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>
