Prevent the default webrtc test machinery from attempting to
create and set an answer when we're just testing rollback
of the offers. Add some locking / waiting to ensure the test
is complete before exiting.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7365>
Use pattern matching against expected error strings that
might include internal element names, where the names
are default assigned with incrementing integers. When running
with CK_FORK=no, there may have been previous tests that
ran in the same process and incremented the counters more
than when running in the default fork-per-test mode.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7365>
This patch addresses the issue where GStreamer would throw an error when
attempting to use bt2100-hlg colorimetry with V4L2, which is not
supported by the current V4L2 kernel. When bt2100-hlg colorimetry is set
from caps, the check for transfer (GST_VIDEO_TRANSFER_ARIB_STD_B67) is
bypassed.
The main improvement is to avoid checking the transfer value in
gst_v4l2_video_colorimetry_matches when it is
GST_VIDEO_TRANSFER_ARIB_STD_B67. This is because the transfer value in
the cinfo parameter comes from gst_v4l2_object_get_colorspace, which
converts the transfer to another value, causing a mismatch.
Since the kernel does not support GST_VIDEO_TRANSFER_ARIB_STD_B67,
gst_v4l2_object_get_colorspace cannot map it correctly from V4L2 to
GStreamer. Therefore, we ignore this check to prevent errors.
changes:
- Added a condition in gst_v4l2_video_colorimetry_matches to bypass the
transfer check when the transfer is GST_VIDEO_TRANSFER_ARIB_STD_B67.
- Ensured that the pipeline does not throw errors due to unsupported
bt2100-hlg colorimetry in V4L2.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7212>
num_backward_references > 0 means we need to cache several frames
after the current frame. But the basetransform class does not
provide any _drain() kind function, so we do not have the chance
to push out our cached frames when EOS or set caps event comes.
Rather than losing the last several frames, we should just give up
the backward reference here.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7348>
The current code forgets to push the first several frames if the forward
reference > 0. They are just cached in history array and will never be
deinterlaced and pushed.
For the first several frames, even the forward reference frames are not
enough, we still need to deinterlace them as normal and push them after that.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7348>
"adobe" in app14 marker seem not a null-terminted string. so, when
we use gst_byte_reader_get_string_utf8, more bytes will be read until
null. and "gst_byte_reader_get_uint8 (&reader, &transform)" will almost fail
to read transform
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7356>
I didn't find the behavior and purpose of streamsynchronizer documented
or intuitive. Eventually I got Edward to explain it to me, which was
very helpful. Now I'm contributing some docs so that the next person
doesn't have to figure it out by asking around and hoping for an answer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7084>
fix playback fail, when some file with length_size_minus_one == 2
According to the spec 2 cannot be a valid value, so that stream has a
bad config record. but breaking the decoding because of that, perhaps is too much.
and ffmpeg seem not check this
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7213>
librtmp allows for attaching arbitrary AMF objects to the end of the
connect packet, and this is commonly used for authenticating with
servers.
Add a new property, extra-connect-args, that mimics librtmp's behavior.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7054>
When glupload generates sink caps based on src caps after determining upload method, src
caps may only contain RGBA format.
In this case, the raw caps on the sink pad generated by glupload will only contain the
RGBA format, which will cause caps negotiation fail, because the filter caps used for
negotiation by the upstream element may only contain other formats, such as xBGR, etc.
Add the formats supported by #GstGLMemory to raw caps to ensure that caps negotiation
succeeds.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7061>
With GLES 2.0 we are forced to use CopyTextImage2D which requires
passing an internal format. With QT6 eglfs, we need to pass GL_RGB
instead, probably because of how the texture has been created. As its
hard to guess, simply fallback to GL_RGB on failure. This fixes usage
or qml6glsrc with eglfs backend, without loosing support for
semi-transparent window on other platforms.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7321>
While analyzing gst_vulkan_get_or_create_image_view_with_info() it
seems obvious that this function returns NULL, and that this should be
covered in the return annotations. However, closer inspection indicates
that this is only a precondition check when the incoming arguments are
incompatible with each other, and should not be considered as a function
that optionally returns a pointer.
Signify this by using precondition checks instead of an opencoded
if-return-NULL.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5736>
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>
