Chiebot-Mirror/FFmpeg - FFmpeg - Gitea: Git with a cup of tea

Author	SHA1	Message	Date
Andreas Rheinhardt	4243da4ff4	avcodec/codec_internal: Use union for FFCodec decode/encode callbacks 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>	3 years ago
Andreas Rheinhardt	ce7dbd0481	avcodec/codec_internal: Make FFCodec.decode use AVFrame* This increases type-safety by avoiding conversions from/through void. It also avoids the boilerplate "AVFrame frame = data;" line for non-subtitle decoders. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>	3 years ago
Andreas Rheinhardt	20f9727018	avcodec/codec_internal: Add FFCodec, hide internal part of AVCodec 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>	3 years ago
Andreas Rheinhardt	a688f3c13c	avcodec/internal: Move FF_CODEC_CAP_* to a new header codec_internal.h 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>	3 years ago
Michael Niedermayer	b59ea948cd	avcodec/dfpwmdec: Check packet size more completely Fixes: out of array access Fixes: 45497/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DFPWM_fuzzer-5239786212818944.fuzz Fixes: 45510/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_DFPWM_fuzzer-4947856883056640 Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg Reviewed-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>	3 years ago
James Almer	441fe2508d	dfpw: convert to new channel layout API	3 years ago
Jack Bruienne	39a33038ce	avcodec: add DFPWM1a codec 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>	3 years ago