With the 2.72 release, glib-networking developers have decided that
TLS certificate validation cannot be implemented correctly by them, so
they've deprecated it.
In a nutshell: a cert can have several validation errors, but there
are no guarantees that the TLS backend will return all those errors,
and things are made even more complicated by the fact that the list of
errors might refer to certs that are added for backwards-compat and
won't actually be used by the TLS library.
Our best option is to ignore the deprecation and pass the warning onto
users so they can make an appropriate security decision regarding
this.
We can't deprecate the tls-validation-flags property because it is
very useful when connecting to RTSP cameras that will never get
updates to fix certificate errors.
Relevant upstream merge requests / issues:
https://gitlab.gnome.org/GNOME/glib/-/merge_requests/2214https://gitlab.gnome.org/GNOME/glib-networking/-/issues/179https://gitlab.gnome.org/GNOME/glib-networking/-/merge_requests/193
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2494>
For formats which we don't have fast-path implementation, compositor
will convert it to common unpack formats (AYUV, ARGB, AYUV64 and ARGB64)
then blending will happen using the intermediate formats.
Finally blended image will be converted back to the selected output format
if required.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1486>
It is entirely possible for the cancellable to be cancelled (and freed)
in gst_rtsp_connection_flush() while there may be an ongoing read/write
operation.
Nothing prevents gst_rtsp_connection_flush() from waiting for the
outstanding read/writes.
This could lead to a crash like (where cancellable has been freed
within gst_rtsp_connection_flush()):
#0 0x00007ffff4351096 in g_output_stream_writev (stream=stream@entry=0x7fff30002950, vectors=vectors@entry=0x7ffe2c6afa80, n_vectors=n_vectors@entry=3, bytes_written=bytes_written@entry=0x7ffe2c6af950, cancellable=cancellable@entry=0x7fff300288a0, error=error@entry=0x7ffe2c6af958) at ../subprojects/glib/gio/goutputstream.c:377
#1 0x00007ffff44b2c38 in writev_bytes (stream=0x7fff30002950, vectors=vectors@entry=0x7ffe2c6afa80, n_vectors=n_vectors@entry=3, bytes_written=bytes_written@entry=0x7ffe2c6afb90, block=block@entry=1, cancellable=0x7fff300288a0) at ../subprojects/gst-plugins-base/gst-libs/gst/rtsp/gstrtspconnection.c:1320
#2 0x00007ffff44b583e in gst_rtsp_connection_send_messages_usec (conn=0x7fff30001370, messages=messages@entry=0x7ffe2c6afcc0, n_messages=n_messages@entry=1, timeout=timeout@entry=3000000) at ../subprojects/gst-plugins-base/gst-libs/gst/rtsp/gstrtspconnection.c:2056
#3 0x00007ffff44d2669 in gst_rtsp_client_sink_connection_send_messages (sink=0x7fffac0192c0, timeout=3000000, n_messages=1, messages=0x7ffe2c6afcc0, conninfo=0x7fffac019610) at ../subprojects/gst-rtsp-server/gst/rtsp-sink/gstrtspclientsink.c:1929
#4 gst_rtsp_client_sink_try_send (sink=sink@entry=0x7fffac0192c0, conninfo=conninfo@entry=0x7fffac019610, requests=requests@entry=0x7ffe2c6afcc0, n_requests=n_requests@entry=1, response=response@entry=0x0, code=code@entry=0x0) at ../subprojects/gst-rtsp-server/gst/rtsp-sink/gstrtspclientsink.c:2845
#5 0x00007ffff44d3077 in do_send_data (buffer=0x7fff38075c60, channel=<optimized out>, context=0x7fffac042640) at ../subprojects/gst-rtsp-server/gst/rtsp-sink/gstrtspclientsink.c:3896
#6 0x00007ffff4281cc6 in gst_rtsp_stream_transport_send_rtp (trans=trans@entry=0x7fff20061f80, buffer=<optimized out>) at ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream-transport.c:632
#7 0x00007ffff4278e9b in push_data (stream=0x7fff40019bf0, is_rtp=<optimized out>, buffer_list=0x0, buffer=<optimized out>, trans=0x7fff20061f80) at ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c:2586
#8 check_transport_backlog (stream=0x7fff40019bf0, trans=0x7fff20061f80) at ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c:2645
#9 0x00007ffff42793b3 in send_tcp_message (idx=<optimized out>, stream=0x7fff40019bf0) at ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c:2741
#10 send_func (stream=0x7fff40019bf0) at ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-stream.c:2776
#11 0x00007ffff7d59fad in g_thread_proxy (data=0x7fffbc062920) at ../subprojects/glib/glib/gthread.c:827
#12 0x00007ffff7a8ce2d in start_thread () from /lib64/libc.so.6
#13 0x00007ffff7b12620 in clone3 () from /lib64/libc.so.6
Fix by adding a cancellable lock and returning an extra reference used
across all read/write operations. gst_rtsp_connection_flush() can free
the in-use cancellable and it will no longer affect any in progress
read/write.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2799>
4x downscaling of chroma with co-sited chroma has never worked
it seems.
Fixes incorrect videotestsrc output and videoconvert conversions
to Y41B, YUV9, YVU9 and IYU9 with co-sited chroma.
e.g.
gst-launch-1.0 videotestsrc ! video/x-raw,format=Y41B,width=1280,height=720 ! \
videoconvert ! autovideosink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2789>
SMPTE 170M and 240M use the same RGB and white point coordinates
and therefore both primaries can be considered functionally
equivalent.
Also, some transfer functions have different name but equal
gamma functions. Adding another colorimetry compare function
to deal with thoes cases at once
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2765>
Raw memory upload should always be the least preferred input
caps, only added by the raw memory uploader as the last thing
in the caps.
Caps negotiation should still choose raw data when it needs to,
and other upload methods that can accept raw data buffers will still do so.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2725>
gst_video_convert_scale_get_fixed_format() receives 'othercaps' from
basetransforms' fixate_caps() vmethod which explicitly mentions that
'`othercaps` may not be writable'.
The gst_caps_intersect() call just before may or may not produce new
caps. Particularly in cases like EMPTY or ANY caps on either of the
inputs, only a ref is taken and returned to the caller.
As a result, gst_video_convert_scale_fixate_format() may have attempted
to modify a non-writable caps structure.
Fix by adding a gst_caps_make_writable().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2709>
There's no need to re-assign the return value of
g_string_append_*() functions and such to the variable
holding the GString. These return values are just for
convenience so function calls can be chained. The actual
GString pointer won't change, it's not a GList after all.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2685>
This reverts commit 6f9ae5d758.
The _transform_caps() function can't tell the difference
between the caller wanting to know the output caps
for the current method, or all possible output caps. If
it includes caps for all possible methods, glupload can
end up negotiating and sending the wrong output caps
downstream.
Partially reverts !2687Fixes#1310
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2699>
If no filter caps are provided with a caps query, always
generate a full set of all caps from all upload methods,
not just the configured one. This is needed to handle
renegotiation when dealing with raw sysmem caps - as the upload
method might accept raw sysmem caps, but only the raw data
uploader adds those to the caps query.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2687>
This reverts commit f3292dc156.
Only the raw data uploader should add sysmem caps to the
actual caps query, because we want them to be at the
lowest priority. If upstream does select to send raw
caps, then the correct upload method will still
be chosen because the accept_caps implementation
will accept them
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2687>
When checking if we need to reconfigure when uploading, check
specifically the output caps of the current method will
result in compatible/incompatible caps, not the full set
of output caps from all upload methods.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2687>
Fixes warnings like:
Received a structure string that contains '="0.5"'. Reading as a gdouble value, rather than a string value. This is undesired behaviour, and with GStreamer 1.22 onward, this will be interpreted as a string value instead because it is wrapped in '"' quotes. If you want to guarantee this value is read as a string, before this change, use '=(string)"0.5"' instead. If you want to read in a gdouble value, leave its value unquoted.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2621>
Some encoders (e.g. Makito) have H265 field-based interlacing, but then
also specify an 1:2 pixel aspect ratio. That makes it kind-of work with
decoders that don't properly support field-based decoding, but makes us
end up with the wrong aspect ratio if we implement everything properly.
As a workaround, detect 1:2 pixel aspect ratio for field-based
interlacing, and check if making that 1:1 would make the new display
aspect ratio common. In that case, we override it with 1:1.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2577>
When collection is updated, decodebin3 exposes pad first and then
streams-selected message is posted.
The condition can cause a situation where playbin3 links non-existing
combiner/playsink pads (since streams-selected is not posted yet) with
new decodebin output pad. This commit will re-check selected/active
streams condition on pad-added and reconfigure output if needed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2482>
zlib is required, and if it isn't found it is checked several ways and
then forced via subproject(). This code was added in commit
b93e37592a, to account for systems where
zlib doesn't have pkg-config files installed.
But Meson already does dependency fallback, and also, since 0.54.0, does
the in-between checks for find_library('z') and has_header('zlib.h') via
the "system" type dependency. Simplify dependency lookup by marking it
as required, which also makes sure that the console log doesn't
confusingly list "not found".
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2484>
It is valid to have the padding set to 1 on the first packet and it
happens very often from TWCC packets coming from libwebrtc. This means
that we were totally ignoring many TWCC packets.
Fix test that checked that a first packet with padding was not valid and
instead test a single twcc packet with padding to check precisely what
this patch was about.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2422>
Background:
Whenever a caps event is received by appsink, the caps are stored in the
same internal queue as buffers. Only when enough buffers have been
popped from the queue to reach the caps, `priv->sample` gets its caps
updated to match, so that they are correct for the following buffers.
Note that as far as upstream elements are concerned, the caps of appsink
are updated immediately when the CAPS event is sent. Samples pulled from
appsink retain the old caps until a later buffer -- one that was sent by
upstream elements after the new caps -- is pulled.
The race condition:
When a flush is received, appsink clears the entire internal queue. The
caps of `priv->sample` are not updated as part of this process, and
instead remain as those of the sample that was last pulled by the user.
This leaves open a race condition where:
1. Upstream sends a new caps event, and possibly some buffers for the
new caps.
2. Upstream sends a flush (possibly from a different thread).
3. Upstream sends a new buffer for the new caps. Since as far as
upstream is concerned, appsink caps are the new caps already, no new
CAPS event is sent.
4. The appsink user pulls a sample, having not pulled before enough
samples to reach the buffers sent in step 1.
Bug: the pulled sample has the old caps instead of the new caps.
Fixing the race condition:
To avoid this problem, when a buffer is received after a flush,
`priv->sample`'s caps should be updated with the current caps before the
buffer is added to the internal queue.
Interestingly, before this patch, appsink already had code for this, in
gst_app_sink_render_common():
/* queue holding caps event might have been FLUSHed,
* but caps state still present in pad caps */
if (G_UNLIKELY (!priv->last_caps &&
gst_pad_has_current_caps (GST_BASE_SINK_PAD (psink)))) {
priv->last_caps = gst_pad_get_current_caps (GST_BASE_SINK_PAD (psink));
gst_sample_set_caps (priv->sample, priv->last_caps);
GST_DEBUG_OBJECT (appsink, "activating pad caps %" GST_PTR_FORMAT,
priv->last_caps);
}
This code assumes `priv->last_caps` is reset when a flush is received,
which makes sense, but unfortunately, there was no code in the flush
code path resetting it.
This patch adds such code, therefore fixing the race condition. A unit
test demonstrating the bug and testing its behavior with the fix has
also been added.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2413>
gst_value_serialize() does more than what's needed to printf-ing
especially when given GValue is already string. Just print string
value as-is without gst_value_serialize() to avoid unreadable
string print, especially for multi-bytes character encoding cases.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2387>
* Remove fields no longer used, or that can be replaced by smaller code
* Rename "channels" to a more meaningful "input pads"
* Directly handle/use combiner pads in the combiners instead of on the playbin3
main structure
Remove the corresponding combiner sinkpad whenever a uridecodebin3 source pad
goes away
* If used, store the corresponding combiner sink pad in the SourcePad helper
structure
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2384>
As implemented, we only support OpenGL 3 API from version 3.2. Though, there
is no issue enabling GLSL 1.30 even if we are going to restrict our API usage
to 2. This allows using texelFetch() on OpenGL 3.0 and 3.1 drivers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2190>
Since the addition of tiling format with subsampled tile size
(NV12_16L32S), getting the tile width/height shifts and tile
size have become more complex. Add a helper to extract and
scale this information for the selected plane and format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2190>
The decision to store the input buffer depends on whether extensions
are to be added to the output buffer, I assume as an optimization.
This creates an issue for subclasses that call negotiate(), where
header_exts is actually populated, from their handle_buffer()
implementation: at chain time, no header extension has been negotiated
yet, which means that we don't add extensions to the first batch of
buffers that comes out.
Keep track of whether negotiate has been called (this is different
from the negotiated field) and always store the input buffer until
then. This fixes the issue while largely preserving the optimization.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2304>
Pipeline such as:
gst-launch-1.0 -vf videotestsrc ! video/x-raw,format=NV12,colorimetry=\(string\)bt709 \
! videoscale ! video/x-raw,format=I420 ! fakesink
Always trigger a error:
ERROR video-info video-info.c:556:gst_video_info_from_caps: no width property given
Because it is called before the fixate_size(), the src caps' resolution
may be absent or not fixed. That causes that the src video info can not
be created correctly and we can not inherit the colorimetry and chroma-site
from the input caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2289>
Fixing this pipeline:
gst-launch-1.0 filesrc location=sample.png ! pngdec ! videorate ! fakesink
- videorate receives a single buffer with pts = 0, duration = invalid;
- then it receives eos triggering this buffer to be pushed downstream;
- the pushing code was assuming that a duration was set, which is
impossible as we received a single buffer and no output framerate was
set either. So the best we can do is to push the buffer without
duration.
Fix#1177
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2296>
GAP events flagged with MISSING_DATA are transformed into GAP buffers
flagged with CORRUPTED.
In these cases, it is preferable to simply keep rendering the previous
buffer (if there was one) instead of flashing the pad in and out of
view.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/708>
Returning TRUE from the `transform_meta` function tells
GstBaseTransform to copy the meta into the new buffer. If videoscale
has already transformed a meta by scaling it, it should always return
FALSE to avoid duplicating the meta.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1630>
Since d0133a2d11 "videoconvert: Allow
passthrough for ANY caps features" videoconvert will always claim that
it supports any kind of memory which is true in very specific case (when
it is running in passthrough mode). To get elements that autoplug
converters depending on the caps running in the pipeline (like
autovideoconvert), we need to have converters no lie about what they can
do when queried `accept_caps` or `query_caps`.
This still accepts any caps feature as before but it introduces
a restriction in the way we handle memory capsfeatures.
We keep previous behaviour in videoconvert and videoscale.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/898>
Now that videoconvert and videoscale's are both based on
GstVideoConverter and are using the exact same code, it makes much more
sense to have one element doing the two operation, and it can be
more efficient in some cases (one single path for both operations).
This removes the `videoscale` and `videoconvert` plugins but keeps the element
but makes them also do both operations (adding some APIs to each element).
There is a small change in API for the `videoscale:dither` property which
was previously a totally unused boolean, it is now an enum and is used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/898>
The format of the caps fields is
ssrc-(SSRC_VALUE)-(ATTRIBUTE_NAME)=(ATTRIBUTE_VALUE)
.
Parsing of the attributes from the caps into the SDP is not implemented
as this depends not only a single stream's caps but on the whole rtpbin
configuration.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2132>
GLib guarantees libintl is always present, using proxy-libintl as
last resort. There is no need to mock gettex API any more.
This fix static build on Windows because G_INTL_STATIC_COMPILATION must
be defined before including libintl.h, and glib does it for us as part
as including glib.h.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2028>
If a file includes a new version of a plugin that exits in the
registry, the output of gst-inspect is incorrect. The output has the
correct version but incorrect filename, and element description.
This seems to have also fixed some documentation issues.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1344>
get_merged_collection() returns an owned stream collection and was
leaked in the else block.
Fix leak when running:
GST_TRACERS=leaks GST_DEBUG="GST_TRACER:7,leaks:6" gst-play-1.0 --use-playbin3 test.mkv
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/954>
Make sure that the requested stream selection isn't identical to the current
one. If that's the case, just carry on as usual.
This avoids multiple `streams-selected` posting ... when the selection didn't
change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2185>
timeapi.h is missing in our MinGW toolchain. Include mmsystem.h
header instead, which defines struct and APIs in case of our MinGW
toolchain. Note that in case of native Windows10 SDK (MSVC build),
mmsystem.h will include timeapi.h
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2153>
The documentation could be read to mean that the caller continuous to
'own' the buffer, and that there is some other mechanism to find out
when to unref it.
Clarify that "not taking ownership" here means "taking a reference",
and specify that you can unref it at any time after calling the
function.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2110>
They are part of gst_dep already and we have to make sure to always have
gst_dep. The order in dependencies matters, because it is also the order
in which Meson will set -I args. We want gstreamer's config.h to take
precedence over glib's private config.h when it's a subproject.
While at it, remove useless fallback args for gmodule/gio dependencies,
only gstreamer core needs it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2031>
The console HANDLE will be keep signalled state unless application
reads console input buffer immediately. So we should read and flush
console input buffer from the thread where the event is signalled,
instead of GMain context thread.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2058>
Configure playsink tried element with the bus of the main pipeline.
That tried element can be a gl video sink, which would benefit from being
able to propagate context messages to the main pipeline and have other
internal pipeline elements configured with it. Having different elements
configured with the same GL context allows them to share buffers with
video/x-raw(memory:GLMemory) caps and achieving zero-copy.
Thanks to Alicia Boya García <aboya@igalia.com> for her work co-debugging
the issue and contributing to find a solution.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2056>
Sources that can internally handle buffering shouldn't have yet-another
buffering element after it. This can be simply detected by checking if it can
answer a TIME BUFFERING query just after creation.
If that is the case, we can expose the element source pads directly
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1905>
Use the return value from gst_element_link_pads() and gst_bin_add()
Fixes:
../ext/gl/gstglmixerbin.c:305:12: error: variable 'res' set but not used [-Werror,-Wunused-but-set-variable]
gboolean res = TRUE;
^
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2038>
Add 5 new navigation event types for touchscreen events, with the same
naming and meaning as in libinput - touch-down, touch-motion, touch-up,
touch-frame and touch-cancel - as well as constructors and parse
functions for them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1633>
Add a function to get x/y coordinates from suitable navigation events,
and one to create a copy with given coordinate values.
For e.g. translating event coordinates, this avoids having to either
switch on the event type to select the right parse function, or
having to rely on implementation details of the underlying event
structure.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1633>
This deprecates the current send_event interface, and the wrapper
functions based on it, replacing it with a send_event_simple interface and
wrapper function. Together with the new event constructors, this avoids
implementations having to directly access the underlying structure.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1633>
