When the subclass attempts to finish without an explicit `out_buffer`,
we take a buffer from our adapter. We need to make this buffer writable
before copying the metadata.
This led to data races such as in the following pipeline, which randomly
messed up the buffer PTS:
gst-launch-1.0 -e audiotestsrc timestamp-offset=5555 num-buffers=100 \
! opusenc ! tee name=t ! queue ! opusparse ! fakesink silent=0 \
t. ! queue ! opusparse ! fakesink silent=0 -v | grep '0000, dur'
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5720>
Whenever that caps changes does not imply that a new segment will start.
Don't reset the last_ts if only the caps have changed. This fixes issues
if you have a stream without only first frame with TS=0, and have resolution
change happening. This was a regression introduced by !3059, which issue was
described in #1352. The reported issue is still fix after this change.
Fixes#1034
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5712>
When seek flush, gst v4l2 buffer pool flush is not atomic which will
lead double enqueue buffer (qbuf) issue, and v4l2 buffer pool qbuf is
also not atomic which will lead no free buffer found in the pool.
1. add lock for calculate enqueue number in streamon function
2. add lock for v4l2 capture end streamoff in pool flush function
3. lock the whole funciton of v4l2 buffer pool qbuf, then the buffer
pool index and qbuf operation are atomic
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5695>
When copying a buffer, for example with gst_buffer_make_writable(), the
new buffer might reference the same GstMemory as the src buffer,
making those memories not writable. If the src buffer gets disposed
first it should return to its buffer pool, but since some of its
memories are not writable it gets discarded and new buffer/memory gets
allocated.
Solves this by making the new buffer keep a reference to the src buffer,
that ensures that by the time the src buffer gets disposed no other
buffer are referencing its memories and it can thus return safely to its
pool.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5696>
gst_buffer_add_parent_buffer_meta() is used when a GstBuffer uses
GstMemory from another buffer that was allocated from a pool. In that
case we want to make sure the buffer returns to the pool when the memory
is writable again, otherwise a copy of the memory is created. That means
the child buffer must drop its ref to the memory first, then drop the
ref to parent buffer so it can return to the pool when it is the only
owner of the memory.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5696>
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 commit makes sure that pads are valid for linking
after the pads has been temporarily unlocked in the linking process.
Not doing this opens up for a race condition where
pads potentially can be linked twice.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5678>
Because we treat raw audio chunks/samples as keyframes, they were interfering
with seek time adjustment.
Became apparent when the accompanying video stream was I-frame only,
for example ProRes.
Since raw audio streams can be seeked freely, it's fine to just ignore them here,
giving priority to the real keyframes in the video stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5674>
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>
If text width ever reached 1px, for example after resizing the output window, the overlay would stop rendering
and never return again. The 1px condition itself does not seem to make much sense here anyway.
This was a chain of events: width reached 1, so the composition was set to NULL. Then, after resizing the output window,
push_frame() was called but would not attempt to renegotiate because composition is NULL. This caused the width/height
to never be updated again, as that only happens during negotiation, so the overlay was gone for good.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5623>
There is a race condition where transfer has not been submitted yet while the
request is cancelled which leads to the transfer state going back to
`DOWNLOAD_REQUEST_STATE_OPEN` and the user of the request to get signalled about
its completion (and the task actually happening after it was cancelled) leading
to assertions and misbehaviours.
To ensure that this race can't happen, we start differentiating between the
UNSENT and CANCELLED states as in the normal case, when entering `submit_request`
the state is UNSENT and at that point we need to know that it is not because
the request has been cancelled.
In practice this case lead to an assertion in
`gst_adaptive_demux2_stream_begin_download_uri` because in a previous call to
`gst_adaptive_demux2_stream_stop_default` we cancelled the previous request and
setup a new one while it had not been submitted yet and then got a `on_download_complete`
callback called from that previous cancelled request and then we tried to do
`download_request_set_uri` on a request that was still `in_use`, leading to
something like:
```
#0: 0x0000000186655ec8 g_assert (request->in_use == FALSE)assert.c:0
#1: 0x00000001127236b8 libgstadaptivedemux2.dylib`download_request_set_uri(request=0x000060000017cc00, uri="https://XXX/chunk-stream1-00002.webm", range_start=0, range_end=-1) at downloadrequest.c:361
#2: 0x000000011271cee8 libgstadaptivedemux2.dylib`gst_adaptive_demux2_stream_begin_download_uri(stream=0x00000001330f1800, uri="https://XXX/chunk-stream1-00002.webm", start=0, end=-1) at gstadaptivedemux-stream.c:1447
#3: 0x0000000112719898 libgstadaptivedemux2.dylib`gst_adaptive_demux2_stream_load_a_fragment [inlined] gst_adaptive_demux2_stream_download_fragment(stream=0x00000001330f1800) at gstadaptivedemux-stream.c:0
#4: 0x00000001127197f8 libgstadaptivedemux2.dylib`gst_adaptive_demux2_stream_load_a_fragment(stream=0x00000001330f1800) at gstadaptivedemux-stream.c:1969
#5: 0x000000011271c2a4 libgstadaptivedemux2.dylib`gst_adaptive_demux2_stream_next_download(stream=0x00000001330f1800) at gstadaptivedemux-stream.c:2112
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5611>
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>
In snapshot mode pngenc should output exactly one frame
and then return FLOW_EOS to upstream. If upstream sends
more input frames before shutting down, it should keep
returning FLOW_EOS but not output any more encoded frames.
After a flushing seek it should output frames again though.
Fixes#3069.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5564>
The propose and decide allocation vfuncs are called directly from
basetransform and need to use the locked accessor function for
retrieving a reliable reference to the GstGLContext (if available)
Fixes spurious crashes on shutdown during pad reconfiguration
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5518>
- 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>
