Currently the element calls abort when failed to prepare reference
picture set. This can happent when the input stream is somehow
corrupted, like a rtsp strem with lost packets. Now it will only
return with GST_FLOW_ERROR instead of terminating whole process.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3505>
An end packet is only produced once for the last subtitle, so multiple
GAP events between subtitles would result only in a single end packet
and nothing else otherwise. This would potentially starve downstream
then, so instead forward the GAP events in that case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3534>
Starting with glib 2.75, `NULL` is `nullptr`, which cannot be
implicitly coerced to `0`, unlike `NULL`. So explicitly pass `0`.
```
[3206/4524] Compiling C++ object subprojects/gst-plugins-bad/sys/directshow/gstdirectshow.dll.p/dshowvideosink.cpp.obj
FAILED: subprojects/gst-plugins-bad/sys/directshow/gstdirectshow.dll.p/dshowvideosink.cpp.obj
"cl" "-Isubprojects\gst-plugins-bad\sys\directshow\gstdirectshow.dll.p" "-Isubprojects\gst-plugins-bad\sys\directshow" "-I..\subprojects\gst-plugins-bad\sys\directshow" "-Isubprojects\gst-plugins-bad" "-I..\subprojects\gst-plugins-bad" "-Isubprojects\gst-plugins-base\gst-libs" "-I..\subprojects\gst-plugins-base\gst-libs" "-Isubprojects\gstreamer\libs" "-I..\subprojects\gstreamer\libs" "-Isubprojects\gstreamer" "-I..\subprojects\gstreamer" "-Isubprojects\orc" "-I..\subprojects\orc" "-I..\subprojects\gst-plugins-bad\sys\directshow\strmbase\baseclasses" "-Isubprojects\gst-plugins-base\gst-libs\gst\video" "-Isubprojects\gstreamer\gst" "-Isubprojects\gst-plugins-base\gst-libs\gst\audio" "-Isubprojects\gst-plugins-base\gst-libs\gst\tag" "-IC:/gst-install/include/glib-2.0" "-IC:/gst-install/lib/glib-2.0/include" "-IC:/gst-install/include" "/MD" "/nologo" "/showIncludes" "/utf-8" "/W2" "/EHsc" "/O2" "/Zi" "/wd4018" "/wd4146" "/wd4244" "/wd4305" "/utf-8" "/we4002" "/we4003" "/we4013" "/we4020" "/we4027" "/we4029" "/we4033" "/we4045" "/we4047" "/we4053" "/we4062" "/we4098" "/we4101" "/we4189" "/utf-8" "-D_MBCS" "/wd4189" "/wd4456" "/wd4701" "/wd4703" "/wd4706" "/wd4996" "-DHAVE_CONFIG_H" "/Fdsubprojects\gst-plugins-bad\sys\directshow\gstdirectshow.dll.p\dshowvideosink.cpp.pdb" /Fosubprojects/gst-plugins-bad/sys/directshow/gstdirectshow.dll.p/dshowvideosink.cpp.obj "/c" ../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp
../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(62): warning C5051: attribute 'noinline' requires at least '/std:c++20'; ignored
../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(123): error C2664: 'LRESULT SendMessageA(HWND,UINT,WPARAM,LPARAM)': cannot convert argument 3 from 'nullptr' to 'WPARAM'
../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(123): note: A native nullptr can only be converted to bool or, using reinterpret_cast, to an integral type
C:\Program Files (x86)\Windows Kits\10\include\10.0.19041.0\um\winuser.h(3690): note: see declaration of 'SendMessageA'
../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(635): error C2664: 'BOOL SystemParametersInfoA(UINT,UINT,PVOID,UINT)': cannot convert argument 2 from 'nullptr' to 'UINT'
../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(635): note: A native nullptr can only be converted to bool or, using reinterpret_cast, to an integral type
C:\Program Files (x86)\Windows Kits\10\include\10.0.19041.0\um\winuser.h(13153): note: see declaration of 'SystemParametersInfoA'
../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(1593): error C2664: 'LRESULT SendMessageA(HWND,UINT,WPARAM,LPARAM)': cannot convert argument 3 from 'nullptr' to 'WPARAM'
../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(1593): note: A native nullptr can only be converted to bool or, using reinterpret_cast, to an integral type
C:\Program Files (x86)\Windows Kits\10\include\10.0.19041.0\um\winuser.h(3690): note: see declaration of 'SendMessageA'
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3528>
The stream selection is done on the currently outputting tracks, but in order to
(de)activate the backing streams we can only do it if the input and output
period are identical.
Fixes crash when doing stream selection during period migration
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3525>
This adds "id" variants to most debugging functions, and allows providing a
string identifier instead of a GObject.
This allows providing unified and clearer debug logs for all the
non-gobject-based items, and opens the way for more unified logging.
As an extension, copying the object name is avoided as much as possible, by
using it directly instead of going through another copy.
* API : gst_debug_message_get_object_id
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3483>
This regression was introduce by fix for making buffer pool thread safe. When
we renegotiate, the pool will be setup after we set the format. But the code
has been simplified to only get the pool once before, which caused a null
pointer deref.
Fixes 94ba019 ("v4l2: Fix SIGSEGV on 'change state' during 'format change'")
Related to !3481Fixes#1626
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3513>
There are cases where user might want to be in full control of the
timeline and not be limited by the checks that are being done by GES
to go from one timeline layout to another, this should be doable as
it is a valid use case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3501>
When coloring is in use, those escape codes are going to be created many times
for almost all debug lines.
Don't create plenty of temporary allocations, and instead build the escape code
ourselves statically
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3498>
Because of the asynchronous resolving of mDNS ICE candidates it is
possible that GstWebRTCICE outlives webrtcbin. This in turn prolongs
the lifetime of the GstWebRTCNiceStream objects via refs in
nice_stream_map. Thus the GstWebRTCICETransport objects held in
GstWebRTCNiceStream may be invalid at the time they are accessed by
the _on_candidate_gathering_done() callback since GstWebRTCNiceStream
doesn't take a reference to them. Doing so would create a circular
reference, so instead this commit introduces weak references to the
transport objects and then we can check if the objects are valid before
accessing them.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3502>
And even that vaav1dec doesn't use vabasedec negotiate vmethod, it should align
with the new scheme of using base's width & height for surface size and
output_info structure for downstream display size negotiation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3480>
This vmethod can be used by decoders with the same VA decoder reopen logic:
same profile, chroma, width and height.
Also a new public method called gst_va_base_dec_set_output_state() with the
common GStreamer code for setting the output state, which is always called by
the negotiate vmethod.
In order to do this refactoring, new variables in vabasedec have to be populated
by the decoders:
* width and height define the resolution set in VA decoder. In the case of H264
would be de coded_width and codec_height, or max_width and max_height in AV1.
* output_info is the downstream video info used for negotiation in
gst_va_base_dec_set_output_state().
* input_state, from codec parent class shall be also held by vabasedec
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3480>
There could be multi-GPU setups where the non-first has more
entrypoints than the first one, and the elements names are not
homogeneous, leading to pipeline building error.
This patch add the render node in the elements names when they belong
to the non-first device.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3491>
To fix the warning on Alderlake
vafilter gstvafilter.c:534:gst_va_filter_ensure_filters:<vafilter0>
vaQueryVideoProcFiltersCaps: list argument exceeds maximum number
Increase the number of caps to 16 as vadumpcaps does.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3473>
We want to make it so that we prefer a higher, not lower, number of
channels. Otherwise, this pipeline would convert from 2 to 1 channels:
gst-launch-1.0 audiotestsrc ! audio/x-raw,channels=2 ! opusenc ! queue ! opusdec ! queue ! opusenc ! fakesink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3494>
In cases where an invalid input packet is submitted to the decoder we emit a
warning but reporting the flow error upstream would also be useful. This came up
with a case were the application interacts directly with the decoder, using a
mechanism similar to GstHarness.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3463>
Whenever the surface is resized before the stream is negotiated, we endup
with an assertion in libgstvideo.
gst_video_center_rect: assertion 'src->h != 0' failed
This fixes it, by following the style aready in place, which is to ensure
surfaces have a minimum size of 1x1.
Fixes#1139
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3467>
gst-launch-1.0 audiotestsrc ! udpsink host=127.0.0.1
gst-launch-1.0 udpsrc ! audioconvert ! autoaudiosink
would crash with a floating point exception when clipping the input
buffer owing to a division by zero because no caps event was received.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3469>
Windows supports various IPC methods but that's completely
different form that of *nix from implementation point of view.
So, instead of adding shared memory functionality to existing
shm plugin, new WIN32 shared memory source/sink elements
are implemented in this commit.
Each videosink (server) and videosrc (client) pair will communicate
using WIN32 named pipe and thus user should configure unique/proper
pipe name to them (e.g., \\.\pipe\MyPipeName).
Once connection is established, videosink will create named shared memory
object per frame and client will be able to consume the object
(i.e., memory mapped file handle) without additional copy operation.
Note that implementations under "protocol" directory are almost
pure C/C++ with WIN32 APIs except for a few defines and debug functions.
So, applications can take only the protocol part so that the application
can send/receive shared-memory object from/to the other end
even if it's not an actual GStreamer element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3441>
Currently in rtp_session_send_rtp(), the existing ntp-64 RTP header
extension timestamp is updated with the actual NTP time before sending
the packet. However, there are some circumstances where we would like
to preserve the original timestamp obtained from reference timestamp
buffer metadata.
This commit provides the ability to configure whether or not to update
the ntp-64 header extension timestamp with the actual NTP time via the
update-ntp64-header-ext boolean property. The property is also exposed
via rtpbin. Default property value of TRUE will preserve existing
behavior (update ntp-64 header ext with actual NTP time).
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1580
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3451>
- Based heavily on the existing Qt5 integration however:
- The sharing of OpenGL resources is slightly different
- The integration with the scengraph is a bit different
- Wayland, XCB and KMS have been smoke tested. Android, MacOS/iOS,
Windows may or may not work.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3281>
Make sure that group-id of a given play item are made consistent from the
start (sources) and all the way through the output.
This ensures that we can reliably detect that we have switched to the next play
item on the output of decodebin3 (and we can therefore properly free/release it)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3457>
When shutting down, we want to remove the urisourcebin blocking probes ... but
we also want to propagate a GST_FLOW_FLUSHING upstream (and not
GST_FLOW_NOT_LINKED) to make the upstream task gracefully stop instead of
posting an error message.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3457>
When `is_selection_done` is called, it checks that all the requested streams are
present in the active stream list ...
... except there could very well be a (about to be removed) stream from the
previous selection present.
Therefore filter the list of streams we add to the message by the streams which
are actually requested.
Fixes issues when switching between different stream types (ex: video-only to
audio-only).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3457>
Previously we tried to route an incoming RTCP FB FIR to the correct ssrc
using the "media source" component of the RTCP FB message. However,
according to RFC5104 (section 4.3.1.2) the "media source" SHALL be set
to 0. Instead the ssrc(s) in use are propagated via the FCI data. Now
a specific GstForceKeyUnit event is sent for every ssrc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3292>
RTP source statistics are tracked for local senders by
treating them as a receiver of their own outbound packets.
Accordingly, track the highest packet seqnum so that the
packets-lost calculation generates a sensible number instead
of always reporting -$number_of_packets_sent
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3454>
There was a drm/drm_mode.h included added recently, drm/ is usually
referencing the linux kernel header, but we only requires the libdrm
headers to be installed. On top of this, including drm_mode.h is never
needed as its already included by drm.h.
Fixes#1596
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3452>
When getting a "404 Not Found" response from the DESCRIBE request, the
source produced a "No supported authentication protocol was found" error
instead of passing on the 404, which was confusing.
Only produce this error message when we're handling a response of "401
Unauthorized" without a compatible WWW-Authenticate header.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3414>
The legacy emulation in DRM/KMS drivers badly interact with GStreamer and
may cause the framerate to be halved. With this property, users can disable
vsync (which is handled internally by the emulation) in order to regain the
full framerate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3303>
The event type for instant-rate-change events was poorly chosen,
leading to them being re-sent too late and even after EOS.
Add a mechanism in GstPad for the sticky event order to be
different to the value of the event type to fix that up.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3387>
The original BUNDLE support commit placed a queue after the
rtpfunnel that combines streams, but I don't see a good reason for
it. It has default settings, so if network output is slow might
accidentally store up to 1 second of pending data, increasing
latency.
Remove it in favour of doing any necessary buffering before
webrtcbin. If it turns out that there is a reason for it to
exist, the limits should probably be configurable and small.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3437>
Currently, when rtspsrc property add-reference-timestamp-metadata=true,
a downstream rtph264depay element will attach multiple copies of the
same GstReferenceTimestampMeta to the depayloaded media buffers. This
can have signficant performance impacts further downstream in a pipeline
like the following:
rtspsrc add-reference-timestamp-metadata=true ! rtph264depay ! h264parse ! ... ! rtph264pay ! ...
For example, if there are 10 packet buffers for a frame of RTP H.264
video, each of those packet buffers will contain the same reference
timestamp meta. The rtph264depay element will then attach all 10
metadata to the depayloaded frame. And then later when we payload the
frame buffer again for proxying, we now have 10 more buffers each with
10 instance of the same metadata. Allocating/deallocating 100+ instances
of metadata @ 30fps for multiple streams has a pretty large performance
impact.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1578
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3431>
The tile width in pixel is not always available. Notably for
8L128 10bit format, the tile is 8x128 bytes, and the pixel
format is fully packed. That means that the tile contains at
least 6 pixels per line, but it also hold some bits of the
pixel from the same line on the previous or next tile.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3424>
In current tile representation, only tiles with power of two
width and height in bytes are supported. This limitation
prevents adding more complex tiles formats.
In this patch, we deprecate tile_ws and tile_hs from GstVideoFormatInfo and
replace if with an array of GstVideoTileInfo. Each plane tiles are then
described with their pixels width/height, line stride and total size.
The helper gst_video_format_info_get_tile_sizes() that depends on the
deprecated API is also being removed. This can simply be removed as it wasn't
in any stable release yet.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3424>
Setting force_live lets aggregator behave as if it had at least one of
its sinks connected to a live source, which should let us get rid of the
fake live test source hack that is probably present in dozens of
applications by now.
+ Expose API for subclasses to set and get force_live
+ Expose force-live properties in GstVideoAggregator and GstAudioAggregator
+ Adds a simple test
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3008>
The only case where we definitely need to write a new trun is when the
data_offset value does not match the end of the list of entries.
Needing multiple trun atoms is required when interleaving multiple
streams together.
All other cases can be covered by adding more entries to the existing
trun atom.
Fixes playback of fragemented mp4 in ffplay and chrome.
Using e.g. mp4mux fragment-duration=1000 fragment-mode=dash-or-mss
and
mp4mux fragment-duration=1000 fragment-mode=first-moov-then-finalise
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3426>
Attribute's value should use returned value from get_attribute for
VAConfigAttribRTFormat, since VAProfileHEVCMain10, in AMD Mesa Gallium,
can process either VA_RT_FORMAT_420 and VA_RT_FORMAT_420_10, which isn't
considered in gstreamer-vaapi design, where encoder's src pads will
expose only 4:2:0 color formats but no 4:2:0 10bit. So, this is a workaround
for this issue while new vah265enc is released.
Signed-off-by: Boyuan Zhang <boyuan.zhang@amd.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3397>
This was the intention from the start, just took me a few years *cough* to
actually implement it properly.
Gapless is handled by re-using as much as possible the same decoders and sinks
if present, and only pre-rolling switching at the sources level (with buffering
if/when needed).
In order to enable "gapless" playback, the "next" uri should be set at any time
between the moment the `about-to-finish` signal is emitted and the moment the
current play item is done. Previously this could only be done with the signal
emission.
This new implementation also allows "Instantaneous URI switching". This allows a
much faster way of switching playback entries while re-using as many elements as
possible. To enable this set `instant-uri` property to TRUE, the default being
FALSE.
API: instant-uri properties
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2784>
When dealing with gapless input (i.e. streams with changing group-id in
GST_EVENT_STREAM_START), we need to take into account the elapsed
running-time (if applicable) in order to properly calculate levels and output
time. Without doing this all incoming data from future groups would be
considered as being "late" and would be consumed immediately.
This does **NOT** modify the actual segment and buffer times, and is only used
internally.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2784>
DecodebinInput (and their backing parsebin or identity) are no longer released
when the corresponding sinkpad is unlinked, but when it's released.
The parsebin element will be resetted:
* If incoming caps are incompatible (was the case before)
* Or when unlinking and it was previously pull-based
This opens the way to use decodebin3 with changing inputs (i.e. gapless)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2784>
Introduce the option to have the streams be parsed with `parsebin` for
compatible sources (i.e. which are eligible for buffering in the same way as
before this commit).
By parsing the inputs directly, this allows more accurate buffering control:
* Instead of relying on potential bitrate information coming from somewhere
* and *without* being linked downstream
If `parse-streams` is activated and the stream is eligible for buffering, then a
`multiqueue` will be used on the output of `parsebin` in order to handle the
buffering.
API: `parse-streams`
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2784>
If the incoming streams are already parsed, there is no need to add yet-another
parsebin to process it *IF* that stream is compatible with a decoder or the
decodebin3 output caps.
This only applies if all the following conditions are met:
* The incoming stream can *NOT* do pull-based scheduling
* The incoming stream provides a `GstStream` and `GstStreamCollection`
* The caps are compatible with either the decodebin3 output caps or a decoder
input
If all those conditions are met, a identity element is used instead of a
parsebin element and the same code paths are taken.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2784>
* Instead of creating temporary `PendingPad` structures, always create a
DecodebinInputStream for every pad of parsebin
* Remove never used `pending_stream` field from DecodebinInputStream
* When unblocking a given DecodebinInput (i.e. wrapping a parsebin), also make
sure that other parsebins from the same GstStreamCollection are unblocked
since they come from the same source
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2784>
Make an explicit topology/tree of structures:
* ChildSrcPadInfo is created for each source element source pad
* ChildSrcPadInfo contains the chain of optional elements specific to that
pad (ex: typefind)
* A ChildSrcPadInfo links to one or more OutputSlot, which contain what is
specific to the output (i.e. optional buffering and ghostpad)
* No longer use GObject {set|get}_data() functions to store those structures and
instead make them explicit
* Pass those structures around explicitely in each function/callback
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2784>
The following problem could happen:
* Thread 1 : urisourcebin gets activated from READY->PAUSED
* Thread 2 : some element causes a pad to be added to urisourcebin , which gets
linked downstream, which decides to activate upstream to pull-based.
* That requires "activating" the pads from PUSH to NONE, and then from NONE to PULL
* Thread 1 : the base class state change handlers checks if all pads are
activated
The issue is that since going form PUSH to PULL requires going through NONE,
there is a window during which:
* Thread 1 : The pad was set to NONE (before being set to PULL)
* Thread 2 : The base class activates that pad (to PUSH)
* Thread 1 : The attempt to "activate" to PULL fails (silently or not)
This is very racy, so in order to avoid that, we make sure that we only add pads
once the transition from READY->PAUSED in the parent classes is done.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2784>
This change allow output caps to be updated even though we stay in
streaming state. This is needed so that any upstream updated to fields
like framerate, hdr data, etc. can result in a downstream caps event
being pushed.
Previously, any of these changes was being ignored and the downstream
caps would not reflect it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3328>
In theory, input caps can be updated anytime at non-keyframe or
sequence boundary, such as HDR10 metadata, framerate, aspect-ratio
or so. Those information update might not trigger ::new_sequence()
or subclass may ignore the changes.
By this commit, input state change will be tracked by baseclass
and subclass will be able to know the non-decoding-essential
update by checking the codec specific picture struct
on ::output_picture()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3328>
This reverts commit fcad4cc646.
This was wrong is so many ways.
* The memcmp was badly used (it should use == 0 to check the data is identical,
and not != 0)
* There was no boundary checks on the present stream section_data when passing
it to memcmp.
* The return value should have been TRUE (i.e. we have done all checks, none of
them failed, therefore the section has been seen before)
* stream->section_data would *always* be NULL if the section had already been
processed
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1559
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3421>
There are cases where upstream will not provide a framerate, or it won't be
fixed. But if there is latency introduced by the decoder we do want to report
it.
Therefore use the framerate stored in the actual decoder, which will have a
default.
Fixes hangs when playing back such streams with decodebin3 (where the multiqueue
will not have been informed of that downstream latency and not grow accordingly)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3391>
Postpone the cleanup of any consecutive sequence of lost frames
which starts at frame 0, until frame 100 is dequeued from driver.
This allows fluster tests JVT/CVWP2_TOSHIBA_E, JVC/CVWP3_TOSHIBA_E
and HEVC/POC_A_Bossen_3 that sends out-of-order frames to successfully
complete (e.g., test of Amphion vpu driver).
Fixes#1569
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3398>
In order to figure out if the "raw" audio contained within the wav
container is actually DTS, wavparse calls the typefinder helper
except that means it runs all typefinders.
Since it only cares about checking for DTS, we should only run the
audio/x-dts typefinder (if present). Commit 858e516 did not really
fix things.
Use the new type helper with the caps to fix this.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3417>
Set udpsrc seqnums on all events sent to the udpsrc's, and before
forwarding events out of rtspsrc set the latest seek seqnum on them if
any.
Also produce a consistent seqnum in rtspsrc from the very beginning.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>
Rewriting GstCudaConverter object, since the old implementation was not
well organized and it's hard to add new features.
Moreover, the conversion operations were not very optimized.
Major change of this implementation:
* Remove redundant intermediate conversion operations such as
any RGB -> ARGB(64) conversion or any YUV -> Y444 (or 16bits Y444).
That's not required most of cases. The only required case is
converting 24bits (such as RGB/BGR) packed format to 32bits format
because CUDA texture object does not support sampling 24bits format
* Use normalized sample fetching (i.e., [0, 1] range float value)
and also normalized coordinates system for CUDA texture.
It's consistent with the other graphics APIs such as Direct3D
and OpenGL, that makes sampling operations much easier.
* Support a kind of viewport and adopt math for colorspace conversion
from GstD3D11 implementation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3389>
GstCudaConverter object can do colorspace conversion and scale at once.
Adding new element "cudaconvertscale" to do that, this can
save unnecessary GPU operation if colorspace conversion and
rescale is required for given input stream format.
Most of codes are taken from d3d11convert element
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3389>
Introduce a new API that can return a GstTypeFind * with helper functions
and data set around buffer data.
While at it, drop factory field from GstTypeFindBufHelper. While it was
useful for logging, it was not passed through function arguments and keeping
it for logging would require an additional API increasing the API surface
and making it harder to use.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3296>
Instead of returning a "const gchar" or a "gchar" that should not be freed, always
return a duplicated string as those functions were used together with g_strdup anyway.
This is needed to prepare support for returning modified strings in next commit.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1147>
This is small regression from commit f7abd81a.
When calling `gst_element_query()` no pad is associated with that query, but the
current code always forwards the query to the associated pad, which is NULL in
previous case. This patch checks for the pad before forwarding the query.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3404>
If we don't receive any data from usrsctp, then there will be no src pad
for the stream id and the stream reset will fail to remove the relevant
src pad. Workaround by first attempting to add the relevant src pad, then
almost immediately removing it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3381>
Replace video_copy with memcpy to fix the issue when the sizes of the
src frame and dst frame don't match. Moreover, for Windows, you have to
do the copy first and call gst_msdk_import_to_msdk_surface later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3231>
Replace video_copy with memcpy to fix the issue when the sizes of the
src frame and dst frame don't match. Moreover, for Windows, you have to
do the copy first and call gst_msdk_import_to_msdk_surface later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3231>
Currently MSDK context does not support d3d11va. Now introduce d3d11va
device to MSDK context, making it able to create msdk session with d3d11
device and to easily share with upstream and donwstream.
Add environment variable to enable user to choose GPU device in multi-GPU
environment. This variable is only valid when there's no context
returned by upstream or downstream. Otherwise it will use the device
that created by upstream or downstream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3231>
Add support for more formats so as to run the libvpx high bit depth test suite.
This means the files under CONFIG_VP9_HIGHBITDEPTH
This also allows running the yuv444p 8bit file in the regular 8 bit vp9 suite.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3356>
If an input is malformed (only produces cea608 field 1 cc_data) then
when in passthrough we would effectively be dropping every second cea608
on output as we would not store any unused cea608 data.
Fix by having all code paths go through the framerate conversion code
which will store and retrieve any relevant data across buffers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
When a tile format is padded and imported as DMABuf, the stride
contains the information about the actual width and height in
number of tiles. This information is needed by the detiling shader
in order accuratly calculate the location of pixels. To fix that,
we also copy the offset and strides into the otuput format and
the converter will ensure that the shader is recompiled whenever
the stride changes.
This fixes video corruptions observed when decoding on MT8195
with videos that aren't not aligned to 64bytes in width.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3365>
... otherwise PAR can be wrongly signalled during the negotiation
Fixing below pipeline when desktop resolution is not 640x480
gst-launch-1.0.exe \
d3d11screencapturesrc ! videoscale !
video/x-raw,width=640,height=480,pixel-aspect-ratio=1/1 ! d3d11videosink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3360>
1. Removes the verification if the internal encoder is not opened
yet to allow the property setting.
2. And toggles on the base class' reconf flag for each property
variable that can be modified at run time.
3. Mark those modifiable properties as mutable while playing.
Currently the run-time modifiable properties are:
qpi, qpp, qpb, bitrate, target percentage, target usage and rate control
Other properties can be enabled too, but they need testing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2466>
Adds an internal function reset() which drains the internal queues and
calls the reconfig() vmethod.
This reset() method is called inconditionally at set_format() and in
handle_frame() if the instance's reconf flag is enabled.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2466>
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, 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/2466>
This method will return the caps configured in the reconstruct buffer
pool, and its maxium number of buffers to allocate.
The caps are needed later to know if the internal encoder has to be
reopened if the stream properties change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2466>
This adds a new boolean property `auto-reconnect`, defaulting to `true`.
Setting it to `false` makes the elements (in caller mode) immediately
report an error to the application instead of trying to reconnect.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3326>
Adding DirecShow video capture filter mode, in addition
to existing MediaFoundation and WinRT(UWP) mode, to support
DirectShow only filters (not KS driver compatible)
such as custom virtual camera filters.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3350>
Use gst_debug_set_threshold_from_string's new reset behavior to undo
GST_DEBUG and ensure the logging tests have a known configuration.
`gst_debug_set_threshold_from_string ("LOG", TRUE)` has the same effect
as `gst_debug_set_threshold_from_string ("", TRUE)` followed by
`gst_debug_set_default_threshold (GST_LEVEL_LOG)`.
Don't bother remembering the default log level set when the test
started. It will get reset by the next test, anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/605>
TLDR: Make `gst_set_threshold_from_string ("", TRUE)` reset *all*
threshold settings, including those set by previous invocations of
`gst_debug_set_threshold_from_string`.
The docs say:
@reset: %TRUE to clear all previously-set debug levels before setting
new thresholds
What actually happens is it sets the default threshold to `ERROR`,
leaves the patterns in place and calls
`gst_debug_category_reset_threshold` on each category.
In effect, any category that is matched by a pattern gets reset to that
threshold if the app changed it by directly invoking
`gst_debug_category_set_threshold`. All other categories are reset to
`ERROR`.
In my opinion this parameter currently has little value, as the same
effect can be achieved by including `ERROR` (without a pattern) in the
string, as in `"foo*:WARNING,*bar:INFO,ERROR"`.
What I actually expect it to do is reset *all* threshold settings,
including those set by previous invocations of
`gst_debug_set_threshold_from_string`, starting off with a clean slate
for the patterns provided with the call.
Otherwise there is no API to do this, besides:
- Painfully removing patterns one-by-one via
`gst_debug_unset_threshold_for_name` *if* you know what the patterns
are.
- Adding a `*:FOO` pattern to affect all categories, which makes the
default threshold useless and practically leaks all the old
patterns.
In my opinion this also makes it fit better into the layers of threshold
config, which is:
1. Temporary:
- `gst_debug_category_set_threshold`
- `gst_debug_category_reset_threshold`
2. Patterns:
- `gst_debug_set_threshold_for_name`
- `gst_debug_unset_threshold_for_name`
- `gst_debug_set_threshold_from_string`
- `GST_DEBUG`
3. Default:
- `gst_debug_set_default_threshold`
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/605>
The scenario is the following:
* Thread 1 is pushing an EOS event on a sinkpad
* Thread 2 is pushing a STREAM_START event on the same sinkpad before Thread 1
returns. Note : It starts pushing the event after Thread 1 took the object lock.
There is a potential race between:
* The moment Thread 1 sets the EOS flag once it has finished sending the
event (via store_sticky_event). When it does that it has both the STREAM and
OBJECT lock
* The moment Thread 2 sends the STREAM_START event (Which should release that
EOS status), but removing the EOS flag is only done while holding the OBJECT
lock and not the STREAM_LOCK, which means it could be re-set by Thread 1 before
it then checks again the EOS flag (without the STREAM lock taken).
The EOS flag unsetting by STREAM_START should be done with the STREAM lock
taken, otherwise it will be racy.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1452
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3320>
In a few cases throughout qtdemux, the results of QT_UINT32 were being
stored in a signed integer, which could cause subtle bugs in the case of
an integer overflow, even allowing the the result to equal a negative
number!
This patch prevents this by simply storing the results of this function
call properly in an unsigned integer type. Additionally, we fix up the
length checking with stsd parsing to prevent cases of child atoms
exceeding their parent atom sizes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3344>
Subtracting a gint from another (or a guint from another) has no guarantees that
it will result in a gint.
Therefore do the actual comparision instead.
Also use the *actual* type for comparing flags (the field value types are different)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3319>
Unlike the legacy elements, GstAdaptiveDemuxStream is a GObject now,
so a bunch of things that were actually stream methods on the
parent demux object can directly become stream methods now.
Move the stream class out to a header of its own.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
Sometimes g_input_stream_read_all_finish() can return
0 bytes, but still succeed (return TRUE) and have more
data available later. Only finish the transfer
if it returns 0 bytes *and* FALSE with no error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
The cancelled flag was only set in the stream finalize()
method, after all activity on the stream has stopped anyway.
Replace uses of cancelled with checks on the stream state.
Remove the replaced flag, which was checked but never set
to TRUE anywhere any more.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3314>
With non-serialized sticky events, such as GST_EVENT_INSTANT_RATE, we both want
to store the event (for later re-linking) *AND* push the event in a non-blocking
way.
We therefore must *not* propagate pending sticky events if the event is "sticky
or serialized" but only if it's "serialized"
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3254>
- Make the srt_epoll_wait loops more uniform.
- Error only via GError when possible; let the element send the error
message. Avoids a second error message.
- Return 0 when cancelled. Avoids an error message from the element.
- Don't send an error message from send_headers when we're a server
sink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3156>
When using the child proxy notation (child::property=value) it may
happen that the target child does not exist at the time of parsing
(i.e: decodebin creates the encoder according to the contents of the
stream). On this cases, we want to delay the setting of the property
to later, when new elements are added. Previous logic performed a
delayed set even if the target child was found but the property
was not found in it. This should be treated as a failure because,
unlike missing elements, properties should not appear dynamically.
By not failing, typos in property names may go unnoticed to the end
user.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2908>
These null checkes are slightly misleading when double-checking
mutability for external language interop. None of the functions in
these files allow the variable at hand to become `NULL` under normal
operation, because they are checked at initialization and never (allowed
to be) reassigned to `NULL`.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1615>
When matching segments across playlists with Program-Date-Times,
use the difference in segment PDTs to adjust the stream time
that's being transferred. This can fix cases where the
segment boundaries don't align across different streams
and the first download gets thrown away once the PTS
is seen and found not to match.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3309>
Check whether the init file / MAP data for a segment
is different to the current data and trigger an
update if so. Previously, the header would only
be checked in HLS after switching bitrate or
after a seek / first download.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3307>
Previously the minimum buffering threshold was hardcoded to a specific
value (10s). This is suboptimal this an actual value will depend on the actual
stream being played.
This commit sets the low watermark threshold in time to 0, which is an automatic
mode. Subclasses can provide a stream `recommended_buffering_threshold` when
update_stream_info() is called.
Currently implemented for HLS, where we recommended 1.5 average segment
duration. This will result in buffering being at 100% when the 2nd segment has
been downloaded (minus a bit already being consumed downstream)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3240>
This is an additional quality parameter. In the default configuration this
quality switch is deactivated because it would cause a workload increase
which might be significant. If workload is not an issue in the application
it can be recommended to activate this feature.
A flush request is done when set_format is called to empty internal bit
buffer maintained by fdk-aac. When this happens, during the explicit
call to handle_buffer, decodeFrame does not return a AAC_DEC_OK. This
gets reported as a decoding error while no decoding error in fact took
place. Since this can be confusing, just return a GST_FLOW_OK and log
that an explicit flush was requested.
In order to figure out if the "raw" audio contained within the wav
container is actually DTS, right now we call the typefinder helper
which runs all typefinders.
Speed up this type finding process by specifying the extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3294>
GST_TRACERS="leaks" GST_DEBUG="GST_TRACER:7,leaks:6" gst-play-1.0 --use-playbin3 test.mkv
When running a pipeline like above, leaks are observed.
0:00:56.882419132 240637 0x5562c528ccc0 TRACE GST_TRACER :0:: object-alive, type-name=(string)GstConcatPad, address=(gpointer)0x7efd7c0d20a0, description=(string)<'':sink_0>, ref-count=(uint)1, trace=(string);
0:00:56.882429131 240637 0x5562c528ccc0 TRACE GST_TRACER :0:: object-alive, type-name=(string)GstConcatPad, address=(gpointer)0x7efd7c0d2be0, description=(string)<'':sink_0>, ref-count=(uint)1, trace=(string);
0:00:56.882437056 240637 0x5562c528ccc0 TRACE GST_TRACER :0:: object-alive, type-name=(string)GstConcatPad, address=(gpointer)0x7efd7c0d3720, description=(string)<'':sink_0>, ref-count=(uint)1, trace=(string);
gst_element_release_request_pad does not unref the pad. It needs to
be followed by gst_object_unref. Doing that fixes the above leaks.
Use g_ptr_array_new_with_free_func with gst_object_unref as the free
function to unref the pad after release.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3177>
Posting latency messages causes a full and potentially expensive latency
recalculation of the pipeline. While subclasses should check whether the latency
really changed or not before calling this function, we ensure that we do not
post such messages if it didn't change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3282>
The picture parameter picture->top_field_first is reused in this mode
to signal the TOP fields. As a side effect, it will change every frame
and current code assumed that if this changes then a renegotiation is
needed. Fixed this by ignoring that change whenever we are decoding one field
only.
Fixes#1523
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3276>
In fact, all the h264 bit writer have byte aligned output except
the slice header. So we change the API from bit size in unit to
byte size, which is easy to use. For slice header, we add a extra
"trail_bits_num" to return the unaligned bits number.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3193>
We use va pool as msdkvpp's bufferpool, which means both va memory
and dma memory will be allocated by va pool. Considering drm modifier
stuff is not ready, we use va memory with higher priortiry than
dma memory when deciding vpp caps.
Besides, this patch removes the specified "interlace-mode" in vpp caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3253>
The gap handling was in place, but there was no event handler to trigger it.
Implement the alpha sink event handler for the gaps. This fixes handling of
valid streams which may not refresh the alpha frames for every video frames.
It will also allow a clean error if the stream was missing the initial
alpha frame, at least until we find a better way to handle these
invalid frames.
Related to #1518
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3264>
Handle when encoder doesn't support rate control, which is set as
VA_RC_NONE, and if the set rate control mode is not supported by the
GStreamer element, the element configuration fails.
Also it logs out max and target bitrate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3063>
The entrypoint is set when the encoder helper is constructed,
nonetheless it was also passed as parameter when opening. That's
buggy.
In order to simplify the code, the entrypoint at construction is
honored.
But gst_va_encoder_has_profile_and_entrypoint() now doesn't rely in
the internal list of profiles since it only contains those that
belongs to codec and entrypoint, thus it queries directly the VA
driver.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3063>
Need to put the actual profile in the output caps otherwise any
capsfilter after the encoder that was used to force the output
profile will fail, such as
fdkaacenc ! audio/mpeg,stream-format=adts,profile=he-aac-v1 ! ..
because we put profile=lc in there to match the profile signaled
in the ADTS header. This is expressed through the base-profile=lc
in the GStreamer caps though, the profile needs to carry the
'real' profile.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1785>
duplicate symbol '__invoke_on_main' in:
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/libgstvulkan-1.0.a(cocoa_gstvkwindow_cocoa.m.o)
/Library/Frameworks/GStreamer.framework/Versions/1.0/lib/libgstgl-1.0.a(cocoa_gstglwindow_cocoa.m.o)
ld: 1 duplicate symbol for architecture x86_64
clang: error: linker command failed with exit code 1 (use -v to see invocation)
Also make the same change in iOS for consistency.
Continuation of https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1132
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3242>
segment events by demuxer.
In order to play nicely with `ffmpeg`, demuxers in `gst-libav` have to make
buffers available to `ffmpeg` while taking the blocking I/O model in `ffmpeg`
into account, which results in buffers not being sent downstream until `ffmpeg`
has processed them in its separate thread.
In constrast, many `gstreamer` events are simply forwarded downstream.
Currently `GST_EVENT_SEGMENT` events are forwarded downstream without any
processing, which can potentially result in:
* `GST_EVENT_SEGMENT` events being out of sync with buffers
* `GST_EVENT_SEGMENT` events going out that are incorrect because they apply
to data seen by the demuxer, but not necessarily seen by downstream elements
I came across this bug when I was attempting to enable G723.1 demuxing/decoding
using the G723.1 demuxer and decoder provided by `ffmpeg`. I wrote tests to
verify support for the functionality, and found that, in push mode,
`GST_EVENT_SEGMENT` events pushed to the demuxer by the upstream `filesrc`
element would be forwarded to the decoder without modification, resulting in
an internal data streaming error. With this patch, tests work in both push and
pull mode.
This patch solves the problem by disabling the forwarding of
`GST_EVENT_SEGMENT` events downstream (an initial `GST_EVENT_SEGMENT` event is
still pushed downstream by the demuxer). It's possible there's a better way to
do this, but, having looked at how a few different `gstreamer` demuxers deal
with `GST_EVENT_SEGMENT` events, it seems like the processing is somewhat
specific to the demuxer implementation, whereas `gst-libav` has one general way
of handling the situation for any `ffmpeg` demuxer. Perhaps there's a better
way to solve this using the `ffmpeg` API to take advantage of specific demuxer
details. IDK.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3218>
Add Windows Graphics Capture (WGC) API based screen capture mode.
The conditions where this mode is used:
* Explicitly requested by user (capture-api property)
* To capture specific window
* When DXGI desktop duplication API does not work on hybrid graphics systems
(e.g., multi-gpu laptop)
Full features of this implementation require Windows 11. And Windows 11
SDK is required to build this feature.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3144>
When the output alignment is smaller than the input alignment, for
example, When the output alignment is "FRAME" and the parse is likely
connecting to a decoder, the current PTS setting for AV1 frames inside
a TU is not very correct.
For example, a TU may begin with non-displayed frames and end with a
displayed frame. The current way will assign the PTS to the first
non-displayed frame, which is a decode-only frame and the PTS will be
discarded in the video decoder. While the last displayed frame has
invalid PTS, and so the video decoder needs to guess its PTS based on
the frame rate and previous frame's PTS. This is not a decent and
robust way. And more important, when the previous frames provide DTS,
the video decoder will also guess the PTS based on the previous frames'
DTS and trigger the warning like:
gstvideodecoder.c:3147:gst_video_decoder_prepare_finish_frame: \
<vavp9dec0> decreasing timestame
It sets the reordered_output and makes the decoder in free run mode.
We should correct the PTS for a TU, let the non-displayed frames have
no PTS while set the correct PTS to the displayed one. Also, when the
AV1 stream has multi spatial layers, there are more than one displayed
frames inside one TU with the same PTS.
Note: If the input alignment is not TU aligned, we can not know the
exact PTS of this TU, and so we just clear the PTS of the decode only
frame and leave others unchanged.
We also correct all the PTS if the output is OBU aligned. All their
PTS and DTS are set to the input buffer's PTS.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3182>
When the incoming data has big alignment than the output, we do not need to
call finish_frame() and exit the current handle_frame() for each splitted
frame. We can push them all at one shot with in one handle_frame(), whcih
may improve the performance and can help us to find the edge of TU.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3182>
These tests relied on setting the name of an element twice to verify
that the last one set took precedence, however name is a CONSTRUCT property
and the parser now errors out when such properties are set twice, in
g_object_new_with_properties .
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3026>
When trying to build the plugin, GCC starts complaining about issues
with one of the cdparanoia headers and it block us from being able
to build the plugin with Werror.
The current warning in the header look like this:
```
[1/2] Compiling C object subprojects/gst-plugins-base/ext/cdparanoia/libgstcdparanoia.so.p/gstcdparanoiasrc.c.o
In file included from ../subprojects/gst-plugins-base/ext/cdparanoia/gstcdparanoiasrc.h:37,
from ../subprojects/gst-plugins-base/ext/cdparanoia/gstcdparanoiasrc.c:31:
/usr/include/cdda/cdda_interface.h:164:3: warning: initialization discards ‘const’ qualifier from pointer target type [-Wdiscarded-qualifiers]
164 | "Success",
| ^~~~~~~~~
...
/usr/include/cdda/cdda_interface.h:163:14: warning: ‘strerror_tr’ defined but not used [-Wunused-variable]
163 | static char *strerror_tr[]={
| ^~~~~~~~~~~
[2/2] Linking target subprojects/gst-plugins-base/ext/cdparanoia/libgstcdparanoia.so
```
Last release of cdparanoia was in 2008, so our best bet for the
time is to ignore the warnings.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2722>
This allows correct handling of wrapping around backwards during the
first wraparound period and avoids the infamous "Cannot unwrap, any
wrapping took place yet" error message.
It allows makes sure that for actual timestamp jumps a valid value is
returned instead of 0, which then allows the caller to handle it
properly. Not having this can have the caller see the same timestamp (0)
for a very long time, which for example can cause rtpjitterbuffer to
output the same timestamp for a very long time.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1500
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3202>
Adding loopback capture mode for specified PID.
Note that this feature requires Windows 10 build 20348
(Windows 11/Windows Server 2022 or later),
and any process loopback related properties will not be exposed
if OS does not support it.
Example launch lines:
* wasapi2src loopback-mode=include-process-tree loopback-target-pid=<PID>
Captures audio generated by an application (specified by PID)
and its child process
* wasapi2src loopback-mode=exclude-process-tree loopback-target-pid=<PID>
Captures desktop audio excluding PID and its child process
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1278
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3195>
If there is an error while connecting, the streaming task will be stopped, and
is_running() will be false, causing a GST_FLOW_FLUSHING to be returned. Instead,
we perform the error check (!self->connection) first, to return an error if
that's what occured.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3189>
We've seen occasional crashes in the `wavparse` module associated with
referencing a buffer in `gst_wavparse_chain` that's already been freed. The
reference is stolen when the buffer is transferred to the adapter with
`gst_adapter_push` and, IIUC, assuming the source doesn't hold a reference to
the buffer, the buffer could be freed during interaction with the adapter in
`gst_wavparse_stream_headers`.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3179>
in certain ways.
In the case that a test is provided for, the size of the `fmt ` chunk is
changed from 16 bytes to 18 bytes (bytes 17 - 20 below):
```
$ hexdump -C corruptheadertestsrc.wav
00000000 52 49 46 46 e4 fd 00 00 57 41 56 45 66 6d 74 20 |RIFF....WAVEfmt |
00000010 12 00 00 00 01 00 01 00 80 3e 00 00 00 7d 00 00 |.........>...}..|
00000020 02 00 10 00 64 61 74 61 |....data|
00000028
```
(Note that the original file is much larger. This was the smallest sub-file
I could find that would generate the crash.)
Note that, while the same issue doesn't cause a crash in pull mode, there's a
different issue in that the file is processed successfully as if it was a .wav
file with zero samples.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3173>
When the alignment is "FRAME" and the parse is likely connecting to
a decoder, the current PTS setting for VP9 frames inside a super
frame is not very correct.
For example, the super frame may begin with non-displayed frames and
end with a displayed frame. The current way will assign the PTS to
the first non-displayed frame, which is a decode-only frame and the
PTS will be discarded in the video decoder. While the last displayed
frame has invalid PTS, and so the video decoder needs to guess its
PTS based on the frame rate and previous frame's PTS. This is not a
decent and robust way. And more important, when the previous frames
provide DTS, the video decoder will also guess the PTS based on the
previous frames' DTS and trigger the warning like:
gstvideodecoder.c:3147:gst_video_decoder_prepare_finish_frame: \
<vavp9dec0> decreasing timestame
It sets the reordered_output and makes the decoder in free run mode.
We should correct the PTS for a super frame, let the non-displayed
frames have no PTS while set the correct PTS to the displayed one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3155>
The scenario is what we try in the tests:
- we have a segment with .stop set
- some frame(s) flow
- we get a CAPS event
- we get an EOS (before getting buffers after the CAPS event)
in that case, without that patch, the segment is not properly closed
which is not correct. In this patch we keep track of previous caps until
a new buffer arrives, this way in that situation we set previous caps
again, and close the segment with the previous buffer.
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1352
in this specific case
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3059>
Apparently we cannot start sending messages from another datachannel
before the previous message was completely sent. usrsctplib will
complain about being locked on another stream id and set
errno=EINVAL.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2454>
The implementation was inconsistent between create and destroy. EGLImage
creation and destruction is requires for EGL 1.5 and up, while
otherwise the KHR version is only available if EGL_KHR_image_base
feature is set. Not doing these check may lead to getting a function
pointer to a stub, which is notably the case when using apitrace.
Fixes#1389
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2925>
The idea is to give the application the possibility to adjust the error
code when responding to a request. For that purpose the pipeline's bus
messages are emitted to subscribers through a signal handle-message.
The subscribers can then check those messages for errors and adjust
the response error code by overriding the virtual method
adjust_error_code().
Fixes#1294
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2972>
The order of the devices iterator from the SDK is undefined and can
randomly change.
Keep the device-number property for backwards compatibility and
simplicity but prefer the persistent-id property and also use it for the
device provider implementation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3078>
GstDXGIGetDebugInterface() is unused when targeting UWP. We directly
call DXGIGetDebugInterface1() in that case.
Fixes build failure:
../gst-libs/gst/d3d11/gstd3d11device.cpp(271): error C2440: '=': cannot convert from 'HRESULT (__cdecl *)(UINT,const IID &,void **)' to 'DXGIGetDebugInterface_t'
../gst-libs/gst/d3d11/gstd3d11device.cpp(271): note: This conversion requires a reinterpret_cast, a C-style cast or function-style cast
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3118>
According to W3C
specification (https://w3c.github.io/webrtc-pc/#datachannel-send) we
should return InvalidStateError exception when trying to send when the
channel is not open. In the world of C/glib/gstreamer we don't have
exceptions but have to rely on gboolean/GError instead. Introducing
these calls for a change in function signature of the action signals
used to send data on the datachannel. Changing the signature of the
existing "send-string" and "send-data" signals would mean an immediate
breaking change so instead we deprecate them. Furthermore, there is no
way to express GError** as an argument to an action signal in a way
that fits language bindings (pointer-to-pointer simply does not work)
and we have to use regular functions instead.
Therefore we introduce gst_webrtc_data_channel_send_data_full() and
gst_webrtc_data_channel_send_string_full() while deprecating the old
functions and corresponding signals.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1958>
Always hold a reference to the soft volume element
provided by the playsinkaudioconvert bin helper, the
same as when volume is provided by a sink element,
or the soft volume element gets unreffed too soon.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3108>
This is a regression that was introduced in
cca2f555d1 (yes, 9 years ago).
The only place where a demuxer streaming thread should be stopped is when the
sinkpad is deactivated from pull mode (i.e. PAUSED->READY).
Attempting to stop the task in this function would cause this to happen when a
FLUSH_STOP or STREAM_START event is received... which can cause deadlocks.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3109>
It would constantly want to renegotiate (and spam the debug log) even
though the channel layout hasn't actually changed. We use the same
fallback in gst_ffmpegauddec_negotiate() already.
This happens with WMA files for example.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3103>
We need to call this to register the MusixBrainz tags before we use
them in an XMP schema.
Fixes this critical when attempting to run jpegparse on a JPEG
containing MusicBrainz XMP tags:
GStreamer-CRITICAL **: 20:41:07.885: gst_tag_get_type: assertion 'info != NULL' failed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3092>
Currently if the user is not able to access the devices under /dev/media*,
either due to no media devices present on the system or simply no permission
to access the device, v4l2codecs initialises with no features or debug messages.
Since calling `GST_DEBUG="v4l2*:7" gst-inspect-1.0 v4l2codecs` is a typical way
to diagnose why element(s) failed to enumerate, we should be more verbose here
when the user is not able to access any /dev/media* device. So print a simple
debug message in this case to aid debugging.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3088>
The purpose of a deep buffer copy is to be able to release the source
buffer and all its dependencies. Attaching the parent buffer meta to
the newly created deep copy needlessly keeps holding a reference to the
parent buffer.
The issue this solves is the fact you need to allocate more
buffers, as you have free buffers being held for no reason. In the good
cases it will use more memory, in the bad case it will stall your
pipeline (since codecs often need a minimum number of buffers to
actually work).
Fixes#283
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2928>
Since commit a79a756b79 we could change to ignore-pcr automatically at 500ms
into a live stream when no PCR is seen by then. However the stream counting in
program change detection was wrongly considering ignore-pcr programs to have a
separate PCR PID, even though we are actually ignoring the PCR PID completely,
resulting in an erroneous program switch getting triggered from the different
stream count. This in turn would send an EOS and switch out the pads for what
actually is still the same program, while we intended to simply apply a
workaround for broken encoders.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3060>
If the SETUP request returns an IPv6 server address in the Transport
field, we would generate an incorrect URI, and multiudpsink would fail
to initialize:
```
rtspsrc gstrtspsrc.c:9780:dump_key_value:<source> key: 'Transport', value: 'RTP/AVP;unicast;source=fe80::dc27:25ff:fe5e:bd13:8080;client_port=62696-62697;server_port=4000-4001'
...
rtspsrc gstrtspsrc.c:4595:gst_rtspsrc_stream_configure_udp_sinks:<source> configure RTP UDP sink for fe80::dc27:25ff:fe5e:bd13:8080:4000
...
multiudpsink gstmultiudpsink.c:1229:gst_multiudpsink_configure_client:<udpsink0> error: Invalid address family (got 23)
```
We can't look at stream->is_ipv6 because we can't rely on the server
returning the right value there. In the issue reported about this,
server reported itself as `KuP RTSP Server/0.1`, and the SDP was:
```
c=IN IP4
m=video 54608 RTP/AVP 96
a=rtpmap:96 H264/90000
```
So we need to parse the string value and figure out the family
ourselves.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1058
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1819>
Fixes warning with meson 0.62:
gst-plugins-bad| subprojects/gst-plugins-bad/meson.build:546: WARNING:
Project targets '>= 0.62' but uses feature deprecated since '0.62.0':
pkgconfig.generate variable for builtin directories. They will be
automatically included when referenced
and more.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3086>
Update unit test for some mpd cases that were reporting
timestamps including the period start time, while
dashdemux2 expects that it needs to add the period
start time itself.
Fix the tests to not expect the period start time
to be included.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3025>
These values will be referred to as timestamp relative to period start
so need to subtract period start time from the values.
Fixes a problem with determining the start position when playing Live content
with SegmentTimeline, presentationTimeOffset and a non-0 period start time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3025>
Starting with Meson 0.62, meson automatically populates the variables
list in the pkgconfig file if you reference builtin directories in the
pkgconfig file (whether via a custom pkgconfig variable or elsewhere).
We need this, because ${prefix}/libexec is a hard-coded value which is
incorrect on, for example, Debian.
Bump requirement to 0.62, and remove version compares that retained
support for older Meson versions.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1245
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3061>
Change the way streams are woken up to download more data.
Instead of checking the level on tracks that are being
output as data is dequeued, calculate a 'wakeup time'
at which it should download more data, and wake up
the stream when the global output position crosses
that threshold.
For efficiency, compute the earliest wakeup time
for all streams and store it on the period, so the
output loop can quickly check only a single value
to decide if something needs waking up.
Does the same buffering as the previous method,
but ensures that as we approach the end of
one period, the next period continues incrementally
downloading data so that it is fully buffered when
the period starts.
Fixes issues with multi-period VOD content where
download of the second period resumes only after
the first period is completely drained.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3055>
When pushing several buffers while the pipeline is in NULL state, meaning
that the action are executed "interlaced", previous code was deadlocking.
This new implementation makes it so the override is always on and we
expect all buffers to go through to be associated to a function, which
is a safe assumption.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3052>
Some servers can return playlists with "old" media playlists and different
Discont Sequence.
In those cases, the segment stream times would be negative when creating a new
time mapping. In order to properly handle such scenarios, shift the values to
stored accordingly to end up with non-negative reference stream time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3054>
This allows users to let videorate fully fill the segments when received
EOS or on new segment, removing an arbitrary limit of 25 duplicates which
might not be what the user wants (for example on low FPS stream in GES,
that sometimes leaded to broken behavior)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3000>
The AV1 support multi spatial layers within one TU with different
resolutions, and only the highest spatial layer need to be output.
For example, there are two spatial layer, base level is 800x600
and higher level is 1920x1080. We need to decode both because the
higher level needs base layer as reference, but we only need to output
1920x1080 frames here.
The current manner always renegotiates the caps once we detect the
current picture resolution changes, so we renegotiate again and
again between different layers. That's a big waste and has very
low performance. We now only do the renegotiation for the highest
output layer. For other non output layers, we just keep a internal
buffer pool which is big enough to handle the surface allocation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2382>
As SPEC says, when multi spatial layer exists, we should only output
one frame with the highest spatial id from each TU. We now store the
highest spatial layer information in the base class in order to let
the sub class handle different layers easily.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2382>
doesn't align on 20 millisecond frame size.
The AMR-WB codec imposes a fixed 20 millisecond frame size. In its current
form, the `voamrwbenc` plugin deals with this limitation by discarding any
audio at the end of the stream that falls short of 20 milliseconds. This patch
keeps the audio data, and appends silence to the end to preserve frame size
alignment.
The patch also adds tests to check for the updated behavior. I noticed that
tests weren't being built, so I changed the build to allow for building the
tests when the `tests` and `voamrwbenc` options are set.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3027>
when expose-all=False
When trying to find an decoder in that case, we loop over the different
decoder factories, and check that it outputs a format that matches the
requested one (through the :caps property), but if we find a decoder
that do match but later on some other don't we end up failing
autopluging. This patch ensures that we still plug the decoder that can
work.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3011>
- Update the docker image we use, starting using the standard one adding
`gtk4-doc` as required by rust plugins
- Update the plugins_doc_caches as required, some more plugins are built
with the new image
- Install ninja from pip as the version from F31 is too old
- Avoid buildings all GSreamer plugins when building the doc as it takes
time and resources for no good reason
- Stop linking to `GInstanceInitFunc` as it is not present in latest GLib
documentation, leading to warnings in hotdoc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2954>
We are supposed to guarantee that pads that are exposed have the caps
set, but for sources that have pad with "all raw caps" templates, we end
up exposing pads that don't have caps set yet, which can break code (in
GES for example).
To avoid that we let uridecodebin plug a `decodebin` after such pads and
let decodebin to handle that for us. In the end the only thing that
decodebin does in those cases is to wait for pads to be ready and expose
them, after that `uridecodebin` will expose those pads.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3009>
We are always building our printf implementation, even when
GST_DEBUG is disabled, since we are exposing api (gst_print*)
that's dependant on our printf behavior.
We don't need to keep __gst_info_fallback_vasprintf around anymore.
Close#640
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/739>
Handle d3d11 device context in set_context() method with
additional device compatibility check so that only NVIDIA GPU
associated d3d11 device can be configured in the element.
And clear old d3d11 device per set_info() for d3d11 device to be
updated as well.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3018>
... and fix d3d11 specific enum type name
GST_CUDA_HAS_D3D is a build time define which indicates whether
GstD3D11 library is available or not, but DirectX SDK headers
must be available on the build system already.
Expose Direct3D related symbols if the build target is Windows
(i.e., if G_OS_WIN32 is defined)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3018>
GLib made the unfortunate decision to prevent libgobject from ever being
unloaded, which means that now any library which registers a static type
can't ever be unloaded either (and any library that depends on those,
ad nauseam).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/778>
GstVA is not currently build by CI, because libva version is lower
than expected. So, the gstva library is not build, thus some symbols
aren't documented, breaking the documentation CI.
To move things forward, let's just remove temporarly the va plugins
from cache. While we decide on how to update the libva package in
the CI.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1025>
When advancing fragment in live, it's normal to return
GST_FLOW_EOS when playing at the live edge of the available
fragments. In that case, we still want to adjust bitrate
dynamically.
Fixes issue with dashdemux2 where the current bitrate of
each adaptation set is changed to the lowest one when
updating the mpd for a live stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3020>
Instead of trying to hardcode site-packages paths for different platforms
just use python.get_install_dir() from meson and let it deal with the rest.
Also no longer try to import pygobject, which would otherwise not be
required at build time.
python.get_install_dir() was at the beginning broken on Windows, but
that was fixed in 0.60 via https://github.com/mesonbuild/meson/pull/9156
and since ges now requires >0.60 this can be ignored.
This change was motivated by the install path being wrong under MSYS2, where
the unix install layout is used and the detection code not taking that into
account.
This MR is a continuation of https://gitlab.freedesktop.org/gstreamer/gst-editing-services/-/merge_requests/230
see the discussion there for extra context.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3012>
Just like for the seconds field, there are no limitations on the hours and
minutes fields. The specification for xml schema duration fields doesn't forbid
specifying durations with only (huge) minutes or hours values.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2951>
When picking an available payload type, we need to pick one that is
available across all media.
The previous code, when multiple media were present, looked at the first one,
noticed it had pt 96 as the media pt, then simply looked at the next media,
noticed it didn't, and decided 96 was available.
Instead, check if the pt is used by any of the media, if it is, decide
it is not available and go to the next pt. I'm fairly sure that was the
original intent.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2984>
When updating a manifest during live playback, preserve the current
representation for each stream.
During update_fragment_info, if the current representation changed
because it couldn't be matched, trigger a caps change and new
header download.
This reverts commit e0e1db212f
and reapplies "dashdemux: Fix issue when manifest update sets slow start
without passing necessary header & caps changes downstream" with
changes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2920>
Timers for RTX packets are dealt with later in update_rtx_timers(), and
timers for non-RTX packets would potentially also be unscheduled a
second time from there so avoid that.
Also don't shadow the timer variable from the outer scope but instead
make use of it directly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2973>
Check back pressure of a stream transport before popping buffer from its backlog.
If the stream transport is not experiencing back pressure, the buffer can be popped from backlog and pushed to client.
Fixes:#1298
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2936>
The address/port is pre-defined by the caller of the function, so
retrying is only going to loop forever.
Ideally the multicast address should be checked after allocating but
this doesn't happen currently, so it's better to error out cleanly then
to loop forever trying the same address.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2975>
This avoids getting in a bunch of corner cases. We'd have to insert
a "rejected" line from the start as a place-holder to get around this,
but the rest of the code just becomes more complicated, so just
disallow it for now.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2439>
If the buffer is not msdk_buffer, we can try to directly import the
attached memory (i.e. va mem and dmabuf mem) by applying the common
uitl function: import_to_msdk_function ().
Here add a flag "from_qdata" in GstMsdkSurface to handle the cropping case,
we should avoid updating the crop values when msdk_surface is from the
memory's qdata, because the crop info from this surface is the already
updated one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2498>
When input buffer is of dmabuf memory but not a msdk buffer (i.e., the
allocator is not msdk_allocator), then we can try to get fd of this mem,
create the corresponding va surface and wrap it as mfx surface.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2498>
If read_one or write_one was called but the stream closed before it could
read/write a whole packet, read_one/write_one would hang indefinitely,
consuming 100% CPU. This commit fixes that by treating a short read/write
as an error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2964>
We were checking possible bind flags for the DXGI format
of the source texture but that's never applied to
the destination texture desc.
Just use the already configured bind (and misc) flags of source texture
for the destination texture allocation without additional check.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2950>
Setting the content-type property shall override internally detected MIME
types, to make it possible to do as following example (where audio/basic to be
used prior to audio/x-mulaw):
gst-launch-1.0 ... ! mulawenc ! audio/x-mulaw,rate=8000,channels=1 !
curlhttpsink location=<url> content-type=audio/basic
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2732>
* Private header name is changed to gstd3d11-private.h to follow
naming convention
* Add Since mark everywhere
* Update member variable names to be consistent with the other
object implementations in this library
* Correct outdated documentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2945>
Theoretically having elements in locked state should not have any effect
at all when the surrounding bin is doing state changes. However
previously a state change error of a locked element would cause the
bin's state change to also fail, which is clearly not intended.
State change failures of locked elements are to be handled by whoever
set the element to locked state. By always returning them here it is
impossible for the owner of the element to handle state change failures
gracefully without potentially affecting the whole pipeline's state
changes.
Non-failure returns are still returned as-is as the distinction between
ASYNC/NO_PREROLL/SUCCESS has big consequences on the state changes of
the bin and overall pipeline. Theoretically SUCCESS should also be
returned in all cases but I can't estimate the effects this would have
on the overall pipeline.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2932>
Current default G_MAXINT is not a correct value under any circumstances.
This creates an issue with screen capture, during which we currently do
not get any framerate info causing G_MAXINT to show up, where elements
downstream can possibly misbehave - for example, `vtenc` causes
a kernel panic.
Replace with 30/1 to avoid such scenarios.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2944>
The current handle_frame() does not return the real error that happens
in decode_scan and decode_frame, which makes the pipeline continue with
the error and may trigger asserting later.
We also return the error when decode_quant_table or decode_huffman_table
fails.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2938>
Add an example to show the usage of present singal.
In this example, a text overlay with alpha blended background
will be rendered on swapchain's backbuffer by using
Direct3D11, Direct2D, and DirectWrite APIs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2923>
The "present" signal will be emitted just before the
IDXGISwapChain::Present() call. The client can perform additional
GPU operation with given GstD3D11Device object and
ID3D11RenderTargetView handle. Or, the client can read back
the scene to be displayed on window using the signal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2923>
libsoup 3.0.x dispatches using a single source attached when the session
is created, so we need to create the session with the same context that
our download thread is later using.
2.74 or 3.1 will dispatch a response using the context which sent the
request. However, for any context other than the one that created the
session, this will also create and destroy sources, so there's still
some slight performance benefit.
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1384
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2913>
This allows an application to provide their own opened DRM device
fd handle to kmssink. For example, an application can lease
multiple fd's from a DRM master to display on different CRTC
outputs at the same time with multiple kmssink instances.
Specifying the fd property is not allowed when driver-name
and/or bus-id properties are specified.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2807>
Handle select-streams and seek events in an element
level send_event() vfunc, so they can be received
before any source pads are created.
This allows preferred streams to be selected before
segment downloading starts.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2912>
Nouveau driver currently only exposes postproc entry. But
vaapidecodebin is registered independent if there are decoders or not,
exposing a segmentation fault.
This patch removes the encoder/decoder/codec arrays if no entries are
found, and if no decoders are found vaapidecodebin is not
registered. Also for vaapipostproc if no postproc entry is found.
Also, if general decoder, used by vaapidecodebin, doesn't have a sink
pad string, don't register the glib type.
Fixes: #1349
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2865>
Without this change cleanup function for g_autoptr is not defined for
GstPlayMediaInfo, GstPlaySignalAdapter, GstPlayVideoRenderer,
GstPlayVideoOverlayVideoRenderer and GstPlayVisualization. Cleanup
function was defined in gstplay.h, but missing in other header files.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2888>
When stopping the element, make sure the pad task
is stopped before destroying the part readers.
Closes a race where the pad task might access
a freed pointer.
Also add a guard against this sort of thing
by holding a ref to the reader in the pad loop.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2901>
When playing live, it's possible that one stream reaches
the end of the available playback window and goes to sleep
waiting for a manifest update, and the manifest update
introduces a new period. In that case, the sleeping
stream needs to wake up and go 'properly' EOS before we
can advance the input to the new period.
Accordingly, make sure that a stream's last_ret value
is not marked as EOS if it's just sleeping waiting for a live
manifest update.
Also fix the output loop to go back and re-check if it's
time to switch to the next period after dequeuing and
discarding an EOS event.
https://livesim.dashif.org/livesim/periods_20/testpic_2s/Manifest.mpd
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2895>
The parent refcount is of the *transformed* buffer, not the input
buffer.
Also update the docs to clarify that @transbuf is the transformed
buffer, and not the buffer on which a transformation is being
performed.
Due to this bug, modifying the structure of a meta that has been
copied to another buffer fails with:
gst_structure_set: assertion 'IS_MUTABLE (structure) || field == NULL' failed
Add a test for the same.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2890>
Newer compilers ( clang 15 ) have turned stricter and errors out instead
of warning on implicit function declations
Fixes
gstssaparse.c:297:12: error: call to undeclared library function 'isspace' with type 'int (int)'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration]
while (isspace(*t))
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2879>
Ideally new() functions should simply call g_object_new() and not much
else, so let's do that here and handle all the construction properly in
a GObject way.
Now a play object created via g_object_new() is actually usable.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2880>
Ideally new() functions should simply call g_object_new() and not much
else, so let's do that here and handle all the construction properly in
a GObject way.
Now a player object created via g_object_new() is actually usable.
In addition, also fix the video-renderer property so that reading it
returns an object of the correct type.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2880>
That is, get rid of unnecessary and wrong special-casing.
This could always use gst_rtsp_url_get_request_uri_with_control() but as
we only have the control base URI as string it is easier to just call
gst_uri_join_strings().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2868>
This was showing up as a memory leak in GTK's
gstreamer media backend:
40 bytes in 1 blocks are definitely lost in loss record 18,487 of 40,868
at 0x484586F: malloc (vg_replace_malloc.c:381)
by 0x50D5278: g_malloc (gmem.c:125)
by 0x50EDBA5: g_slice_alloc (gslice.c:1072)
by 0x50EFBCC: g_slice_alloc0 (gslice.c:1098)
by 0x51F2F45: g_type_create_instance (gtype.c:1911)
by 0x51DAE37: g_object_new_internal (gobject.c:2011)
by 0x51DC080: g_object_new_with_properties (gobject.c:2181)
by 0x51DCB20: g_object_new (gobject.c:1821)
by 0x9855F86: UnknownInlinedFun (gstplayer-wrapped-video-renderer.c:109)
by 0x9855F86: gst_player_new (gstplayer.c:579)
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1374
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2875>
Otherwise we won't send the protection packets for the last few
packets when a stream ends.
Also send EOS on the FEC src row pad immediately, and on the FEC src
column pad after draining is complete. This makes it so that the FEC
src pads on rtpbin behave the same way as the RTCP src pads on rtpbin
when EOS is received on the send_rtp_sink pad.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2863>
When a new segment event arrives, it immediately updates
the current stored segment, which was used for calculating
the running time of the current text buffer for every
passing video frame. This means a segment that arrives
after the text buffer might get used to (mis)calculate
the running times subsequently.
Instead, calculate and store the right running time
using the current segment when storing the buffer. Later
the stored segment can get freely updated.
This fixes the case where pieces of video and text streams
are seamlessly concatenated and fed through the text overlay.
Previously, it could lead to the current text buffer suddenly
have a massive running time and blocking all further input.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2802>
Radeon mesa gallium driver has a bug which adds P010_10LE sink caps
format. This patch removes formats which arent 420 chroma.
gst_caps_set_format_array() wasn't used because the fix traverse
several structures with potential different formats.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2844>
When returning GST_ADAPTIVE_DEMUX_FLOW_RESTART_FRAGMENT
for the first segment data, we might need to requeue the
header.
This was leading to occasional prerolling stalls on
HLS live streams with renditions.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2849>
Make sure gst_adaptive_demux_loop_cancel_call()
never tries to operate on an invalidated main context. Make
sure to clear the main context pointer while holding the lock,
and to check it in gst_adaptive_demux_loop_cancel_call()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2847>
GLib's GRecMutex will allocate another heap memory for CRITICAL_SECTION
struct and g_rec_mutex_lock/g_rec_mutex_unlock use WIN32 APIs actually.
We don't need such intermediate function calls and redundant heap allocation.
Just call WIN32 APIs directly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2845>
Media playlist updates and fragment downloads happen in an interleaved
fashion. When a media playlist update fails *while* a segment is being
downloaded, this means we lost synchronization.
Properly propagate and handle this
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2839>
There is now only a single case where we setup the initial playlist to 0, which
is for the very first variant stream.
Rendition streams will have the initial playlist "synchronized" against the
variant stream media playlist.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2839>
Loss of synchronization happens when the updated media playlist has no
relationship to the previous ones. This could happen because of network issues,
server issues, etc...
When this happens, we take no chance and "reset" ourselves so that we can "seek
back to live" against the new updated playlists.
Since this happens at the "media playlist update" level, make sure the custom
flow return is propagated up.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2839>
We are already in the main scheduler thread, therefore we can do the "seek back
to live" directly. This also avoids other pending actions to take place.
Also handle the loss of sync when doing manifest updates.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2839>
Close some race conditions in switching to the next period,
by ensuring the tracks are completely drained first and by
not outputting EOS events to the output source pad
if there is another period pending.
Fixes Manifest_MultiPeriod_1080p.mpd some more.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2838>
Before sending EOS, update the period's has_next_period
flag and/or create the next period. This closes a race
where the output loop might receive the EOS event
and either push it downstream (causing premature EOS),
or receive it and try and switch to the next period
before that period is completely set up.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2838>
When combining stream flows, ignore streams that
are not selected, instead of checking whether
the stream state has changed yet.
Fixes another issue with dashdemux2 where it fails to
change to the next period when playing content with
several video, audio and text streams, as with
Manifest_MultiPeriod_1080p.mpd when seeking to 730
just before the end of the first period.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2838>
The is_gst_mini_object_check would sometimes detect a proper GObject
as a mini object, and then bad things happen.
We know whether a pointer is a proper GObject or a MiniObject here
though, so just pass that information to the right code paths and
avoid the heuristics altogether.
Eliminates all remaining uses of object_is_gst_mini_object().
Fixes#1334
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2832>
The is_gst_mini_object_check would sometimes detect a proper GObject
as a mini object, and then bad things happen.
We know whether a pointer is a proper GObject or a MiniObject here
though, so just pass that information to the right code paths and
avoid the heuristics altogether.
There are probably more cases where the check should be eliminated.
Fixes#1334, maybe
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2832>
This is based on gtksink, but similar to waylandsink uses Wayland APIs
directly instead of rendering with Gtk/Cairo primitives.
Note that the long term plan is to move this into the existing extension
in `-good`, which requires the Wayland library to move the as well.
For this reason several files like `gstgtkutils.*` and `gtkgstbasewidget.*`
are straight copies and should be kept in sync.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1515>
This can be important for instance when a container holds multiple
tracks with the same media type, with no indication (eg tags) of
which track is the default one.
In that case, players usually pick the first track by default.
This is especially useful when using smart editing with GES, as
it will result in the same ordering as the input file that was
used as a template.
For reference, this yields the same order as ffprobe.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1549>
The previous code was storing container children in reverse
addition order, this was mitigated by the fact that track elements
were also stored in reverse order, thus restoring the original
order, but it seems more consistent to preserve order throughout,
the extra cost of append operations is negligible.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1549>
when creating a profile from a discoverer info.
There is no justification for the existing code, and talking with
Thibault he cannot remember why the sort was in place.
On the other hand, this allows GES users to not have to implement
a callback for the select-tracks-for-object callback when using
it to trim a single clip, which the output profile was built from:
track elements will be placed in the appropriate track by default,
that is the one that will be connected to the matching profile.
For multi-clip timelines, the situation doesn't change, users will
still have to implement a callback and do the leg work of placing
track elements (if any) in a matching track (if any).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1549>
chroma-format, bit-depth-chroma, bit-depth-luma are all informative
fields set by the H265 and H265 parser upon receiving an SPS.
They shouldn't be constrained downstream of the parser, instead
if a user wants those to ultimately match certain values they
should do so by constraining a profile.
In this case however, we also always remove the profile constraint
in order to let encoders pick a suitable one as a function of the
raw input video format and their own capabilities.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1549>
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>
Forgot to change the wrap type in e0014ef4fe which broke the
subproject. Wasn't noticed by CI because the subproject cache wasn't
regenerated.
The accompanied patch was included in 2.8.2, so it is not needed. It
was originally needed with 2.8.1
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2812>
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>
1.21.0.1 should not satisfy a check for 1.22.0.
If someone needs more control they should do a feature check for
the symbol in the headers or lib.
Based on a similar patch by Tim-Philipp Müller for libnice.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2501>
The caps negotiation should respect the selected method to the test pipeline below works properly.
gst-launch-1.0 videotestsrc ! video/x-raw,width=320,height=600 ! videoflip method=clockwise ! video/x-raw,width=600,height=320 ! fakesink
Signed-off-by: Adrian Fiergolski <adrian.fiergolski@fastree3d.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2803>
In the trick mode, driver may queue a valid buffer follow by an
empty buffer which has no valid data to indicate EOS.For the empty
buffer whose memory is multi-plane, need to resize it before
unreference it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2731>
Depending on device feature level, d3d11 runtime can support
ID3D11Fence which is equivalent to ID3D12Fence.
Waiting using fence has performance-wise benefit over pulling
ID3D11Query status. If ID3D11Fence is not supported by device,
then ID3D11Query will be used instead.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2790>
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>
It may happens that bitstream doesn't provided SPS in decoding order
(like in VPSSPSPPS_A_MainConcept_1 conformance test file).
To be sure that the decoder got the correct SPS parameters process
SPS just before start decoding the frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2575>
