Handle when encoder doesn't support rate control, which is set as
VA_RC_NONE, and if the set rate control mode is not supported by the
GStreamer element, the element configuration fails.
Also it logs out max and target bitrate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3063>
The entrypoint is set when the encoder helper is constructed,
nonetheless it was also passed as parameter when opening. That's
buggy.
In order to simplify the code, the entrypoint at construction is
honored.
But gst_va_encoder_has_profile_and_entrypoint() now doesn't rely in
the internal list of profiles since it only contains those that
belongs to codec and entrypoint, thus it queries directly the VA
driver.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3063>
Need to put the actual profile in the output caps otherwise any
capsfilter after the encoder that was used to force the output
profile will fail, such as
fdkaacenc ! audio/mpeg,stream-format=adts,profile=he-aac-v1 ! ..
because we put profile=lc in there to match the profile signaled
in the ADTS header. This is expressed through the base-profile=lc
in the GStreamer caps though, the profile needs to carry the
'real' profile.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1785>
duplicate symbol '__invoke_on_main' in:
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/libgstvulkan-1.0.a(cocoa_gstvkwindow_cocoa.m.o)
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/libgstgl-1.0.a(cocoa_gstglwindow_cocoa.m.o)
ld: 1 duplicate symbol for architecture x86_64
clang: error: linker command failed with exit code 1 (use -v to see invocation)
Also make the same change in iOS for consistency.
Continuation of https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1132
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3242>
Add Windows Graphics Capture (WGC) API based screen capture mode.
The conditions where this mode is used:
* Explicitly requested by user (capture-api property)
* To capture specific window
* When DXGI desktop duplication API does not work on hybrid graphics systems
(e.g., multi-gpu laptop)
Full features of this implementation require Windows 11. And Windows 11
SDK is required to build this feature.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3144>
When the output alignment is smaller than the input alignment, for
example, When the output alignment is "FRAME" and the parse is likely
connecting to a decoder, the current PTS setting for AV1 frames inside
a TU is not very correct.
For example, a TU may begin with non-displayed frames and end with a
displayed frame. The current way will assign the PTS to the first
non-displayed frame, which is a decode-only frame and the PTS will be
discarded in the video decoder. While the last displayed frame has
invalid PTS, and so the video decoder needs to guess its PTS based on
the frame rate and previous frame's PTS. This is not a decent and
robust way. And more important, when the previous frames provide DTS,
the video decoder will also guess the PTS based on the previous frames'
DTS and trigger the warning like:
gstvideodecoder.c:3147:gst_video_decoder_prepare_finish_frame: \
<vavp9dec0> decreasing timestame
It sets the reordered_output and makes the decoder in free run mode.
We should correct the PTS for a TU, let the non-displayed frames have
no PTS while set the correct PTS to the displayed one. Also, when the
AV1 stream has multi spatial layers, there are more than one displayed
frames inside one TU with the same PTS.
Note: If the input alignment is not TU aligned, we can not know the
exact PTS of this TU, and so we just clear the PTS of the decode only
frame and leave others unchanged.
We also correct all the PTS if the output is OBU aligned. All their
PTS and DTS are set to the input buffer's PTS.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3182>
When the incoming data has big alignment than the output, we do not need to
call finish_frame() and exit the current handle_frame() for each splitted
frame. We can push them all at one shot with in one handle_frame(), whcih
may improve the performance and can help us to find the edge of TU.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3182>
Adding loopback capture mode for specified PID.
Note that this feature requires Windows 10 build 20348
(Windows 11/Windows Server 2022 or later),
and any process loopback related properties will not be exposed
if OS does not support it.
Example launch lines:
* wasapi2src loopback-mode=include-process-tree loopback-target-pid=<PID>
Captures audio generated by an application (specified by PID)
and its child process
* wasapi2src loopback-mode=exclude-process-tree loopback-target-pid=<PID>
Captures desktop audio excluding PID and its child process
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1278
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3195>
If there is an error while connecting, the streaming task will be stopped, and
is_running() will be false, causing a GST_FLOW_FLUSHING to be returned. Instead,
we perform the error check (!self->connection) first, to return an error if
that's what occured.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3189>
When the alignment is "FRAME" and the parse is likely connecting to
a decoder, the current PTS setting for VP9 frames inside a super
frame is not very correct.
For example, the super frame may begin with non-displayed frames and
end with a displayed frame. The current way will assign the PTS to
the first non-displayed frame, which is a decode-only frame and the
PTS will be discarded in the video decoder. While the last displayed
frame has invalid PTS, and so the video decoder needs to guess its
PTS based on the frame rate and previous frame's PTS. This is not a
decent and robust way. And more important, when the previous frames
provide DTS, the video decoder will also guess the PTS based on the
previous frames' DTS and trigger the warning like:
gstvideodecoder.c:3147:gst_video_decoder_prepare_finish_frame: \
<vavp9dec0> decreasing timestame
It sets the reordered_output and makes the decoder in free run mode.
We should correct the PTS for a super frame, let the non-displayed
frames have no PTS while set the correct PTS to the displayed one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3155>
Apparently we cannot start sending messages from another datachannel
before the previous message was completely sent. usrsctplib will
complain about being locked on another stream id and set
errno=EINVAL.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2454>
The order of the devices iterator from the SDK is undefined and can
randomly change.
Keep the device-number property for backwards compatibility and
simplicity but prefer the persistent-id property and also use it for the
device provider implementation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3078>
GstDXGIGetDebugInterface() is unused when targeting UWP. We directly
call DXGIGetDebugInterface1() in that case.
Fixes build failure:
../gst-libs/gst/d3d11/gstd3d11device.cpp(271): error C2440: '=': cannot convert from 'HRESULT (__cdecl *)(UINT,const IID &,void **)' to 'DXGIGetDebugInterface_t'
../gst-libs/gst/d3d11/gstd3d11device.cpp(271): note: This conversion requires a reinterpret_cast, a C-style cast or function-style cast
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3118>
According to W3C
specification (https://w3c.github.io/webrtc-pc/#datachannel-send) we
should return InvalidStateError exception when trying to send when the
channel is not open. In the world of C/glib/gstreamer we don't have
exceptions but have to rely on gboolean/GError instead. Introducing
these calls for a change in function signature of the action signals
used to send data on the datachannel. Changing the signature of the
existing "send-string" and "send-data" signals would mean an immediate
breaking change so instead we deprecate them. Furthermore, there is no
way to express GError** as an argument to an action signal in a way
that fits language bindings (pointer-to-pointer simply does not work)
and we have to use regular functions instead.
Therefore we introduce gst_webrtc_data_channel_send_data_full() and
gst_webrtc_data_channel_send_string_full() while deprecating the old
functions and corresponding signals.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1958>
Currently if the user is not able to access the devices under /dev/media*,
either due to no media devices present on the system or simply no permission
to access the device, v4l2codecs initialises with no features or debug messages.
Since calling `GST_DEBUG="v4l2*:7" gst-inspect-1.0 v4l2codecs` is a typical way
to diagnose why element(s) failed to enumerate, we should be more verbose here
when the user is not able to access any /dev/media* device. So print a simple
debug message in this case to aid debugging.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3088>
Since commit a79a756b79 we could change to ignore-pcr automatically at 500ms
into a live stream when no PCR is seen by then. However the stream counting in
program change detection was wrongly considering ignore-pcr programs to have a
separate PCR PID, even though we are actually ignoring the PCR PID completely,
resulting in an erroneous program switch getting triggered from the different
stream count. This in turn would send an EOS and switch out the pads for what
actually is still the same program, while we intended to simply apply a
workaround for broken encoders.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3060>
Fixes warning with meson 0.62:
gst-plugins-bad| subprojects/gst-plugins-bad/meson.build:546: WARNING:
Project targets '>= 0.62' but uses feature deprecated since '0.62.0':
pkgconfig.generate variable for builtin directories. They will be
automatically included when referenced
and more.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3086>
Starting with Meson 0.62, meson automatically populates the variables
list in the pkgconfig file if you reference builtin directories in the
pkgconfig file (whether via a custom pkgconfig variable or elsewhere).
We need this, because ${prefix}/libexec is a hard-coded value which is
incorrect on, for example, Debian.
Bump requirement to 0.62, and remove version compares that retained
support for older Meson versions.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1245
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3061>
The AV1 support multi spatial layers within one TU with different
resolutions, and only the highest spatial layer need to be output.
For example, there are two spatial layer, base level is 800x600
and higher level is 1920x1080. We need to decode both because the
higher level needs base layer as reference, but we only need to output
1920x1080 frames here.
The current manner always renegotiates the caps once we detect the
current picture resolution changes, so we renegotiate again and
again between different layers. That's a big waste and has very
low performance. We now only do the renegotiation for the highest
output layer. For other non output layers, we just keep a internal
buffer pool which is big enough to handle the surface allocation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2382>
As SPEC says, when multi spatial layer exists, we should only output
one frame with the highest spatial id from each TU. We now store the
highest spatial layer information in the base class in order to let
the sub class handle different layers easily.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2382>
doesn't align on 20 millisecond frame size.
The AMR-WB codec imposes a fixed 20 millisecond frame size. In its current
form, the `voamrwbenc` plugin deals with this limitation by discarding any
audio at the end of the stream that falls short of 20 milliseconds. This patch
keeps the audio data, and appends silence to the end to preserve frame size
alignment.
The patch also adds tests to check for the updated behavior. I noticed that
tests weren't being built, so I changed the build to allow for building the
tests when the `tests` and `voamrwbenc` options are set.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3027>
- Update the docker image we use, starting using the standard one adding
`gtk4-doc` as required by rust plugins
- Update the plugins_doc_caches as required, some more plugins are built
with the new image
- Install ninja from pip as the version from F31 is too old
- Avoid buildings all GSreamer plugins when building the doc as it takes
time and resources for no good reason
- Stop linking to `GInstanceInitFunc` as it is not present in latest GLib
documentation, leading to warnings in hotdoc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2954>
Handle d3d11 device context in set_context() method with
additional device compatibility check so that only NVIDIA GPU
associated d3d11 device can be configured in the element.
And clear old d3d11 device per set_info() for d3d11 device to be
updated as well.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3018>
