CodecProfile will be set in MFXVideoDECODE_DecodeHeader() to match
the input stream. Setting the hard-coded profile here will mislead
user that msdkh265dec supports a special profile only.
Previously alloc_info is initialized when both thiz->initialized
and thiz->allocation_caps are true, but only thiz->initialized is
checked when alloc_info is used.
Instead of relying on buffers after a state change to PLAYING to always start
from 0, track the amount of time we have spent outside playing but not changed
state to PAUSED.
Note that, since Nvidia does not provide nvEncodeAPI.lib file,
find_library() couldn't be used for build on Windows.
This patch changes to load nvEncodeAPI(64).dll or libnvidia-encode.so
in runtime
dynlink_* was introduced since CUDA Toolkit 9.x but it's deprecated from 10.0.
Instead of using #ifdef hack, shipping nvidia headers of NVIDA CODEC SDK
can make build/code simple
MediaSDK has been released as open source [1], but the directories
where it installs its files, are different from the binary only
distribution.
This patch adds to the libraries path the directory /lib. Also it
is defined in meson if the include directory has the mfx/ prefix,
something that is already handled in autotools.
1. https://github.com/Intel-Media-SDK/MediaSDK
"required" keyword is not a valid argument for has_header()
WARNING: Passed invalid keyword argument "required".
WARNING: This will become a hard error in the future.
These function are not used at all, using them together with the
transport-volume property from avdtpsrc may end up in a binding loop so
we better remove the functions.
If properties are proxied through GBinding this can work only if the
proxied property keeps it's own value. The previous implementation will
read the original value if the proxied property signals a change and
thus nothing will happen.
Right now this only works for video. An attempt was made at adding
monitoring following the example winks, but it seems the only devices that
can be easily detected are KS sources, which winks already handles.
The previous behaviour had issues when setting one of the device properties
after _get_caps had been called. The device shouldn't be locked in until after
_start has been called.
the 2018.3.1 intel sdk release places libraries into /lib64 instead of
/lib/lin_x64 or /lib/x64, this commit adds /lib64 to the libdir
locations list
Fixes#815
Before this patch, if mode=auto and video-format!=auto, video-format would
always be ignored, and get set to 8bit-yuv, or if detected to be RGB444, then
it would be set to 8bit-argb. This change respects video-format if it is set
to 10bit-yuv (v210) or 8bit-bgra, even when mode=auto.
Closes#772
There's a race condition when is-live is set to true and the shmsrc
element releases the pipe in the transition from PLAYING to PAUSED.
To avoid it this change ensures that _create method takes the pipe
and increases the use_count in one operation protected by object lock.
Also perform apropriate protections when releasing the pipe.
https://bugzilla.gnome.org/show_bug.cgi?id=797203
../sys/decklink/gstdecklinkvideosink.cpp:1006:11: error: ‘GstDecklinkVideoSink {aka struct _GstDecklinkVideoSink}’ has no member named ‘scheduled_stop_time’
self->scheduled_stop_time = start_time;
^
Decklink sometimes does not notify us through the callback that it has
stopped scheduled playback either because it was uncleanly shutdown
without an explicit stop or for unknown other reasons.
Wait on the cond for a short amount of time before checking if scheduled
playback has stopped without notification.
https://bugzilla.gnome.org/show_bug.cgi?id=797130
This is part of a much larger goal to always keep the frames we schedule to
decklink be always increasing. This also allows us to avoid using both the
sync and async frame display functions which aren't recomended to be used
together.
If the output timestatmsp is not always increasing decklink seems to hold
onto the latest frame and may cause a flash in the output if the played
sequence has a framerate less than the video output.
Scenario is play for N seconds, pause, flushing seek to some other position,
play again. Each of the play sequences would normally start at 0 with
the decklink time. As a result, the latest frame from the previous sequence
is kept alive waiting for it's timestamp to pass before either dropping
(if a subsequent frame in the new sequence overrides it) or displayed
causing the out of place frame to be displayed.
This is also supported by the debug logs from the decklink video sink
element where a ScheduledFrameCompleted() callback would not occur for
the frame until the above had happened.
It was timing related as to whether the frame was displayed based
on the decklink refresh cycle (which seems to be 16ms here),
when the frame was scheduled by the sink and the difference between
the 'time since vblank' of the two play requests (and thus start times
of scheduled playback).
https://bugzilla.gnome.org/show_bug.cgi?id=797130
This is now handled directly in gstaudiosrc/sink, and we were setting
it in the wrong thread anyway. prepare() is not the same thread as
sink_write() or src_read().
This adds "restore-crtc" property using which one
can restore previous crtc mode.
By default it is enabled, if CRTC was already
active with a valid mode and kmssink set a new mode
on CRTC using force-modesetting.
This helps user restore previous crtc mode and get
the previous session back after running a kmssink
pipeline involving a force-modesetting.
For e.g. When running a kmssink pipeline on rpi
using force-modesetting on tty console, it was giving
a blank screen after pipeline, and now with help of restore-crtc
functionality, CRTC is set with previous crtc mode
previously active on tty console.
Edited-by: Nicolas Dufresne <nicolas.dufresne@collabora.com>
https://bugzilla.gnome.org/show_bug.cgi?id=797025
This allow setting properties that contains spaces. The spaces are
replaced with '-'. As an example, one can set the connector proper
"scaling mode" with the following:
... ! kmssink connector-properties="s,scaling-mode=1"
https://bugzilla.gnome.org/show_bug.cgi?id=797027
Can be used to pass custom connector properties to DRM. Properties can
be enumerated using modetest tool. These properties can then be applied
with the following gst-launch-1.0 syntax. Note that the name of the
structure is ignored.
... ! kmssink connector-properties="s,props1=value,props2=value"
https://bugzilla.gnome.org/show_bug.cgi?id=797027
drmModeGetFB returns -EINVAL for multi-planar framebuffers. Instead of
depending on the framebuffer dimensions to select the mode, use width
and height from GstVideoInfo, which was used to create the framebuffer
in the first place. This enables kmssink to display multi-planar
formats such as I420 or NV12 with modesetting enabled.
https://bugzilla.gnome.org/show_bug.cgi?id=796985
Since both audio and video capture devices declare the KSCATEGORY_CAPTURE interface,
plugging a camera that supports both could result in an audio device being mistaken
for a video one.
https://bugzilla.gnome.org/show_bug.cgi?id=796958
With the Windows 8.1 SDK, the v1 of the AUDCLNT_STREAMOPTIONS enum is
defined which only has NONE and RAW, so it's not only defined when
AudioClient3 is available.
Add a meson check for the symbol. This is not needed for Autotools
because there we build against the MinGW audioclient.h which is still
at v1 of the AudioClient interface.
The mxsfb-drm driver has been added to the kernel long ago and will now
be the default display driver for NXP i.MX28, i.MX6SX and i.MX7D
processors so now is a good time to add it to kmssink.
Also, this is used in the upcoming i.MX8MQ and i.MX8MM processors.
https://bugzilla.gnome.org/show_bug.cgi?id=796873
Otherwise decklink seems to hold onto the latest frame and may cause a
flash in the output if the played sequence has a framerate less than the
video output.
Scenario is play for N seconds, pause, flushing seek to some other position,
play again. Each of the play sequences would normally start at 0 with
the decklink time. As a result, the latest frame from the previous sequence
is kept alive waiting for it's timestamp to pass before either dropping
(if a subsequent frame in the new sequence overrides it) or displayed
causing the out of place frame to be displayed.
This is also supported by the debug logs from the decklink video sink
element where a ScheduledFrameCompleted() callback would not occur for
the frame until the above had happened.
It was timing related as to whether the frame was displayed based
on the decklink refresh cycle (which seems to be 16ms here),
when the frame was scheduled by the sink and the difference between
the 'time since vblank' of the two play requests (and thus start times
of scheduled playback).
Use async_depth for latency calcuation instead of
the length of Tasks array which could be NULL since we
don't do the msdk decoder init in set_format().
According to MediaSDK specification,
Width must be a multiple of 16 and Height must be a multiple
of 16 for progressive frame sequence and a multiple of 32 otherwise.
This patch sets a 16 bit alignment for width and 32 bit alignment
for height as default.
https://bugzilla.gnome.org/show_bug.cgi?id=796566
In cases where we do hard resest, the current code destroys the frame
which has new resolution bit early and this causes buffer_unmap
warnings. Keep an extra ref to the frame internally to avoid this.
The gst-msdk decoders only support packetized formats for
all codecs except VC1. For VC1, it supports codec_data for advanced
profiles and this codec_data wan't submitting to MSDK's DecodeHeader APIs.
Make sure the subclass deocders correctly configured so that
the codec_data buffers are in place in the internal adapter for
MediaSDK's DecoderHeader usage.
Currently we use the gst_video_decoder_get_oldest_frame()
to get the old pending frame to output. But this is not correct
if pts re-ordering required. This patch uses a custom made
get_old_frame() which accounts the PTS too similar to the
v4l2decoder.
https://bugzilla.gnome.org/show_bug.cgi?id=796699
The patch adds a serios of changes to support dynamic resolution
change and efficient utilization of resources.
Major changes:
-- Use MSDK's apis to retrieve the headers instead of only relying
on upsteram notification. For eg: avc decoder requires SEI header
information for dpb count calculation which we don't get from caps.
-- For all codecs other than VP9, we force the reset of decoder
if resoultion changes to fit with gstreamer flow. VP9 enfource
the hard reset only if the new resolution is bigger.
-- delay the src caps setting till msdk api's invokation in
handle_frame to avoid caching multiple configuration values
-- ensure pool negotiation is based on decoder's allocation_caps.
--dynamic resoluttion change use an explicit allocation_query
to reclaim the buffers before closing the decoder (thanks to v4l2dec)
--In case if we don't get upstream notification of res change (for eg,
this can can happen for vp9 frames with ivfheader where ivfparse
is not able to notify the dynamic changes), we handle the the case
based on MFX_ERR_INCOMPATIBLE_VIDEO_PARAM which is the return value
of MFXVideoDECODE_DecodeFrameAsync
-- calculate the minimum surfaces to be preallocated based on
msdk suggestion, downstream requirement, async depth and scratch surface
count for smooth display.
https://bugzilla.gnome.org/show_bug.cgi?id=796566
The transform from mediacodec applies to the texture coords, but
GStreamer affine meta applies to the video geometry, which is the
opposite - so invert it to get display correct for decoders
that require transforming
According to msdk spec, there are two ways to enable filters:
1: Filters can be enabled by adding a filter ID
to mfxExtVPPDoUse. In this case, default filter parameters are used
2: Add filter configuration structures directly to mfxVideoParam.
Using 1 with 2 is optional but legal. Unfortunately it won't work
with some specific use cases like Detail/EdgeEnhancement.
Let's stick with option2 which works fine for all VPP operations.
https://bugzilla.gnome.org/show_bug.cgi?id=796468
Since we do the MSDK initializing in set_caps(), a FALSE
return may still cause the invokation of set_caps() again
and this will leads to buffer allocation and other mess-up.
So make sure the msdk initialized correctly before trying
to do any buffer allocation.
https://bugzilla.gnome.org/show_bug.cgi?id=796465
Make sure all the enabled filter structures are added in the
mfxVideoParm before doing the VPPQuery so that msdk
can do the input param validation
https://bugzilla.gnome.org/show_bug.cgi?id=796465
Update the license blurb in camconditionalaccess.[hc] from GPL to LGPL.
The plugin is LGPL and the GPL header in those two files was just a
copy/paste mistake.
Using NV12 layout in dmabuf mode giving mis-aligned
VPP output with the media-driver. Keep the NV12 support
(so that we can file the bug agianst msdk or mediadriver),
but lower the ordering so that BGRA picks as default.
NV12 issue can be reproduced with explicit capfilter:
vidoetestsrc ! msdkvpp ! video/x-raw\(memory:DMABuf\),format=NV12 ! glimagesink
Added a utility method to replace the MemID (interanl VASurfaceID)
associated with the mfxFrameSurface. This is usefull for dmabuf-import
where we need to replace the memID dynamically
https://bugzilla.gnome.org/show_bug.cgi?id=794817
Exporting DRM_PRIME fd to VASurface requires direct
invocation of VA api VACreateSurface with
VASurfaceAttribExternalBufferDescriptor and other
necessary surface attributes.
https://bugzilla.gnome.org/show_bug.cgi?id=794817
In case the wasapi buffer levels got low in shared mode we would still wait until
more buffer is available until writing something in it, which means we could never
catch up and recover.
Instead only wait for a new buffer in case the existing one is full and always write
what we can. Also don't loop until all data is written since the base class can handle
that for us and under normal circumstances this doesn't happen anyway.
This only works in shared mode, as in exclusive mode we have to exactly
fill the buffer and always have to wait first.
This fixes noisy (buffer underrun) playback with the wasapisink under load.
https://bugzilla.gnome.org/show_bug.cgi?id=796354
The calculation for the frame count in the non-aligned case resulted in
a one too low buffer frame count.
This resulted in:
1) exclusive mode not working as the frame count has to match
exactly there.
2) Buffer underruns in shared mode as the current write() code doesn't
handle catching up to low buffer levels (fixed in the next commit)
To fix just use the wasapi API to get the buffer size which will always
be correct.
https://bugzilla.gnome.org/show_bug.cgi?id=796354
S_FALSE is a valid return value which does not indicate an error.
For example IAudioClient_Stop() returns S_FALSE when it is already stopped.
Use the FAILED macro instead which just checks if an error occured or not.
This fixes spurious warnings when using the wasapisink element.
https://bugzilla.gnome.org/show_bug.cgi?id=796280
This is a new warning introduced by gcc 8
We already check just before that we have enough space, just do a regular
memcpy with the full string size.
camswclient.c:87:3: error: ‘strncpy’ specified bound depends on the length of the source argument [-Werror=stringop-overflow=]
'cuDeviceComputeCapability' was deprecated as of CUDA 5.0
gstnvenc.c: In function ‘gst_nvenc_create_cuda_context’:
gstnvenc.c:290:9: error: ‘cuDeviceComputeCapability’ is deprecated [-Werror=deprecated-declarations]
&& cuDeviceComputeCapability (&maj, &min, cdev) == CUDA_SUCCESS) {
^
https://bugzilla.gnome.org/show_bug.cgi?id=796203
The new property "output-order" can be set to either "display" order
which is the default where frames will be outputting in display order,
or "decoded-order" which will be outputting the frames in decoded order.
The "decoded order" output is generally useful for debugging. But there
are few
customers who use it for low-latency streaming. For eg if the customer
already knows that the stream doesn't have b-frames (which means no
algorithm requires for display order calculation), then they can use
"decoded-order"
output to skip some of the DPB logic to avoid the frame accumulation at
start-up.
The root cause of the above issue is a bit of unclarity in h264 spec +
lazy implementation of many H264 encoders; This is well handled in
gstreamer-vaapi using "low-latency" property:
https://bugzilla.gnome.org/show_bug.cgi?id=762509https://bugzilla.gnome.org/show_bug.cgi?id=795783
For packetized input, inform the msdk that the buffer has
a complete frame or complementary field pairs. For decoding,
this means that the decoder can proceed with this buffer without
waiting for the start of the next frame, which effectively reduces
decoding latency.
https://bugzilla.gnome.org/show_bug.cgi?id=795783
Currently we use an async depth of 4 as default (based on
recommendations
in msdk apps), which indicates how many asynchronous operations an
application performs
before the application explicitly synchronizes the result. As a result,
we
queue four frames in decoder which might not be good approach for
live streaming.
This patch reset the async-depth to 1 as default so that we do sync for
each frame we decode without queuing. Customer can play with already
exposed "async-depth" property for other use cases
https://bugzilla.gnome.org/show_bug.cgi?id=795783
So far msdk produced dmabuf fds are non-mappable.
If user wants to download the content of underlined surfaces,
dmabufcapsfeature negotiated pipeline will fail. So if the input surface
is dmabuf and downstream doesn't have support for dmabuf capsfeatures,
we do the vpp (no passthrough) and produce the mappable videomemory
buffers.
https://bugzilla.gnome.org/show_bug.cgi?id=794946
prpose_allocation:
-- always instantiate a pool for for upstream
-- use async_depth + 1 as min buffer count
decide_allocation:
-- always create a new bufferpool for source pad.
Each of the msdk element has to create it's own mfxsurfacepool
which is an msdk contraint. For eg: Each Msdk component (vpp, dec and
enc)
will invoke the external Frame allocator for video-memory usage
So sharing the pool between gst-msdk elements might not be a good idea.
https://bugzilla.gnome.org/show_bug.cgi?id=793705
If the "output-cc" property is set to TRUE and there is CC present
in the VBI Ancillary Data, they will be extracted and set on the
outgoing buffer as GstVideoCaptionMeta.
Only CDP packets are supported.
https://bugzilla.gnome.org/show_bug.cgi?id=773863
This adds entry for new DRM driver from xilinx
called "xlnx" which supports atomic modesetting.
We have kept entry for older DRM driver "xilinx_drm"
for backward compatility with a note describing
deprecation.
Signed-off-by: Devarsh Thakkar <devarsht@xilinx.com>
https://bugzilla.gnome.org/show_bug.cgi?id=795228
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.
Since cuda-tools 9.0, nvcuvid.h is replaced by dynlink_nvcuvid.h.
This patch changes nvdec to use run-time dynamic linking if
cuda-tools version >= 9.
nvenc does not require any change since its necessary headers are
still available.
https://bugzilla.gnome.org/show_bug.cgi?id=791724
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
Now, when you set loopback=true on wasapisrc, the `device` property
should refer to a sink (render) device for loopback recording.
If the `device` property is not set, the default sink device is used.
This patch includes:
1\ Implements MsdkDmaBufAllocator and allocation of msdk dmabuf memroy.
2\ Each msdk dmabuf memory include its own msdk surface kept by GQuark.
3\ Adds new option GST_BUFFER_POOL_OPTION_MSDK_USE_DMABUF
https://bugzilla.gnome.org/show_bug.cgi?id=793707
There needs to be generalized for the parameter from
GstVideoMsdkVideoMemory to GstMemory.
Thus we can call these functions if using DMABuf memory.
https://bugzilla.gnome.org/show_bug.cgi?id=793707
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.
Since there is already an "adaptive-B" option, just
use boolean property for B-pyramid enabling.
Fixme: Not sure whether this can be supported in vp8 and vp9.
It could be possible through GPB (b without backward ref) but
can't verify currently. We can move this as common property
once verified with vp8 and vp9 without breaking any backward
compatibility.
https://bugzilla.gnome.org/show_bug.cgi?id=791637
Add a new property "trellis" to enable trellis quantization.
Keeping trellis as a flag value (which is boolean for gst x264 enc element)
since it is possible to enable/disable this seperately for
I,P and B frames through MediaSDK ext option headers.
The subclass implementations always need to inform base-encoder
if it requires the inclusion of Extend Header buffers (mfxExtCodingOption2
and mfxExtCodingOption3).
https://bugzilla.gnome.org/show_bug.cgi?id=791637
This option controls down sampling in look ahead bitrate
control mode. According to spec it is only supported in AVC.
Fixme: Probably HEVC also have support for this in recent
MSDK versions. We could move the enumeration types to common
header usable for multiple codecs.
https://bugzilla.gnome.org/show_bug.cgi?id=791637
MediaSDK has support for a number of rate control algorithms.
Adding all possible options to the property rate-control.
Fixme1: In case of failure, currently we don't have a proper method
to show which rate-control has been failed. It could be better
to add some extensive validation on EncQuery output in case of error.
Unfortunately, not all ratecontrol methods are supported by every codecs
and we don't have the dynamic detection of supported ratecontrol methods yet.
https://bugzilla.gnome.org/show_bug.cgi?id=791637
We have the property "i-frames" to set the IDR interval in a
gop. Unfortunately MSDK HEVC encoder behaves bit differently
for IdrInterval field, IdrInteval == 1 indicate every
I-frame should be an IDR (which is IdrInterval == 0 for other codecs),
IdrInteval == 2 means every other I-frame is an IDR
(which is IdrInterval == 1 for other codecs) etc.
So we generalize the behaviour of property "i-frames" by
incrementing the value by one in each case (only for HEVC).
https://bugzilla.gnome.org/show_bug.cgi?id=791637
The base encoder common properties are not valid for
mjpeg encoder where there is no motion compensation or rate control.
Delaying the property installation on the base gobject
untill the subclass class_init get invoked.
https://bugzilla.gnome.org/show_bug.cgi?id=791637
The gst-msdk decoders prefer packetized streams as input
and in this case we can avoid unnecessary input bitstream copy
to mfxBitstream. This works fine for codecs like h264 where
we only support byte-stream with au alignment. Other format
conversions should be done thorugh parsers. But this won't work
for codecs like vc1 where we don't have an autoplugged parser.
Even the parser is not capable to do format conversions.
Packetizing through base decoders parse() routine will bring a
lot of uncecessary of complexities and codecparser libraray dependency.
So we just use an interal gst_adaper to keep track of bitstream
which is not consumed by msdk durig AsynchronusDecoding.
This adapter will get used only if subclass implementations
set the "is_packetized" to FALSE for msdk base encoder.
https://bugzilla.gnome.org/show_bug.cgi?id=792589
Adding Simple and Main profiles decode support.
Currently msdkvc1dec is not capable to handle the codec_data,
only instream headers are supported. Also msdk vc1 decoder
expecting instream with Sequence header as per SMPTE 421M Annex L.
Most of the decdoebin/playbin pipeline won't work with the above
constraints
because vc1parse is still not an autoplug element.
Only way to make mskdvc1dec work is by connecting a vc1parse
as an upstream element.
https://bugzilla.gnome.org/show_bug.cgi?id=792589
Use drm render node as the first choice of device node file.
Fall backs to use drm primary (/dev/dri/card[0-9])
if there is no render node available
Basic logic is inherited from gstreamer-vaapi, but using
gudev API rather than libudev directly.
Added gudev library as dependency for msdk.
https://bugzilla.gnome.org/show_bug.cgi?id=791599
1\ If downstream's pool is MSDK bufferpool,
2\ If there's shared GstMsdkContext in the pipeline,
a decoder decides to use video memory.
This policy should be improved to handle more cases.
https://bugzilla.gnome.org/show_bug.cgi?id=790752
In case that pipeline is like ".. ! decoder ! encoder ! ..." with using
video memory,
decoder needs to know the async depth of the following msdk element so
that it could
allocate the correct number of video memory.
Otherwise, decoder's memory is exhausted while processing.
https://bugzilla.gnome.org/show_bug.cgi?id=790752
How to share/create GstMsdkcontext is the following:
- Search GstMsdkContext if there's in the pipeline.
- If found, check if it's decoder, encoder or vpp by job type.
- If it's same job type, it creates another instance of
GstMsdkContext
with joined-session.
- Otherwise just use the shared GstMsdkContext.
- If not found, just creates new instance of GstMsdkContext.
https://bugzilla.gnome.org/show_bug.cgi?id=790752
According to the driver's instruction, if there are two or more encoders
or decoders in a process, the session should be joined by
MFXJoinSession.
To achieve this successfully by GstContext, this patch adds job type
specified if it's encoder, decoder or vpp.
If a msdk element gets to know if joining session is needed by the
shared context,
it should create another instance of GstContext with joined session,
which
is not shared.
https://bugzilla.gnome.org/show_bug.cgi?id=790752
1\ In decide_allocation, it makes its own msdk bufferpool.
- If downstream supports video meta, it just replace it with the msdk
bufferpool.
- If not, it uses the msdk bufferpool as a side pool, which will be
decoded into.
and will copy it to downstream's bufferpool.
2\ Decide if using video memory or system memory.
- This is not completed in this patch.
- It might be decided in update_src_caps.
- But tested for both system memory and video memory cases.
https://bugzilla.gnome.org/show_bug.cgi?id=790752
1\ Proposes msdk bufferpool to upstream.
- If upstream has accepted the proposed msdk bufferpool,
encoder can get msdk surface from the buffer directly.
- If not, encoder get msdk surface its own msdk bufferpool
and copy from upstream's frame to the surface.
2\ Replace arrays of surfaces with msdk bufferpool.
3\ In case of using VPP, there should be another msdk bufferpool
with NV12 info so that it could convert first and encode.
Calls gst_msdk_set_frame_allocator and uses video memory only on linux.
and uses system memory on Windows until d3d allocator is implemented.
https://bugzilla.gnome.org/show_bug.cgi?id=790752
Implements 2 memory allocators:
1\ GstMsdkSystemAllocator: This will allocate system memory.
2\ GstMsdkVideoAllocator: This will allocate device memory depending
on the platform. (eg. VASurface)
Currently GstMsdkBufferPool uses video allocator currently by default
only on linux. On Windows, we should use system memory until d3d
allocator
is implemented.
https://bugzilla.gnome.org/show_bug.cgi?id=790752
Implements msdk frame allocator which is required from the driver.
Also makes these functions global so that GstMsdkAllocator could use
the allocated video memory later and couple with GstMsdkMemory.
GstMsdkContext keeps allocation information such as mfxFrameAllocRequest
and mfxFrameAllocResponse after allocation.
https://bugzilla.gnome.org/show_bug.cgi?id=790752
Makes GstMsdkContext to be a descendant of GstObject so that
we could track the life-cycle of the session of the driver.
Also replaces MsdkContext with this one.
Keeps msdk_d3d.c alive for the future.
https://bugzilla.gnome.org/show_bug.cgi?id=790752
Same changes as done for wasapisink in cbe2fc40a. Turns out this is
sometimes also needed for capture. Reported by Mathieu_Du.
Also improve logging in that case for easier debugging.
Sometimes the minimum period advertised by a card results in an
unaligned buffer size error during initialization in exclusive mode.
In that case, we can fetch the actual buffer size in frames and
calculate the period from that.
We can't do this pre-emptively because we can't call GetBufferSize
till Initialize has been called at least once.
https://bugzilla.gnome.org/show_bug.cgi?id=793289
This reduces the chances of startup glitches, and also reduces the
chances that we'll get garbled output due to driver bugs.
Recommended by the WASAPI documentation.
https://bugzilla.gnome.org/show_bug.cgi?id=793289
So far, we have been completely discarding the values of latency-time
and buffer-time and trying to always open the device in the lowest
latency mode possible. However, sometimes this is a bad idea:
1. When we want to save power/CPU and don't want low latency
2. When the lowest latency setting causes glitches
3. Other audio-driver bugs
Now we will try to follow the user-set values of latency-time and
buffer-time in shared mode, and only latency-time in exclusive mode (we
have no control over the hardware buffer size, and there is no use in
setting GstAudioRingBuffer size to something larger).
The elements will still try to open the devices in the lowest latency
mode possible if you set the "low-latency" property to "true".
https://bugzilla.gnome.org/show_bug.cgi?id=793289
This requires using allocated strings, but it's the best option. For
instance, a call could fail because CoInitialize() wasn't called, or
because some other thing in the stack failed.
https://bugzilla.gnome.org/show_bug.cgi?id=793289
This is particularly important when running in exclusive mode because
any delays will immediately cause glitching.
The MinGW version in Cerbero is too old, so we can only enable this when
building with MSVC or when people build GStreamer for MSYS2 or other
MinGW-based distributions.
To force-enable this code when building with MinGW, build with
CFLAGS="-DGST_FORCE_WIN_AVRT -lavrt".
https://bugzilla.gnome.org/show_bug.cgi?id=793289
This provides much lower latency compared to opening in shared mode,
but it also means that the device cannot be opened by any other
application. The advantage is that the achievable latency is much
lower.
In shared mode, WASAPI's engine period is 10ms, and so that is the
lowest latency achievable.
In exclusive mode, the limit is the device period itself, which in my
testing with USB DACs, on-board PCI sound-cards, and HDMI cards is
between 2ms and 3.33ms.
We set our audioringbuffer limits to match the device, so the
achievable sink latency is 6-9ms. Further improvements can be made if
needed.
https://bugzilla.gnome.org/show_bug.cgi?id=793289
We will use ->device for storing a pointer to the IMMDevice structure
which is needed for fetching the caps supported by devices in
exclusive mode.
https://bugzilla.gnome.org/show_bug.cgi?id=793289
This will set the actual-latency-time and actual-buffer-time of the sink
and source.
We completely ignore the latency-time/buffer-time values set
on the element because WASAPI is happiest when it is reading/writing at
the default period. Improving this will likely require the use of the
IAudioClient3 interfaces which are not available in MinGW yet.
https://bugzilla.gnome.org/show_bug.cgi?id=792897
Currently only does probing and does not handle messages from
endpoints/devices. In the future we want to do proper monitoring which
is well-supported in WASAPI.
https://bugzilla.gnome.org/show_bug.cgi?id=792897
We need to parse the WAVEFORMATEXTENSIBLE structure, figure out what
positions the channels have (if they are positional), and reorder them
as necessary.
https://bugzilla.gnome.org/show_bug.cgi?id=792897
There is no fixed limitation for the number of devices on the
decklink API side according to BlackMagic. Many PC motherboards
are able support 6 decklink cards each with up to 8 inputs so
a limit of 16 might well be too low.
https://bugzilla.gnome.org/show_bug.cgi?id=777239
Both the source and the sink elements were broken in a number of ways:
* prepare() was assuming that the format was always S16LE 2ch 44.1KHz.
We now probe the preferred format with GetMixFormat().
* Device initialization was done with the wrong buffer size
(buffer_time is in microseconds, not nanoseconds).
* sink_write() and src_read() were just plain wrong and would never
write or read anything useful.
* Some functions in prepare() were always returning FALSE which meant
trying to use the elements would *always* fail.
* get_caps() and delay() were not implemented at all.
TODO: support for >2 channels
TODO: pro-audio low-latency
TODO: SPDIF and other encoded passthroughs
Three new properties are now implemented: role, mute, and device.
* 'role' designates the stream role of the initialized device, see:
https://msdn.microsoft.com/en-us/library/windows/desktop/dd370842(v=vs.85).aspx
* 'device' is a system-wide GUIDesque string for a specific device.
* 'mute' is a sink property and simply mutes it.
On my Windows 8.1 system, the lowest latency that works is:
wasapisrc buffer-time=20000
wasapisink buffer-time=10000
aka, 20ms and 10ms respectively. These values are close to the lowest
possible with the IAudioClient interface. Further improvements require
porting to IAudioClient2 or IAudioClient3.
https://docs.microsoft.com/en-us/windows-hardware/drivers/audio/low-latency-audio
Sometimes we might get an audio packet without a corresponding video
frame. In these cases, the stream and hardware reference timestamps
would be missing, because they're called on the video frame. Instead of
potentially breaking stuff downstream that might depend on these, we now
extrapolate them.
https://bugzilla.gnome.org/show_bug.cgi?id=792042
When we receive a video or audio buffer, we calculate the next stream
time based on the current stream time + buffer duration. If the next
buffer's stream time is after that, we issue a warning.
This happens because the stream time incoming from Decklink should be
really constant and without gaps. If there is a gap, it means that
something went wrong, e.g. the internal buffer pool is empty (too many
buffers queued up downstream).
https://bugzilla.gnome.org/show_bug.cgi?id=781776
Sometimes we might get an audio packet without a corresponding video
frame. In these cases, the stream and hardware reference timestamps
would be missing, because they're called on the video frame. Instead of
potentially breaking stuff downstream that might depend on these, we now
extrapolate them.
https://bugzilla.gnome.org/show_bug.cgi?id=792042
