My Prism dream came true...but m`y go'i instead of rice for awhile

I was never a fan of Apogee or Digidesign converter. There are...boring! You just almost could never guess what's going to happen to your beloved analog sources once "inject" into them. The one I trust and many of the master engineers love to rely on are the Prism stuffs. Very expensive...but now check this out, the Orpheus is here. Firewire converter and multi out, can you beat that? Price is not announce yet but words on the street is that it will be accessible.
Well read for yourself drool.
rpheus is a FireWire interface for personal recording and sound production, for professional musicians, songwriters, engineers and producers. Orpheus is ideal for music and sound recording, production & monitoring, stem-based mastering and analogue summing.

Orpheus provides Prism Sound's renowned performance and sound quality in a dedicated FireWire unit compatible with Windows XP & Vista and MAC OS X 10.4x (Intel & PPC).

Orpheus has line, microphone and instrument inputs, good foldback and stereo or surround monitoring capabilities, ADAT and SPDIF digital I/O plus support for external MIDI devices. Microphone inputs include MS matrix processing and high-performance digital sampling-rate conversion (SRC) is available for digital inputs or outputs.

Orpheus signal path
Eight analogue input channels and up to 10 digital input channels are available (SPDIF on RCA/coax plus ADAT optical) as DAW inputs through the host's audio driver. Similarly, eight analogue output channels, up to 10 digital output channels and stereo headphone outputs can play 22 different channels. For low-latency foldback or monitoring to headphones or main outputs, each output pair (1-2, 3-4 etc) can be driven with an individual local mix of any selection of inputs through the controller applet. All inputs are electronically balanced with automatic unbalanced operation. Outputs are electronically balanced with 'bootstrapping', i.e. level is maintained if one leg is grounded.


No-compromise, full Prism Sound audio quality
Dedicated FireWire interface
ASIO and WDM drivers provided for Windows XP and VISTA
Directly compatible with CORE AUDIO on Mac OS X 10.4+ (Intel & PPC)
Eight "Prism Sound" AD and DA channels, plus SPDIF, ADAT & headphones
Four high-end integrated mic preamps (typ.-130dBu EIN), switchable phantom
MS Matrix processing on mic inputs
Two instrument inputs
Prism Sound "Overkillers" on every channel to control transient overloads
Fully-floating (isolated) balanced architecture for optimum noise rejection
Mono or stereo input configurations
Outputs arranged as stereo pairs, each with individual mixer
Low-latency "console-quality" 8-bus digital mixer for foldback monitoring
Fader, pan, cut, solo on every mixer channel
Dual headphone outputs each with its own front-panel volume control
Front-panel master volume control, assignable to selected channels
Configurable for stereo, 5.1 or 7.1 or surround monitoring
Built-in sample rate conversion (SRC) on DIO channels
Prism Sound 4-curve SNS noise shaping on digital outputs
State-of-the-art clock generation with proprietary hybrid 2-stage DPLL
MIDI in/out ports


Luke Ehret

---

Why Tool's music sound so so good and yours not...listen to what they have to say.

check the interview and listen to how Tool approach their music, arrangement, and rehearsal...and it has nothing to do with being a star.
Tool let their music does the talking, not the newspaper, and talk it does...very well!

I hate Limiter...but not this one!

Chandler Limited Zener Limiter...damm you!

The main reason for Zener Limiter reissue was to add features and flexibility to the powerful and vintage sounding TG limiter circuits. To that end, Wade has added many new controls including switch able input impedance for hard of soft driving of the unit, 11 position attack, 21 position release, side chain filtering, and Comp1, Comp2, and Limit settings.

chandler zener limiter features

The EMI TG12413 Zener Limiter is the ultimate TG Limiter issued in celebration of the 75th birthday of Abbey Road Studios. The Zener Limiter was conceived by Chandler Limited designer Wade Goeke and is based on the vintage EMI circuits used to record The Beatles and Pink Floyd.

This newest EMI Limiter continues the tradition of EMI Limiters started in 1956 with the RS114 tube limiter, and continued with the RS168 Zener Limiter in 1966, part of the TG12345 console channel in 1968 and TG12413 in 1974. The newest version borrows from the RS168 Zener Limiter and TG12345 console strip to make a new fully featured and flexible unit for modern use. It is interesting to note that the TG Limiter was originally designed to replace both the Fairchild 660 (Limit Mode) and the Altec 436/RS124 (Comp 1 Mode).

• 21 position input switch with audio taper
• 21 position output switch in 1dB steps, ±10dB of level control
• Input Impedance: 300/1200 switchable
• Side chain filter
• Hard wire bypass
• Stereo Link
• THD Function bypasses the limiter and drives the circuit to 2% THD and higher
• Attack Controls
  Limit: -5, 8, 12, 18, 25, 37, 50, 62, 75, 87, 100ms.
  Comp. 1: -5, 8, 12, 18, 25, 37, 50, 62, 75, 87, 100ms.
  Comp. 2: 28, 47, 68, 100, 125, 210, 285, 350, 425, 495, 570
• Release Controls
  Limit: 10, 25, 50, 65, 100, 125, 150, 200, 250, 300, 250, 400, 450, 500, 625, 750ms and 1, 1.25, 1.5 and 2 seconds
  Comp. 1: 10, 25, 50, 65, 100, 125, 150, 200, 250, 300, 250, 400, 450, 500, 625, 750ms and 1, 1.25, 1.5 and 2 seconds
  Comp. 2: 50, 125, 250, 375, 425, 545, 700, 850ms and 1, 1.2, 1.6, 2, 2.25, 2.5, 2.8, 3.5, 4.25, 5, 7.1, 8.5 and 10 seconds

COOKING CRYSTAL...no no not that one, digital clock crystal!

This is so freak'n crazy. I found this video from DAS on youtube.com and this dude is saying to keep the clock stable you need to...cook it!
Yeah, no kidding. This company claimed that their digital clock are much more stable than others by building an oven inside the unit, then "cook" the crystal to keep it at certain temperature so it is better as a clocking unit for your digital system.

You decide for yourself. By the way, if any of you bought a Apogee Big Ben to clock your Protool system, you are an idiot...email me, I tell you why.



Tran Duy

Small Room-Low Freq Control

Cool article on Modal mode in your small project studio.

It is no surprise that industry figures show that there might be over 100,000 "home / project" studios in existence today. This number will grow. In the last 10 years, an audio revolution has occurred, which allows "home studios" to rival the sounds of pro level studios. "Desk-top" audio, as I call it, is possible due to several factors:

1. reduced equipment prices;
2. integration with the ever more powerful home computer;
3. the digital revolution in general; and
4. an associated social change in the methodology of producing and distributing music.

While these technical changes have affected the way music is recorded, they have not affected the way music is played back (listened to). Songwriters, musicians and producers still listen to music in indoor environments and depend on some order of true acoustic response from a monitoring system in order to discern content and production values.

There are many acoustic and architectural issues that we will discuss in this column during the next year. Where do we begin? This is the challenge that TAXI has presented to me. The most common question I am asked is usually something like, "How big (or what size) should my room be?" During 34 years of designing studios, I have been asked this question thousands of times. It's a little bit like asking how big should a car be. (Of course the next question is how much will all this cost, which is a little bit like asking "How much does a car cost?" We'll get to that later!). This size question is the hardest. But a good answer will make the rest of the studio design and building process (or remodeling process) easier.

Most "home / project" studios are small. They often have less equipment than many larger commercial studios, plus the client accommodation factor is typically less or non-existent (you don't have to have a fancy lobby!). Small room design is not a black art, although certain acoustic issues are in fact more complicated in smaller rooms than larger ones.

Most studio acoustic issues can be reduced to two large areas of concern:

1. Sound isolation (prevention of sound transfer from one space to another)
2. Internal room acoustics (what happens to sound in a playback or recording environment)

It is the second issue we are concerned with today. (Regarding issue #1: A great number of "home / project" studios do not have serious isolation issues or elect to not deal with them for financial reasons).

Since music has a wide frequency range (as opposed to speech, for instance) and since the level of accuracy of affordable mid-size speakers has become amazing in the past few years, even our small room acoustically gets complicated very quickly. Rooms behave differently at different frequencies.

Simply presented, at mid and high frequencies (those for the most part above 300hz) individual reflection patterns will cause sound to be perceived cleanly or not at the listening position. At these frequencies, sound can be "viewed" a bit like rays of light. At lower frequencies, however, reflection control becomes less and less relevant (since wave lengths become larger) and what counts more is the ratio of raw room dimensions and the position of speakers and listener. Sound at these frequencies behaves more like waves. The overall dimensions of the room will effect the natural distribution of eigentones (fancy term for "standing waves").

A common misunderstanding is that standing waves are bad. This is ridiculous. That's like saying that wheels on a car are bad. Four different size wheels on a car are bad! Standing waves always exist in a closed environment. What we strive for is as even a spacing of these frequencies as possible. Think of these standing wave frequencies as the ability of the room to "ring out" or reinforce tones naturally. And so one can imagine that if the proportions of a room are chosen correctly, then there will be a more natural spacing of the tones and the room will tend to reinforce lower frequency tones more evenly. This is a good thing. The opposite, of course, would be harmful and tend to cause uneven response at the listening position. Not a good thing for audio playback.

So, the first step in room acoustic design (after making sure that all equipment, furniture, etc. fits) is to try to choose a room shape that has as good a chance of even low frequency eigentones spacing (organization) as possible. That is what we will discuss in the remainder of this article. (We'll look at high frequency reflection control in the next article).

By way of example, I recount this story. A former student called me up a couple of years ago and ask,

" . . . I have a 20 ft. by 20 -ft. basement room that I want to use as a control room for my home studio; what should I do to make it sound good?"

That's a big question. Half of me wanted to hang up, but half of me accepted the challenge of trying to give him a one-minute answer. After a minute of thinking, I answered,

"Build a closet."

He probably thought I was joking, but I still believe this answer was a good one. The square room (20 ft. by 20 ft.) is almost the worst shape you could have (only thing worse would be a 20 ft. cube). The width and length being the same dimensions would cause the lower frequency eigentones (standing wave frequencies) to be identical, thus causing harsh frequency anomalies -- pile up of energy -- as well as voids at other frequencies. These frequencies are not that hard to calculate, doing some very easy math. (My promise to TAXI was no math in this column, which is hard, since the language of acoustics is physics and one of the languages of physics is math!) By building a closet, the future TAXI driver would have possibly created a room that was 20 ft. wide and (more or less) 15.5 ft. deep -- much better room ratio. He also gets a closet for storage and possibly a good location for noisy equipment and other devices. Notice that I suggested in his new room that 20 ft. was the WIDTH of the room. By having the side walls further away from the listening position we help mid / high frequency reflection control (see article illustration 1). Again, we will discuss high frequency reflection control in next month's article. Choosing room ratios can also be easily analyzed by using a very well known industry "pictogram" of accepted room ratios (see article illustration 2).

Illustration 1
Before and after low frequency modal distributions for a 20' x 20' room and a revised design.

Note the improved room acoustics due to better modal distribution.

Ill. 1a - shows 20' x 20' room in plan

Ill.1b - shows newly created 16.5' x 20' room in plan

Illustration 2
Acceptable room ratio "pictogram" with basic user steps:
a. divide all room dimensions by the height (this will then set the height as 1)
b. plot width and length on horizontal and vertical scales
c. determine acceptability by noting whether plot is in or out of "the zone"

Ill. 2a - shows 12' x 24' x 36' room - poor low frequency room mode acceptability


Ill.2b - shows 21.5' x 15' x 9.5' room - good low frequency room mode acceptability

Reminder: the easiest way to begin to have good low frequency response in your room is start off with good room ratios. There are other acoustic "treatments" that can be added to the room if needed, particularly if circumstances do not allow good room ratios. We will look at them later.

Have fun!

PS. aaah...some of you will wonder what the hell is room modal and how to calculate your own space to what is the best room mode. That is what coming next....so come back!

SoundPure LLC is on the team!

After a long working relation, Bflatmultimedia inc. would like to welcome Soundpure LLC onto the team in VN as part of Bflatmultimedia pro audio and equipment consultant services which had been the strongest part of our company.
Duy is now officially working for Soundpure LLC and coordinate with Doug Wesling and Todd in the US for sound contract in VN.

here are some of our latest and hot items...


Toft new analog mixer which can be config in 16, 24, and 32 channels. Great EQ and line input, and fatten your cold DAW.


DWFEARN channel strip...must we say more!



For a more of our consultancy on Pro Audio gears, check the listing http://www.soundpure.com/showManufacturers.do

Tran Duy

Thinking building your own studio lately?

While in VN the most interest bunch of folks I met were the Vietnamese architects. Most of these guys came out of school all had knowledges about sound and the reaction of building materials associated with their design works. I often get asked by them of the applications and how it may applied in the real world situations.
It is an interesting topics to get into for me, I also do want to learn a thing or two regarding architect designs and how it may help me in studio design situations.

Ok, I will start to submit things I found to be associate with sound and architect designs for studio or home studio in the future.

so far here is the start....

With most architect text I found in VN, all mentioned STC as way to determine material transmission value of building material...it is cool but wait, it's 2007 now, so let's clear this issues up.

The STC - or Sound Transmission Class - is a recognized standard and is, by far, the common sound isolation standard in use in North America today. Virtually every commentary that one reads focuses on STC, yet STC is not without significant limitations, and for a great many applications it is not a good measure of sound isolation at all. Before we talk about the limitations of STC as a rating, let’s take a look at what STC is.

What is STC?

Is STC a measure of how many decibels of sound a wall can stop? - No, it is not.

Is STC a ranking of how good a wall is? - No, for most applications it is not.

So what the heck is it? - It is a very old (1961) method for ranking walls over the frequency range of 125 - 4000 Hz, assuming that the noise the wall is trying to stop is generally even across the frequency spectrum.

The problems with the STC system

The three basic limitations of STC are apparent from the description of the system above.

1. It only considers frequencies down to 125 Hz.
The first, and most severe, problem with the STC system is that it only considers frequencies down to 125 Hz. What noise exists below 125 Hz?

• Most of the sonic energy generated by the average home theater
• A large percentage of the sonic energy generated by traffic, your neighbors in VN, and music
• Much of machinery noise

If you have sound isolation problems, there is a very good chance that it is low frequency noise you are having trouble with, so one could say that the STC calculation completely ignores the frequencies that are most problematic. That’s not good.

2. It assumes even energy dispersion. It is accurate within its frequency range only for noise sources that have approximately even energy levels across the frequency band. Most noise sources do not meet this criterion and some (like the average home theater) are worlds away from this criterion.

3. Its calculation system is archaic. STC dates back to 1961; a time before computers made complex calculations easy, and the method of determining STC reflects this. In today’s world more complex, vastly superior calculations can easily be done. OK, that’s great, but do any real problems actually occur?

So why is the STC system used at all?

Well, there are some very good reasons why the STC system is in use.

• It’s been around for so long that essentially every law, regulation, and piece of legislation relating to sound control is based on it. Old habits are hard to break.
• As frequency falls, the ability of the different labs to get consistent results also falters. +/- 3 STC points from lab to lab is typical, but if the STC system were extended down to, say, 40 Hz, this might increase to +/- 10 STC points or more, making the results basically meaningless.
• Its easier for companies marketing commercial products to attain a huge STC increase than it is to attain a huge increase across the full frequency range. This leads to a lot of focus on STC, and less discussion of critical things like low frequency performance.

Are there better rating systems?

Yes, the best standardized rating system in North America is called OITC, and is typically used for exterior wall elements. OITC features a modern calculation system and considers frequencies down to 80 Hz.

In Europe at times full-range standards are applied. An assessment of existing standards is given in the appendices of this document.

until next time....

Tran Duy

Hard Floor for studio?

Very cool post I found on the web....

SIDEBAR: HARD FLOOR, SOFT CEILING

The following is from an exchange that took place in the rec.audio.pro newsgroup in May, 2003:

Bill Ruys asked: Why it is recommended to have bare (un-carpeted) floors in the studio? One web site I visited mentioned that a bare floor was a prerequisite for the room design with diffusors and absorbers on the ceiling, but didn't say why. I'm trying to understand the principal, rather than following blindly.

Paul Stamler: Carpet typically absorbs high frequencies and some midrange, but does nothing for bass and lower midrange. Using carpet as an acoustic treatment, in most rooms, results in a room that is dull and boomy. Most of the time you need a thicker absorber such as 4-inch or, better, 6-inch fiberglass, or acoustic tile, and you can't walk around on either of those. Hence the general recommendation that you avoid carpet on the floor and use broadband absorbers elsewhere.

Lee Liebner: the human ear is accustomed to determining spatial references from reflections off of side walls and floor, and a low ceiling would only confuse the brain with more early reflections it doesn't need. Everywhere you go, the floor is always the same distance away from you, so it's a reference that your brain can always relate to.

John Noll: Reasons for having wood floors: they look good, equipment can be rolled easily, spills can be cleaned up easily, provide a bright sound if needed, sound can be deadened with area rugs.

Ethan Winer: In a studio room, versus a control room, a reflective floor is a great way to get a nice sense of ambience when recording acoustic instruments. Notice I said reflective, not wood, since linoleum and other materials are less expensive than wood yet sound the same. When you record an acoustic guitar or clarinet or whatever, slight reflections off the floor give the illusion of "being right there in the room" on the recording. It's more difficult to use a ceiling for ambience - especially in a typical home studio with low ceilings - because the mikes are too close to the ceiling when miking from above. And that proximity creates comb filtering which can yield a hollow sound. So with a hard floor surface you can get ambience, and with full absorption on the ceiling you can put the mike above the instrument, very close to the ceiling, without getting comb filtering.

Dave Wallingford: I've always preferred wood floors for a few reasons: 1) It's easier to move stuff around, 2) You can always get area rugs if you need them, And the main reason: 3) Pianos sound like crap on carpet.

Cho may guitar gods tai Vietnam phan 4...dam Boss pedals

BOSS PEDAL!!!!


Roland cube Amp...yeah Phuc!



Enough Boss shit, here comes Marshal



JCM2000DSL



And finally Petucci's rig and his boss demo.

Souxsie: Dreamshow DVD live concert!

This went back to my day as a punk rocker (dam I am old), I had followed Souxsie and her band the Banshees every step of the way. I saw almost every US concerts and like NIN + Tool, this is a must see show.


Two years ago in SF, I managed to attend her show at the Warfield which she took the entire Japanese Taiko troop on stage with her, I had been waiting for the dam DVD for this show ever since. It is finally here and here is the glimpse of that show. Please buy the dam DVD, you will not regret it...if you know music.

New toy from Cranesong, API, and Manley

EGRET

Scott at Cranesong just send me this baby to test out. Basically it would be release soon after Cranesong works out all the bugs. So far I found nothing and the dam thing sound...Warm and the wow factor went to 10!

Egert is a highly flexible workstation back end. It contains 8 channels of high quality D/A
converters and a stereo line mixer with color options to help bring analog summed digital mixes
to life.
The stereo mixer has a level control, a cue send, a color control, and a pan control on each
channel. Each channel also contains an analog / digital source button, and solo - mute buttons.
By using the balanced direct outputs and the balanced analog inputs you can insert analog
processing into individual channels.
The built in cue bus with its master level control can be used as an effects send. A balanced
stereo effects return is built into the system.
The master bus level control, a stepped attenuator, has 1 Db steps for most of its range, This
allows for accurate gain control, repeatability, and stereo gain matching to better than .05 db.
The headphone system allows a monitor mix to be created when Egert is being used in multi
channel location recording. Thus providing fail safe knowledge that all channels being recorded
contain proper audio
The D/A converters support sample rates up to 192K and have sample rate converters on each
channel for input jitter reduction. There is a front panel switch to disable the SRC for cases
where lower latency is required. The system is built so that the converters and the interface can
be upgraded as the technology changes.

API 5500 EQ and A2D Mic pre w/ digital out

The lineage of the 5500 circuit can be traced back to the original 550 equalizer designed by Saul Walker, the founder of Automated Processes Inc. The 550 was designed as a console equalizer which, due to the architecture of the recording console, uses an unbalanced input. Because the ergonomics of a console dictate that the controls take up little space, sometimes the number of included functions can be limited. The 5500 is specifically designed to address these limitations. It has a balanced input, a true straight-wire bypass, an integrated power supply with noiseless muting, and a range control that expands it versatility to mastering applications.

Like all API products, the 5500 contains no integrated circuits in its signal path. The gain comes from two hand-built 2520 operational amplifiers in each channel. The balanced input is handled by a 2510 discrete operational amplifier, which is similar to the 2520, but without the high current output stage. Besides being a key component of the API sound, the 2520 coupled with the API 2503 output transformer is capable of delivering +30dBm before clipping. With this much headroom, it is unlikely that the 5500 can be driven to distortion unless perhaps another API unit is driving the input!

A new feature in the 5500 is the range control. The range of the amplitude controls can be reduced to 1/2 or 1/4 of their stated scale, providing a means of adjusting the tonal balance with finer resolution in an even gentler manner. This should be especially useful for complex program material as contained in stems or submixes, and is ideally suited for mastering purposes.

Also new on the 5500 is a true hard-wire bypass. In this mode the output connector is wired directly to the input. Shortly after the power is first applied, or immediately after it is lost, a special circuit enables bypass mode so signal is never lost and power thumps are never heard.

The input XLR is connected to an active balanced circuit. The output XLR is driven from a transformer coupled output and can drive any load from 600 ohms or greater to full output capability. The polarity is such that there is no change from input to output, so it is suitable in studios using either pin 2 or pin 3 as the "hot" connection. In addition, there is a 1/4 inch input connector that interrupts any signal that is present on the input XLR. It is balanced and can be driven from either balanced or unbalanced sources. A positive signal on the tip will deliver a positive signal on pin 2 of the output XLR. Using the 1/4 inch input does not bypass any internal circuitry and does not change gain or operating level.

A2D Micpre with Digital out

The A²D represents a landmark achievement for API; it is the first product ever produced by the company with an integrated digital output. Recognizing the desire for a high quality solution to raise microphone signals to a workable line level, we engineered one of our most popular mic preamp designs into an enclosure with our new proprietary analog to digital conversion. The result is a complete package - a pair of superb mic preamps feeding an A/D section with a set of standard digital interface outputs, multiple sample-rate choices, and internal or external clock options. Precise control of both the analog gain as well as the level feeding the digital section means that the mic signal can be maintained super-clean if desired, or driven harder on the analog side for that whallop and impact that can only be achieved by driving the transformer into saturation for that historic API sound!

The A²D analog section consists of the circuitry contained in two API 312 mic preamps, or one half of the popular 3124 quad mic preamp rack product. A level control pot with an expanded 20-segment metering system shows precise mic gain levels. Control switches include polarity, 48v phantom power, input pad, 2:1 transformer routing, and Mic/Line select. Each analog input contains a balanced low-level XLR connector and a high-level line input on a 1/4 inch jack.

The Digital section of the A²D includes two level pots to adjust the feed to the A/D converters, each with a corresponding 20-segment calibrated meter. A multi-position switch selects the desired sample frequency. The External Sync LED will light when an external clock signal is used.

Rear panel Analog connectors include an XLR input and output for each channel, as well as a balanced 1/4" TRS insert to the A/D converter. This allows the A²D to have external analog processing devices inserted in the audio chain (EQ, compression, etc.)

Digital interface connectors provided are SPDIF single ended RCA connector and AES digital out on XLR. A coaxial BNC accepts external "Super Clock" which will illuminate the front panel EXT SYNC LED when locked.

In addition, there are a pair of 9-pin D-sub connectors which allow the linking of multiple A²D units, all operating off the first sample-rate clock in the chain.

Manley SLAM!

Eva Manley was showing this thing way back on 2006 AES show and the way it was priced, make everyone clinched. Yeah, this thing is expensive but does it live up to the hype. Manley refused to send me one to "play" with cause Eva told me, "Why? It's rock and Duy, you will keep it anyway."
So it took me awhile to find the money to get one but she was right, I kept mine...hidden from all my friends. If they knew, they would ran over and borrow. How can I refuse?

But secret out now, I owned one...yum boy this thing is so DELICIOUS!

here is the spec. and don't you hate me.

SLAM! stands for "Stereo Limiter And Micpre" and it pretty much describes what it will do to most VU meters. We had to put a switch on this limiter to drop its internal VU meters down 3 & 6dB to keep the poor little needles from bending– it'll get LOUD fast (hence the exclamation mark). And on top of being an amazing pair of (actually four) Limiters, and Class A tube mic preamps, it also has masters degrees in DI, AtoD and DtoA, VU and PPM but that was too much for an acronym.

When it comes to classic gear, especially classic limiters, we can all agree numbers like LA2s, 1176s, 2264s, and others come to mind. What if all those sounds were to be found in one stereo tube unit. Interested? We combined our favorite Electro-Optical circuit (ELOP^®) with a damn fast FET based brick-wall limiter reminiscent of some cool classics. And we added a sidechain filter that can remove low frequencies that makes it more useful for a variety of tasks, while retaining that easy, "it just works" quality that has made it a favorite for vocals everywhere. The FET-based limiter has different desirable characteristics that both optimise the signal for digital recording, due to the brick-wall capability, and let you dial in how clean or crunchy, or punchy, and how loud and proud you want it. Manley is excited to announce the SLAM!

GET IN: The SLAM! starts with tube Mic Preamps on both channels with switchable phantom power and phase reverse. We’ve got plenty of gain for you in this new tube circuit, up to 60dB with 20dB more in the limiter– plenty for soft singers using low output ribbon mics. The input attenuator is right up front, like a variable pad so your loud rock ‘n’ rollers won’t cause a problem either. Track direct with the Direct Instrument input or bring your line level inputs into the SLAM! for processing. Come on in!

Got some Digital to deal with? No problem. There is a slot in the back for our optional 24 Bit/96KHz Analog to Digital converter and 24 Bit/192KHz Digital to Analog converter card. This allows you to record directly to your computer or workstation and provides a convenient way to use a digital insert to jump into the analog realm and back. An AES digital input receives your stereo digital signal to convert to analog for processing by the SLAM! when the DAC input is selected. Or just take advantage of this killer DAC and use the DAC Output for monitoring, listening, or to go out to some other analog processors or tape machines.

The ELOP^® LIMITER: Our beloved Electro-Optical Levelling Amplifier circuit, the ELOP^®, still the favorite for vocal tracking, shows up here with its new switchable side-chain high pass filter making it much more useful for mixes and other tracks as well. One of the most appreciated aspects of our original ELOP^® is the simplicity of a two knob limiter, which is optimum for demanding pro recording pressures. You can still grab two knobs and have complete control of level and dynamics and be ready to hit that record button. Intuitive controls and a straight ahead signal path through quality components simply works.

The FET LIMITER: Here’s a new fast FET based "brick wall" limiter that spans the range of clean predictable limiting to the grunge and gravel that other FET based limiters are desired for. The FET RELEASE control allows you to dial up a variety of colors. There is even a "CLIP" setting that provides as round a shape as has ever been available. (Read more below about the CLIP settings.) We aimed at "sound" first, then experimented with circuits until we heard what we wanted discovering fresh approaches that provide a very wide range of useful dynamic control. For example, you can set up for a fast brick wall limiter with slower releases for minimum artifacts, or go with the faster releases to get some crunch and then get extreme loudness. The Attack control is unusual too providing the sonic control common with its historic brothers but very unusually retaining the zero overshoot “none-shall-pass” brick wall that defines a real limiter. And it’s punchy! As the saying goes, “Louder is better.” The SLAM! optimizes levels to analog to digital converters with ballz, warmth and loudness verging on silly or just clean and safe.

What is especially nifty is the ability to use the ELOP^® and FET limiters together at the same time dialing in whatever amounts of each effect you like. Every limiter has its own character or sound and here we have combined two powerful world-class tools that can be uniquely blended and mixed to taste and purpose. Even the metering is appropriate for each limiter so you can easily see how each is contributing.

Metering on the SLAM! is exceptional. The two full size real VU meters show you the raw input levels, the final output levels, and the ELOP^® Limiter gain reduction in the traditionally preferred format for visually displaying apparent loudness. VU’s are cool. Noticing how mastering engineers often need a switchable pad on their VU's because the "optimised" levels tend to pin the needles in the red, the SLAM! VU’s also have a 0, -3dB, -6dB switch to help with this situation. There are also very fast and super accurate multi-color LED ladder peak meters that display the input and output levels and the FET gain reduction, and some combinations. They will do Peak, Peak Hold timed, and Peak Hold until reset, and also allow you to manually set where the individual bars change colors to match your other gear. And with maximum clean I/O levels of +30 dBu, it is a lot more likely you will use these meters to see what’s clipping downstream. It probably won’t be the Manley tube line drivers inside!

The back panel should keep everybody happy. All the usual XLR’s for transformer balanced I/O and 1/4" phone jacks for direct unbalanced I/O. There are four TRS jacks for inserting external EQ into the sidechains and another connector for linking multiple units for surround. It’s all here. The optional 24 bit converters are a brand new design that insures the lowest possible conversion jitter even with questionable sources. It will lock to an external clock via the digital input and is 44.1, 48, 88.2, 96, and 192KHz capable. The DAC output is always available. There is even a twisted way (called "Expert Mode") to patch the SLAM! so that you can have the MIC PRE/ELOP feeding an EQ, (or some other thing), returning back to the SLAM! to drive the FET LIMITER, meters, and the A to D converter. Sometimes we think of just about everything...

The SLAM! also seems a natural as a mastering tool. The FET limiter works great on mixes, and the ELOP^® now has a high pass filter switch that is spot-on for mastering. We offer a new mastering version which eliminates the mic preamps, adds detented controls replacing the pots, and provides extra metering options. Manley compressors and EQs seem to be the most popular pieces of gear ever when it comes to mastering. The ultimate analog rack for the final important touches to the mix would have include the SLAM!, The Variable Mu® and The Massive Passive. Combine this with some digital processors of similar calibre and one has a respectable and very effective set of mastering tools. That Manley rack plus some Manley Microphones would make an ultimate stereo recording chain too. Gear you’ll want to keep for a lifetime.

...So the Egret price for around $4500, the API 5500 for $3000, A2D for $2000, and SLAM will slams for around $6500...but that that is all depend right???

Duy Tran

Cho may guitar gods tai Vietnam, phan 3

The Worm



The Wiggler



16-seconds delay



...ky` toi Boss pedal.

Roger Waters Plays Dark Side Down Under with Meyer Sound MILO

Legendary Pink Floyd frontman Roger Waters recently brought his "World Tour 2007" to Australia and New Zealand, performing songs from Dark Side of the Moon and other Pink Floyd classics to a backdrop of spectacular visual effects. To power Waters's captivating live show, Auckland, New Zealand-based Oceania Audio provided a state-of-the-art sound system built on the Meyer Sound MILO high-power curvilinear array loudspeaker.

Two front hangs of 10 MILO cabinets and two MILO 120 high-power expanded coverage curvilinear array loudspeakers each, augmented by six UPA-1P compact wide coverage loudspeakers and four UPA-2P compact narrow coverage loudspeakers for front fill, comprised the main system.


"The MILO system sounded superb," says Oceania Audio's technical director, Paul Jeffery, who tuned the system with the help of FOH engineer Trip Khalef and his team. "I was very impressed with the way the sound held together, even when the system was being thrashed to within an inch of its life. The system provided clean, loud, high-fidelity sound through the entire concert. It was a thrill and a pleasure to listen to the live rendition of Dark Side of the Moon with such clarity and detail."

Jeffery notes that a concert review published in Christchurch, N.Z.'s The Press referred to the audio quality of the surround sound performance as "breathtaking." "Generally speaking, if reviewers feel compelled to remark on the sound quality at all, it's usually negative. If they ignore it, that means it was acceptable," says Jeffery. "So I think we can infer from that comment that MILO was indeed performing brilliantly."

Oceania Audio's MILO arrays have had a very busy first quarter. In one month alone, the systems performed back-to-back dates with Waters and another rock legend, Eric Clapton. Oceania's hectic tour schedule also includes dates with Dionne Warwick and the Parachute outdoor music festival.

Cho may guitar gods tai Vietnam, phan 2

ELECTRO HARMONIX!



MINI Q-tron



Bass Harmonic Synth



Harmonic graphic fuzz

Dream Theater's The Dance of Eternity

How Mike lay it out

Cho may guitar gods tai Vietnam, phan 1

Truoc het so sanh PRS voi Strat.



Di tour nha may cua PRS



Zack Wylde cho Mini Marshal stack.



Ok, Zeek Overdrive test!

The 'Transient' behind SYG drum sound

Till this day, whenever I travel to Vietnam and meet up with my musician friends there. The same questions had always been what did I do to get the drum sound for the Vietnamese American band SYG (SUM YUNG GUYZ). How did I brought out the transient attack on Meo's snare and the kick out front with all the distortion from Bau's Mesa stack? The difficulty for this also with Dan's vocal, the band music carry a low distortion, almost fuzzy color. During the mix for this CD, I need to bring Meo's out, sort of outside the heavy jungle of sound. Meo's playing is fast a furious but he often got drown out by his band mates easily and this was the issue for me during mixing.


Fortunately I manage to borrow a SPL Transient design 4 unit and used it on Meo's drum tracks. The entire Cd was recorded on 2-inch 24 track analog tape deck, transfer to Protool Mix3 system for mixing via Benchmark AD. The TD 4 was the only dynamic I used for Meo's drum tracks with some minor digital editing initially.






Transient Design 4 help brought out front all the attack of the drum strike for the snare and the kick with just enough compression without killing or muddy up everything.

I went out and bought one a month later and love everything about. Ever wonder why you hardly see one selling on Ebay, no one want to let it go.

here are the spec:

The Transient Designer provides a revolutionary concept for dynamic processing rendering controls such as Threshold, Ratio and Gain superfluous. The Transient Designer's automation is highly developed, so that while the processing going on inside the box may be very complex, the user has to deal with just a few intuitive controls.

Differential Envelope Technology

SPL's Differential Envelope Technology is the first analog solution for level-independent shaping of envelopes allowing transients to be accelerated or slowed down and sustain prolonged or shortened. The degree of dynamic processing required to do this couldn't be duplicated even using a chain of several conventional compressors, yet only two controls per channel are required to allow the user to completely reshape the attack and sustain characteristics of a sound. Attack can be amplified or attenuated by up to 15dB while Sustain can be amplified or attenuated by up to 24dB, enabling weak drum sounds to be made much more percussive and powerful, or for over-percussive transients to be softened. All necessary time constants (Attack, Decay and Release) are automated and optimised adaptively in a musical manner according to the characteristics of the input signal. This results in natural sounding signal processing and fast operation.

How does the Transient Designer work?

The Transient Designer uses envelope followers to track the curve of the natural signal so that optimum results are guaranteed regardless of the input signal's dynamics (for more information refer to Tech Talk). Because of the level-independent processing inherent in Differential Envelope Technology (DETª), manual threshold adjustments are not required. In order to maintain the cleanest possible signal path, the Transient Designer uses the excellently specified THAT 2181-VCAs, which are especially natural sounding, transparent and create minimal distortion. High amplitudes are processed without damping of high frequencies or reducing bass.

Sound benefits

• The attack of a bass drum or a loop can be emphasised with a single control to increase the punch and the penetration in the mix
• Sustain of snares or toms can be shortened very musically for more transparency
• Beats any gate for drums – must-have for live engineers!
• Acoustic or electronic instruments get "in the face" sounds or can be smoothened with lower attacks
• De-verb capability! Less Sustain shortens reverb flags of any sound (more elegant than gates do)
• New panorama effects based on different dynamic effects per channel – great for loops

Other features and connectivity

For stereo operation the Link function connects channel pairs (1 and 2 and/or 3 and 4). Linked channels are controlled by the same side-chain voltage so as to maintain a coherent and stable stereo image. When operating in Link-mode, the control elements of the first (or third) channel, including the Active switch, control the second (or fourth) channel, too.

Each channel is equipped with a relay hard bypass circuit to ensure a minimum signal path when the process is bypassed and to provide power fail safety. The Signal-LEDs allow to quickly monitor the signal flow, which is particularly important if the four channels are connected to a patchbay. The Transient Designer is fitted with XLR-connectors for balanced operation. To improve signal quality, SPL has developed a new hybrid-component balanced input/output stage using all laser-trimmed resistors with a tolerance of 0.01%. This approach has resulted in an exceptionally high CCMR (common mode rejection) better than -80dB at 1kHz.

Transient Designer rear panel view

Specifications

Input & Output

• Instrumentation amplifier, electronically balanced (differential), transformerless
• Nominal input level: +6dB
• Input impedance: = 22kOhms
• Output impedance: <>
• Max. input level : +24dBu
• Max. output level: +22,4dBu
• Minimum load ohms: 600Ohms
• Relay Hard Bypass
• Power Fail Safety

Measurements

• Frequency response: 20 Hz - 100 kHz
  (100 kHz = -3 dB)
• CCMR (common mode rejection): - 80dBu @1kHz
• THD & N: 0,004% @1kHz
• S/N CCIR 468-3: -89dBu
• S/N A-weighted: -105dBu

Power Supply

• Torroidal transformer: 15VA
• Fuse: 315mA
• Ground-Lift switch
• Voltage selector

Dimensions

• Housing: Standard EIA 19"/1U, 482 x 44,45 x 237mm
• Weight: 3,4 kg

Videos before the storm

You are awsome!

Once again, I could not say thanks enough to the people I met in Vietnam for the past month. I was sad to leave good friends behind but I will see you again soon.

I would like to thank you for your wonderful friendship and smiles+ice tea and coffee...

-The gang at Galaxy Movie theater, casts and crew of the movie 'the Rebel' :
Nhan (mr. dictionary): you are harmless when you drunk
Nam (Sumo): "...from now on, you're no longer a person. You are a number, no mother, no father." line from 'The Rebel'.
Son Oi(Asst. Director): I will see you in UCLA in June.
Ut Chang, Phuong (Just Men haircut), be Phuong, Bao (architect)
-Hero In Danger:
Phuc (mr. crazy), Quoc (curly), Cuong (battery Cuong), Bao (mr. safety net)
-Atmosphere:
Truong flying V, Hoang, Tung, Linh for boiled quail eggs, Saigon Beers, kho muc, thuoc la Con Meo, and motorcycle rides.
-Unlimited:
Thanh Bo...the man with the great voice and skill on the Amek mixer at Viet Tan Studio and tickets to HTV shows.
-Gat Tan Day
Long...quick ice tea but good conversation.
-Lazy Dolls
"...you think you know, why don't you get on stage and see for yourself." the drummer girl said.

Special thanks for the wisdoms and advices with great respect for:

-anh Trang va chi Hue: 'Yes, I will gladly become your adopted brother.'
-Huong and Doan Khoa: There will not be enough Bun bo Hue to keep us apart...or maybe pho, xoi, hay la cam tom bi? How can Vietnam do without you two?
-The great artist family of Dao Minh Tri: 2 bac, Hung va Tung...the wonderful arts, friendship, and teaching.
-Thien and Trang: advices, strategies, friendship, coffee, dinners, company, art, film, and American Idol.
-Quynh and Rob of Quynh's art gallery: thank you for the smokes.
-the gang at 3T for a great evening at your place.
-Anh Tai at Con Ca Vang
-Bao and Vi: you two rocks!
-RMIT
-Chu Tan at Viet Tan studio: For the great conversation in audio, coffee, ice tea and wisdom. I will see you again in 3 weeks.
-the gang at Long Thanh golf course :
Anh Bac Nam: laugh and Mercedes rides.
Nghiep va Quan for your money!
-the gang at Song Be golf course:
Hung of Zone Golf: golf, rides, lunch, boozes, more boozes, and dirty stories.
Phi: for rides, golf, money, and more dirty stories
Thinh: for golf and things not to do when on the golf course in Vietnam.
Jimmy: Rides and Kut Kit
Dragon Alex: Mr. Aussie for the lighter and cig.
Lam (le Chevral), Lam (IDG), Alex, Tuan, anh Chanh (Cargill), Tung (HSBC).

...Let's not miss Duc Tri, Ho Ngoc Ha, Thuy Tien, Cam Van, Khac Trieu, Tung Duong, Linh Microwave, Hai audio, chi Oanh at Ti Mau, anh Hien & chi Ha, Phuoc Sofitel.
...and lastly Mike, Steve, Anita, Gavin, Db Dave at Meyer Sound for constant support for Vietnam. Eva Manley of Manley labs, Scott at Cranesong, Mark of API, Scott at LAchapel Audio, Fletcher of Mercenary Audio.

I may miss somebody but do know I thank you and appreciate all of you. Will see you in 3 weeks.