I've been listening to digital music for many years through head phone's. And I've always understood that MP3 are good quality but if you want the best then you use Aiff or Wav.

I'm now starting to get a bit more into it, wanting to convert my CD's to digital for listening through my seperates system.

Clearly the best would be to use Aiff or Wav files, however massive file sizes put me off this option. Below this there are other lossless formats FLAC, Wavepac or apple lossless(which is the only one that works in iTunes), great sound quality but still a heafty file size.

Now there is the option between MP3 and AAC. I believe (but don't know) that AAC is better, I've heard it has a wider frequency range.

What about conversion bitrates? is 128kbps good, or just acceptable (for both formats). I'm using iTunes on a Mac to convert my music which gives the option of bitrates upto 320kbps, if I were to use these bit rates would the improvments be noticable through a decent sound system (yamaha amp, wharfedale speakers). And would the file sizes at the really high bitrates mean that i might as well have used a lossless format?

Could one of the resident experts give me some sound advice please.


Thanks for any help you can offer.

Depends if you need to transcode or not. If not, You could use an all-in-one solution like mp3 / aac for all listening enviroments. You need to do some abx testing to find the point where common annoyances start to resolve then stick with that or use some more headroom (if 128k is good then use 160k~190k or so). This doesn't take into account rare bad cases (killer samples) where even those settings might not be enough.

Another 'PC based' quasi-lossless solution is a hybrid encoder like wavpack or optimfrog dualstream. Performance is very good at 350k and also suitable for transcoding to other lossy formats. You can also store a correction file on DVD for future lossless restoration. My other option is an 'End To All - Armaggedon' bitrate of 500~550k with hybrid encoder, Otherwise its lossless.

Any modern format (AAC, MP3, Vorbis etc) using a good recent encoder will sound very good at 160 Kbps VBR or above. This setting will be transparent for most listeners when using most real world music samples. Most of the challenge these days is making encoders even better at 128 Kbps and below. Good modern encoders make good files at 160 Kbps and above, so there is less attention paid at such settings.

Hence a decision between MP3 or AAC really just comes back to where you intend to play the files. MP3 is much more broadly supported than AAC, but if you are using iTunes and / or an iPod then AAC is a viable choice.

Another thing to think about is that the while the AAC encoder in iTunes is excellent quality, the MP3 encoder in iTunes isn't. The best quality MP3 encoder is still probably LAME. The current stable version is 3.97 you can use it within Max. Or there is an older version, 3.95.1 inside the iTunes-LAME plugin. However, the 3.97 version is better quality, and is recommended by this forum, at least until a stable version of 3.98 is released.

Regarding the bitrate. For most people 128 Kbps VBR is transparent (indistinguisable from the original source file). For LAME MP3, the setting -V5 --vbr-new will produce files around this bitrate. If you want larger files, then use a LOWER -V preset. For example, -V4 targets around 165 Kbps, and -V3 targets around 175 Kbps.

For iTunes AAC, select 128 Kbps, then check the VBR encoding box. By default, iTunes uses CBR encoding, but VBR encoding is a bit more efficient for about the same bitrate. It will average around 135 Kbps. To turn on VBR, you need to go to the custom encoding menu, which lets you change the bitrate. Just check the box next to Use Variable Bitrate Encoding.



If you want better quality with iTunes AAC, you'll have to change the bitrate to 160 Kbps. With the VBR mode on, this will make files that average around 165 Kbps VBR.

128k isn't trasparent to me in abx tests. With mp3 only at -V2 I start to get stability and -v4 is also nice. Using --lowpass 16 or -Y , I can get a bitrate of 170~190k with -v2 and usually only killer samples are abxable and what version encoder you use doesn't matter much. At lower settings a wide range of samples is abxable with moderate concentration. I suggest for the clueless to run abx tests for a while starting at 128k.

If you're on a Mac, you basically have two choices: iTunes or Max from Stephen Booth.

Max extends your options somewhat, since it will also allow you to encode into FLAC and LAME MP3.

Uncompressed -- Uncompressed is only necessary for actually editing the audio data itself, which you probably don;t want to do. Otherwise, it's pointless, since losless formats really are lossless: all the information is there; it's just compressed (rather like a zipped Word document).

Lossless -- If you want lossless, FLAC may be an attractive alternative, because there's wide support across every OS you might be likely to use. It's becoming something of a de facto standard for lossless. However, on OS X--currently--you would need a third party player to play it back, such as Play. That's an interesting program (with some nice features in the pipeline) but in its early days yet. I guess here you'd need to make make a format choice based on whether you'd prefer a perhaps slightly more future-proof format versus a more mature and full-featured player.

Lossy -- the CoreAudio AAC encoder in general gives better results than MP3 as you noted. But the LAME MP3 encoder (which Max uses) is very good--comparable to AAC. So either AAC or LAME Mp3 would be fine. AAC is MP3's successor, so I suppose it's slightly more future-proof but currently not as widely supported across platforms/devices. Again, it's a toss-up based on what devices you have or what player/s you want to use.

Bitrates -- There will be people here who'll tell you to do listening tests to determine whether or not you can hear a difference. Done properly that's an involved process; also I don't know of any software on the Mac it's easily done with. Frankly, I tend to go with the defaults. With LAME MP3 you'd just set the encoder to "Transparent" and it will give the material what the engineers who developed the encoder consider an appropriate bitrate for it. That varies, because it's meant to, but will be around 190kbps.

The iTunes encoder defaults to 128kbps, but that's probably because they've chosen a setting optimized for the iPod--and in the days when those had smaller drives at that. The Nero AAC encoder for Windows defaults to a quality setting that's equivalent to 160-170kbps, and that's what I'd use for iTunes, too. Just set the bitrate to 160kbps and check the VBR box, so that the encoder will use a slighty higher bitrate where it determines it needs it.

Personally, I would be happy to use either 160kbps VBR AAC iTunes or MP3 encoded at the "Transparent" (190kbps) setting in Max. If you do hear deficiences, you can encode at higher bitrates/quality levels and see whether those are gone. I wouldn't use very high bitrates like 320kbps with a lossy codec myself, because those codecs are intended to give you a lot of bang for your buck and not waste space. And if space is not a problem, one might as well use lossless as use very high bitrate lossy.

Caveat: You say resident experts. I'm anything but an expert; these are amateur observations.

I would stick to mp3 for an all-in-one solution. AAC doesn't hold an major advantage at higher bitrates nor hardware compatibility.

Thank you gents, for all your advice. You've been a big help.

Really?? "Most people"? I'm a 55-year-old professional musician - a brass player - who has suffered the usual occupational hearing losses over the years, and I still don't need ABX to tell me that 128kbs is not transparent any more than I need ABX to distinguish an oboe from a trumpet! That is one of the two main reasons I never buy music from iTunes (the other is that nasty DRM). Through decent or excellent buds (buds that cost as much or more than the iPod itself), I find 192kbs to be acceptable - unless I ABX with Apple Lossless. The difference is not only mathematically real, it is audible - through the aforementioned high-quality buds, that is. But even through the crappy buds that come with iPods, the quantization noise is noticeable and annoying with any decent quality source material.

That is NOT to say that others may not find 128kbs to be "acceptable." But I just have a very hard time believing that anyone with moderately decent hearing could not hear the difference between 128kbs and say, Apple Lossless, in an ABX type of test if not just casual listening.

Hard drive space is cheap. I rip to lossless. That way, I can stream the files through my home stereo without fear of cringe-worthy audio.

Based on the listening test http://www.hydrogenaudio.org/forums/index....showtopic=40607 it seems that 128kbps is almost transparent for most people. If you didn't participate in that listening test then it is really bad, since you would probably contribute to better ratings (this is of concern for me because I am working on software for objective grading of audio quality and precise results from listening test are precious to me).

Possibly not, but where ABX is really useful is to back up statements like...


Not all ~128kbps samples are equal to all sets of ears. There are too many variables excluding equipment or listening environment: what samples are used? What codec and encoder is being used? CBR or VBR? how much training has the listener had to detect artifacts (be it through profession or self-study)?

For the record, I have never ABXed Nero AAC @ -q .4 (~125) or Vorbis @ -q 4 (~128) with samples from my own collection. I've managed Vorbis once or twice with test samples. For said test, I was using a pair of Sennheiser CX-300s out of an m-audio Revolution 5.1 sound card. The self-study training I have done is minimal, and I have yet to get past 'Difficult' in the first column here. When I revisit 'Harder', it is still a considerable effort.

All that said, I agree with the peace-of-mind that comes with lossless (some samples are not transparent at *any* bitrate for a given encoder, and who knows what's lurking in my 16000+ tracks of goodness?), but I've achieved said peace-of-mind with Vorbis @ -q 5 (~160) and lossless backups.

It's worth mentioning that one can easily upgrade the encoder used by the iTunes-LAME plug-in. You just need to download the 3.97 binary for Mac and stick it in the "Resources" folder that you find when you right-click the iTunes-LAME plug-in in Finder and choose "show package contents."