Check out the [[url=http://[/URL]="http://www.studiopr…"]Studio Projects SP828[/]="http://www.studiopr…"]Studio Projects SP828[/] also.

what about using optical/spdif? are there any 8 ch pres like that?

Thanks to David French... looks like a nice piece of kit. And if the pan pots don't alter the direct outs, then there are my sub-group / monitor mix outs ... Sold!

To miks: all of the 1R units I mentioned including the new Mackie offer ADAT lightpipe outs, AES digital, and I think the PreSonus is switchable over to the pro-sumer S/PDIF.

A/D conversion in the 002R is known to be less than stellar (crap :)), but I'm not sure if the others are much better in that Dept. It's a matter of convenience, too, since the 002 only includes four (4) mic pres and I'd need more than that for live drum and bass.

i have an 002 and a ISA428...just 4 pres but 8 A/D converters (with optional card) so, you can add another 4 pres from other devices...around 2300$...but i can tell you it´s a BIG step... 8-)

PD: like the converters too...above 002s

Was also thinking that the Mackie 1620 console is a good value since I could use its post-EQ and channel direct outs to my 002R

I'm not sure exactly what you're saying here, but the direct outs on the new mixers are pre-EQ.

A/D conversion in the 002R is known to be less than stellar (crap )

I'd agree that its conversion is less than stellar, but I wouldn't call it "crap". It's decent.

-Duardo

Thanks to Duardo. I should have checked the specs on the Onyx I was used to the signal path on the 1604.

Think after all of these suggestions I am going with the SP828. I'm maybe one or two channels short but I can make that up by using the S/PDIF out of an existing mic pre.

Now all I have to do is figure out if the 002 can handle this much audio reliably. I'm already looking at another FW bus. Cheers, mm

The 002 can easily provide 18 tracks of audio to your computer. The question should be can your CPU keep up.

What do you mean by another FW bus?

Steve

I've been using the 001 and an Ai3 for track laying on location for a couple of years now.

It's all good on a simple XP1800 based machine.
Editing can also be peformed on the XP machine with reliability,
Then tracks are transferred to a OSX TDM HD system for mixing.

I tend not to use the 17 and 18th input as it means another device needs to clock.
The Ai3 is quite well behaved and reliable. I'm sure there are now a couple of other units that can perform as well and I don't see why the newer 002 and a newer XP based machine wouldn't perform just as well.

Yes,
the move from PCI to Firewire could have some issues.

Kev, do you have some more information on the Ai3? Sounds like a reliable unit.

I should have mentioned in my orig post, I'm on a PowerBook G4 1.33GHz with 1GB mem.

What I'm going to buy soon is a second FW interface to create one bus for incoming audio and one for the external hard drive(s).

--> http://www.epowermac.com.au/pages/Macpower_PCMCIA_Firewire_400_161_1.htm

michael_midfi wrote: A/D conversion in the 002R is known to be less than stellar (crap :))

Less than stellar? deffinately!
Crap? no way! They are a big step above the 001..... now THAT is crap ;)

Market update: Gear I want isn't even on the market yet! The Studio Projects SP828 is not in dealer's hands and manufacturer's delivery dates are "up in the air".

Also, Mackie's Onyx 800R is nowhere to be found.

Choice seems to be down to the OctoPre versus the Digimax LT (since I don't need 96K).

Someone at scam ca$h suggested I use a Venice 160 PA desk for recording which I thought was a fine idea solving the issue of enough channels with EQ and aux sends for monitoring *until* I saw it's nearly $3,000 price tag!

michael_midfi wrote: Kev, do you have some more information on the Ai3? Sounds like a reliable unit.

:shock:
like what sort of info ?

It has ligthpipe in and out and this allows it to chase the incoming clock signal and so the computer is the master. I prefer this as it keeps things a little more simple. Simple often leads to reliability.

The audio I/O is on TRS and the front panel is switchable -10 and +4

Not sure I completely believe their headroom specs and so I often end up at -10 so my mic-pre's can easily clip PT.
Long explaination but I have made my mic-pres so I can do some internal trimming to get the desired result.

As with many of the Alesis units there is a certain hyped sound but I find it useful and does contrast with the 001 analog inputs.

I use this for locational recording and sometimes in the studio with my old 888-24. I haven't used it with a 96I/O yet.

As with all things there is a compromise but I have never been sorry I bought the Ai3.

Markd102, you are correct I think I should take back my "crap" comment above.

My point really was, do you think that the mic pres inside the 002R could stand out compared with, say, a Mackie 1402-VLZ Pro?

Those are pretty transparent high headroom and while they'd be no one's premier choice on here, I think they would do the job for recording drums with < 6 mics.

"Kev, do you have some more information on the Ai3? Sounds like a reliable unit."

and i would like to add.... one of the few 16bit converters claiming to be 24bit!!!!!

pretty respectable!!! :? :? :? :shock: :shock: :-? :-?

This is the second thread where you have tied the 16bit / 24 bit and signal to noise ratio in such a way.

Seems as you only have part of the story. There are many factors when making any recommendation.
I have said repeatedly that the Ai3 has given reliability and been well behaved on a few ligthpipe devices.
Most of the new ones will probably sound better but you will need to wait for 2 years before you can give an opinion on reliability.

Second hand this unit will be cheap and leave money to buy better Mic's and Mic-pres. These will be in use long after the Ai3 and the next generation of AD's is long gone.

we will then be arguing over the merits of 192k ... fascinating stuff over at PSW in Dan Lavry's forum
http://recforums.prosoundweb.com/index.php/t/2133/0
and many others
like word clock
http://recforums.prosoundweb.com/index.php/t/2133/0
this has a direct bearing on how the unit performs in a chase mode

all too complicated to for a simple Lightpipe interface question ... especially when it is for a unit at the bottom of the food chain.

As for the signal to noise and operating levels ... and bits ...
we could go around in circles
so
flip the lid and tell my what chip is being used ... then flip the lid on a few of the competitors ...
:wink:

The performance of the AI3 (or any other converter) is reliant on whether or not it is in slave mode or clocking on it's own. The internal clock is not very good but when the AI3 is clocking off computer at 24 bits the performance improves. Much of what makes some converters better than others is the clocking. On the other hand, if I had an Apogee converter that could send clock, I would slave the computer to that. Use whichever clock is the most stable.

Almost all converters that utilize the ADAT lightpipes, uses the same proprietary ADAT chip from Alesis ... I believe that was what Kev was saying at the end of his last post ...

When you look at what Kev wrote as to cost and obsolescence, the argument for the AI3 gets better. My feeling is that mics and pres really make much more of a difference than converters and I concur with Kevs observations. I've made some pretty successful albums with 16 bit ADAT and at the time those machines were at the edge of the technology, I was asking myself, "How much better does it need to be?"

My take on converters is the same as mic pres, the mid priced products are still a compromise in one way or the other. The best are expensive. The rest are pretty much samo sameo excluding clocking. So if you clock a cheap converter with a better source, the performance improves to that of something that might be two or three times the price. But why pay for better clocking in a converter that you are going to sync to a computer?

Converters does always utilies the least jitter when used with it's internal clock. By using an external clock in a converter the best performance you will get will always contain more jitter than the internal clock... unles you have a funky designed unit.

Best Regards

Kurt Foster wrote:
Almost all converters that utilize the ADAT lightpipes, uses the same proprietary ADAT chip from Alesis ...

And this means...?
Do you believe that they sound the same then?

Oh ! ... this is just too simplistic to say sound is better or sound is the same ...
the application of a chip ... any chip digital or analog can have great bearing on both sound and associated performances.

Lets leave the Ai3 just for a minute and look at another very popular budget ADDA.
yes it is a B unit ... :shock:
... even so this is fun to look at
It has been said by some that, it is a direct copy of an RME.
It might only be that both companies have looked at the application notes for the relevant Wavefront AL1101 and 1201 chips and landed with similar results.
:(
it could be more devious, it is not for me to say.

get to the point Kev.

One of these units have a TL074 chip at both the input and the output. This TL074 could have great bearing on both sound and ultimate signal to noise ratio. It might also be interesting to know what operating level the AD and DA chips run at and how both companies chose to apply this to the outside world.

I think this is where Alesis made some choices and ended up at lower that what many of you would call pro. ... -14 or -13dB or there abouts.

side step
another relatively unknown converter from :
Digital Audio Denmark ADDA 2402 24-bit Audio Converter
they don't tell the whole story in their brochure but they do go a bit deeper that some
http://www.digitalaudio.dk/reviews/adda_2402/audiomedia.PDF

A/D conversion is 24-bit, and the resolution can be decreased to 20, 18 or 16 bits by adding psychoacoustic dither. This is done by implementing the algorithms and coefficients derived in Robert Wannamaker's Psychoacoustically Optimum Dither AES paper, which a friend of mine who designs converters read when it came out and described as a mathematical nightmare. This advanced dither feature is ‘free' with the CS5397 — fortunately there are talented individuals who are able to transform advanced psychoacoustic research into silicon, and manufacturers who are willing to implement it, as it adds a great deal to the 2402's capabilities.

Robert Wannamaker's Psychoacoustically Optimum Dither AES paper ... :shock:
No I haven't read it and had no idea it existed.

bottom line is
... now I have no idea whether to describe this unit as 24 bit, 20bit ,18 bit or 16 bit.

but the Crystal Semiconductor CS5397 does support 96k 24bit.

what other features are lurking inside these chips and do manufacturers make use if them ?

All of this and the original question was how to get more inputs for drums to a 002.
:(
there are many possibilities
a second hand Ai3 might be cool and cheap

enjoy

"... now I have no idea whether to describe this unit as 24 bit, 20bit ,18 bit or 16 bit."

it is a 24bit device where the last 8 bits are noise....

i know that it is probably good avlue for money... but when the specs are so wack all my alarm goes off

i will pop the hood of it soon and see what hides inside... i'm too curios....

the tl 074 is an awful just as the 5534 and of course the original 741 but nobody uses that...

5532/4 is inside so many respectable units as ssl consoles and focusrite preamp..... and my 22 year old pro one synth.... and on the delta audiophile 2496!!!!

tl 074 makes up most of an old trident mixer....

they are really pretty responsible for the sound of alot of things.....

i wish people would use less and better opamps

iznogood wrote: "... it is a 24bit device where the last 8 bits are noise....

mmm ... :? ... 8 bits of noise

but the Digital Audio Denmark ADDA2402 have a quoted
Dynamic range (A) > 117 dB for both A to D and D to A
and
Dynamic range (A) > 112 dB for DD conversion

but getting back to the earlier types of units like the BADDA8000 and perhaps the Ai3
....
:roll:
....
say you were to start within the the digital world and create an encoded signal
then send it through am ADAT Lightpipe system

from the Wavefront AL401AD Encoder application notes :

The AL1401A OptoGen™ interface has been designed for ease of use and flexibility in systems designed to interface to the ADAT. protocol. It supports both left and right justified 16, 18, 20, 22 and 24-bit data formats for ease of integration into existing devices as well as new devices. These formats allow it to operate in parallel with many standard DACs.
The designer uses the WDCLKNEG, Format0, Format1, Format2 and Format3 pins to select the desired format.
If WDCLKNEG is high, the falling edge of WDCLK signals the start of a new sample period. If low, the rising edge of WDCLK signals the start of a new sample period. In both cases, the first sample data sent is the odd numbered (left) channel. The second is the even numbered (right) channel.
The format pins are summarized in the Formats table. The AL1401A provides support for both the ADAT Type I format (16-bit) and the ADAT Type II format (20-bit).
USER0 is used to transmit the ADAT format 32-bit timecode. USER1 is used to transmit MIDI data. USER2 and USER3 are reserved and should be tied low. User bits are sampled at the
WDCLK edge that indicates the end of right channel data.

and from the AL402DS Decoder application notes :

The AL1402 OptoRec interface has been designed for ease of use and flexibility in systems
designed to interface to the ADAT. protocol. It supports both left and right justified data formats
for ease of integration into existing devices as well as new devices. These formats allow it to
operate in parallel with many standard ADC’s.
The designer uses the FMT0, FMT1, MODE0 and MODE1 pins to select the desired format and mode.
The format pins are summarized in Table 3, Formats. The AL1402 provides support for both the ADAT. Type I format (16-bit) and the ADAT. Type II format (20-bit).
Data output is 24 bit.
Data input lengths up to 24 bits is supported.
USER0 is used to receive the ADAT. format 32-bit timcode; USER1 is used to receive MIDI data (if the source device supports these features). USER2 and USER3 are reserved and should not be used.

So an important factor could be :
ADAT Type 1 format is 16 bit
ADAT Type II format is 20 bit

Even though an A to D converter like an AL1101 is 24-bit and 107dB dynamic range (A-wt)
and the associated D to A might also be 24 bit and being fed by an AL1402 OptoRec interface with a 24 bit data output the resultant signal to noise may be dominated by the ADAT format on the optical link.

opamps ... :shock:
well, that's a whole 'notha' thread.

oh sorry kev... :?

i thought you were still talking 'bout the ai-3

it another thing about the d.a.d.

it's a great sounding converter.... used it for years

almost 20 bits...

Also, Mackie's Onyx 800R is nowhere to be found.

They're shipping now.

As for the signal to noise and operating levels ... and bits ...
we could go around in circles
so
flip the lid and tell my what chip is being used ... then flip the lid on a few of the competitors ..

The chip used is a very small part of the overall sound of a converter (which is why those ads saying that "Box x uses the same converters as box y[/y] at a fraction of the price" are so deceptive...sure, it may be true technically, but it certainly doesn't mean that they'll sound the same)...the analog circuitry on the front or back end of the chips, the power supply, the clock, and so on all play a huge part in the audio quality of a converter.

This is the second thread where you have tied the 16bit / 24 bit and signal to noise ratio in such a way.

They are tied together. That's what 24 bits give us...more dynamic range. Sure, none give us a true 24 bits' worth of dynamic range, but when a 24-bit converter only gives us 16 bits' worth of usable information, that just shows how important all of the other parts of the converter are.

My feeling is that mics and pres really make much more of a difference than converters and I concur with Kevs observations. I've made some pretty successful albums with 16 bit ADAT and at the time those machines were at the edge of the technology, I was asking myself, "How much better does it need to be?"

You seem to be very picky about audio quality, so it still baffles me why you don't see the value in better converters. Sure, they don't hold their value like higher-quality analog gear does (although higher-end converters, like Apogee, do tend to hold their value much better than cheaper converters), but if you're spending good money on the best preamps and processors, I can't understand why you'd cut corners in when it comes to the device that actually converts those sounds to digital.

... now I have no idea whether to describe this unit as 24 bit, 20bit ,18 bit or 16 bit.

It's a 24-bit converter that can dither its output signal down to 20, 18 or 16 bits.

but the Digital Audio Denmark ADDA2402 have a quoted
Dynamic range (A) > 117 dB for both A to D and D to A

It won't give you >117 dB when you have the output set to 16 or 18 bits, or 20 for that matter.

-Duardo

Duardo wrote: The chip used is a very small part of the overall sound of a converter (which is why those ads saying that "Box x uses the same converters as box y at a fraction of the price" are so deceptive...sure, it may be true technically, but it certainly doesn't mean that they'll sound the same)...the analog circuitry on the front or back end of the chips, the power supply, the clock, and so on all play a huge part in the audio quality of a converter.

:roll:

I think people are not reading my posts
... in this thread or the other thread currently talking about the Ai3 and the ada8000

I haven't given opinions just presented some facts of what is contained inside.

Apart from PCB layout which IS important
... the choice of chips does have great bearing on the ultimate sound and performance.
(not sure that can be counted as my opinion :roll: )
I have already stated that one of the budget units uses the TL074 as an input and output op-amp and that it is very likely that a class unit may have an expensive chip here.
Whether an increase of 60c to $4 across about 8 op-amps can turn into $1000 at the retail counter is for you to decide.

Chips like the Wavefront range are well sorted and the scope to implement variations is low.
The ada8000 has some transistor and two stages of TL074 before the A to D and then onto the Opto Encoder and send.
From the Opto Decoder the signal goes to the Wavefront D to A and is then buffered with a single stage of TL074 with the second stage of TL074 used to create an inverted signal for a balanced output.

As far as clock is concerned,
the ability to chase external clock could be more important than the it's internal clock.

Yes as always the power supply is important that it is spec'd correctly both Digial and Analog supply rails.
This is one area that I believe the cheap units can be improved.
(finally an opinion from Kev)
Supply rail smoothing at each analog op-amp it something you do see on class gear of the past.
BUT
The ada8000 does look like it has some electrolytic caps at the TL074's but I would have to have a closer look to be sure. Perhaps some low ESR 105deg caps can make a marginal difference here.

Only four stages of TL074 in and out ... but then a TL074 is a quad opamp. (yes input and output are probably shared across multiple chips blah blah ...)
Simply a better quality opamp may achieve the maximum change to the sound.
I did say that opamps was a whole 'notha' thread

No one seemed to pick up on the
ADAT Type 1 format is 16 bit
ADAT Type II format is 20 bit

please do check out the Wavefront products page and read the PDFs
http://www.wavefrontsemi.com/products.html
A total of 8 in the range and only 4 that concern us here.

enjoy

Hi,
just passing very first impressions on the Behringer unit. (I´m not afraid of writing that name here). Got one at a passable low price, and thought, what the h..., could probably be useful for something. Sort of like getting one more of these lowprice chinese mics just for the fun of it. One or another of them seems to get used now and then, so might as well have a few.

Anyway, opened the box and looked inside. Very impressed by what I see as far as build. Neat layout, surface mount chips, very little hand assembly. Definetely not a "boutique" job, looks truly "industrial mass market" to me. Sort of like looking a modern mass market build car. (Not like a hand built sports car, but they are out of my budget, and doesn´t get me a second faster to the job in the morning anyway).

Next thing is to hear what it sounds like and how well it stands up to usage over time. Will do a bit of that in the future, but nothing really bad has cropped up so far.

Anyway, to the best of my knowledge the single most important factor of sound is the artist. Next comes mic placement. Third is mic. Fourth is probably mixing (might be third place). Fifth is probably mic preamp (could be more important in some cases). On sixth place comes converter, remember this is today, not ten years ago when converters where black art. I don´t expect much, but in the right hands this unit will not destroy the sound -- it might not improve it but I expect it to do a decent job.

Gunnar.

I'm slowly collecting information before I purchase a multichannel preamp for my use. I have a Mackie 8-24 console right now. If I buy the Mackie Onyx 800R will it be the same (or only marginally different) from the pres that are already in my console? Primarily planning to use external multichannel external pres for tracks from the drum booth. Might also use it for other tracks after experimenting. Will eventually get something nicer for recording vocals.

Kurt was helpful earlier and recommended that I look into the JLM TMP-8 and that's got my main attention for now. He wasn't impressed with the Presonus M80. Now I'm curious about the new Mackie multi-pre unit that I keep reading about. Any thoughts?

Spectrumband wrote: ... recommended that I look into the JLM TMP-8 and that's got my main attention for now.

Joe ... JLM .. is looking into Lightpipe connectivity for this a other JLM mic-pres