* RED_OR_ALPHA8 will map value to alpha for OpenGL, use R8 to avoid
2nd shader
* Determine texel size for proper texture memory preparation
* QByteArray::fromRawData() does shallow copy and thus leads to use of
corrupted memory
* Make sure RGBA dummy texture is fully opaque
* QRhiTexture::create() must be called to allocate texture resources
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6581>
During RTP-Info synchronization, clock_base was temporarily switched
from the actual clock-base to the base RTP time and then back some lines
later.
Instead directly work with the base RTP time. The comment about using a
signed variable for convenience doesn't make any sense because all
calculations done with the value are unsigned.
Similarly, rtp_clock_base was overridden with the rtp_delta when
calculating it, which was fine because it is not used anymore
afterwards. Instead, introduce a new variable `rtp_delta` to make this
calculation clearer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6575>
It's not in the same period as the current RTP base time but always in
the very first period. This avoids using it again at a much later time.
The code in question is only triggered with rtcp-sync=rtp-info.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6575>
It is compared to other extended RTP timestamps all over rtpjitterbuffer
and since 4df3da3bab the initial extended RTP timestamp is not equal
anymore to the plain RTP time.
Continue passing a non-extended RTP timestamp via the `sync` signal for
backwards compatibility. It will always be a timestamp inside the first
extended timestamp period anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6575>
When the buffer DTS is estimated based on arrival time at the
jitterbuffer (rather than provided on the incoming buffer itself),
it shouldn't be used for skew adjustment. The typical case is
packets being deinterleaved from a tunnelled TCP/HTTP RTSP stream,
and the arrival times at the jitter buffer are not well enough
correlated to usefully do skew adjustments.
This restores the original intended behaviour for the 'estimated dts'
path, that was broken years ago during other jitterbuffer refactoring.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6561>
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/6539>
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/6519>
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/6507>
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/6492>
The attempt to free the domain data is happeing twice during the ptp deinit.
Once while iterating through the list domain_data and second while iterating
through the list domain_clocks, so this is crashing the application
trying to gst_ptp_deinit
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6457>
And handle the case of a NULL buffer being returned cleanly, which is
valid as long as a buffer list is returned instead. Previously this
would cause an assertion because of calling gst_buffer_unref() with
NULL.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6463>
If we can calculate timestamps for buffers, then set the duration
on outgoing buffers based on the number of samples depayloaded.
This can fix the muxing to mp4, where otherwise the last packet
in a muxed file will have 0 duration in the mp4 file.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6456>
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/6454>
Some driver doesn't implement enum_framesize. The maximum supported
size can be got by trying format with a very large size. Also need
to set max_width/max_height for this case, otherwise default encoded
buffer size 256kB is too small.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6430>
The --atleast-version implies --exists, but the implementation in
earlier commits had the version check applied any time the --exists was
checked, and the default value of the major and minor versions were set
to the GStreamer major and minor versions. The resulting behavior would
have gst-inspect return '1' if the plugin's version didn't match
gstreamer's even when --atleast-version was not specified in the command
line args. The change in this patch removes that behavior and adds
tests to verify that if --exists is specified WITHOUT --atleast-version
the version check will NOT be applied. If both arguments are specified
and the version does not match the arg-supplied version number, a new
return code of '2' is used to uniquely identify the failure.
Fixes#3246
Signed-off-by: Thomas Goodwin <thomas.goodwin@laerdal.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6414>
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/6411>
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/6411>
`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/6387>
This reverts commit 8e923a8e2d.
This caused regressions, see #3303.
Without this commit, osxaudiosrc ! osxaudiosink won't work
right, but since that hasn't really been a huge problem
for years it's probably best to revert this until a proper
solution can be figured out.
(cherry picked from commit f04f86f3ee)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6384>
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/6376>
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/6376>
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/6376>
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/6375>
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/6339>
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/6344>
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/6350>
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/6345>
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/6343>
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/6342>
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/6338>
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/6334>
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/6332>
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/6331>
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/6330>
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/6298>
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/6295>
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/6295>
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/6277>
