When the size of V4L2 capture or output is changes with VIDIOC_S_FMT,
the device is only required to update the compisition window to fit
inside the new frame size. This can result in captured data only being
updated on a portion of the frame after a resize.
Update the composition window to the default value determined by the
V4L2 device driver whenever the format is changed to make sure that
all image data is composed to its full size.
Fixes#765
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1806>
As specified formally in RFC8851
Each rid description is placed in its own caps field in the structure.
This is very similar to the already existing extmap-$id sdp<->caps
transformations that already exists.
The mapping is as follows:
a=rid:0 direction ';'-separated params
where direction is either 'send' or 'recv'
gets put into a caps structure like so:
rid-0=(string)<"direction","param1","param2",etc>
If there are no rid parameters then the caps structure is generated to
only contain the direction as a single string like:
rid-0=(string)direction
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1760>
Not having this field is equivalent with it being 1/1 so consider
it like that. The generic caps functions are not aware of these
semantics and would consider the caps different, causing a negotiation
failure when caps are changing from caps with to caps without or the
other way around.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1826>
Not having these fields is equivalent with them being mono/0 so consider
them like that. The generic caps functions are not aware of these
semantics and would consider the caps different, causing a negotiation
failure when caps are changing from caps with to caps without or the
other way around.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1826>
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>
```
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>
```
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
While this is slightly more expensive (~48% slower per random number) it
does not cause any measurable difference when running through a complete
audio conversion pipeline.
On the other hand its random numbers are of much higher quality and on
spectrograms for 32 bit to 24 bit conversion the difference is clearly
visible.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1729>
The instant-rate value in the TrickMode enum is a
flag, but the other values are not. Move instant-rate
to the end of the enum and give it a value large enough
for it to be used without modifying the trick-mode
setting.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1788>
Update x264enc long-name to be more than just the name. Then the
description also was updated to be longer than the long-name, and
similar to the plugin description.
Finally, as I am here, H264 was replaced by H.264 and x264 is only a
plugin (not plugins).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1771>
They can't be used in any useful way. The type of every GstMemory is
always GST_TYPE_MEMORY and the subtyping relationship has to be
implemented on top of that via the associated allocator and mem_type
string.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1764>
Scenario:
- Source 1 requesting and waiting a clock id
- Source 2 requesting and waiting on a clock id
- Test attempting to crank both sources in the same GstHarness
gst_test_clock_crank() originally dropped locks between the retrieving
of the next clock id and advancing to the next clock id. This would
mean that both sources would race each other attempting to complete
their clock waits. Sometimes the operations would be performed in the
correct order, other times they would not and a FALSE return value would
be produced.
This would lead to an assertion in gst_harness_push_from_src() expecting
that all clock cranks to succeed.
Fix by ensuring that the clock wait produced is dealt with before
processing the next by not dropping the relevant locks after retrieving
the next clock id.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1299>
This is a minimal unit test the show that the stride extrapolation can work
with all pixel format we support. This minimal verify that the extrapolation
match the stride we set into GstVideoInfo with 320x240 for all the pixel
format we support. The tiles formats are skipped, since their stride is
set as two 16bit integers, and we also skip over palette planes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1567>
As this element is single threaded, we only need to stop the objects to
allow changing the format again. Fixes assertion notably on shutdown and
on some other situation where the format may be set twice without
actually activating the element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1567>
