In order to install the color balance interface, a GstVaFilter is
instantiated and queried to know if it supports color balance
filter. It was done just after the GObject was registered. Now, it's
done before.
The reason of this change is that deinterlace element has to be
registered only if deinterlace filter is available, using only one
instantiate of GstVaFilter.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2495>
Having an unlimited input queue is very bad if the
encoder can't run at real-time. Eventually it will
consume all RAM. I don't really see any reason to
have more than 1 outstanding encoded frame, so
remove the queue and limit things to 1 pending frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2499>
There was a race when going to PAUSED while pushing a buffer to the
pipeline process (where we weren't even cancelling anything).
This rework base all the cancellation around the GCancellable
"cancelled" signal trying to ensure that the streaming thread will not
block once a cancel operation happens.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2504>
During the output process, if there are multiple frames in a TU (i.e. multi-spatial
layers case), only one frame with the highest spatial layer id should be selected
according to av1 spec. The highest spatial layer id is obtained from idc value of
the operating point.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2475>
Commit bc09d8cc changed gstmpdparser to put the entire
<ContentProtection> element in the "value" field, so that DRMs
other than PlayReady could make use of the data inside this
element.
However, the data in the "value" field does not include any
XML namespace declarations that are used within the element. This
causes problems for a namespace aware XML parser that wants to
make use of this data.
This commit modifies the way the XML is converted to a string
so that XML namespaces are preserved in the output.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2487>
Some closedcaption elements like sccenc except input buffers
to have timecode metas. The original use case is to serialize
closed captions extracted from a video stream, in that case
ccextractor copies the video time code metas to the closed
caption buffers, but no such mechanism exists when creating
a CC stream ex nihilo.
Remedy that by having timecodestamper accept closedcaption
input caps, as long as they have a framerate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2490>
GstWebRTCSCTPTransport is now made into into an abstract base class
that only contains property specifications matching the
RTCSctpTransport interface of the W3C WebRTC specification, see
https://w3c.github.io/webrtc-pc/#rtcsctptransport-interface. This
class is put into the WebRTC library to expose it for applications and
to allow for generation of bindings for non-dynamic languages using
GObject introspection.
The actual implementation is moved to the subclass WebRTCSCTPTransport
located in the WebRTC plugin.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2214>
Being able to access the SCTP Transport object from the application
means the application can access the associated DTLS Transport object
and its ICE Transport object. This means we can observe the ICE state
also for a data-channel-only session. The collated
ice-connection-state on webrtcbin only includes the ICE Transport
objects that resides on the RTP transceivers (which is exactly how it
is specified in
https://w3c.github.io/webrtc-pc/#rtciceconnectionstate-enum).
For the consent freshness functionality (RFC 7675) to work the ICE
state must be accessible and consequently the SCTP transport must be
accessible for enabling consent freshness checking for a
data-channel-only session.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2214>
This encoder advertises alignment=au as output format, which means
each output frame should contain a full decodable access unit.
The video encoder base class is not aware of our output alignment
and will output spurious buffers with just the SPS/PPS inside when
we call gst_video_encoder_set_headers(), which is broken because
each buffer is supposed to contain a full decodable access unit
in our case.
Just don't tell the base class about our headers, they will be
sent at the beginning of each IDR frame anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2178>
User can get the required buffer size by using buffer pool config.
Since d3d11 implementation is a candidate for public library in the future,
we need to hide everything from header as much as possible.
Note that the total size of allocated d3d11 texture memory by GPU is not
controllable factor. It depends on hardware specific alignment/padding
requirement. So, GstD3D11 implementation updates actual buffer size
by allocating D3D11 texture, since there's no way to get CPU accessible
memory size without allocating real D3D11 texture.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2482>
Split fields ends up on multiple picture and requires accessing the
other_field to complete the information (POC).
This also cleanup the DPB from non-reference (was not useful) and skips
properly merge field instead of keeping them duplicated. This fixes most
of interlace decoding seen in fluster.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2474>
When a frame is composed of two fields, the base class now split the
picture in two. In order to support this, we need to ensure that picture
buffer is held in VB2 queue so that the second field get decoded into
it. This also implements the new_field_picture() virtual and sets the
previous request on the new picture.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2474>
Originally, if a buffer arrives with crop meta but downstream doesn't
handle crop allocation meta, vapostproc tried to reconfigure itself to
non pass-through mode automatically. Sadly, this behavior was based on
the wrong assumption that propose_allocation() vmethod would bring
downstream allocation query, but it is not.
Now, if vapostproc is in pass-through mode, the cropping is passed to
downstream. Pass-through mode can be disabled via a parameter.
Finally, if pass-through mode isn't enabled, it's assumed the buffer
is going to be processed and, if cropping, downstream already
negotiated the cropped frame size, thus it's required to do the
cropping inside vapostproc to avoid artifacts because of the size of
downstream allocated buffers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2443>
There are three framerate conversion algorithms described in
<https://github.com/Intel-Media-SDK/MediaSDK/blob/master/doc/mediasdk-man.md>,
interpolation is not implemented so far and thus distributed timestamp algorihtm
is considered to be more practical which evenly distributes output timestamps
according to output framerate. In this case, newly generated frames are inserted
between current frame and previous one, timestamp is calculated by msdk API.
This implementation first pushes newly generated buffers(outbuf_new) forward and
the current buffer(outbuf) is handled at last round by base transform automatically.
A flag "create_new_surface" is used to indicate if new surfaces have been generated
and then push new outbuf forward accordingly.
Considering the upstream element may not be the msdk element, it is necessary to
always set the input surface timestamp as same as input buffer's timestamp and
convert it to msdk timestamp.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2418>
The mapping between an RTP session and the SDP m= line is not always the
same, especially when BUNDLEing is used.
This causes a failure in a specific case where if when bundling,
if mline 0 is a data channel, and mline 1 an audio/video section,
then retrieving the transceiver at mline 0 (rtp session used) will fail
and cause an assertion.
This fix is actually potentially a regression for cases where the remote
part does not provide the a=ssrc: media level SDP attributes as is now
becoming common, especially when simulcast is involved.
The correct fix actually requires reading out header extensions as used
with bundle for signalling in the actual data, what media and therefore
transceiver is being used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2467>
