Commit graph

1133 commits

Author SHA1 Message Date
Seungha Yang
c4ac657364 qsv: Add H.264 decoder
Initial decoder implementation with baseclass

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1786>
2022-03-01 21:24:07 +00:00
Nirbheek Chauhan
4de365b31c webrtc_sendrecv.py: Sync element props with C version
Also add indentation to make it easier to read

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1821>
2022-03-01 16:33:28 +00:00
Nirbheek Chauhan
5ca5a83e75 webrtc_sendrecv.py: Ensure that gst-python overrides are installed
Otherwise fetching of the offer will fail with a cryptic error:

```
Traceback (most recent call last):
  File "/../gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py", line 56, in on_offer_created
    offer = reply['offer']
TypeError: 'Structure' object is not subscriptable
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1821>
2022-03-01 16:33:28 +00:00
Nirbheek Chauhan
e9a02a7380 webrtc_sendrecv.py: Don't try to set state on a None pipe
```
ERROR peer '5762' not found
Traceback (most recent call last):
  File "/../gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py", line 190, in <module>
    res = loop.run_until_complete(c.loop())
  File "/usr/lib64/python3.10/asyncio/base_events.py", line 641, in run_until_complete
    return future.result()
  File "/../gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py", line 155, in loop
    self.close_pipeline()
  File "/../gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py", line 142, in close_pipeline
    self.pipe.set_state(Gst.State.NULL)
AttributeError: 'NoneType' object has no attribute 'set_state'
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1821>
2022-03-01 16:33:28 +00:00
Nirbheek Chauhan
78f8505b9a webrtc_sendrecv.py: Fix SSLError when connecting to websocket server
```
  File "/../gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py", line 189, in <module>
    loop.run_until_complete(c.connect())
  File "/usr/lib64/python3.10/asyncio/base_events.py", line 641, in run_until_complete
    return future.result()
  File "/../gstreamer/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py", line 40, in connect
    self.conn = await websockets.connect(self.server, ssl=sslctx)
  File "/home/nirbheek/.local/lib/python3.10/site-packages/websockets/legacy/client.py", line 650, in __await_impl_timeout__
    return await asyncio.wait_for(self.__await_impl__(), self.open_timeout)
  File "/usr/lib64/python3.10/asyncio/tasks.py", line 445, in wait_for
    return fut.result()
  File "/home/nirbheek/.local/lib/python3.10/site-packages/websockets/legacy/client.py", line 654, in __await_impl__
    transport, protocol = await self._create_connection()
  File "/usr/lib64/python3.10/asyncio/base_events.py", line 1080, in create_connection
    transport, protocol = await self._create_connection_transport(
  File "/usr/lib64/python3.10/asyncio/base_events.py", line 1110, in _create_connection_transport
    await waiter
  File "/usr/lib64/python3.10/asyncio/sslproto.py", line 631, in _on_handshake_complete
    raise handshake_exc
  File "/usr/lib64/python3.10/asyncio/sslproto.py", line 676, in _process_write_backlog
    ssldata = self._sslpipe.do_handshake(
  File "/usr/lib64/python3.10/asyncio/sslproto.py", line 116, in do_handshake
    self._sslobj = self._context.wrap_bio(
  File "/usr/lib64/python3.10/ssl.py", line 526, in wrap_bio
    return self.sslobject_class._create(
  File "/usr/lib64/python3.10/ssl.py", line 865, in _create
    sslobj = context._wrap_bio(
ssl.SSLError: Cannot create a client socket with a PROTOCOL_TLS_SERVER context (_ssl.c:801)
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1821>
2022-03-01 16:33:28 +00:00
Nirbheek Chauhan
e453e43e5a webrtc_sendrecv.py: Fix deprecation warning with Python 3.10
asyncio.get_event_loop() will not implicitly create a new event loop
in a future version of Python, so we need to do that explicitly.

```
webrtc_sendrecv.py:188: DeprecationWarning: There is no current event loop
  loop = asyncio.get_event_loop()
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1821>
2022-03-01 16:33:28 +00:00
Nirbheek Chauhan
4c2fd7f104 webrtc_sendrecv.py: Fix styling errors
These are now enforced by the pre-commit python style hook.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1821>
2022-03-01 16:33:28 +00:00
Nirbheek Chauhan
d6799b069a webrtc: Update Makefile for building webrtc-sendrecv
This now needs the RTP library.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1821>
2022-03-01 16:33:28 +00:00
Jan Schmidt
cad9f8fb36 uridecodebin3: Remove dead variables
Leftover junk from original port

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1820>
2022-03-01 14:21:43 +00:00
Jan Schmidt
cebf769725 matroska-mux: If a stream has a TITLE tag, use it for the name.
If a title tag is pushed to a pad, store it as the Track name.
This means that players will use it as the human readable
description of the track, instead of something generic like 'Video'
or 'Subtitle'

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1798>
2022-03-01 13:17:40 +00:00
Jan Schmidt
7efdc9c7f5 matroskademux: Don't parse Tracks element twice
If the tracks element was parsed from the SeekEntry, don't
parse it a second time and recreate tracks, as this
loses any tags that were read using the seek table.

If a genuinely new Tracks element is found, do read that
as it is needed for MSE support.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1798>
2022-03-01 13:17:40 +00:00
Sebastian Fricke
7063aa1471 docs: Fix typos in documentation
In building-from-source-using-meson.md:
s/implicitely/implicitly/

In README.md:
s/uncompatible/incompatible/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1743>
2022-03-01 11:33:10 +00:00
Sebastian Fricke
5a421886b4 docs: Extend documentation for the GStreamer development environment
Add more extensive documentation for the development environment.
Document how the tool works, how to use it and common use cases.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1743>
2022-03-01 11:33:10 +00:00
Sebastian Fricke
0b6bbce012 Remove the uninstalled term
Remove the symbolic link `gst-uninstalled` which points to `gst-env`.
The `uninstalled` is the old name and the project should stick to a
single name for the procedure.
Remove the term from all the files, exceptions are variables from
dependencies like `uninstalled_variables` from pkgconfig and
`meson-uninstalled`.
Adjust mentions of the script in the documentation and README.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1743>
2022-03-01 11:33:10 +00:00
Sebastian Fricke
c999d2c3a9 Maintain build instructions at a single location
Do not maintain similar build instructions within each gst-plugins-*
subproject and the subproject/gstreamer subproject. Use the build
instructions from the mono-repository and link to them via hyperlink.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1743>
2022-03-01 11:33:10 +00:00
Sebastian Fricke
7702072a6f Add documentation for GST_VALIDATE_APPS_DIR
Add documentation for the environment variable, explaining what it is
used for and the default search locations.

Fixes: 4d569b51ed add a way to specify an application directory.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1743>
2022-03-01 11:33:10 +00:00
Sebastian Fricke
278f4a6418 Improve environment variable documentation
At GST_VALIDATE_FILE:
s/will be outputed/are output/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1743>
2022-03-01 11:33:10 +00:00
Sebastian Fricke
0a65c87e8c Add documentation for GST_VALIDATE_PLUGIN_PATH
Add documentation for the environment variable, explaining what it is
used for and the default search locations.

Fixes: 83d6978f80 Implement fault_injection as a Gs(tValidate)Plugin
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1743>
2022-03-01 11:33:10 +00:00
Víctor Manuel Jáquez Leal
c769a089ea docs: Add vah264enc metadata.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1051>
2022-03-01 10:53:50 +01:00
He Junyan
1f2f135cdb va: enable the H264 encoder.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1051>
2022-03-01 10:53:50 +01:00
He Junyan
f17357f759 va: Add H264 encoder.
This a new VA-API implementation of a H264 encoder.

It can control the GOP and parameter settings, while the MV searching,
VCL and the rate control algorithm are implemented by VA drivers and HW.
It supports most of the common usage options in H264, but still lacks
of look ahead, field, B frame weighted prediction, etc.

Co-authored-by: Victor Jaquez <vjaquez@igalia.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1051>
2022-03-01 10:53:50 +01:00
He Junyan
736a0ac9b0 va: Add a common encoder object.
As the counterpart of the va decoder, this class handles all the
common logic for the encoding routine and miscellaneous queries about
encoding.

Co-authored-by: Victor Jaquez <vjaquez@igalia.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1051>
2022-03-01 10:53:50 +01:00
He Junyan
fb644e84fa va: Add vacompat.h to wrap glib functions.
The g_queue_clear_full() and g_array_copy() functions in the glib
may not be available for the current glib version check, so we add
helper functions to wrap it.
This should be deleted after the glib version bumps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1051>
2022-03-01 10:53:50 +01:00
He Junyan
57d50a941f va: Add the profile string name into the profile_map.
We also add a helper function of gst_va_profile_from_name to get
the VA profile value by its profile string name.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1051>
2022-03-01 10:53:50 +01:00
He Junyan
e0b6c6678b va: Change the H264 profile string order in the profile_map.
The first one should be the one that matches the VA profile's name
most precisely.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1051>
2022-03-01 10:53:50 +01:00
He Junyan
83408cfdc8 va: caps: Expose gst_va_create_coded_caps as helper function.
And allow free indentation for array declaration.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1051>
2022-03-01 10:53:50 +01:00
He Junyan
1defc9ce6b test: Add test cases for the H264 bitwriter.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1051>
2022-03-01 10:53:49 +01:00
He Junyan
d68d3b9a0d codecparsers: bitwriter: Add the common bit writer functions for H264.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1051>
2022-03-01 10:53:49 +01:00
He Junyan
ca914f4ac2 codecparsers: nalutils: Add nal_writer_reset_and_get_data help function.
We not only want to create a NAL gstmemory, but also need to create and
get the raw data of a NAL writer for the later usage.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1051>
2022-03-01 10:53:49 +01:00
Guillaume Desmottes
1f02f24828 gs: look for google_cloud_cpp_storage.pc
storage_client.pc was legacy and has been removed:
df6fa3611c (diff-bc35ad7c2fe631fd5578a06092412dba81c7ddd27bb25df7e17bb13771799afcL743)

No need to keep looking for storage_client.pc as a fallback as 1.25.0,
our minimum version, already ships google_cloud_cpp_storage.pc

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1815>
2022-03-01 08:10:39 +00:00
Vivia Nikolaidou
b699feefee yadif.asm: Fix improper usage of LOAD macro
LOAD macro relies in m7 being zero for interleaving purposes. Using LOAD
on the m7 register makes it interleave with its new content instead of
with 0.

The effect of this bug was bobbing on some static lines that appeared
over fast-moving content.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1816>
2022-03-01 07:22:10 +00:00
Vivia Nikolaidou
d499342f0d yadif.asm: Typo fixes in comments
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1816>
2022-03-01 07:22:10 +00:00
Vivia Nikolaidou
087ca88213 yadif: Fix bug in C implementation of CHECK
It was different compared to the corresponding part in both ffmpeg and
the asm implementation. Fixing this makes videotestsrc pattern=spokes
not jump at all when not using the asm optimisations.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1816>
2022-03-01 07:22:10 +00:00
jinsl00000
ef4cc9e637 ipcpipeline: fix crash and error on windows with SOCKET or _pipe()
The fd was in different meanings on windows:
POSIX read and write use the fd as a file descriptor.
The gst_poll use the fd as a WSASocket.

This patch use WSASocket as default on windows. This is a temporary measure, because IPC has many different implement. There may be a better way in the future.

See #1044

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1791>
2022-03-01 06:31:51 +00:00
Ming Qian
24eb35f113 v4l2videodec : enable resolution change
The dynamic resolution changes when
the sequence starts when the decoder detects a coded frame with one or
more of the following parameters different from those previously
established (and reflected by corresponding queries):
1.coded resolution (OUTPUT width and height),
2.visible resolution (selection rectangles),
3.the minimum number of buffers needed for decoding,
4.bit-depth of the bitstream has been changed.

Although gstreamer parser has parsed the stream resolution.
but there are some case that we need to handle resolution change event.
1. bit-depth is different from the negotiated format.
2. the capture buffer count can meet the demand
3. there are some hardware limitations that the decoded resolution may
be larger than the display size. For example, the stream size is
1920x1080, but some vpu may decode it to 1920x1088.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1381>
2022-03-01 00:00:50 +00:00
Ming Qian
fe56af607b v4l2videodec : refactor the setup process of capture
v4l2videodec do some refactoring so that it can support
dynamic resolution change event.

1.wrap the setup process of capture as a function,
as decoder need setup the capture again when
dynamic resolution change event is received.

2.move the function "remove_padding"

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1381>
2022-03-01 00:00:50 +00:00
Célestin Marot
cabff7a20f video-info: encoded format can have RGB color-matrix (Fixes #1435)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1435>
2022-02-28 16:05:43 +00:00
Wu Tong
c60ac7a04b MSDK: Add _context_query() and avoid compile error on Windows
To avoid compile error on Windows, macro definitions are added to suppress va
variables. In the meantime, add function _context_query() to query
context on Windows.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1693>
2022-02-28 12:54:23 +00:00
Sebastian Dröge
5849570fe8 buffer: Clarify that the MARKER flag maps to the corresponding RTP header flag
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1776>
2022-02-28 10:13:11 +00:00
Sebastian Dröge
b0afaffc5d rtp: In payloaders map the RTP marker flag to the corresponding buffer flag
This allows downstream of a payloader to know the RTP header's marker
flag without first having to map the buffer and parse the RTP header.

Especially inside RTP header extension implementations this can be
useful to decide which packet corresponds to e.g. the last packet of a
video frame.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1776>
2022-02-28 10:13:11 +00:00
Joseph Donofry
630dbea94c osxaudiosrc: Support a device as both input and output
osxaudiodeviceprovider now probes devices more than once to determine
if the device can function as both an input AND and output device.

Previously, if the device provider detected that a device had any output
capabilities, it was treated solely as an Audio/Sink.  This causes issues
that have both input and output capabilities (for example, USB interfaces
for professional audio have both input and output channels).  Such devices
are now listed as both an Audio/Sink as well as an Audio/Source.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1385>
2022-02-28 06:51:21 +00:00
Sebastian Dröge
2fc91717cb registry: Fix multi-line #warning compiler warning
subprojects/gstreamer/gst/gstregistry.c:1593: unexpected character `"'

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1802>
2022-02-28 06:01:24 +00:00
Sebastian Dröge
6c3642da49 video-format-info: Use correct parameter name in gst_video_format_info_extrapolate_stride() docs
../subprojects/gst-plugins-base/gst-libs/gst/video/video-format.c:7570: Warning: GstVideo: gst_video_format_info_extrapolate_stride: unknown parameter 'info' in documentation comment, should be 'finfo'

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1803>
2022-02-27 13:19:49 +02:00
Sanchayan Maity
cc3419daf6 rtp: ldac: Set frame count information in payload
The RTP payload seems to be required as it carries the frame count
information. Also, gst_rtp_base_payload_allocate_output_buffer had
the second argument incorrect.

Strangely some devices like Shanling MP4 and Sony XM3 would still
work without this while some like the Sony XM4 do not.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1797>
2022-02-26 21:09:57 +05:30
Sanchayan Maity
7c9a315578 ldac: Set eqmid in caps
We set the eqmid in caps to be usable downstream by rtpldacpay for
knowing the frame count.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1797>
2022-02-26 17:05:22 +05:30
Vivia Nikolaidou
7cebd5b359 tsmux: Skip empty buffers
They can be created e.g. by aggregator when there is a gap. Such buffers
should not be muxed at all.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1611>
2022-02-25 21:29:43 +00:00
Xavier Claessens
3d8372cc50 devenv: Add some missing GStreamer specific env variables
This should make "meson devenv" closer to what "gst-env.py" sets.

- GST_VALIDATE_SCENARIOS_PATH
- GST_VALIDATE_APPS_DIR
- GST_OMX_CONFIG_DIR
- GST_ENCODING_TARGET_PATH
- GST_PRESET_PATH
- GST_PLUGIN_SCANNER
- GST_PTP_HELPER
- _GI_OVERRIDES_PATH

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1768>
2022-02-25 20:35:26 +00:00
Sebastian Dröge
3941eb7dbd audioconvert: Add dithering-threshold property
By default, no dithering is applied if the target bit depth is above 20
bits. This new property allows to apply dithering nonetheless in these
cases.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1730>
2022-02-25 19:32:28 +00:00
Jan Alexander Steffens (heftig)
e10bd02e1d fdkaacdec: Support arbitrary channel configs
Try to match the config to GStreamer positions. If something doesn't
fit, fall back to a set of unpositioned channels.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1561>
2022-02-25 18:20:52 +00:00
Jan Alexander Steffens (heftig)
d4b4ffc944 fdkaacdec: Use predefined channel layouts
This limits the decoder to the layouts predefined for the encoder
(including the MPEG standard layouts) but greatly simplifies the
implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1561>
2022-02-25 18:20:52 +00:00