This patch will add MPEG4 encoder using Android Mediacodec
Signed-off-by: Samuel Mira <samuel.mira@qt.io<mailto:samuel.mira@qt.io>>
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
The only encoders avaliable using mediacodec were h264 and hevc. This
patch adds the vp9 encoder.
Signed-off-by: Samuel Mira <samuel.mira@qt.io<mailto:samuel.mira@qt.io>>
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
This patch add another ARIB caption decoder using libaribcaption
external library.
Unlike libaribb24, it supports 3 types of subtitle outputs:
* text: plain text
* ass: ASS formatted text
* bitmap: bitmap image
Default subtitle type is ass as same as libaribb24.
Advantages compared with libaribb24 on ASS subtitle are:
* Subtitle positioning.
* Multi-rect subtitle: some captions are displayed at different
position at a time.
* More stability and reproducibility.
To compile with this feature:
* libaribcaption external library has to be pre-installed.
https://github.com/xqq/libaribcaption
* configure with `--enable-libaribcaption` option.
`--enable-libaribb24` and `--enable-libaribcaption` options are
not exclusive. If both enabled, libaribcaption precedes as
order listed in `libavcodec/allcodecs.c`.
Signed-off-by: rcombs <rcombs@rcombs.me>
The encoder seems to be trading blows with hevc_nvenc.
In terms of quality at low bitrate cbr settings, it seems to
outperform it even. It produces fewer artifacts and the ones it
does produce are less jarring to my perception.
At higher bitrates I had a hard time finding differences between
the two encoders in terms of subjective visual quality.
Using the 'slow' preset, av1_nvenc outperformed hevc_nvenc in terms
of encoding speed by 75% to 100% while performing above tests.
Needless to say, it always massively outperformed h264_nvenc in terms
of quality for a given bitrate, while also being slightly faster.
Fixes: member access within null pointer of type 'const FFCodec' (aka 'const struct FFCodec')
Fixes: 45726/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-6554445419249664
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Up until now, codec.h contains both public and private parts
of AVCodec. This exposes the internals of AVCodec to users
and leads them into the temptation of actually using them
and forces us to forward-declare structures and types that
users can't use at all.
This commit changes this by adding a new structure FFCodec to
codec_internal.h that extends AVCodec, i.e. contains the public
AVCodec as first member; the private fields of AVCodec are moved
to this structure, leaving codec.h clean.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This avoids unnecessary rebuilds of most source files if only the
list of enabled components has changed, but not the other properties
of the build, set in config.h.
Signed-off-by: Martin Storsjö <martin@martin.st>
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>