Archive for the ‘Lossless Audio’ Category

A look on Perfect Clarity Audio

Friday, March 28th, 2025

Since Paul asked me to look at it (and what’s equally important, provided the binary to look at), I did exactly that.

It turned out to be a rather simple codec, originally known as Sonic Foundry Lossless. The only really interesting detail is that it has intra- and inter-frames. Intraframes start with 0xFFFFFFFF and four last samples for each channel (for filtering purposes). Each frame stores the number of samples (I suppose), coded data size, Rice code parameter (per channel), filter method/order (ditto), LPC coefficients (same, only if global flags enable it though) and probably CRC.

Data is coded as fixed-parameter Rice codes (low bit is used for the sign)—unless it’s filter method 5 which means all zeroes, then optional LPC phase (LPC has order four and 8-bit coefficients) and then, depending on filter order, fixed prediction as in Shorten.

Finally there may be mid/stereo reconstruction (if global config signals it) and clipping for 24-bit mode (but not for 16-bit apparently).

I don’t know if they’ve added some improvements in the newer versions but this looks rather simple compared to other lossless codecs. There’s the final bit of weirdness: it’s usually stored in WAV which means it’s one of those codecs with variable-size frames in WAV. Not something I’d like to support.

P.S. I’ll document it for The Wiki (as well as DVC from the previous post) a bit later.

Posted in Lossless Audio | 2 Comments »

Looking at the (un)Original Sound Quality format

Tuesday, June 27th, 2023

I was asked to look at it and since I have nothing better to do, I finally did.

OSQ was a format used by WaveLab mastering software for storing lossless compressed audio (they seem to have dropped support for it in the more recent versions). Internally it’s a clone of ages-old Shorten which does not even have LPC mode, only fixed predictors.

Of course the format somewhat differs and there are more prediction modes but they still use maximum three previously decoded samples and filter coefficients are constant. There are other original additions such as stereo decorrelation or adaptive Rice coding with the parameter selected based on previous statistics and some floating-point calculations that look more complex than they should be. Overall though it looks like somebody took Shorten and decided not to do the hard part (LPC), maybe in order to make files to save faster, and tried to compensate it by a couple of simple tricks.

I’ll probably document it at my leisure but overall it’s a rather silly format.

Posted in Lossless Audio | 3 Comments »

A quick glance at WA

Saturday, January 21st, 2023

A certain Paul B. asked me to look at WA (aka WavArc) and it turned out to be rather interesting. For starters, it’s the only lossless audio archiver and not compressor I’m aware of (the difference is an ability to store multiple files in single archive). Of course there are things like Rar or WinZip with special multimedia compression modes but they’re general-purpose archives with content-specific compression methods and not audio-only archivers.

There are two versions of the executable: 32-bit DOS one and 16-bit DOS one (where all 32-bit integer and FPU operations are emulated). The latter turned out in showcase what Ghidra lacks in supporting old DOS executables, so eventually I tried 32-bit version. Even if I had to load it manually, luckily it turned out to have PE-like header for the loader so it was no problem figuring out segment mapping. After that it was a piece of cake.

Essentially there are three compression modes: store (mode 0), Shorten-like fixed prediction and fixed Rice codes (modes 1-4) and mode 5 with LPC prediction and residue coding using either fixed Rice codes or arithmetic coder (with fixed model transmitted before residues).

Overall, it’s a curious piece of software that was interesting to look at.

Posted in Lossless Audio | 2 Comments »

Looking at a-Pac

Monday, September 19th, 2022

Since currently there’s a preparation phase for the next Ukrainian counteroffensive and I don’t know what other features to add to NihAV (beside improving video player), I was bored enough to do this.

After a comment from Paul, I’ve looked at some random lossless audio codec from 90’s, namely a-Pac. While the codec by itself it very simple (no LPC or IIR filter, no variable-length codes except in block header, no arithmetic coder either) as you can see from its description in the Wiki, it was a bit tricky to RE.

The main issues being: apac.exe is in NE format (which means 16-bit code and Ghidra sucks at figuring out which segment registers should be used when) and, which is worse, written in Delphi with all the quirks of its object model implementation. So it’s after some search I’ve found the virtual table corresponding to the APAC format handler (both coding and decoding) and after trying one function after another I’ve finally found the one responsible for encoding—after which finding a decoder function was easy.

The code itself (beside the weirdness introduced by Delphi compiler like using both positive and negative offsets in vtable calls) is very old-fashioned too: bit reading is done by keeping bitreader state as global variables, reading n-bit values by reading a single bit in a loop and refilling that bit buffer every byte (with a callback function).

It’s still fun to see how we’ve moved from simple formats like this to overcomplicated formats like LA or OptimFrog while settling down on in-between complexity FLAC, Monkey’s Audio and WavPack.

Posted in Lossless Audio | Comments Closed

NihAV: now with lossless audio encoder

Tuesday, October 26th, 2021

Since I wanted to try my hoof at various encoding concepts it’s no wonder that after lossy audio encoder (IMA ADPCM with trellis encoding), lossless video encoder (ZMBV, using my own deflate implementation for compressing), lossy video encoder (Cinepak, for playing with vector quantisation, and VP6, for playing with many other concepts) it was time for a lossless audio encoder.

To remind you, there are essentially two types of lossless audio compressors—fast and asymmetric (based on LPC filters) and slow and symmetric (based on adaptive filters, usually long LMS ones). The theory behind them is rather simple and described below. (more…)

Posted in Lossless Audio, NihAV | 2 Comments »

A brief look at Sonarc

Tuesday, September 29th, 2020

Recently The Mike asked if I can look at this format. In case you didn’t know, The Multimedia Mike is one of the under-appreciated founders of opensource multimedia, involved both in reverse engineering codecs and maintaining infrastructure for about two decades (for example this particular blog has been here for fifteen years thanks to him and his maintenance efforts). So of course I had to look at it even if out of sheer respect.

Sonarc is probably the first known lossless audio codec as the copyright mentions year 1992 as the first date (Shorten and VocPack appeared in 1993). Spoiler: it turned out to be closer to Shorten in design.

This was harder to RE because it was larger (decompressor was three to four times larger than VocPack) and the original was written in Borland Pascal with all the peculiarities it brings. By those peculiarities I mean mostly Pascal strings. Well, the code for manipulating them is annoying to parse but not too bad, the main problem is that they are put in the same segment with code right before the function that uses it and that confuses Ghidra which for some reason selects the segment with standard library routines for them instead (and uninitialised variables are not assigned to any segment at all). The write() implementation is also no fun.

Side note: back in the day Turbo Pascal was probably the best programming language for DOS and back in school at least two my schoolmates were doing crazy things with it (and Delphi later) which I couldn’t (and I was writing in C as I still do today). Yet somehow the popularity of the language vanished and I haven’t heard anything about them becoming famous programmers (neither did I but they had better chances). And the only modern project written in Pascal that I’m aware of is Hedgewars.

Anyway, let’s talk about the format itself. Sonarc can compress raw PCM, .voc and .wav into either its own format or into .wav and it supports both 8- and 16-bit audio.

From what I saw it uses the same approach: optionally applying the LPC filter and coding the residues. Residues can be coded with two different approaches: old one for 8-bit audio and new one for 8- and 16-bit audio. Old 8-bit audio coding uses one of eight different static Huffman codebooks or can code residues as raw bytes (and I can’t remember that many other codecs doing the same except for MLP and DT$-HD Lossless probably because why compress audio in that case). New 8-/16-bit coding still uses fixed codebooks but in a different fashion: now they simply code the number of bits for the residue. It does not look like the data is split into segments but I may be wrong (I/O is still not the easiest thing to get around there).

Overall it’s not a bad codec for its time and e.g. FLAC has not come that far away from it in concepts (except that it uses Rice codes and has independent frames plus partitioning inside individual frame for better compression). I hope though there are no older lossless audio codecs out there to be discovered (CCITT G.711 infinite-law with its fixed 1:1 compression does not count).

Posted in Lossless Audio | 2 Comments »

A look at VP1 and VP2

Saturday, September 26th, 2020

One of the issues with On2 VPx family is that they started it from VP3 while having four different TrueMotion codecs before that (it’s like the company was called Valve and not Duck at that time). But I wanted to look at some lossless audio codecs and there’s VocPack or VP for short which has versions 1 and 2. Bingo!

This is a very old lossless audio codec that appeared in 1993 along with Shorten and, as it turns out, originated the second approach to lossless audio compression. While Shorten was a simple format oriented on fast decoding and thus used fixed prediction (either LPC filter or even fixed prediction scheme) and Rice codes for residues (the same scheme used in FLAC and TAK), VocPack employed adaptive filter and arithmetic coding (the approach carried by LA, Monkey’s Audio, OptimFROG and such). And it was made for DOS and 8-bit audio! Well, version 2 added support for 16-bit but it seems to compress only high 8 bits of the sample anyway while transmitting low bits verbatim.

And it turned out to be my first real experience of using Ghidra with DOS executables. The main troubles were identifying library functions and dealing with pointers. Since it was compiled with Borland C++ 3.0 (who doesn’t remember it?) it was rather easy to decompile but library functions were not recognized (DOS executables don’t get much love these days…) but searching the disassembly for int 21h with Ralf Brown’s interrupts list at hand it was easy to identify calls for file operations (open/read/write/seek) and from those infer the stdio library functions using them and finally the code using all those getc()s. And of course segmented model makes decompiling fun, especially when decompiler can’t understand segment/offset variables being used separately. In result sometimes you recognize offset but you have to look at the data segment yourself to see what it refers to; even worse, for some local variables Ghidra seemed to assume wrong segment which resulted in variables in disassembly and decompiled output pointing to non-existent locations. Despite all of that it was rather easy to understand what unpacker for VP1 does. VP2 has only packer and no unpacker publicly available (feel free to trace the author and buy a copy from him that supports unpacking) plus it depends on those wrongly understood global variables more which prevented me from understanding how encoding a residue works there. In theory you should be able to set data segment manually but I don’t see a point on spending more than a couple of hours on REing the format.

It was a nice distraction though.

Posted in Lossless Audio | 4 Comments »

Lossless audio codecs were more advanced than I thought

Wednesday, September 23rd, 2020

As I’d mentioned in a previous post on lossless audio codecs, I wanted to look at some of them that are still not reverse engineered for documentation sake. And I did exactly that so now entries on LA, OptimFROG and RK Audio are not stubs any more but rather contain some information on how the codecs work.

And if you look at LA structure you see a lot of filters of various sizes and structure. Plus an adaptive weight used to select certain parameters. If you look at other lossless audio codecs with high compression and slow decoding like OptimFROG or Monkey's Audio you’ll see the same picture: several filters of different kinds and sizes layered over each other plus adaptive weights also used in residuals coding. Of course that reminded me of AV2 and more specifically about neural networks. And what do you know, Monkey's Audio actually calls its longer filters neural networks (hence the name NNFilter.h in the official SDK and you can spot it in the version history as well leaving no doubts that it’s exactly the neural networks it is named after).

Which leads me to the only possible conclusion: lossless audio codecs had been using neural networks for compression before it became mainstream and it gave them the best compression ratios in the class.

And if we apply all this knowledge to video coding then maybe in AV4 we’ll finally see some kind of convolution filters processing whole tiles and then the smaller blocks removing spatial redundance maybe with some compaction layers like many neural network designs have (or transforms for largest possible block size in H.265/AV1/AVS2) and expansion layers (well, what do you think motion interpolation actually does?) and using RNNs to code residues left from all the prediction.

Posted in Lossless Audio, Useless Rants | 3 Comments »

Revisiting lossless codecs…

Sunday, September 6th, 2020

I’ve decided to add a couple of lossless audio formats in a preparation for a long-term goal of having a NihAV-based player (the debug tool nihav-player that I currently have can’t really count for one especially considering how it does not play pure audio files and tends to deadlock in SDL audio thread).

So I’ve added nihav-llaudio crate with four most common formats for music I have, namely FLAC, Monkey’s Audio, TTA and WavPack. And I guess it’s time to revisit my opinion about various lossless audio formats now that I’ve (re)implemented support for some of them (I tried to summarise my views about them almost ten years ago). Let’s see what has changed since then:

• I had a closer look at MPEG-4 ALS and it turned out to be rather interesting (and probably the only lossless audio codec with P-frames) but it also has somewhat insane options (like maximum prediction order of 1023 for LPC; or coding the whole file with just one I-frame and the rest being P-frames so no seeking is possible) and RLSLMS being broken (the reference decoder can’t decode the official reference samples) and it got no popularity at all;
• TTA turned out to be very simple with a baffling rationale

The sample count in a TTA1 frame is a multiple to 576 (sound buffer granule). Based on this, the “frame time” is defined as a constant 1.04489795918367346939. Thus, the sample count in a regular TTA1 frame determined as: regular TTA1 frame length = frame time * sample rate.

I’m no mathematician so this does not form a coherent logical chain for me, I’d use something like “frame length in samples is sample rate rounded up to multiple of 576” instead of “sample rate multiplied by 256/245”. The main irritating point is that last frame contains less samples and you need to signal that it’s last frame (or merely check if you have enough bits left to decode a full frame after you decoded enough samples for the last frame). Oh, and TTA2 seems to be still in development.

• And speaking about codecs in development, I don’t see new lossless audio codecs appearing after 2010. Either I got too old and don’t spot them or the interest has finally faded out. This might be because most people don’t buy music any more but rather rent it in some online store or use some streaming service. And those who still do probably use one of the old established codecs like FLAC.
• And since I’ve mentioned it, my opinion on it has not changed and only got a bit more refined not that I have a decoder for it as well. Previously I thought FLAC is a simple format with a bad bitstream format that makes seeking hard. Now I know that FLAC is a simple format (fixed predictor or LPC up to order 32 and fixed Rice codes; the only thing that improves compression is splitting residues into partitions where optimal k for coding them can be selected) with horribly designed bitstream format.

Normally lossless audio formats either store offsets for each frame or have an easily recognizable header, but FLAC is different. It’s obvious that the author was inspired by MPEG audio header design but those actually had frame sizes coded. Here in order to find where the frame ends you need either to decode it or calculate CRC for the data you read (and in the likely case of false positives also check that the data is followed by a valid header). One could argues that there’s often a seek table in FLAC file but for e.g. in luckynight.flac those entries are for multiples of ten seconds positions, making seeking to a more precise position a task of skipping frames (which is fun—see above).

• WavPack is still the best designed format in my opinion though it would be nicer to have some initial header with various metadata instead of having it stored in the first block. Other than that still no objections.
• And it turns out there’s lossless AAC compression that employs wavelet transform before LPC (it’s Chinese AAC though so who cares).

I remember reading somewhere (on Hydrogenaudio most likely) a brief story about development of several popular lossless audio codecs (even told by the author of one but I might be wrong). Essentially it’s not a NIH syndrome but very close: somebody develops a format, another guy finds a minor flaw the original developer refuses to address (my memory is hazy but I think there were such things mentioned as no plugin for some player or not supporting some tags) and develops another format. The amount of formats that came to existence because somebody wanted to create a format and could not keep it to himself is pretty large too.

But those days seem to be over and maybe I’ll reverse engineer some of those old codecs for documenting reasons as there’s very little risk that somebody would pick them up and make widespread now. Alternatively I can rant on newer formats sucking as well. Though why wait, let’s do it now:

• AAC sucks because of the countless extensions and attempts to bundle various coding approaches under the same name (fun fact – “xHE-AAC” is actually pronounced as “MPEG-D you-suck”);
• AV1 sucks because of the organisational structure and their decisions during (and after) the design stage;
• AV2 is not here yet but it sucks for the same reason;
• BlueTooth audio codecs suck in various ways (except SBC, it’s okay for the purpose), especially because of marketing them as high-definition and robust while in reality they rarely are;
• Chinese codecs suck for being rip-offs of better-known codecs. It’s especially gross that one of them got standardised as IEEE 1857.2 AAC;
• H.264 sucks because of countless extensions;
• H.265 inherited some from H.264 and added the licensing situation on top of that;
• MPEG-5 EVC sucks because it’s a Frankenstein monster constructed from bits from H.263-H.265;
• Opus sucks for being designed for streaming case and used elsewhere;
• Vector-based codecs suck because current tools are still not good enough to autovectorise complex shapes and recognize gradients.

Now back to doing nothing.

Posted in Lossless Audio, NihAV | 5 Comments »

OptimFROG

Saturday, March 26th, 2016

You know, the greatest reverse engineer I know is Derek B. He’s managed to RE such codecs as Canopus HQX and Cineform HD in the most efficient manner ever—saying he’ll do it and patiently waiting until somebody else does it.

So here are some words about his favourite lossless audio codec. The most interesting thing about it is that it was actively developed in 2001-2006 and then it was suddenly resurrected in 2015. Also it’s one of few non-standard codecs (i.e. not made into standard) that has several articles written about it.

The codec actually consists of two different formats, seemingly an old one and a newer one (that looks like it supports all range of sample type). The former is notable for having signal reconstruction stage using floating point math (a thing you don’t see in codecs every day), the latter seems to employ various parameter reading and reconstruction methods. Coding is done using low precision range coder (large values are decoded using chunks of 8 or 12 bits). So nothing really interesting there.

P.S. I’m definitely not going to write a decoder for it. There are too many lossless audio codecs already, let all proprietary ones (in custom containers too) die in peace.

Posted in Lossless Audio | 7 Comments »