| Commit message (Collapse) | Author | Age | Files | Lines |
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Related to https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1141
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2236>
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Updating for removed d3d11videosink wrapper bin and the change of
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2113
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2169>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2171>
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This element can be useful for CI purposes on machines not running any system
audio daemon. The element implements the GstStreamVolume interface.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2125>
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Instead of requiring interlaced video, simply skip CC detection
when the input is progressive.
This allows placing line21decoder unconditionally in pipelines,
without having to worry about whether the input stream will be
interlaced, or even worse interlacing just in case!
+ update doc cache
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1885>
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Useful when having a service that runs a GStreamer pipeline
or application in Google Cloud to avoid storing the inputs
and outputs in the running container or service. For example
when analyzing a video from a Google Cloud Storage bucket
and extracting images or converting the video and then uploading
the results into another Google Cloud Storage bucket.
- gssrc allows to read from a file located in Google Cloud
Storage and it supports seeking.
- gssink allows to write to a file located in Google Cloud
Storage. There are 2 modes, one similar to multifilesink and
the other similar to filesink.
Example:
gst-launch-1.0 gssrc location=gs://mybucket/videos/sample.mp4 ! decodebin ! glimagesink
gst-launch-1.0 playbin uri=gs://mybucket/videos/sample.mp4
gst-launch-1.0 videotestsrc num-buffers=5 ! pngenc ! gssink object-name="img/img%05d.png" bucket-name="mybucket" next-file=buffer
gst-launch-1.0 filesrc location=sample.mp4 ! gssink object-name="videos/video.mp4" bucket-name="mybucket" next-file=none
When running locally simply set GOOGLE_APPLICATION_CREDENTIALS. But
when running in Google Cloud Run or Google Cloud Engine, just set the
"service-account-email" property on each element.
Closes #1264
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1369>
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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>
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Especially specify the field-order in the interleaved mode. Otherwise it
might cause the negotiation to fail, because
GST_PAD_SET_ACCEPT_INTERSECT is not set on the sinkpad, and the
field-order is missing in the sink template but can be present in the
outside caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2062>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2054>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2050>
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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>
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Fix vkcoloconvert and vkviewconvert long names.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2034>
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* Update class metadata
* for wrapper bin elements to be distinguishable from internal element.
* D3D11 -> Direct3D11 for consistency
* Add missing Since mark everywhere
* Update plugin cache
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2029>
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Those are supported (to a certain extent) so we should not limit
ourself to baseline
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1789>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1614>
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Various software, including ffmpeg's Decklink support, fails parsing CDP
packets that contain anything but CC data in the CDP packets.
Based on this property, timecodes are not written into the CDP packets
even if they're present.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1833>
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rtpsrc tries to do a lookup of the caps based on the encoding-name. For
not so standard encodings, the caps can be set, avoiding the lookup.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1406>
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Makes the plugin a tad more useful :)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1845>
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The same way as playbinX does it as it is often quite useful
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This allows supporting muxing sinks like hlssink2 or splitmux
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The way it is usable by encodebin2. This is what splitmux does already.
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They're just subsets of the high profile.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1634>
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They're subsets of the high profiles with no interlacing and
no B-frames for constrained
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1634>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1621>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1621>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1771>
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This causes no changes to the profile but keeps the existing settings.
The profile can also be changed from e.g. the card's configuration
application and in that case probably should be left alone.
The default is the new value as it keeps the profile setting as it is,
which is consistent with the previous behaviour in 1.18.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1721>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1151>
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forward alignment and num-stripes caps properties
Use caps height when setting caps for subframe
We want downstream to use full frame height, not subframe height
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1653>
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Thanks to @kazz_naka on Twitter
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1691>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1665>
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One was forgotten in 309f6187fef890c7ffa49305f38e89beac3b1423.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1617>
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Currently for buffer splitting only output duration can be specified.
Allow specifying a buffer size in bytes for splitting.
Consider a use case of the below pipeline
appsrc ! rptL16pay ! capsfilter ! rtpbin ! udpsink
Maintaining MTU for RTP transfer is desirable but in a scenario
where the buffers being pushed to appsrc do not adhere to this,
an audiobuffersplit element placed between appsrc and rtpL16pay
with output buffer size specified considering the MTU can help
mitigate this.
While rtpL16pay already has a MTU setting, in case of where an
incoming buffer has a size close to MTU, for eg. with a MTU of
1280, a buffer of size 1276 bytes would be split into two buffers,
one of 1268 and other of 8 bytes considering RTP header size of
12 bytes. Putting audiobuffersplit between appsrc and rtpL16pay
can take care of this.
While buffer duration could still be used being able to specify
the size in bytes is helpful here.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1578>
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Adding vp9parse element to parse various stream information such as
resolution, profile, and so on. If upstream does not provide resolution and/or
profile, this would be useful for decodebin pipeline for autoplugging
suitable decoder element depending on template caps of each decoder element.
In addition, vp9parse element supports unpacking superframe into
single frame for decoders. The vp9 superframe is a frame which consists
of multiple frames (or superframe with one frame is allowed) followed by superframe
index block. Then unpacked each frame will be considered as normal frame
by decoder. The decision for unpacking will be done by downstream element's
"alignment" caps field, which can be "super-frame" or "frame".
If downstream specifies the "alignment" as "frame",
then vp9parse element will split an incoming superframe into single frames
and the superframe index (located at the end of the superframe) data
will be discarded by vp9parse element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1041>
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Similar to #GstCCExtractor:remove-caption-meta
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1554>
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We can only determine a correct placement for the CC line
with:
* height == 525 (standard NTSC, line 21 / 22)
* height == 486 (NTSC usable lines + 6 lines for VBI, line 1 / 2)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1554>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1256>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1506>
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MJPEG Tools may reencode pictures in a second pass to stick
closer to the target bitrate. This can result in slower than
real-time encoding for full HD content in certain situations,
as entire GOPs need reencoding when the reference picture is
reencoded.
See https://sourceforge.net/p/mjpeg/bugs/141/ for background
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1491>
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The PAL/NTSC widescreen modes were added after 1.16 but inserted before
the HD modes, which changed the integer values of the enums.
Move them to the very end instead to keep backwards compatibility.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1048
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1492>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1480>
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This property currently only supports a 'strict' that checks that
all the input streams have the exact same number of frames.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1424>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1424>
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Add va plugin
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1387>
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We don't want to expose all of the webrtcbin internals to the world.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1444>
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https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/753
https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/754
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1441>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1433>
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I thnk w cn spre the xtra lttrs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1397>
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The commit that added that was reverted. Need to remove this
from docs cache manually.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1422>
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