Take the case into account when user filters have been set before the
source gets updated.
Note that the further linking of the filters, if present, happens below
in the `gst_camera_bin_check_and_replace_filter()` calls.
The audio filter is still affected by the same issue but left out for
now.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5682>
Even if IDXGIOutput6 says current display colorspace is HDR,
captured texture via IDXGIOutputDuplication::AcquireNextFrame()
is converted frame by OS unless we use IDXGIOutput5::DuplicateOutput1()
with DXGI_FORMAT_R16G16B16A16_FLOAT format, in order for captured
frame to be scRGB color space. Then application should perform
tonemap operation based on reported display white level, color primaries, etc.
Since we don't have any tonemapping implementation, ignores colorimetry
reported by IDXGIOutput6.
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3128
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5679>
This is how it was documented and how it worked before the port to GstPlay.
Without this, applications expecting signals to be emitted directly
without anything running the main context will simply not receive any
signals.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5673>
The code seems to validate that the media-level fingerprint matches
the fingerprint of the previous media or of the whole session. There
is no such requirement in any RFC I found. The session-session one
is just meant to act as a fallback when there is no media-level
fingerprint.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5663>
The src caps of the libde265 is now fixed to I420, and so if the
stream is other format, such as 4:4:4 or 10 bits format, the pipeline
will crash because the dowstream element accesses the video buffer as
I420 format.
We now restrain the input caps to "main" profile, which only contains
4:2:0 8 bits stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5596>
When decoding stream using hardware V4L2 decoder element, in any of the
currently supported formats, the decoding will fail once frame number
1000000 is reached. The reported error clearly indicates a wrap-around
occured, instead of receiving decoded frame 1000000, frame 0 is received
from the hardware V4L2 decoder driver.
The problem is actually not in the driver itself, but rather in gstreamer,
which uses `struct v4l2_buffer` member `.timestamp` in a special way. The
timestamp of buffers with encoded data added to the SINK (input) queue of
the driver is copied by the driver into matching buffers with decoded data
added to the SOURCE (output) queue of the driver. In fact, the timestamp
is not a timestamp at all, but rather in this special case, only part of
it is used as an incrementing frame counter.
The `.timestamp` is of type `struct timeval`, which is defined in
`sys/time.h` [1]. Only the `tv_usec` member of this structure is used
for the incrementing frame counter. However, suseconds_t tv_usec [2]
may be limited to range [-1, 1000000]:
"
[XSI] The type suseconds_t shall be a signed integer type capable of
storing values at least in the range [-1, 1000000].
"
Therefore, once frame 1000000 is reached, a rollover occurs and decoding
fails.
Fix this by using both `struct timeval` members, `.tv_sec` and `.tv_usec`
with matching modular arithmetic, this way the failure would occur again
just short of 2^84 frames, which should be plenty.
[1] https://pubs.opengroup.org/onlinepubs/9699919799/basedefs/sys_time.h.html
[2] https://pubs.opengroup.org/onlinepubs/9699919799/basedefs/sys_types.h.html
A test case using stateless hardware h264 decoder, the WARN/ERROR output
in gstreamer log indicates a failure occurred. With this change, that
error no longer occurs and the WARN/ERROR are not present:
```
pc$ gst-launch-1.0 videotestsrc num-buffers=1001001 pattern=6 ! \
video/x-raw,width=16,height=16,format=I420 ! \
x264enc ! filesink location=/tmp/test.h264
dut$ GST_DEBUG="*:3" gst-launch-1.0 filesrc location=/tmp/test.h264 ! \
h264parse ! v4l2slh264dec ! fakesink
...
0:03:51.393677606 12111 0x370df400 WARN \
v4l2codecs-decoder gstv4l2decoder.c:1157:gst_v4l2_request_set_done:<v4l2decoder2> \
Requested frame 1000000, but driver returned frame 0.
0:03:51.394140597 12111 0x370df400 WARN \
v4l2codecs-decoder gstv4l2decoder.c:1157:gst_v4l2_request_set_done:<v4l2decoder2> \
Requested frame 1000001, but driver returned frame 1.
0:03:51.394425216 12111 0x370df400 WARN \
v4l2codecs-decoder gstv4l2decoder.c:1157:gst_v4l2_request_set_done:<v4l2decoder2> \
Requested frame 1000002, but driver returned frame 2.
0:03:51.394665211 12111 0x370df400 WARN \
v4l2codecs-decoder gstv4l2decoder.c:1157:gst_v4l2_request_set_done:<v4l2decoder2> \
Requested frame 1000003, but driver returned frame 3.
0:03:51.394785833 12111 0x370df400 WARN \
v4l2codecs-h264dec gstv4l2codech264dec.c:1059:gst_v4l2_codec_h264_dec_output_picture:<v4l2slh264dec0> \
error: Failed to decode frame 1000000
ERROR: from element /GstPipeline:pipeline0/v4l2slh264dec:v4l2slh264dec0: Failed to decode frame 1000000
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5603>
This can happen with the dummy "noopenh264" library that the freedesktop
flatpak runtime ships, and Fedora is planning on shipping as well. In
both cases the dummy implementation gets replaced with the actual
openh264 library that's downloaded directly from Cisco, but just to be
on safe side, this patch makes it careful to check the return values to
avoid crashing if the underlying library hasn't been swapped out yet.
The patch is taken from freedesktop-sdk and was originally written by
Valentin David <valentin.david@codethink.co.uk>.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5586>
ISimpleAudioVolume controls volume of corresponding audio session
and there would be only single input/output audio session
in case of share-mode, which means that it controls audio volume of the
process. Instead, use IAudioStreamVolume interface which controls
volume of the stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5579>
Other Windows applications allow window switching even when
an application window is in fullscreen mode. Also fixing
regression introduced in 15248d8b84
which makes restored window is always located at topmost
since we do not call SetWindowPos() anymore when restoring
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5578>
Ignore alpha component of source (mouse cursor texture)
when blending alpha channel, otherwise the background area of source
(which has zeros) will be written to render target. Then it will result
in black rectangle if output texture is converted to premultiplied alpha
texture
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5577>
- Don't try to make the parameters match `GHFunc`. Use a dedicated
callback for `g_hash_table_foreach`.
- Don't try to be clever with buffer memories. We're allocating a full
packet anyway, might as well memcpy and save on a lot of complexity.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5516>
The v4l2codecs H.265 decoder uses the
GstH265SliceHdr::entry_point_offset_minus1 array so make sure that it is not
freed before decoding the frame.
Before this patch, some H.265 input would segfault in
gst_v4l2_codec_h265_dec_fill_slice_params() when executing the line:
guint32 entry_point_offset = slice_hdr->entry_point_offset_minus1[i] + 1;
Make sure that the array is not freed before using it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5503>
While adding arbitrary tile support, a round up operation was badly
converter. This caused the Y component of the stride to be 0. This
eventually lead to a crash in glupoad preceded by the following
assertion.
gst_gl_buffer_allocation_params_new: assertion 'alloc_size > 0' failed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5463>
The interaudiosrc might take buffers of different sizes from the audio adapter,
so keeping metas consistency would be an issue. So the sink now strips the audio
metas away and the src adds them back (for non-interleaved layouts only) when
taking buffers from the adapter.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5416>
Issue is that when amc was producing a codec-data buffer, a
GstVideoCodecFrame was being popped off the internal queue. This meant
that the codec-data was being associated with the first input frame and
the second (first encoded buffer) output buffer with the second input
frame. At the end (assuming one input produces one output which seems
to hold in my testing and how the encoder is currently implemented)
there would be an input frame missing and would be pushed without any
timing information. This would lead to e.g. muxers rejecting the buffer
without PTS and failing to mux.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5335>
Even if the segmentation feature value is not updated,
the parsed "segmentation_update_map" and "segmentation_temporal_update"
values should not be cleared as it's referenced during lower
level bitstream parsing. Also, don't use assert() in parser
unless it's clearly impossible condition.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5336>
Setting the surface source rectangle has been omitted so far. As a side effect
surface created with padded width/height are being scaled down. Fix this using
the viewporter source rectangle configuration. This can later be enhanced
to support crop meta.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5298>
Since d3d11convert and its variant elements does not enable basetransform's
passthrough, passthrough allocation query needs to be handled
manually in order to respect downstream element's min/max buffer
requirement.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5297>
It's only malformed data in APP when its length is less than 6 chars,
because it should have at least an id string. Otherwise, if the id string
is not handled, no warning is raised, only a debug message noticing it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5053>
Fix compiler warnings about not using the return value when
freeing the GString segment with g_string_free(.., FALSE):
ignoring return value of ‘g_string_free_and_steal’ declared with attribute ‘warn_unused_result’
which we get with newer GLib versions. These were all harmless.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5015>
There's no reason to release GstMemory manually at all.
If we do release GstMemory, corresponding GstBuffer will be
discarded by GstBufferPool baseclass because the size is changed
to zero.
Actual cause of heavy CPU usage in case of fixed-size pool
(i.e., decoder output buffer pool) and if we remove GstMemory from
GstBuffer is that GstBufferPool baseclass is doing busy wait in acquire_buffer()
for some reason. That needs to be investigated though, discarding
and re-alloc every GstBuffer is not ideal already.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4943>
The framerate should only be replaced (and corrected for alternating field)
when it is parsed from the bitstream. Otherwise, the upstream framerate
from caps should be trusted and assumed correct.
Related to gst-plugins-bad!2020
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4352>
If two senders use the same multicast IP and port then new_session_pad()
may try to add a srcpad to the same stream twice.
stream->srcpad is updated but gst_element_add_pad() fails the second
time. As a result stream->srcpad points to a deleted object and
access in gst_sdp_demux_stream_free() fails with a segfault.
Just ignore the second pad. Nothing useful can be done with it anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4873>
