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>
This is the maximum amount supported by aacenc. 8-channel output fully works.
16-channel also encodes fine, but codec-utils isn't able to parse its channel config,
so output level will not be shown in caps. For that to work, GASpecificConfig parsing
needs to be implemented. It's not a critical issue and can be worked on at a later date.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6359>
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>
Adds the `atenc` element capable of encoding AAC-LC audio, using the AudioToolbox framework.
It's able to encode up to 7.1 channel configurations.
Comes with basic knobs for rate control (bitrate for CBR, quality for VBR).
Support for more profiles (LD, HE-AAC) should be simple, but is not included here because of bugs
with parsing of the AudioSpecificConfig.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6254>
None of the GL allocators actually offer a generic alloc() implementation. As a
side effect, they cannot be offered as they don't work with generic video
buffer pool.
Our specialized buffer pool can be dropped by tee or alphacombine as sharing the
same buffer pool over two branch is not supported by the pool API.
Fixes#3372
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6327>
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>
On systems using UsrMerge (like openSUSE or Fedora), /lib64 is
a symlink to /usr/lib64. So dladdr is returning the path to
the gstreamer library in /lib64 in priv_gst_get_relocated_libgstreamer.
Later gst_plugin_loader_spawn tries to build the path to the
gst-plugin-scanner helper from /lib64 and ends up trying to use
/lib64/../libexec/gstreamer-1.0/gst-plugin-scanner which doesn't exist.
By canonicalizing the path with a call to realpath, gst-plugin-scanner
is found correctly under
/usr/lib64/../libexec/gstreamer-1.0/gst-plugin-scanner
Similar change applied to gstreamer/libs/gst/net/gstptpclock.c
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6322>
This matches autoplug in other places such as decodebin, otherwise we
will pick "randomly" based on the order in which plugins are
registered, which is mostly dependent on the order in which readdir()
returns items.
So let's make it predictable.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6227>
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>
Memory from gst_adapter_map() could live shorter than the GstMemory that the GstBuffer wraps around it, which in lucky
cases 'just' caused a re-use of the same memory for multiple (potentially still in use!) input buffers, but could easily
end up pointing to an already-freed memory.
Manifested when an AudioToolbox encoder kept getting silence inserted in seemingly random circumstances, turned out
to be the memory being re-used by GStreamer at the same time that the AT API was processing it...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6320>
This will mimic the playbin2 behaviour, which sets the "next" entry to be
NULL.
The biggest impact this has is that when going back to READY the current play
entry will be discarded (instead of being kept around for when you go back to
PAUSED/PLAYING).
Fixes#3371
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6324>
Since commit 4d86f994, when setting an RTSP media both shared and
reusable, streaming cannot be restarted after the first time all the
clients disconnect. That happens because the sockets (unlike
addresses) of GstRTSPStream are not cleared in
gst_rtsp_stream_leave_bin, and on restart sockets and addresses are
not allocated in gst_rtsp_stream_allocate_udp_sockets, and then the
check in create_sender_part fails. Fix this by clearing sockets in
gst_rtsp_stream_leave_bin.
Fixesgstreamer/gst-rtsp-server#113
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6325>
find_slot_for_stream_id() will return a slot which has the request stream-id as
active_stream *or* pending_stream (i.e. the slot on which that stream is
currently being outputted or will be outputted).
When figuring out which slot to use (if any) we want to consider stream-id
which *will* appear on a given slot which isn't outputting anything yet the same
way as if we didn't find a slot yet.
Fixes races when doing intensive state changes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6270>
With the new copy_opaque system, the corresponding frame is stored in the
picture opaque ref.
This also handles the case where the "regular" opaque might be empty in the
case of "DECODE_ONLY" frames, since it that field is set in `get_buffer2()`
which might not be called for those frames
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6301>
When we're doing a state change from PLAYING to NULL, first we invoke
gst_rtspsrc_loop_send_cmd_and_wait (..., CMD_CLOSE, ...) during
PAUSED_TO_READY which will schedule a TEARDOWN to happen async on the
task thread.
The task thread will call gst_rtspsrc_close(), which will send the
TEARDOWN and once it's complete, it will call gst_rtspsrc_cleanup()
without taking any locks, which frees src->streams.
At the same time however, the state change in the app thread will
progress further and in READY_TO_NULL it will call gst_rtspsrc_stop()
which calls gst_rtspsrc_close() a second time, which accesses
src->streams (without a lock again), which leads to simultaneous
access of src->streams, and a segfault.
So the state change and the cleanup are racing, but they almost always
complete sequentially. Either the cleanup sets src->streams to NULL or
_stop() completes first. Very rarely, _stop() can start while
src->streams is being freed in a for loop. That causes the segfault.
This is unlocked access is unfixable with more locking, it just leads
to deadlocks. This pattern has been observed in rtspsrc a lot: state
changes and cleanup in the element are unfixably racy, and that
foundational issue is being addressed separately via a rewrite.
The bandage fix here is to prevent gst_rtspsrc_stop() from accessing
src->streams after it has already been freed by setting src->state to
INVALID.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6302>
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>
And also re-timestamp them with the current buffer's PTS.
Not doing so keeps the timestamps of event packets as
GST_CLOCK_TIME_NONE or the timestamp of the previous buffer, both of
which are bogus.
Making sure that (especially) the first packet has a valid timestamp
allows putting e.g. the NTP timestamp RTP header extension on it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5173>
Parse the speed and scale in the server's response
*before* the range, so that the range start/stop
are swapped (or not swapped) correctly based
on the server's actual chosen values. Otherwise,
the old rate from the segment is used - what the
last seek asked for, but not necessarily what
the server chooses.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6248>
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>
Cocoa version of glwindow only checks the preferred size upon window creation. glimagesink sets the size right before
calling gst_gl_window_show(), which might be way after the window is created in some cases. If the size was set too
late, glimagesink on macOS would remain 320x240 unless manually resized.
This change makes sure to resize the existing window when _show() is called.
Curiously, this has always been an issue, but went from manifesting every once in a while to being almost completely
broken once old event loop workarounds were removed and gst_macos_main() was introduced.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6185>
Don't use g_return_val_if_fail() to catch the
open-ended segment or empty segment cases in
gst_segment_to_running_time_full()
g_return_val_if_fail() is for programmer errors,
and can be compiled out with a flag.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6219>
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>
Making it possible to properly handle compositors that have those
properties as doubles and handle antialiasing.
Internally we were handling those values as doubles in framepositioner,
so expose new properties so user can set values as doubles also.
This changes the GESFramePositionMeta API but we are still on time for 1.24
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6241>
The `G_DECLARE_FINAL_TYPE` macro does not need to be terminated with a
semicolon and the extra semicolon breaks building e.g. libcamera with
clang because `-Wextra-semi` is used which produces the following
error in conjunction with `-Werror`:
```
gstreamer-1.0/gst/allocators/gstdrmdumb.h:61:43: error: extra ';' outside
of a function is incompatible with C++98 [-Werror,-Wc++98-compat-extra-semi]
61 | GST, DRM_DUMB_ALLOCATOR, GstAllocator);
| ^
1 error generated.
```
Fix this by removing the extra semicolon
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6239>
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>
In the case where the queue shrinks due to a property change and the queue
becomes full, we would set the waiting_del flag, which would prevent posting the
100% buffering message on the bus. Since the pipeline is not aware of the new
buffering value, in the common case where the pipeline is paused during
buffering, it would never resume.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6165>
Remove the percent_changed check to determine whether a buffering message should
be posted. The check on the last posted buffering value is sufficient, and the
removal doesn't introduce additional complexity.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6165>
If input height and parsed one are identical, do not consider it as interlaced
Fixing below pipeline:
gst-launch-1.0 videotestsrc ! video/x-raw,format=I420,width=640,height=10 \
! jpegenc ! jpegparse ! jpegdec ! videoconvert ! autovideosink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6181>
When dealing with demuxers which aren't streams-aware, we need to handle the
old-school "stream replacement" dance from `parsebin` and hide that in such a
way that output pads are re-used (if compatible).
By analyzing the collection posted by parsebin, we can:
* Identify whether some output slots are no longer used (because the stream they
currently handle is not present in the collection)
* Decide if some upcoming streams could re-use the existing slot
This supports both buffering and non-buffering modes.
Fixes#1651
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6201>
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>
When switching urisourcebin, ensure that we first unlink *all* pads from
decodebin3 before linking them again.
This is to ensure that decodebin3 completely knows that all previous pads are no
longer needed and can prepare itself to being re-used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6179>
The value is stored as an 8 bit integer, with 0 meaning that there is
not data for this extension. That means that the maximum length is 255
bytes and not 256 bytes.
On the other hand, the one-byte RTP header extensions are storing the
length as a 4 bit integer with an offset of 1 (i.e. 0 means 1 byte
extension length), so here 16 is the correct maximum length.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6180>
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>
If we drop all messages with the same clock id as ours we will also
drop all messages coming from a PTP clock on our host since both clock
ids are build from the same MAC address.
At least for Linux we do not see our own messages anyway since the
network stack can well distinguish between multicast send from our
socket or from another socket on the same machine. To make sure that
this works for all supported platforms just drop delay requests since
this is the only message that is sent from the GStreamer PTP clock.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6172>
If we don't specify a path for loading, the runtime linker will search
for the library instead, which will use the usual mechanisms: RPATHs,
LD_LIBRARY_PATH, PATH (on Windows), etc.
Also try harder to load a non-devel libpython using INSTSONAME, if
available.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6159>
After a flushing seek, rtspsrc doesn't reset the last_ret value for
streams, so might immediately shut down again when it resumes pushing
buffers to pads due to a cached `GST_FLOW_FLUSHING` result
Prevent a stored flushing value from immediately stopping
playback again by resetting pad flows before (re)starting
playback.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6137>
This can be used to store informational messages, errors or
warnings which can later be shown to the user in gst-inspect-1.0,
which can be useful for plugins that expose elements dynamically
based on external libraries or hardware capabilities.
Status messages can then provide an indication as to why a
plugin doesn't have any elements listed, for example.
Plus unit test to make sure code paths are exercised a little.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3832>
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>
Because this depayloader may build several output buffers within one
process run we push them all into a GstBufferList and push them out at
once to make sure that each buffer gets notified about each header
extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
Because this depayloader may build several output buffers within one
process run we push them all into a GstBufferList and push them out at
once to make sure that each buffer gets notified about each header
extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
Because this depayloader may build several output buffers within one
process run we push them all into a GstBufferList and push them out at
once to make sure that each buffer gets notified about each header
extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
Because this depayloader may build several output buffers within one
process run we push them all into a GstBufferList and push them out at
once to make sure that each buffer gets notified about each header
extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
Because this depayloader may build several output buffers within one
process run we push them all into a GstBufferList and push them out at
once to make sure that each buffer gets notified about each header
extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
Because this depayloader may build several output buffers within one process
run we push them all into a GstBufferList and push them out at once to
make sure that each buffer gets notified about each header extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
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>
When allocating buffers with alignment parameters specified, it
may be necessary to overallocate memory to adjust to the requested
alignment. Previously the padding length was not included in the mmaped
buffer size, leaving unmapped bytes at the end of the buffer.
This caused intermittent SEGV faults and valgrind failures when running
the wayland_threads example.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6104>
The srt unittest test_src_listener_sink_call will sometimes fail under
valgrind with the following splat:
Memcheck, a memory error detector
Copyright (C) 2002-2017, and GNU GPL'd, by Julian Seward et al.
Using Valgrind-3.18.1 and LibVEX; rerun with -h for copyright info
Parent PID: 14579
HEAP SUMMARY:
in use at exit: 799,848 bytes in 2,182 blocks
total heap usage: 64,090 allocs, 61,908 frees, 37,891,032 bytes allocated
120 bytes in 1 blocks are definitely lost in loss record 1,563 of 1,681
at 0x4842FF5: operator new(unsigned long) (vg_replace_malloc.c:422)
by 0x6031E29: srt::sync::SetThreadLocalError(CUDTException const&) (sync_posix.cpp:461)
by 0x5FCD77E: CUDT::epoll_wait(int, std::set<int, std::less<int>,
std::allocator<int> >*, std::set<int, std::less<int>,
std::allocator<int> >*, long, std::set<int, std::less<int>,
std::allocator<int> >*, std::set<int, std::less<int>, std::allocator<int> >*) [clone .cold] (api.cpp:3796)
by 0x5FE2F79: UDT::epoll_wait2(int, int*, int*, int*, int*, long, int*, int*, int*, int*) (api.cpp:4277)
by 0x5F0C626: gst_srt_object_read (gstsrtobject.c:1569)
by 0x5F0F978: gst_srt_src_fill (gstsrtsrc.c:180)
by 0x5F5A2A1: gst_base_src_default_create (gstbasesrc.c:1620)
by 0x5F5C9AE: gst_base_src_get_range (gstbasesrc.c:2630)
by 0x5F5EF5A: gst_base_src_loop (gstbasesrc.c:2959)
by 0x4918B13: gst_task_func (gsttask.c:399)
by 0x4A60B33: g_thread_pool_thread_proxy.lto_priv.0 (gthreadpool.c:354)
by 0x4A5DC41: g_thread_proxy (gthread.c:826)
by 0x4F532A4: start_thread (pthread_create.c:481)
by 0x4C71322: clone (clone.S:95)
An issue has been started against libsrt here:
https://github.com/Haivision/srt/issues/2867
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6098>
External plugin loader support for Windows is introduced
in this dev cycle. Since helper binary was not required (useless)
before this version, people may not ship the binary
with new GStreamer version, then they will observe warning message.
Instead of displaying the warning at plugin loading time,
checks helper bin earlier and disable external plugin loader
if helper binary is not installed.
Fixes: https://gitlab.freedesktop.org/gstreamer/cerbero/-/issues/448
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6083>
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>
