Commit graph

1627 commits

Author SHA1 Message Date
Haihao Xiang
c778686a3c test: enlarge the number
This is to make sure the case can pass after adding new video formats.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1141>
2021-05-11 12:24:41 +08:00
Sebastian Dröge
26b8a96b84 appsrc: Add test for testing the max-* and leaky-type properties
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1133>
2021-05-05 15:13:33 +00:00
François Laignel
ca7a964fb1 Use gst_element_request_pad_simple...
Instead of the deprecated gst_element_get_request_pad.
2021-05-05 11:55:54 +03:00
Doug Nazar
27c392bda3 tests/tcp: Fail if unable to start pipelines.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1105>
2021-04-20 09:49:23 +00:00
Doug Nazar
a273573d1e overlaycomposition: Fix test for big endian.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1103>
2021-04-12 04:39:49 -04:00
Jakub Adam
1a87a6572e rtpbasedepayload: handle caps change partway through buffer list
While preparing a blist for pushing, some RTP header extension may
request caps change for a specific buffer in the list. When this
happens, depayloader should immediately push those buffers from the list
that precede the currently processed buffer (for which the caps change
was requested) and only then apply the new caps to the src pad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1011>
2021-03-12 18:45:04 +01:00
Vivia Nikolaidou
2527c8f9f8 libs: audio: Handle meta changes in gst_audio_buffer_truncate
Set timestamp and duration to GST_CLOCK_TIME_NONE unless trim==0,
because that function doesn't know the rate and therefore can't
calculate them. Set offset and offset_end to appropriate values. Make it
clear in the documentation that the caller is responsible for setting
the timestamp and duration.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/869

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1039>
2021-02-18 11:25:32 +02:00
Vivia Nikolaidou
278b10dd2e videoconvert,videoscale: Add alternate-field negotiation tests
Make sure buffers with alternate-field interlacing mode can be
negotiated

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1027>
2021-02-04 21:47:27 +02:00
Guillaume Desmottes
df9064fdc6 rtpbasedepayload: set attributes on newly requested extensions
Users were supposed to configure the extension themselves but it was
impossible to do so as they didn't have access to the caps.

Fix #864

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1021>
2021-01-27 09:48:49 +01:00
Guillaume Desmottes
912cf46b83 rtpbasepayload: set attributes on newly requested extensions
Users were supposed to configure the extension themselves but it was
impossible to do so as they didn't have access to the caps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1021>
2021-01-27 09:48:49 +01:00
Marijn Suijten
abb026ec6a gl,video: Make ptrs to VideoInfo and (GL)AllocationParams immutable
These parameters are incorrectly regarded as mutable in G-IR making them
"incompatible" with languages that are explicit about mutability like
Rust. In order to clean up the code and expected API there, update the
signatures here, right at the source (instead of overriding them in
Gir.toml and hoping for the best).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1005>
2021-01-14 11:53:10 +00:00
Seungha Yang
410efd196a video-chroma: Add support for any combination of chroma-site flags
We've been allowing only a few known chroma-site values such as
jpeg (not co-sited), mpeg2 (horizontally co-sited) and
dv (co-sited on alternate lines). That's insufficient for
representing all possible chroma-site values. By this commit,
we can represent any combination of chroma-site flags.
But, an exception here is that any combination with
GST_VIDEO_CHROMA_SITE_NONE will be considered as invalid value.

For any combination of chroma-site flags,
gst_video_chroma_to_string() method is deprecated in order to
return newly allocated string via a new gst_video_chroma_site_to_string()
method. And for consistent API naming, gst_video_chroma_from_string()
is also deprecated. Newly written code should use
gst_video_chroma_site_from_string() instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/927>
2020-12-08 07:21:28 +00:00
Matthew Waters
7a53fbad68 rtp/basepayload: implement support for rtp header extensions
New signals are added for managing the internal list of rtp header
extension implementations read by a specific depayloader instance.

If the 'extmap-$NUM' field is present in the src caps, then an
extension implementation will be requested but is not required to be able
to negotiate correctly.  An extension will be requested using the
'request-extension' signal if none could be found internally.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/748>
2020-12-03 10:19:32 +00:00
Matthew Waters
092ea647bb rtp/basedepayload: implement support for rtp header extensions
New signals are added for managing the internal list of rtp header
extension implementations read by a specific depayloader instance.

If the 'extmap-$NUM' field is present in the sink caps, then an
extension implementation will be requested but is not requited to be
able to negotiate correctly.  An extension will be requested using the
'request-extension' signal if none could be found internally.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/748>
2020-12-03 10:19:32 +00:00
Matthew Waters
427c3f4442 rtp: add base object for reading/writing rtp header extensions (RFC5285)
Facilitates the creation of rtp header extension implementations that
can be reused across applications.

Implementations are registered into the GStreamer registry as elements
(idea from GstRTSPExtension) and can be retrieved by URI or filtered
manually.  RTP header extensions must have the classification
"Network/Extension/RTPHeader" to be considered as a RTP Header
extension.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/777
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/748>
2020-12-03 10:19:32 +00:00
Marijn Suijten
3ec795f613 audio: Move fill_silence into audio_format_info
With the function named gst_audio_format_fill_silence it would get
associated to the GstAudioFormat type in .gir which is incorrect and
confusing. See [1] for the discussion sparking this change.

https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/630#note_694795

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/940>
2020-11-25 19:18:25 +01:00
Mathieu Duponchelle
c50f4477ec video-converter: switch to using a task pool ..
.. and make use of that API in videoaggregator.

When setting certain properties, such as cropping or the scaled
size of pads, a new converter is created by videoaggregator.

Before that patch, this implied spawning new threads, potentially
at each aggregate cycle when interpolating pad properties. This
is obviously wasteful, and re-using a task pool removes that
overhead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/896>
2020-11-12 17:38:34 +00:00
Thibault Saunier
d268c193ad videoaggregator: Guarantee that the output format is supported
In the case `videoaggregator` is set as allowing format conversions,
and as we convert only on the sinkpads, we should ensure that the
chosen format is usable by the subclass. This in turns implies
that the format is usable on the srcpad.

When doing conversion *any* format can be used on the sinkpads, and this
is the only way that we can avoid race conditions during renegotiations
so we can not change that fact, we just need to ensure that the chosen
intermediary format is usable, which was not actually ensured before
that patch.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/834

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/909>
2020-11-03 00:10:31 +00:00
Nicolas Dufresne
db4567152d tests: allocator: Fix FDMemory portability issue
This fixes few issues in the test but mainly some portability issue reported
on Ubutun. The test now uses a randomly name tempory file located into system
default tempory location and uses glib wrappers when available.

Fixes !895

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/901>
2020-10-29 09:45:25 +00:00
Tobias Ronge
e2a1aa44df fdmemory: Allow for change of protection mode
After a memory has been unmapped, protection mode can now be changed
when mapping it again.

See https://bugzilla.gnome.org/show_bug.cgi?id=789952.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/895>
2020-10-28 17:11:05 +00:00
Seungha Yang
615b1ac579 tests: appsrc: Fix unstable test case
Wait all buffers to be consumed before sending flush seek event,
so that checking timestamp and segment as expected.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/816>
2020-10-14 10:57:19 +00:00
Will Miller
ac72a6adaa gstrtpbuffer: fix header extension length validation
We validate the header extensions length of an RTP buffer by comparing
it against the block size. Since we multiply the length in words by 4 to
get the length in bytes, a suitably large length could cause a wrapround
of the uint16, giving a lower length which erroneously passes the check
and allows the buffer to be mapped.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/864>
2020-10-12 15:01:22 +01:00
Matthew Waters
52793dbfca tests: add gl structs to abi check
Tested on x86, x86_64, armv7l, aarch64.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/854>
2020-10-09 06:12:30 +00:00
Marijn Suijten
d0f36c7e13 video: Rename video_color_transfer to video_transfer_function
Rename remaining `gst_video_color_transfer_{encode,decode}` functions on
the `GstVideoTransferFunction` enumeration to
`gst_video_transfer_function_{encode,decode}` permitting
gobject-introspection to turn these into associated functions and place
them under the respective `<enumeration>` block in gir XML files.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/805>
2020-09-12 09:46:44 +03:00
Sebastian Dröge
40a1e01740 glmixer: Fix unit test to actually work reliably
Don't run the harness in live mode, or otherwise it would output frames
already in the very beginning before a buffer was provided to it due to
timeout.

Also send EOS/a second buffer before pulling a buffer as videoaggregator
has one frame of latency.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/812>
2020-09-10 14:19:04 +03:00
Sebastian Dröge
91ec4e06d7 video: Rename gst_video_color_transfer_*() to gst_video_transfer_function_*() in new API
The type is called GstVideoTransferFunction so the function names should
match, otherwise gobject-introspection is keeping the functions as
global functions instead of methods on the type.

The same mistake was also made in lots of other APIs over the years, but
here we can at least fix it for 1.18 still.

Thanks to Marijn Suijten for noticing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/807>
2020-09-07 13:04:20 +03:00
Sebastian Dröge
61064257ef videoaggregator: Update for additional info parameter to the "samples-selected" signal
See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/590

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/780>
2020-08-07 09:34:37 +03:00
Guillaume Desmottes
dd5f7f1bf9 gl: move each gl platform specific API to its own gir
With contributions from:
Thibault Saunier <tsaunier@igalia.com>
Matthew Waters <matthew@centricular.com>

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/651

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/661>
2020-08-06 04:09:09 +00:00
Mathieu Duponchelle
1de8af6f8b videoaggregator: update to new samples selection API
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/778>
2020-08-05 20:09:52 +02:00
Jordan Petridis
66ff1eedca tests/check/elements/audioresample.c: avoid implict int ot float conversion
Also use doubles instead so the calculation won't overflow

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/773>
2020-08-04 17:32:31 +03:00
Mathieu Duponchelle
2faeb7d394 videoaggregator: implement samples selection API
Call gst_aggregator_selected_samples() after filling the queues
(but before preparing frames).

Implement GstAggregator.peek_next_sample.

Add an example that demonstrates usage of the new API in combination
with the existing buffer-consumed signal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/728>
2020-07-31 07:54:56 +00:00
Matthew Waters
a1e9f4e37b rtpbasepayload: place twcc-ext-id behind environment variable
Adding properties for each and every rtp header extension is not
scalable and a new interface will be implemented for the general case
(https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/777).

Set the environment variable "GST_RTP_ENABLE_EXPERIMENTAL_TWCC_PROPERTY"
to any value to reenable the short-lived twcc-ext-id property.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/761

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/756>
2020-07-21 11:57:55 +00:00
Olivier Crête
cb6edaf6f8 videorate: Error out on streams with no way to guess framerate
This is better than going into an infinite loop.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/761>
2020-07-20 22:05:57 +00:00
Olivier Crête
323554a31a videorate: Add test that reproduces infinite loop
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/761>
2020-07-20 22:05:57 +00:00
Havard Graff
36fec290a3 test/rtp: use the proper _INIT for initializing rtp/rtcp buffer structs.
Fixes -Wmissing-field-initializers in Clang.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/757>
2020-07-15 16:57:01 +02:00
Havard Graff
c488fd74a0 rtpbasedepayload: test warning fixes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/757>
2020-07-15 16:57:01 +02:00
Nicolas Dufresne
98b44fdb46 video: Add support for linear 32x32 NV12 tiles
This adds linear 32x32 NV12 based tiles. This format is notably used by
Allwinner VCU and exposed in V4L2 as being "SUNXI Tiled" format. In this
patch we generalize the plane info calculation so we can share this part
with the 4L4 variant.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/754>
2020-07-14 21:43:56 -04:00
Nicolas Dufresne
7d1028424c video: Add NV12_4L4 tile format
This format is produced by Verisillicon VC8000D VPU decoder, it is a simple 4x4
tiling layout in a linear way.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/753>
2020-07-14 17:33:31 +00:00
Santiago Carot-Nemesio
93cb325fa1 rtcpbuffer: Notify error in case packet can not be added to an RTCP compound packet
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/476>
2020-07-10 14:16:10 +00:00
Seungha Yang
cb34faaa17 tests: appsrc: Add unit test for custom segment
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/663>
2020-07-10 07:52:53 +00:00
Vivia Nikolaidou
1d0ccf8baa video-color: Add bt601 transfer function
Functionally the same as 709 but technically has a different value, and
external software (e.g. ffmpeg) finds "wrong" values produced by
GStreamer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/724>
2020-07-03 11:57:49 +03:00
Hosang Lee
f84f7a2cec tests: subparse: add test for webvtt without hour component
Test for webvtt without hour component.
mm:ss.000
2020-06-18 09:06:32 +09:00
Jan Schmidt
205bb066ed video-converter: Add checks for configuration sanity.
If the cropping or scaling input or output rects put us completely
outside the input/output frame respectively, we can't draw anything
except black safely. Check for those conditions and don't set up a
configuration that attempts to access out of bounds memory outside
the input/output framebuffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/696>
2020-06-12 06:49:56 +00:00
Jan Schmidt
bf5d51c5da video-converter: Guard against invalid frame input
If the frames passed in to gst_video_converter_frame()
have a different layout than was configured for, the
conversion code might go out of bounds and crash.

Do a sanity check on each frame passed in, and in the
absence of a return value in the API, just
refuse the conversion in invalid cases and leave the
destination frame untouched so it's obvious to
users that it was broken.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/696>
2020-06-12 06:49:56 +00:00
Guillaume Desmottes
1b4ab9f033 tests: enforce I420 format
Tests are assuming video is I420 but are not actually enforcing it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/689>
2020-06-09 08:09:58 +00:00
Edward Hervey
78444fc622 tests: Avoid hang with decodebin test
When adding elements dynamically to a pipeline one should never guess what the
curren/target state is, and instead use `gst_element_sync_state_with_parent()`.

Fixes racy hang when running within valgrind

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/692>
2020-06-08 08:11:00 +02:00
Sebastian Dröge
954a314ca8 videoencoder: Add test for min-force-key-unit-interval property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/684>
2020-06-05 10:04:43 +00:00
Sebastian Dröge
76364ebfe7 videoencoder: Also don't request a new key-unit if we already got one after the requested running time
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/684>
2020-06-05 10:04:43 +00:00
Sebastian Dröge
931b5ad996 videoencoder: Add test for correct force-keyunit event handling
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/684>
2020-06-05 10:04:43 +00:00
Sebastian Dröge
01eecc69bd videoencoder: Fix force-keyunit handling in test
This now behaves according to the videoencoder API instead of some other
signalling.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/684>
2020-06-05 10:04:43 +00:00