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1394 commits

Author SHA1 Message Date
Tim-Philipp Müller
8e04651b8b Use gst_buffer_new_memdup() 
Update for function rename in core.

https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/827

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2281>
 2021-05-24 19:05:27 +01:00
Tim-Philipp Müller
0151276d7f Use new gst_buffer_new_copy() 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2279>
 2021-05-23 17:20:16 +01:00
Olivier Crête
8b595e7c8b webrtc test: Print content of error GstMessage 
Makes it easier to interpret the result of the CI!

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
 2021-05-13 16:37:31 -04:00
Olivier Crête
78d2d6cf6f webrtcbin tests: Add test for intersection src pad caps 
This checks that the codec preferences are intersected also with what
the src pad can handle.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
 2021-05-13 16:37:31 -04:00
Olivier Crête
cc556452ce webrtc test: Add explicit test clock 
This way the test clock is not linked to the multiple harnesses

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
 2021-05-13 15:05:00 -04:00
Olivier Crête
2aa7efedd3 webrtc test: Add test for codec preferences negotiation 
Validate that it does the intersection with the caps from
the sink pad and rejects the offer creation otherwise.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
 2021-05-13 15:05:00 -04:00
Olivier Crête
70befc0b21 webrtcbin: Implement caps queries on sinkpad based on codec preferences 
Also includes a unit test.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
 2021-05-13 15:05:00 -04:00
Olivier Crête
6a3a62abae webrtcbin tests: Use properties to access the inside of the transceiver object 
This will allow hiding the insides from unsafe application access.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
 2021-05-13 15:05:00 -04:00
Johan Sternerup
caefc3a831 webrtcbin: Add unit test for closing of data channels 
Add test for verifying that the data channel "close" action signal
triggers an SCTP_RESET_STREAMS request that is propagated to the other
side and eventually leads to both sides closing properly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2186>
 2021-05-12 03:02:27 +00:00
François Laignel
ad3d7d34cc Use gst_element_request_pad_simple... 
Instead of the deprecated gst_element_get_request_pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2180>
 2021-05-05 06:17:14 +00:00
Doug Nazar
be2996c48e tests/netsim: Set src caps before creating buffers 
GstHarness requires the source pad caps to be set before
buffer allocations.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2179>
 2021-04-21 09:05:44 +00:00
Seungha Yang
817544860d d3d11: Add support for BGRx and RGBx formats 
For such formats, we can re-use existing BGRA/RGBA implementations
but ignoring alpha channel

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2174>
 2021-04-21 05:45:59 +00:00
Olivier Crête
bc817f340c webrtcbin test: Don't fail if data channel is created 
In tests that voluntarily create a data channel.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2168>
 2021-04-21 03:21:55 +00:00
Doug Nazar
edbf0a6622 tests/avtp: increase timeout of test_depayloader_fragmented_big 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2160>
 2021-04-14 07:05:13 +00:00
Doug Nazar
6faff99596 check: fix dash_mpdparser_check_mpd_client_set_methods test. 
Setting guint64 valist properties without type specifier fails
on 32bit archs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2161>
 2021-04-14 06:35:25 +00:00
Doug Nazar
63b5ae0ffe line21enc: fix remove-caption-meta property test 
It's possible for the same address to be allocated to the decoded
metadata. Switch test to actual detect if it was removed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2159>
 2021-04-13 16:34:15 -04:00
Doug Nazar
a1535a4dc3 tests: fix shm test deadlock 
Stopping the consumer first would occasionally allow the producer
to fill the shm segment causing it to block in send() and unable
to be stopped.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2158>
 2021-04-13 11:59:35 +00:00
Doug Nazar
a930b62afc check: Fix test dash_mpdparser_xlink_period 
Test used http://404/ERROR/XML.period as an invalid url. Curl now
interprets that as an 32bit int and tries an actual connect which
timesout. Use .invalid as an IANA reserved domain for invalid DNS.

curl -v http://404/ERROR/XML.period
*   Trying 0.0.1.148:80...

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2157>
 2021-04-13 10:17:47 +00:00
Olivier Crête
474c4bf08f webrtcbin test: Wait for set-local-desc & set-remote-desc to continue 
To avoid racing betwen the SDPs being set and the next step of the
test, let's wait for setting the SDP both locally and remotely to succeed.
of the test

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
 2021-04-12 18:37:27 -04:00
Olivier Crête
4a48e291ff webrtcbin test: Add for the case where a second m-line is renegotiated 
This is for the case where there answerer forces a specific media type
for a m-line, but he origin offer only has the other media type. In this
case, we will create a second transceiver on receiving the offer and add
the desired media type using renegotiation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
 2021-04-12 18:37:27 -04:00
Olivier Crête
2bd647e999 webrtc test: Verify that forcing different kinds on peers fails 
If the offer contains an audio kind and a video kind, forcing them both
at m-line zero will fail.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
 2021-04-12 18:37:27 -04:00
Olivier Crête
8df5b9f974 webrtc tests: Verify that create-offer is rejected when needed 
Verify that it gets rejected if a m-line at index 1 is requested but
there is no m-line 0.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
 2021-04-12 18:37:27 -04:00
Olivier Crête
913d308e22 webrtcbin test: Add test for various cases where get_request_pad is meant to fail 
This should ensure that the recently added code works.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
 2021-04-12 17:55:07 -04:00
Olivier Crête
5971a96109 webrtcbin: Try to match an existing transceiver on pad request 
This should avoid creating extra transceivers that are duplicated.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
 2021-04-12 17:55:07 -04:00
Olivier Crête
d49e664c84 webrtcbin test: Test adding a stream to a stream+datachannel 
This use-case was previously broken by the expectation of having
a 1-1 match between the pad id and the m-line index

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
 2021-04-12 17:55:07 -04:00
Sebastian Dröge
ffa4d84e54 h2645parser: Catch overflows in AVC/HEVC NAL unit length calculations 
Offset and size are stored as 32 bit guint and might overflow when
adding the nal_length_size, so let's avoid that.

For the size this would happen if the AVC/HEVC NAL unit size happens to
be stored in 4 bytes and is 4294967292 or higher, which is likely
corrupted data anyway.

For the offset this is something for the caller of these functions to
take care of but is unlikely to happen as it would require parsing on a
>4GB buffer.

Allowing these overflows causes all kinds of follow-up bugs in the
h2645parse elements, ranging from infinite loops and memory leaks to
potential memory corruptions.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2103>
 2021-03-24 09:22:48 +00:00
Matthew Waters
640a65bf96 gst: don't use volatile to mean atomic 
volatile is not sufficient to provide atomic guarantees and real atomics
should be used instead.  GCC 11 has started warning about using volatile
with atomic operations.

https://gitlab.gnome.org/GNOME/glib/-/merge_requests/1719

Discovered in https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/868

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2098>
 2021-03-22 14:34:36 +11:00
Matthew Waters
e463bcfadf tests/webrtc: check for more sdp things across the board 
e.g.

- test for a=setup:$val and direction attributes in all tests
- test number of media sections
- test number of formats in each m= section (for audio/video)
- test no duplicate formats

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2093>
 2021-03-19 18:02:21 +11:00
Mathieu Duponchelle
08442cc792 cccombiner: implement scheduling 
Prior to that, cccombiner's behaviour was essentially that of
a funnel: it strictly looked at input timestamps to associate
together video and caption buffers.

This patch instead exposes a "schedule" property, with a default
of TRUE, to control whether caption buffers should be smoothly
scheduled, in order to have exactly one per output video buffer.

This can involve rewriting input captions, for example when the
input is CDP sequence counters are rewritten, time codes are dropped
and potentially re-injected if the input video frame had a time code
meta.

Caption buffers may also get split up in order to assign captions to
the correct field when the input is interlaced.

This can also imply that the input will drift from synchronization,
when there isn't enough padding in the input stream to catch up. In
that case the element will start dropping old caption buffers once
the number of buffers in its internal queue reaches a certain limit
(configurable).

The property is exposed so that existing users of cccombiner can
revert back to the original behaviour, but should eventually be
removed, as that behaviour was simply inadequate.

This commit also disallows changing the input caption type, as
this would needlessly complicate implementation, and removes
the corresponding test.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2076>
 2021-03-17 22:00:25 +00:00
Stéphane Cerveau
451c875d40 zxing: update to support version 1.1.1 
Support new API in 1.1.1
Update the supported input video format.
Update tests to use parse_launch

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2037>
 2021-03-12 01:03:49 +00:00
Philippe Normand
fae7c8dd7e play: tests: Switch user-agent test to a real HTTP server 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2061>
 2021-03-09 18:03:48 +00:00
Philippe Normand
3eec2f4be8 play: tests: Refactor to use new Message bus API 
Instead of relying on an extra GMainLoop, the messages are poped from the player
bus and handled synchronously. This should avoid flaky behaviors.

Fixes #608

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2061>
 2021-03-09 18:03:48 +00:00
Matthew Waters
2bed220771 webrtc: don't generate duplicate rtx payloads when bundle-policy is set 
It was possible to generate a SDP that had an RTX payload type
that matched one of the media payload types when providing caps via
codec_preferences without any sink pads.

Fixes

m=video 9 UDP/TLS/RTP/SAVPF 96
...
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 nack pli
a=fmtp:96 apt=96

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2046>
 2021-03-09 02:22:35 +00:00
Vivia Nikolaidou
4ccad5336f tests: Add negotiation tests for the interlace elements 
Many complicated cases exist. Would be good to have some checks.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2062>
 2021-03-08 21:02:13 +02:00
Ilya Kreymer
92626535c7 webrtc ice: Add 'min/max-rtp-port' props for setting RTP port range 
default min port == 0, max port == 65535 -- if min port == 0, uses existing random port selection (range ignored)
add 'gathering_started' flag to avoid changing ports after gathering has started
validity checks: min port <= max port enforced, error thrown otherwise
include tests to ensure port range is being utilized (by @hhardy)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/119>
 2021-03-01 14:42:17 +00:00
Michael Olbrich
5a03862fca h264parse: don't invalidate the last PPS when parsing a new SPS 
When a SPS is received then any previous PPS remains valid. So don't clear
the PPS flag from the parser state.

This is important because there are encoders that don't generated a PPS after
every SPS.

Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/571

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2019>
 2021-02-17 16:22:18 +00:00
He Junyan
be7a9e29df test: Add more test cases for the av1parse obu aligned output. 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1979>
 2021-01-26 12:22:31 +00:00
He Junyan
3e82c1f88e test: Add test cases for av1parse element. 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1614>
 2021-01-19 18:38:03 +00:00
Seungha Yang
d1e7290109 d3d11: Add support for packed 4:2:2 and 4:4:4 10bits formats 
Add support for Y210 and Y410 formats which are commonly used format
for en/decoders on Windows. Note that those formats cannot be used for
render target (output) of shader.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1821>
 2020-11-20 02:28:54 +09:00
He Junyan
12af439c58 test: av1parser: update the test result because of bug fixing. 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1464>
 2020-11-17 19:31:09 +00:00
Jan Schmidt
be131dba6a tests: Don't set dtlsenc state before linking. 
Link the dtlsenc in the testsuite before setting it to paused, as it
starts a pad task that can generate a not-linked error otherwise.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1744>
 2020-10-31 21:46:16 +11:00
Jan Schmidt
c1be9c53e1 dtls: Catch bus errors and fail instead of hanging. 
If the DTLS elements fail, they post a bus error and don't signal any
key negotiation. Catch the bus error and fail the test early instead
of letting it hang and time out.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1741>
 2020-10-31 01:47:06 +00:00
Seungha Yang
f62ecc1625 tests: Add CUDA filter unit tests 
Adding a test for buffer meta and colorspace conversion

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1633>
 2020-10-16 15:56:49 +00:00
Jan Alexander Steffens (heftig)
3ea6387f96 tests: svthevcenc: Fix test_encode_simple 
Pick the same I420 format the other test use. Without this, the source
picks AYUV64, which fails.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1573>
 2020-10-10 04:34:56 +00:00
Ederson de Souza
8335039ecd tests/avtp: Fix coverity issues 
Fixes sign extension issues, unchecked return values and some constant
expression results.

CID: 1465073, 1465074, 1465075, 1465076, 1465077
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1398>
 2020-09-28 18:40:43 +00:00
Seungha Yang
ea24a2e527 d3d11: Add support for packed 8bits 4:2:2 YUV formats 
Note that newly added formats (YUY2, UYVY, and VYUY) are not supported
render target view formats. So such formats can be only input of d3d11convert
or d3d11videosink. Another note is that YUY2 format is a very common
format for hardware en/decoders on Windows.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1581>
 2020-09-18 14:47:21 +00:00
Haihao Xiang
4a93f6e651 h265parse: recognize more HEVC extension streams 
There are streams which have the right general_profile_idc and
general_profile_compatibility_flag, but don't have the right extension
flags. We may try to use chroma_format_idc and bit_depth to
recognize these streams.

e.g.
https://www.itu.int/wftp3/av-arch/jctvc-site/bitstream_exchange/draft_conformance/SCC/IBF_Disabled_A_MediaTek_2.zip

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1328>
 2020-09-16 16:51:45 +00:00
Haihao Xiang
626af12498 h265parser: select the right profile for high throughput SCC stream 
Currently screen-extended-high-throughput-444 is recognized as
screen-extended-main-444, screen-extended-high-throughput-444-10 is
recognized as screen-extended-main-444-10 because they have the same
extension flags, so without this patch, it is possible that a decoder
which supports SCC but doesn't support throughput SCC will try to decode
a throughput SCC stream.

e.g.
https://www.itu.int/wftp3/av-arch/jctvc-site/bitstream_exchange/draft_conformance/SCC/HT_A_SCC_Apple_2.zip

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1328>
 2020-09-16 16:51:45 +00:00
Jordan Petridis
e4732fbbd5
validate: plug leak in gssdp 
These are triggered by the webrtcbin tests

https://gitlab.gnome.org/GNOME/gssdp/-/issues/10
 2020-09-14 14:42:36 +03:00
Matthew Waters
e2d88f0569 webrtc: propagate more errors through the promise 
Return errors on promises when things fail where available.

Things like parsing errors, invalid states, missing fields, unsupported
transitions, etc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1565>
 2020-09-14 04:04:29 +00:00