Mathieu Duponchelle [Thu, 18 Feb 2021 15:16:33 +0000 (16:16 +0100)]
compositor: fix drawing of transparent background
When drawing the background multithreaded, y_start needs to be
scaled to obtain the correct byte offset from which to start
memsetting (yoffset).
Fixes #871
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1042>
Kristofer Björkström [Thu, 25 Feb 2021 10:03:31 +0000 (11:03 +0100)]
gstrtspconnection: correct data_size when tunneled mode
gst_rtsp_connection_send_messages_usec in tunneled mode does base64
encode messages. When calculating data_size 1 bytes is added, which
results in ending the base64 with a NULL.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1051>
Robert Rosengren [Wed, 24 Feb 2021 12:46:04 +0000 (13:46 +0100)]
audio: Use GST_BUFFER_PTS instead of deprecated GST_BUFFER_TIMESTAMP
GST_BUFFER_PTS already used in audio code base (e.g. gstaudiodecoder),
so migrate completely from deprecated GST_BUFFER_TIMESTAMP for better
readability, as gstcompat.h defines GST_BUFFER_TIMESTAMP directly to PTS
anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1048>
Sebastian Dröge [Wed, 24 Feb 2021 17:51:40 +0000 (19:51 +0200)]
audioaggregator: Log if the sample rate of one sinkpad is not accepted
Otherwise this can silently cause not-negotiated errors without any
direct hint about what went wrong.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1049>
Francisco Javier Velázquez-García [Tue, 23 Feb 2021 13:40:02 +0000 (14:40 +0100)]
videotestsrc.c: Correct left shift operator
Use the left shift operator '<<' instead of the mistakenly typed less
than operator '<'.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1047>
Vivia Nikolaidou [Fri, 19 Feb 2021 14:44:35 +0000 (16:44 +0200)]
video-converter: Don't upsample/downsample/dither invalid lines
This is a fallout from the conversion to support multiple threads.
convert->upsample_p is never NULL now, it's always an allocated array of
n_threads potentially-null pointers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1043>
Jeongki Kim [Mon, 22 Feb 2021 06:36:53 +0000 (15:36 +0900)]
audioresample: Respect buffer layout when drain
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1045>
Jan Schmidt [Thu, 18 Feb 2021 15:27:07 +0000 (02:27 +1100)]
videoconvert: Only prefer upstream chroma-site with same subsampling.
If converting YUV formats with different chroma-subsampling, there's
probably no good reason to prefer the upstream chroma-siting so just use
the default for the output format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1033>
Jan Schmidt [Mon, 15 Feb 2021 14:24:15 +0000 (01:24 +1100)]
videoconvert: Implement more sophisticated colorimetry caps transfer
Implement a more sophisticated transfer of colorimetry and
chroma-site fields to output caps when fixating.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1033>
Jan Schmidt [Mon, 15 Feb 2021 07:34:33 +0000 (18:34 +1100)]
videoconvert: Forward colorimetry and chroma-site from upstream.
If downstream has expressed no preference for particular colorimetry
and chroma-site configuration, transfer them from the input caps.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/614
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1033>
Stéphane Cerveau [Tue, 19 Jan 2021 14:56:18 +0000 (15:56 +0100)]
decodebin3: change stream selection message owner
In order to select the streams on GST_MESSAGE_STREAM_COLLECTION,
the app needs to send the select-streams event
to the decodebin and not to the parsebin.
The message should be always owned by the decodebin.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1014>
Vivia Nikolaidou [Wed, 17 Feb 2021 10:41:06 +0000 (12:41 +0200)]
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>
Tim-Philipp Müller [Mon, 15 Feb 2021 16:05:30 +0000 (16:05 +0000)]
uridecodebin3: make caps property work
The caps set on uridecodebin3 via the "caps" property
were never passed to the internal decodebin3, so did
absolutely nothing.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/837
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1034>
Alicia Boya García [Fri, 12 Feb 2021 23:27:04 +0000 (00:27 +0100)]
videodecoder: Fix racy critical when pool negotiation occurs during flush
I found a rather reproducible race in a WebKit LayoutTest when a player
was intantiated and a VP8/9 video was loaded, then torn down after
getting the video dimensions from the caps.
The crash occurs during the handling of the first frame by gstvpxdec.
The following actions happen sequentially leading to a crash.
(MT=Main Thread, ST=Streaming Thread)
MT: Sets pipeline state to NULL, which deactivates vpxdec's srcpad,
which in turn sets its FLUSHING flag.
ST: gst_vpx_dec_handle_frame() -- which is still running -- calls
gst_video_decoder_allocate_output_frame(); this in turn calls
gst_video_decoder_negotiate_unlocked() which fails because the
srcpad is FLUSHING. As a direct consequence of the negotiation
failure, a pool is NOT set.
gst_video_decoder_negotiate_unlocked() still assumes there is a
pool, crashing in a critical in gst_buffer_pool_acquire_buffer()
a couple statements later.
This patch fixes the bug by returning != GST_FLOW_OK when the
negotiation fails. If the srcpad is FLUSHING, GST_FLOW_FLUSHING is
returned, otherwise GST_FLOW_ERROR is used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1031>
Jan Alexander Steffens (heftig) [Mon, 15 Feb 2021 16:22:47 +0000 (17:22 +0100)]
libs: audio: Fix gst_audio_buffer_truncate meta handling
In the non-interleaved case, it made `buffer` writable but then changed
the meta of the non-writable buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1035>
Alejandro González [Sat, 13 Feb 2021 21:25:18 +0000 (21:25 +0000)]
audioencoder: Fix gst_audio_encoder_get_audio_info return ownership GTK-Doc
GTK-Doc specifies that, by default, the caller owns returned objects, so that the caller should free them when it is done. However, in the case of this function, the returned GstAudioInfo is owned by the decoder, so this default choice is incorrect. This creates double free problems when using GStreamer Rust bindings, because they are generated using the information contained in the docs.
Fix this by correctly specifying that the caller does not own the returned object.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1032>
Alejandro González [Sat, 13 Feb 2021 17:24:37 +0000 (17:24 +0000)]
audiodecoder: Fix gst_audio_decoder_get_audio_info return ownership GTK-Doc
GTK-Doc specifies that, by default, the caller owns returned objects, so that the caller should free it when it is done. However, in the case of this function, the returned GstAudioInfo is owned by the decoder, so this default choice is incorrect. This creates double free problems when using GStreamer Rust bindings, because they are generated using the information contained in the docs.
Fix this by correctly specifying that the caller does not own the returned object.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1032>
Thibault Saunier [Fri, 8 Jan 2021 11:39:32 +0000 (08:39 -0300)]
encoding-profile: Plug a leak of factory list
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1002>
Thibault Saunier [Fri, 20 Nov 2020 21:35:49 +0000 (18:35 -0300)]
encodebin: Add APIs to set element properties on encoding profiles
User often want to set encoder properties on encoding profiles,
this introduces a way to easily 'preset' properties when defining the
profile. This uses GstStructure to define those properties the same
way it is done in `splitmux` for example as it makes simple to handle.
This also defines a more complex structure type where we can map a set
of properties to set depending on the muxer/encoder factory that has
been picked by EncodeBin so it is quite flexible.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1002>
Thibault Saunier [Fri, 20 Nov 2020 21:35:49 +0000 (18:35 -0300)]
encoding-profile: Cleanup profile serialization documentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1002>
Alexander Vandenbulcke [Fri, 8 Jan 2021 10:51:36 +0000 (11:51 +0100)]
gl/dispmanx: fix deadlock triggered by `set_render_rectangle`
When the gstglimagesink is started with the option `glimagesink
render-rectangle="<0,0,1920,1080>"`, the pipeline reaches a deadlock.
The reason the deadlock occurs is that the
`gst_gl_window_set_render_rectangle` takes locks on the window, in
addition it calls `window_class->set_render_rectangle(...)` which
executes the `_on_resize` function. Since the `_on_resize` function also
takes locks on the window the deadlock is achieved.
By scheduling the adjustment of the render rectangle through an async
message for `gst_gl_window_dispmanx_set_render_rectangle`, the actual
resize happens in another context and therefore doesn't suffers from the
lock taken in `gst_gl_window_set_render_rectangle`.
This solution follows the same approach as gl/wayland. The problem was
introduced by b887db1. For the full discussion check #849.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1030>
Vivia Nikolaidou [Thu, 4 Feb 2021 16:51:27 +0000 (18:51 +0200)]
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>
Vivia Nikolaidou [Thu, 4 Feb 2021 16:22:15 +0000 (18:22 +0200)]
videoscale: Support for alternate-field interlacing
Accept the negotiation, video-converter.c is aware of the half-height
already
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1027>
Vivia Nikolaidou [Tue, 2 Feb 2021 18:33:27 +0000 (20:33 +0200)]
videoconvert: Support for alternate-field interlacing
Treat the data just like normal data with half the height. Also treat it
as progressive when converting from/to I420 because it requires
different handling for chroma subsampling.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1027>
Havard Graff [Thu, 5 Apr 2018 09:59:52 +0000 (11:59 +0200)]
audio: add GstAudioLevelMeta
Will be used to implement RTP extension https://tools.ietf.org/html/rfc6464
Co-authored-by: Guillaume Desmottes <guillaume.desmottes@collabora.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/706>
Guillaume Desmottes [Tue, 26 Jan 2021 08:37:44 +0000 (09:37 +0100)]
rtpbasedepayload: add auto-header-extension property
Same property as the one I just added on rtpbasepayload.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1022>
Guillaume Desmottes [Tue, 26 Jan 2021 08:37:44 +0000 (09:37 +0100)]
rtpbasepayload: add auto-header-extension property
Using RTP header extensions is currently not convenient. Users have to
handle signals from the RTP payloader and instantiate the extension
element themselves, making it impossible to use with gst-launch.
Adding a property allowing the payloader to automatically try creating
extensions. This should help simple use cases and testing using
gst-launch.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1022>
Sebastian Dröge [Tue, 19 Jan 2021 08:23:12 +0000 (10:23 +0200)]
typefindfunctions: Consider the number and types of atoms found in a row for suggesting a probability
If there are 3 or more known atoms in a row, it's likely that this is
actually MOV/MP4 even if we don't find any other known atoms. If 5 or
more are found then this is most certainly MOV/MP4 and we can return.
Also if a moov and mdat atom is found, this is definitely a MOV/MP4 file
and can be used as such, independent of anything else following the
mdat.
Fixes typefinding of various MOV files that have no `ftyp` atom but
otherwise a valid file structure followed by some garbage.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1013>
Marijn Suijten [Fri, 29 Jan 2021 20:40:33 +0000 (21:40 +0100)]
gstaudiostreamalign: Pass self as const pointer in getter functions
It was noticed in [1] that `GstAudioStreamAlign` is a simple boxed type
that is passed as const in the copy function, but not as such in the
getters. These functions turn out to be the only users of `const = true`
overrides in `gstreamer-rs`. Since there is no locking or other advanced
caching/sharing going on (as happens with miniobjects) these functions
can safely take self as const pointer.
[1]: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/683#note_783129
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1025>
Jakub Adam [Mon, 27 Apr 2020 18:52:34 +0000 (20:52 +0200)]
video-hdr: Add API to check content light level equality
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/969>
Guillaume Desmottes [Tue, 26 Jan 2021 09:39:34 +0000 (10:39 +0100)]
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>
Guillaume Desmottes [Tue, 26 Jan 2021 09:39:34 +0000 (10:39 +0100)]
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>
Guillaume Desmottes [Mon, 25 Jan 2021 13:25:45 +0000 (14:25 +0100)]
rtpbasedepayload: fix clear-extensions signal definition
Typo as we were using the wrong enum.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1021>
Knobe, Daniel [Tue, 26 Jan 2021 13:05:48 +0000 (14:05 +0100)]
overlay/example: added qt core dependency for qt overlay example
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1020>
Guillaume Desmottes [Mon, 25 Jan 2021 13:25:45 +0000 (14:25 +0100)]
rtp: fix clear-extensions signal definition
Typo as we were using the wrong enum.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1018>
Guillaume Desmottes [Fri, 22 Jan 2021 09:10:05 +0000 (10:10 +0100)]
rtphdrext: fix typo in doc
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1018>
Matthew Waters [Fri, 22 Jan 2021 03:02:29 +0000 (14:02 +1100)]
gl/examples: fix recordgraphic example
Not ported to proper modern OpenGL though but that is the case for a lot
of the GL examples.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/859
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1016>
Marijn Suijten [Mon, 4 Jan 2021 22:25:10 +0000 (23:25 +0100)]
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>
Marijn Suijten [Thu, 14 Jan 2021 09:09:40 +0000 (10:09 +0100)]
audio/audio-buffer: @buffer in audio_buffer_map is out caller-allocates
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1005>
Marijn Suijten [Wed, 13 Jan 2021 10:07:47 +0000 (11:07 +0100)]
video/video-frame: @frame in video_frame_map is out caller-allocates
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1005>
Marijn Suijten [Tue, 12 Jan 2021 09:34:41 +0000 (10:34 +0100)]
audio,video: Add `out caller-allocates` to init and from_caps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1005>
Sebastian Dröge [Mon, 11 Jan 2021 10:25:33 +0000 (12:25 +0200)]
videosink: Add new GstVideoSink::set_info() virtual method
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/986>
Sebastian Dröge [Sat, 26 Dec 2020 12:16:34 +0000 (14:16 +0200)]
videosink: Implement more complete BaseSink::get_times() based on the framerate
This will only make use of the framerate if the subclass is chaining up
BaseSink::set_caps(). Otherwise it will have the same behaviour as the
basesink default.
Doing so is useful if video buffers don't contain a duration to
calculate a default duration, and various video sinks already implement
a custom version of this.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/986>
Marijn Suijten [Tue, 12 Jan 2021 09:36:34 +0000 (10:36 +0100)]
video: Convert info_to_caps to take self as const ptr
This requires a slight modification to the function itself because it
was overwriting a member locally.
However, now this side-effect cannot be observed outside the function
anymore.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1008>
Matthew Waters [Wed, 6 Jan 2021 00:16:18 +0000 (11:16 +1100)]
gl: add get_type() implementations for all of our memory types
Otherwise, various bindings can't really know the type of an object as
required.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/999>
Jakub Adam [Tue, 12 Jan 2021 21:08:14 +0000 (22:08 +0100)]
rtpbasepayload: fix header extension length calculation
Since ternary operator has the lowest precedence in the expressions at
hand, wordlen would always incorrectly yield 0 or 1.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1009>
Thibault Saunier [Thu, 7 Jan 2021 19:27:25 +0000 (16:27 -0300)]
giosrc: Ensure that an error is posted when underlying file is deleted
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1001>
Jakub Adam [Tue, 14 Jul 2020 18:44:18 +0000 (20:44 +0200)]
video-blend: fix blending 8-bit and 16-bit frames together
Replace hardcoded 255s with the correct max value for the given color
depth. Use 64-bit integer in calculations where overflow may occur.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1000>
Thibault Saunier [Mon, 7 Dec 2020 14:23:49 +0000 (11:23 -0300)]
encodebasebin: Ensure that parsers are compatible with selected encoders
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/845
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/963>
Matthew Waters [Tue, 5 Jan 2021 09:18:24 +0000 (20:18 +1100)]
gl: document some GL caps specifics
Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/854
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/996>
Tim-Philipp Müller [Mon, 4 Jan 2021 13:40:20 +0000 (13:40 +0000)]
tagdemux: resize and trim buffer in place to fix interaction with oggdemux
Elements operating in pull mode may optionally pass a buffer to
pull_range that should be filled with the data. The only element
that does that at the moment is oggdemux operating in pull mode.
tagdemux currently creates a sub-buffer whenever a buffer pulled
from upstream (filesrc, usually) needs to be trimmed. This creates
a new buffer, however, so disregards any passed-in buffer from a
downstream oggdemux.
This would cause assertion failures and playback problems for
ogg files that contain ID3 tags at the end.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/848
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/994>
Mathieu Duponchelle [Thu, 31 Dec 2020 16:25:40 +0000 (17:25 +0100)]
appsrc: fix signal documentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/992>
Edward Hervey [Wed, 30 Dec 2020 13:56:54 +0000 (14:56 +0100)]
videoaggregator: Pop out old buffers on timeout
This situation happens in the situation where an input stream has a framerate
exceeding the timeout latency (Ex: 1fps with a latency of 500ms) and an input
stream greater than output framerate (ex: 60fps in, 30 fps out).
The problem that would happen is that we would timeout, but then buffers from
the fast input stream would only be popped out one by one.... until a buffer
reaches the low-framerate input stream at which point they would quickly be
popped out/used. The resulting output would be "slow ... fast ... slow ... fast"
of that input fast stream.
In order to avoid this situation, whenever we detect a late buffer, check if
there's a next one and re-check with that one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/990>
Mathieu Duponchelle [Tue, 29 Dec 2020 22:44:42 +0000 (23:44 +0100)]
compositor/blend.c: fix MT checker pattern
When filling the checker pattern from multiple threads, y_start
needs to be taken into account to determine the shade of the
current pixel.
Example pipeline:
gst-launch-1.0 videotestsrc ! video/x-raw, width=1920, height=1080, format=I420 ! \
queue ! compositor sink_0::xpos=200 ! video/x-raw, format=I420 ! videoconvert ! \
xvimagesink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/988>
Mathieu Duponchelle [Tue, 29 Dec 2020 22:42:08 +0000 (23:42 +0100)]
compositor/blend.c: fix MT fill_checker for YUY2
The initial byte offset should be calculated from the stride,
not from the dest_add variable
Example pipeline:
gst-launch-1.0 videotestsrc ! video/x-raw, width=1920, height=1080, format=YUY2 ! \
queue ! compositor sink_0::xpos=200 ! video/x-raw, format=YUY2 ! xvimagesink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/988>
Mathieu Duponchelle [Tue, 29 Dec 2020 19:24:51 +0000 (20:24 +0100)]
compositor/blend: fix blending of subsampled components
The correct way to determine the byte offset at a certain yoffset
in a subsampled component is to shift the yoffset by the component's
hsub
This fixes out-of-bounds memory accesses and visible artefacts,
example pipeline with the samples from #802:
gst-launch-1.0 compositor name=vmixer sink_1::xpos=1910 sink_1::ypos=1080 ! \
videoconvert ! videorate ! xvimagesink \
filesrc location=VID_20200723_203606.mp4 ! decodebin name=demux1 ! \
queue ! videoflip method=vertical-flip ! vmixer. \
filesrc location=bridgeoverstubbledwater.mp4 ! decodebin name=demux2 ! \
queue ! vmixer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/988>
Sebastian Dröge [Thu, 24 Dec 2020 09:44:27 +0000 (11:44 +0200)]
decodebin3: When reconfiguring a slot make sure that the ghostpad is unlinked
This was only taken care of previously if there was a decoder before.
However if previously a decoder was not needed then the ghostpad
would've been linked directly to the slot's srcpad.
Reconfiguring the slot requires this to be undone so that linking can
happen normally.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/985>
Seungha Yang [Sun, 20 Dec 2020 13:35:45 +0000 (22:35 +0900)]
examples: overlay: Add an example for Win32 window handle with playbin
Demonstrate video overlay handling on Windows when playbin is in use
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/983>
Stéphane Cerveau [Tue, 15 Dec 2020 21:05:11 +0000 (22:05 +0100)]
hdr: update doc
update GST_VIDEO_HDR10_PLUS_MAX_ROWS_MD_APL and
GST_VIDEO_HDR10_PLUS_MAX_COLS_MD_APL
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/979>
Stéphane Cerveau [Fri, 3 Jul 2020 10:06:14 +0000 (12:06 +0200)]
videodecoder: Forward hdr-format info downstream
By default the hdr-format detected by a parser should
be passed to the downstream element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/650>
Stéphane Cerveau [Fri, 8 May 2020 09:06:42 +0000 (11:06 +0200)]
video-hdr: add hdr formats
Provide enum and helper method to set the hdr format
name in caps by example.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/650>
Stéphane Cerveau [Mon, 19 Oct 2020 13:36:58 +0000 (15:36 +0200)]
video-hdr: introduce HDR10+ parser
Video can now parse a HDR10+ data structure
coming from a SEI message.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/650>
Stéphane Cerveau [Mon, 27 Jan 2020 09:12:05 +0000 (10:12 +0100)]
video-hdr: add HDR10+ structure
Provides structure and GstVideoMeta
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/650>
Sebastian Dröge [Tue, 15 Dec 2020 13:07:31 +0000 (15:07 +0200)]
audiorate: Make buffer writable before changing its metadata
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/977>
Fabrice Fontaine [Mon, 14 Dec 2020 06:42:55 +0000 (07:42 +0100)]
gst-libs/gst/video/gstvideoaggregator.c: fix build with gcc 4.8
Fix the following build failure with gcc 4.8 which has been added with
https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/commit/
d268c193ad39fb970351ed62898be806ebd0a71e:
../gst-libs/gst/video/gstvideoaggregator.c: In function 'gst_video_aggregator_init':
../gst-libs/gst/video/gstvideoaggregator.c:2762:3: error: 'for' loop initial declarations are only allowed in C99 mode
for (gint i = 0; i < gst_caps_get_size (src_template); i++) {
^
Signed-off-by: Fabrice Fontaine <fontaine.fabrice@gmail.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/974>
Jordan Petridis [Thu, 10 Dec 2020 17:06:00 +0000 (19:06 +0200)]
libvisual: workaround clang warning
libvisual api expects a priv data pointer to be passed, though we know its
going to be `GstDebugLevel`.
```
../subprojects/gst-plugins-base/ext/libvisual/plugin.c:33:39: error: cast to smaller integer type 'GstDebugLevel' from 'void *' [-Werror,-Wvoid-pointer-to-enum-cast]
GST_CAT_LEVEL_LOG (libvisual_debug, (GstDebugLevel) (priv), NULL, "%s - %s",
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/975>
Edward Hervey [Fri, 11 Dec 2020 09:13:59 +0000 (10:13 +0100)]
decodebin3: Release selection lock when pushing EOS
We can't keep the lock otherwise this would lock other actions. In order to keep
it safe, we grab a list of peer pads to send EOS to with the lock taken, then
send to the peer pads with the lock released.
Also make sure the selection lock is taken for another call to this function
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/847
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/971>
Edward Hervey [Fri, 11 Dec 2020 10:40:22 +0000 (11:40 +0100)]
typefind: Fix debug category usage
Only register it in one place and use it everywhere from the header. Also make
sure it doesn't clash with the debug category from core.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/972>
Stéphane Cerveau [Fri, 4 Dec 2020 15:55:53 +0000 (16:55 +0100)]
typefindfunctions: allow per features registration
Split plugin into features including
typefind functions which can be indiviually
registered during a static build.
More details here:
https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/900>
Stéphane Cerveau [Tue, 1 Dec 2020 12:34:15 +0000 (13:34 +0100)]
pbtypes: allow per features registration
Split plugin into features including
dynamic types which can be indiviually
registered during a static build.
More details here:
https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/900>
Stéphane Cerveau [Mon, 30 Nov 2020 10:00:30 +0000 (11:00 +0100)]
subparse: allow per feature registration
Split plugin into features including
elements and device providers which
can be indiviually registered during
a static build.
More details here:
i
https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/900>
Julian Bouzas [Tue, 11 Aug 2020 15:47:02 +0000 (11:47 -0400)]
ogg: allow per feature registration
Split plugin into features including
elements and device providers which
can be indiviually registered during
a static build.
More details here:
https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/900>
Stéphane Cerveau [Thu, 22 Oct 2020 09:07:26 +0000 (11:07 +0200)]
playback: allow per feature registration
Split plugin into features including
elements and device providers which
can be indiviually registered during
a static build.
More details here:
https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/900>
Stéphane Cerveau [Fri, 16 Oct 2020 12:35:04 +0000 (14:35 +0200)]
alsa: allow per feature registration
Split plugin into features including
elements and device providers which
can be indiviually registered during
a static build.
More details here:
https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/900>
Thibault Saunier [Wed, 9 Dec 2020 13:40:40 +0000 (10:40 -0300)]
tests: Fix races in the read-growing-file test
First: the buffer is pushed on the encoding filesink only *after* the
checkpoints, leading to wrong ordering in expectations.
The second one is about EOS being posted on the bus *before* the
stop action is executed, leading to the action never being executed.
That action was plain useless as the default behavior of validate
scenario is to internally create a "stop" action on EOS (unless
disabled by user).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/968>
Thibault Saunier [Thu, 19 Nov 2020 22:17:18 +0000 (19:17 -0300)]
giosrc: Add support for growing source files
Add a way for applications to specify that the underlying file is
growing which implies that the source won't EOS when reaching the end
of the file but instead start monitoring it and start reading it again
whenever a change is detected.
Also add a validate test to check the behavior
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/937>
Seungha Yang [Wed, 25 Nov 2020 19:04:48 +0000 (04:04 +0900)]
video: Make use of gst_video_chroma_site_{from,to}_string() API
Replace deprecated gst_video_chroma_{from,to}_string()
to newly added gst_video_chroma_site_{from,to}_string()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/927>
Seungha Yang [Tue, 10 Nov 2020 09:01:12 +0000 (18:01 +0900)]
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>
Jakub Adam [Mon, 30 Nov 2020 20:38:08 +0000 (21:38 +0100)]
rtpbasepayload: pass optional caps fields in a GstStructure
For more flexibility, allow to pass the extra output caps fields as
a GstStructure.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/952>
Ratchanan Srirattanamet [Thu, 5 Dec 2019 13:08:36 +0000 (20:08 +0700)]
audiobasesrc: always acquire if not acquired in _setcaps
audiobasesrc's setcaps contains an optimization that makes it not re-
acquire the ringbuffer if the caps have not changed. However, it doesn't
check if it has successfully acquired it or not. It's possible to have
the caps set but not having ringbuffer acquired if the previous attempt
to acquire fails.
This commit replaces the caps existence check with whether the
ringbuffer is acquired or not. There's no need to check for caps
existence because 1.) it's unlikely to be NULL if the ringbuffer is
acquired, and 2.) _setcaps shouldn't be called with a NULL caps.
This should also let the element retry on acquiring ringbuffer after an
error by re-setting the element's state to READY and back to PLAYING.
Whether this behavior is correct is up for debate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/512>
He Junyan [Fri, 6 Nov 2020 06:05:39 +0000 (14:05 +0800)]
glimagesink: Avoid assert in query.
The sink_query just uses context, other_context and display to query info.
But all these objects can be changed or distroyed in state_change() func
and other places.
This patch is not very perfect. The condition race still exists in other
places in this element. All the functions directly access these objects
without protection. Most of them are executed when the data is pushing and
draw context/window have already been established, so they should not have
problems. But the sink_query and propose_allocation functions are the query
-like functions and executed in query context, which can be called in any
state of the element. So it can cause some crash issues because of destroyed
context object.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/922>
He Junyan [Fri, 6 Nov 2020 08:18:14 +0000 (16:18 +0800)]
glcolorconver: Return empty caps in transform_caps if fails.
We should not return a NULL in transform_caps() function. The NULL
will generate a assert of:
"transform_caps returned caps which are not a real subset of the
filter caps"
in transform base class.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/922>
He Junyan [Fri, 6 Nov 2020 07:36:01 +0000 (15:36 +0800)]
glbasefilter: Need to check the display before lock it.
In find_gl_context_unlocked(), the display of filter may be NULL
and can cause crash if we directly access and lock it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/922>
He Junyan [Fri, 6 Nov 2020 07:36:21 +0000 (15:36 +0800)]
glbasefilter: Delete the un-paired unlock in change_state().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/922>
Arun Raghavan [Thu, 3 Dec 2020 11:50:07 +0000 (06:50 -0500)]
audioencoder: Fix incorrect GST_LOG_OBJECT usage
GstBuffer is not a GstObject, so this causes a warning to be emitted.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/956>
Matthew Waters [Fri, 10 Jul 2020 05:33:46 +0000 (15:33 +1000)]
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-$NUM' field is present in the src caps, then an
extension implementation will be requested but is not required to be able
to negotiate correctly. An extension will be requested using the
'request-extension' signal if none could be found internally.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/748>
Matthew Waters [Fri, 10 Jul 2020 05:30:57 +0000 (15:30 +1000)]
rtp/basedepayload: implement support for rtp header extensions
New signals are added for managing the internal list of rtp header
extension implementations read by a specific depayloader instance.
If the 'extmap-$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>
Matthew Waters [Fri, 10 Jul 2020 05:14:09 +0000 (15:14 +1000)]
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>
Mart Raudsepp [Thu, 26 Nov 2020 12:52:26 +0000 (14:52 +0200)]
gl/eagl: Fix automatic resize behaviour
https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/681
added a layoutSubViews, which never gets called, because it should have been
called layoutSubviews (non-capital "v"). However after fixing that, it still
doesn't work correctly, because window_width/height values are immediately
updated and then draw_cb will never trigger the resize path, because the
values are already up to date.
Update the values inside the resize path again instead, so the check for
entering the resize path is logically always correct.
This makes the layoutSubviews unnecessary, as it only updated the internal
size values prematurely, so it is deleted instead of method naming fixed.
These changes were originally done to avoid accessing UIKit objects on the
main thread, but no additional accesses are added here, only internal
private variable assignments under the same draw_lock, so there should be
no threading issues reintroduced.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/945>
Mart Raudsepp [Thu, 26 Nov 2020 12:45:05 +0000 (14:45 +0200)]
gl/eagl: Fix resize condition check in draw_cb to not get called unnecessarily
A CGSize contains CGFloat values (a typedef to double or float), which means
that the values aren't equal, despite it being equal after they are cast to
int by assigning them to window_height/width private members. This leads to
excessive gst_gl_window_resize calls on each frame, at least if the CGFloat
value has a .5 decimal value, e.g. 103.5.
Fix it by storing them as CGFloat instead of gint.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/945>
Matthew Waters [Fri, 20 Nov 2020 05:51:09 +0000 (16:51 +1100)]
video/converter: increase the number of cache lines for resampling
The exising hardcoded max default does not account for the possible
-1 offset when retrieving lines for resampling. As a result, when
another chain has the same number of cache lines (4), the resample
operation would be attempting to generate 5 lines with a cache size
of 4 and would overwrite the first cache line.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/821
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/938>
Sebastian Dröge [Tue, 1 Dec 2020 11:13:40 +0000 (13:13 +0200)]
compositor: Don't crash in prepare_frame() if the pad was just removed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/950>
Guillaume Desmottes [Tue, 1 Dec 2020 08:31:48 +0000 (09:31 +0100)]
videometa: gir annotate the size of plane array in new API
Fix #838
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/949>
Thibault Saunier [Thu, 19 Nov 2020 22:06:12 +0000 (19:06 -0300)]
encodebin: Enhance algorithm to check if profiles can be plugged to muxer
In particular muxing sinks have specific set of sinkpads and we need
to handle that.
Thibault Saunier [Thu, 19 Nov 2020 22:03:29 +0000 (19:03 -0300)]
encodebin: Ignore element classification when a muxer name specified
Instead of going through the list of known muxers go ahead and
instantiate the muxer specified as 'preset name' as this specifies
the exact element factory name to use.
Thibault Saunier [Fri, 13 Nov 2020 19:32:45 +0000 (16:32 -0300)]
pbutils: Add support for muxing sinks usage in encoding profiles
Thibault Saunier [Fri, 13 Nov 2020 19:32:45 +0000 (16:32 -0300)]
encoding: Implement encodebin2 with an ON_REQUEST src pad
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/304
Thibault Saunier [Fri, 13 Nov 2020 16:06:32 +0000 (13:06 -0300)]
encodebin: Split implementation into a base class
Create EncodeBaseBin as a base class for the existing
encodebin to allow other implementations.
Co-authored with Jan Schmidt <jan@centricular.com>
Seungha Yang [Fri, 27 Nov 2020 19:14:14 +0000 (04:14 +0900)]
glcontext: wgl: Implement check_feature vfunc
There are several WGL specific extenstions such as WGL_NV_DX_interop.
Currently we have no WGL specific extension support and
this commit is also only for debugging purpose.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/947>
Sanchayan Maity [Fri, 27 Nov 2020 06:11:36 +0000 (11:41 +0530)]
audiodecoder: Move max_errors out of GstAudioDecoderContext
Currently max-errors gets set during init to default or via property.
However, if a decoder element calls gst_audio_decoder_reset with 'full'
argument set to TRUE, it would result in all the fields of context being
zeroed with memset. This effectively results in max-errors getting a
value of 0 overriding the default or user requested value set during
init.
This would result in calls to GST_AUDIO_DECODER_ERROR which track error
counts and allow max-errors, to be ineffective.
To fix this move max-errors out of GstAudioDecoderContext, as changes to
context should not affect this. The error_count is anyways also in
GstAudioDecoderPrivate and not in context.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/946>