Audio Compression Primer

Hack Jandy writes "For those of you with a little extra time this afternoon, check out Sudhian's primer to all things concerning audio compression. The article details everything from DRM to CRC matrixes (with a healthy dosage of Ogg)."

Comments

Is FLAC worth it?

[Megaweapon]

"FLAC is the Linux users lossless audio codec of choice"

Unless your doing some form of audio editing or "production" recording, is lossless really worth the extra size compared to a 192kbps Ogg or MP3? I usually have more problems with static from the stupid 3.5mm jack than a lossy format.

Re:Is FLAC worth it?

[stratjakt]

Most of us aren't exactly audiophiles.

I'll go stereo to mono and reencode at 22khz for my tv captures. It sounds the same to me.

As for mp3s, etc, the only time I ever listen to it in the car, and there's so much ambient noise, it's not worth bothering. Hell, 128k joint stereo sounds like the CD to me, I don't know any better.

I don't listen to much music anymore. All the bullshit and RIAA and this is legal and blah blah blah, it's all killed music as an artform for me. I used to play guitar in bands, and love playing music. It's just dead to me now. White noise.

Re:Is FLAC worth it?

[itp]

I keep my entire CD collection on disk as FLAC, and then transcode to the lossy format(s) most useful to me at the time (currently Vorbis to play in my Rio Karma). If I ever need a new format I can go back to the FLAC and reencode without transcoding from another lossy format.

Virtually dismisses lossy compression

[Sanity]

This article doesn't seem to talk much about ogg at all, unless I am missing something, in fact, it virtually dismisses all lossy algorithms in favour of lossless algorithms which achieve only 50% compression (instead of 90% compression with lossy).

Each to their own, but I am more than satisfied with oggs or mp3s encoded at a reasonable bitrate - I think the popularity of hardware such as iPods suggest that most other people are too.

Re:128K should be enough for everyone

[aceh0]

FM Radio is far from CD quality hence there isnt really a need to use very high bitrate MP3s or whatever

Waste of time . . .

[barryman_5000]

Not very informative for slashdot ppl. I think we should have had an article more about code or something. I think most slashdotters understand codecs and the differences in lossless and lossy compressions. Waste of 15 minutes.

AAC

[sometwo]

So what about AAC used by Apple in their music store?

I did a little googling and found this (http://www.teamcombooks.com/mp3handbook/13.htm):

AAC (Advanced Audio Coding) is not a MPEG layer, although it is based on a psycho-acoustic model. Sometimes referred to as MP4, AAC provides significantly better quality at lower bit-rates than MP3. AAC was developed under MPEG-2 and also exists under MPEG-4.

AAC supports a wider range of sampling rates (from 8 kHz to 96 kHz) and up to 48 audio channels, plus up to 15 auxiliary low frequency enhancement channels and up to 15 embedded data streams. AAC works at bit rates from 8 kbps for mono speech and up to in excess of 320 kbps for high-quality audio. Three profiles of AAC provide varying levels of complexity and scalability.

AAC software is much more expensive to license than MP3 because the companies that hold related patents decided to keep a tighter reign on it. Most AAC software is geared towards professional applications and secure music distribution systems, so it may be a while before you see AAC in consumer-oriented products.

Re:AAC

[Skuto]

AAC is *much LESS* expensive than MP3. Just compare the licensing costs from Vialicensing (AAC) vs Thomson (MP3).

The parent is plain wrong. ("Don't believe all you read on the internet, kids")

more algorithms

[barik]

While the article is a primer, I was a little disappointed in the algorithmic treatment given in the article itself. Right now I know of two excellent free publications: Introduction to Sound Processing and The Sounding Object, which both treat the theoretical, DSP side of things. Any other resources that Slashdot readers can recommend for those who are interested in the subject of audio compression and representation?

128/192 kbps is enough for everyone...

[katharsis83]

I second that.

On repeated double-blind tests on very expensive equipment, even audiophiles are unable to distinguish between CD quality and LAME encoded 192 kbps MP3 files. Those who say they are able to aren't using double-blind tests or have super-human mutant ears. If you go check over at Hydrogen-Audio (where audiophiles and people who care far too much about LAME settings hang out), most of the forum posts indicate that anything above 192 kbps is transparent even to their equipment, which is pretty above average.

On regular equipment, PC World did a small test a while ago on standard equipment: http://www.pcworld.com/reviews/article/0,aid,64123 ,pg,1,00.asp.
Their results found that ~192 kbps is pretty much transparent as well.

mp3-tech.org also has a listening test availible. On their run, they found 192 CBR kbps to be nearly transparent (*feels* different, but don't know why), and 256 kbps CBR to be completely transparent (can't tell compressed from source CD).

"The listening equipment is the following :

* Teac VRDS 25 CD reader
* MIT T2 cables
* Yamaha AX 1050 amplifier
* Denon PMA 960 amplifier (for frequencies 50Hz)
* Celestion speakers"

This test was also done a while ago on an older mp3 compression program( c. 1998), so current LAME encoding probably allows for complete transparency at 192kbps or so.

Re:128K should be enough for everyone

[pthisis]

especially when listening to music on hi-quality speakers a la Bose

Bose is doesn't make high-quality speakers, they make expensive speakers that don't perform nearly as well as alternatives (for instance, the Acoustimass satellites use crappy paper cones that perform poorly in the upper frequencies). A $300 pair of B&W DM302's will thrash anything Bose makes soundly for sound quality. Also investigate Hale, Thiel, or Paradigm. If you really want to spend thousands, spend it on Magnepan (Magneplanar 1.6Q) or Vandersteen (2ce signature) or the higher end speakers from the companies I already mentioned. But those DM302's are good enough to be highly rated by places like Stereophile magazine and they're an incredible deal.

If you really want a bunch of little satellite speakers, Energy makes a much better sounding (and somewhat cheaper) system like that. I hear from people I trust that Tannoy makes an incredible one as well, but I haven't heard it.

Re:I still hear MDCT distortions

[radish]

Not wanting to get some award for pedantry, but all music recording is "lossy". If you listen to a CD, you're not hearing the exact same sound you'd here in the studio, those cymbals sound diffrent due to sampling, quantization etc. So when it comes to "lossy compression" causing "artifacts" - it's only creating different artifiacts, there already were some.

Of course this doesn't go against what you're saying at all, other than calling FLAC "perfect" is wrong. It might be the same as the CD, but that has it's own problems.

ARRRG! He gets Nyquist WRONG!

[wowbagger]

According to the "Nyquist Theorem," you need to have twice as many digital samples as the frequency of the analog signal you are trying to represent to have enough data to accurately build it.

WRONG!

Nyquist's criterion is "You must have at least twice as many samples as the largest BANDWIDTH of the signal in order to correctly reconstruct it."

You can take a 10.7 MHz signal, and sample it at 10000 samples per second, and correctly reconstruct it, so long as the signal is guaranteed to be bandwidth limited to 10.7 MHz +/- 2.5 kHz. This is often done in software defined radio to aquire the signal from the intermediate frequency (IF) of the analog front end.

You also have to have an appropriate reconstruction filter at the output of the system in order to correctly recover the signal - if you don't have the right reconstruction filter, you will NOT reconstruct the signal correctly.

You also have to take into account the effects of any signal modulation - take a 20 kHz sine wave, and burst it for 10 msec, and you widen the bandwidth of the signal by about 100 Hz (depending upon the exact shape of the burst - a perfect square burst will widen the signal as a sinc function and will, in effect, increase the bandwidth to infinity, which is why square bursts are generally Considered Harmful in communications work).

Also, you don't oversample a signal in time to account for "rounding errors" - you oversample in time because the frequency response of sampling a system in time introduces a sinc response in frequency - by moving the sampling rate up you reduce the impact of this response on the recovered signal's frequency response. You also greately ease the requirements on the reconstruction filter - the filter can be wider (have fewer poles in the transfer function - thus fewer parts needed).

FLAC is often worth it.

[Venner]

I personally think 256kbps or even 192kbps is good. But it depends on your output (speakers, headphones) and more importantly your ears. Some people don't mind 92kbps while others won't settle for anything less than vinyl (usually people with $30k+ wrapped up in their setups...)

I have a decent mid-range receiver & set of speakers*. I had a friend of mine administer a blind listening test on me. I could pick out the FLAC encode vs. the Ogg "higher quality" (I think it was -q7 or -q8) encode about 75% of the time.

Most of the time I am content with a good Ogg encode (I mean, hell, I'd never have heard the difference if the samples weren't played back to back!) I generally only use FLAC for a) my favorite albums and b) classical music. Size wouldn't be an issue... but for the fact that I keep an oft-updated mirror of the data on a second computer. As drive space is become rather inexpensive, I forsee a time when lossless will be the way to go, except for portables.

*Ascend Acoustics CBM-300 stereo pair, HSU sub, and a HK AVR-325 receiver.

Re:Make the right choice. Choose Vorbis.

[valkraider]

I had a Vorbis listening party this past Summer at my home.

But no one came.

Hey! Me too!

[Just Some Guy]

I do the exact same thing, except that I keep my entire CD collection on CD. If I ever need a new format, I can go back to the CD and reencode without transcoding from another lossy format.

I've got about 350GB of lossless audio goodness in a set of nice oak bookshelves built into my wall. Considering that the time it takes to get up, get a CD, rip it, and encode it is not much longer than it takes to locate a FLACed album on my fileserver and encode it - that is, the encoding stage is several times longer than the "get up and rip the first track before starting to encode" phase - I think I'll stick with my current system.