Because of the asynchronous resolving of mDNS ICE candidates it is
possible that GstWebRTCICE outlives webrtcbin. This in turn prolongs
the lifetime of the GstWebRTCNiceStream objects via refs in
nice_stream_map. Thus the GstWebRTCICETransport objects held in
GstWebRTCNiceStream may be invalid at the time they are accessed by
the _on_candidate_gathering_done() callback since GstWebRTCNiceStream
doesn't take a reference to them. Doing so would create a circular
reference, so instead this commit introduces weak references to the
transport objects and then we can check if the objects are valid before
accessing them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3502>
And even that vaav1dec doesn't use vabasedec negotiate vmethod, it should align
with the new scheme of using base's width & height for surface size and
output_info structure for downstream display size negotiation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3480>
This vmethod can be used by decoders with the same VA decoder reopen logic:
same profile, chroma, width and height.
Also a new public method called gst_va_base_dec_set_output_state() with the
common GStreamer code for setting the output state, which is always called by
the negotiate vmethod.
In order to do this refactoring, new variables in vabasedec have to be populated
by the decoders:
* width and height define the resolution set in VA decoder. In the case of H264
would be de coded_width and codec_height, or max_width and max_height in AV1.
* output_info is the downstream video info used for negotiation in
gst_va_base_dec_set_output_state().
* input_state, from codec parent class shall be also held by vabasedec
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3480>
There could be multi-GPU setups where the non-first has more
entrypoints than the first one, and the elements names are not
homogeneous, leading to pipeline building error.
This patch add the render node in the elements names when they belong
to the non-first device.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3491>
To fix the warning on Alderlake
vafilter gstvafilter.c:534:gst_va_filter_ensure_filters:<vafilter0>
vaQueryVideoProcFiltersCaps: list argument exceeds maximum number
Increase the number of caps to 16 as vadumpcaps does.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3473>
Windows supports various IPC methods but that's completely
different form that of *nix from implementation point of view.
So, instead of adding shared memory functionality to existing
shm plugin, new WIN32 shared memory source/sink elements
are implemented in this commit.
Each videosink (server) and videosrc (client) pair will communicate
using WIN32 named pipe and thus user should configure unique/proper
pipe name to them (e.g., \\.\pipe\MyPipeName).
Once connection is established, videosink will create named shared memory
object per frame and client will be able to consume the object
(i.e., memory mapped file handle) without additional copy operation.
Note that implementations under "protocol" directory are almost
pure C/C++ with WIN32 APIs except for a few defines and debug functions.
So, applications can take only the protocol part so that the application
can send/receive shared-memory object from/to the other end
even if it's not an actual GStreamer element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3441>
There was a drm/drm_mode.h included added recently, drm/ is usually
referencing the linux kernel header, but we only requires the libdrm
headers to be installed. On top of this, including drm_mode.h is never
needed as its already included by drm.h.
Fixes#1596
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3452>
The legacy emulation in DRM/KMS drivers badly interact with GStreamer and
may cause the framerate to be halved. With this property, users can disable
vsync (which is handled internally by the emulation) in order to regain the
full framerate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3303>
The original BUNDLE support commit placed a queue after the
rtpfunnel that combines streams, but I don't see a good reason for
it. It has default settings, so if network output is slow might
accidentally store up to 1 second of pending data, increasing
latency.
Remove it in favour of doing any necessary buffering before
webrtcbin. If it turns out that there is a reason for it to
exist, the limits should probably be configurable and small.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3437>
In current tile representation, only tiles with power of two
width and height in bytes are supported. This limitation
prevents adding more complex tiles formats.
In this patch, we deprecate tile_ws and tile_hs from GstVideoFormatInfo and
replace if with an array of GstVideoTileInfo. Each plane tiles are then
described with their pixels width/height, line stride and total size.
The helper gst_video_format_info_get_tile_sizes() that depends on the
deprecated API is also being removed. This can simply be removed as it wasn't
in any stable release yet.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3424>
This change allow output caps to be updated even though we stay in
streaming state. This is needed so that any upstream updated to fields
like framerate, hdr data, etc. can result in a downstream caps event
being pushed.
Previously, any of these changes was being ignored and the downstream
caps would not reflect it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3328>
In theory, input caps can be updated anytime at non-keyframe or
sequence boundary, such as HDR10 metadata, framerate, aspect-ratio
or so. Those information update might not trigger ::new_sequence()
or subclass may ignore the changes.
By this commit, input state change will be tracked by baseclass
and subclass will be able to know the non-decoding-essential
update by checking the codec specific picture struct
on ::output_picture()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3328>
This reverts commit fcad4cc646.
This was wrong is so many ways.
* The memcmp was badly used (it should use == 0 to check the data is identical,
and not != 0)
* There was no boundary checks on the present stream section_data when passing
it to memcmp.
* The return value should have been TRUE (i.e. we have done all checks, none of
them failed, therefore the section has been seen before)
* stream->section_data would *always* be NULL if the section had already been
processed
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1559
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3421>
Rewriting GstCudaConverter object, since the old implementation was not
well organized and it's hard to add new features.
Moreover, the conversion operations were not very optimized.
Major change of this implementation:
* Remove redundant intermediate conversion operations such as
any RGB -> ARGB(64) conversion or any YUV -> Y444 (or 16bits Y444).
That's not required most of cases. The only required case is
converting 24bits (such as RGB/BGR) packed format to 32bits format
because CUDA texture object does not support sampling 24bits format
* Use normalized sample fetching (i.e., [0, 1] range float value)
and also normalized coordinates system for CUDA texture.
It's consistent with the other graphics APIs such as Direct3D
and OpenGL, that makes sampling operations much easier.
* Support a kind of viewport and adopt math for colorspace conversion
from GstD3D11 implementation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3389>
GstCudaConverter object can do colorspace conversion and scale at once.
Adding new element "cudaconvertscale" to do that, this can
save unnecessary GPU operation if colorspace conversion and
rescale is required for given input stream format.
Most of codes are taken from d3d11convert element
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3389>
Instead of returning a "const gchar" or a "gchar" that should not be freed, always
return a duplicated string as those functions were used together with g_strdup anyway.
This is needed to prepare support for returning modified strings in next commit.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1147>
If we don't receive any data from usrsctp, then there will be no src pad
for the stream id and the stream reset will fail to remove the relevant
src pad. Workaround by first attempting to add the relevant src pad, then
almost immediately removing it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3381>
Replace video_copy with memcpy to fix the issue when the sizes of the
src frame and dst frame don't match. Moreover, for Windows, you have to
do the copy first and call gst_msdk_import_to_msdk_surface later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3231>
Replace video_copy with memcpy to fix the issue when the sizes of the
src frame and dst frame don't match. Moreover, for Windows, you have to
do the copy first and call gst_msdk_import_to_msdk_surface later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3231>
Currently MSDK context does not support d3d11va. Now introduce d3d11va
device to MSDK context, making it able to create msdk session with d3d11
device and to easily share with upstream and donwstream.
Add environment variable to enable user to choose GPU device in multi-GPU
environment. This variable is only valid when there's no context
returned by upstream or downstream. Otherwise it will use the device
that created by upstream or downstream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3231>
Add support for more formats so as to run the libvpx high bit depth test suite.
This means the files under CONFIG_VP9_HIGHBITDEPTH
This also allows running the yuv444p 8bit file in the regular 8 bit vp9 suite.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3356>
If an input is malformed (only produces cea608 field 1 cc_data) then
when in passthrough we would effectively be dropping every second cea608
on output as we would not store any unused cea608 data.
Fix by having all code paths go through the framerate conversion code
which will store and retrieve any relevant data across buffers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
... otherwise PAR can be wrongly signalled during the negotiation
Fixing below pipeline when desktop resolution is not 640x480
gst-launch-1.0.exe \
d3d11screencapturesrc ! videoscale !
video/x-raw,width=640,height=480,pixel-aspect-ratio=1/1 ! d3d11videosink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3360>
1. Removes the verification if the internal encoder is not opened
yet to allow the property setting.
2. And toggles on the base class' reconf flag for each property
variable that can be modified at run time.
3. Mark those modifiable properties as mutable while playing.
Currently the run-time modifiable properties are:
qpi, qpp, qpb, bitrate, target percentage, target usage and rate control
Other properties can be enabled too, but they need testing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2466>
Adds an internal function reset() which drains the internal queues and
calls the reconfig() vmethod.
This reset() method is called inconditionally at set_format() and in
handle_frame() if the instance's reconf flag is enabled.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2466>
If parameters remain similar enough to avoid either encoder reopening
or downstream renegotiation, avoid it.
This is going to be useful for dynamic parameters setting.
To check if the stream parameters changed, so the internal encoder has
to be closed and opened again, are required two steps:
1. If input caps, profile, chroma or rate control mode have changed.
2. If any of the calculated variables and element properties have
changed.
Later on, only if the output caps also changed, the pipeline
is renegotiated.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2466>
This method will return the caps configured in the reconstruct buffer
pool, and its maxium number of buffers to allocate.
The caps are needed later to know if the internal encoder has to be
reopened if the stream properties change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2466>
This adds a new boolean property `auto-reconnect`, defaulting to `true`.
Setting it to `false` makes the elements (in caller mode) immediately
report an error to the application instead of trying to reconnect.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3326>
Adding DirecShow video capture filter mode, in addition
to existing MediaFoundation and WinRT(UWP) mode, to support
DirectShow only filters (not KS driver compatible)
such as custom virtual camera filters.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3350>
- Make the srt_epoll_wait loops more uniform.
- Error only via GError when possible; let the element send the error
message. Avoids a second error message.
- Return 0 when cancelled. Avoids an error message from the element.
- Don't send an error message from send_headers when we're a server
sink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3156>
These null checkes are slightly misleading when double-checking
mutability for external language interop. None of the functions in
these files allow the variable at hand to become `NULL` under normal
operation, because they are checked at initialization and never (allowed
to be) reassigned to `NULL`.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1615>
This is an additional quality parameter. In the default configuration this
quality switch is deactivated because it would cause a workload increase
which might be significant. If workload is not an issue in the application
it can be recommended to activate this feature.
A flush request is done when set_format is called to empty internal bit
buffer maintained by fdk-aac. When this happens, during the explicit
call to handle_buffer, decodeFrame does not return a AAC_DEC_OK. This
gets reported as a decoding error while no decoding error in fact took
place. Since this can be confusing, just return a GST_FLOW_OK and log
that an explicit flush was requested.
In fact, all the h264 bit writer have byte aligned output except
the slice header. So we change the API from bit size in unit to
byte size, which is easy to use. For slice header, we add a extra
"trail_bits_num" to return the unaligned bits number.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3193>
We use va pool as msdkvpp's bufferpool, which means both va memory
and dma memory will be allocated by va pool. Considering drm modifier
stuff is not ready, we use va memory with higher priortiry than
dma memory when deciding vpp caps.
Besides, this patch removes the specified "interlace-mode" in vpp caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3253>
The gap handling was in place, but there was no event handler to trigger it.
Implement the alpha sink event handler for the gaps. This fixes handling of
valid streams which may not refresh the alpha frames for every video frames.
It will also allow a clean error if the stream was missing the initial
alpha frame, at least until we find a better way to handle these
invalid frames.
Related to #1518
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3264>
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>
