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
Those fields have been introduced in version 2 and later to define new
profiles like the format range extensions profiles (A.3.5).
NOTE: This patch breaks the parser ABI, rebuild needed.
https://bugzilla.gnome.org/show_bug.cgi?id=793876
We used to have the same enum to represent H265 profiles and idc values.
Those are no longer the same with extension profiles defined from
version 2 of the spec.
Split those enums so the semantic of each is clearer and we'll be able
to add extension profiles to GstH265Profile.
Also add gst_h265_profile_tier_level_get_profile() to retrieve the
GstH265Profile from the GstH265ProfileTierLevel. It will be used to
implement the detection of extension profiles.
https://bugzilla.gnome.org/show_bug.cgi?id=793876
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.
Measures the audio latency between the source pad and the sink pad by
outputting period ticks on the source pad and measuring how long they
take to arrive on the sink pad.
Very useful for quantifying latency improvements in audio pipelines.
This plugin was particularly useful during development of the
low-latency features of the wasapi plugin.
https://bugzilla.gnome.org/show_bug.cgi?id=793839
Strictly speaking, the TTML spec requires that text backgrounds extend
only to the font height of the related text, rather than to the vertical
distance between lines. The result of this is that there will typically
be vertical gaps between line backgrounds through which moving video can
be seen. Since this was unnacceptable to some content providers, v1.0.1
of the IMSC spec (which profiles TTML) adds a new attribute,
itts:fillLineGap[1], that allows content authors to specify that clients
should extend text backgrounds such that there are no gaps between
lines. This attribute is also going to be included in the next release
of EBU-TT-D.
This patch adds support for fillLineGap to ttmlparse and ttmlrender.
[1] https://www.w3.org/TR/ttml-imsc1.0.1/#itts-fillLineGaphttps://bugzilla.gnome.org/show_bug.cgi?id=787071
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.
There is nothing in the spec that state that framerate is not valid in
that case. This aligns GStreamer with FFMPEG behaviour for similar
streams.
https://bugzilla.gnome.org/show_bug.cgi?id=793284
add_global_arguments() can't be used in subprojects. It's
entirely possible that -bad is a subproject but gstreamer
is picked up from an installed location, so we should
really use add_project_arguments() in both cases.
