For some cameras `gst_jpeg_parse_app0()` fails on a invalid segment.
While this is likely a driver or firmware bug that should be addressed
accordingly, it's not fatal and likely does not deserve a bus message on
every frame, flooding journals.
Turn down the volume of the warnings by turning them into object
warnings. If we conclude that in some cases we'd still want bus
warnings, they can be done more fine-grained in the
`gst_jpeg_parse_appX()` functions.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6490>
GstD3D12Device objetct's internal resources are singletons per adapter
already though, the object itself is not a singleton.
Due to the singleton design (unlike other APIs such as d3d11),
d3d12 device context sharing is not a strict requirement
for zero-copy, but handles context ones to make things less noisy.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6513>
Adding RGBA, RGBx, BGRA, BGRx, VUYA and RGB10A2_LE format support for performance.
However, these formats are not still recommended if upstream can support
native YUV formats (e.g., NV12, P010) since NVENC does not expose
conversion related optiones. Note that VUYA format is 4:4:4 YUV format
already but NVENC runtime will convert it to 4:2:0 format internally
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6417>
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, format, 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/6441>
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, format, 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/6441>
The last frame which has the smallest diff should be consider as
the first choice rather than the golden frame. Especially when only
one reference available, this way can improve the BD rate about 5
percentage.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6379>
Some extreme case such as "videotestsrc pattern=1" can generate pure
white noise videoes, for which encoder may generate too big output
for current coded buffer size. We now consider the qindex and bitrate
to avoid that.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6483>
It might happen that the key event arrives when the d3d11videosink
is stopping. In case of GstD3D11WindowWin32 it can raise a
navigation event even when the sink is already freed, because the
window object's refcount may reach 0 in the window thread. In
other words sometimes the GstD3D11WindowWin32 lives few ms more
then the GstD3D11VideoSink, because it's freed asynchronously.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6476>
In the case of multi-channels transcoding, a context with child
sesseion can be parent for others, so we need to check if the
msdkcontext has any child session in the list to avoid session
leaks. Otherwise, we will see the failure of closing a parent
session because one of its child's child session not released.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6259>
Calling gst_pad_peer_query_caps() without a filter can give us EMPTY caps, whereas all the code below
assumes that's not the case. Replacing query+intersect with a filtered query ensures we always get a subset
of the template caps back.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6429>
There was a potential busy loop occuring because when we were taking
data from the internal ccbuffer, we were not resetting which field had
written data. This would mean that the next time data was retrieved
from ccbuffer, it was always from field 0 and never from field 1.
This only affects usage of cc_buffer_take_separated() which is only used
by cdp->raw cea608.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6423>
In an early non-linked scenario, this was causing a ton of criticals about the queue array,
because the output callback would still fire for leftover frames that were still being processed by VT
at the time the output loop stopped. This makes sure they're flushed correctly as well.
Also renames gst_vtdec_loop to gst_vtdec_output_loop for consistency with related functions.
wip
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6397>
Sometimes a call to negotiate (and thus drain) can happen from the output loop
(via finish_frame()), which will tell VT to output all internal frames, but that won't succeed
if we happen to decide to wait for the queue to empty (because the loop is waiting for draining to finish and
will not make space in the queue!). This commit adds an override for the queue size limit if we're draining/flushing.
This bug could happen for any formats, but was especially obvious for ProRes, which has dpb_size of 0.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6397>
Because ID3D12Device objects are singletons per adapter,
GstD3D12Device was following the API design, that is, keep track
of global GstD3D12Device objects and reuses it.
That means ID3D12Device object can be released at the time
when GstD3D12Device is destroyed.
But exetrnal APIs such as NVENC does not seem to be happy
with the released ID3D12Device, that could be a driver bug though.
Let's hold already opened ID3D12Device permanently without releasing
it for now.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6395>
`on_error()` can be called with a NULL details structure, so in that situation
the `gst_structure_copy()` would raise a critical warning. Create an empty
structure instead of attempting to copy a NULL one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6385>
In order to simplify caps negotiations for clients and, notably, be more
compatible with va* decoders.
Crucially this allows clients to know ahead of time whether buffers will
actually be DMABufs.
Similar to GstVaBaseDec we only announce system memory caps if the peer
has ANY caps. Further more, and again like va decoders, we fail in
`decide_allocation()` if DMA_DRM caps are used without VideoMeta.
Apart from buggy peers this can happen e.g. when a peer with ANY caps
is used in combination with caps filters.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5890>
Most importantly rely on video info helpers instead of manual parsing
of caps, which will allow us to use additional helpers in the future.
While on it, tighen the check for supported formats - failing that
indicates a bug in caps negotiation - and make some style changes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5890>
This ensures we don't create filter caps that are not supported by the
individual codec implementations, as well as that the resulting caps
have the required fields so they can be turned into a GstVideoFormat.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5890>
When this error gets caught the GstD3D11Device object raises the new
"device-removed" signal. This allows to handle the error from outside:
stop the playback, re-create the player, replace the catched GstContext by
the new one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6193>
Do not chain up to parent's GstBufferPool::start() which will do
preallocation. We don't want it to be preallocated
since there are various cases where negotiated downstream buffer pool is
not used at all (e.g., zero-copy decoding, IPC elements).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6326>
This fixes a crash in `gst_va_h264_enc_class_init` and `gst_va_h265_enc_class_init`
(and probably also in gst_va_av1_enc_class_init) when calling
`g_object_class_install_properties (object_class, n_props, properties);`
When rate_control_type is 0, the following code is executed in :
```
} else {
n_props--;
properties[PROP_RATE_CONTROL] = NULL;
}
```
n_props has initially a value of N_PROPERTIES but PROP_RATE_CONTROL
is not the last element in the array, so it's making
g_object_class_install_properties fail to iterate over the
properties array.
This applies the same fix to gstvah264enc.c, gstvah265enc.c and
gstvaav1enc.c.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6319>
osxaudio has a few helper methods potentially useful in atdec (or future atenc), like GStreamer -> CoreAudio
channel mapping. Doesn't make sense to duplicate them in applemedia, and atdec is the only audio-oriented
element there anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6223>
Cea608 (valid) padding removal is available on the input side of ccconverter
or configurable on cccombiner. cccombiner can now configure whether
valid or invalid cea608 padding is used and for valid padding, how long
after valid non-padding to keep sending valid padding.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6300>
The transport stream only returned the CAPS for the first matching PT entry
from the `ptmap`. Other SSRC with the same PT where not included. For a stream
which bundled multiple audio streams for instance, only the first SSRC was
knowed to the SSRC demux and downstream elements.
This commit adds all the `ssrc-` attributes from the matching PT entries.
The RTP jitter buffer can now find the CNAME corresponding its SSRC even if it
was not the first to be registered for a particular PT.
The RTP PT demux removes `ssrc-*` attributes cooresponding to other SSRCs
before pushing SSRC specific CAPS to downstream elements.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6119>
In the situation where playback starts from a keyframe before
the target playback segment, then the first buffers will be
outside the configured segment and gst_segment_to_stream_time()
will return GST_CLOCK_TIME_NONE unconditionally.
If drop-out-of-segment is false, the RTP buffers will not be
dropped, but will be sent witout ONVIF extension timestamps
and given GST_CLOCK_TIME_NONE timestamps on the receiver.
Instead, use gst_segment_to_stream_time_full() to extrapolate
stream time outside the segment so that such buffers still
get assigned their correct timestamps on the receiver.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6248>
Don't accidentally include the stuffing byte (if present)
into the bottom field size. It should only be included in the
total segment length.
Fixes problems with FFmpeg not rendering the subtitles
with a stuffing byte, giving a "Invalid object location!" error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6250>
Provide a clock from the source that is a monotonic system clock with
the rate corrected based on the measured and ideal capture rate of the
frames.
If this clock is selected as pipeline clock, then provide perfect
timestamps to downstream.
Otherwise, if the pipeline clock is the monotonic system clock, use the
internal clock for converting back to the monotonic system clock.
Otherwise, use the monotonic system clock time calculated in the above
case and convert that to the pipeline clock.
In all cases this will give a smoother time than the previous code,
which simply took the difference between the driver provided capture
time and the current real-time clock time, and applied that to the
current pipeline clock time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6208>
Otherwise there's a small window between querying the state and doing
the transfer in which a frame could be dropped, and we would then output
the frame right after the dropped one as if it was the dropped frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6208>
In low delay B mode, the P frame is converted as B frame with forward
references. For example, One P frame may refers to P-1, P-2 and P-3 in
list0 and refers to P-3, P-2 and P-1 in list1.
So the num in list0 and list1 does not reflect the forward_num and
backward_num. The vaapi does not provide ref num for forward or backward
so far. In this case, we just consider the backward_num to be 1 conservatively.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6249>
In b_pyramid mode, B frames can be ref and prevPicOrderCntLsb can
be the B frame POC which is smaller than the P frame. This can cause
POC diff bigger than MaxPicOrderCntLsb/2 and generate wrong POC value.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6249>
Gets being released memory back to queue even if allocator is flushing
in order to count the number of outstanding memory objects.
Also, clear queue if there's no outstanding memory object and
allocator is flushing
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6240>
Syncrhonizing buffer commits to the streaming thread can lead to
dropped frames when frame callbacks are not processed before the
next frame is ready for rendering. Depending on the drift between
the wayland compositor and buffer source timings, this can lead to
periods of significant frame drop, especially when the media frame
rate is close to the display frame rate.
Cache buffers in the streaming thread and peform commits on the
display thread to eliminate the buffer commit racing.
The implementation is the same for both waylandsink and gtkwaylandsink,
so move it to the common wayland library under gst-lib.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6133>
Add synchonized versions of wl_display_sync() and wl_callback_destroy()
that will ensure that to callbacks can be managed in a thread safe way
on the display queue even when they are dispatched from a separate
thread.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6133>
Unprepare method posts WM_GST_D3D11_DESTROY_INTERNAL_WINDOW
command to the window queue, and from that moment considers
internal_hwnd to be released, and so it sets it to null.
The problem is that it's possible that right at that moment
the window thread might be already processing some other
command, or just another command might be already in the queue.
On practice we met a crash when WM_PAINT got processed in between
(unprepare already finished and WM_GST_D3D11_DESTROY_INTERNAL_WINDOW
was not handled yet)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6187>
When the conversion is only caps feature from memory:VAMemory to system memory,
it's possible to optimize by doing a pseudo pass-through since the va-backed
buffers are the same for system memory buffers.
This change will also mitigates #2940
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6174>
If the allocation query received from downstream doesn't handle GstVideoMeta but
it requests memory:DMABuf caps feature, it's incomplete, so we rather reject the
negotiation.
Both in base decoder, base transform and compositor.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6155>
This is a simplification of the venerable
gst_va_base_dec_get_preferred_format_and_caps_features() function, which
predates since gstreamer-vaapi. It's used to select the format and the
capsfeature to use when setting the output state. It was complex and hard to
follow. This refactor simplifies a lot the algorithm.
The first thing to remove _downstream_has_video_meta() since, most of the time
it will be called before the caps negotiation, and allocation queries make sense
only after caps negotiation. It might work during renegotiation but, in that
case, caps feature change is uncommon. Better a simple and common approach.
Also, for performance, instead of dealing with caps features as strings, GQuarks
are used.
The refactor works like this:
1. If peer pad returns any caps, the returned caps feature is system memory and
looks for a proper format in the allowed caps.
2. The allowed caps are traversed at most 3 times: one per each valid caps
feature. First VAMemory, later DMABuf, and last system memory. The first to
match in allowed caps is picked, and the first format matching with the
chroma is picked too.
Notice that, right now, using playbin videoconvert never return any.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6154>
Some subtitle "decoders" had a wrong category of "Parser", which `parsebin`
relies on to identify elements which do not *decode* streams but *parse* them.
This would cause such subtitle decoders to be plugged in within parsebin,
preventing the original stream to be properly used by (more efficient)
downstream decoders or subtitle renderers.
Fixes#1757
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6153>
This inherits from the same rule as gst_buffer_add_meta
```
gst-mpegtspesmetadatameta.h:98: Warning: GstMpegts:
gst_buffer_add_mpegts_pes_metadata_meta: return value: Invalid non-constant
return of bare structure or union; register as boxed type or (skip)
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6146>
This reverts questionable commit 009bc15f33
which looks completely wrong.
The GstWasapi2RingBuffer:buffer_size variable is used to
calculate available buffer size we can write
(i.e., available size = buffer_size - padding_size).
But the commit makes the size to be exactly same as buffer period.
Then, it can confuse this element as if the endpoint buffer is full on
I/O event callback (if padding size is equal to buffer period)
but it's not true.
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2870
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6132>
- Add the missing field parameter and put the output parameter at the
end.
- Use a switch to verify valid values instead of hard-to-follow range
checks.
- Don't consider bad values a programming error, just a regular failure.
- Set all data fields at the end so we can pass a pointer to an
uninitialized structure without GCC complaining.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5450>
The global semaphore was never closed/unlinked, causing permission
denied issue if the device is later used by another user. Properly
removing the semaphore when stopping the pipeline would still leave it
open in case of a crash.
With a GStreamer specific name, it was also not preventing other apps to access
the device concurrently.
Finally, if the system has multiple cards, the lock should be per card
and not global (to be confirmed).
Fixes: #3283.
Sponsored-by: Netflix Inc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6117>
MaxDpbSize specified in A.4.2 tells upper bound of decoded picture
buffer size but does not tell actual required size.
Use max_dec_pic_buffering value as a dpb size. Some backends
such as DXVA and NVDEC might require pre-allocated DPB buffer
and unnecessary large DPB size will result in waste of GPU memory.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6101>
In rtpbin we already systematically check for all property names
except latency, correct that.
In webrtcbin we need to check before trying to use the do-retransmission
property.
This is useful for the case where an element like identity gets passed
to rtpbin's request-jitterbuffer property, when the application wants
to use webrtcbin in an SFU situation, with no reordering and no added
latency
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6112>
According to recommendation from MS, IDXGIOutputDuplication::ReleaseFrame()
needs to be called just before IDXGIOutputDuplication::AcquireNextFrame()
for performance reasons, so that driver can accumulate dirty rects
and update texture at once. But it seems to cause choppy output.
Do release acquired frame immediately once processing done,
like d3d11 implementation does.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6092>
* Bump the rank of the musepack v7/v8 FFmpeg demuxers to SECONDARY
* Bump the rank of the musepack v7/v8 FFmpeg audio decoders to SECONDARY
* Demote the rank of the musepackdec element to MARGINAL
This is for two reasons:
* The musepack library is no longer maintained, whereas the FFmpeg
implementation can/will receive fixes
* The `musepackdec` implementation was a all-in-one "parsing and decoding" blob
which doesn't play nicely with decodebin3 and others
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3033
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6074>
Sends a gap event if nothing to output for a given input buffer.
For example, an input buffer might not contain any caption data
for downstream requested field, then we need to inform downstream
of the case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6073>
WebKit commit b12e7ed2ad3a ("[WPE] Upstream the new WPE platform API
https://bugs.webkit.org/show_bug.cgi?id=265286")[1] added a `WPEView` typedef
which clashes with our `WPEView` class.
Rename the `WPEView` class to `GstWPEThreadedView` to avoid the collision.
Also prefix the `WPEContextThread` class with `Gst` and rename the
source files to reflect the new class name and use lowercase while at it
for consistency
[1] b12e7ed2ad
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6065>
Previously, the path lock was held even while issuing caps queries to
other elements. This can lead to deadlocks in more complex pipelines.
Avoid this by reworking gst_switch_bin_get_allowed_caps() to acquire
references to switchbin paths and then releasing the path lock.
Subsequent operations in that function then act on the acquired
references, thus eliminating the need for holding the path lock for
the entirety of that function.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4632>
The caps query specifies _all_ caps that the element can handle, not just
caps from the current path element. If for example a switchbin has two
paths, with one having an element that handles video/x-h264, and another
path whose element handles video/x-raw, and the second path is the
current path, then the existing code would report only video/x-raw as
supported. Fix this by report all allowed caps, even if there is a
current path defined.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4632>
The rationale is that a passthrough path (= one with no element) behaves
as if the switchbin's sink- and srcpad were one. In particular, internal
caps queries (needed for computing the allowed caps) then go to the peers
instead to path elements. Rework gst_switch_bin_get_allowed_caps () for
a clear handling of NULL path elements and for proper dataflow passthrough
and caps & accept-caps query handling.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4632>
The drop probe was present in early switchbin versions to implement paths
that drop dataflow. However, this feature turned out to be too problematic
and thus was removed. Some bits remained though. This commit removes those
bits and clarifies that in the current switchbin version, a NULL path
element instead means passthrough.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4632>
If the current segment has a configured stop point, detect
when when pad timestamps proceed past that point and mark
them as EOS. Otherwise, tsdemux continues streaming
the whole input downstream (unless something downstream detects
and returns EOS for us)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6023>
