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In the past three years, I have been converting shows from DAT and MD to CD. I have even spent some time doing analog conversions to CD. Since I started, this particular aspect of taping/trading has truly exploded. When I started, there were virtually _no_ DMB shows on CD. Now, there are probably a couple dozen people who are actively and rather prolifically converting shows of DMB alone from DAT to CD. It's quite impressive.
Accordingly, it has become very important to disseminate information about how to _properly_ convert shows.
I do not claim to be an authority by any means on anything, much less on dat>cd conversions specifically. However, having used several digital audio cards on both PC's and Macintosh machines, I felt I should at least write something up. That, and a friend asked me to :-).
Getting Started
Let's define some terms
First, what you need
How do I hook this stuff up?
What kind of digital connections are there?
How can I recognize one from the other(s)?
What kinds of cables are used to connect things, digitally?
Okay, so I got a digital output. What is a digital audio
card and how is it different from my Soundblaster?
Do I have to open up my computer and put in a digital audio
card? Are there alternatives that are less complicated?
What about my laptop? Can I use something with that?
How do I convert between the coax and optical (ie-my cd player
has an optical output, but the soundcard has coax)?
How do I connect my DAT deck up to the sound card?
Philosophical and nuts/bolts side to conversion
I've heard I shouldn't convert from MD/analog to CD. Is
this true?
Is it bad to convert analog>CD? Is there a "right" way to
do it?
Why can't I just use my regular sound card?
What is the sampling rate and why is it important to conversions?
How can I convert the sampling rate?
Which is the best?
What digital audio cards can resample in real-time?
What about after-the-fact resampling?
Can I buy an external sampling rate converter?
What are some good sound editing programs?
How much does all this stuff cost?
Okay, so it's hooked up and I understand the philosophy.
What next?
How should I transfer the audio to my hard drive?
How much space will I need to record this audio?
How do I cut the show into tracks?
What's the best approach to cutting tracks?
How do I determine what tracks will fit on a 74 minute
disc?
What if the levels are too low after I've put them onto
the hard drive?
What else do I need to know?
First, let's go over some terms you will need to know. Converting from one kind of media to another is a complicated process involving all kinds of differing media, standards, connections, cables, and tons of other things. All of these are defined on my Taping & Trading Glossary page. It would be wise to familiarize yourself with these terms as I will not go over them again during the rest of the FAQ. Some of them are:
- Media
The main types of media being dealt with in this FAQ are Digital Audio Tape (DAT), Compact Disc Recordable (CDR), and Minidisc (MD). CDRW procedure can be considered basically identical to that of CDR. Analog media, such as cassettes, will also be discussed to some extent. - Connections
There are different types of connections that, to a degree, greatly complicate the process of converting shows. The major ones are coaxial and optical digital connections and unbalanced RCA connections. Also of importance is the Sony "7-pin" connector.
As a matter of habit, when I say "tape," I am refering actually to any recording of a show, whether it is on MD, DAT, analog, or CD. They are all "tapes."
This FAQ assumes conversion is being done on a computer. This FAQ is also basically a conversion guide for PC's and Windows.
First, you need a source with a digital output. This means a dat deck, an md home deck (portables don't have digital out), or an analog-to-digital converter with your analog source plugged into it.
Next, you need a computer with a digital input. That means a digital audio card or some other digital audio input device. Put a bunch of cables in there and start up some wave editing/recording software and you're all set. Well, it's not quite that simple...
What kind of digital connections are there?
There are three major kinds of digital connections. The first two are
part of the Sony/Phillips Digital Interface Format, of S/PDIF. They are
the two most common connections you'll find.
- Coaxial - this is the most common. Electrical signal with a charge being a 1 and the absence of a charge being a 0.
- Optical - Popular with Sony. It uses a red LED and fiber optics to send a kind of morse-code signal to transmit the data.
- AES/EBU - uses XLR connectors and is borderline compatible with S/PDIF. The nice thing about this one is that it does not transmit SCMS, and is especially popular among "pro" audio engineers and tapers.
- Coaxial - the connector looks like an RCA jack, except it's usually not color-coded and there is only one jack.
- Optical - the most common connector is a Toslink socket, which is square-ish. You can also tell an optical output because it's emitting red light.
- AES/EBU - Big XLR connectors, and there is only one per direction (where as analog XLR's usually come in pairs)
- Coaxial - a coaxial cable with rca plugs at the end. Basically, any 75 ohm cable with RCA's will work. Coincidentally, 75 ohms is the standard for video cable...so you can get a "coaxial digital cable" for about $3 a meter. Or you can spend $40 on the Monster Cable one...
- Optical - needs a fiber optic line, usually with Toslink connectors on it. Places like Core-Sound and The Sound Professionals carry them at good prices.
- AES/EBU - to be honest, I'm not sure. These need to be wired properly so the right signal is being sent down the right pin. Any help, anyone?
Okay, so I got a digital output. What is a digital
audio card and how is it different from my Soundblaster?
A digital audio card is an expansion card, installed in your computer,
that can accept digital audio. As far as digital audio goes, it acts very
similarly ti a regular sound card, except that the data going through
it is already digitized audio - 1's and 0's. The data goes directly to
the hard drive without going through any (noise-prone) conversions to
and from analog.
A regular soundcard, like a Creative Labs Soundblaster, takes an analog signal and converts it to digital inside the computer.
Most digital audio cards, by the way, also have analog inputs. So they are both digital and analog soundcards.
Do I have to open up my computer and put in a digital
audio card? Are there alternatives that are less complicated?
Recently, several products have come out that are alternatives to the
above scenarios. The proliferation of the Universal Serial Bus (USB) and
Firewire ports on computers has opened the door to new ways of getting
data into computers.
While the only Firewire device I've seen remains the MOTU 828, which is clearly a studio-level device, there are a couple USB devices on the market. The Opcode DATPort was the first, though I think the company went out of business recently. Roland and Tascam have also recently come out with products.
The nice thing about USB and (potentially) Firewire devices is that they are external. The inherent flexibility of the USB and Firewire protocols means that all you have to do is install the appropriate drivers and software, plug the device in, and you're set. It's really quite convenient, though the viability of these interfaces for large, 2-3 hour audio files (1.2-1.8GB of 16/44.1 data) remains tricky.
What about my laptop? Can I use something with that?
Along with the above mentioned items (assuming your laptop has a USB
port), the VXPocket device has recently been released by [fill in]. This
is a truly powerful PC Card device, supporting high quality analog and
digital input with multi-track support. It's pricey, though not overly
so by audio standards at $500. Considering how powerful it is, there really
isn't anything holding one back from having a truly portable recording
and CD mastering station.
How do I convert between the coax and optical (ie-my
cd player has an optical output, but the digital audio card has coax)?
You need a converter somewhere in line between the source and the sound
card.
Places such as The Sound Professionals carry converters for about $60-$150.
Alternatives would be to get some kind of digital deck, like a DAT or MD, that has different connections. Plug the optical from, say your CD player, into the DAT deck, and plug the DAT deck's coaxial output into the soundcard. Off the top of my head, Sony's have both of these, and the Fostex D5 actually has optical and AES/EBU connections.
How do I connect my DAT deck up to the sound card?
Well, I'm assuming that, if you got an optical output with a soundcard
that can take optical, you can figure that out. Or, if you got an optical
out and a soundcard with coax, that you now know how to get a converter.
So, the real issue becomes the "non-standard" connections. Basically,
that means the Sony Portable DAT 7-pin digital i/o socket.
Sony, in its great and utterly illogical wisdom, decided that, even though they helped design the S/PDIF standard, they would use a non-standard plug in their portable DAT recorders. To interface this 7-pin outlet with a digital sound card, you need a conversion cable. These can be gotten from places such as Oade Bros., or American-Digital.
Now, not only is the socket itself a different connector, but the voltages aren't quite right, either - it is possible that the receiving device may not be able to lock into the signal. In reality, most devices that obey S/PDIF are able to reconcile this problem, but...you may want to consider getting an active cable versus a passive one.
Passive cables do nothing to the signal voltages. Core-Sound has these, and they tend to be less expensive. Active ones will convert to S/PDIF levels. Oade Bros is the only one I know that makes a fully active one.
Having just explained how to hook a sony portable up to a soundcard, I am now going to say that you probably do not want to use your portable sony for conversions. Invest in a good home deck - it'll last longer, honestly.
Philosophical and nuts/bolts side to conversions
I've heard that I shouldn't convert from MD/analog
to CD. Is this true?
This comment is usually made on the grounds that MD and analog are inferior
to DAT. Bottom line, they are. However, the other bottom line is
how the show sounds, and how the tape compares to other sources out there.
If it's the best sounding one and you properly document its lineage, then
I think it's fine to convert it.
If you have a digital source, then you just need to transfer the bits over. That means a digital audio card.
If you have analogs, it's a different story....see the next question.
Is it bad to convert analog>CD? Is there a "right"
way to do it?
It is not inherently bad to convert analogs>CD, no. However, if there
are DAT copies out there, making CD's from an analog source is only going
to pollute the market, because there is almost always a better sounding
copy out there. Therefore, only extremely rare analogs, and very high
quality ones at that, should be ever considered for conversion.
Another big problem with analog signals is that they are susceptible to the electronic noise and interference found in a computer. The best solution to that problem is to use an outboard analog-to-digital converter. These can be found from many different companies and in a wide range of prices.
Why can't I just use my regular sound card?
Computers are very noisy machines, electrically. When I plug headphones
into my SoundBlaster AWE64 Gold, one of their "high-end" cards, I can
hear a noticeable hum in the background from all the electrical interference
in the machine. If you were to convert an analog signal to digital inside
a computer, you pick up all that noise and get a hissy copy.
What is the sampling rate and why is it important
to conversions?
The sampling rate is the number of samples a digital recording has per
second. There are different ones out there - CD's and MD's are standardized
at 44.1KHz, DAT's can be 32, 44.1, or 48KHz, and DVD-Audio, a new standard,
can be all the way up to 96KHz.
The benefits of higher rates is greater resolution of sound - technically, the high end comes out a bit better, and the sound is more nuanced.
The sampling rate is very important with doing conversions, especially since DATs can be one of three different rates, but CD's only one. Very simply - rates are not compatible with each other. Machines that can only read one cannot read others. A CD must be 44.1KHz or it will not play in cd players.
How can I convert the sampling rate?
There are a few ways to get the sampling rate from one to another.
- "Real-Time" - this is when the conversion is done while the audio is streaming into the computer through the card. This can be done either with hardware (with a sampling rate converter or by the audio card itself) or with software (Samplitude, from SEK'd, is a software application that can resample in real-time)
- After the fact - this means the audio is recorded onto the computer at whatever sampling rate the tape originally had, then converted afterwards using software algorithms.
Which way is the best?
The jury is still out as far as a final answer goes, as there are
varying degrees of quality with both approaches.
Most hardware resampling, with two glaring exceptions, is generally considered to be of "pro" quality and is fine. The two major exceptions are the chips on the Creative Labs Soundblaster Live! and Turtle Beach Montego II sound cards, neither of which are "bit-accurate" - they do not capture all the bits while working with digital audio. Samplitude's software real-time resampling, however, is considered to be of "pro" quality.
After-the-fact resampling is technically always going to be better than
real-time because more complex algorithms can be applied. However, this
also leads to a greater variance in quality. I have read that Samplitude's
and Cool Edit Pro's resampling are the best, while Sound Forge is somewhat
lacking in its qualities.
What digital audio cards can resample in real-time?
The major digital audio cards that can resample in real-time are:
- Zefiro Acoustics ZA2
ISA, 20-bit, coax, optical, aes/ebu connections.
For the longest time, this was the standard. Tremendous resampling quality and handled just about anything, though it had a few glaring bugs with video conflicts - SEK'd Prodif XX
PCI, 24bit, coax, optical, adat
This covers the entire Prodif line of cards. Using the Samplitude software, these cards can resample incoming digital audio. Very robust. This is the card I personally use and recommend. - Soundblaster Live! and Turtle Beach Montego II
PCI (don't know anything else about them)
These cards are probably the most prominent digital audio cards, in terms of mass consumer market. These cards do resample, which is actually very nice, but both cannot capture bit-accurate streams, meaning data is lost. The Live!, in addition, also adds noise to the signal as it comes in through some sort of default resampling process. These cards should be avoided if possible.
What about after-the-fact resampling?
The advantage of resampling after transfering the audio to the computer
is that acoustically better-sounding algorithms can be used. The downsides
are that it can take quite a while, especially at the highest quality
levels. I have a friend who converted after recording using Pro Tools
on a Mac - at the highest setting (better than the ZA2 can do in real-time)
it took over 17 hours. Even if it takes 3 hours, that's time spent after
you have already recorded the music onto your drive.
Can I buy an external sampling rate converter?
Yes - there are a few. I knew Behringer makes one that also happens to
be an SCMS-stripper, but I don't have their website handy. It's the SRC-2000.
Alternatively, it is interesting to note that MD home decks have converters built right into them. You could, conceivably, plug the source into the MD deck, then the MD deck into the sound card. Just set it up so the data goes right through the MD deck, gets resampled, and is then sent to the computer at 44.1KHz.
What are some good sound editing programs?
Technically, a digital audio card acts very much like a regular sound
card, except the audio is digital. Therefore, provided the card is installed
properly, any sound recording program, even Windows' Sound Recorder, will
be able to capture audio.
Software I have used:
- SEK'd Samplitude
Powerful program, and some audio professionals say it even sounds better than others (not sure about that). Can resample in real-time. Not as intuitive, though. Multi-tracking and mixing tools are poweful. - Sonic Foundry's Sound Forge
This is almost industry standard in some respects for 2 channel, stereo work. Very powerful and robust, and easy to use. - Syntrillium's Cool Edit Pro
One of the most user-friendly programs I've seen. Does everything you need and has some tools that just make a lot of sense.
How much does all this stuff cost?
Off the top of my head:
- Zefiro ZA2 - $420
- DAT Deck - Sony R300 is the least expensive at $600. "Pro" decks are usually over $1000.
- cables - you can spend $10, or $100, depending on how much you believe the hype
- 7-pin cables - $50-$150, depending on passive or active
- Full-featured sound editing software - usually around $300
Okay, so it's hooked up and I understand the philosophy. What next?
How should I transfer the audio to my hard drive?
Obviously, you need to get audio on your hard drive, but should you
record the whole show? A track at a time?
Since you can do all kinds of editing with digital audio, it's best to start big and then go small. So go ahead and dump that whole 2.5 hour show onto your hard drive. Hit <Record> and go watch a movie.
How much space will I need to record this audio?
1 minute of digital audio takes up exactly 10mb of space. So if the
show is 120 minutes (2 hours), then it will take up 1.2gb of hd space.
How do I cut the show into tracks?
Your main two options are to either just do a lot of cutting and pasting
with the main sound program (such as Sound Forge), or to use CDWav.
The former is a bit tricky both because it's tedious and because of "sector-boundaries." Basically, one needs to cut tracks along sector-boundaries, which must be a multiple of 588 samples. If your program is set for this (74 frames of audio per minute is cd audio standard) then your edits will be fine. But if it's not set that way, you will get little clicks in between tracks. Cool Edit Pro and Sound Forge do NOT cut on sector boundaries by default.
CDWav is this great little program that takes an audio file, lets you put in "splits," and automatically cuts up the big file into smallers ones along the exact sector boundaries needed for a perfect audio CD. It's great, and handles even big files with aplomb. It can be downloaded here.
Just as an aside - Cool Edit Pro allows you to save selections. You highlight a song, then hit <Save Selection>. Much nicer than constantly doing cut/paste. But you still deal with the sector-boundaries issue.
What's the best approach to cutting tracks?
This is very much a personal thing. While one would like to have standards,
it's really up to the converter how he/she best feels tracks should be
cut. I can only say what I do:
- Following my friend Mike Vernal's lead, I make an "intro" track before the first song. That way, you can jump immediately to the first song but the disc doesn't start right off with it, and there isn't any fading into the song, only into the ambiant noise before the song.
- Songs are cut with about a three second lead. This is both because CD players can sometimes be a bit off when locating a track and because it's a bit disrupting to advance a track and have the music just explode right off the bat.
- I cut a track if an extended solo ends a song. For instance, if Victor Wooten goes into a 15 minute bass solo at a Flecktones show that is 1) not just a jam in the middle of a song and 2) is long enough to basically be a song unto itself, then it gets its own track.
- Medleys are consolidated medleys, not consecutive mini-tracks.
- I fade in/out only for the beginning and end of the entire show, leaving the encore break in, ambiant noise and all. The only times I crossfade on the encore break is if it's exceptionally long and/or there is excessive crowd noise during it.
How do I determine what tracks will fit on a 74
minute disc?
First, use your best judgement; if it makes sense to you, then it
probably will to others, as well.
Second, these are my general rules:
- if there are multiple sets, try to put an entire set on a disc, even if it means only the encore on a third disc
- make sure you don't split discs across an intro>song. You don't want the intro on disc 1 and the song on disc 2. Segues and long transition jams should be treated the same way.
- if you've covered the above, then try to keep the discs with an equivalent amount of empty space. Try not have a full disc 1 and disc 2, then a virtually empty disc 3. Spread the show out over all three discs.
- I personally never use 80 minute discs since they are not Red Book standard. If you choose to utilize them, make a note in the info file, and try to include an alternative tracking for 74 minute discs.
What if the levels are too low after I've put them
onto the hard drive?
First, understand that since you are using digital audio, the levels
on the DAT will be the same on the computer. There are some recorders
out there with "digital level control" but unless you are running
the DAT through that the only way to control the levels before or as they
are entering the digital audio card is by doing a conversion into the
analog domain. No good.
Okay - so the process of amplifying the sound is named different things in different programs.
- Amplification
I'm pretty sure Cool Edit Pro calls it "Amplify." This means you are increasing the signal by a certain amount across the board, without any reference point. - Normalization
Sound Forge calls its amplification process Normalization. This is amplifcation in reference to a specific point or statistic. For instance, you can "peak normalize" the file so that the absolute loudest point reaches 0db or whatever level you wish. "RMS normalization" refers to normalizing the average power of the entire file, which roughly equates with overall loudness.
When you amplify, you must be careful not to clip the signal - increase its intensity to the point where it distorts. One solution to this problem is to employ compression - basically, as quieter moments are made louder, louder moments are made quieter. The overal dynamic range of the file, or the distance between its highs and lows in volume, is compressed. This is a much more significant alteration of the show than peak normalization, however.
The major concern with any kind of normalization or amplification is that you are, after all, altering the original show from the recording. Generally, I avoid doing this if the show was recorded properly with hot levels. However, if there is a clear and justifiable need to increase levels, then I do so, but with care to not alter the show's characteristics too much.
What else do I need to know?
Honestly, not much. I say this partly because there are so many different
products out there that I could not possibly cover everything, and partly
because it's important to try it out and find things out on your own.
I have a very good knowledge of Sound Forge because I use it almost every
day. Basically, get the parts together, get them all working, then get
them all working together, then just get going...
General
Digital Audio Card Manufacturers and Resources