Klarinet Archive - Posting 000408.txt from 2002/09

From: "Benjamin Maas" <benmaas@-----.com>
Subj: RE: [kl] MP3 - WAV - CD
Date: Fri, 20 Sep 2002 13:17:10 -0400

Well, Mark is the closest here, but not completely right.

> -----Original Message-----
> From: Mark Charette [mailto:charette@-----.org]
>
> Not quite. WAV is 16 bits of digital bits at (normally) 44.1
> Kbs/sec/channel. It's processed via a DAC (digital-to-analog converter) to
> get the sound back into an analog form. That gives 22.05 Khz
> analog/channel
> at near 98 db S/N ratio.

Let's start with WAV: WAV has become a generic term for any non-compressed
sound file on a computer. It started as a term for an 8 or 16 bit file on a
PC (as opposed to .aiff which started on the Amiga and was quickly picked up
by Macintosh). The sampling rate of that file could be at 44.1KHz or 48
KHz. (for those that don't know, this number refers to the number of
"pictures" of the waveform that are being taken each second-obviously the
higher, the more accurate and the ability to capture higher frequencies).

Bit-wise, all these formats are pretty much the same, except for a header
that tells the computer how to read the information. Today, WAV is not the
only format out there... BWF (broadcast wav) is the same header but includes
time information, SDII (sound designer II) is a mac format developed for the
ProTools digital workstation. It also contains time information. The
sampling rates for all of these files can go as high as 384 KHz with 32 bit
resolution.

To listen to any digital sound in any format, the signal will have to go
through a DAC... It is not just for WAV files. The Nyquest theorem does
indeed say that the sample rate is twice the highest frequency that can be
reproduced. In reality, few DACs (and ADCs) actually reproduce this as they
have filters built in to prevent any sort of aliasing at higher frequencies.
In reality, 44.1KHz sampling rates will accurately reproduce 20-21KHz but
not all the way up to 22.05KHz.

This is part of the argument for using high sampling rates on recordings
these days. The DVD-Audio format addresses these issues with a possible
sampling rate of 192KHz and SACD with the Direct Stream Digital 1 bit
3.884MHz (can't remember the exact number) sampling rate.

> CD format is very, very similar - the track formats are different, but the
> raw bits are just about the same. There's some extra guard tracks that are
> read to help guide a laser along (things that you don't have to
> worry about
> on a WAV file on a hard drive) along with land/pit variance (jitter, the
> bug-eyed monster of CD-R recordings. The CD-R can work fine for
> data and be
> unacceptable for sound playback).

The CD is identical information as your 16 bit 44.1KHz WAV or AIFF file. If
you have a WAV or AIFF file that is in any other bit depth or sample rate,
it will have to be dithered and/or resampled so that it fits this criteria.

Compared to the quality that most recordings are being made these days, CDs
are indeed lower fidelity as they are at a lower bit and sample rate than
the master. I personally do all of my recordings and processing at 24 bits
(or higher) and usually at 44.1KHz sampling rate.

> You description of MP3 is close - there's a lot of adaptive encoding going
> on that supposedly throws out frequencies that wouldn't be discernable.
> With the new encoders it's quite impressive. With the older
> encoders it was
> pretty dismal.

This is pretty much right.... I'd add that for you to hear the MP3, it will
have to be uncompressed and played through a DAC. The signal that is going
through that is most likely a 16 bit signal, like a CD.

That MP3 will *NEVER* sound as good as the original, though. MPEG-1 Layer
3 is a Lossy compression scheme. That means that the signal that goes in to
be compressed will not be the identical to the signal that comes out
uncompressed. The damage done to the signal depends on a couple things:

1. What data rate are you compressing to? Extremely low data rates will
decimate the sound. High ones (ie. 384KB/Sec) will generally sound quite
decent. While I can hear a great difference on my monitoring setup (very
high end), when I listen to MP3's that I've encoded on lesser systems, I
can't hear a difference. The trick is to figure out how much you can get
away with and still have it sound decent. I personally find that cutoff to
be roughly 128KB/sec.

2. Which encoder are you using? There are several different encoders out
there. I tend to stick to Frauhoffer (spelling?) encoding for my MP3s.
X-ing is another popular one. Each uses its own process to compress the
sound and have it come out as close as possible. This is largely a
subjective choice... I like purple, but you may like orange...

3. Are you doing anything to your sound before you compress into MP3?
There are processes that you can do to a sound to lessen the artifacts of
compression, especially when you are going to a low bit rate... If you want
these done to your MP3, hire me <G> (sorry... not going to let all my trade
secrets out...)

The good news is that while MP3 is not good for your music, there are other
formats that are available that will hopefully be adopted that sound much
better. MP2-AAC (Mpeg Layer 2, Advanced Audio Coding) is a large step up
from MP3. Half the bit rate can sound just as good... Meridian Lossless
Packaging is a Lossless data compression that is used for DVD-Audio.
Lossless compression means that what goes in comes out... Nothing is lost in
the compression process. For movies, you may also see Dolby Digital (AC-3)
or DTS. These are surround sound/theater compression schemes that take 6
channels of sound and fold them down to a managable data stream. In the
uncompression, they return to the 6 discrete channels. These, like MPEG
formats, are lossy compression schemes too...

Well, I hope I haven't confused anybody ;-)

Back to the clarinet....

-Ben

Benjamin Maas
Fifth Circle Audio
Los Angeles, CA
benmaas@-----.com
http://www.fifthcircle.com

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