Archive for the ‘Audio’ Category

Preserving extinct formats

Wednesday, February 27th, 2013

By the request of one guy (he has provided samples as well) I shall work on supporting old Monkey’s Audio versions (before 3.95).

Why? Because the latest official version of Monkey’s Audio has dropped support for those files, because I wanted to support such files since really long time (just didn’t have a good opportunity to do that) and because I definitely need a distraction from Go2Insanity codec (I shan’t blog about it anymore).

Well, let’s see what the old versions of the worst (known) designed lossless audio codec have to offer me.

Teasing

Saturday, October 6th, 2012

In the recent month I was not very productive, so I’d like to talk about codecs that I’m not likely to finish soon (not that I’m going to finish any codec soon anyway).

GoToMeeting 2/3

G2M2 decoder output (the best I could get)

Here’s the best output I could get from G2M2 or G2M3 data by decoding JPEG part of the tiles. ELS part still needs some work since it’s boring — 10-neighbour prediction, differential pixel decoding and other wonders of binary coder.

Certain Intermediate Codec

I managed to reverse engineer some parts of it. First you have so-called fixed header, then you have strip sizes and then strip data with some header as well. The way it’s coded is also more or less clear. But some connecting details — like how those strips are divided (now it looks like 96×1 macroblocks or equally ridiculous).

Since it’s QuickTime it’s hard to say where are the entry points to the codec and what functions are invoked.
Also the only usable binary (with debug symbols) is PowerPC only. It’s nice platform but I still need to learn some of its peculiarities.

VoxWare MetaSound

It turns out that it is slightly simplified variant of TwinVQ. It does not have variable-length codes, all values are read as fixed bits (depending on sampling rate and bitrate of course). The only catch is that it’s hard to find where such description is retrieved or generated. And existing codebooks are somewhat different.

Some Notes on Un-RE’d Codecs

Saturday, June 23rd, 2012

If I haven’t REd a codec that doesn’t mean I haven’t looked at them at all.
So today I want to talk a bit about some un-REd codecs and what peculiarities they have.

Looks like that all interesting codecs can be divided into three groups: screen codecs, intermediate codecs and speech codecs.
Since I don’t understand the latter group I shan’t give details on it.

Screen codecs

We have lots of them and they can be divided into two categories: simple and monsters.
Simple codecs usually employ some standard data compression library (zlib, FastLZ, LZO or LZF) or Huffman coding with standard median prediction and interframe difference.
I.e. boring, let’s talk about monsters.

  • Windows Media Video 9 Screen (aka MSS2) — combines palettised regions coded like in its predecessor with WMV9-coded regions.
  • M$ Expressions Encoder Screen (aka Titanium Screen codec) — it uses variable-length codes and codes frames with one of two methods. One of them is DCT exactly the same as in M$ ATC Screen codec.
  • MSU Screen Lossless Codec — this one seems simply code R,G,B values with some arithmetic coder and lots of context modeling and prediction.
  • Go2Meeting codecs — a good demonstration of the fact that the best strategy against REing is employing shitty coding monkey.
    Version 4 of decoder was monolithic 8 MB .dll file, version 4 is 15 MB already, all in “fine” C++.
    There are two compression methods known.
    Version 2 employs some weird arithmetic coder substitution (suspiciously like ELS-coder by Wm.D. Withers).
    Version 3 employs libjpeg and zlib for coding image blocks somehow, frame data doesn’t look like it at all.

Intermediate codecs

Cineform — looks like they use Huffman coding and wavelets and it codes 10-bit video.

Fruit Intermediate Codec — looks a lot like its successor (ProRes) but with different bitstream format and fixed coding scheme instead of adaptive ones.

BitJazz SheerVideo — the main problem with it is that most of the codec code performs conversion between any of couple dozens of formats (8- and 10-bit YUV and RGB packed in any possible way). Actual decompression code gets lost somewhere.

Some Notes on Indeo Audio (samples needed BTW)

Friday, June 1st, 2012

I’ve been working on this codec for a while and somewhat got it working.

Good news — it employs the same algorithms as its predecessor, except that it has stereo mode.

Bad news — it feeds slightly different values to those algorithms. So some tables used in calculations and number of free bits in the block (for allocation) differ. I’ve almost got it and hacked version of our IMC decoder outputs almost perfect sound. My suspicions are that it modifies original IMC tables for stereo mode case (since it codes audio in mid-side stereo mode it makes sense).

The problem is that there’s only one sample with this codec and it’s extremely short. So if someone has more files with Indeo Audio please provide them to us.

Codebook Hell

Tuesday, March 27th, 2012

There’s one codec I’d like to have reverse-engineered and implemented as an opensource decoder (well, lots of other codecs as well but this one particularly). Its name is VoxWare MetaSound, that’s an old codec which was used as an alternative to MP3 in old days of DiVX 3 😉 and its clones.

It’s definitely based on TwinVQ and is probably closer to the variant that got into MPEG-4 Audio standard (I suspect that mostly to make that standard even more bloated than before). That figures from having such modes like 8kHz/6kbps which is not present in VQF but present in ISO 14496-3 draft.

This codec probably has more data tables than TwinVQ (in binary decoder the section with codebooks is more than 256kB large, in TwinVQ it’s about 200kB) and should set a new record if we ever get a decoder for it.

Decoding looks very simple in theory: decoder initialises codebooks for given samplerate and bitrate (it’s actually signaled in extradata: VOXq for 44.1kHz/32kbps, VOXk for 16kHz/16kbps, VOXz for 44.1kHz/48kbps), for every frame it reads window type and an array of some values and performs reconstruction.

So far I was able to identify only some codebook information. Bark tables seems to be identical, but shape and whatever codebooks seem to be different.

I’ve spent a couple of evenings finding out that information and I dare someone (especially you, Vitor!) to write a decoder for it. I don’t know a thing about TwinVQ except one fact and it’s stated in the title.

Call for Intel Codecs

Monday, March 19th, 2012

I’ve spent two weekends and finally REd and wrote decoder for Re* Audio Lossless Format. With news like these I can deliberately call it Intel Audio Lossless Format.

So, what codecs we’re lacking so far?

  • Intel Audio Coder — it’s quite similar to IMC (Music Coder) but not identical.
  • Intel Layered Video Codec — probably it’s just h.263 variant, the only thing I know is that RealVideo 2 decoder was based on it (it’s mentioned in doxygen for Helix SDK I saw once in Internet somewhere and this supports that theory indirectly).
  • ClearVideo — a licensed fractal-based codec. It’d be rather simple DCT-based codec if not for one catch: it uses domain search to generate codes that then are used for block unpacking (and in decoder too, it seems). Maybe these patents will help?
  • Intel NGV — we’ll deal with it when it’s ready 🙂

Feel free to send any useful information about them, preferably working decoders of course.

After that we can claim full support of Real and Intel codec family.

Why Lossless Audio Codecs generally suck

Saturday, November 27th, 2010

Why there are so many lossless audio codecs? Mike, obviously, had his thoughts on that subject and I agree with my another friend who said: “it’s just too easy to create lossless audio codec, that’s why everybody creates his own”.

Well, theory is simple: you remove redundancy from samples by predicting their values and code the residue. Coding is usually done with Rice codes or some combination of Rice codes and an additional coder — for zero runs or for finer coding of Rice codes. Prediction may be done in two major ways: FIR filters (some fixed prediction filters or LPC) or IIR filters (personally I call those “CPU eaters” for certain property of codecs using it). And of course they always invent their own container (I think in most cases that’s because they are too stupid to implement even minimal support for some existing container or even to think how to fit it into one).

Let’s iterate through the list of better-known lossless audio codecs.

  1. ALAC (by Apple) — nothing remarkable, they just needed to fit something like FLAC into MOV so their players can handle it
  2. Bonk— one of the first lossless/lossy codecs, nobody cares about it anymore. Some FFmpeg developers had intent to enhance it but nothing substantial has been done. You can still find that “effort” as Sonic codec in libavcodec.
  3. DTS-HD MA — it may employ both FIR and IIR prediction and uses Rice codes but they totally screwed bitstream format. Not to mention there’s no openly available documentation for it.
  4. FLAC — the codec itself is good: it’s extremely fast and features good compression ratios. The only bad thing about it is that it’s too hard to seek properly in it since there’s no proper frame header and you can just hope that that combination of bits and CRC are not false positive.
  5. G.711.0 — have you ever heard about it? That’s its problem: nobody cares and nobody even tries to use it.
  6. MLP/Dolby True-HD — it seems to be rather simple and it exists solely because there was no standardised lossless audio codec for DVD.
  7. Monkey’s Audio — well, the only good thing about is that it does not seem to be actively developed anymore.
  8. MPEG-4 ALS — the same problem: it may be standardised but nobody cares about it.
  9. MPEG-4 SLS — even worse since you need bitexact AAC decoder to make it work.
  10. OggSquish — luckily, it’s buried for good but it also spawned one of the worst container formats possible which still lives. And looking at original source of it one should not wonder why.
  11. RealAudio Lossless Format — I always say it was named after its main developer Ralph Wiggum. This codec is very special — they had to modify RM container format specially for it. A quick look inside showed that they use more than 800 (yes, more than eighty hundred) Huffman tables, most of them with several hundreds of codes (about 400 in average). That reminds me of RealVideo 4 with its above-the-average number of tables for context-dependant coding.
  12. Shorten — one of the first lossless audio codecs. Hardly anyone remembers it nowadays.
  13. TAK — it was originally called YALAC (yet another lossless audio codec) for a reason. Since it’s closed-source and fortunately not widespread (though some idiots use it for CD rip releases), it just annoys me time from time but I don’t think someone will work on adding support for it in FFmpeg.
  14. TrueAudio (TTA) — I can say anything about it except it seems to be quite widespread and it works. Looks like they’re still alive and work on TTA2 but who cares?
  15. WavPack — that’s rather good codec with sane bitstream format too. Looks like its author invested some time in its design. Also he sent patches to implement some missing features in our decoder (thank you for that!).
  16. WMA Lossless — from what I know, it uses IIR filter based on least minimum squares method for finding its coefficients. It has two peculiarities: that filter is also used for inter-channel decorrelation and bitstream format follows WMA9 format, i.e. it has something like interframes and frame data starting at arbitrary point (hello, MP3!).

P.S. I still hope this post won’t encourage anybody to write yet another useless lossless audio decoder.

Notes on AAC quantisation

Thursday, March 19th, 2009

I should have written this earlier if not for non-FFmpeg work I have to do here. BTW, are some linguists around there that can explain a relation between bureaucratic and textile (“bureaucracy” comes from a sort of cloth used to cover tables, “red tape” is rather obvious, Russian “????????”, “????????” and “??????????” are also related to a process of obtaining thin threads). Ahem.

AAC coding has two computationally costly operations — MDCT and coefficient quantisation. While the former takes more cycles per one call, the latter is usually called several times for each frame, so those times tend to sum up and outweigh MDCT in bad encoders (like mine). From rate distortion theory we know how to determine proper quantizers for AAC – distortion caused by that quantisation multipled by lambda plus number of bits needed to code that band with this quantiser should be minimal for given value of lambda.

How could we achieve this? Well, use one of three approaches:

  1. Assign some fixed quantizers
  2. Use some ad hoc rule to determine quantiser and then refine its value a bit (aka heuristic, since it gives good speed, it is widely used)
  3. Try all possible quantisers by brute force or Viterbi method (optimal but very slow)

With heuristic you have one catch: if your primary guess on quantiser is not good then refining either takes a lot of time or gives you far from optimal result. Trellis-based search is implemented in my decoder and results in around 20x slower than realtime encoding speed (i.e. encoding one second of audio takes 20 seconds of CPU time) on modern CPUs. I’m playing with something heuristical and fast.

Now to quantising itself.

Each coefficient is quantised as out = (int)pow(in / quantiser, 0.75);. Division of floating-point numbers is slow, taking power of a number is even slower. You can convert MDCT coefficients to the power of three fourths (and quantisers are also converted in precomputed table), thus getting rid of power. FAAC also multiplies coefficients so they are always quantised except for taking integer part. My decoder just multiplies possible codebook vectors by that quantiser and compares it with input coefficients leaving them intact. I also had an idea to present MDCT coefficients in base pow(2, 0.25) making it easy to manipulate but someone still has to test it where base conversions won’t eat all of the gain. I have also tried several optimisations like not trying to match coefficients against all codebook vectors using only close enough vectors. More approaches to try.

(I hope these notes will form “How I Wrote the Best Opensource AAC Encoder Around (to Accompany x264)” memoirs :-S )

General psychoacoustic <-> coding interaction principles

Thursday, March 5th, 2009

OK, let’s suppose we have some abstract subband coder. What it does? It performs some transform on input block of data (like MDCT or QMF filterbank) then obtained frequencies are grouped, quantized and coded.

There could be many approaches but usually there are two general principles employed:

  • Some frequencies matter more than another.
  • Energy carried by subbands matters too.

Psychoacoustic model gives us a list of subband weights meaning their importance. Now what encoder could do with them? Quantize input data and code it. There are three approaches:

  1. Perform optimal coding using psychoacoustic data (good but slow)
  2. Do some heuristics to get some quick and dirty approximation (most popular approach)
  3. Ignore psychoacoustics completely (seems to be popular too)

Optimal coding may be done by employing Vitterbi method in one form or another. Heuristics are usually done in that way: give some initial prediction value for quantizer then refine it a bit until result is close enough to desired one.

More on AAC-specific coding later.

AAC encoder and psy model

Thursday, March 5th, 2009

As you may know, I am working (mostly NOT working though :(, but still remember about it) on AAC encoder. This morning I’ve made simpler psychoacoustic model inspired by FAAC (yes, Dark Shikari, FAAC has some sort of hardcoded psy model) work with my encoder.

I’ll try to use this blog with its original purpose — to formalize my thoughts on subject at hand. I thinks many posts on different aspects of psychoacoustics will follow before more or less suitable encoder will appear. “More or less suitable” means it should be at least a good audio encoding counterpart for x264 (while “fully suitable” means total world domination).

Too bad there’s not enough time (always).