When the sink goes from PLAYING to READY and then back to PLAYING,
the initialization of the audioclient in prepare() fails with the
error AUDCLNT_E_ALREADY_INITIALIZED. As a result, the playback
stops.
To fix this, we need to drop the AudioClient in unprepare() and
grab a new one in prepare() to be able to initialize it again
with the new buffer spec.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2096>
The functionality now resides in
gst_wasapi_util_get_device() and
gst_wasapi_util_get_audio_client().
This is a preparatory patch. It will be used in the following
patch to init/deinit the AudioClient separately from the device.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2096>
The current setting of color properties are not very correct and
we will get some kind of "unknown Color Standard for YUV format"
warnings printed out by drivers. The video-color already provides
some standard APIs for us, and we can use them directly.
We also change the logic to: Finding the exactly match or explicit
standard first. If not found, we continue to find the most similar
one.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2385>
Qualcomm GPU works fine with current implementation now.
Noticeable difference between when it was disabled and current
d3d11 implementation is that we now support GstD3D11Memory
pool, so there will be no more frequent re-binding decoder surface anymore.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2377>
Adding the compatile profiles when we decide the final profile used for decoding.
The final profile candidates include:
1. The profile directly specified by SPS, which is the exact one.
2. The compatile profiles decided by the upstream element such as the h265parse.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2322>
The segmentation is stateful, its information may depend on the previous
segmentation setting. For example, if loop_filter_delta_enabled is TRUE,
the filter_level[GST_VP9_REF_FRAME_INTRA][1] should inherit the previous
frame's value and can not be calculated by the current frame's segmentation
data only. So we need to maintain the segmentation state inside the vp9
decoder and update it when the new frame header comes.
We also fix the CLAMP issue of lvl_seg and intra_lvl because of their wrong
uint type here.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2369>
The current way of DMA buffer mapping is simply forwarding the job
to parent's map function, which is a mmap(). That can not handle the
non-linear buffers, such as tiling, compressed, etc. The incorrect
mapping of such buffers causes broken images, which are recognized
as bugs. We should directly block this kind of mapping to avoid the
misunderstanding.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2353>
If the decoder has crop_top/left value > 0(e.g. the conformance
window in the H265). Which means that the real output picture
locates in the middle of the decoded buffer. If the downstream can
support VideoCropMeta, a VideoCropMeta is added to notify the
real picture's coordinate and size. But if not, we need to copy
it manually and the other_pool is needed. We always assume that
decoded picture starts from top-left corner, and so there is no
need to do this if crop_bottom/right value > 0.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2298>
The base va decoder's video_align is just used for calculation the
real decoded buffer's width and height. It does not have meaning
for the VideoMeta, because it does not align to the real picture
in the output buffer. We will use VideoCropMeta to replace it later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2298>
The base va decoder's video_align is just used for calculation the
real decoded buffer's width and height. While the gst_video_info_align
just calculate the offset and stride based on the video_align. But
all the offsets and strides are overwritten in gst_va_dmabuf_allocator_try
or gst_va_allocator_try, which make that calculation useless.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2298>
By this commit, following formats will be newly supported by d3d11 elements
* Y444_{8, 12, 16}LE formats:
Similar to other planar formats. Such Y444 variants are not supported
by Direct3D11 natively, but we can simply map each plane by
using R8 and/or R16 texture.
* P012_LE:
It is not different from P016_LE, but defining P012 and P016 separately
for more explicit signalling. Note that DXVA uses P016 texture
for 12bits encoded bitstreams.
* GRAY:
This format is required for some codecs (e.g., AV1) if monochrome
is supported
* 4:2:0 planar 12bits (I420_12LE) and 4:2:2 planar 8, 10, 12bits
formats (Y42B, I422_10LE, and I422_12LE)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2346>
When B-frame is enabled, encoder seems to adjust PTS of encoded sample
by using frame duration.
For instance, one observed timestamp pattern by using B-frame enabled
and 30fps stream is:
* Frame-1: MF pts 0:00.033333300 MF dts 0:00.000000000
* Frame-2: MF pts 0:00.133333300 MF dts 0:00.033333300
* Frame-3: MF pts 0:00.066666600 MF dts 0:00.066666600
* Frame-4: MF pts 0:00.099999900 MF dts 0:00.100000000
We can notice that the amount of PTS shift is frame duration and
Frame-4 exhibits PTS < DTS.
To compensate shifted timestamp, we should
calculate the timestamp offset and re-calculate DTS correspondingly.
Otherwise, total timeline of output stream will be shifted, and that
can cause time sync issue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2354>
As per SDK doc, IDeckLinkInputCallback::VideoInputFrameArrived
method might not provide video frame and it can be null.
In that case, given stream_time can be invalid.
So, we should not try to convert GST_CLOCK_TIME_NONE
by using gst_clock_adjust_with_calibration()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2337>
Track kernel and VA driver dependencies so gstreamer
will re-inspect the plugin if any of them change.
Also, do not blacklist the plugin if !msdk_is_available
since it could be a transient issue caused by one or
more external dependency issues (e.g. wrong/missing
driver specified, but corrected by user later on).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2335>
We add 12 bits entries into this default mapping. And the old mapping
is not precise. For example, the NV12 should not be used as the default
mapping for VA_RT_FORMAT_YUV422 and VA_RT_FORMAT_YUV444, it is even not
a 422 or 444 format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2332>
alphacombine element is a simple element that assumes buffers are always
paired, or at least that missing buffers are signalled with a GAP. The QoS
implementation in the GstVideoDecoder base class allow decoders dropping
frames independently and that could lead to stall in alphacombine.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2326>
... based on MediaFoundation work queue API.
By this commit, wasapi2 plugin will make use of pull mode scheduling
with audioringbuffer subclass.
There are several drawbacks of audiosrc/audiosink subclassing
(not audiobasesrc/audiobasesink) for WASAPI API, which are:
* audiosrc/audiosink classes try to set high priority to
read/write thread via MMCSS (Multimedia Class Scheduler Service)
but it's not allowed in case of UWP application.
In order to use MMCSS in UWP, application should use MediaFoundation
work queue indirectly.
Since audiosrc/audiosink scheduling model is not compatible with
MediaFoundation's work queue model, audioringbuffer subclassing
is required.
* WASAPI capture device might report larger packet size than expected
(i.e., larger frames we can read than expected frame size per period).
Meanwhile, in any case, application should drain all packets at that moment.
In order to handle the case, wasapi/wasapi2 plugins were making use of
GstAdapter which is obviously sub-optimal because it requires additional
memory allocation and copy.
By implementing audioringbuffer subclassing, we can avoid such inefficiency.
In this commit, all the device read/write operations will be moved
to newly implemented wasapi2ringbuffer class and
existing wasapi2client class will take care of device enumeration
and activation parts only.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2306>
