Commit graph

689 commits

Author SHA1 Message Date
Tim-Philipp Müller
e539f0cd67 Use new gst_buffer_new_memdup()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1170>
2021-05-26 11:46:27 +00:00
Matthew Waters
f03071439f gl/api: improve the to/from string for GstGLAPI/GstGLPlatform
With unit tests now!

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/809>
2021-05-13 15:35:23 +10:00
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
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
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
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
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
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
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
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
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
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
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
Guillaume Desmottes
02fd2f12f9 audio: add gst_audio_make_raw_caps()
More binding friendly version of GST_AUDIO_CAPS_MAKE().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/676>
2020-06-02 11:57:42 +00:00
Guillaume Desmottes
84e0689d58 video: add gst_video_make_raw_caps()
More binding friendly version of GST_VIDEO_CAPS_MAKE().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/676>
2020-06-02 11:57:42 +00:00
Seungha Yang
7d7108f35d tests: audiosink: Test class extension struct
Test a vfunc which belongs to GstAudioSinkExtension struct.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/547>
2020-05-28 19:14:29 +09:00
Matthew Waters
6fc33560e1 tests/gl: add test for GL context removal
Tests functionality fixed by:

https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/654

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/656>
2020-05-08 15:10:17 +10:00
He Junyan
5bb8bdf90d test: pbutils: Add check for high throughput scc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/645>
2020-04-28 21:27:36 +08:00
Seungha Yang
1dee0f05a7 video-hdr: Rework for GstVideoMasteringDisplayInfo and GstVideoContentLightLevel struct
This commit modifies GstVideoMasteringDisplayInfo and GstVideoContentLightLevel
structs so that each value is to be more like hdr_metadata_infoframe struct
of linux drm header and DXGI_HDR_METADATA_HDR10 struct of Windows.
So each value is no more fraction but normalized one as per CTA 861.G spec.
Also the unit of each value will be consistent with H.264, H.265
specifications, hdr_metadata_infoframe struct for linux and
DXGI_HDR_METADATA_HDR10 struct for Windows.
2020-04-01 11:11:15 +00:00
Miguel Paris
f265e5cbd5 rtpbuffer: add_extension_onebyte_header: fix the proper wordlen
The wordlen ("length") MUST represent the total "number of 32-bit words
in the extension, excluding the four-octet extension header" (rfc3550).
There are cases where already existent padding is reused for adding
the new extension. So the new wordlen should be updated if the new
added extension makes it to increase.
2020-03-19 14:18:20 +01:00
Philippe Normand
7240cad9c5 navigation: Mouse scroll events support
This patch introduces a new API to send and parse mouse scroll events. Mouse
event coordinates are sent relative to the display space of the related output
area. This is usually the size in pixels of the window associated with the
element implementing the GstNavigation interface.
2020-03-19 09:59:47 +00:00
Guillaume Desmottes
ea2619aadc video: fix GST_VIDEO_FRAME_IS_BOTTOM_FIELD()
GST_VIDEO_FRAME_FLAG_BOTTOM_FIELD is a subset of
GST_VIDEO_FRAME_FLAG_TOP_FIELD so needs to be checked accordingly.

Fix #726
2020-02-26 16:15:59 +00:00
Guillaume Desmottes
26f386ce8b video: add macros checking for GST_VIDEO_BUFFER_FLAG_TOP/BOTTOM_FIELD flags
The GST_VIDEO_BUFFER_FLAG_TOP_FIELD flag is a superset of
GST_VIDEO_BUFFER_FLAG_BOTTOM_FIELD as they are defined using other
flags. As a result we can't use GST_BUFFER_FLAG_IS_SET() to check for
those flags.
2020-02-26 16:15:59 +00:00
Håvard Graff
85e201fe30 rtpbasepayload: add property for embedding twcc sequencenumbers
By setting the extension-ID for TWCC (Transport Wide Congestion Control),
the payloader will embed sequencenumbers as a RTP header-extension
according to https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01#section-2

The negotiation of this being enabled with downstream elements
is done with caps reflecting the way this is communicated using SDP.
2020-02-14 09:40:59 +00:00
Havard Graff
7283a45afe rtpbasepayload: fix test warnings
Compiling with MSVC and Clang.
2020-02-11 14:06:45 +00:00
Kristofer Björkström
4152b0c840 rtpbasepayload: timestamp bug, if rate control=no
With commit "basepayload: Expose onvif-no-rate-control property" the rtp
timestamp changed behaviour when rate control is disabled.

When disabling rate control, we must take care of the stream time to
avoid the timestamps to begin from zero again.
2020-02-11 12:30:49 +00:00
Nicolas Dufresne
104458071a tests: rtpbasedepayload: Test flow return whith push/push_list
This validate that the base class properly save and return the flow
return value received when gst_rtp_base_depay_push/push_list() helper is
being used.
2020-01-11 19:39:55 -05:00
Aaron Boxer
807418894b rtspurl: add API method to create request uri combined with control url
code logic very similar to gst_rtsp_url_get_request_uri ()
2019-12-27 16:57:08 +00:00
Stéphane Cerveau
66df967dab tests: add video encoder test with subframes API 2019-12-21 02:59:14 +00:00
Olivier Crête
6b283d9e78 Revert "videoencoder: factor out logic from gst_video_encoder_finish_frame()"
This reverts commit b1ec312b8e.
2019-12-19 17:52:12 -05:00