With the necessary pixel formats defined, we can now expose support for
the remaining 10/12bit combinations that VAAPI can handle.
Specifically, we are adding support for:
* HEVC
** 12bit 420
** 10bit 422
** 12bit 422
** 10bit 444
** 12bit 444
* VP9
** 10bit 444
** 12bit 444
These obviously require actual hardware support to be usable, but where
that exists, it is now enabled.
Note that unlike YUVA/YUVX, the Intel driver does not formally expose
support for the alphaless formats XV30 and XV360, and so we are
implicitly discarding the alpha from the decoder and passing undefined
values for the alpha to the encoder. If a future encoder iteration was
to actually do something with the alpha bits, we would need to use a
formal alpha capable format or the encoder would need to explicitly
accept the alphaless format.
Sufficiently recent Intel hardware is able to do encoding of 8bit 4:4:4
content in HEVC and VP9. The main requirement here is that the frames
must be provided in the AYUV format.
Enabling support is done by adding the appropriate encoding profiles
and noting that AYUV is officially a four channel format with alpha so
we must state that we expect all four channels.
Now that we have a combination of capable hardware (new enough Intel)
and a mutually understood format ("AYUV"), we can declare support for
decoding 8bit 4:4:4 content via VAAPI.
This requires listing AYUV as a supported format, and then adding VAAPI
as a supported hwaccel for the relevant codecs (HEVC and VP9). I also
had to add VP9Profile1 to the set of supported profiles for VAAPI as it
was never relevant before.
The -shortest option (which finishes the output file at the time the
shortest stream ends) is currently implemented by faking the -t option
when an output stream ends. This approach is fragile, since it depends
on the frames/packets being processed in a specific order. E.g. there
are currently some situations in which the output file length will
depend unpredictably on unrelated factors like encoder delay. More
importantly, the present work aiming at splitting various ffmpeg
components into different threads will make this approach completely
unworkable, since the frames/packets will arrive in effectively random
order.
This commit introduces a "sync queue", which is essentially a collection
of FIFOs, one per stream. Frames/packets are submitted to these FIFOs
and are then released for further processing (encoding or muxing) when
it is ensured that the frame in question will not cause its stream to
get ahead of the other streams (the logic is similar to libavformat's
interleaving queue).
These sync queues are then used for encoding and/or muxing when the
-shortest option is specified.
A new option – -shortest_buf_duration – controls the maximum number of
queued packets, to avoid runaway memory usage.
This commit changes the results of the following tests:
- copy-shortest[12]: the last audio frame is now gone. This is
correct, since it actually outlasts the last video frame.
- shortest-sub: the video packets following the last subtitle packet are
now gone. This is also correct.
GSoC'22
libavfilter/vf_chromakey_cuda.cu:the CUDA kernel for the filter
libavfilter/vf_chromakey_cuda.c: the C side that calls the kernel and gets user input
libavfilter/allfilters.c: added the filter to it
libavfilter/Makefile: added the filter to it
cuda/cuda_runtime.h: added two math CUDA functions that are used in the filter
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>
Support for VDPAU accelerated AV1 decoding was added with libvdpau-1.5.
Support for the same in ffmpeg is added with this patch. Profiles
related to VDPAU AV1 can be found in latest vdpau.h present in
libvdpau-1.5.
Add AV1 VDPAU to list of hwaccels and supported formats
Added file vdpau_av1.c and Modified configure to add VDPAU AV1 support.
Mapped AV1 profiles to VDPAU AV1 profiles. Populated the codec specific
params that need to be passed to VDPAU.
Signed-off-by: Philip Langdale <philipl@overt.org>
This enables printing to a resource specified with -o OUTPUT.
In case the output is not specified, prints to stdout as usual.
Address issue: http://trac.ffmpeg.org/ticket/8024
Signed-off-by: Marton Balint <cus@passwd.hu>
Noticed-and-suggested-by: Mark Gaiser <markg85@gmail.com>
Reviewed-by: Mark Gaiser <markg85@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Required to remux m2ts to mkv
Minor changes and porting to FFBitStreamFilter done by the committer.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Calculate Spatial Info (SI) and Temporal Info (TI) scores for a video, as defined
in ITU-T P.910: Subjective video quality assessment methods for multimedia
applications.
This patch builds on my previous DFPWM codec patch, adding a raw
audio format to be able to read/write the raw files that are most commonly
used (as no other container format supports it yet).
The muxers are mostly copied from the PCM demuxer and the raw muxers, as
DFPWM is typically stored as raw data.
Please see the previous patch for more information on DFPWM.
Signed-off-by: Jack Bruienne <jackbruienne@gmail.com>
From the wiki page (https://wiki.vexatos.com/dfpwm):
> DFPWM (Dynamic Filter Pulse Width Modulation) is an audio codec
> created by Ben “GreaseMonkey” Russell in 2012, originally to be used
> as a voice codec for asiekierka's pixmess, a C remake of 64pixels.
> It is a 1-bit-per-sample codec which uses a dynamic-strength one-pole
> low-pass filter as a predictor. Due to the fact that a raw DPFWM decoding
> creates a high-pitched whine, it is often followed by some post-processing
> filters to make the stream more listenable.
It has recently gained popularity through the ComputerCraft mod for
Minecraft, which added support for audio through this codec, as well as
the Computronics expansion which preceeded the official support. These
both implement the slightly adjusted 1a version of the codec, which is
the version I have chosen for this patch.
This patch adds a new codec (with encoding and decoding) for DFPWM1a.
The codec sources are pretty simple: they use the reference codec with
a basic wrapper to connect it to the FFmpeg AVCodec system.
To clarify, the codec does not have a specific sample rate - it is
provided by the container (or user), which is typically 48000, but has
also been known to be 32768. The codec does not specify channel info
either, and it's pretty much always used with one mono channel.
However, since it appears that libavcodec expects both sample rate and
channel count to be handled by either the codec or container, I have
made the decision to allow multiple channels interleaved, which as far
as I know has never been used, but it works fine here nevertheless. The
accompanying raw format has a channels option to set this. (I expect
most users of this will not use multiple channels, but it remains an
option just in case.)
This patch will be highly useful to ComputerCraft developers who are
working with audio, as it is the standard format for audio, and there
are few user-friendly encoders out there, and even fewer decoders. It
will streamline the process for importing and listening to audio,
replacing the need to write code or use tools that require very
specific input formats.
You may use the CraftOS-PC program (https://www.craftos-pc.cc) to test
out DFPWM playback. To use it, run the program and type this command:
"attach left speaker" Then run "speaker play <file.dfpwm>" for each file.
The app runs in a sandbox, so files have to be transferred in first;
the easiest way to do this is to simply drag the file on the window.
(Or copy files to the folder at https://www.craftos-pc.cc/docs/saves.)
Sample DFPWM files can be generated with an online tool at
https://music.madefor.cc. This is the current best way to encode DFPWM
files. Simply drag an audio file onto the page, and it will encode it,
giving a download link on the page.
I've made sure to update all of the docs as per Developer§7, and I've
tested it as per section 8. Test files encoded to DFPWM play correctly
in ComputerCraft, and other files that work in CC are correctly decoded.
I have also verified that corrupt files do not crash the decoder - this
should theoretically not be an issue as the result size is constant with
respect to the input size.
Signed-off-by: Jack Bruienne <jackbruienne@gmail.com>
