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>
