Fixes: signed integer overflow: 529008646 * 8 cannot be represented in type 'int'
Fixes: 383379145/clusterfuzz-testcase-minimized-ffmpeg_dem_CAF_fuzzer-6674045107503104
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The "progress2" API in pthread_slice.c currently associates a progress
value with a thread rather than a job, relying on the broken assumption
that a job's thread number is equal to its job number modulo thread
count.
This removes this API entirely, and changes hevcdec to use a
ThreadProgress-based implementation that associates a
mutex/cond/progress value with every job.
Fixes races and deadlocks in hevdec with slice threading, e.g. some of
those mentioned in #11221.
thread.h currently contains both API for decoder use and functions
internal to lavc generic layer. Move the latter to avcodec_internal.h,
which is a more appropriate place for them.
The snow encoder uses block based motion estimation which can read out of array if
insufficient alignment is used
It may be better to only apply this for the encoder, as it would safe a few bytes of memory
for the decoder. Until then, this fixes the issue in a simple way.
Fixes: out of array access
Fixes: 68963/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SNOW_fuzzer-4979988435632128
Fixes: 68969/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SNOW_fuzzer-6239933667803136.fuzz
Fixed: 70497/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_SNOW_fuzzer-5751882631413760
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
All versions of MSVC that support C11 (namely >= v19.27)
also support the restrict keyword, therefore av_restrict
is no longer necessary since 75697836b1.
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
It is unnecessary since the removal of non-thread-safe callbacks
in e0786a8eeb. Since then, the
AVCodecContext has only been used as logcontext.
Removing ff_thread_release_buffer() allowed to remove AVCodecContext*
parameters from several other functions (not only unref functions,
but also e.g. ff_h264_ref_picture() which calls ff_h264_unref_picture()
on error).
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Avoids allocations and error checks and allows to remove
cleanup code for earlier allocations. Also avoids casts
and indirections.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Instead, use forward declarations; and in order not to affect
any user include these headers for them, but not internally.
This has the advantage of removing implicit inclusions of these
headers from almost all files providing codecs.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
These defines are also used in other contexts than just AVCodecContext
ones, e.g. in libavformat. Furthermore, given that these defines are
public, the AV-prefix is the right one, so deprecate (and not just move)
the FF-macros.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Fixes: signed integer overflow: 256 * 668003712 cannot be represented in type 'int'
Fixes: 59819/clusterfuzz-testcase-minimized-ffmpeg_dem_MATROSKA_fuzzer-4674636538052608
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: out of array access
Fixes: 45982/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_IFF_ILBM_fuzzer-5124452659888128
Fixes: 45982/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_IFF_ILBM_fuzzer-6362836707442688
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: out of array read on 32bit
Fixes: 54857/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_VC1_fuzzer-5840588224462848
The chroma MC code reads over the currently allocated frame.
Alternative fixes would be allocating a few bytes more at the end instead of a whole
line extra or to adjust the threshold where the edge emu code is activated
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
bink supports 16x16 blocks in chroma planes thus we need to allocate enough.
Fixes: out of array access
Fixes: 55026/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_BINK_fuzzer-6013915371012096
Reviewed-by: Peter Ross <pross@xvid.org>
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
avcodec_enum_to_chroma_pos() and avcodec_chroma_pos_to_enum()
deal with enum AVChromaLocation which is defined in lavu.
These functions are therefore replaced by
av_chroma_location_enum_to_pos() and av_chroma_location_pos_to_enum().
This commit provides the necessary deprecations. Also already make
these functions wrappers around the corresponding lavu functions
as not doing so would force one to disable deprecation warnings.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
APTX decodes four bytes of input to four stereo samples; APTX HD
does the same with six bytes of input. So it can be easily supported
in av_get_audio_frame_duration().
This fixes invalid durations and (derived) timestamps of demuxed
APTX HD packets and therefore fixed the timestamp in the aptx-hd
FATE test.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Actually, ff_slice_thread_allocz_entries() always already
allocates zeroed entries, so ff_reset_entries() was already
unnecessary. Make this more clear by renaming it to
ff_slice_thread_allocz_entries().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
It results in undefined behaviour. Instead initialize the mutexes
and condition variables once during init (and check these
initializations).
Also combine the corresponding mutex and condition variable
into one structure so that one can allocate their array
jointly.
Reviewed-by: Tomas Härdin <tjoppen@acc.umu.se>
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This adds the exact bits per sample for DFPWM to
av_get_exact_bits_per_sample.
Previously, the DTS and PTS were set to 0 because the codec never
reported them, but adding this allows libavformat to automatically
set DTS and PTS from the byte position of the stream.
Signed-off-by: Jack Bruienne <jackbruienne@gmail.com>
This is possible, because every given FFCodec has to implement
exactly one of these. Doing so decreases sizeof(FFCodec) and
therefore decreases the size of the binary.
Notice that in case of position-independent code the decrease
is in .data.rel.ro, so that this translates to decreased
memory consumption.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This increases type-safety by avoiding conversions from/through void*.
It also avoids the boilerplate "AVSubtitle *sub = data;" line
for subtitle decoders. Its only downside is that it increases
sizeof(FFCodec), yet this can be more than offset lateron.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
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>
Also move FF_CODEC_TAGS_END as well as struct AVCodecDefault.
This reduces the amount of files that have to include internal.h
(which comes with quite a lot of indirect inclusions), as e.g.
most encoders don't need it. It is furthemore in preparation
for moving the private part of AVCodec out of the public codec.h.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Since the request_channel_layout is used only by a handful of codecs,
move the option to codec private contexts.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: James Almer <jamrial@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>
