When checking for renegotiation against a local offer,
reverse the remote direction in the corresponding answer
to fix falsely not triggering on-negotiation needed when
switching (for example) from local sendrecv -> recvonly
against a peer that answered 'recvonly'.
In the other direction, when the local was the answerer,
renegotiation might trigger when it didn't need to -
whenever the local transceiver direction differs from
the intersected direction we chose. Instead what we want
is to check if the intersected direction we would now
choose differs from what was previously chosen.
This makes the behaviour in both cases match the
behaviour described in
https://www.w3.org/TR/webrtc/#dfn-check-if-negotiation-is-needed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7303>
In order to use oes-external, the qml6glsink needs a fragment shader that uses
the samplerExternalOES.
The qsb tool is not able to handle shaders that contain samplerExternalOES since
this feature is not supported by all target shading languages. The qsb tool is
able to replace a shader in the qsb file to handle this use case. Use it to
generate a shader variant that uses samplerExternalOES for OpenGL ES and select
that variant if the qml6glsink negotiated texture target oes-external.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7319>
Fixes for basic rollback (from have-local-offer or have-remote-offer to
stable). Allow having no SDP attached to the webrtc session description
in that case, and avoid all the transceiver and ICE update logic
normally applied when entering the stable signalling state
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7304>
release_frame() can be useful for manually dropping frames without posting QoS messages like finish_frame() would.
Matches the same kind of API on the decoder side of things.
Modifies the behaviour of release_frame() to make sure events from released frames are stored as 'pending'
and pushed before the next non-dropped frame. This is needed because now release_frame() can be called outside of
finish_frame(), so we would potentially just lose events and bad things would happen.
drop_frame() was also added to match the decoder API. It functions almost identically to finish_frame() without a buffer
attached to the frame, except instead of immediately pushing the frame's events, it will store them as pending.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7190>
In case when conn->input_stream is NULL and glib was built with
"glib_checks" enabled, g_pollable_input_stream_read_nonblocking()
returns -1, but does not set the "err".
The call stack:
read_bytes() ->
fill_bytes() ->
fill_raw_bytes()
The return value -1 passed up to read_bytes() and incorrectly
processed there after "error:" label.
This changes the return value to EINVAL.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7210>
Fix an inverted condition when checking if sink pad caps match
the codec-preference of an unassociated transceiver, and
fix a condition check for transceiver media kind to
avoid matching sinkpad requests where caps aren't provided
against unassociated transceivers where the caps might
not match later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7237>
According to the ffmpeg documentation[1] the read_packet function should never
return 0. ffmpegdata_peek returns 0 when the stream is EOF causing us to fail
detecting EOF and never close the pipeline, continually spinning on more data.
ffmpeg instead wants an AVERROR_EOF code for to signal EOF.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4999>
With MR 7156, transceivers and transports are created earlier,
but for sendrecv media we could get `not-linked` errors due to
transportreceivebin not being connected to rtpbin yet when incoming
data arrives.
This condition wasn't being tested in elements_webrtcbin, but could be
reproduced in the webrtcbidirectional example. This commit now also
adds a test for this, so that this doesn't regress anymore.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7294>
A previous fix, a275e1e029, is correct but was too
permissive since it treats all un-matched NAL units the same as AU delimiters
even though some other NAL unit types can be encountered in the processing loop.
The problem this can cause is that some hardware decoders experience bad
performance when handling FD units that precede the SPS.
This change restores the original behavior for FDs so that they're ignored until
the SPS is received and it preserves the codec conformance test gains that the
fix has achieved.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7166>
glCheckFrameStatus() can fail by returning 0, and otherwise return a
status. Fix the trace to make it clear when we get an unkown status
compare to having an error, in which case we also trace the error code.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7291>
Parts may emit bus messages that want to take the splitmuxsrc
lock and prevent the downward state change. Avoid a deadlock
after a part sends an error message by taking a ref and
dropping the lock around the unprepare call
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
Publish fragment-id in the messages that splitmuxsink and splitmuxsrc
send, so when they are received out of order (due to async finalization,
for example), they can still be identified / ordered correctly.
Fix a race in the splitmuxsink unit test where messages might be
received out of order
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
Add a `num-lookahead` property that will 'prepare' a number of
fragments in advance of the playhead if they have been deactivated
or closed by a limited number of `num-open-fragments`. It can help
to avoid any play stalls reading the indexes or headers of the next
file from high-latency media or on resource limited machines.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
Publish the playback offset for and duration into the
splitmuxsink-fragment-closed bus message as each fragment
finishes.
These can be passed to splitmuxsrc via the 'add-fragment'
signal to avoid splitmuxsrc measuring all files on startup
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
Add a reasonably large default for the number of simulataneous
files to open, that won't affect users that split recordings into
a few large files, but will help prevent fd exhaustion for users
that make recordings with lots of small fragments
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
When calculating the timestamp offset to apply to
media streams in a fragment, ensure that all fragments
are offset "together" to preserve alignment in cases
where there might gaps in a recording at a fragment boundary.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
Add a signal that allows adding fragments with a specific offset
and duration directly to splitmuxsrc's list. By providing the
fragment's offset on the playback timeline and duration directly,
splitmuxsrc doesn't need to measure the fragment making for faster
startup times.
Add a bus message that's published when fragments are measured,
reporting the offset and duration, so they can be cached by an
application and used on future invocations.
Add examples for handling the bus message and using the 'add-fragment'
signal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
Add a property to limit the number of parts splitmux will open
simultaneously. Modify the part handling to support deactivating
and reactivating the demuxing for each part.
The default is '0', to preserve the existing behaviour of opening
all parts at the beginning.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
According to https://w3c.github.io/webrtc-pc/#set-the-session-description
(steps in 4.6.10.), we should be creating and associating transceivers when
setting session descriptions.
Before this commit, webrtcbin deviated from the spec:
1. Transceivers from sink pads where created when the sink pad was
requested, but not associated after setting local description, only
when signaling is STABLE.
2. Transceivers from remote offers were not created after applying the
the remote description, only when the answer is created, and were then
only associated once signaling is STABLE.
This commit makes webrtcbin follow the spec more closely with regards to
timing of transceivers creation and association.
A unit test is added, checking that the transceivers are created and
associated after every session description is set.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7156>
If a downstream buffer pool is offered, vulkanupload checks its allocation
parameters to honor them. Only adds to usage the TRANSFER bits, which are
required to upload buffers.
Also, fail if the buffer pool cannot be configured with the current parameters.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7219>
If the stream has a special colorimetry that is not in the colorimetry
list, it will cause negotiation to fail. We should allow passing any
colorimetry, so add an extra structure without the colorimetry field.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7029>
video-info supports encoded format to have RGB color-matrix, while
v4l2object just leave the v4l2 matrix to default when mapping
GST_VIDEO_COLOR_MATRIX_RGB. It causes gst matrix changed to be
GST_VIDEO_COLOR_MATRIX_BT601 when mapping v4l2 colorimetry.
So add support for encoded format with RGB color-matrix in v4l2object.
Note that for M2M encoders, we should in theory assume that that we can
transfer this value from OUTPUT to CAPTURE queues, though its only true
if the drivers does not do CSC. For now, we don't support any RGB
codecs, but leaving a note for the future.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3952>
The V4L2_MAP_QUANTIZATION macro has been fixed to something a lot saner,
fix our replica accordingly. The new macro now simply set the quantization
to full range is the pixel formats is RGB based, or if the JPEG
colorspace is used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3952>
Now that driver version is expected to be equal or superior to 1.3.275 the bug
in NVIDIA and RADV regarding usage is solved, we can revert commit b7ded81f7b.
Also this patch sets the internal usage variable after all the validation are
run, thus the state don't keep an invalid usage.
Finally, the now unused supported_usage variable is dropped.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7247>
Virtual method set_config() can be called several times, and if the number of
profiles counter isn't reset the pool will reach an error state.
The purpose of number of profiles is to check the number of valid vulkan video
profiles (two in the case of transcoding use-case, for example) so it's local to
set_config() virtual method.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7247>
Fixing warnings
GStreamer-CRITICAL **: 01:21:25.862: gst_value_set_int_range_step:
assertion 'start < end' failed
Although when QSV runtime reports a codec is supported, resolution query
fails sometimes, espeically VP9 encoder case on Windows.
Don't try to register an element if resolution query returned an error
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7250>
A fence configured in GstD3D12Memory should be used only for
write access to be completed. And because d3d12 -> d3d11 copy path
is read access to d3d12 resource, we should not set fence to
memory. Otherwise another read access to the d3d12 resource
will wait for d3d11 device context's copy operation although
simultaneous read access is allowed.
Use background thread to keep d3d12 resource and wait for d3d11 device's
copy operation instead.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7243>
When configured in constant bitrate mode, the muxer computes timing information
using the configured bitrate and the byte counter (now = bytes sent / byterate).
When an application changes the bitrate in CBR mode during playback, the
relationship between bytes sent and bitrate is no longer valid so new timing
values will be off by the ratio of the old bitrate to the new bitrate.
Furthermore, it will upset the way that padding is generated.
pad_stream() works by trying to fit the byte counter to now * byterate.
The result is that when decreasing bitrate, the muxer stalls, waiting until the
byte counter is in agreement with now * byterate. Also, when increasing
bitrate, the padding will spike in volume until the byte counter fits with
now * byterate.
If the byte counter is scaled by the ratio of new bitrate / old bitrate when
adjusting bitrate, then padding is generated in a way that applications would
more likely expect.
One detail this change doesn't yet address is whether the next PCR will match up
optimally with the previous PCR right after the byte counter is scaled. In that
case, some correction may be necessary. Also, perhaps the user should be
prevented from changing from bitrate=0 to bitrate=nonzero during playback since
it's not straightforward how to scale the byte counter in that case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7158>
We would previously register a whole bunch of encoder/decoder for which the caps
were ... "unknown/unknown".
Add a function to quickly check (without generating caps) whether a given
AVCodecID has a known mapping (which can include the {video|audio}/x-gst-av-*
ones) without generating the caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6237>
videoscale does not have convert function, so remove the convert
description in it's classification. Otherwise, if we want use
autovideoconvert to convert colorsapce, autovideoconvert will select
videoscale to do convert and this will cause to fail.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7215>
VLC counts METADATA as 1 even if the specification states you must not.
This leads to asfdemux failing since there are no bytes left when asfdemux
tries to extract the "last" header.
Do not fail hard in this case and try to proceed when everything else went
fine.
So at least gst-discoverer will see what's in the file.
Closes#3684
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7209>
This is to avoid a regression in validation layer (introduced by commit
916c4e70cd) when using vulkandownload
VUID-VkImageMemoryBarrier2-srcAccessMask-03914 .. vkCmdPipelineBarrier2():
pDependencyInfo->pImageMemoryBarriers[1].srcAccessMask (VK_ACCESS_TRANSFER_READ_BIT)
is not supported by stage mask (VK_PIPELINE_STAGE_2_VIDEO_DECODE_BIT_KHR)
since vulkandownload set DPB memories' access mask to
VK_ACCESS_TRANSFER_READ_BIT, while they are retain by the DPB queue, so when
they are used as DPB after been shown, this validation error is raised.
Must of the barrier values are set ignoring the previous state of the vulkan
images.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7211>
None of the symbols in webrtc-audio-coding-1 are marked with
`__declspec(dllexport)`, rendering the library usable only if
it was built with GCC/Clang.
The only fix available (as the pulseaudio copy has not been updated
with Google's upstream) is to ensure the fallback builds statically.
Although this change will also affect webrtcdsp's dependency on
webrtc-audio-processing-1, it does not break its compilation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6407>
The access flags are kept around the operations, but when the buffer is
released, the access flag should be reset to its original value, since queue
transfers can be done along the pipeline and, when reusing the buffer, the new
queue might not support the latest access flag.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7165>
Instead of dragging the last destination pipeline stage as current barrier
source pipeline stage (which isn't a valid semantic) this patch adds a parameter
to gst_vulkan_operation_add_frame_barrier() to set the source pipeline stage to
define the barrier.
The previous logic brought problems particularly with queue transfers, when the
new queue doesn't support the stage set during a previous operation in a
different queue.
Now the operation API is closer to Vulkan semantics.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7165>
Doing so resets the stride from the VideoMeta and it wasn't done before
the commit below. While on it, drop the plane size check as we can't
reliably predict the correct size when using DRM modifiers.
Fixes: 89b0a6fa23 ("va: refactor buffer import")
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7187>
null event NT handle to ID3D12Fence::SetEventOnCompletion()
will block the calling CPU thread already, thus it has no point that
creating an event NT handle in order to immediate wait for fence at CPU-side.
Note that passing a valid event NT handle to the fence API might be useful
when we need to wait for the fence value later (or timeout is required),
or want to wait for multiple fences at once via WaitForMultipleObjects().
But it's not a considered use case for now.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7176>
GstD3D12Window.priv.input_info is referenced by mouse event handler
in order to calculate corresponding original position
if scene is rotated/flipped by the videosink.
Fixing regression introduced by recent d3d12videosink refactoring
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7177>
The driver - AKA intel-vaapi-driver - has been unmaintained for four years
now and encoding appears to be broken in various cases. As it's unlikely
that the situation will improve, blocklist the driver for encoding.
Decoding appears to be stable enough to keep it enabled.
The driver can still be used by setting the `GST_VA_ALL_DRIVERS` env
variable.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7170>
GstFFMpegAudDec.context can be nullptr if decoder got closed
without opening new context. Note that we don't need to clear
AVCodecContext.extradata there since avcodec_free_context()
will do clear the data if needed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7180>
The driver frame counters (processed, dropped, buffer level) are not
always correct apparently, and don't allow reliably assigning a frame
number to captured frames.
Instead of relying on them, count the number of frames directly here and
detect dropped frames based on the capture times of the frames: if more
than 1.75 frame durations are between two frames, then there must've
been a dropped frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7163>
Define a default usage and use it instead of repeating the same bitwise
addition.
Therefore, when usage is defined as zero, the usage is defined with the
format's supported usage and the default usage, now without the storage
bit, but with color and input attachment bits.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6798>
Not doing so would mean that tags would be overidden by any tag events sent by
upstream. Also only send a tag event directly if upstream never sent one.
By default use GST_TAG_MERGE_REPLACE to override tags that exist in both the
upstream event and this element with the ones from this element, but provide a
new "merge-mode" property to adjust the behaviour.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7145>
avcodec_close() is deprecated and it's not supported anymore to re-open
a codec, so we only ever allocate the codec in set_format() now and
always free it after usage.
As part of this, also fix various memory leaks in related code paths.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6505>
Direct access to AVInputFormat::read_probe() is not possible anymore
with ffmpeg 7.0, and the usefulness of this typefinder seems limited
anyway. An alternative implementation around av_probe_input_format3() or
similar would be possible but it would be going over all possible ffmpeg
probes at once.
Having a typefinder here means that basically every application will
load the gst-libav plugin when typefinding is necessary, which has
unnecessary performance impacts. If a typefinder from here was indeed
missing from typefindfunctions in gst-plugins-base then it would be
better to add it there directly.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3378
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6505>
If a d3d12 memory holds non-direct-queue fence but the fence was
created with D3D12_FENCE_FLAG_SHARED flag, use the fence instead of
waiting for fence at CPU side. Note that d3d12ipcsrc or
d3d12screencapture elements will hold such sharable fence.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7139>
This needs significant work to use the new Metal→Vulkan integration
extension `VK_EXT_metal_objects`
```
MoltenVK/mvk_deprecated_api.h:132:1: note: 'vkGetMTLDeviceMVK' has been explicitly marked deprecated here
MVK_DEPRECATED_USE_MTL_OBJS
^
MoltenVK/mvk_deprecated_api.h:74:52: note: expanded from macro 'MVK_DEPRECATED_USE_MTL_OBJS'
#define MVK_DEPRECATED_USE_MTL_OBJS VKAPI_ATTR [[deprecated("Use the VK_EXT_metal_objects extension instead.")]]
^
../sys/applemedia/videotexturecache-vulkan.mm:303:20: error: 'vkSetMTLTextureMVK' is deprecated:
Use the VK_EXT_metal_objects extension instead.
VkResult err = vkSetMTLTextureMVK (memory->vulkan_mem.image, texture);
^
MoltenVK/mvk_deprecated_api.h:151:1: note: 'vkSetMTLTextureMVK' has been explicitly marked deprecated here
MVK_DEPRECATED_USE_MTL_OBJS
^
MoltenVK/mvk_deprecated_api.h:74:52: note: expanded from macro 'MVK_DEPRECATED_USE_MTL_OBJS'
#define MVK_DEPRECATED_USE_MTL_OBJS VKAPI_ATTR [[deprecated("Use the VK_EXT_metal_objects extension instead.")]]
^
2 errors generated.
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7091>
With clang on macOS:
```
error: format specifies type 'long' but the argument has type 'uint64_t' (aka 'unsigned long long')
...
error: format specifies type 'unsigned long' but the argument has type 'VkImageView' (aka 'unsigned long long')
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7091>
When building for iOS in Cerbero, as of MoltenVK SDK 1.3.283, we have
to statically link to libMoltenVK since it no longer ships a dylib.
This requires linking to libc++, so we find the dep with the objc++
compiler to ensure that meson uses the right linker.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7091>
${libdir}/gstreamer-1.0/include is only valid after installation, but
extra_cflags are added unconditionally, so we can't use that for
include flags.
Instead, let's add the include flag via variables, which are different
for installed and uninstalled pc files.
This is particularly bad for consuming GStreamer via CMake which barfs
on non-existent include paths.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7130>
since the encoded output is changing based on version
it does not make sense to check the output bitstream with a fixed
bytearray since the version in the target might vary. So sticking
to checking the number of output buffers and encoded frame size
similar to the other tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7141>
The code is simplified by using GQuarks for looking for caps features, and
removing inner loops.
Also, it's used the pad template caps to compare with the incoming caps because
is cheaper at the beginning of negotiation, where the pad template caps is used.
And, since the ANY caps where removed, there's no need to check for an initial
intersection.
Finally, the completion of caps features is done through a loop.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6698>
The video meta API now is a mandatory request for DMA kind negotiation. In
current code, the raw method always adds that video meta API in the query
of propose_allocation(), so we do not need to do the duplicated task here.
Just adding a comment to declare that.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6698>
The ANY caps in pad template caps seems to mess up the DMA negotiation.
The command of:
GST_GL_API=opengl gst-launch-1.0 -vf videotestsrc ! video/x-raw,format=NV12 !
vapostproc ! "video/x-raw(memory:DMABuf)" ! glimagesink
fails to negotiate, but in fact, the vapostproc can convert the input NV12
formant into the RGBA format to render.
The ANY may help the passthough mode, but we should make the negotiate correct
first.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6698>
gst_element_send_event(FLUSH_START / FLUSH_STOP) returns FALSE in cases
where any of the most downstream elements have unlinked pads, even if
the pipeline is successfully flushed.
Currently this is considered expected behavior in GStreamer. This patch
updates gst-validate to treat it as such and therefore not fail the test
for a "failing" flush.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7064>
After signal recovery the capture times for the next frames are simply
wrong. Experimentally this affected 2-3 frames and seemed to be related
to the buffer fill level after signal recovery, so drop at least 5
frames and up to fill level + 1 frames in this situation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7106>
Since a buffer resource will occupy at least 64KB,
allocating upload resource per decoding command might not be
an optimal approach. Instead, use sub-region of a upload resource
for multiple decoding command if sub-regions are not overlapped
each other.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7108>
- Align `glib_debug`, `glib_assert` and `glib_checks` options with GLib,
otherwise glib subproject won't inherit their value. Previous names
and values are preserved using Meson's deprecation mechanism.
- Add `extra-checks` and `benchmarks` options in the main project so it
can be inherited in GStreamer subprojects.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1165>
When enable parallel encoding, it is possible that the unshown frame
is not output but it is already be marked as a repeated frame header.
So we need to use a dedicated buffer to hold the repeat frame header,
don't mix it with the orignal frame data.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6867>
Schedule (semi-)static resource upload at converter creation time.
And use single resource for all vertex, index, and constant
buffers, since separate resources will waste GPU memory.
Note that size and address of a committed resource are 64K aligned
even if requested buffer size is small.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7081>
If the first buffers have no timestamp then the sink position would be
initialized to 0. The source pad might output this buffer, which would then
initialize the source position to 0 too.
Afterwards two buffers with a valid but huge timestamp might arrive before any
of them are output on the source pad. The first one would set the sink position
to a huge value, the second one would notice that the difference between the
huge value and 0 is certainly larger than max-size-time and consider the queue
as full.
Instead, simply don't update the times from buffers without timestamps and
assume whatever was set before is still valid, i.e. the buffer has the same
timestamp as the previous one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7071>
We were storing the probe id in a different structure (DecodebinOutputStream)
than the pad it is targetting (which is in MultiQueueSlot).
The problem is that when re-targetting outputs (to a different slot)... we would
end up having an invalid probe id, or not have a reference to an existing one.
Instead, store the probe id in the same structure as the pad it's targetting
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7069>
As NVIDIA Amperium. In this case the each output buffer is also a DPB,
but using a different view layer.
Still pending a validation layer issue:
VUID-VkVideoBeginCodingInfoKHR-flags-07244
Co-authored-by: Victor Jaquez <vjaquez@igalia.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6954>
query_result_status can be optional so we should not create
the query pool if the queue does not support it,
ie, AMD does not support VK_QUERY_TYPE_RESULT_STATUS_ONLY_KHR
In other use case such as VK_QUERY_TYPE_VIDEO_ENCODE_FEEDBACK_KHR, the
query pool must be created.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7043>
As only queryResultStatusSupport can be optional,
the variable name should be more specific.
queryResultStatusSupport reports VK_TRUE if query type
VK_QUERY_TYPE_RESULT_STATUS_ONLY_KHR and use of
VK_QUERY_RESULT_WITH_STATUS_BIT_KHR are supported.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7043>
According to the SPEC:
The frame id numbers (represented in display_frame_id, current_frame_id,
and RefFrameId[ i ]) are not needed by the decoding process, but allow
decoders to spot when frames have been missed and take an appropriate action.
So we should just print out warning and should not return error in parser when
mismatching. The decoder itself is already robust to handle the reference missing.
Fixes#3622
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7047>
Specify "layout" field in src template to make sure it's
set and gets fixated properly if the downstream element
supports both interleaved and non-interleaved caps.
Fixes
gst_pad_set_caps: assertion 'caps != NULL && gst_caps_is_fixed (caps)' failed
critical with e.g.
gst-launch-1.0 rtpdtmfsrc ! rtpdtmfdepay ! audioconvert ! fakesink
Not that the layout really matters in our case since we always
output mono anyway, but non-interleaved requires adding AudioMeta,
so this is the easiest fix.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7036>
Because DXGI flip mode swapchain will disallow GDI operation
to a HWND once swapchain is configured, videosink has been creating
child window of application's window. However, since window creation
would take a few milliseconds, it can cause performance issue such as
UI freezing. Adding a property so that videosink can attach
DXGI swapchain diretly to application's window in order to improve
performance.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7013>
A large refactoring commit for adding features and improve performance
* Reuse internal converter and overlay compositor:
Converter can be reused as long as input and display formats are not
changed. Also overlay compositor reconstruction is required only if
display format is changed
* Don't wait for full GPU flush on resize or close:
D3D12 swapchain requires GPU idle in order to resize backbuffer.
Thus CPU side waiting is required for swapchain related commands
to be finished. However, don't need to wait for full GPU flushing.
* Support multiple sink on a single external window
Keep installed subclass window procedure even if there's no associated
our internal HWND. This will make window procedure hooking less racy.
Then parent HWND's message will be transferred to our internal HWNDs
if needed.
* Adding support for window handle update
Application can change target HWND even when videosink is playing or
paused state. So, users can call gst_video_overlay_set_window_handle()
against d3d12videosink anytime. The videosink will be able to update
internal state and setup resource upon requested.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7013>
Observed Intel GPU driver crash when multiple decoders are
configured in a process. It might be because of frequent
command queue alloc/free or too many in-flight decoding commands.
In order to make command queue persistent and limit the number of
in-flight command lists, holds global decoding command queue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7019>
This was already being used in handle_frame() for errors that happen when queueing a frame for decoding,
let's do the same when a frame is flagged with an error in the output callback.
From quick testing, this makes seeking more reliable (previously, it would sometimes cause a decoding error
and shut the whole decoder down due to GST_FLOW_ERROR).
Also manually sets the max error count to actually stop processing if too many errors occur.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6446>
ReferenceMissingErr is not critical and the simplest solution is to just ignore it. The frame has
the FrameDropped flag set when it occurs, so we can just drop it as usual.
BadDataErr is also not immediately critical, but in its case let's set the ERROR flag,
so the output loop can use GST_VIDEO_DECODER_ERROR to count and error out if it happens too many times.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6446>
ensure_input_parsebin() has a top comment saying it must be called with
INPUT_LOCK taken, but 2 out of 3 usages of the function call it without
taking that mutex.
This patch adds locking in these two remaining usages.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5279>
Upon fatal errors the loop function will first post an error message
then push out an EOS event.
An application may react immediately to the error message by setting the
state of the pipeline to NULL, meaning by the time we push out the EOS
event PAUSED_TO_READY may have reset the seek seqnum to -1.
While this is harmless, the assertion when setting an invalid seqnum
isn't tidy, fix this by simply not resetting to INVALID as it serves no
practical purpose and the next READY_TO_PAUSED will select a new seqnum
anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7032>
Adding NVIDIA nvCOMP library based plugin for lossless raw video
compression/decompression. To build this plugin, user should
install nvCOMP SDK first and specify the SDK path via
"nvcomp-sdk-path" build option or NVCOMP_SDK_PATH env.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6912>
Musl does not implement GNU basename and have fixed a bug where the
prototype was leaked into string.h [1], which resullts in compile errors
with GCC-14 and Clang-17+
| sys/uvcgadget/configfs.c:262:21: error: call to undeclared function 'basename'
ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration]
| 262 | const char *v = basename (globbuf.gl_pathv[i]);
| | ^
Use glib function instead makes it portable across musl and glibc on
linux
[1] https://git.musl-libc.org/cgit/musl/commit/?id=725e17ed6dff4d0cd22487bb64470881e86a92e7a
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7006>
Certain V4L2 drivers can report that a video receiver is seeing
some signal, but that it is unable to synchronize to it. IOW: the driver
can sometimes report V4L2_IN_ST_NO_SYNC and not report V4L2_IN_ST_NO_SIGNAL.
In particular, I've seen the tc358743 (HDMI-to-CSI2 converter) driver
sometimes report this when deployed to a fleet of embedded Raspberry Pis.
The relevant kernel code is in [1]. The video output is not practically
usable when V4L2_IN_ST_NO_SYNC is reported (only visually corrupted frames,
sometimes with random "snow", are received). I assume that this happens when
either the HDMI cable is poorly plugged in or damaged or when a CSI2 FFC
cable is used and is damaged.
The change in this commit is useful for detecting this working-but-not-really
condition in application code. Applications already listening for the "Signal lost"
message will gain the ability to handle this condition.
There seem to be more V4L2 error flags like this, see [2]. However, I do not
have practical experience with them and adding only V4L2_IN_ST_NO_SYNC seems
like a safer option.
[1]: https://github.com/raspberrypi/linux/blob/be8498ee21aa/drivers/media/i2c/tc358743.c#L1534
[2]: https://www.kernel.org/doc/html/v6.6/userspace-api/media/v4l/vidioc-enuminput.html
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7021>
The initial goal was to support the case where we are paused watching a live
stream, and when we resume we can no longer resume from the previously
downloaded position. In that case we internally do a flushing seek back to the
"current live head position". This was also extended since to be able to
handle (utterly broken) servers when we can't really figure out where we are
anymore and therefore trigger that lost sync so we can try to get back on our
feet.
This does fix the issue... but results in spurious FLUSH_{START|STOP} events
being sent downstream. While that's fine for regular playback scenarios, it's a
bit of a wild scenario since a lot of pipelines/applications don't expect such
events when it wasn't triggered by downstream/application.
Fixes#3605
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7005>
This fixes a regression introduced by 6c4f52ea20
There are cases where the input stream will be push-based, time-segment and not
have a collection nor caps. This means the event-based checks are not sufficient
to decide when/where to plug in a identity or parsebin to process the input.
For those corner cases we setup a buffer probe to ensure we always end up with
at least a parsebin
Fixes#3609
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7010>
It can be seen as a WA in the case of multi-channel transcoding (like
decoder output to two channels, one for encoder and one for vpp).
Normally, encoder sets min pts of a huge value to avoid negative dts,
while vpp set pts without this addtional huge value, which are likely to
cause input surface pts does not fit with encoder (since both encoder
and vpp accept the same buffer from decoder, means they modify the timestamp
of one mfx surface). So we add this huge value to vpp to ensure enc and
vpp set the same value to input mfx surface meanwhile does not break
encoder's setting min pts for dts protection.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6971>
Adding a property to control error reporting behavior when output
window is closed in playing or paused state. This can be useful
for apps where an app wants to close window even if it's playing
a stream, and the closed window is expected.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6939>
The typefind code was rejecting content smaller than 128 bytes making it
impossible to play files with very small srt files.
But those can actually be properly detected so fix typefind to allow
smaller content and try its best with it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6937>
When measuring video latency, one mechanism involves taking a photo
with a camera of two screens showing the test video overlayed with
timeoverlay or clockoverlay. In these cases, if the display's pixel
response time is crappy, you will see ghosting due to which it can be
quite difficult to discern what the current timestamp being shown is.
This commit adds a property that *also* shows the timestamp in
a different (sequentially predictable) location every frame, which
makes it easy to tell what the latest rendered timestamp is.
For bonus points, you can also use the fade-time of the previous frame
to measure with sub-framerate accuracy when the photo was taken, not
just clamped to the framerate, giving you a higher precision latency
value.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6935>
This attempts to implement the gtkglsink element on Windows using WGL,
as there were some more gotchas that are along the way, since we need to
juggle with libepoxy along the way, meaning that we need a recent
GTK+-3.24.x for this to work properly, i.e. the upcoming GTK+-3.24.43.
Since we are essentially using an overlay compositor only during
rendering, move its initialization and destruction into the
gtk_gst_gl_widget_render() function, so that things are safer as we are
doing things across threads between gstreamer (gst-gl) and GTK, as GL
operations, as above, have more gotchas on Windows.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4289>
When dealing with push-based inputs, we are now delaying the creation of
parsebin/identity until we get all pre-buffer events.
We therefore can simplify the handling of new pads being linked and only have to
check if upstream can handle pull-based or not.
Avoids creating parsebin for parsed upstream data altogether
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6953>
Simple gst_gl_sync_meta_wait() is not sufficient to ensure GL commands
are executed before dma-buf devices get to see the buffer.
This is the first step that should make the code behave correctly for
everybody, although there may be performance penalty. In the future we
should introduce a more general sync meta that would allow to move the
waiting from gldownload (the producer) to the sink elements (the
consumers).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6968>
Not all GPUs can support arbitrary offset of
D3D12_PLACED_SUBRESOURCE_FOOTPRINT when copying GPU memory between
texture and buffer. Instead of calculating size/offset per plane,
calculate the entire size and offsets at once.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6967>
drm_fourcc.h should be picked up via the pkgconfig include, not the
system includedir directly.
Also consolidate the libdrm usage in va and msdk.
All this allows it to be picked up consistently (via the subproject,
for example).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6932>
This patch fixes this critical warning when registering MSDK:
_dma_fmt_to_dma_drm_fmts: assertion 'fmt != GST_VIDEO_FORMAT_UNKNOWN' failed
It was because the HEVC string with possible output formats has an extra space
that could not be parsed correctly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6853>
The name is already passed via the signal parameters so it doesn't have
to be retrieved again via GstObject API, which would crash on other
GObjects. Child proxy child objects can be any kind of GObject and the
code here otherwise handles this correctly already.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6938>
When transforming from unknown alignment to frame or obu, the TU timestamp
was not properly transferred. Fix this by saving the TU DTS as the first
DTS seen within the the TU data, and the PTS as the last PTS seen in that
TU data. Finally, reset the TU timestamp after each TU have completed.
Fixes#1496
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6895>
Some servers might not provide 100% matching PDT when doing updates, or accross
variants. This would cause the code matching segments using PDT to fail if the
segment PDT was 1 microsecond (or whatever small value) before the candidate
segment. And would pick the (wrong) following segment as the matching one.
In order to be more tolerant when matching, we instead check whether the
candidate segment is within the first segment of the segment we are trying to
match.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
If we end up with a segment with an internal time that varies from the supposed
one, this could be for two reasons:
* We guess-timated the wrong segment to go to when advancing or switching
variants. In that case we try to find the actual segment to go to (just before
this change).
* There was a complete playlist change (for whatever reason) and we can't find a
replacement. In that case we want to carry on playback from this position but
need to remember that we moved (by setting the stream to DISCONT, and
resetting the new mapping).
Fixes playback on several broken stream
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
Since the default value of `m3u8->discont_sequence` (before parsing of the
playlist data) was 0 .. we would never properly detect the presence of that
field if it was present with a value of 0.
This would later on cause havoc in playlist synchronization where we would
assume it didn't have a discontinuity sequence specified (whereas it did, and it
was 0).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
A lot of streams will do a poor job of estimating proper duration of fragments
in the playlist, but over several fragments have it correct.
Instead of constantly trying to realign the estimated stream time, allow for a
more realistic tolerance of 3-4 video frames
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
When updating playlists, we want to know whether the updated playlist is
continuous with the previous one. That is : if we advance, will the next
fragment need to have the DISCONT buffer set on it or not.
If that happens (because we switched variants, or the playlist all of a sudden
changed) we remember that there is a pending discont for the next fragment. That
will be used and resetted the next time we get the fragment information.
Previously this was only partially done. And it was racy because it was set
directly on `GstAdaptiveDemux2Stream->discont` when a playlist was updated,
instead of when the next fragment was prepared.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
When dealing with live streams, the function was assuming that all segments of
the playlist had valid stream_time. But that isn't TRUE, for example in the case
of failing to synchronize playlists.
Fixes losing sync due to not being able to match playlist on updates
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
Even if no new synchronization information is available.
This is necessary because the timestamp offset logic in rtpbin depends
on the base RTP time that is determined by the jitterbuffer, but this
changes all the time (especially in mode=slave) and the timestamp
offsets have to be updated accordingly. Doing so is especially important
if they're only determined by the RTP-Info, which never changes from the
very beginning.
The interval can be configured via the new min-sync-interval property.
Synchronization happens at least that often, but at most as often as the
old sync-interval property allows.
Both intervals are now based on the monotonic system clock.
Additionally, clean up synchronization code a bit, only emit either
inband NTP or RTCP SR synchronization at the same time, based on which
one has the more recent time information, and only emit RTP-Info
synchronization if it wasn't provided previously at the same time as the
NTP-based synchronization information.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
There is generally no requirement to ignore RTCP SR if the RTP time of
the SR differs a lot from the last received RTP packet. The mapping
between RTP and NTP time stays valid until there was a stream reset, in
which case we wouldn't use that information anyway.
When using rtcp-sync-send-time=false the default of 1s difference can
easily be exceeded, e.g. if encoding of the stream after capture adds
more than 1s of latency.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
Never is useful for some RTSP servers that report plain garbage both via
RTCP SR and RTP-Info, for example.
NTP is useful if synchronization should only ever happen based on RTCP
SR or NTP-64 RTP header extension.
Also slightly change the behaviour of always/initial to take RTP-Info
based synchronization into account too. It's supposed to give the same
values as the RTCP SR and is available earlier, so will generally cause
fewer synchronization glitches if it's made use of.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
Instead of switching on the very first stream, require that all streams
have switched before switching to the different synchronization
mechanism.
Without this there will be a noticeable gap during the switch. E.g. when
going from RTP-Info to NTP-based association, first the first stream
only would get an offset, then the first two, ... then all of them.
Depending on the order of streams this will cause a lot of changes in
ts-offset during the transition.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
Previously these parameters were randomly changed in the body of the
function to avoid having to declare a new variable, which made the code
very hard to follow. By marking them as const this won't be possible
anymore in the future.
Also the RTP clock-base (RTP time from RTSP RTP-Info) is an unsigned
64 bit integer as it's an extended RTP timestamp.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
Both were entangled previously and very hard to follow what happens
under which conditions. Now as a very first step the code decides which
of the two cases it is going to apply, and then proceeds accordingly.
This also avoids calculating completely invalid values along the way and
even printing them int the debug output.
Also improve debug output in various places.
This shouldn't cause any behaviour changes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
This simplifies the code as it's a much simpler case than the normal
inter-stream synchronization, and interleaving it with that only
reduces readability of the code.
Also improve some debug output in this code path.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
Adds a separate vtenc_h265a element (with a _hw variant as usual) for the HEVCWithAlpha codec type.
Decided to go with a separate element to not break existing uses of the normal HEVC encoder.
The preserve_alpha property is still only used for ProRes, no need for it here because we explicitly say we want alpha
when using the new element.
For now, the HEVCWithAlpha has an issue where it does not throttle the amount of input frames queued internally.
I added a quick workaround where encode_frame() will block until enqueue_frame() callback notifies it that some space
has been freed up in the internal queue. The limit was set to 5, which should be enough I guess? Hopefully this is not
too prone to race conditions.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6664>
When we are dealing with parsed inputs (i.e. using identity), we need to ensure
that we have a valid stream collection (and therefore DBCollection) before
anything flows dowsntream.
In those cases, we hold onto those events until we get such a collection.
Fixes#3356
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>
This commit separates collection and selections into a new separate structure:
DecodebinCollection.
This provides a much cleaner/saner way of dealing with collections being
updated, gapless playback, etc...
There is now a list of DecodebinCollection in flight, of which two are special:
* input_collection, the currently inputted/merged collection
* output_collection, the currently active collection on the output of multiqueue
Handling GST_EVENT_SELECT_STREAMS is split, by looking for the collection to
which it applies. And the requested streams are stored in it. IIF that
collection is output_collection we can do the switch, else it will be updated
when it becomes active.
Detecting which collection/selection is active is done by looking at the
GST_EVENT_STREAM_START on the output of the multiqueue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>
* Move the handling of GST_EVENT_STREAM_START on a slot to a separate function
* There was a lot of usage of `gst_stream_get_stream_id()` for the slot
active_stream. Cache that instead of constantly querying it.
* Rename the variables in `handle_stream_switch()` to be clearer
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>
* Centralize associating an output to a slot in one function, including properly
resetting those fields
* Rename functions to be more explicit
* Move code to "reset" an output stream into a dedicated function (will be used
later)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>
* Rename the function names to be clearer, with prefixes
* Pass the input (or stream) directly where appropriate
* Document usage, inputs, ownership
* Rename variables for clarity where applicable
* Avoid double lock/unlock if callee can handle it directly
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>
Simplify its usage by having it directly create the message if the collection
changed. This is what caller were always doing and avoids releasing selection
locks yet-another-time
Also use it in more places to avoid code repetition
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6774>
`StdVideoDecodeH265PictureInfo.flags.IsReference` refers to section 3.132 ITU-T
H.265 specification:
reference picture: A picture that is a short-term reference picture or a
long-term reference picture.
`GstH265Picture.ref` doesn't reflect this, but we need to query the NAL type of
the processed slice.
This patch fixes the validation layer error
`VUID-vkCmdBeginVideoCodingKHR-slotIndex-07239` while using the NVIDIA driver.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6901>
Conceptually identical to the present signal of d3d11videosink.
This signal will be emitted with current render target
(i.e., swapchain backbuffer) and command queue. Signal handler
can record GPU commands for an overlay image or to blend
an image to the render target.
In addition to d3d12 resources, videosink will send
d3d11 and d2d resources depending on "overlay-mode"
property, so that signal handler can render by using
preferred/required DirectX API.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6838>
The idea of using separate command queue per videosink was that
swapchain is bound to a command queue and we need to flush the
command queue when window size is changed. But the separate
queue does not seem to improve performance a lot.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6838>
On various 32 bit systems, time_t is actually 64 bits while long is
still only 32 bits. The macro would wrongly trigger its assertion in
this case if a value with more than 68 years worth of seconds is
converted.
Examples are various newer 32 bit platforms and old ones that are
compiled with -D_TIME_BITS=64.
Also statically assert that time_t is either 32 or 64 bits. Other values
might need adjustments in the macro.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6869>
The prime_fds for multi planes may be the same. For example, on Intel's
platform, the NV12 surface may have the same FD for the plane0 and the
plane1. Then, the DRM_IOCTL_GEM_CLOSE will close the same handle twice
and get an "Invalid argument 22" error the second time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6914>
From the spec (chapter 34, v1.3.283):
````
UNORM: the components are unsigned normalized values in the range [0, 1]
SRGB: the R, G and B components are unsigned normalized value that represent
values using sRGB nonlinear encoding, while the A component (if one
exists) is a regular unsigned normalized value
```
The difference is the storage encoding, the first one is aimed for image
transfers, while the second is for shaders, mostly in the swapchain stage in the
pipeline, and it's done automatically if needed [1].
As far as I have checked, other frameworks (FFmpeg, GTK+), when import or export
images from/to Vulkan, use exclusively UNORM formats, while SRGB formats are
ignored.
My conclusion is that Vulkan formats are related on how bits are stored in
memory rather their transfer functions (colorimetry).
This patch does two interrelated changes:
1. It swaps certain color format maps to try first, in both
gst_vulkan_format_from_video_info() and gst_vulkan_format_from_video_info_2(),
the UNORM formats, when comparing its usage, and later check for SRGB.
2. It removes the code that check for colorimetry in
gst_vulkan_format_from_video_info_2(), since it not storage related.
1. https://community.khronos.org/t/noob-difference-between-unorm-and-srgb/106132/7
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6797>
This mirrors the behaviour in vp8enc / vp9enc and is generally more
useful than using any framerate from the caps as it provides some degree
of accuracy if the stream doesn't have timestamps perfectly according to
the framerate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6891>
Turns out AudioConvertHostTimeToNanos and AudioGetCurrentHostTime are macOS-only APIs, which prevents apps using
GStreamer on iOS from being accepted into App Store.
This commit replaces those functions with a manual version of what they do - mach_absolute_time() for the current time,
and data from mach_timebase_info() at the beginning to convert host timestamps to nanoseconds.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6789>
The `videosink` refernce in main() is a floating one, so it should not
be unref'ed (the playbin practically takes ownership of it).
This prevents a "gst_object_unref: assertion '((GObject *)
object)->ref_count > 0' failed" at runtime.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6883>
Using g_error() crashes the application, producing a coredump e.g. when
the user closes the video window, which can be confusing (especially on
the very first tutorial).
Let's change this to log the error message without crashing, using
g_printerr(), like subsequent tutorials.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6883>
To simplify the description, I'm assuming we only have two streams: video and audio.
For the video stream, we have the following events :
- STREAM_START => stream->wait set to true
- NEW_SEGMENT(1) => blocked waiting in gst_stream_synchronizer_wait
- FLUSH_START => unblocked
- FLUSH_STOP => stream->wait reset to false
- NEW_SEGMENT(2) => not waiting, since stream->wait is false
Then for the audio stream, we have the following events :
- STREAM_START => stream->wait set to true
- NEW_SEGMENT(2) => blocked waiting in gst_stream_synchronizer_wait for ever.
Note: The first NEW_SEGMENT event and the FLUSH_START, FLUSH_STOP events of the audio stream
are dropped before being received by the streamsynchronizer element, because the decodebin audio pad src
is not yet linked to the playsink audio pad sink.
To fix this deadlock, we don't reset stream->wait to false in the FLUSH_STOP event when it is not
waiting for the EOS of the other streams.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6763>
If we have constant duration buffers, set the duration on
outgoing buffers, like rtpmp4adepay does. This fixes
problems with (for example) muxers like mp4mux not writing
the duration of the final sample into the index.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6878>
_get_osfhandle() expects valid fd and CRT will abort program
if given paramerter is invalid. The fd can be invalidated
in various way, file was deleted by other process after
we open a file. To avoid it, our own exception
handler must be installed so that _get_osfhandle() can return
INVALID_HANDLE_VALUE if fd is invalid.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6877>
In some cases you want to ensure that a specific element factory is used
while requiring some specific caps but this was not possible. You can
now do `qtmux:video/x-prores,variant=standard|factory-name=avenc_prores_ks`
to ensure that the `avenc_prores_ks` factory is used to produce the
'standard' variant of prores video stream.
This also enhances a bit the documentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6875>
The encoder can specify the a preferred_output_delay value to get better throughput
performance. The higher delay may get better HW performance, but it may increases
the encoder and pipeline latency.
When the output queue length is smaller than preferred_output_delay, the encoder
will not block to waiting for the encoding output. It will continue to prepare and
send more commands to GPU, which may improve the encoder throughput performance.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4359>
This counter is incremented once for every segment, meaning it would
e.g. overflow after 24 days when using 1ms segments. Once that happens,
completely wrong positions are reported and invalid memory is handed out
for writing/reading the next segments.
As the affected variables are unfortunately part of the public API of
the struct, a second set of variables is added together with accessor
functions and both variables are kept in sync for backwards
compatibility.
All existing users of the two variables are moved to the new ones but
external code might still run into the overflow.
This also slightly breaks API as external code updating the variables
will have no effect anymore but the only known user of this is
pulsesink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6740>
If the muxer times out because of the latency deadline it can happen
that some pads have no caps yet. In that case skip creation of streams
for these pads and create updated section tables once the first buffer
arrives later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6823>
This makes sure that for sparse streams (KLV, DVB subtitles, ...) the
muxer does not wait until the next buffer is available for them but
times out on the latency deadline and outputs data.
For non-live pipelines it will still be necessary for upstream to
correctly produce gap events for sparse streams.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6823>
* gst_ffmpegviddec_frame() is the only caller of gst_ffmpegviddec_video_frame()
and has the same signature. Just move the checks into a single function and
use that.
* Make it clear which frames are the input and output ones in
gst_ffmpegviddec_video_frame() to make issues like the one fixed in the previous
commit more obvious.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6851>
The FORCE_KEYFRAME frame which has GST_VIDEO_CODEC_FRAME_FLAG_FORCE_KEYFRAME
bit set should be the sync point. So we should let it be an IDR frame to begin
a new GOP, rather than just promote it to an I frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6619>
The FORCE_KEYFRAME frame which has GST_VIDEO_CODEC_FRAME_FLAG_FORCE_KEYFRAME
bit set should be the sync point. So we should let it be an IDR frame to begin
a new GOP, rather than just promote it to an I frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6619>
__STDC_NO_ATOMICS doesn't seem to exist. In fact the only compiler
I've found that sets any of those is msvc, but it sets
__STDC_NO_ATOMICS__, not __STDC_NO_ATOMICS.
__STDC_NO_ATOMICS__ is the one documented by C11 standard.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6848>
This fixes the code regarding dropping "ghost frames", that is to say input
frames which ended up not producing any decoded frame.
The iteration itself makes sense.. but it was stopping at the "input" frame and
not the decoded frame we just got back.
When dealing with I-frame codecs, ffmpeg will decode frames in separate frames,
so there is no guarantee that they are decoding in order.
Fixes playback issues with such codecs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6842>
Set the P and B frame qp to I frame value to avoid generating delta
QP between different frame types. For ICQ and QVBR modes, we can
only set the qpi value, so the qpp and qpb values should be set to
the same value as the qpi.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6841>
Address below message reported by SDK debug layer.
ID3D12Device::CheckFeatureSupport: Unsupported Decode Profile Specified.
Use ID3D12VideoDevice::CheckFeatureSupport with D3D12_FEATURE_VIDEO_DECODE_PROFILES
to retrieve a list of supported profiles
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6839>
There's nothing requiring <= 64 channels except for getting the reorder
map and creating a channel mixing matrix, but those won't be possible to
call anyway as channel positions can only express up to 64 channels.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6819>
Certain V4L2 fourccs don't (yet) have DRM counter parts, in which case
we can't create DMA_DRM caps for them. This is usually the case for
specific tilings, which are represented as modifiers for DMA formats.
While using these tilings is generally preferable - because of e.g.
lower memory usage - it can result in additional conversion steps when
interacting with DMA based APIs such as GL, Vulkan or KMS. In such cases
using a DMA compatible format usually ends up being the better option.
Before the addition of DMA_DRM caps, this was what playbin3 ended up
requesting in various cases - e.g. prefering NV12 over NV12_4L4 - but
the addition of DMA_DRM caps seems to confuse the selection logic.
As a simple and quite robust solution, assume that peers supporting
DMA_DRM caps always prefer these and reorder the caps accordingly.
In the future we plan to have a translation layer for cases where
there is a matching fourcc+modifier pair for a V4L2 fourcc, ensuring
optimal results.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6645>
Doing it in gst_play_new() means that bindings that directly call
g_object_new() with the GType wouldn't end up initializing both.
This affects at least the Python and GJS bindings.
gst_init() is nonetheless only called from gst_play_new() once because
calling it from class_init would likely lead to problems as that's
called from somewhere in the middle of GObject.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6801>
Don't add an extra ref if non-floating as that ref will never be
unreffed.
gst_bin_add() is transfer floating (alias to transfer none).
Fixes a leak when a non-floating ref was provided as a return value in
the request-aux-sender signal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6807>
Use of VK_PIPELINE_STAGE_2_ALL_COMMANDS_BIT instead of
specific VK_PIPELINE_STAGE_2_VIDEO_DECODE_BIT_KHR
Fix for VUID-vkCmdPipelineBarrier2-srcStageMask-03849
pDependencyInfo->pImageMemoryBarriers[0].srcStageMask
(VK_PIPELINE_STAGE_2_VIDEO_DECODE_BIT_KHR) is not compatible with
the queue family properties
(VK_QUEUE_GRAPHICS_BIT|VK_QUEUE_COMPUTE_BIT|VK_QUEUE_TRANSFER_BIT|
VK_QUEUE_SPARSE_BINDING_BIT|VK_QUEUE_PROTECTED_BIT) of this
command buffer. The Vulkan spec states: The srcStageMask member
of any element of the pMemoryBarriers, pBufferMemoryBarriers, or
pImageMemoryBarriers members of pDependencyInfo must only
include pipeline stages valid for the queue family that was
used to create the command pool that commandBuffer was allocated
from (
https://www.khronos.org/registry/vulkan/specs/1.3-extensions/
html/vkspec.html#VUID-vkCmdPipelineBarrier2-srcStageMask-03849)
The frame barrier should use a compatible srcStageMask for all
the queues.
Remove reset_pipeline_stage_mask as it is redundant
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6780>
Add pitch tests with different forward and backward playback rates.
Those tests depend on the libSoundTouch version to validate the buffers
checksums. The actual version uses libSoundTouch 2.3.2, use the
`--force-fallback-for=soundtouch` meson option to build using the same
version.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6247>
- fully protect accesses to the libsoundtouch API that is not
thread-safe.
- fully protect accesses to GstPitch members that could be read by a
downstream query thread while written by an upstream streaming thread
or a user thread.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6247>
- use the `GST_PITCH_GET_PRIVATE` accessor when needed
- rename `out_seg_rate` to `output_rate` to use the same name as the parameter
- rename `seg_arate` to `segment_applied_rate` to improve readability
- apply gst-indent to gstpitch.hh/cc
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6247>
When changing playing rate, the output segment was not correctly
calculated because the stream time ratio was computed using the previous
input segment rate instead of using the actual rate. This was producing
wrong results for the output segment start and end values.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6247>
The h26xdecoder 'stop' method was not called
as the vulkan h26x class rewires the video decoder
'stop' base method to its own one.
It was causing some memory leaks such as dangling parser
and dpb in h26xdecoder base class.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6782>
Modifying the `avoid-reencoding` property of `encodebin` could potentially cause
it to reconfigure itself, in which case the source pad will be removed and then
re-added.
Therefore set that property *before* attempting to link to that pad.
Fixes smart-render
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6757>
when pipline is
glvideomixerelement->glcolorconvertelement->gldownloadelement and
glcolorconvertelement is not passthrough, the gl bufferpool between
glvideomixerelement and glcolorconvertelement will not add gl sync meta
during allocating buffer. This will cause that glcolorconvert's inbuf
has no sync meta to wait for.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6756>
1. The PTS of all frames should not be changed.
2. Just update the DTS based on the PTS. For the frame which is not
reordered, the DTS is equal to PTS. For frame which is reordered,
the DTS is equal to previous DTS. For example:
Input: F0[D0, P0] -- F1[D1, P1] -- F2[D2, P2] -- F3[D3, P3]
Output: F0[I, D0, P0] -- F3[P, D0, P3] -- F1[B, D1, P1] -- F2[B, D2, P2]
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6703>
1. The PTS of all frames should not be changed.
2. Just update the DTS based on the PTS. For the frame which is not
reordered, the DTS is equal to PTS. For frame which is reordered,
the DTS is equal to previous DTS. For example:
Input: F0[D0, P0] -- F1[D1, P1] -- F2[D2, P2] -- F3[D3, P3]
Output: F0[I, D0, P0] -- F3[P, D0, P3] -- F1[B, D1, P1] -- F2[B, D2, P2]
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6703>
Fixing deadlock in below case
* GC lock is taken by background thread, and the background thread calls
gst_d3d12_ipc_client_release_imported_data() which takes ipc lock
* ipc lock is already taken in ipc thread and trying to pushing GC data
via gst_d3d12_command_queue_set_notify()
* gst_d3d12_command_queue_set_notify() is trying to take GC lock
but it's already taken by background thread
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6749>
Ideally, GPU waiting should be scheduled just before executing command list.
But handling the case outside of converter is a bit complicated.
Under an assumption that constructed command list will be executed
immediately, schedules GPU-side waiting inside of conversion method
to simplify the flow.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6749>
Neither LIBDIR nor LIBPL are set with the native windows Python
(unlike MSYS2), so we need to use `prefix` which takes us to the
rootdir of the Python installation.
The name is also different: it's python312.dll, not python3.12.dll.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6734>
`python.get_variable('FOO', [])` becomes `python.get_variable('FOO')`
due to how Meson treats empty arrays in arguments, which breaks the
fallback feature of get_variable().
So we need to actually check whether the variable exists before trying
to fetch it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6734>
Since we don't want to expose video decoding API outside of GStreamer, the
header is removed from installation and both source files are renamed as
-private.
The header must remain in gst-libs because is referred by GstVulkanQueue,
which's the decoder factory.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6723>
First it derived mapping was disabled for P010 formats, but also there's an
issue with interlaced frames.
It would be possible to disable derived mapping only for interlaced (H.264
decoder and vadeinterlace) but it would spread the hacks along the code. It's
simpler and contained to disable derived completely for Mesa <23.3
Fixes: #3450
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6729>
Instead of duplicating the GStreamer format to DRM fourcc mapping, this patch
uses the GstVideo library helpers. This duplicates the big O of looking for,
since the two lists are traversed, but it's less error prone.
Partially reverts commit 547f3e8622.
Fixes: #3354
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6731>
The assertion that was present before is a bit too harsh, since there is now
a (understandable) use-case where this could happen.
In gapless use-case, with two files containing the same type (ex:audio). The
first one *does* expose a collection with an audio stream, but decoding
fails (for whatever reason).
That would cause us to have configured a audio combiner, which was never
used (i.e. not active).
Then the second file plays and we (wrongly) assume it should be activated
... whereas the combiner was indeed present.
Demote the assertion to a warning and properly handle it
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3389
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6737>
If waylandsink received buffer rate is high which causes frame
drop, the cached staged buffer will be replaced when next buffer
needs to be rendered and be freed after redraw. But there is
chance to get memory leak if ended without redraw. So need to
free staged buffer when do gst_wl_window_finalize().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6670>
`rsvg_handle_get_dimensions()` and `rsvg_handle_render_cairo()` are
deprecated, and the replacement librsvg functions as specified in the
migration guide are `rsvg_handle_get_intrinsic_size_in_pixels()` and
`rsvg_handle_render_document()`.
However, those are not drop-in replacements, and actually have
breaking semantics for our use-case:
1. `intrinsic_size_in_pixels()` requires SVGs to have width+height or
the viewBox attribute, but `get_dimensions()` does not. It will
calculate the geometry based on element extents recursively.
2. `render_cairo()` simply renders the SVG at its intrinsic size on
the specified surface starting at the top-left, maintaining
whatever transformations have been applied to the cairo surface,
including distorted aspect ratio.
However, `render_document()` does not do that, it is specifically
for rendering at the specified aspect ratio inside the specified
viewport, and if you specify a viewPort that does not match the
aspect ratio of the SVG, librsvg will center it.
Matching the old behaviour with the new APIs is a lot of work for no
benefit. We'd be duplicating code that is already there in librsvg in
one case and undoing work that librsvg is doing in the other case.
The aspect ratio handling in this element is also kinda atrocious.
There is no option to scale the SVG while maintaining the aspect
ratio. Overall, element needs a rewrite.
Let's just disable deprecations. The API is not going anywhere.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6726>
There was an issue with this equality check, which was to figure out what to do
with PCR pids (whether they were part of the streams present or not) and whether
we ignore PCR or not.
Turns out ... we already took care of that further up in the function.
The length check can be simplified by just checking whether the length of
the *original* PMT and the new PMT are identical. Since we don't store "magic"
PCR streams in those, we can just use them as-is.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6713>
Set as much information as possible on the slot (including the associated
track) *before* the associated source pad is added to the element.
We need this so that incoming event/queries can be replied to if they are
received when adding the pad
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6690>
