This commit removes the ineffective FF_MPV_DEPRECATED_ options,
namely mpeg_quant (this is only an option for MPEG-4), a53cc
(this is only an option for MPEG-2), force_duplicated_matrix
(applies only to MJPEG) and b_strategy, b_sensitivity and brd_scale
(these options only make sense for encoders supporting B-frames,
which currently means the MPEG-1/2 and MPEG-4 encoders).
Given that these options never changed the outcome of encoding,
they are removed at once.
Notice that the options for the encoders for which it made sense
are not affected by this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This avoids including version.h in all source files, avoiding
unnecessary rebuilds when the version number is bumped. Only
version_major.h is included by the main header, which defines
availability of e.g. FF_API_* macros, and which is bumped much
less often.
This isn't done for libavutil/version.h, because that header needs
to be included essentially everywhere due to LIBAVUTIL_VERSION_INT
being used wherever an AVClass is constructed.
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>
They correspond to the relevant fields from the packet that follows the
one where the expressions are being applied.
Signed-off-by: James Almer <jamrial@gmail.com>
This is done a second time for 5.0 because master was
merged into 5.0 so that it contains the recent DOVI additions.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Most of user data unregistered SEIs are privated data which defined by user/
encoder. currently, the user data unregistered SEIs found in input are forwarded
as side-data to encoders directly, it'll cause the reencoded output including some
useless UDU SEIs.
I prefer to add one option to enable/disable it and default is off after I saw
the patch by Andreas Rheinhardt:
https://patchwork.ffmpeg.org/project/ffmpeg/patch/AM7PR03MB66607C2DB65E1AD49D975CF18F7B9@AM7PR03MB6660.eurprd03.prod.outlook.com/
How to test by cli:
ffmpeg -y -f lavfi -i testsrc -c:v libx264 -frames:v 1 a.ts
ffmpeg -y -i a.ts -c:v libx264 -udu_sei 1 b.ts
ffmpeg -y -i a.ts -c:v libx264 -udu_sei 0 c.ts
# check the user data unregistered SEIs, you'll see two UDU SEIs for b.ts.
# and mediainfo will show with wrong encoding setting info
ffmpeg -i b.ts -vf showinfo -f null -
ffmpeg -i c.ts -vf showinfo -f null -
This fixes tickets #9500 and #9557.
Reviewed-by: "zhilizhao(赵志立)" <quinkblack@foxmail.com>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
There is no reason to wrap them in #ifndef guards, they should only be
defined here and nowhere else. The define guards just add the
possibility to accidentally use the same FF_API name in different
libraries.
When a color indexing transform with 16 or fewer colors is used,
WebP uses "pixel packing", i.e. storing several pixels in one byte,
which virtually reduces the width of the image (see WebPContext's
reduced_width field). This reduced_width should always be used when
reading and applying subsequent transforms.
Updated patch with added fate test.
The source image dual_transform.webp can be downloaded by cloning
https://chromium.googlesource.com/webm/libwebp-test-data/
Fixes: 9368
Signed-off-by: James Zern <jzern@google.com>
Unused since 1f63665ca5.
Found-by: Soft Works <softworkz@hotmail.com>
Reviewed-by: Soft Works <softworkz@hotmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
It is supported only by a few decoders (h263, h263p, mpeg(1|2|)video
and mpeg4) and is entirely redundant with parsers. Furthermore, using
it leads to missing frames, as flushing the decoder at the end does not
work properly.
Co-authored-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
When encoding yuva420 (alpha) frames, the vpx encoder uses a second
vpx_codec_ctx to encode the alpha stream. However, codec options were
only being applied to the primary encoder. This patch updates
codecctl_int and codecctl_intp to also apply codec options to the alpha
codec context when encoding frames with alpha.
This is necessary to take advantage of libvpx speed optimizations
such as 'row-mt' and 'cpu-used' when encoding videos with alpha.
Without this patch, the speed optimizations are only applied to the
primary stream encoding, and the overall encoding is just as slow
as it would be without the options specified.
Signed-off-by: Adam Chelminski <chelminski.adam@gmail.com>
Signed-off-by: James Zern <jzern@google.com>
ff_pnm_parser and ff_vp3_parser already hit the current limit;
an addition to the former (to handle pfm) is planned.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
this prevents some mismatches in config values for realtime and all
intra modes, avoiding failures like:
[libaom-av1 @ ...] Failed to initialize encoder: Invalid parameter
[libaom-av1 @ ...] Additional information: g_lag_in_frames out of
range [..0]
Signed-off-by: James Zern <jzern@google.com>
These bits are reserved in earlier versions of the H.264 spec, and
some poor hardware decoders require they are zero. Thus, it is useful
to be able to zero these on streams that may have them set. The result
is still a valid H.264 bitstream.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
HDR10+ metadata is stored in the bit stream for HEVC. The story is
different for VP9 and cannot store the metadata in the bit stream.
HDR10+ should be passed to packet side data an stored in the container
(mkv) for VP9.
This CL is taking HDR10+ from AVFrame side data in libvpxenc and is
passing it to the AVPacket side data.
Reviewed-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Signed-off-by: James Zern <jzern@google.com>
Un-hardcode the 200ms minimum latency between emitting subtitle events
so that those that wish to receive a subtitle event for every screen
change could do so.
The problem with delaying realtime output by any amount is that it is
unknown when the next byte pair that would trigger output will happen.
It may be within 200ms, or it may be several seconds later -- that's
not realtime at all.
With these triggering a lot of crashes recently, an option to globally
disable all of them is added as a tool to work around those crashes in
case the SEI data is not needed by the user.
Also re-enables s12m for hevc_nvenc, since the issue is not specifically
with that, but it affects all SEI data.
Signed-off-by: Timo Rothenpieler <timo@rothenpieler.org>