers saying that it can expect
this buffer, when it would never arrive. The resulting effect being no
more buffer pushed out of the jitterbuffer, and it would buffer
incoming data indefinitely.

By instead inserting the buffer in the gap_packets queue, the _reset()
function will take responsibility for using that as the first buffer
of the new stream.

Fixes #703
2020-03-04 12:55:52 +01:00
Jan Schmidt
f490c38416 splitmux: Avoid negative DTS
In order to concatenate fragments, splitmuxsrc offsets
the start of each fragment PTS to 0 to align it with the
previous file. This means that DTS can go negative for
the first fragment, with really bad results.

Add a fixed offset to outgoing timestamp ranges to
avoid that.
2020-03-04 05:42:21 +00:00
Yeongjin Jeong
830db205f6 tests: flvmux: Instead of using the testclock, just send eos event for drain
When using the testclock for determining clock in test, it is sometimes observed
that the clock entry is not registered in time by the aggregator. So deadlock occurs
between the aggregator and the test thread.
2020-03-02 01:37:27 +09:00
Håvard Graff
fdf002d069 rtpsession: fix crash when no extension-header present for twcc 2020-02-24 13:06:27 +00:00
Yeongjin Jeong
9feb35638a tests: flvmux: Add test for rollover timestamp
The timestamps that exceed uint32 maximum value should be handled to rollover.
2020-02-18 18:39:34 +09:00
Havard Graff
1df706448c rtpmanager: Google Transport-Wide Congestion Control RTP Extension
Generating and parsing the RTCP-messages described in:
https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01
2020-02-14 10:09:02 +00:00
Håvard Graff
9ba9837058 rtpfunnel: various cleanups
* Organize GstRtpFunnelPad and GstRtpFunnel separately
* Use G_GNUC_UNUSED instead of (void) casts
* Don't call an event "caps"
* Use semicolons after GST_END_TEST (helps gst-indent)
2020-02-14 10:08:05 +00:00
Mikhail Fludkov
57b0522cd7 rtpptdemux: set payload to caps inside gst_rtp_pt_demux_get_caps
Refactoring to remove duplicate code and add test
2020-02-11 18:39:22 +00:00
John Bassett
16d750bc01 rtpssrcdemux: Handle RTCP APP packets
Fix crash when processing RTCP APP packets.
2020-02-11 15:12:07 +01:00
John Bassett
5800950a2d rtpssrcdemux: Bad RTP/RTCP packet is not fatal
When used for processing bundled media streams within rtpbin the rtpssrcdemux element may
receive bad RTP and RTCP packets, these should not be treated as a fatal error.
2020-02-11 15:10:12 +01:00
Mikhail Fludkov
35596e7fac rtpssrcdemux: introduce max-streams property
The property is useful against atacks when the sender changes SSRC for
every RTP packet. The property with the same name introduced in rtpbin
was not enough, because we still can end up with thousands of pads
allocated in rtpssrcdemux.
2020-02-11 15:10:12 +01:00
Havard Graff
94e10d522e rtpssrcdemux: fix test warnings 2020-02-11 15:07:45 +01:00
Olivier Crête
c00796eaa5 rtpsession: Add test for packet rate maths 2020-02-06 14:01:38 -05:00
Mathieu Duponchelle
a245e85fb1 rtprtxsend: allow generic input caps
When connected to an upstream rtpfunnel element, payload-type,
ssrc and clock-rate will not be present in the received caps.

rtprtxsend can already deal with only the clock rate being
present there, a new property is exposed to allow users to
provide a payload-type -> clock-rate map, this enables the
use of the max-size-time property for bundled streams.
2020-01-28 15:44:13 +00:00
Kristofer Björkström
9c86414279 rtph265pay: TID for NALU type 48 was always set to 7
A typo bug: | instead of & resulted in TID alwasy being set to 7
for the aggregated NALU of type 48
2020-01-13 15:41:30 +01:00
Havard Graff
8b96d8ee8d rtpbin: fix shutdown crash in rtpbin
The key is to make sure the jitterbuffer is set to NULL *before* the
ptdemux.

The race that existed would basically happen when ptdemux had reached
READY, and the jitterbuffer would then push a buffer, triggering a new
pad with a new payloadtype being added and ghosted to the rtpbin itself.

However, the srcpad of the ptdemux would now be inactive, and all the
sticky-event pushed on it would be swallowed, not allowing any to reach
the ghost-pad. Then the buffer in-flight would come to the ghostpad,
and we would assert that a buffer arrived before the necessary
events.

By simply re-ordering the state-changes, we ensure that there will be
no buffer racing into the ptdemux while its state is being changed,
and the problem disappears completely.

Notice also that there is not point in disconnecting the signals on the
ptdemux before this point, since we need the push-thread to settle
down before we can do this in a non-racy way.
2019-12-20 08:27:07 +00:00
Mathieu Duponchelle
e2462005fb qtmux: port to GstAggregator 2019-12-16 14:17:38 +00:00
Havard Graff
eb6d8791e8 rtpsession: add test for requesting FIR after having requested PLI 2019-11-29 20:17:29 +00:00
Havard Graff
54524b08dc rtpjitterbuffer: make test more stable 2019-11-29 18:33:10 +00:00
Havard Graff
f997859913 rtpsession: add locking for clear-pt-map
...or it will segfault from time to time...
2019-11-29 14:23:49 +01:00
Havard Graff
87457a862d rtpjitterbuffer: make sure not to drop packets based on skew
One of the jitterbuffers functions is to try and make sense of weird
network behavior.

It is quite unhelpful for the jitterbuffer to start dropping packets
itself when what you are trying to achieve is better network resilience.

In the case of a skew, this could often mean the sender has restarted
in some fashion, and then dropping the very first buffer of this "new"
stream could often mean missing valuable information, like in the case
of video and I-frames.

This patch simply reverts back to the old behavior, prior to 8d955fc32b
and includes the simplest test I could write to demonstrate the behavior,
where a single packet arrives "perfectly", then a 50ms gap happens,
and then two more packets arrive in perfect order after that.

# Conflicts:
#	tests/check/elements/rtpjitterbuffer.c
2019-11-02 23:05:32 +00:00
Tim-Philipp Müller
c9a47c0c8d Remove autotools build system 2019-10-14 11:04:18 +01:00
Aaron Boxer
46989dca96 documentation: fix a number of typos 2019-10-05 22:38:11 +00:00
Simon Arnling Bååth
8173596ed2 gstrtpjitterbuffer: Custom messages when dropping packets
This commit adds custom element messages for when gstrtpjitterbuffer
drops an incoming rtp packets due to for example arriving too late.
Applications can listen to these messages on the bus which enables
actions to be taken when packets are dropped due to for example high
network jitter.

Two properties has been added, one to enable posting drop messages and
one to set a minimum time between each message to enable throttling the
posting of messages as high drop rates.
2019-10-04 20:31:56 +00:00
Olivier Crête
a24596423a rtpjitterbuffer: Cancel timers instead of just unlocking loop thread
When the queue is full (and adding more packets would risk a seqnum
roll-over), the best approach is to just start pushing out packets
from the other side.  Just pushing out the packets results in the
timers being left hanging with old seqnums, so it's safer to just
execute them immediately in this case. It does limit the timer space
to the time it takes to receiver about 32k packets, but without
extended sequence number, this is the best RTP can do.

This also results in the test no longer needed to have timeouts or
timers as pushing packets in drives everything.

Fixes #619
2019-09-28 07:47:54 -04:00
Nicolas Dufresne
d4c6c335c5 rtpjitterbuffer: No need to wake the timer thread on head changes
If the jitterbuffer head change, there is no need to systematically
wakeup the timer thread. The timer thread will be waken up on if
an earlier timeout has been pushed. This prevent some more spurious
wakeup when the system is loaded. As a side effect, cranking the clock
may set the clock at an earlier position.
2019-09-27 17:34:04 -04:00
Nicolas Dufresne
37742cd36d rtptimerqueue: Consolidate a data structure for timers
Implement a single timer queue for all timers. The goal is to always use
ordered queues for storing timers. This way, extracting timers for
execution becomes O(1). This also allow separating the clock wait
scheduling from the timer itself and ensure that we only wake up the
timer thread when strictly needed.

The knew data structure is still O(n) on insertions and reschedule,
but we now use proximity optimization so that normal cases should be
really fast. The GList structure is also embeded intot he RtpTimer
structure to reduce the number of allocations.
2019-09-27 17:34:04 -04:00
Nicolas Dufresne
a53ffb6e11 tests: jitterbuffer: Demacroify some helpers
There is no reason for these to be macros anymore. This makes the
test helper much more readable.
2019-09-27 13:02:16 -04:00
Jan Schmidt
31be44c47f splitmux: Add muxer-pad-map property
Add a property which explicitly maps splitmuxsink pads to the
muxer pads they should connect to, overriding the implicit logic
that tries to match pads but yields arbitrary names.
2019-09-06 12:38:56 +00:00
Mathieu Duponchelle
e58ca79741 valgrind: suppress Cond error coming from gnutls
taken from fb4a8dda21
2019-08-08 14:39:17 +00:00
Antonio Ospite
1c5c90ea23 test: rtpbin_buffer_list: add a test for invalid packets in buffer list
Upstream elements can send all kinds of data in a buffer list, so cover
the case of an invalid RTP packet mixed with valid RTP packets.
2019-08-07 15:32:30 -04:00
Antonio Ospite
12b420168c test: rtpbin_buffer_list: add a test for multiplexed RTP and RTCP
RTP and RTCP packets can be muxed together on the same channel (see
RFC5761) and can arrive in the same buffer list.

The GStreamer rtpsession element support RFC5761, so add a test to cover
this case for buffer lists too.
2019-08-07 15:32:30 -04:00
Antonio Ospite
6a5f38a325 test: rtpbin_buffer_list: add a test for different timestamps in buffer list
Buffers with different timestamps (e.g. packets belonging to different
frames) can arrive together in the same buffer list,

Add a test to cover this case.
2019-08-07 15:32:30 -04:00
Antonio Ospite
b158be0d98 test: rtpbin_buffer_list: add function to check timestamp 2019-08-07 15:32:30 -04:00
Antonio Ospite
31f221f89d test: rtpbin_buffer_list: add a test about reordered or duplicated seqnums 2019-08-07 15:32:30 -04:00
Antonio Ospite
1d337b704e test: rtpbin_buffer_list: add a test for lange jump in seqnums with recovery 2019-08-07 15:32:30 -04:00
Antonio Ospite
ae5d4b8cf0 test: rtpbin_buffer_list: add a test for large jump in sequence numbers 2019-08-07 15:32:30 -04:00
Antonio Ospite
fb46c6bf08 test: rtpbin_buffer_list: add a test for wrapping sequence numbers 2019-08-07 15:32:30 -04:00
Antonio Ospite
d9d0461aeb test: rtpbin_buffer_list: add a test for permissible gap in sequence numbers 2019-08-07 15:32:30 -04:00
Antonio Ospite
92be4121e8 test: rtpbin_buffer_list: add a test for the case of failed probation
When a new source fails to pass the probation period (i.e. new packets
have non-consecutive sequence numbers), then no buffer shall be pushed
downstream. Add a test to validate this case.
2019-08-07 15:32:30 -04:00
Antonio Ospite
0537925c3a test: rtpbin_buffer_list: add function to check sequence number 2019-08-07 15:32:30 -04:00
Antonio Ospite
ec56e2fb78 test: rtpbin_buffer_list: add test to verify that receiving stats are correct
Add a test to verify that stats about received packets are correct when
using buffer lists in the rtpsession receive path.

Split get_session_source_stats() in two to be able to get stats from
a GstRtpSession object directly.
2019-08-07 15:32:30 -04:00
Antonio Ospite
80daea90f0 test: rtpbin_buffer_list: add a test for buffer lists on the recv path 2019-08-07 15:32:30 -04:00
Jan Schmidt
436d33b288 splitmuxsink: add the ability to mux auxilliary video streams
The primary video stream is used to select fragment cut points
at keyframe boundaries. Auxilliary video streams may be
broken up at any packet - so fragments may not start with a keyframe
for those streams.
2019-07-15 11:46:36 +00:00
Olivier Crête
efed059b9a rtp session: Add test for collision loopback detection
Ignore further collisions if the remote SSRC change with ours, it's
probably because someone is sending us back the packets we send out.
2019-07-06 14:23:20 +00:00
Olivier Crête
b5faed910e rtpsession tests: Add test for third-party collision detection
Add tests to validate the code that ignores the same packets coming
from 2 different sources (an third-party collision).
2019-07-06 14:23:20 +00:00
Olivier Crête
3574e6c176 rtpsession: Add test for collision on incoming packets
Make sure that the collision is properly detected on incoming packets.
2019-07-06 14:23:20 +00:00
Olivier Crête
4e18567863 rtpsession test: Verify that on-ssrc-collision message is emitted 2019-07-06 14:23:20 +00:00
Olivier Crête
9d9d543d5c rtpsession: Also send conflict event when sending packet
If the conflict is detected when sending a packet, then also send an
upstream event to tell the source to reconfigure itself.

Also ignore the collision if we see more than one collision from the same
remote source to avoid problems on loops.
2019-07-06 14:23:20 +00:00
Olivier Crête
061afa33ee rtph265pay: Also immediately send packet if it is a suffix NAL
Immediately send packet if it contains any suffix NAL, this is required
in case they come after the VCL nal to not have to wait until the next frame.
2019-07-03 19:05:29 +00:00
Olivier Crête
97f2fb4cc8 rtph26xpay: Wait until there is a VCL or suffix NAL to send
With unit tests.
2019-07-03 19:05:29 +00:00
Olivier Crête
4810c93222 rtph265pay test: Add unit tests for aggregation 2019-07-03 19:05:29 +00:00
Olivier Crête
1b32cb1eae rtph265pay: Implement Aggregation packets
Align with rtph264pay
2019-07-03 19:05:29 +00:00
Olivier Crête
42d775dfbe rtph264pay test: Add unit tests for aggregation 2019-07-03 19:05:29 +00:00
Olivier Crête
cede4f993d rtph264pay: Default to not adding latency when aggregating
Send the bundle as soon as there is one VCL unit in the packet at
the end of an incoming buffer.

The DELTA_UNIT flag is not reliable, so ignore it.
2019-07-03 19:05:29 +00:00
Olivier Crête
0c094612be rtp-payloading test: Fix working to 1.0 buffers instead of groups 2019-07-03 19:05:29 +00:00
Jan Alexander Steffens (heftig)
b46dab13d2 rtph264pay: Support STAP-A bundling
Add a new property "do-aggregate"* to the H.264 RTP payloader which
enables STAP-A aggregation as per [RFC-6184][1]. With aggregation enabled,
packets are bundled instead of sent immediately, up until the MTU size.
Bundles also end at access unit boundaries or when packets have to be
fragmented.

*: The property-name is kept generic since it might apply more widely,
   e.g. STAP-B or MTAP.
[1]: https://tools.ietf.org/html/rfc6184#section-5.7

Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/434
2019-07-03 19:05:29 +00:00
Havard Graff
4d4e8f99b9 rtpjitterbuffer: Add unit test for unsolicited rtx affecting skew 2019-07-03 06:23:07 -06:00
Thomas Bluemel
8d955fc32b rtpjitterbuffer: Only calculate skew or reset if no gap.
In the case of reordered packets, calculating skew would cause
pts values to be off. Only calculate skew when packets come
in as expected. Also, late RTX packets should not trigger
clock skew adjustments.

Fixes #612
2019-07-03 06:23:07 -06:00
Jan Alexander Steffens (heftig)
91e858dcbe
test: flvmux: Test changing caps with one sinkpad
These tests segfault without the preceding crash fix.
2019-06-19 14:36:21 +02:00
Jan Alexander Steffens (heftig)
daafce54ac
test: flvmux: Use gst_harness_sink_push_many
And check its return value.
2019-06-19 14:36:21 +02:00
Mathieu Duponchelle
ebe2756434 jitterbuffer: unset DTS on output buffers 2019-06-14 16:02:59 +02:00
Mikhail Fludkov
ec5fa49631 rtpjitterbuffer: late packets shouldn't affect PTS of the following packet
If, say, a rtx-packet arrives really late, this can have a dramatic
effect on the jitterbuffer clock-skew logic, having it being reset
and losing track of the current dts-to-pts calculations, directly affecting
the packets that arrive later.

This is demonstrated in the test, where a RTX packet is pushed in really
late, and without this patch the last packet will have its PTS affected
by this, where as a late RTX packet should be redundant information, and
not affect anything.
2019-06-13 11:55:10 +02:00
Mikhail Fludkov
b9c3e354ee rtpjitterbuffer: fix rtx delay calulation when large packet spacing 2019-06-12 11:39:32 +02:00
Stian Selnes
6269ed49ab rtpjitterbuffer: Fix delay for EXPECTED timers added by gaps
This patch corrects the delay set on EXPECTED timers that are added when
processing gaps. Previously the delay could be too small so that
'timout + delay' was much less than 'now', causing the following retries
to be scheduled too early. (They were sent earlier than
rtx-retry-timeout after the previous timeout.)
2019-06-12 11:39:32 +02:00