Commit graph

97 commits

Author SHA1 Message Date
Julien
e9f5d94c93 gs: add source and sink for Google Cloud Storage
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>
2021-03-18 22:32:48 +00:00
Mathieu Duponchelle
08442cc792 cccombiner: implement scheduling
Prior to that, cccombiner's behaviour was essentially that of
a funnel: it strictly looked at input timestamps to associate
together video and caption buffers.

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

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

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

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

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

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

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2076>
2021-03-17 22:00:25 +00:00
Vivia Nikolaidou
cb55d30b3c interlace: Specify interlace-modes in the sink pad template
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>
2021-03-08 21:01:50 +02:00
Tim-Philipp Müller
766bd655fc interlace: add more formats, esp 10-bit, 12-bit and 16-bit ones
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2054>
2021-03-03 18:34:26 +00:00
Víctor Manuel Jáquez Leal
b61b3d833d docs: plugins update VA elements
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2050>
2021-03-02 15:33:54 +00:00
Ilya Kreymer
92626535c7 webrtc ice: Add 'min/max-rtp-port' props for setting RTP port range
default min port == 0, max port == 65535 -- if min port == 0, uses existing random port selection (range ignored)
add 'gathering_started' flag to avoid changing ports after gathering has started
validity checks: min port <= max port enforced, error thrown otherwise
include tests to ensure port range is being utilized (by @hhardy)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/119>
2021-03-01 14:42:17 +00:00
Víctor Manuel Jáquez Leal
771645e445 vulkan: Fix elements long name.
Fix vkcoloconvert and vkviewconvert long names.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2034>
2021-02-24 20:15:52 +01:00
Seungha Yang
8794f4b713 d3d11: Documentation update
* 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>
2021-02-23 03:34:11 +09:00
Thibault Saunier
927bd289e5 openh264enc: Add support for main and high profiles
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>
2021-02-11 14:58:35 +00:00
He Junyan
44b7e9268c doc: Add the av1 parse element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1614>
2021-01-19 18:38:03 +00:00
Sebastian Dröge
0243afcb9d ccconverter: Add property to specify which sections to include in CDP packets
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>
2020-12-07 19:23:42 +02:00
Marc Leeman
102c60f82c rtpmanagerbad: allow setting caps on rtpsrc
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>
2020-12-04 14:51:38 +00:00
Edward Hervey
d137171f03 opencv: Expose retinex parameters
Makes the plugin a tad more useful :)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1845>
2020-12-03 17:04:07 +01:00
Thibault Saunier
530f694366 uritranscodebin: Add setup-source and element-setup signals
The same way as playbinX does it as it is often quite useful
2020-11-30 17:31:48 -03:00
Thibault Saunier
142e571c28 transcode: Port to encodebin2
This allows supporting muxing sinks like hlssink2 or splitmux
2020-11-30 17:31:48 -03:00
Thibault Saunier
f1cf5d0683 hlssink2: Mark as Muxer
The way it is usable by encodebin2. This is what splitmux does already.
2020-11-30 15:16:01 -03:00
Olivier Crête
03d710bd40 openh264dec: Accept constrained-high and progressive-high profiles
They're just subsets of the high profile.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1634>
2020-11-18 15:47:36 -05:00
Olivier Crête
e53da20938 nvdec: Accept progressive-high and contrained-high profiles
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>
2020-11-18 15:46:52 -05:00
Arun Raghavan
8aa6db2c8d bluez: a2dpsink: Add support for LDAC to a2dpsink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1621>
2020-11-11 22:19:33 +05:30
Arun Raghavan
ef3085c743 bluez: avdtpsink: Add support for LDAC to avdtpsink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1621>
2020-11-11 22:19:33 +05:30
Thibault Saunier
3a554f6d62 qroverlay: Generate documentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
2020-11-11 00:18:32 +00:00
Jason Pereira
cba368785b decklink: correct framerate 2KDCI 23.98
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1771>
2020-11-05 14:28:28 +00:00
Sebastian Dröge
56b2130300 decklink: Add a default profile id
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>
2020-10-30 16:23:31 +02:00
Thibault Saunier
b254c0d5fe transcodebin: Port to decodebin3
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1151>
2020-10-29 13:30:07 +00:00
Aaron Boxer
db13dc9d02 jpeg2000parse: support frame and stripe alignment in caps
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>
2020-10-27 08:26:23 +01:00
Vivia Nikolaidou
94e1623434 cameracalibrate: Improve gst-inspect documentation
Thanks to @kazz_naka on Twitter

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1691>
2020-10-13 17:21:59 +03:00
Seungha Yang
9279326d8a decklink: Update doc
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1665>
2020-10-08 20:05:03 +00:00
Sebastian Dröge
97e648a738 decklink: Correctly order the different dependent mode tables
One was forgotten in 309f6187fe.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1617>
2020-10-01 06:29:19 +00:00
Sanchayan Maity
248d2bb795 audiobuffersplit: Add support for specifying output buffer size
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>
2020-09-21 15:17:18 +00:00
Seungha Yang
2b152eae69 videoparsers: Add vp9parse element
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>
2020-09-10 14:56:52 +00:00
Mathieu Duponchelle
c58357fb66 line21enc: add remove-caption-meta property
Similar to #GstCCExtractor:remove-caption-meta

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1554>
2020-09-09 22:11:28 +02:00
Mathieu Duponchelle
c07e2a89ba line21enc: heavily constrain video height
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>
2020-09-09 19:38:58 +02:00
Nazar Mokrynskyi
8c37eea410 rtmp2: fix code style, update documentation cache
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1256>
2020-09-09 05:53:08 +00:00
Matthew Waters
2d31aba78d vulkan: docs annotation updates
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1506>
2020-08-15 02:55:30 +00:00
Mathieu Duponchelle
93a54093ec mpeg2enc: add disable-encode-retries property
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>
2020-08-06 17:13:03 +00:00
Sebastian Dröge
309f6187fe decklink: Re-order modes enum for backwards compatibility with 1.16
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>
2020-08-06 12:22:04 +03:00
Seungha Yang
c7da86665f docs: Update wasapi2 and mfvideosrc doc
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1480>
2020-07-31 21:50:22 +09:00
Thibault Saunier
7db147e9aa iqa: Add a 'mode' property
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>
2020-07-23 17:14:08 +00:00
Thibault Saunier
0349f032bf iqa: Implement child proxy
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1424>
2020-07-23 17:14:08 +00:00
Víctor Manuel Jáquez Leal
30a588f31e docs: update plugins doc cache
Add va plugin

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1387>
2020-07-21 16:15:47 +00:00
Matthew Waters
597c1b4ec6 webrtc: remove private properties/signals from the now public ice object
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>
2020-07-20 15:56:20 +10:00
Tim-Philipp Müller
99a0615592 docs: update for new pixel formats
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>
2020-07-15 12:43:20 +01:00
Nicolas Dufresne
2778fd7e31 doc: Updated cache file for the new pixel format
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1433>
2020-07-14 17:06:59 -04:00
Tim-Philipp Müller
395ecb3d2f avtp: rename tstamp-mode to timestamp-mode
I thnk w cn spre the xtra lttrs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1397>
2020-07-11 00:14:44 +01:00
Tim-Philipp Müller
e86a549b5d docs: fix up for errorignore convert-error signal removal
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>
2020-07-08 23:50:35 +01:00
Jan Schmidt
46cc64e09f mpegtsmux: Fix handling of MPEG-2 AAC
The audio/mpeg,mpegversion=2 caps in GStreamer refer to
MPEG-2 AAC (ISO 13818-7), not to the extended MP3 (ISO 13818-3),
which is audio/mpeg,mpegversion=1,mpegaudioversion=2/3

Fix the caps, and add handling for MPEG-2 AAC in both ADTS and raw
form, adding ADTS headers for the latter.
2020-07-08 12:24:13 +00:00
Philippe Normand
8900f2d2f9 wpe: Update plugin's doc cache
This was forgotten in !1392.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1402>
2020-07-02 17:07:46 +00:00
Seungha Yang
c72ccded6c docs: Update plugin cache for Windows plugins 2020-07-02 17:21:33 +02:00
Seungha Yang
76793ffabc nvcodec: Update for documentation
* Add Since marks
* Make use of GST_PARAM_CONDITIONALLY_AVAILABLE flag
* Add documentation template caps
2020-07-02 17:21:24 +02:00
Tim-Philipp Müller
c229127b43 avtp: documentation fixes
Unclear why hotdoc wants 'gstavtp' as the plugin name here,
that's just wrong.

Add since marker and mark private subclasses as plugin API
so hotdoc knows they belong to the plugin and aren't external.

Fix GstAvtpAafTstampMode get_type() function.
2020-07-01 18:41:25 +01:00