The clock seems to have a lot of drift (or we're using it incorrectly)
which causes buffers to be late on the sink and get dropped.
Disable till someone can investigate whether our usage of the API is
incorrect (it looked correct to me) or if something is wrong.
Using the default value (InterleavedDec == MFX_SCANTYPE_UNKNOWN)
causing issues with non-interleaved sample decode. Ideally the usage
of MFXVideoDECODE_DecodeHeader should fix these type of issue, but
it seems to be not. But hardcoding the InterleaveDec to
MFX_SCANTYPE_NONINTERLEAVED
is fixing the problem and fortunately msdk seems to be taking care of
Interleaved samples
too .So let's hardcode it for now.
https://bugzilla.gnome.org/show_bug.cgi?id=793787
We can just return the template caps till the device is opened when
going from READY -> PAUSED. This fixes a CRITICAL when calling
ELEMENT_ERROR before the ringbuffer is allocated.
Also fixes a couple of leaks in error conditions.
https://bugzilla.gnome.org/show_bug.cgi?id=794611
For example, if framerate 0/1 is provided from upstream, the driver
fails to configure and complain about it.
We can let it go and make the driver assuming framerate itself.
https://bugzilla.gnome.org/show_bug.cgi?id=789752
There is no log of gst_decklink_com_thread () which initializes COM.
The initialization part is not valid with #ifdef MSC_VER.
Windows binaries are built with gcc.
As with other codes, it was avoidable by setting it to G_OS_WIN32
instead of MSC_VER.
https://bugzilla.gnome.org/show_bug.cgi?id=794652
There was not handling the end of encoding sequence in encoder.
This patch does drain any remaining internal streams while decoder
already does this.
Document says:
"To mark the end of the encoding sequence, call this function with a
NULL surface
pointer. Repeat the call to drain any remaining internally cached
bitstreams—one
frame at a time—until MFX_ERR_MORE_DATA is returned."
https://bugzilla.gnome.org/show_bug.cgi?id=793236
Sometimes parent context is released before its children get released.
In this case MFXClose of parent session fails.
To make sure that child sessions are closed before closing a parent
session,
Parent context needs to manage child sessions and close them first when
it's released.
https://bugzilla.gnome.org/show_bug.cgi?id=793412
Currently a gst buffer has one mfxFrameSurface when it's allocated and
can't be changed.
This is based on that the life of gst buffer and mfxFrameSurface would
be same.
But it's not true. Sometimes even if a gst buffer of a frame is finished
on downstream,
mfxFramesurface coupled with the gst buffer is still locked, which means
it's still being used in the driver.
So this patch does this.
Every time a gst buffer is acquired from the pool, it confirms if the
surface coupled with the buffer is unlocked.
If not, replace it with new unlocked one.
In this way, user(decoder or encoder) doesn't need to manage gst buffers
including locked surface.
To do that, this patch includes the following:
1. GstMsdkContext
- Manages MSDK surfaces available, used, locked respectively as the
following:
1\ surfaces_avail : surfaces which are free and unused anywhere
2\ surfaces_used : surfaces coupled with a gst buffer and being used
now.
3\ surfaces_locked : surfaces still locked even after the gst buffer
is released.
- Provide an api to get MSDK surface available.
- Provide an api to release MSDK surface.
2. GstMsdkVideoMemory
- Gets a surface available when it's allocated.
- Provide an api to get an available surface with new unlocked one.
- Provide an api to release surface in the msdk video memory.
3. GstMsdkBufferPool
- In acquire_buffer, every time a gst buffer is acquired, get new
available surface from the list.
- In release_buffer, it confirms if the buffer's surface is unlocked or
not.
- If unlocked, it is put to the available list.
- If still locked, it is put to the locked list.
This also fixes bug #793525.
https://bugzilla.gnome.org/show_bug.cgi?id=793413https://bugzilla.gnome.org/show_bug.cgi?id=793525
Directsoundsrc/sink have multiple issues, most of which cannot be
fixed at all because the API is deprecated and is implemented as a
compatibility wrapper around WASAPI since Vista.
Users and developers should now use the wasapisrc/sink elements, and
future development efforts should go towards that.
The low-latency property is *always* safe to enable, so applications
that do realtime communication should set it, and the elements will
automatically configure WASAPI to use the lowest possible device
period, and the audioringbuffer in audiobasesink will also be
configured accordingly.
Applications can also use exclusive mode during capture and playback
for the lowest possible latency if they know that the device will not
be used by any other application.
In this mode, the latency-time and buffer-time properties will be
completely ignored.
The AudioClient3 API is only available on Windows 10, and we will
automatically detect when it is available and use it.
However, using it for capturing audio with low latency and without
glitches seems to require setting the realtime priority of the entire
pipeline to "critical", which we cannot do from inside the element.
Hence, we can only enable that by default for wasapisink since
apps should be able to safely set the low-latency property to TRUE if
they need low-latency capture or playback.
This allows us to request ultra-low-latency device periods even in
shared mode. However, this requires good drivers and Windows 10, so
we only enable this when we detect that we are running on Windows 10
at runtime.
You can forcibly disable this feature on Windows 10 by setting
GST_WASAPI_DISABLE_AUDIOCLIENT3=1 in the environment.
