The new PS website requires you to login to enjoy its many features that are specific to each user.  For example, if you wish to take advantage of PS Audio’s extended warranty program by registering your product, or managing and controlling your products, getting updates, email subscriptions, Community forum access and notifications of new products and events through the PS Audio Newsletter, then you must register with the site.

Many of you were previously registered on the old website.  When we switched sites it was not possible to move users over to the new site for security reasons.  Each user has a unique password that you created and we do not have access to those passwords and cannot move them.  This requires us to ask you to register as a new user.  You only need do this once and we’re good to go.

If you were formerly registered all your information is still saved, including your product registration information.  Once you register on the new site you’ll have access to all your registered products and be able to control and manage them online.

If you have yet to register as a new user on this new website, you are not able to login.  You cannot use your previous login to access the site.

Here is the process.  It’s rather simple and quick.

Click on the Register New User link on the top of the website.

Once at the registration page all you need to do is fill in the fields and you’re almost finished.  Couple of notes:

• We encourage you to use your email address you used to register your products (if you did).  Product registrations are all tied to the email address you originally used to register those products.
• You may choose any user name you wish (as long as it is not taken) but it must be all lower case and contain only letters.
• The Captcha form is a pain but we need to make sure the web robots don’t pollute our site.  Fill in the crazy words as asked. 

If you registered and didn’t receive an email with your temporary password, please contact us at support@psaudio.com and email us your details and we’ll get you handled.

I already registered my products on the old PS Audio site and now they do not appear in My PS Audio page under My Registered Products.

How do I locate them and do I have to register again?

No, your products are safely stored with us and transferred from the old site database to the new one.  To find your products simply click on the link at the bottom of the box on My Registered Products page marked “Missing Something?” and enter the email address you registered your products under and the system will locate your registered products for you.

My PS is where you go to set your personal preferences, register your products, get help and service, control your equipment.  It is your personal webpage for everything you need to do with PS Audio and PS Audio equipment.

My PS is is located in the main menu bar of the website next to Power.

If you do not see the My PS category available as shown in the picture, it most likely means that you are not logged into the website.  For the My PS section to work the website must know who is trying to access their personal webpage.  To do this you must be logged in with your user name and password.

If you do not have a user name and password or you are a new or returning PS customer you must register on the site as a new user (even if you were registered before on the old website).  It only takes a minute and when you’re done, you will have full access to the My PS section of the website.

The PS Audio Forums are now called Community and the Community Forums are located in this same Resources section of the website you are in now.

Once registered on the main site you should be able to go join the forums. PS Tracks is a separate login so if that’s where you are trying to comment on, that needs another registration and login.

The PS Community forums however should be together with your main registration. If you are having trouble with this, please let us know and we’ll investigate.

Once you’ve registered your product on the website, go to My PS and then go to Updates.  If there is an update available it will appear there.

Make sure you can see your registered product in My Registered Products in your My PS section of the website.

If you do not see My PS in the top menu bar of the website, chances are good you are either not registered on the website or have not logged in.

NativeX is our product name for the PS Audio digital lens technology.

 NativeX will reduce incoming jitter levels to below 1 pico second.

The DL3 usb is synchronous. The PWD usb is asynchronous.

Yes this is the new usb driver for PC

The new website Community forums section is a great way to get involved in the high-end community as well as PS.  To access the Community forums, you go to the Resources tab and click on Community.

You must be logged in to post or to change your user settings, signature or picture.  If you need help registering or logging in, refer to the Knowledge Base for help.

Once you are logged in to the PS website, it’s easy to manage your email subscriptions for the forums, your signature and your picture.  Simply look at the top blue bar of the forum and you’ll note your name (user name) in the heading.  Click your name  and you will see all the settings area as in the picture below.  Have fun and let us know if you have any questions.

The VTA (Vertical Tracking Angle) of a tone arm on a record player is a critical adjustment for good sound.  Many people would encourage turntable owners to play with the angle until the vinyl sound right.  We would agree.  However, there is a bit more science to it than just listening.

The object is to match, as closely as possible, the same VTA as the original cutting head for the master was set.  Typically, there is an easy standardized method – that relates to the degree of angle used by almost all cutting masters.  The Vertical Tracking Angle was not always standardized. But since the stereo disc was launched, the angle was defined at 15 º and was changed in the nineteen seventies to 20 º. That is why the Ortofon SL-15 became SL-20.The

To adjust your VTA properly, you need to find the adjustment on the base (post where the arm is mounted to the turntable) of your pickup arm that allows you to raise or lower the back (the end opposite of where your cartridge is) of your tonearm. Look at your owners manual (if you still have it), go to the audio dealer from whom you purchased it or contact the manufacturer for help in finding out how to make this adjustment. CAUTION: On most tonearms, you will NOT be able to adjust the VTA while playing a record or with the stylus even resting on a record (without destroying the record and/or cartridge cantilever or stylus). YOU HAVE BEEN WARNED!

For initial setup of your VTA, place a medium thickness album (no 180 gram re-issues or flabby RCA Dyna-flex Red Seals) on the turntable and place the stylus on the record (do not have the turntable rotating for these adjustments). With the stylus resting on this medium thickness album, the bottom of the cartridge should be parallel to the album. By this, I mean the flat area near the front of the cartridge where the cantilever / stylus assembly protrudes from the bottom of the cartridge. CAUTION: Make all adjustments on the tonearm with it sitting on the tonearm rest. You now have a good starting point to find where the nominal VTA setting is located for your arm / cartridge combination. Select 3 records from your collection with which you are familiar. You will use them to find fine tune the nominal starting point for your VTA adjustment. One of them should be what I will call a normal thickness album (London CS 6xxx or STS 15xxx (orange – silver label), RCA Shaded Dog, non 180 gm. Chesky, etc.). The next should be a thick album (Decca or EMI reissue, Mobile Fidelity 2-xxx series, etc.). The third album should be a thin album like an RCA Dyna-flab.

After setting the starting point of your VTA session using your eyesight, listen to a section of all 3 albums. What you want to l sisten for is the seniority of the strings, the “air” around the instruments and the width of the hall. If you set your VTA correctly for nominal thickness albums, you will hear the following:

1.The medium thickness album will have extended stage width, a hint of air or rich harmonics around the individual instruments and singing in the upper strings without any stridency.

2.The thin album will have good stage width but the strings will sound unnatural, edgy and irritating.

3.The thick album will sound slightly muffled, with a lack of high frequencies and air around the instruments.

If this is not what you hear in your comparison, your VTA is not set properly for medium thickness albums. If the thin albumounds correct, the back or base of your tonearm needs to be raised about 0.010″ (0.4mm) (the thickness of a cover of Ultimate Audio) for medium thickness albums. If the thick album sounds correct, the base of your tonearm needs to be lowered about 0.010″ (250 micrometers) for medium thickness albums. A few passes at this and you will learn what to listen for when you adjust your Vertical Tracking Angle.

To reiterate, once you have found the correct VTA setting for a medium thickness album, you can use this starting point when you want to adjust your VTA for best sonics. For very thin albums (flabby Red Seal), or Angel and late Columbia, you will have to lower the back of the tonearm by as much as 0.005″. For very thick albums and many of the Decca, Classic or EMI reissues, you will have to raise the back of the tonearm by as much as 0.010″ or 0.015″. Also remember that during the course of the life of your cartridge, the nominal setting will change as the cantilever ages and flexes making it sound as if the back of the tonearm is too low. After a short period of time of focusing on the sound (and not the music), you will learn to identify when the VTA is adjusted properly. After this adjustment is correctly made, listen and enjoy the music.

Sedrick Harris, Paul McGowan

Typically the longest connection in our AV system is between the receiver or power amplifier and the loudspeaker. The quality of this connection can play a large part in the way your system sounds. We would first recommend that you purchase high quality Audiophile speaker cables to make sure this connection is handled properly. However, these can range from several dollars per foot to several thousand dollars per foot.

Here’s an inexpensive solution: as is true with power cords, thicker is usually better. If you are currently using thin lamp cord for your speaker connection to your receiver or power amplifier, you can make a difference by doing two things: purchase thicker, multi-stranded wire from your hardware store, and separate the two conductors.

Visiting your hardware store, acquire the appropriate length of either 14 gauge or 12 gauge multi-stranded copper wire, such as a heavy lamp cord. Then, before you strip the two ends to connect to your speakers and power amplifier, separate the two conductors by peeling them away from each other. Now, instead of one length of two conductor wire (one for the plus and one for the minus), you will have two lengths of single conductor wire. Connect up your speakers to your amplifier using these two separate conductors.

The reason that I suggest separating the two conductors is to lower the capacitance of the cable.

Keep the two conductors apart from each other by a couple of inches.

Clearly some of the decision is purely personal if wattage requirements are not an issue.  The smaller full size G Series amplifier and Control amplifiers have the same output power rating as the Trio A and C100’s do, and they can both power the same loudspeakers equally.  Most loudspeakers are just fine with a couple hundred watts per channel and either series will be just fine with most loudspeakers.

The decision points come down to budget and system expectations.  For equal wattage ratings (as mentioned) the Trio Series is perfect for a less critical music application such as an office system, study, bedroom or even a dedicated high end setup.  The more revealing nature of the G Series would be appropriate for any two channel system serving a customer with extremely high expectations.

At the end of the day, both the Trio and G Series are based on the same technologies and engineered to the same high standards.  Either would be great in almost any setting one could imagine.

Most DAC’s (Digital to Analog converters) regardless of their price use very similar D to A chipsets from only a handful of manufacturers like TI, Cirrus Logic and ESS (Sabre). 

While there are certainly differences in these DAC’s it’s unlikely any one would be singled out as significantly better sounding than the others.  It turns out that the big differences we hear between DACs has far more to do with their analog output stages than anything else.  A DAC has two parts as you might imagine.  A digital side and an analog side.  For a digital side with the same resolution and bit depth (44kHz to 192kHz and 16 bit to 24 bit) the digital sides will be very similar.

The analog side is where the big differences lie.  The type of circuit (solid state or tube), the implementation of that circuit (class A, high feedback, low feedback, IC or discrete) have the majority of  effect on how a DAC sounds.

In the DLIII PS Audio leveraged its long history of DAC designs to great advantage (we released the world’s first DAC to the high end).  By eliminating all IC’s in the analog output stage and implementing a fully discrete, class A, low feedback JFET design, the PS Audio DLIII manages to sound more open, sweeter,  warmer and more like music than just about any DAC at any price.  To pull every last bit of music out of the DLIII we also use a very large power supply.

The secret is in the analog stage and power supply and when evaluating DACs this is where you should be looking first.

The term “digital amplifier” is, typically, a misnomer.  What most people refer to as a digital amplifier is properly refered to as a class D or PWM (pulse width modulated) amplifier.  PWM class D amplifiers are, for the most part, analog based designs.  They are referred to as digital amplifiers because of the way they amplify; using pulses to create the audio output in a fashion similar to a digital audio product.

A class D power amplifier’s primary advantage over a linear amplifer is efficiency.  Class D amplifiers typically enjoy an efficiency of over 90%, vs. 50% for a linear amplifer.  The bottom line is that class D amplifiers produce far less heat for any given output power they deliver.

To understand how a class D PWM amplifiers works, it is first important to understand how a classic linear amplifier works.

Let’s look at a traditional power amplifier

A traditional tube or solid state power amplifier uses the output devices like a constantly adjusted valve.  Picture water going through a pipe with a faucet valve regulating the flow of the water.  Your hand (in this example) is the musical signal, and the water is power to your speakers.

If you could move your hand on the valve quickly enough, you could smoothly turn the water faucet up and down creating a flow of water that mimicked the action of your hand.

With very little force from your hand, a lot of high pressure water can appear at the output of the faucet.  This is the nature of an amplifier, a very small force controlling a very large force.  The small force is the output from your preamplifier, and the very large force is the output of your power amplifier.

The problem with this approach is two fold: it isn’t very accurate and it isn’t very efficient.

In a standard analog power amplifier, the musical signal at the output of the power amp is not an exact copy of the musical signal at the amp’s input.  When an amplifier’s output does not match its input we would say it has a linearity problem.  There are a number of ways this can be compensated for and that fact, in a nutshell, is what power amp designers have been struggling to correct for – in one way or another – for years.

Good for cold weather

Ever noticed the use of big heat sinks on power amplifiers?  These are necessary because traditional power amplifiers generate lots of heat and that heat needs to be dissipated away from the output transistors.

The heat that is generated in a power amplifier is wasted energy.  A typical class AB amplifier is only 50% efficient.  This means that 50% of the energy consumed goes into powering your speakers and the remaining 50% is converted into heat.

Why the heat?

Traditional power amplifiers produce heat when the output of the amplifier is somewhere between one of two possible states: all the way on or all the way off.  In a class A or class AB amplifier this is virtually always.  There is rarely ever a time that a power amp is all the way off or all the

way on, so they are always generating heat.

Remember our valve explanation above?  The entire time the faucet is anywhere other than all the way on or all the way off, it is producing heat.  So too with a traditional power amplifier.

Now, let’s look at a Class D power amplifier

One way to create an amplifier that produces little to no heat is to have the output devices either all the way on or all the way off.  Sound familiar?  Yup, the use of all-the-way-on (1) or all-the-way-off (0) is what we are all becoming familiar with: PWM.

But while this is a great idea, how the heck do you make all of the musical signals in between all-the-way-off and all-the-way-on?

The answer to this perplexing question has been around for many years, but when it was first invented it wasn’t used for producing music, rather, it was used to regulate the speed of electric motors in trains and industry.

A bit of history on PWM amps

The first usage of a Class D approach to power amplification we could find was produced by a man who gave Bill Gates a shot in the arm, Sir Clive Sinclair.

Now, here’s a bright guy who invented the world’s first pocket calculator and is credited with being amongst those that started the PC revolution.  The Timex/Sinclair computer and the Executive calculator were genuinely innovative products from a genuinely innovative kind of guy.

In 1964, Clive Sinclair started a company called Sinclair Radionics Ltd. that produced a Class D PWM power amplifier known as the X-10, designed by engineer Gordon Edge.  This tiny power amplifier was purported to produce a whopping 10 watts of power but, in reality, produced only a couple of watts.  This was later to be replaced by the Z-12 that actually produced a real 12 watts.

PWM is born

What engineer Gordon Edge did, to solve the problem of heat, was to use a system of 1’s and 0’s in varying lengths.  By controlling how long each of the two states remained on (or off), a signal could be produced that, after the appropriate manipulation, would actually produce music.  That process is known as Pulse Width Modulation or PWM.

The first Audiophile version of the PWM process that we are aware of was an Infinity systems product known as the SWAMP amplifier.  The SWAMP was a revolutionary product as envisoned by Arnie Nudell and his then partner in Infinity, John Ulrich (now running Spectron).  As the story goes, research efforts for this product nearly killed the company in its infancy so expensive and back breaking was its design challenges.  The SWAMP was finally produced in the mid 1970’s and despite some horrific reliability problems, the amp received critical acclaim.

Following the cancellation of the Infinity switching amplifier, Infinity’s very next attempt at a power amplifier was the Infinity Hybrid Class A , an artful blending of tubes and a traditional Class AB solid state design.

Modern day uses of PWM include the Laser disc, DLP projectors (Digital Light Processing displays), DSD (Direct Stream Digital) or SACD, Jeff Rowland Designs 501 mono blocks and of course, the PS Hybrid Class A power amplifier and its more modern counterparts the GCA and GCC series.

How PWM works

Think of PWM as a digital data stream that does not require a D to A processor to convert its signal to analog.  In fact, the PWM process is 100% analog.

A digital, or PCM signal (Pulse Code Modulation), which is what is used in a CD or DVD, converts the musical signal into a series of numbers – each number representing a voltage level.  The higher or lower the number, the higher or lower the voltage.  The CD disc itself merely stores these numbers in a computer like memory.

To play back a CD or DVD using PCM you need another computer to read these numbers and process the conversion.  This “computer” is called a D to A processor.

PWM does not need another conversion process, so it does not require another device to interpret its code into audio.  PWM is a far more direct approach than PCM, and as mentioned is not digital in nature.

The long and short of it

PWM is simple to understand.

At the output of a PWM amplifier there is an electronic switch, rather than an analog “valve”.  The switch has one of two states: on or off.

In order to duplicate a musical signal, the output switch turns on or off in longer and shorter time periods.  If the volume of the music is low, the on/off times are very short.  The louder the music gets the longer the switch stays on.

The switch is connecting your loudspeakers to the amplifier’s power supply (like a battery).  So, when the switch is on, the full power of the amplifier is applied, and when the switch is off the power supply is disconnected from the loudspeaker.

The longer the power supply is connected to the loudspeaker, the more your woofer or tweeter move, thus creating sound.

If you do this fast enough, then the speaker doesn’t know it is being turned on and off and it sounds smooth, like music.

Going to the movies

A good example of how this works is a moving picture or video.  What you view on the screen looks like it is fluid, smooth and not full of one after another transitions.  We all understand that a movie is really just a series of individual still photographs, transitioning quickly enough to fool our eye into believing they are “moving”.  Our eye can be fooled if we present it images 24 times a second.

So too with a PWM amplifier.  Moving from on and off quickly, we can fool the loudspeaker and our ear/brain mechanism into believing it is smooth and lifelike (because actually it is).  In this instance, we need to go a bit faster than the movies and turn these on/off transitions at a much quicker rate: 350 thousand times a second.

Let’s look at an example

In this example, time is represented in the horizontal plane from left to right.  Amplitude (volume) is represented in the vertical plane, or up and down.  Note how the individual bars, which represent the on time of the output switch, stay on longer as the amplitude goes up and down.  This is the code that PWM uses.

To convert this code into a smooth output, we need to remove the transition marks and fill in the gaps.
To do this we require a filter.

A filter uses elements like capacitors and inductors (coils of wire) to store and release energy.  At the beginning of a switch transition, some of the very quick energy is used to charge up the filter elements, thus making it smoother.

When the output switch turns off, the energy that we previously stored in our filter is released, thereby filling in the gap between the on and off.  This is the smoothing action that we need to complete the conversion.

And, that is basically it!

The advantages

Remember when we started this discussion we mentioned that analog amplifiers have two problems: they are not linear and they produce heat.  Well, guess what?  PWM amplifiers are absolutely linear and produce almost no heat.

With respect to linearity, which is the ability of the power amplifier’s output to accurately follow the amplifier’s input, traditional power amplifiers aren’t very good because they rely on the valve’s characteristics to function perfectly.  In a PWM amplifier we don’t care how accurate the valve or switch is, because it is only turning either on or off.

Perhaps a good analogy would be the difference between an artist and a non-artist.  Anyone can draw a simple line on two planes, while only an artist can draw everything in between and make it look like a photograph.  How close the drawing looks like the real thing is completely dependant on the skill of the artist.

So too is the problem with using a transistor or a tube as a perfect valve.  At either end of their amplifying limits they become non linear – they are NOT perfect valves.

So, the linearity of the PWM or switching amplifier is nearly perfect.  Voltage in equals voltage out.

And when it comes to heat, a PWM amplifier loses only between 5% and 10% to heat, the rest of the energy being sent directly to the loudspeaker.

The disadvantages

PWM amplifiers are rather tricky to design, they can radiate airborne noise, and they are very sensitive to different loudspeaker impedances (the later problem of output impedance finally solved in the GC series of power amplifiers from PS Audio).

A design challenge

Designing a traditional power amplifier is almost trivial to designing a proper PWM based power amp.

The disciplines are entirely different and require a unique knowledge of RF and digital techniques not required in a traditional power amplifier.

PWM circuits are also pushing the envelope of technology.  The output devices used in these designs are typically MOSFET’s and in a full range digital power amplifier these MOSFET’s are required to operate under conditions that semiconductor manufacturers couldn’t even dream of meeting even just a few years ago.

A question of noise

Perhaps one of the most daunting of tasks with a PWM power amplifier is noise; radiated noise.

The on/off switching of the power supply as described above can create a radio signal.  In fact, if you wanted to build a radio station transmitter you would build something very close to a digital power amplifier.  Instead of an output filter you would use an antenna.

While radio station engineers do their best to get as much of the energy developed by their digital power amplifiers into the air, Class D audio engineers do their best to keep as much as they can out of the air.  This task is quite a challenge for every engineer that tackles the dream of a high efficiency power amplifier and can be absolutely daunting in its successful completion.

PS Audio GCA and Trio series amplifiers solve this classic noise problem quite well and are quiet enough to place even the most sensitive of equipment near the amplifier with no ill effects.  In a PS power design, noise is not a problem.

Load problems

From a practical standpoint load variations of loudspeakers, as powered by a PWM amplifier, are one of the biggest hurdles that digital amplifier designers face in making an audio amplifier that sounds good on every system.

In fact, the PS Audio GCA and Trio series of power amplifiers, integrateds and multi channel amps are perhaps one of the very few amplifiers in the world to really solve this problem (in an Audiophile friendly way).

Remember the output filter we spoke of used to smooth out the switching transitions of the PWM circuit?  This is where the problem resides.

The problems with filters

All filters are designed to work into a specific load.  That means that a filter’s action will be consistent only if the input and the output to the filter remain constant.  This is true for most every filter.

The problem is the loudspeaker itself.  4 Ohms?  8 Ohms?  The loads are different.  To make matters worse, a 4 Ohm or 8 Ohm speaker is not really what it claims.

Ever notice in loudspeaker manufacturer’s specifications the word “nominal” impedance?  This refers to the fact that the speaker has an overall impedance of whatever they specify (4 or 8 ohms).  Unfortunately, the actual impedance of a loudspeaker varies with frequency.

If you are playing a bass note, for example, the loudspeaker impedance is anywhere from 2 Ohms to 30 Ohms!  This is quite a swing.  What’s a poor output filter on a digital power amp to do?  Remember, its performance is dependant on the load being constant.

The results can range from a dull muddy sound to a hard and bright sound all from the same amplifier.  It is very dependant on the type of speaker you have.  Beware of Class D amplifiers and their interaction with speakers!

15:06:38

A Digital to Analog Converter. Simply stated, it is a separate device used to convert the digital signal, created by a CD transport, music server or computer, into an analog signal we can hear as music. The first separate DAC used for High End Audio was invented by PS Audio. Later introductions were by Arcam (the black box) and Theta.

DAC’s have four main component groups that make up their workings: a receiver that takes in the digital data and separates out the clock signal from the transport, the digital filter that gets rid of unwanted digital signals, the D to A processors where the actual conversion from digital to analog takes place, and the analog output stage that amplifies the converted signal and gets it ready for the outside world.

There are two types of DAC’s, one bit and multi bit, with the standard being the multi-bit.

Digital audio has been around for many years but was first popularized by the introduction of the CD player in 1982.  Up to that point, most consumers were unaware you could convert the music we heard from vinyl records and tape recordings to a language understood only be a machine.  Digital audio in its raw form is nothing more than a series of bits ( a bit is a two-state on and off marker referred to as 1 and 0), the binary language used by all computers.

How do these bits work and record and playback music?

First, let’s consider how music is expressed in analog terms. When a microphone pickups up sound, it converts it into electricity or voltage. Voltage is what comes out of a battery. So, let us envision that we have a battery with 10 volts available, and that it is hooked up to a microphone (somehow). Let us further envision that when we speak or play music into this microphone, it takes the voltage out of the battery in small or great amounts. Small amounts when we whisper into the microphone, and large amounts when we yell. Therefore, the louder we speak into the microphone, the more voltage is produced out of it (amplitude).

Now, we have a rising and falling level of voltage (amplitude) out of our battery caused by any change in volume of speech into the microphone. Also, this voltage is turning on and off faster when we hit a high note (AC), slower when we hit a lower note, therefore the speed at which this voltage moves (frequency) is determined strictly by the pitch of one’s voice.

Putting it another way, we have a voltage that its bigger and smaller with the level we speak (amplitude): those same voltages are turning on and off (AC) in rythym with one’s speech; and the speed at wh ich those voltages turn on and off (frequency) varies in direct proportion to the pitch of one’s voice.

If we then took this varying voltage and plugged it into a power amplifier that was connected directly to a pair of loudspeakers, we would hear ourselves over those same speakers. And, if we put this varying voltage (the output of our microphone) into a tape recorder, it would record all of these changes in amplitude and frequency so we could play it back later.

OK. Let’s go digital. In its simplest form, we’re going to change the higher and lower voltages into numbers, and then record those numbers. That’s basically it. The rising and falling voltages (amplitude) are electronically measured and those measurements converted to their numeric value, and that numeric value then recorded computer style.

The 1’s and 0’s you hear so much about are merely a counting scheme used to record bigger and smaller numbers. How? By a counting method that is bonehead easy to understand. Let’s use a simple 4 bit system to count, for example, and note the drawing below. There are 3 vertical lines and 16 horizontal lines. Along the top of our box (now full of squares) going in a horizontal direction we’ve marked #‘s 1,2, 4, 8 (one number in each box). These 4 numbers (1,2,4,8) are the digital’meaning’ or representation of the selected squares below. The squares below use ‘X’ and ‘0’ to represent ‘1’ and ‘0’ that we hear so much about. Using only ‘1’s’ and ‘0’s’, let’s start counting in binary fashion.

In the first horizontal row (below our written numbers) note 0,0,0,0 in each of the 4 horizontal squares. This represents zero.

The next row has three ‘0’s’ and one ‘X’. Note the ‘X’ is in the same column as the digital representation for 1 (one).

If we need 3 (three), then we’ll place an ‘X’ in both the ‘1’ and the ‘2’ column. The computer knows to add these two together, so we get ‘3’. Study the chart below and it will become obvious.

See how, using only 1’s and 0’s a computer can count? Instead of a 4 bit system that can only count 16 numbers, 16 and 20 bit and 24 bit systems can count numbers as in the millions.

Classes of biasing of an amplifier.  Biasing refers to a constant power being consumed by an amplifier or its output stage.

The term class A refers to biasing of the output stage (in a transistor power amp). Bias is an engineering term that actually means what it implies (i.e. Leaning in one direction). By that, we refer to the amount of current flowing through the output stage.

Class A is when both devices conduct at all times while Class B is when only one device is on at any instant. These are both limiting cases and there is thus a setting between these two limits where, for small signal levels, both devices conduct but for larger levels, only one conducts. This is termed Class AB.

Class AB is not a true class, but is a very common term applied to the biasing levels used in most audio amplifiers. The output stage is biased to carry a quiescent current significantly less than half the maximum output current, (as needed for full Class A) but sufficient to keep both devices running in Class A for small output signals. However as the output signal increases, the amplifier becomes Class B with one device cutting off on each half cycle. This biasing scheme effectively moves the linearity curves toward one another resulting in a transfer curve that is more linear as it passes through the origin. This scheme approaches the efficiency of class B yet offers class A distortion levels (nearly) especially for small output levels where class B suffers most.

It is particularly significant that the distortion is reduced for small signal levels, as it is at low levels that the human ear is most sensitive to distortion.

With the exception of Nelson Pass’s (Pass Labs) amplifier that essentially has one ‘sex’ (polarity) of output device handling the entire signal, all transistor amplifiers have (in essence) two output devices. One output device handles the positive (plus) half of the signal and the other handles the negative half of the signal. If you can envision such an arrangement, a possible problem might come to mind – the transition area. Like a relay runner handing off the baton to the next runner, the first half of the two-device output stage ‘hands over’ the signal to the second half of the output pair ;this transition is the problem. As the signal goes from the positive device, to the negative device there is a moment when neither device is handling the signal, creating a ‘gap’ in the music.

Class A biasing in its truest form means that there is a lot of current running through these two devices all the time. When you run current through the devices, it creates heat, and the more you run through the stage the greater the amount of heat. A true Class A amplifier runs the same amount of current through its output stage as it is expected to deliver to the load (the speaker). So, for instance, a 100 watt (rated) class A amplifier draws 200 watts per channel at all times (even when there is not a signal). When the amplifier is asked to deliver 100 watts of power to the loudspeaker, 100 watts (or half of the current) goes into the load (speaker) and the other half continues to go through the output stage.

One interesting fact is that under full power delivery conditions (when the amp is putting its full 100 watts into the speaker), the Class A power amp actually runs cooler. This is because only half of the power is being converted into heat in the amplifier, while the other half is busy driving the loudspeaker.

Preamplifiers control the volume of your system and select the appropriate input to listen to.  Consider them the control center for your hifi.  The two main types of preamplifiers are passive and active but both perform the same function.  A passive preamplifier has input switching and volume attenuation control but has no active amplification inside.  Passive preamps require no electricity to operate.  An active preamp, which is the most common type, is a passive preamp with an active amplification stage added to its output and can produce volume levels greater than the preamp’s input.

A preamplifier can be broken down into its component sections quite easily. Typically, there is the input selection stage first. The input selection stage is usually nothing more that a number of input connectors, either RCA or Balanced (XLR), and a switch arrangement set up to choose one of the inputs (such as phono, tuner, tape, etc.). Once an input has been selected, it next goes to the volume and balance control.

These simple variable resistive elements (potentiometers or pots) vary the resistance the signal goes through thus making the level of the signal go higher or lower depending on the direction the control is turned.

If the preamp in question is a passive preamp (or ‘pots in a box’), the signal is then routed either through an electronic buffer or straight out to a set of jacks to meet the outside world. This type of arrangement has no gain. In other words, whatever level of signal goes into the passive preamp is the highest level that can come out. even if the volume control is turned all the way up.

A standard preamplifier, though, has a gain stage. This can be either a tube gain stage or a transistor gain stage, and there are literally hundreds of different designs to accomplish this. These gain stages provide amplification of signal, so output signals can be stronger than the signal fed into the inputs, depending on the volume control settings.

PS Audio has manufactured many different models of preamplifiers over the years, the current day analog preamps are the Trio A100 and the G Series GCP200.

Using iTunes Music Manager to play to a USB audio device like the PerfectWave DAC, DLIII DAC or any USB enabled DAC is easy once you set it up properly.
As it turns out, the settings you can make for iTunes cannot be found in iTunes itself, but in the technology that it uses: the good old Quicktime Player! All filters and codecs that iTunes can use are actually ‘hidden’ in Quicktime.   This is a necessary step for high-end performance because iTunes doesn’t sound very good running on Windows XP or Windows 7.

In order to provide the most fool-proof method of installation, iTunes picks whatever settings are specified in Windows’ default sound settings. This means that it probably picks the integrated soundcard on the motherboard. And that is a bad thing. Why? First off, it will use kmix, Windows’ internal mixer that resamples and levels all sounds into the same stream in order to be compatible with any sound you could possibly try to reproduce. Second, the built-in soundcard may not be the best card you have installed.

There are some settings you can make for iTunes but you need to look for them in the Quicktime player as can be seen in the image above. Choose preferences and then Quicktime preferences. This will give you the following window.

Contrary to many other applications, there is not an awful lot you can set here but still this step is very important. You can choose between Safe Mode (waveout only) or Direct Sound. Now, even though officially DirectSound should be better and it is therefore used by default, we strongly advise to use Safe mode via Wave Out because it sounds an unbelievable amount better than the Direct Sound option. While the latter sounds thin, flat, slow and unexciting, the Safe mode sounds much more musical, fuller, more fluid, grander and simply more like real music.

This was a very important setting to be made here. But as you’ve probably already gathered, there’s also the bitrtate/samplerate option. This one you can use to your own insight. In general: if your source files are 44/16 (CD-based mp3′s or wav’s) you best pick the same settings. They should already be set like that by default as it is the most common used setting. If you choose a higher rate, the program will simply upsample. And it does that quite well. In my case I have set 24/96 as I have many LP-recordings made in that resolution and playing 44k files through the upconversion provides a nice airyness that I like. Opinions may vary though and if you are after bit-perfection, you better choose your rates accordingly.
As a side note, iTunes does not play FLAC. It does Apple Lossless however so maybe you could convert your files if you need to. This I have not confirmed yet.

USB audio devices are just like any other audio device to XP, and have the same options (DirectSound, WaveOut, ASIO/WDM, Kernel Streaming). With iTunes and XP, since ASIO and kernel streaming are NOT options, then, even with USB, you are not bit perfect.

There are other programs that can help you get bit perfect results if you wish such as Virtual Cable, Pure Music, and Amarra.  Please refer to their individual web pages for more information.  (thanks to Christian Punter from Hi Fi Advice)

If you are using iTunes or any of the iTunes based programs such as Amarra, Pure Music or Virtual Cable and wish to control the music experience with a hand held remote, an Apple iPod Touch, iPhone or iPad device and a free program from Apple are all you need.

The Remote app from Apple, available for free download from the Apple App Store provides a full music server quality experience for those of you using the USB inputs on your DACS, such as the PerfectWave Mark II or the DLIII from PS Audio.

In formation on the features of the Apple Remote app can be located here: http://www.apple.com/itunes/remote/

Fortunately with a MAC, as is the norm, it’s easy. Simply plugin your DLIII to the MAC and it should auto connect and be available immediately.

To play from the MAC in iTunes you will need to make sure Quicktime player is installed. Quicktime Player is how your MAC will play to the DLIII or the PWD PS product.

Should the MAC not auto configure the PS DAC, you may need to go to the Audio Midi application and click on the PS Audio DAC. The Audio Midi application is located in Applications->Utilities folder of the Finder.

Cct burn in can be a contentious issue. As long as you have a signal flowing

in the unit the cct will burn in.

There are two inputs on the PS Audio 200C and 200CX power amplifiers, produced in the mid 1980′s.  The standard input is a capacitor coupled input and the DC input is a direct DC coupled input.  The preferred input is the DC input as it sounded better without the capacitor in the signal path.

Years ago some preamplifiers were prone to DC on their outputs but more modern preamps are not so the standard input is really no longer necessary to use.

The PS Audio Soloist is the world’s first and only in-wall power conditioner offering full AC filtering, surge and spike protection.  They are not dificult to install and can make a world of improvement.  In this How To article, PS customers explain in great detail their experience with installing a Soloist and the results they got once installed.

Bill Harris

Today I installed my new PS Audio Soloist.  If I was installing this in a double gang box, it probably would have only taken a few minutes.  But my stereo was plugged into a single gang outlet and as you may know the Soloist is a double gang box size.  To assist me in the install, I bought a Goldblatt wallboard saw, since I knew that the new Soloist conditioner would require a larger space than my single gang outlet and somehow the hole in my wall would have to be enlarged.

Of course, before beginning, I located the breaker switch and ensured that the power was off, by testing it with a lamp plugged into the socket that I wanted to remove.

I removed the wall plate and the socket, which is when I realized that I needed to call my friendly electrician (an audiophile, it turns out).  I was unable to tell how the box was attached to the wall and felt that someone with experience should be involved.  We talked for a bit and he came over.

He tested for the stud position and I measured the distance from the end of the wall, thus we assumed that the stud was to the right of the box.  He used a box knife to cut around the single gang box to relieve it from the paint and with my new saw, enlarged the hole to the left side of the outlet to provide a way to feel for the nails securing the box to the stud.  He said he only cut when pulling, in order to prevent the wallboard from coming apart in chunks.

Then he began the task of prying the box from the stud.  Once it was loose from the stud, he managed it retrieve it through the hole (without dropping it in between the walls).

He then marked an outline in pencil to guide his enlargement of the hole to fit the new larger two gang box, packaged with the Soloist.  Rather than using the wallboard saw, he went back to using the box knife to slice a smooth cut through the wallboard.  He felt that this would further prevent the wallboard from breaking up.

Once the hole was sliced to size, he began fishing the wires through the back of the new box, by pushing open two of the slots.  The slots are designed to provide some grip to the wires to prevent them from slipping out.

Once the box was in the wall and the built-in flip out securing hooks were tightened, he drilled through the box into the stud and used three square head screws to secure the box to the stud, so that removing tightly gripped plugs from the outlet would not cause the box to move.

We twisted the wires together and secured them to the wires of the Soloist with the supplied wire twists.  The wires were squeezed into the back of the box as far as they would go and the Soloist was slipped into place.  Finally the supplied hex bolts were installed to secure the Soloist into the box in the wall.

We plugged in the lamp again and reset the breaker to restore the power.  That worked and the small green LED on the Soloist lit up to show that it was working.  Total installation cost $20 (I told you he was a friend, but we paid him more).

I vacuumed, picked up the tools and plugged in the power cables from the SimAudio amp and the Wadia player.

This is where it got interesting.  I played the Modern Times CD from Spyro Gyra and noticed that my speakers suddenly had more bass.  I had always thought that they sounded a bit thin and that I might need a bigger amplifier.  I guess I was wrong.  The instruments had better separation from each other and I felt like I was closer to hearing was laid down in the studio where it was recorded.

My wife, who is a singer, thought that the highs had more sparkle and were clearer.  The electrician was also impressed.  We played a Mozart piano concerto next and realized that our impressions were still valid.  My wife also plays piano and thought that the sound of the piano was more realistic with the Soloist.  We also tried voices with some Selah, the “In My Life…” title, and found that the voices had a fuller presence.  We switched to Bob Dylan’s Blood on the Tracks and she listened to the whole thing, especially enjoying the acoustic guitar sound.  We finally had to eat, but left the music playing.  Later, we went to Paul Simon’s Graceland.

I have to admit that I did not think that a Surge Protector/Conditioner would make a difference in the sound.  I knew about Paul McGowan’s pedigree and the reputation of his products.  But I had no idea that something this simple would sound so good.  In fact, I had used something like this before when recording vocal groups and heard no difference with it or without it.  But then, it wasn’t a PS Audio product, was it?

This Soloist product was a system upgrade for me.  At $200 US, it has to be the cheapest way to better sound of anything that I can think of.

It is not a tweak.  It is a bonafide system upgrade, a way to purer, more truthful sound.  You don’t have to do a blind AB test, switching back and forth to determine if there is a difference.  It is patently obvious.

I actually felt that I was hearing things that I had never before heard on these recordings.  I have heard people talk about more blackness (in the background) and a lack of haze or grunge, but you know, I don’t think that I heard it that way.  What I heard was more information.  I heard a sweet, beautiful, non-fatiguing sound (also a surprise).  I heard music.

Run to your dealer or logon to PSAudio.com right away and get some of these Soloist.  Otherwise you will be deprived of some beautiful music.  Forget about the cord elevators, tip toes and magic wooden blocks.  The Soloist from PS Audio will deliver an extremely obvious audible upgrade to your music system.  Run.’

Jerry Powell writes: 

‘I actually got the Soloist installed the day after I received it, and documented the installation with the attached pictures (you’ll see that I started with a standard outlet and expanded it to accept the new duplex format–piece of cake all the way around). Unfortunately, I then had to leave town for several days and it kind of slipped on me–till I saw Paul’s newsletter the other day, which reminded me…..

I’ve been enjoying the Soloist ever since I got back to town–I’ve got a Duet Power Center plugged into the outlet now so I can connect all of the components of my little office stereo system into a single ‘star’ grounding setup.

As you’ve heard many times before, no doubt, it dropped the noise floor noticeably, which heightens the apparent dynamics, brings out the low-level detail, and so forth–just as one would expect. It may be particularly noticeable in my home office environment, where I’ve got multiple computers running all of the time, including a Qsonix media server, which is essentially just another, specialized, computer. But I also feed the system from a Squeezebox and Internet Radio sources, all of which sound great.

Thank you for the opportunity to try a Soloist–very easy installation and very nice, clean appearance after-the-fact, which is perfect for my office environment. Of course I LOVE my PP Premiers (in my main A/V setup) and Quintessence (tv room), but this is a much better solution for my office.’

Richard Hassler writes:

’1.1    Introduction
Let’s start with a bit of history… A year and a half ago I decided to try to improve the power in my system. I had two Adcom power switchers that also “filtered” the power. They helped with switching but I had come to realize that they did not really help to improve the power seen by my system. I bought two UPC 200′s and they seemed to improve the system performance. The UPC 200′s were feeding only part of my system: the Yamaha Pre/pro/amp, HDTV, DVD and D-VHS player.

Then I bought a Soloist, to help my Vacuum Tube Logic amp see cleaner power. The improvements I heard impressed me; but I needed to have more stable AC going to my amp. It doesn’t have the greatest bias regulation and I hoped that stabilizing the AC would reduce the bias drifts that I was seeing, due to fluctuations in the AC mains. This prompted the purchase of my first Power Plant Premier.

OK, so now I am intrigued. The PPP really changed the sound of my system for the better. It was downstream from the Soloist, feeding the VTL amp and the tuner. Then I replaced the UPC 200′s with a second PPP. It provided another step function improvement of my system.

Given the level of performance that I was experiencing at this point, I was very skeptical that I would see much improvement by merely adding a Soloist in front of the second PPP. It should just lower the noise floor that is seen by the components a bit more, right? You will see that I was pleasantly surprised.

NOTE: The new Soloist is in the chain for the pre/pro/amp, BluRay, D-VHS and HDTV which are all fed by the second PPP.

1.2    Installation
As the accompanying photos show, my system lives inside alcoves in my family room wall. This makes changes of any kind, a real pain. It took a bit of work to access the Hubble outlet that the Soloist would replace. Luckily, I was able to do it without removing my Yamaha integrated amp, because that would have required a rework of many system connections. Breaking and re-making all of those connections would render “before and after” comparisons less meaningful. Ultimately, I removed only a PPP and a shelf, to install the Soloist. It went smoothly except for a phenomenal brain cramp that resulted in cutting the hole in the drywall too wide to fit the double gang box. It mounted firmly but the result looks half-baked. Luckily this outlet is hidden by shelves, curtain, amp, … . The embarrassing part is that I am a surveyor and supposed to be an expert at measurements. So much for any chance at a reputation as a home improvement specialist!

1.3    Music Listening
I installed the Soloist immediately after I picked it up. I have listened to a wide variety of music for the last week. Acoustic jazz, chamber music, hard rock, solo acoustic guitar, pipe organ, … I listened to it all. Using the Denon BluRay player, I listened to CD’s, multichannel music DVD’s (using DD and DTS encoding) and a few BluRay discs containing HD music. Most of the discs that I brought out were old favorites that I have heard through my system hundreds of times in the past. I expected subtle differences from the Soloist at best, but I heard more.

1.3.1    Highs
I didn’t detect further extension to the already extended highs that my system can achieve. I did hear a reduction in grain that significantly improved the reproduction of cymbals, bells, violins and the singing voice. I could hear more of the underlying sound without an excess in sibilance. These improvements let me hear how grainy and unarticulated the highs had been before the change. I heard these changes while listening to CD after CD. The improvements were real and obvious. As you see frequently described in the audio press… cymbals sounded like cymbals instead of like a shaken can of bb’s. The highs just sounded more natural than before.

1.3.2    Upper Midrange
It is difficult for me to describe the improvements that I heard in this critical part of the music spectrum. Simply put, I heard more information here. The harmonic structure that fills out guitar, voice, cymbals and other instruments, was more obvious. I heard of lot of sound where before it seemed a bit threadbare. A beautiful pallet of midrange notes revealed themselves from the mass of sound. My system was exposing music on my CD’s that I had simply never heard before. Acoustic guitar took on a new life through my system. Wow there is a lot of harmonic info that had been missing. Again, the words “more natural” come to mind.

1.3.3    Mid bass -Low bass
The improvements here are less difficult to describe. I could hear kick drum and electric bass significantly better than before. The leading transients were more sharply defined and provided a workout for the woofers (mid and sub) in my Quatros. Plucked acoustic bass sounded more lifelike. You could clearly detect the fingers move across the strings to provide the natural “thump” that you expect to hear. My system seemed to loose a bit of the midrange plangency when I added the Denon last spring. It is back with a vengeance, making the sound so much more full than I have been hearing. I am not talking about one-note, boomy bass here. I am talking about a well-controlled full-bodied foundation that extends smoothly down to the depths that my Quatro subs are able to achieve. There is plenty of definition and overtone present.

1.3.4    Soundstage
My system can throw a wide and moderately deep soundstage. The phase-correct Quatros are great at imaging as well. However, lately I have noticed a bit of clumping of images within the sound field. The Soloist seems to have tamed this malady in my system. The sound stage isn’t wider. I don’t think that it is deeper either although the rear boundaries seem better delineated. The sound field is more smoothly filled with images of the instruments that stand their ground and hang in the air as though there are tangible bodies there. This is probably the most pronounced improvement that the Soloist has wrought. How a power filter can bring about this kind of change blows my mind. The spatial images that a simply miked CD can provide, is startling. The improvements in the highs, which I described above, accentuate the effect. Bells, claps, and cymbal strikes now leave me looking around the room for the source.

1.4    Movie Viewing
I have watched several DVD’s and BluRay discs since I added the Soloist. I can’t say that I have noticed any profound improvements to the viewing experience other than the sound improvements listed above. The Denon player is amazing and the quality of its picture through my Sony CRT can be a “you are there” experience. If anything, perhaps I can see a slight improvement in the video dynamic range.

1.5    TV Viewing
I have watched a fair amount of TV since adding the Soloist. For digital television shows (including high def sports), there are slight improvements to the video dynamic range, as I have reported with the movie discs, but it is subtle at best.

On the other hand, analog TV has jumped up significantly in quality. A vast reduction in grain and noise has made the presentation more viewable. The color saturation has also improved. Unfortunately, the analog sources have a limited life span but at least I can enjoy them while they are still around. TCM was all but unwatchable through my system before the Soloist installation. Now the picture is nearly noise free and the black and white movies on that station show the great grayscale that my system can provide.

1.6    System Description

• Vandersteen Speakers

o    (2) Quatro main speakers
o    VCC-5 center channel
o    (2) VLR-1 surrounds

• Vacuum Tube Logic Stereo 90 amplifier (driving Quatros)
• Denon 3800 BDCI BluRay player (analog outputs only)
• Yamaha DSP-A1 Integrated Pre/Pro/Amp

o    internal amps drive center channel and surround speakers
o    Internal decoders not used for: CD, BluRay, DVD and analog TV
o    Internal decoders used for digital TV

• Sony KD-34XBR960 HDTV
• Magnum Dynalab FT-101 FM tuner
• Audioquest interconnects and video cables

•    PS Audio power cables, Duet, Soloist, and (2) Power Plant Premiers

• Finite Elemente vibration control
• ASC 18″ Tube traps

1.7    Conclusions
I am known as a bit of a complainer. I manage to find fault with most things that come across my plate. Much to my chagrin, I have little to complain about from the addition of the Soloist to my home theater. It has incrementally improved the smallest of features and greatly improved others. Given that I already had the best power filtering that you can get, I expected very little change from the Soloist. I found just the opposite. Improvements were in abundance.

Thanks to PS Audio for this chance to add the Soloist to complete my Power Delivery System. It has opened my eyes one more time, to the importance of clean power.’

Larry Rasmussen writes: 

The offer to install the soloist came just when I have been thinking I should ad some kind of device to protect the items at the right side of my system. The core of my system is on a rack  to outside of my main left speaker in the corner and well served by a Quintet Power Center. However that leaves the video monitor in the middle, a DVD recorder serving as tuner for the Fuji Plasma, one of my newly acquired Channel Islands mono blocks over on the right and a pending added amp for a center channel all without protection. The rest of my system is a Cary Theater 11 Processor as pre amp for music and movies, KEF Reference 203 main speakers, and a JLTi combo DVD player  The area is served by 2 dedicated lines, one for the main rack and one for the video monitor.

I used one of the existing house circuits for the Soloist. Instead of replacing the outlet I just punched a hole in the Sheetrock to the left of it. Measure the height of the box you want to match from the floor, draw a level line along that height, put the outlet box face against the wall and trace the outline. I use a tiny level that I use for speaker leveling on outlet boxes and level at each step. I just did a half dozen or so boxes in the house and it is super fast. Draw box, punch a one inch chisel through on each of the four lines, use double Sheetrock saw in so you don’t have to pull out to turn around. Do each line separately and it’s done in minutes clean, square, level, no ragged corners. I would have thought the  chisel would make a mess but tried it on a whim, really just cuts a slot. Its very easy to tie in to an adjacent box and I really needed the extra outlets.

So I went back and forth a bit between an unconnected house circuit and the soloist with the plasma monitor and digital recorder/channel tuner. Nothing extraordinary leaped out. I’m on Comcast cable in a suburb of Seattle by the way. I did discern cleaner lines along the clothing during the Emmy show, a bit better differentiation or sharpness. I just watched a few shows over the weekend changing back and forth occasionally. Sometimes a bit of wavy interference was reduced with monitor and tuner on the Soloist, sometimes it looked the same.

My 60′s track home exhibits none of the problems you read about in reviews of power equipment but I do appreciate the unobtrusive protection and bit of additional picture or audio clarity the Soloist and then more so the Quintet ad. I would be inclined to recommend the Soloist as a starting point for just about anyone.

Inexpensive, easy to install and your gear is safe whether you are a pretty casual mid fi guy like me or are going to get really serious about conditioning. It’s quality would be consistent with anything you would want to put downstream, gear or conditioning equipment.

A dedicated AC line is simply a separate AC wire going back to your home’s circuit breaker box.  Instead of the standard AC feed with multiple AC outlets available for many pieces of equipment, a dedicated line is a “home run” single use wire from the equipment it powers all the way back to the circuit breaker box where the AC power comes into our homes.

Stereophile editor John Atkinson writes “For less than the cost of a budget power amplifier-a mere $373.45-the electrician ran two new 30A lines to the listening room, one with the hot on one side of neutral, the other on the other. Each had its own circuit breaker and each fed two hospital-grade wall sockets. (These orange receptacles grasp the prongs of AC plugs with a clasp akin to the Vulcan death grip.) All source components and the system preamplifier were plugged into an Inouye AC conditioner, in turn plugged into one of the new lines; power amplifiers were plugged into the other new line.

The sonic effect was nothing short of stunning. Within the context of a power amplifier’s characteristic sound quality, bass fundamentals relatively dropped away to minus infinity, such was the increase in their weight, while the WATT/Puppy’s “hump” in the upper bass became considerably less bothersome. Yes, the characteristic sounds of components were not changed-black was not rendered white-but the differences between those characters was heightened, the overall quality of each enhanced. The sonic contrast knob was turned up a notch, if you will, the blacks becoming a deeper black, the whites becoming more brilliant”

Strong praise indeed for such an inexpensive addition to your home audio or theater room.

If you’re unfamiliar with AC wiring or nervous about doing this yourself, we highly recommend hiring an electrician to do it for you.  The cost will probably be less than $500 and may be worth every penny.  However, if you feel good about tackling this yourself, here are some tips and help.

In its simplest form, we want to run an AC wire from the target outlet in our home AV system area, back to our circuit breaker.  The wire will be attached to an AC receptacle on the equipment side and a newly added circuit breaker on the feed side.  A dedicated line will have its own circuit breaker as well as AC outlet.

Most homeowners, including many DIYers, are intimidated by home wiring and electricity in general. This is understandable. Electrical current is difficult to grasp as a concept. It can’t be seen but it can certainly be felt. Be advised that you will be dealing with potentially lethal voltages so any work you do must be done when the power is completely off.

Here is an overview of what needs to be done with a bit of info at first.

Electricity Distribution Point: The Circuit Breaker Box

Electricity enters the home from the electric company into the circuit breaker box. Older homes may have a fuse box. Sometimes the term load center service box is used. In any event, the power comes in to the main circuit breaker at 220 Volts. Usually, it is 100 Amps.

But most applian ces in the US run on 110 Volts. What gives? Well, the 220 V can be split. So, in our example, two 110V 20A circuits are the same as one 220V 20A circuit. In reality, in almost all homes, the electrical power is broken down at the circuit breaker box into a combination of 110V and 220V circuit breakers.

The purpose of the circuit breaker (or fuse) is to break the circuit when too high an amperage demand is put on the circuit. This should only happen if too many high demand appliances on a circuit are running at the same time.

So the power has been delivered to the home and distributed to circuit breakers at the panel or fuse box. What next? The juice has to be delivered to the our receptacle where it will do its work and make life good. This, obviously, is done via wiring.

Just any wiring? No. Before running any new wiring, check with the local building codes to know what to use. In most cases it will be specified as 12 or 14 gauge nonmetallic sheathed cable, known simply as Romex. Inside this cable are plastic coated copper wires which are color coded. For a few years, aluminum wire was used, but like lead-based paint, it was found to be dangerous and was discontinued.

We would always recommend going to the heaviest gauge copper Romex you can use, never less than 12 gauge and typically 10 gauge (the lower the gauge number the thicker the wire conductors).

We would recommend that you install a PS Audio Power Port or Power Port Premier AC receptacle at the end of the dedicated AC line.  This is a superior power delivery receptacle which can accept up to a 10 gauge wire for best performance.

The most difficult task 
Running the wire from the circuit breaker box to the receptacle.  Sometimes you get lucky and can run the Romex under the house or through the attic, thus making it easier than having to place it inside the walls on an existing home.  Once you manage to run the wire from the circuit breaker to the intended AC outlet, which should be mounted in an outlet boxes screwed or nailed to the studs, the worst is over.

Running Wire in a Permanently Unfinished Space   
Before you proceed, be certain you won’t change your mind later and decide to finish the space, because you’ll have to redo the wiring. This method requires considerably more wire than what you’d use on a space that will be finished because you’re essentially following the path of the wood framing up, down and across. This keeps the wire from spanning open spaces, where it could be snagged and pulled by yard tools or sports equipment in the garage, for example.

What You’ll Need 

* Both municipal and national codes require insulated wire, and most commonly used is 12- or 14-gauge nonmetallic sheathed cable known as Romex.  We recommend 10 gauge.
* Plastic cable straps, also known as plastic staples.
* Hammer

How to Run the Wire 
1. Plan your wiring layout. First, decide where you want your new electrical box that will house the Power Port AC receptacle. Attach the box, whether metal or plastic, to the studs with nails.
2. Run the wire through the box and out the side hole in the direction of the breaker panel. Leave enough wire to reach the breaker box, plus at least another foot, for later connection.
3. Secure the cable to the very center of the wider face of the studs within 12 inches of each box and at least every 4 feet with plastic cable straps, also called plastic staples. Be extremely careful not to nail through the wire itself. Wherever the cable doesn’t snug up to the wood, secure it with an additional strap. All wires must be at minimum 1-1/4 inches from the front and back edges of all studs and joists.
4. At the top of the stud, route the wire up to the header (the piece of wood that runs across the top of the studs) and across its face.
5. To move horizontally, bring the wire up to the face of the header atop the stud, then staple it as you carry it across until you reach the stud that bears the box you want. Choose the shortest route possible, of course, running the wire against wood. Keep the wire as protected as possible – and as visible and accessible as possible.
6. Continue stapling the wire until you reach the box.
7. Snake the wire into the box and out through the front about 10 to 12 inches, then double back , leaving about 20 to 24 inches total, and run the wire out of the box toward the next one, keeping the wire securely stapled to the wood.
8. To cross the ceiling, travel with the joists, not across them, securing to the wider, vertical side of the joist. You don’t want wire without wood to support it.

Running Wire in a Space to be Finished 

It takes less wire and less effort to run wire on a wall that’s going to be finished. The major difference in this case is that it’s perfectly fine to span the space between the studs with the wire, since it will be enclosed and won’t be in danger of being snagged or pulled.

To run wire across the studs: 

1. With a 1-inch spade bit mounted on a right-angle power drill or standard drill with a right-angle attachment, bore a hole through the wide face of the stud. The hole needs to be at least 1-1/4 inches from the front edge of the stud to meet code requirements and to prevent accidental contact when the drywall goes up. There are no rules regarding how high you place the hole and wiring – the best route is the one that leads directly to the box.
2. Following the steps above, install the electrical box. Instead of following the perimeter of the framing, just run the wire through each hole, spanning the space between the studs.
3. If you’re going to insulate the wall before installing drywall, be sure to leave enough slack in the wiring between the studs so there’s no tension when the insulation is put in. Insulation is commonly sliced so the wiring is encased in it, but check with your insulation’s manufacturer for their recommendations regarding installation around the wiring.

Tips 

* It’s imperative that you contact your municipality and ask for its specific codes regarding electrical wiring. There may be differences from national code, and you don’t want unpleasant surprises later.
* Find out if permits are required.
* Metal nail guard plates can be placed over the edges of studs to protect the drilled hole and wiring inside it.

Installing the AC receptacle

Installing the AC receptacle is quite easy.  Most receptacles are clearly marked as to hot (black wire) nuetral (white wire) and ground (bare copper wire.

Connect the Romex you have installed to the AC receptacle, install it in the wall and place the protective cover for it.

Recommendations 
We recommend you use a 30 amp breaker for even the lowest draw source equipment feed.

We recommend you use the heaviest gauge Romex wiring for the dedicated line, preferably 10 gauge.

We recommend you use a PS Audio Power Port or Power Port Premier AC receptacle to feed the power from your new dedicated line to your equipment.  It’s a bit of work to install a dedicated line, so make sure you take full advantage of the improved connectivity available with a PS Power Port.

Electrical Do’s and Don’ts 
DO always turn off the breaker before working on anything, such as changing a ceiling fan, replacing a light switch or installing a home theater.  We recommend being ultra safe and turning off the main breaker for the entire home.  Always test to see if there is no AC power by using a meter.  Remember,  AC can kill you.

DO test wires with a meter.

DON’T work off a metal ladder.

Very few audio or video systems are dead quiet. There are usually always a few hum related problems. If your system has a bit of hum , is it the transformer or not? How do you determine the source of hum and what can you do about it?

Sometimes hums and buzzes are quite obvious, sometimes not. The ‘hum noise’ usually comes in two flavors, a low non-irritating drone (50 or 60 Hz) or a slightly higher pitched buzz or raspy/irritating ‘angry insect’ sound (100 or 120 Hz).  Video hum is usually seen as diagonal bars across the TV or screen of a projector.

Every audio/video system has some degree of either audible or visual hum or buzz. If your system has some of these noises, and they are at a level that is noticeable or bothersome to you, there are a few things you can do to fix these problems.

Find out what’s making the noise

We first need to divide our search into two categories; electrically or mechanically induced hum.

To see if it is an electrical problem, make sure your system has been on and warmed up for at least 10 minutes, then simply place your ear near the loudspeaker (with no music playing) and listen to determine if the hum or buzz is coming from your speaker. If it is, then at least one component of your problem is electrical.

A mechanically induced hum or buzz is equally easy to determine. Place your ear very near to each piece of your electrical equipment and again, listen for hum and buzz. If you hear a hum emanating from within your equipment, we would refer to this as mechanically induced noise (as opposed to an electrically induced noise).

Mechanical hum

If it’s mechanically induced hum/buzz, what can be done?  Plenty, and PS offers two off-the-shelf solutions: the Power Plant AC regenerator and the Humbuster AC .

Mechanically induced hum is caused, almost entirely, by the transformer and is nearly impossible to get rid of (unless you own the PS Audio Power Plant or Humbuster AC). If you suffer from this noise problem, you’ve probably also noticed that it’s intensity varies depending on the time of day, sometimes even the time of month. The reason it varies is due, in large part, to the quality of the AC line voltage and how much DC is on it.

Why do transformers hum?

We could use the tired saying ‘because they don’t know the words,’ but that might get us sidetracked.

The short and simple answer is that transformers hum because of an effect known as ‘lamination rattle’ caused by DC voltage on the line or poor construction or both. ‘Lam’ rattle occurs in all transformers to some degree, that degree being related to the quality of the transformer and the quality of the line voltage.

Does my equipment have a transformer?

Most likely, yes. With but few exceptions, all powered electronics have a power transformer that is driving them.

Why does it have a transformer and what the heck does a transformer do?

Transformers serve two main functions: isolation and voltage change.

Isolation is essential for your safety. You certainly don’t want the metal parts of your equipment to be connected directly up to the wall socket do you?  Some of us may remember as a small child the effects of sticking a butter knife or similar types of objects into the wall socket. But connect we must if we’re to draw power from the utility company.

A transformer connects the equipment to the power but there’s no physical connection between the two, they are isolated via a magnetic field which passes the energy.  The isolation provided by the magnetic field helps keep us safe.

Voltage change is another great feature of the transformer. Most solid state equipment runs on relatively low voltages that range from 5 volts to 30 volts. What comes out of the wall socket is 120 volts (or 220 volts depending on your country) so the transformer reduces the voltage out of the wall to something manageable for our equipment.

How does a transformer work?

Transformers work with AC power. They will not work with DC power (like what comes out of a battery).  This is due to the fact that transformers work on magnetic fields and the magnetic fields need to change their polarity (North to South and back again) to make the transformer work properly.

What comes out of your home’s wall socket is AC power. AC stands for Alternating Current, which simply means that power alternates between plus and minus voltage, unlike power delivered from a battery which has a fixed plus and minus voltage.  Batteries deliver DC which stands for Direct Current.

Complicated as all that may sound, it is really quite simple. Think of AC power this way; 50 or 60 times a second (depending on what country you live in) the power in your home goes positive, then negative – like a battery that flips over once every 60th of a second.  This polarity change is needed to operate motors and transformers.

What’s inside a transformer?

They’re really quite simple devices, consisting of two main elements; wire and laminations.

In a typical power transformer, there are two coils of wires wrapped around pieces of iron.  The coils are in close physical contact with each other but are electrically isolated.  There’s an input coil that connects to the homes AC receptacle, and an output coil that feeds your equipment. These coil mce_style=”float: left;” mce_src=”http://www.psaudio.com/articles/images/181_15.jpg” border=0 v:shapes=”_x0000_s1027″> s are nothing more than a length of copper wire wound in a circle and when you apply power to them they become magnets.

The pieces of iron are called laminations.  The laminations are basically small sheets of steel or iron, stacked one on top of the other. They are used to focus the magnetic field, created by the coils of wire.

So, why do transformers hum?

As mentioned earlier, transformers hum because of lamination rattle caused by either DC voltage on the AC line or poor construction or both.

Remember when we mentioned that this humming problem was due to either power line conditions or the quality of the transformer? Here’s why: when there’s DC on the line, we have an asymmetrical field which causes greater vibrations.  The laminations are ‘pushed’ together in one direction because of the DC.

Because transformers work when the coil magnets switch poles from North to South and back again, 60 times a second, DC forces the input coil to always sit in one direction and this makes the transformer a little crazy so it hums.

To reduce these noises, transformer manufacturer have several tricks up their sleeves: they can varnish, or use super glue to stick the laminations together so they rattle less, and they can make bigger transformers that don’t have to work so hard, even in the presence of DC. The harder a transformer has to work, the more stress and strain is placed on the laminations.

But these measures don’t entirely solve the problem because you need to do that at the source of the problem, the DC on the line.

Why this is bad

When a transformer hums, it is actually physically vibrating or shaking inside of the chassis. This, in turn, shakes and vibrates everything else inside the chassis. Many components in the chassis are sensitive to vibrations, including tubes, semiconductors and capacitors. In an even moderate case, this vibration can effect sound and picture quality as many of these internal components are microphonic and reproduce the humming into the audio or video signal.

The magnetic fields produced by transformers create electrical energy in more than just the transformer. They can easily generate an electrical current in a capacitor, for instance, which is essentially a coil of ‘wire’ itself. It can even generate an unwanted electrical current in any wiring or PC board traces and is why transformers have to be strategically placed away from all other components.

So, what can we do?

We have several options: power the system with a Power Plant AC regenerator and eliminate the hum, or purchase and connect a PS Audio Humbuster to eliminate it as well.

We can buy equipment with better transformers.  PS and many high-end audio video manufacturers go to great lengths to design and build near perfect transformers that have very low levels of mechanical hum even under the worst conditions coming out of the wall.  But regardless of how well the transformer’s made, we can’t do better than fixing the problem properly in the first place.  That’s where the Humbuster AC or the Power Plants come into play: they fix the problem at the source.

It’s best to fix the problem at the source

Regardless of how well a transformer is made it’s best to keep the DC out of it.  Transformers with DC on them have core saturation problems, some amount of mechanical noise and lowered efficiency.

PS offers several solutions for lowering hum, eliminating DC and improving performance.  The Power Plant series of AC regenerators and the Humbuster AC device .  Do your system a favor and eliminate its AC problems.

What about electrical noises?

Electrical noises are usually caused by one of two main problems: proximity or ground loops.  These hums can be iden tified by listening to the speaker for a low humming sound (as opposed to a buzz) or in video can be seen as a distortion of a TV tube or diagonal bars across the set.

Proximity refers to how close one piece of equipment is to another.  Since transformers work by generating magnetic fields, these fields can be rather large and if the field gets too close to another audio or video product, noise (hum) can be induced into the product from the transformer.  This type of sensitivity is typically restricted to high gain pieces of equipment like phono stages, but even preamplifiers sitting in close proximity to a power amplifier can have hum induced into it.

Solving proximity problems is relatively easy: simply move the equipment further apart.

Ground loops hums are perhaps the most difficult to track down.  Ground loops are a result of differing ground potentials.  This means that the ground of one AC source or equipment source is at a different level than the ground of another AC source or equipment.  This difference is usually amplified in the form of audible or visible hum.  Visible hum is usually seen as diagonal bars across the video screen.

Tracking these types of hums down is more difficult and below we have assembled some helpful tips.  At the end of the day it may make more sense to speak with your dealer for help.

Tracking down ground loop problems

The easiest way to figure out where ground loop problems lie is by the process of elimination. You need to determine where the hum or buzz is coming from within your system.  If it’s a video hum problem, use a known good source like a DVD player rather than cable or satellite.  In video, it’s best to always assume that it’s either a connection problem or, more likely, a cable problem.  Our experience has shown that poorly shielded video cables cause more hum problems than just about anything else.

In an audio situation, the first suspect in our hunt would be the power amp or the receiver that is driving the loudspeaker. To see if the power amp or the receiver is the culprit, turn them off, disconnect its inputs and turn it back on again. Go back to the speaker and place your ear in close proximity to see if the hum is still there. If it is, then you have a problem with your power amp or receiver and you should seek help from its manufacturer.

1. If the hum/buzz goes away when you remove the inputs to the power amp, your next step will be to reconnect the amp and move further down the chain. If you were working with a receiver or an integrated amplifier, you will need to jump to step 4. If you have a preamp, or processor that is feeding the power amp, your next step would be to disconnect all inputs to the preamplifier or processor. Once these are disconnected, and the preamp or processor is connected only to the power amplifier, turn the system on and again, listen for hum. Should the hum now appear, it is a problem with your preamp or processor or their interaction with the power amp. Before returning the preamp or processor to the manufacturer, try a cheater plug to break a ground loop. Cheater plugs are simple devices that convert a three prong AC plug into a two prong AC plug and in the act of converting three prongs, to two prongs, they disconnect the ground from the wall socket. Try one of these on the preamp, or the power amp, or both.
2. If you determine that there is still no hum present when the preamp, processor or receiver is connected with no inputs, then selectively begin plugging in your various inputs one at a time. After each connection, check for hum until you discover the humming culprit.
3. VCR’s, surround processors, and any device that is connected to a television cable or satellite dish can cause a loud buzz and should always be suspect. If, by the process of elimination described above, you determine it is a component like a VCR that is causing the hum/buzz to occur, and using a cheater plug or removing the ground pin on a PS xStream Power Cable doesn’t help matters, it may be necessary to isolate the cable connection (CATV) with an isolation transformer. This inexpensive device is available at most Wal Mart, Radio Shack or department store type outlets and is sometimes called a ‘matching transformer’. If you have problems finding one, call your local cable TV company for advice. The matching transformer will be placed between the cable TV cord and the VCR, TV or processor.

Just remember, take the system down to its simplest level of connection.  Find a way to hook the system up with as many pieces of the system missing or not connected.  Keep it simple and get it to the point where the hum’s gone.  Then start adding back components one at a time until the hum returns.

Finding the problem is 9/10th of the work in finding a solution.

There aren’t many inexpensive upgrades you can add that are as easy to install and effective as a Power Port .

Everything in our AV systems needs power to operate.

The quality of this power is directly related to the performance of our equipment.

Getting the power out of the home’s AC receptacle is the first place we should concern ourselves with proper connectivity.

Most homes have a $0.99 “contractor’s special” installed in the wall socket.  While these are perhaps adequate for a toaster, a lamp or a bathroom utility, they are hardly appropriate for a high-end stereo or video system.

Typical contact materials found in 99% of all home electrical systems are brass which can and does lose connectivity over just a few months.

Power Ports replace these sub-standard power connections and can contribute greatly to the performance and long term reliability of your AV system.

Customers who have installed Power Ports report nearly unanimous praise for their results.

‘For me the Power Port’s $49.95 x 3 is the best money I have ever spent for audio or video gear. Bar none. ‘

Why risk poor connectivity?

‘I received the Power Port I ordered from PS Audio on Friday. I replaced my existing receptacle and plugged in my ATI 2505 power amp. The amp is on a dedicated 20-amp line. I powered up the system and turned on the TV as a test. My wife happened to be walking through the living room at the same time. She stopped and asked me what I had done to the sound on the TV. She was amazed. ‘It’s sounds clearer, like you took a blanket away from the speakers. ‘

We would recommend replacing every AC receptacle in your home that has any form of audio/video equipment connected to it.

The list would include power amplifiers, preamplifiers, digital source equipment, televisions, RPTV’s, projectors, receivers.  In short, any piece of home AV equipment you care about should be connected through a Power Port.

Installing a Power Port couldn’t be easier

We estimate the average time to install a Power Port is approximately 15 minutes or less.

We replaced 5 sets of duplex sockets in the PS Listening room on a Saturday morning in about one hour.

After the Power Ports were installed we reconnected the equipment, let it warm up for another hour and were simply stunned at the level of improvement.

Step one

First things first.  Make sure the power is off to the receptacle you want to replace.  This is really easy to do.

Perhaps the quickest method is to plug a lamp into the receptacle and make sure the lamp is illuminated.

Have a friend watch the lamp while you go and find the circuit breaker that is powering the receptacle.
As soon as the lamp goes out, have the friend make sure it does not power the lamp in either the top or bottom plug.

Step two

Remove the cover plate.

This is achieved by simply removing the small screw in the middle of the plate and pulling the cover off, thus exposing the old receptacle.

As an added precaution, if you have one of the AC noise sniffers we recommended in the PS Tips section , place this inside the receptacle area just to make sure the power is completely off.  These AC noise sniffers are readily available at any hardware store.

Now, remove the top and bottom screws of the receptacle to move it from the wall.

Step three

Remove the wires from the old receptacle.

Using a Phillips screwdriver, remove the white wires from the old receptacle, then the black wires, and then the ground wires.  Actually, you can do this in any order you wish just as long as you keep track of the wire colors.  Black is hot, white is nuetral, and green should be ground.

Step four

Remove the Power Port from its package and attach the wires to the Power Port.

The Power Port is marked “white” and “black” so you know exactly where to place the wires and keep the polarity correct and safe.  Look closely at the front of the Power Port and you’ll notice the nomenclature that tells you.

Simply attach the wires in the proper place, screw them down tightly.

The ground wire goes on the green screw on the side of the Power Port.

Step five

Place the Power Port back into the wall cavity.

Position the Power Port so it lines up with the female threads of the wall box and use the two included screws to re-attach it to the electrical box.

Basically just reverse the procedure used to remove the original receptacle.

Once the screws are in place, line up the Power Port so it is straight.

Step six

Replace the cover plate and you are done!

Now, continue replacing each receptacle associated with any AV equipment in the same way and you will be guaranteed a perfect connection every time you use your equipment.

That’s about it!

As we said, it doesn’t get a lot easier when it comes to making a serious improvement to your AV system.

Power Ports are affordable, they provide an excellent long-term level of connectivity to your system and are fully UL approved.

If you have any questions our customer service people stand ready to help.

Why depend on an aging and corroding brass receptacle to power your equipment?

We realize that there will be those skeptics in the audience that don’t really believe this will make much of an improvement.  Certainly that is one of the reasons we offer a full money back promise on the Power Port.

Perhaps it is time we spend as much care with our AC power connections as we do with our AV signal connections.  It isn’t hard to imagine the need for a quality interconnect cable and equally it should not be difficult to imagine the benefits of properly connecting your equipment to its source of power.

If you are interested in outfitting your listening room or AV room with a great receptacle, we would encourage you to consider the Power Port.

Would you like to try one in your home?

Ordering a Power Port is easy.

First, determine how many receptacles in your home have items like a television, a computer, surround sound system, audio or video sources or AV equipment of any kind attached to it.

Second, click here and order the number of Power Ports you are interested in.

Again, we suggest one Power Port replacing each of the duplex sockets in your listening room, extension strip, AV room or anywhere you are concerned about the quality of the connectivity of your equipment.

Once ordered, the Power Ports will arrive in a few days and you are good to go.

Simply follow the simple instructions on the back of the Power Port package, download the same instructions here , or follow the easy step by step instructions on this page.

It is truly one of the easiest and quickest upgrades you can make to your system and it’s an upgrade that will last the lifetime of your home.

Don’t compromise the qualtiy of your power.

Get it right in the first place with a Power Port!

As you will no doubt read in many areas of this web site, the quality of the AC power coming out of the wall is important to your system’s performance. If you can’t get it out of the wall properly, you are handicapped from the very beginning of the chain.

One key factor in this chain is the AC power cord itself. Many pieces of stereo/video equipment have detachable AC cords. Most AC cords that are supplied to you by the manufacturer are woefully inadequate. Since the AC power cord is the first link in the system, it is very important that you have a proper cord.

A relatively expensive solution is to purchase a high end “Audiophile” cord. We sell our low cost entry PC, the Punch, for $49.99. While this may sound very expensive for a simple AC power cord, comparing the performance between this power cable and your stock cable can be a rather big shock.

One simple, inexpensive solution (if you are hesitant to purchase a high-end power cable) is to buy a thicker cord.  Most manufacturer supplied power cords are 18 gauge thick. A better gauge to have would be 16, or 14, or even 12 gauge. The lower the number of the gauge, the heavier the wire.

If you’d like to purchase an inexpensive heavier gauge detachable cord, you can do so over the web.

http://www.alliedelec.com/catalog/indices/indices.asp

Look under Belden and acquire the shielded version.

We would certainly recommend the PS Audio solution first.

The PowerPlay™ web accessible power conditioners, as well as the PowerPack™ UPS , use ultra modern Cloud Computing to manage, control and communicate with every PowerPlay unit in the field.

Cloud Computing is fundamentally different than the way all other web and network accessible power conditioners work.

A tradional network accessible power conditioner, like those from Panamax™, APC™ and Richard Gray™, rely on a built in web page for each power conditioner in the field.  All the control, features and functions available to the power conditioner or UPS are embedded into the unit’s inernal web page.

To communicate and control the power conditioner or UPS in this classic scheme, a fixed or static IP address is required and the router must be configured to open the correct port to access the internal web page in the power conditioner or UPS.

To communicate, the installer or dealer must enter in the specific IP address of the equipment.  This creates a direct link to the unit’s internal web page, through the assigned port, which appears on the client’s browser and the unit can be controlled.  In some cases, it is necessary to install additional control software on the accessing client for this system to work.

PS Audio’s PowerPlay line of web accessible power conditioners relies on a far superior system known as Cloud Computing.

In this system, an internet connected remote server (PS GlobalNet™), operated by PS Audio, has all the software and communication access located in the central server.  The power conditioners are controlled by Global Net rather than their internal web pages.  Access to the power conditioners does not require a fixed or static IP address and can be controlled from any web browser anywhere in the world.

The advantages of GlobalNet for this system are many:

• No static or fixed IP required
• Location of the power conditioner can be changed at any time without notification
• No software required
• Unlimited control levels are achieved due to the power of the server
• Not limited to an internal web page
• Updates to both equipment and the Cloud happen instantly
• Power quality data accessible on the Cloud for years

Similar yes, but ceratinly not the same.  The network accessible power conditioners from Panamax™, APC™ and Richard Gray™ differ from the PS Audio PowerPlay ™ series in several major ways.

Network setup.   All three units from these other manufacturers require a static IP to access them over the web.  A static IP is a fixed and unique web address that must be preordered, maintained and paid for on a monthly basis over the life of the service.  Once installed and activated, installers must also setup these devices with a specific port setting and other network complications.  Should the client move, this must be redone from the beginning.

Web access and control.   Once these units are setup, the dealer or installer must record the IP address and maintain a record of the address in order to access and control the power conditioner.  If the installer is on the road and receives a request for service, it can be very difficult (impossible actually without the IP address)  to access the customer’s power conditioner from the field.

Limited control features.   These three units have limited control features that allow simple reboot and control over zones, input naming (in some cases) and minimal performance readings.

PowerPlay solves the problem.   The PowerPlay units are entirely different.  PowerPlay requires no static IP , no port setting, no network configuration whatsoever. Instead, PowerPlay utilizes GlobalNet™ Cloud Computingto solve this problem.

Any internet accessible connection will be perfect for PowerPlay.  Just plug it in, register the unit, and you are done.  Everything is completely transparent to the installer and the user.

The PowerPLay units are registered under the customer’s name and address and control of their unit is accessed easily by name over any web enabled browser.

Installers in the field can respond to a customer request for service using their iPhone, Blackberry or any web enables public terminal.  The only information required is the customer’s name and the installers login password.

Once connected, PowerPlay offers a wealth of user interface options including all the basic reboot and zone management features along with IR commands, full measurement and records of power line events by date, intelligent UPS control by priority, auto reboot ping feature, green scheduling of equipment shutdown and power up, email and text alerts.

PowerPlay is a full featured, easy to connect system that requires no network setup whatsoever.

One major advantage of the Soloist in-wall power conditioner is the lack of a power cord.  All power conditioners, regardless of their design, require an extra power cord to connect them to the AC power source.

In many cases, this power cord becomes a limiting factor in a high end AV system unless you choose one that is up to the task; which can be expensive.

By placing the Soloist in the wall, you start to build a solid AC power foundation for your system and place the protection and the first stage cleaning where it belongs; in the wall.

If you are using the Soloist in partnership with a television of any type, especially on-wall mounted plasma or LCD or ceiling mounted projectors, the benefits are enormous.  Protection from surges and spikes, achieving a better picture through AC cleaning and all without the degradation and clutter of an additional power cable.

Lower noise to put it simply.  The NH is a parallel device which means it is not in the path of the AC power.  Rather, a NH works across the line, drawing energy only at around 10kHz and above where the majority of AC line noise from dimmers, cell phones and appliances live.  This noise power the lightn inside the NH and by doing so, the noise amplitude (amount) is reduced by the number of parallel Noise Harvesters at any one point.

Two Harvesters reduces the line noise in half, four Harvesters in half again, etc.  If you have serious line noise you wish to eliminate for ever, or even bothersome hum you can’t eliminate, try more than one Harvester to fix the problem.

Using a Juice bar II extender, you can add up to 8 Harvesters at any one location for significant noise reduction.

Not in the least.  In fact, depending on the power cable you use to feed the Juice Bar II, you can consider this high end plug extender as having zero impact on either power delivery or connected equipment.

The way the Juice Bar II is built should say it all.  The actual input AC prong is machined from the very same 1/4 inch copper bar that feeds every outlet.  There are no wires, no messy joints to restrict the power along the power bus bar delivery system inside the Juice Bar II.  This product far exceeds any plug extender we have ever made.

The only cautionary note is the input power cable to the Juice Bar.  This is the one place where you can restrict the JB’s input power and great care should be taken with respect to length and quality if you expect to maintain zero impact on connected equipment.

“Component Level performance” is a term we use to make the point that our power conditioning units like the Quintet and the Duet perform as well as a full chassis power conditioner.  One of the problems we have getting people to understand just how powerful the Quintet and Duet are is their form factor; which is both a benefit and a problem.  The benefit is a convenient and easy to place piece of equipment, the problem is people have trouble believing they are just as powerful (if not more) than a larger full size chassis model.

The fact is that despite the big chassis, the insides and functionality of these two power centers is every bit as robust as any of their competitors.  Winners of the Absolute Sound’s product of the year award, both the Quintet and Duet embody the idea of “good things come in small packages”.

For years we have judged products by their size and weight.  With today’s compact technology such as the Nano Crystalline filters used throughout our award winning power product line, we’re able to deliver far more than ever before; in convenient packages.

The Duet, Quintet and Juice Bar II extender were designed specifically to go behind equipment, rack shelves and equipment shelves.  The Duet, Quintet and Juice Bar deliver true “Component level performance” and will stand up to any full size passive power conditioner or extender strip in the world.

Yes.  Both the Duet and Quintet Power Centers come with a reasonably nice stock cord as do all of our products so you can fire them up right away.

We would encourage you to upgrade as soon as you can to a better cable however.  While we fully understand that some cables cost as much as the Duet and Quitet themselves, this is more of a testament to the great value that these two award winning power centers are than anything else.  The power cable is a key element in connecting equipment us.  While we would love to include a high end power cable with every power center, we cannot maintain their high value by doing so.

It is absolutely worth the extra bank to get the best performance you can by getting a great aftermarket power cable connecting your system components together.

Power cables are the connections between the AC power source and your equipment.  Think of these in the same way you might envision a water pipe.  If you have a restricted flow water pipe, you will not be able to deliver as much water as may be required.  The same is true for AC power.  Great care must be taken to make sure the power delivery system to each piece of equipment in your setup is unrestricted at a minimum.

How big a problem is this?  Most stock power cables have relatively small wire gauges and can restrict power demands of equipment, in particular instaneous dynamic demands that exceed the ability of the stock power cord to allow the needed power’s passge, especially to power amplifiers and video projection devices.

The quality of the connectors on power cables is yet another big factor in delivering power properly.  Most connectors on off-the-shelf stock power cables are made from stamped brass.  Brass has a tendency to “barnacle” or build up small protrusions that limit the conducting area of a plug’s connector thus limiting current flow.  PS Audio (and other high end manufacturer’s products) take extraordinary care to use polishing and plating of non intrusive materials (such as gold, silver, nickel, rhodium) to their connectors in an effort to improve connectivity.  Stay away from any products with unpolished raw brass connectors.

Shielding.  Most power cables are either not shielded or are poorly shielded.  Shielding is important both from incoming noise and from outgoing noise.  Most equipment produces EMI on its own and without a shielded power cable these emissions radiate to other equipment.

Quality of the copper inside the cable.  Most stock power cords use the lowest possible copper quality to reduce costs.  Use of PCOCC and OFC copper can enhance current flow and improve sound and video.

Active cleaning.  Properly engineered, ferrite (an iron based material) can reduce noise levels in the AC powerline by small amounts.  PS xStream cables are the only power cables in the world with impregnated active ferrite noise reduction materials in their jacket.
Some power cables use a lump or ‘slug’ of ferrite to reduce noise.  While this is effective for noise reduction we have found it restricts the sound when not distributed.

There are three types of categories for AC power cleaning and connection devices: passive, passive transformer, and active. 

Passive power conditioning is, by far, the largest category of power conditioning available.  In a passive power conditioner (of which there are two types: series or parallel) passive components are used (capacitors, resistors and coils of wire) to act as filters to reduce unwanted noises on the AC line.  The most effective passive power conditioners are series devices, the least effective are parallel devices.

All parallel devices have one major drawback in their performance: they cannot add meaningful levels of energy back onto the line.  Passive power conditioners can only remove unwanted artifacts on the power line.  While the system’s performance can benefit from the removal of these unwanted artifacts, passive conditioners cannot address the major problems found on the AC line such as voltage regulation, clipped sine wave, dynamic power loss.  These major power problems are shared by all AC power consumers and are only addressed with an active power device.

The second most common AC power conditioner is the passive isolation transformer.  These are generally large AC power transformer connected in series with the AC line.  Line isolation transformers have no physical connection to the AC power line.  Instead, these devices couple the AC power through a magnetic field, thus providing 100% physical isolation.  Their benefits are decreased potential for hum, decreased interaction from one piece of equipment to another.  Their downsides are many, including large size and weight, restricted power delivery abilit (depending on their size), increased waveform clipping and distortion under load.

Active power conditioning properly executed is the best solution of all three methods.  Active devices like PS Audio Power Plant and Exact Power’s products, are essentially large power amplifiers delivering the AC power to equipment.  These products have many benefits including instantaneous voltage regulation, dynamic energy replacement, elimination of waveform clipping and so forth.  Their downsides are higher costs to build, heat, power restrictions (depending on the size of the active device).  Consumers should be aware that not all active power devices are the same.  Some, like the APS regenerator, are based on older load restricted double conversion UPS products that can actually perform with higher distortion levels under dynamic loads than the wall itself.

Yes.  In fact a parallel powe r conditioner like those manufactured by Shunyata, Richard Grey, Transparent and Audience (among others) have only the added power cable feeding the power conditioner as a restriction.  The power cleaning circuitry itself does not restrict the AC power at all.

There are, however, little evidence that these parallel devices provide any measurable AC cleaning benefits although many customers find they do add a perceived improvement to the system they are powering.  There is a great deal of debate ton this subject with some who would suggest the added power cable itself may be a major contributor to the change in perceived performance.

A series power conditioner like those from Monster Cable, Panamax, Fuhrman, PS Audio and others have easily measured results in AC power cleaning.  The downside to a series power conditioner is that there will always be some level of restriction to the AC power by the very nature of the series design itself.  Great care and engineering skill must be employed to have minimal dynamic restriction and maximum measured benefit in this type of conditioner.

PS Audio power products have both measurable and published test results showing their performance and demonstrate over 60dB of AC cleaning for passive devices and 80dB for our active regenerators (Power Plants).

PS series devices, such as ther Quintet, Duet and Quintessence (among others) are engineered with wire gauges heavy enough that they bordering on overkill, coupled with solid copper bus bars etc. in a strong engineering effort to reduce possible dynamic restriction as much as humanely possible while still being extremely effective in reducing AC noise.

Power Plant AC regenerators are both effective as providing improved dynamics over even the wall AC power itself.  This is due to the onboard energy storage capabilities of an active regenerator.

Many customers want to understand why a properly designed high-performance aftermarket power cord makes any difference when that very power cord is attached to your home’s internal wiring that is perhaps 100 feet in length and acting like a superb antenna for noise!

There are several reasons.  The first is best understood by rethinking the power cable’s role in the system.  Most of us think that a power cable is the last length of cable at the end of the power chain.  In fact, from the point of view of the equipment, it is actually the first piece in the chain.  When our equipment receives AC power it is typically through a power transformer.  The first thing a power transformer “sees” is the connecting AC power cable and its characteristic electrical “signature” (inductance, capacitance, shielding) affects the interaction between the transformer and the circuit feeding it power.

Secondly, noise and its source.  While the  wiring in your home is subject to lots of noise from sources like cell phones, computers, terrestrial broadcasting sources and so on, the single beiggest noise source can often be your AV system itself.  Every piece of equipment in your system radiates RF in the air as well as places switching noise from your equipment’s power supplies onto the line.

The walls in your home pickup noise, but the strength of that noise is quite low relative to one of the single biggest sources of radiated noise in your home; your AV system.  Protecting your power from radiation with high quality, well shielded aftermarket power cables, helps reduce noise at the source of the most noise; the AV equipment itself.

Secrets of a coil of wire and a magnet

httpv://youtu.be/OeA3MRtNaX4

Absolutely.  While high end power cables like the AC Series from PS handle the most exposed last 6 feet of the run, where the noise levels are generated from the equipment itself and are the highest, in-wall is also important.  Consider, however, that rewiring your home with expensive power cable is an expensive and tough option.  One of the less expensive options available to you is to run 10 gauge power cables and place them in grounded metal conduit.  You will need an electrician for this unless you’re very handy, but this is the best sounding option we are aware of, keeping noise low and voltage drops low as well.

This fixes the problem where the unit prematurely reboots itself and doesn’t display wattage properly.

Online update

The online update process is the easiest if you have internet access connected to your new Power Plant.  You must register the unit before you are able to login to your command center.

To update, simply go to your command page through Globalnet, http://perfectwave.psaudio.com/power/ , click on Command Center and you’ll see “Firmware Update”.  Click on this and then go have a beer.  It might take a while.  Under no circumstance should you remove the power.  If you have a problem, just let it go and call us before doing anything rash.  It should go very smoothly.

If you don’t have your Power Plant connected to the internet, you’ll then need to update via the SD card we supplied with the unit.  This is also what you’ll do if the internet update gets corrupted for any reason.

SD Card update

At your computer, please use the following link to unzip and place the unzipped files on the SD card that came with your P5/P10.

http://updates.psaudio.com/perfectwave/11-056-01-X-FMC-13.zip
To prepare the SD card we suggest doing a quick format prior to loading the new firmware (erase all the old files) then drag and drop the files from the above link onto the SD card.

Once the SD card contains the new firmware you will need to power off the P5 (or P10) with the toggle switch in the back, insert the SD card into the Power Plant, then return power to the unit. The update may take several minutes, but should take no longer than 10 minutes.

Once it’s completed you should be good to go.

Certainly.  This information applies to problems encountered when upgrading P300, P600 and P1200 Power Plants to the new Plus software.

You must have MultiWave II installed for this to work properly.

There are three known common problems with the Plus upgrade:

• No markings on the two chips
• Placing the chips in backwards
• Not getting the bottom row of chip legs into the socket of the front panel board in properly .

No markings on the two chips

Our entire first batch of MultiWave II Plus upgrade kits went out of the factory without proper labeling of the two chips.  There is a front panel chip and an oscillator card chip.

Chip label beginning with number 11 goes to front panel display board.

Chip label beginning with number 12 goes to oscillator card.

Placing the chips in backwards

There is a half moon ‘pin 1’ mark on each chip that must go into the board properly.  Failure to do this will result in the chip being destroyed.  Looking at either the front panel board or the oscillator card, there are several chips soldered in place on each board.  These chips also have the half moon marking on them.  Make sure the new chip you are installing has its half moon marking going exactly the same direction as the other chips on the board.

All chips on each board face the same direction.

P300 Display Board

• Half moon faces to LEFT (toward middle of board)
• Unit turned over resting on it’s top side, bottom panel removed for access

P600 V1 Display Board

• Half moon faces directly UP
• Large square shape, serial number range: P6-01378 and higher or any serial number containing a letter after the P6

P600 V2 Display Board

• Half moon faces to RIGHT (toward middle of board)
• Small rectangular shape, serial number range: P6-0367 – P6-01377

P600 V3 Display Board

• Half moon faces to RIGHT (toward middle of board)
• Small L-Shape, serial number range: P6-0001 – P6-0366

P300/P600/P1200 MWII Oscillator Card

• Half moon faces to LEFT

P500 Display Board

• Half moon faces to LEFT

P1000 Display Board

• Half moon faces directly UP

*In every situation, orientation is being assumed that you are facing each board directly.  In other words, looking at the Display Board from the rear of the unit.  Or, looking directly at the component side of the Oscillator card when holding in your hand.
**Chip socket also has a half moon cutout to help match chip direction.  Silk screen on board also has half moon drawing outlining the chip socket to help match chip direction.
***All chips face the same direction on each board.

Not getting the chip legs into the socket properly

The majority of installation problems we have seen are caused by improperly inserting the chip into the socket.  In particular, the front panel display board.  This problem manifests itself as a series of ‘888’ on the front panel display board if you put them in wrong.

To avoid this problem, always insert the bottom row of pins of the chip into the socket first.  Then, put the top row in second.  Using this simple method makes sure that the chip is in properly.  After you are done, make sure the half moon is going the correct way as well as using your mirror to make sure each pin is properly in place.

Questions?  Call or email us.  We’re here to help

We place a THD analyzer on the front panel of the Power Plant Premier (PPP) to show the THD levels on the line.  Their elimination is important to good performance and all the engineering that goes into the PPP is designed to eliminate harmonics.  When you engage the MultiWave feature, a bit of third harmonics is added top the sine wave on the output.  This added third harmonic increases the charging time for equipment and improves the performance.  So if we went to all the trouble of building a THD analyzer on the front panel to show how the PPP eliminates harmonics, why add them back in with MultiWave?

The answer lies in a little understanding.  The harmonics we work so hard to get rid of are not, in themselves, a problem.  What causes the harmonics in the first place is.  When the AC sine wave is “flat topped” meaning the peak of the AC sine wave has been cut off by too many users on the AC line (a common problem) this causes harmonics and we measure them and display it for you.

To fix this problem, we add the missing energy back into the AC and replace the flat top sine wave with a perfect form that is harmonic free.  So it isn’t the harmonics generated that are bad, they are just an indication of the problem of flat topping.  The real problem is the missing energy used to charge your equipment – which the PPP handles by rebuilding new power.

Now, with MultiWave, we actually create more energy by extending the charging time of the peak of the sine wave once we replace the missing energy.  We do this by adding in a bit of pure third harmonic – but the key here is it’s full energy – there’s nothing missing in the sine wave.

The flat topping we were concerned with is now replaced with the missing energy, and the charging time for the equipment has been extended.

The peak of the sine wave is where most of our equipment draws its power and when that voltage is missing, the average level going into our power amplifiers or connected equipment is lower, thus we have both higher ripple current (more noise) and lower overall volts.

So where does the energy we add back into the sine wave come from?

Because we are building new AC from DC, we store the needed energy in big power supply capacitors in the PPP.  When we build the new sine wave, we draw the energy from those capacitors to feed the connected equipment what it wants.  This can be rather extreme and in some cases as we need to deliver up to 50 amps!  That’s a big challenge for any piece of equipment and it’s one of the reasons the PPP is tough to design and build.

Compare all that to a power conditioner – which cannot store any energy at all – and you quickly see the advantages of the PPP.  Plus, we’re only talking about flat topping.  Remember, one of the bigger problems with power amps is when they draw current to reproduce loud dynamics through your speakers, the whole AC line voltage can drop and often does.

Thus if you were to look at the AC line voltage feeding a power amplifier with a scope you’d actually see the music on the line in the form of a dynamic voltage drop.  The fact that the PPP provides fully regulated voltage on an instantaneous (dynamic) basis is one of its main benefits – and even a bigger one that correcting the flat topping.

Very simply, a power conditioner fliters a small amount of high frequency noise on the AC line while a Power Plant generates new power free of any noise or problems.

The easiest way to think about the difference between a power conditioner and a Power Plant is using a water filter as an example.

A simple water filter is capable of removing some of the dirt found in water.  Simple water filters are not capable of turning dirty contaminated water into pure clean water.

To convert dirty, contaminated water into pure and clean drinking water, it is necessary to first boil the water and convert it to steam; then reversing the process, convert the steam back to water leaving all the contaminants behind.

In the same way, a Power Plant converts the AC to pure DC, then back again to pure AC leaving any problems behind.

A power conditioner cannot perform these functions and can only have a minor improvement on the AC power.  The basic problems in all power lines remain with a power conditioner: unstable voltage, dynamic loss, distortion, noise etc.

You bet!  Because the Juice Bar II (JB II) has no filtering, protection or anything to get in the path of the AC, it’s perfect to increase the number of outlets on a Power Plant of any vintage.

Other uses for the JB II is to place up to 8 Noise Harvesters in the sockets and use this as a huge noise sink.  8 Harvesters in a JB II makes a significant impacto on destroying AC line noise at its source by converting it to light.

Since Noise Harvesters work in parallel, the more you add the greater the impact.

Probably not.  In our experience the P-500 and P-300 Power Plants are not suitable for use powering any TVs.  To reliably power TVs one would need a Power Plant Premier, or, perhaps, a P-600 or P-1000 Power Plant which is not powering a lot of additional load.  TV’s are pretty hungry devices.

Both are based on the same principals of operations but the way they do it makes all the difference in the world.  Online UPS use a class D (digital amp) amplifier to produce the sine wave and are considerably noisier because of it.  Their distortion is rarely better than 1% and typically 2%.  Having said that, 1% is still better than what comes out of the wall.

Here’s the biggest problem, they have a very difficult time with a low power factor load.  This load, unfortunately, is the most common type and found on just about every power amplifier ever made – and most other products as well.  With this type of load, they can’t deliver the peak power required to keep the sine wave from distorting at the peak of the sine wave and under even moderate levels we get distortions approaching 10% – which is far worse than what comes out of the wall.

They do a good job of regulating the voltage but they are very poor sine wave regenerators in the real world.

The Power Plant AC regenerator, on the other hand, can deliver peaks of current in excess of 50 amps (three times what is available from the wall receptacle) on each half cycle of the sine wave, which means there is no distortion at the peak of the sine wave under low power factor loads.  The ability of the Power Plant to generate these huge peak currents, coupled with its CleanWave and MultiWave capabilities, set it apart from any other attempt at regenerating power.

Better dynamics, far more open sound, better imaging for audio and richer, more saturated colors for video.

Everything we hear and see in our AV systems starts out as raw AC power from the wall.  The quality of that AC power has a considerable impact on your system’s performance, whether it’s audio or video related.  Building a solid AC power foundation is a critical aspect of any high-performance audio or video system.

It’s important that you approach the addition of power products with the proper expectations.  Too high an expectation and you might be disappointed.  No expectations and chances are you may not know what to listen or look for.

A Power Plant can make a dramatic improvement in video performance.  This is because perhaps even more than audio equipment, video demands near perfect AC to look its best.  Most video products have switching power supplies to convert the the incoming AC into power for the set and these supplies place a tremendous burden on the incoming power.  when the AC power is cleaner and unrestricted, as it would be only with a regenerator like a Power Plant, video performance will improve.

No, it won’t look like a new three dimensional picture in place of the original two dimensional picture, but the color will be more vibrant, the black levels will be deeper and the overall appearance of the video will be improved.  In many cases, the improvement is quite dramatic and in nearly all cases, a clear and obvious improvement is gained.

With an AC system, the more pieces of the system you can supply with pure regulated power from a Power Plant, the greater the benefit to your system’s performance.  There is a clear cummulative benefit to power an entire system from a Power Plant.

You should hear immediate improvements in a lowered noise floor, removal of any glare and harshness riding on the music and improved depth and spatial qualities to the musical soundstage.

Instruments to listen to, in particular, are stringed instruments such as pianos, guitars and violins.  Power Plants, in particular, make improvements to the upper harmonics and body of these particular instruments.

Passive power products from PS Audio have much the same attributes as Power Plants do, just not quite as pronounced.

No.  In fact quite the opposite. 

Up to their rated power, they have excess energy storage on board (50 amps vs. the 15 or 20 available from the wall socket) that increases the instantaneous power available to your equipment and maintains a steady regulated voltage.

Without a Power Plant feeding your equipment, you will experience dynamic voltage sags when the peak demand gets heavy, say from a power amplifier trying to feed a loud dynamic to your loudspeakers.  There are no power conditioners on the market today, either parallel or series, that can add energy back into the equipment to make up for this dynamic loss of power under heavy demand.

Only an active device that has large amounts of onboard energy storage available, such as the Power Plant Premier, can supply this dynamic power need to your equipment.

Connecting the P5 or P10 to the network and internet allows you to control the Power Plants through your control page on this website as well as view power performance graphs and schedule events.

Connecting to the internet is relatively straightforward. If you have a hard wired Ethernet connection to your network or home router, simply plug the P5 or P10 in and it’ll find its own way out. If you must do it wirelessly you’ll need an Ethernet Bridge. In this Knowledge Base there’s a tutorial on how to connect a wireless Bridge.

The bottom line is this: the P5 or P10 will connect to the internet on any Ethernet setup that you could also plug a laptop in and access a webpage. If you can provide a connection that does that, then you’re good to go.

The quick answer is no you are not missing anything, and in fact gaining quite a bit with the new Power Plant designs, as long as you follow a few simple steps.

First a bit of background.  Following the P500 we invented a new, far more efficient means of regenerating AC power.  The original Power Plant technology was less than 50% efficient on a good day – meaning that less than half the energy consumed by the Power Plant actually went to the connected equipment and the other half was wasted on producing heat.  With the introduction of the Power Plant Premier we achieved better than 85% efficiency losing less than 15% of the energy to heat.  This new technology also gained an order of magnitude lower output impedance in the process.

One aspect of the older and less efficient technology that was lost was balanced power.  The newer Power Plants cannot produce balanced power.  This was felt to be quite an acceptable tradeoff by PS Engineering because balanced power really only buys you two things: lower noise on the connecting power cable and the ability to lower common mode distortion in connected equipment transformers.

Power cables can pickup unwanted noise if they are not properly shielded and balanced power can help lower this noise by as much as 10dB.  However, the noise isn’t an issue and balanced power offers no benefits to lowering the noise if customers use a properly shielded power cable like those PS Audio as well as a number of other manufacturers make.

Connected power transformers can reduce common mode distortion products by as much as 10dB if balanced power is used and nothing if it is not.  This is valuable if you have distortion products on the AC power, which you do not in the case of a Power Plant.

So, summing up, the tradeoff between higher efficiency and lower output impedance vs. balanced power is an easy choice because if customers use a decent power cable between the Power Plant and the connected equipment, then balanced power offers no performance improvement – while the lowered impedance and improved efficiency offer tremendous performance benefits.

I’ve moved to the US from a country which has 230v mains voltage. It was a complete move and as a result I have all my 230v hifi gear which includes a 230v P600 and P300. I need more conditioned 220V output power, so I was wondering if I could somehow convert a 120v P600 to output 230v?

Unfortunately, no because while the earlier Power Plants can be set to accept any input voltage, they cannot produce anything but 120 volts on their outputs.  To produce an export model, we added a step up transformer inside the export units.

Yes you can control the power plants by both global net and IR codes.

A PPP does have the facility to make  small adjustments to output voltage.

This is achieved using a  potentiometer on the underside of the unit.

Please consult your dealer or contact PS Audio for detailed instructions for adjusting the output voltage.

Dimentions of P5   H4″ W17″ D14″

Dimentions of P10 H8.5″ W17″ D14″

Actually no, the fuses are going to be different. For 120v P5/P10s, the fuses used are going to be 5A 5 x 20mm slow blow. For 240v P5/P10s, the fuse used is going to be a 3A 5 x 20mm  fuse. For all Perfect Wave Power Plants, the fuses are going to be physically located on the back panel of the unit itself.

One of the beauties of the PerfectWave DAC and Transport products is the ease of which we can add features, update software and keep your device current and operating flawlessly.  This is accomplished through updates to the firmware inside each unit.  The firmware is the operating instructions the PerfectWave system uses to run the device.

Installing new firmware on the PerfectWave products is a really simple procedure.  The entire update is stored on an SD memory card (similar to what you might find in a digital camera for memory).  The necessary files are loaded onto the SD card, placed in the rear of the PerfectWave product when the AC power is off, then the files are automatically loaded as soon as the AC power is connected.  It’s really that easy.

The simplest way to handle the update is to ask your dealer to either assist you or actually perform it for you. If you want the dealer to handle everything, this will most likely require you to return the unit to the dealer for the few minutes it takes to upgrade the software.

You can easily handle the upgrade yourself if you wish.  To do this upgrade yourself, there are two options: download the software, add it to the SD memory card and install it.  Or, you can purchase a pre-programmed SD card directly from PS Audio for $19.95 (shipped anywhere in the world) and simply install it in the unit.  Either way you choose, the update is easy.  If you wish to purchase the pre-loaded SD card, please click here to go to our shopping cart and order the SD card package to be sent to you.

Installing the firmware upgrade yourself

Most customers will simply install the firmware upgrade themselves because it is so easy to do.  Here is an overview of the steps and then we’ll do it step by step.

Overview

Download the files, copy the files to a blank SD memory card using your computer to do so, making sure the power is off on the PerfectWave product, insert the SD card into the PerfectWave and turn the power back on.  The front panel logo light will blink indicating the upgrade is taking place and when the unit is ready to play again, simply remove the SD card and you’re done.

Check to see what version you have installed

To check what firmware is currently on your PerfectWave product you need to access the setup screen.  Turn the power to the PerfectWave off through the rear panel switch.  Then turn it back on.  As soon as you see the PS logo appear on the front panel touch screen, touch and hold your finger on the logo until the setup screen appears.  From here you can see the unit ID, the firmware installed and set the front panel touch screen brightness if you wish.

Download the files

You should have received an email giving you a link to download the appropriate files for the particular update.  If you did not, contact us atsales@psaudio.com and we’d be happy to make sure you get the correct links to download.

Download the files onto your computer.  It’s easiest in many cases to place them on the desktop of either your MAC or Windows computer.  This just simply helps locate them.  If you received the files in a zip folder you must first unzip the files.  Do NOT try and add the zip file itself to the SD card.  The PerfectWave cannot read a zip file.

You will need a standard SD memory card to load the files onto.  You can only use a standard SD card that does not exceed 2gB in size.  Ultra, Ultra II SDHC SD cards cannot be read in the PerfectWave.  Make sure the SD card is a standard version.

Place the SD card in an SD card reader connected to either your MAC or Windows computer.  SD card readers and low cost and easy to acquire.  They typically plug into your computer’s USB connection and self install.  Most computer setups already have access to this type of card reader. Some cards have a lock on them that prevents data being removed or added – make sure this is unlocked.

Once the computer recognizes the SD card in the reader, double click to see the contents.  Make sure the SD card is empty and formatted.  Most are.  If there is content on the SD card, it’s best to erase the content and clear the card.

With a cleared SD card, simply drag the PS Audio update files you downloaded from our servers onto the card.  Make sure you see all the files on the SD card.  Make sure there is not a zip file on the SD card.

Load the firmware

Turn off your system’s power amplifier.  This will make sure that when you turn the power off on either PerfectWave product you don’t get any unwanted pops and noises.

Turn power off on the PerfectWave product.  To do this, reach around the back of the unit and switch off the power.  Check to see if there is an SD card installed already.  Typically, the PWT is shipped with an SD card either included or installed.  The PWD is typically shipped without an SD card.

If there is an SD card installed, push on the card to get it out.  The card will pop out for you.  Note that the card is installed upside down.

Install the new card with the firmware on it.  Make sure the card goes in upside down as shown in the picture.  Make sure it click into place.

Turn the rear panel power switch on.  If everything is going properly, the front panel logo light will begin to blink, indicating the firmware is being loaded.  It will keep blinking until the entire code has been loaded.  Once fully loaded, the PerfectWave will restart and be ready for operation.

Remove the card and you’re done

You can now remove the SD card.  If it is a PWT, install the original SD card back into the slot so you can get cover art and song titles to appear on the front panel (if you’re connected to the internet).  The PerfectWave DAC does not have an SD card installed.

You can check the front panel setup screen to verify you have the latest update completed successfully if you wish.  If the logo light on the front panel did not start blinking then the firmware was not installed.  Call us for help or email our service department.

As always, we are here to help you in any way we can.  Thanks for the support!

The fuse values for these units are 1 amp slow blow and each will use two (2) 5x20mm 1 amp slow blow fuses

1.   UNPLUG unit

2.      Place a soft protective cloth on the work table to prevent scratching the PWD/PWT top

3.      Place the unit upside down on the work table

4.      Attach a ground strap from your wrist to the center lug of the AC input power connector so you don’t damage the PWD/PWT with a static discharge.

5.      Remove the four flathead screws shown in the red circles with a small Phillips screwdriver

6.      Tilt the unit up vertically on its side with the AC input power connector closest to the workbench

7.      Using a small screwdriver, push the lid equally from each of the four holes where the cover screws were removed.  This will work off the lid until you can grab it with your fingers or it will fall off of the chassis.  Lower the lid to the workbench.

8.      Disconnect the ground wire from the lid using pliers.

9.      Lay the PWD/PWT chassis on the workbench.

10.  Lift each fuse out of the socket by the fuse cover. One is shown removed, below, next to the Bridge guide rail.  There are two fuses per PWD/PWT.

11.  Replace the fuse with a 5 x 20 mm, 1 amp slow blow, 250V fuse, Schurter part number 0034.3117 or equivalent.

12.  Reinstall the fuses.

13.  Tilt the unit on its side with the AC input power connector facing down against the workbench.

14.  Reattach the ground wire to the lid.

15.  Slide the lid into the chassis.

16.  Lay the chassis upside-down on the soft cloth on the work bench.  Seat the lid into the chassis with any pressure as required. Reinstall flathead screws.

Absolutely it will.  The MK II upgrade is an internal PC board that gets replaced and inside the PWD the voltages are always the same regardless of what country you’re from.  So feel free to order the upgrade kit without any worries about voltage.

httpv://youtu.be/QKm6zDAb7G4

Occasionally we’ve received reports of people who connect their PWDs to their computer and – nothing happens. The computer continues to play music over its internal speakers and the DAC just sits there.   In that case do the following:

1)     Power down the computer fully.

2)     Power down the Perfect Wave DAC via the rocker switch located on the back of the unit.

3)     If present, remove the SD card from the PWD and set it aside.

4)     Disconnect the USB cable from the computer.

5)     Turn your computer back on, letting its OS fully boot.

6)     Restore power to your Perfect Wave DAC, letting it fully initialize.

7)     Reconnect the USB cable from the PWD to the computer. The DAC should now show up as a player on your computer.

So, in the rare event that your PWD doesn’t show up on your computer, the technique described above should be just the ticket to solve this problem.

The fuses in both the PWD and PWT are 1 amp 20mm size fuses.  If you wish to replace them with upgraded versions, we recommend you watch the Mark II upgrade video which gives you a view as to how to get into the unit and replace them.

To remove the top cover: http://youtu.be/542U4yTFzJQ

To replace the fuse itself: http://youtu.be/tAy5wFioyS0 near the end of the video

Gapless play has been finished and is working well.  To use gapless, you must be running eLyric Music Manager Server software which is available free at http://www.elyric.com .  Gapless music needs to be tagged as gapless, something you can handle in a tagging program such as iTunes or eLyric.  If you are using eLyric it is possible to go to the Preferences area of the program and tag all albums as gapless (the spaces on non-gapless albums will be preserved) or you can individually tag albums as gapless through eLyric (or iTunes) tagging editors.  On eLyric the tag editor is available by right clicking any track or album.

Under 1 Ohm. Because the balanced and single ended outputs have, essentially, a small power amp-like output, the connection between the PerfectWave DAC and a power amplifier is ideal for use without a preamp. We designed the PWD output to be identical to the high-current, low impedance outputs we would use for a preamplifier. Highly recommended.

The new 192kHz asynchronous USB input on the PWD Mark II will require a driver to work.

If you are running on a MAC, this driver is already built in.  Just plug the USB from the MAC into the PWD Mark II and you should be fine.

For Windows users, download this file to your desktop, then follow the instructions:

http://updates.psaudio.com/MarkIIUSBDriver/PS Audio – USB Audio 2.0 Driver V1.22.0.zip

A couple things to consider here regarding this driver:
- This driver is specifically designed to work with your MKII. If you haven’t upgraded yet the USB 2.0 driver won’t be of any use to you.
- This driver is designed to work on all Windows XP, Vista and Windows 7 computers.
- If you have a Mac running OS X Snow Leopard 10.6.3 or later, your computer already has this driver built in, and you’re good to go. 10.6.3 was released in March of 2010.

To make life a little easier for everyone, I’ve included an explanation of the download and installation process here. If anyone has any questions don’t hesitate to get in contact with us!

Please note that this walk-through is only for computers running Windows 7.

1) Download the PS Audio USB Audio 2.0 Driver file located at www.psaudio.com

- Note that this driver will download as a zipped (compressed) folder that contains 15 individual files.

2) Once downloaded, unzip the folder. This is done by right-clicking the folder and sellecting ‘extract all’ from the list of options.

- This will create a new folder containing the uncompressed files you will need.

3) Open the new folder you just created and double-click the file called ‘Setup’. This will begin the installation process.

- There are two files named ‘Setup’. The one that needs to be clicked has an icon that looks like a computer with a black screen. It is the only file with this appearance.

4) After a couple of minutes, you will be prompted to connect the device you want to install. At this time, connect your Perfect Wave DAC to your computer. Shortly after, your USB driver installation will finish.

5) To complete the entire process and begin listening to music, click Start > Control Panel > Sound. Select the speaker icon that says ‘PS Audio PerfectWave’ as its description. Now click ‘Set Default’ and check ‘Default Device’. Click OK.

6)Next and while still in the Control Panel click on-> Speaker Properties so the window appears, click the Advanced tab and select the highest resolution you will be playing over the USB connection and click OK. The display on the PWD should then change to reflect this resolution when the USB input is selected.

Yes indeed, there are a few good Droid (Android) apps to control the PWD Bridge combination on your mobile device.

While PS Audio does not currently have its own Droid app there are others that work well.  Simply search fro UPnP controller and you’ll find several.  If you need some help, we can provide specific suggestions if you need.

PS Audio will release it’s own Droid controller in the future.

The best way to do this is through the Bridge, not USB.  The reason is simple: only through the Bridge can you effectively and easily transcode the FLAC file format so the PWD can play it.  If you go USB you will need another program like Amarra or Pure Music which will cost money and require you to use the computer for a player.  If you use the Bridge you can simply install the free eLyric program we provide and then use an iPad or iPhone to control the music selection.

Indeed there is. To bypass the volume control on the Perfect Wave DAC and instead use an existing pre amp to control the volume of your system, simply set the gain on your PWD to 100. It’s as easy as that.

Pl

Dear Sir, sorry to hear you’re having a problem getting the f/w loaded.

First thing to check is the lock button on the card, it should be unlocked.

So essentially when you insert the card with the contacts in the upward position the

grey or white lock button on the left hand side of the sd card will be in the forward

position closest to the contacts.

Secondly, there can sometimes be a slight miscontact between the sd card and the

card reader terminals, this is often remedied by inserting the card in and out a few times.

Please contact support@psaudio.com for the 2.2.0 firmware.

You can upgrade your firmware via the community forum here

http://www.psaudio.com/forum/

First, BE SURE you’ve converted your files to WAV.  If you haven’t already, see http://www.psaudio.com/ps/how-to/how-to-convert-flac-high-resolution-files-to-wav-for-homemade-dvd-discs-for/

Be sure you know where these files are as you’ll need to retrieve them in a bit.

You can get ImgBurn at http://www.imgburn.com  There’s a lot of spurious other stuff available for downloads on this page, so click “downloads” in the upper left corner and you’ll see this page.

Click on the big blue box that says “BetaNews” and the “Download now”  Ignore all the other stuff or you’ll be busy all night.

Let IMGburn download and install.  Once it’s done, you’ll see a box pop up.

Click the upper left “Write image file to disc” command button.

Go to “Mode”  and select “Build”

Drag the files from the folder where they’re stored to the “source”  box.  You can search from this window, but at the right of the “source” box you’ll see a white search button right above the yellow search button (above the red “X”)  USE THE WHITE BUTTON to get the files, the yellow won’t work.  Don’t ask me why.

Make SURE these are WAV files.

Set up the Options tab:

Options tab:
o Data Type: MODE1/2048
o File System: UDF
o UDF Revision: 1.02
o No other boxes checked on this tab

Set the label tab with the title, artist, or whatever you like.

Date tab: Use File Date & Time selected

o Restrictions tab: defaults are fine, on the UDF tab nothing is selected

o Bootable Disc tab: no selected options

Click the “build” button in the lower left corner (shown with the file and arrow to the disc)

Pick a spot to store the build file.  Remember where you put it, you’ll use it in the next steps.  This file is stored with a file extension of “.ISO”   Let ImgBurn save the build file, it will take a few minutes.

Go back to ImgBurn Mode tab and select “Write” and make sure the ‘verify’ checkbox is selected rught there under the ‘destination’ box.

Drag the .ISO file you made above to the “Source” box as shown above.  Once that’s done, make sure there’s a DVD in the drive and click the “burn” button in the lower left corner as highlighted with the blue arrow in the middle.

The disc will start to burn.

Don’t pull the disc out yet.  The tray will pop open, and then reclose.  Don’t interrupt as it need to verify the disc.

Let it verify the disc.

It will confirm the burn is complete and you can eject the disc.  You can play this on your PWT.

Enjoy.

Trader’s Little Helper is available at http://tlh.easytree.org

Copy the above link into your browser and go to the web page.  Scroll down and download the program from the highlighted section shown below.

It’s a straightforward installation.  Let it install normally.

Before you start, be sure you know where your source (FLAC) files are and the folder where you want to put the finished WAV files.  It was easier to create the destination folder prior to starting Trader’s Little Helper.

When it’s ready start the program.  You’ll get what looks like a blank window.  Select “Format” and “Decode Audio Files”

Press “Add” so you can select the folders/ files where the original FLAC files are stored.

Press “open” when you’re done.  The files will appear in the upper window.

Next, select the destination folder by picking the “decode to this location” button and clicking the little unlabeled button under the top window shown in blue, below.  Scroll through the file network and select your destination folder.

Press “OK” and then press “Decode”

The program will take a while and give you a status update as it completes each track.  When it’s done you should see all green in the process log.

That’s all there is to it!

If you use this procedure to decode FLAC into WAV for use on the Bridge, click the “Keep foreign metadata” to maintain album art and other metadata tagging.

This How To document assumes you have already downloaded high resolution files to your computer in the popular FLAC format availble from popular high resoution vendors.

Step by Step Procedure:

1.  Download the free Ashampoo Burning Studio 10 from:
http://download.cnet.com/Ashampoo-Burning-Studio-Free/3000-2646_4-10776287.html

If the files are already in WAV format:  Create a new folder on your desktop, named WAV.  Transfer the WAV files to it.  Go to step 7

IF the files are in FLAC format:

2.  Download dbPoweramp file conversion software from:
http://dbpoweramp-music-converter-dmc.en.softonic.com/
Pictorial instructions how to convert files are actually posted on their home website but we will also go over it in detail below.

3.  Create and name two new folders on your desktop. Name one folder FLAC, name the other folder WAV.

CONVERT FLAC FILES TO WAV
4.  Copy and Paste the FLAC music files you wish to burn to DVD into the folder named FLAC. Remember The Perfect Wave transport will only read files and cannot open folders to read the files inside. So only place the actual song files into the FLAC folder.

5.  Open dbPoweramp software and select “dbPoweramp Music Converter”.

Point the software browser to the FLAC folder on your desktop.

Notice the FLAC files appear in the software window.

6.  Highlight all the FLAC files that will be on the DVD and select “OPEN” near bottom of window.

A new window appears:

Select the : “Converting to:” bullet and select Wave
Select the “Uncompressed” bullet, and leave all options in default mode, which you will see (as source)
Select the “Folder” bullet, click Browse, and select your new WAV folder on your desktop as your Output Location. (This is where your converted files will go.)
Select “Convert>>” at bottom of screen
You will notice the conversion taking place. When complete, you will have your music files converted to the WAV format in the WAV folder and ready to burn to DVD.

7.  Burn the WAV files to DVD
Open Ashampoo Burning Studio 10
Select “Expert Functions” from the left side menu
Select “Create a Data CD/DVD/Blu-ray Disc using advanced settings”
A new window opens: choose No ISO (disabled), No Joliet (disabled), UDF 1.5
Leave all other settings at default and boxes unchecked, click NEXT
A new window opens, click ADD
An “Add files and Folders” window appears
At “Look in:” find your WAVfolder, click on that WAV folder
Notice your songs appear in the box, select them all and click “add” then “Finish” at bottom of screen
Notice a new window appears ready to burn the DVD
Click “Next”
Window appears prompting you to insert a DVD
Click “Write”, the DVD will start burning and notify you once complete.

A user has had success using http://www.castudio.org/dvdaudioextractor/ to rip two channel audio files from DVDs.

This is untested from PS Audio.  Best of luck.

If you own a MAC and downloaded or have acquired either FLAC (Free Lossless Audio Codec) or ALAC, MP4 AAC (Apple Lossless) music files and wish to convert them to WAV so you can then burn a DVD and play them in your PWT, there’s an easy to use program that’s free.  It’

s called xACT and can be downloaded by going here: http://www.macupdate.com/info.php/id/14246/xact .  While we don’t have anything to do with this program’s creation or use, we have tested it out and it does work.

When burning a DVD with the appropriate WAV files, be sure and use the UDF burning system.  If you’re on a MAC, then it’s easy, just drag the WAV files onto the DVD and click on burn.  MAC’s use the UDF format natively and will produce excellent results without any intermediary program.  If you are not using a MAC, we recommend NEROhttp://www.nero.com/enu/index.html as the DVD burning software of choice.

Convert hi-res files to WAV using XLD.
(XLD is a shareware program available at:http://tmkk.hp.infoseek.co.jp/xld/index_e.html )

(1) open Preferences, click ‘General” tab, set ‘Output format’ to ‘WAV’
(2) under ‘Output directory’ check the ‘Specify’ button, then click ‘set’ and choose folder you require for output
(3) click ‘Open’ and then close the preferences
(4) click File menu heading and choose ‘Open…‘ from the drop-down menu
(5) choose the file or folder of files you wish to convert and click ‘Open’
N.B. Remember that the PWT only displays track numbers so at this stage it is worth numbering the tracks in the order you require on the DVD. I put all the tracks for a DVD in a single folder and start numbering at ‘01’, ‘02’ etc.

B. Burn a UDF 2.0 DVD data disc using Express Burn.
(Express Burn is a freeware program available at: http://www.nch.com.au/burn/ )

(1) click ‘New Disc’ icon, type in name required for disc, check the ‘Data DVD’ button and then ‘OK’
(2) click ‘OK’ to the info box and add files either by dropping them in the box or by using the ‘Add Files’ button
(3) in the ‘FileSystem’ drop-down menu choose ‘UDF’
(3) click the ‘Burn DVD’ button
(4) in the ‘Burner Drive’ options EITHER choose your disc-burning drive into which you have put a disc OR choose ‘Image File’
(I use the latter option and do the disc burning separately in Toast using the ‘Copy’, ‘Image File’ menu item. In this way I can first burn a DVD-RW to test my hi-res music selections and then burn a DVD-R once I’m happy.)

I have had no success with two other obvious means of burning data DVDs on a Mac:

C. Toast
Under the ‘Data’ drop-down menu there is a ‘DVD-ROM (UDF)‘ entry. I tried this by dropping the files I wished to burn into the box but the resulting disc was declared to be ‘Invalid’ by the PWT.

D. Finder
If you Cntrl-click on a folder using Finder there is an option under ‘More’ to ‘Toast it’. This launches Toast and brings up the ‘DVD-ROM (UDF)‘ entry in the ‘Data’ window i.e. exactly the same as C. above.
I am currently using a relatively old version of Toast (8.05) whereas the latest is 10. I’m upgrading to Toast 10 so I’ll find out whether the UDF 2.0 files necessary for the PWT can now be burnt to DVD.

While it is very easy to generate data DVDs with a Mac, I feel others may find it useful if I share my conclusions. I was looking for a simple process that I could use to reliably create data DVDs to hold the LP albums I am digitising.

As noted elsewhere in this Forum the simplest method is to use ITunes:

(1) create a playlist by hitting the ‘+‘ button at bottom left
(2) drag and drop tracks from your library list to this playlist
(3) hit the burn button at bottom right
(4) in the pop-up box that appears check ‘Data CD or DVD’
(5) insert a blank DVD.

As the PWT can only accept WAV files at the moment, you need to ensure the tracks you wish to choose at (2) are WAV files. If like me you store your music as Apple Lossless or AIFF you will first need to create WAV versions as follows before (2):

(1-1) go to iTunes ‘Preferences
(1-2) under ‘General’ hit the ‘Import Settings’ button
(1-3) in the pop-up box choose ‘WAV Encoder’ from the drop-down menu at ‘Import Using’, choose ‘Automatic’ at ‘Setting’ and hit the ‘OK’ button
(1-4) hit the ‘OK’ button in the ‘General’ menu
(1-5) select all the tracks in your library you need to convert to WAV (hold down Apple key and click on tracks)
(1-6) under top-level menu heading ‘Advanced’ choose “Create WAV version’

Idiots guide to Hi-Res burning on DVD for the Perfect Wave Duo using windows Vista”
Rudimentary step by step
This will assume to are starting with FLAC (most common download format right now)
1: Download and install “Nero 9.0” free trial version (huge program, takes forever!)
2: Download free “Traders Little Helper” software .
3: Create 2 folders that you can easily find and access. Name one FLAC and the other WAV
4: Download FLAC file into pre-determined Flac folder .
5: Open “Traders Little Helper”
6: Select “Decode Audio Files” from the File menu .
7: Import/select the FLAC files to be converted using the “Add” button
8: Make sure the box next to “over-write existing file is NOT CHECKED
9: Make sure to select the WAV folder created above as the destination of new WAV files
10:Click the “Decode” button
You now have usable WAV files. Time to burn the DVD-R
11: Open Nero software
12: Select the “Rip and Burn” option tab
13: Select “Burn Data Disc” button
14: Do not use “Express Burn”. Select the “Burning ROM” option instead.
15: Select “DVD-ROM (UDF)” option from the left side column
16: Click the “UDF” tab on the top
17: Change options box to “Manual Settings” (this allows you to access UDF formats)
18: Leave Partition Type as “Physical “ and select “UDF 2.0” in the box below.
19: Select the “New” button from below
20: Find the folder with your WAV files.
21: Select, drag and drop files you want burned
22: Click the Burn Button from the toolbar.

You’re done!

Stick it into your PW Duo and prepare yourself for a listening experience of a lifetime!

Here’s a tip…Use the I2S connection. I strongly recommend it. I’ll also give a thumbs up to PSA’s pure silver I2S-12 . Incredible detail and sound with this connection.

My motivation for writing it is two fold. First, I wish I had found something like this a few days ago. It would have saved me many hours of frustration. Secondly, I don’t want other newbie’s to walk away from the PWT/PWD because the thought of burning hi-res DVD’s scares them. PSA has created an outstanding product here and you owe it to yourself to give it a try.  Even “idiots” like myself can/should enjoy it’s cutting edge technology and sound. Hope this helps a few folks do just that.

Levi

httpv://youtu.be/UEtBMOa5NfQ

The PWT will load the data for you automatically as long as you are connected to the internet.  Do not load the data yourself as it will not be recognized.

Make sure the SD Card in the back of the PWT is a clean 1gB SD card – and keep in mind the card must be inserted upside down.  Check to make sure you are connected to the internet and cover art will appear on the front panel touch screen if the art is available from our servers.  If not, you can go to our website and go to My PS.  While in My PS, go to View to see what’s been played on your PWT.  Here you can add cover art if it is missing and edit your track information.

Your PWT must be registered in My Registered Products in the My PS section of the website for this to work.

You must also have the latest firmware version installed on the PWT.  To check, go to My PS and then go to Updates to see if you have the latest version of firmware for your PWT.

Please make sure your PWT has the latest firmware installed, which is 1.18. The older original firmware, 1.16 won’t work for cover art. There are instructions in the Knowledge Base on how to check and install new firmware. If you still have questions, certainly contact us.

Unfortunately at this time the Perfect Wave Transport is not going to be able to display Unicode characters on its display. This may however be something that can be incorporated in a future iteration of firmware.

One of the beauties of the PerfectWave DAC and Transport products is the ease of which we can add features, update software and keep your device current and operating flawlessly.  This is accomplished through updates to the firmware inside each unit.  The firmware is the operating instructions the PerfectWave system uses to run the device.

Installing new firmware on the PerfectWave products is a really simple procedure.  The entire update is stored on an SD memory card (similar to what you might find in a digital camera for memory).  The necessary files are loaded onto the SD card, placed in the rear of the PerfectWave product when the AC power is off, then the files are automatically loaded as soon as the AC power is connected.  It’s really that easy.

The simplest way to handle the update is to ask your dealer to either assist you or actually perform it for you. If you want the dealer to handle everything, this will most likely require you to return the unit to the dealer for the few minutes it takes to upgrade the software.

You can easily handle the upgrade yourself if you wish.  To do this upgrade yourself, there are two options: download the software, add it to the SD memory card and install it.  Or, you can purchase a pre-programmed SD card directly from PS Audio for $19.95 (shipped anywhere in the world) and simply install it in the unit.  Either way you choose, the update is easy.  If you wish to purchase the pre-loaded SD card, please click here to go to our shopping cart and order the SD card package to be sent to you.

Installing the firmware upgrade yourself

Most customers will simply install the firmware upgrade themselves because it is so easy to do.  Here is an overview of the steps and then we’ll do it step by step.

Overview

Download the files, copy the files to a blank SD memory card using your computer to do so, making sure the power is off on the PerfectWave product, insert the SD card into the PerfectWave and turn the power back on.  The front panel logo light will blink indicating the upgrade is taking place and when the unit is ready to play again, simply remove the SD card and you’re done.

Check to see what version you have installed

To check what firmware is currently on your PerfectWave product you need to access the setup screen.  Turn the power to the PerfectWave off through the rear panel switch.  Then turn it back on.  As soon as you see the PS logo appear on the front panel touch screen, touch and hold your finger on the logo until the setup screen appears.  From here you can see the unit ID, the firmware installed and set the front panel touch screen brightness if you wish.

Download the files

You should have received an email giving you a link to download the appropriate files for the particular update.  If you did not, contact us atsales@psaudio.com and we’d be happy to make sure you get the correct links to download.

Download the files onto your computer.  It’s easiest in many cases to place them on the desktop of either your MAC or Windows computer.  This just simply helps locate them.  If you received the files in a zip folder you must first unzip the files.  Do NOT try and add the zip file itself to the SD card.  The PerfectWave cannot read a zip file.

You will need a standard SD memory card to load the files onto.  You can only use a standard SD card that does not exceed 2gB in size.  Ultra, Ultra II SDHC SD cards cannot be read in the PerfectWave.  Make sure the SD card is a standard version.

Place the SD card in an SD card reader connected to either your MAC or Windows computer.  SD card readers and low cost and easy to acquire.  They typically plug into your computer’s USB connection and self install.  Most computer setups already have access to this type of card reader. Some cards have a lock on them that prevents data being removed or added – make sure this is unlocked.

Once the computer recognizes the SD card in the reader, double click to see the contents.  Make sure the SD card is empty and formatted.  Most are.  If there is content on the SD card, it’s best to erase the content and clear the card.

With a cleared SD card, simply drag the PS Audio update files you downloaded from our servers onto the card.  Make sure you see all the files on the SD card.  Make sure there is not a zip file on the SD card.

Load the firmware

Turn off your system’s power amplifier.  This will make sure that when you turn the power off on either PerfectWave product you don’t get any unwanted pops and noises.

Turn power off on the PerfectWave product.  To do this, reach around the back of the unit and switch off the power.  Check to see if there is an SD card installed already.  Typically, the PWT is shipped with an SD card either included or installed.  The PWD is typically shipped without an SD card.

If there is an SD card installed, push on the card to get it out.  The card will pop out for you.  Note that the card is installed upside down.

Install the new card with the firmware on it.  Make sure the card goes in upside down as shown in the picture.  Make sure it click into place.

Turn the rear panel power switch on.  If everything is going properly, the front panel logo light will begin to blink, indicating the firmware is being loaded.  It will keep blinking until the entire code has been loaded.  Once fully loaded, the PerfectWave will restart and be ready for operation.

Remove the card and you’re done

You can now remove the SD card.  If it is a PWT, install the original SD card back into the slot so you can get cover art and song titles to appear on the front panel (if you’re connected to the internet).  The PerfectWave DAC does not have an SD card installed.

You can check the front panel setup screen to verify you have the latest update completed successfully if you wish.  If the logo light on the front panel did not start blinking then the firmware was not installed.  Call us for help or email our service department.

As always, we are here to help you in any way we can.  Thanks for the support!

Sometimes it may be necessary to open the transports drive drawer manually to recover
a disk if the unit cannot be powered up or is not functioning properly. You will need a
thin piece of shim stock (0.010”) or a small paper clip will work to perform this operation.
1. Make sure the power switch on the rear panel is off and AC line is disconnected.
2. Insert the tool into the location shown in the photo below and give it a push. It
can take a knack to do this so it may take more than one try.
3. The drawer will slide out roughly to the position shown in the lower photo.
4. Gently pull the drawer out to recover the disk. Push the drawer in when done.

First, for the novices at PSAudio, a “Playlist” is a series of radio stations arranged in a list.  You can select the radio station in tNp as if it was a track in an album, but the radio will stream continuously.

Follow the instructions at http://www.psaudio.com/ps/how-to/how-to-create-playlists-in-james-river-media-center-15/ to create a new playlist (F8, right click the ‘playlists’ header in the left window or click on the button in the center window to make a new playlist)  Give it a name.  For this example I’m going to use a file called “JR Radio.”  It will appear in the left window along with the other playlists.

In the left window go back to the audio area, select ‘connected media’ and select the radio station you want to add to the playlist  or find the URL on the web with your browser.

Right click the URL and copy the shortcut

Go to the “file” menu in JRiver and select “Open URL”  A dialog box will appear.  Right-click-paste your desired radio station into the window in the dialog box

Select “OK”

After a moment, the radio will start streaming through JRiver.  You’ll see the activity at the top of the screen.

Right click on the radio station information, select “Send to” and then the name of the radio playlist, in this case “JR Radio”

Repeat for as many radio stations as you like.

Once you’re done creating the playlist, export it to the M3U the same way we did before—Right-click the playlist name (In this case JR Radio)  and select “Export “  then save it as an extended M3U to the same library location that your server uses. (eLyric, Asset, Twonky, or James River).  The upcoming version of tNp will see this as a playlist and you can choose the radio station as if it was a track in an album.

Repeat for as many playlists you’d like to make.

Enjoy!

First, for the novices at PSAudio, a “Playlist” is a series of song or track titles arranged in a list in the order you’d like to hear them.  It’s the same thing as the tagNplay Queue, only you can store the file on your computer.

New MediaMonkey Installation

If you haven’t already done so, get Media Monkey from www.Mediamonkey.com and install it.  You don’t need the MM “Gold” to do this task, so you can use the free version if you like.  It’s a straightforward install, nothing will appear out of the ordinary.

Near the end, it will ask you what kind of files you want MediaMonkey to use.  ( Make sure the CODEC(s) you use for your tunes are selected in the list.)

Make Playlist

Launch MediaMonkey.  If this is your first time using it tell it where your music files are located   Make sure it’s the same location as your server music library.  Let them load.  It will take a while

Launch MediaMonkey and go to the library where your server music is stored.

To create a new playlist, click on the EDIT menu and “new playlist”

A new box will appear and you can name the playlist as you like.  In this case, I’m going to create a playlist that I’ll use to test the subwoofer called “Bass Test”.

You’ll note the new playlist “Bass Test” appears in the left window below the library in, of all things, the ‘playlist’ section.

In the left window go back to the library and select the folder where the album/ track is located.  The tracks will appear in the window on the right.  Select the track you want to add to the playlist and just drag it to the playlist; in this case “Bass Test.”

Select new albums and tracks and drag them to the playlist.    Drag as many tracks as you like into the playlist.

Once you’re done creating the playlist, export it to the M3U.  Select the playlist you want to export in the “playlists” section on the left (In this case “Bass Test” is highlighted) and then go to the “File” menu from the upper toolbar and choose “Export to Playlist”

A new screen will appear asking for the name of the file.  Make sure you save it as an M3U file, and make sure the location is in the folder your server uses (eLyric, Asset, Twonky, or James River).

Repeat for as many playlists you’d like to make.

Note that you can go to the M3U file in its folder on your hard drive, click on it, and MediaMonkey will play it on your computer. (MediaMonkey can NOT serve it to the Bridge, however, because MM is NOT a UPnP DLNA server)

TagNplay will see these playlists.  You can then use tNp to play the list through your PWD/Bridge as if it were an album.

Have fun!

First, for the novices at PSAudio, a “Playlist” is a series of song or track titles arranged in a list in the order you’d like to hear them.  It’s the same thing as the tagNplay Queue, only you can store the file on your computer.

Make Playlist

Launch iTunes and go to the library where your server music is stored.

REMINDER:  iTunes doesn’t recognize FLAC or many other CODECs.  It does like WAV or MP3 or other Apple formats

To create a new playlist, click on the FILE menu and “new playlist” or CTL-N

A new box will appear and you can name the playlist as you like.  In this case, I’m going to create a playlist called “TechDemo1”.

You’ll note the new playlist “TechDemo” appears in the left window below the library in the ‘playlist’ section.

In the left window go back to the library and select the folder where the album/ track is located.  The tracks will appear in the window on the right.  Select the track you want to add to the playlist and just drag it to the playlist; in this case “TechDemo1.”

Select additional albums and tracks and drag them to the playlist.

Once you’re done creating the playlist, export it to the M3U.  Right-click the playlist name and select “Export “

A new screen will appear asking for the name of the file.  Make sure you save it as an M3U file, and make sure the location is in the folder your server uses (eLyric, Asset, Twonky, or James River).

Repeat for as many playlists you’d like to make.

INTERNET RADIO

The playlist creation process is almost identical for internet radio in iTunes.  Make a new playlist with the file or Ctrl-N command and name it “Radio”  then go up to the “Radio” section of the Library, select your favorite station, and drag it to the “Radio” playlist.

If you don’t like the default radio playlist, you can add your own radio.   Go to “Advanced” in the upper menu and select “Open Audio Stream”  (Or CTRL-U)   Another dialog box appears.

Get your URL from your browser and select the actual streaming URL, not the host web page.   Right click and “copy shortcut”

Paste this into the new dialog box in iTunes that we created in the previous step   and click “OK.”  The radio will start streaming.

To save it to the Radio playlist, right click on the pale yellow box showing the station just under the “iTunes” header.  Select “Add to Playlist” and then the “Radio” Playlist.  The new URL will appear in the playlist.

Save the “Radio” playlist the same way we did before—Select the “radio” playlist in the playlist section, right click and “export”, then save it as an M3U to the same library location that your server uses.  The upcoming version of tNp will see this as a playlist and you can choose the radio station as if it was a track in an album.

Enjoy!

First, create a new playlist.  Right click “playlist” or press the F8 key.  A new playlist box appears.  Give it a name.  In this case this playlist will be for my Battlestar Galactica tracks.  So I’ve called the playlist “BSG”.

Next, pick an album or artist.  (Select “File” under the album artwork)  Select a track and drag it to the “BSG” box.

Load up the playlist with the tracks you want.  Select another album as you like and add as many tracks as you want.

Whenever you want to play the playlist, just select it from the playlist section, pick a track, click on it, and it will start to play on your selected “playing now” player.  Or you can push the big “Play” button at the top of the screen.

Shuffle the tracks with the other big button at the top of the right window.  You can remove duplicates with its respective button, also.

Enjoy!

Start in the upper right of the Twonky Manager screen.  You’ll see a tiny icon that looks like a pair of gears sandwiched in between the “manage” button and the “?” help button.  Click on the gear icon.

The settings submenu will appear.

At the top of the settings screen, click on “Servers”

Note the “add” button  just undre the content paths.  Click the “add” button.  Another submenu appears and you can browse for the files.

Find the files where the new music is stored and click “OK”  Exit out of the setings menu when you’re done.

First, go to http://www.twonky.com/products/twonkymanager/tryit.aspx to download the free 30-day trial (click on the ‘download’ link.)  If you want to buy it, go to http://www.twonky.com/store/index.aspx and buy the “twonky manager”.  The install instructions should be the same.

Click on the link and ‘save’ the download to a file location you wish.  “Run” the installer program.

Select your language

Accept the license agreement

If you purchased the program, enter the code key

If you are updating form a previous version, perform a “clean install”

Enter a name for the media server and select “express install” unless you’re using a network storage device (NAS) or want to specify specific file paths at which point you need to do a “custom install.”

NOTE:  For those with the USB drive or an external NAS, be sure and do a ‘custom install.’

If custom install, uncheck the box if you’re using a NAS, or select the media root directories if you have a specific file path you only want Twonky to use.

Click “install”  If you’re using ‘express’ install, it may take a while for Twonky to find your music.

Let it install Twonky “Beam” only if you want to stream music from the internet to renderers on your network.

Finish the installation by pressing the ‘finish’ button.

THE FIRST TIME you launch Twonky from the desktop icon, it will search for music. If you’re using an external NAS or USB drive MAKE SURE you connect the USB drive or NAS before you click the icon to launch the program. Let it scan

From the installation manual: After scanning your hard disk, TwonkyMedia manager should display music, photos and video located in your shared media folders. If you don’t see any media when you open TwonkyMedia manager. Go to the Server Settings page and make sure the Content Paths point to folders that contain media.
Next, you should right-click on all of the icons that appear in the Play Here section and name and assign icons to all of your devices.

Once the music has been found by Twonky, tagNplay will find Twonky as a Library.  Select it on the Library list, and your PWD (or other renderer) as the player and run tagNplay normally.

Another PC/MAC DLNA server available that works withthe PWD DAC Bridge is dBpoweramp’s Asset.
Available here at http://www.dbpoweramp.com/asset-upnp-dlna.htm

It’s avaialable free as the DLNA server only or for an additional charge if you want to run Internet Radio, Dynamic Playlists, Jukebox Random Selection, Browse by File & Folder, and PC Sound Card Streaming

Note: You will need tagNplay or some other controller to make it work.

Go to the link above and download the version that’s applicable to your computer.  There’s a version available for MAC, Windows 7, Vista, and Windows XP

When installing, the program will show you a dialog box and ask if you want to implement the program in ‘local account’ mode, which is recommended.  Let it load.

You’ll see a dialog box titled “Asset UPnP Configuration”  Click on the ‘edit’ button.  A new box pops up. In the upper left corner you’ll see the “Audio Library” section of the new screen.  If your music files are located there, you’re good to go.  If not (say your files are on an external USB drive) then click on the red “[add folder]” text.  Another dialog box will pop up that you can browse through your files and tell Asset where your music is located.  Click “OK” and you’ll get back to the ‘advanced settings’ dialog box.

The other areas’ defaults looked OK to me (if anyone has better settings please let us know) so I left those alone.  Click the “OK” at the bottom of the ‘advanced settings’ box.

You’ll go back to the ‘Asset UPnP configuration’ box and it will take a few minutes to load all the titles into the library.

When it’s done, click “OK”  this dialog box will then disappear.

Don’t panic.  Asset is running.

Start tagNplay.  In the library tab, you’ll see “Asset XXXXX” where ‘XXXX’ is your computer’s name.  Select it as the library.  (Select your PWD as the ‘player’)

Let it load titles into tagNplay.  You can then select titles from tagNplay and get music playing.

Enjoy.

James River Media Center 15. 

Go to http://www.jriver.com/download.html and download the program.  You can give it a try for 30 days, so you don’t have to buy it straight away.

Install the program on the computer you want to act as your server.  In my case it was a small Netbook.

Once it loads and you get the main screen, you have to first set up the program to make it work as a DLNA server.  Go to the “Tools” menu and select ‘options’ at the bottom of the list.  A new screen will pop up.

There is a list of selections at the left side of this new screen.  Select “Media Network”

In the screen that appears for ‘media network’, you’ll see a check box, labeled “DLNA Server”  in the upper right corner.  Click on this box and make sure it’s checked.  This allows sharing of the media with other DLNA devices.

On the lower right of the screen is the “DLNA Renderer” check box.  Set it also. This lets tagNplay to control Media Center.

Immediately under that box is the “DLNA Controller”  Check it, too.   This will allow you to stream music directly to the Bridge/PWD if you don’t want to use tNp or if you lose your iDevice.

I’ve checked the box in the upper left “Library Server” which allows the other devices to share the library and playlists.  I don’t know if it does anything, but it seems to work.

Before you leave this page the following is important to get optimum performance from the Bridge:

On the right side of this screen, you’ll see a menu of things starting with ‘customize views,’ ‘Display name,’, ‘Audio, ‘Images,’ etc.

Under ‘audio,’ set the ‘Conversion’ (it’s a drop-down menu) to “Never Convert”

Under that, set the ‘encoder’ to “Uncompressed Stream”

If these last two steps aren’t done, the JR will send MP3s to the Bridge and you’ll get subpar performance from the Bridge.

Exit the options menu.  If it asks you if you want to save changes, answer ‘yes.’

Next, find your music.

On the main screen, at the left you’ll see a list.  Select “Drives and Devices”  Click on ‘my computer’ and click down until you get to the area where your music files are stored.  If they’re all in one folder,  highlight the master folder.  You’ll note that on the middle upper side of the screen is a button labeled “Import Into Library”  When you’ve found your music files click on “Import Into Library” and wait for the files to load into the JR.  Repeat as needed until your music files are loaded.

Now JR is ready to stream music to the Bridge.  Make sure this computer is linked to the same router as the Bridge.

You can verify the Bridge and JR are on the same network if you go to the left menu on the main screen and selct “Playing Now.”  You should see “PerfectWave DAC xxxxxx” as an option on that list.  If you don’t,  the units aren’t talking to each other on the network.

Play music!

1) Start tagNplay.  Select the ‘Library’ tab at the bottom of the screen.  You should see the computer with the JR files on it labeled like “[Computer Name] (Generic DLNA)”  where [Computer Name] is the name you gave your computer when you first bought it.  Then select the ‘Player’ tab and you should see ‘Perfectwave DAC xxxx.’  Select it.  Go to the ‘Music’ tab.  Select ‘Audio.’  Choose ‘Artists’ or ‘Albums’  (depending on how you want to find your music) and let the files load into tNp.  Pick your album, select it or a track to play, adjust the volume and let ‘er rip.

2) If you lose your iDevice or the spouse/kids swipe it for their own use, you can stream music directly from the JR comptuer to the Bridge.   With Jr up and running go to the ‘Playing Now’ option on the list on the left.  Open it, and click on the ‘Perfectwave DAC’  (All you have to do is click it, this tells JR where you want to send the music.  ‘Zone 1’ will be the speakers on the computer where JR is running)  Then go up to the ‘Audio” section on the list.  Open the ‘Audio’ section and you’ll see ‘Albums’, ’ Artists,’ ‘Files,’ etc.  If you select ‘Albums’  they should appear in the center of the screen (It may take several weeks for JR to find all the album art and get it to the right place,  but it will get there eventually.)  Scroll over the album you want to play, and the word “play” will appear under the album.  Click it. The music will stream to the Bridge.

Enjoy.

Best of luck.

—SSW

If you’re on a MAC and are using iTunes to manage your music library, you may have noticed that it will not handle FLAC (Free Lossless Audio CODEC) encoded files.  FLAC encoded music is the defacto standard for reducing file sizes without loss of quality.  FLAC files are bit-for-bit perfect copies at about half the file size.

iTunes will not recognize FLAC files and offer their own method of lossless encoding called Apple Lossless.  Unfortunately, Apple Lossless doesn’t work with anything other than iTunes so that vast majority of people use FLAC and are forced to forego iTunes as their music player and library organizer for these files.

Now there is an easy way to fix this.  There is a free program you can download called Fluke that fixes everything.  Download FLUKE

Fluke is a plug in for your MAC.  It’s simple to use.  Once installed, you simply choose any FLAC encoded file or group of files and choose to open with Fluke.  Fluke quickly converts the FLAC file identification tags to fool iTunes into thinking they are Quicktime files (Quicktime is Apple’s video format).  This allows iTunes to recognize these and play them.

One last trick you may wish to know.  If you want to copy these files onto a CD, iTunes won’t allow you to do this without yet another trick.  While iTunes will now play the FLAC files for you, if you wish to copy these files onto a CD you’ll first need to convert them to WAV.  To do this, simply highlight the files you wish to convert, right click the highlighted files and click on “convert to WAV”.  Now you have a bit perfect copy of the FLAC files in WAV.  You can then add these to your playlist in iTunes and burn them onto a CD.

If you are building a library of music in iTunes, it’s important to setup the CD transfer (RIP) process just the way you want.  iTunes default setting is to compress the music you copy from a CD using their AAC encoder at 128kbs.  Many of us building music libraries do not want to compress the CD because of the loss of audio quality.

Before you spend all the time required to import your music library, decide what you want to do with your library in the long term.  For our library setup, we prefer FLAC (Free lossless Audio Encoding) for a number of reasons, compatibility with nearly all music server systems being chief among them.  See our How To tip on FLAC and Exact Audio Copy for details.

An Apple based system is essentially a closed system although an increasing number of music servers now support Apple libraries.  If you wish to use iTunes to import your library, check with the manufacturer of your music server to see if it is compatible with the server.  Most are.

However, if you choose to stick with iTunes for your library, this article will help guide you how to setup iTunes properly.

Go to the Edit tab.  Click on Preferences .  In the general preference tab you will see a menu item labeled ‘when you import a CD ‘ and a window that gives you a set of choices.  Next to the window is the ‘Advanced ‘ option button.  Click this button and the advanced option menu will open.

From here you can select your quality level options.  Using the drop down box labeled ‘import settings ‘ you will see a number of options, including WAV, AIFF, Apple lossless, MP3 and AAC.

WAV and AIFF are lossless full 1:1 copy settings.  Since a typical audio CD has approximately 750mB of data stored on it, using these settings will require approximately 750mB of hard drive storage space to copy the CD.

WAV files are difficult to attach cover art and meta data to as they are not designed to carry this.  So if uncompressed music is your choice, the AIFF would be a better choice in iTunes.  The two uncompressed formats are identical in performance.

Apple lossless will reduce the size requirements of a CD in half without any loss of data.  Thus, a 750mB CD is reduced in size to 375mB.  Our recommendation would be to use Apple lossless for your entire library.  It is sonically indistinguishable from either WAV or AIFF and easy to attach cover art to.  Further, Apple lossless takes up only half the space of either WAV or AIFF.

AAC and MP3 are lossy compression methods.  MP3 is the lowest quality copy method available on iTunes and we recommend avoiding it.  AAC (Advanced Audio Coding) is a better lossy coding method and if you really need the extra hard drive space, choose this instead.

A music server for around $1500?  At that price it’s hard to imagine that it’s even worth writing about but, in fact, you can build one for that price and it is one of the better music servers available today regardless of price.  We’ll show you how.

First, a little background on music servers.  A music server is a recent development.  Think of it as a readily available library of all your music, accessible from the palm of your hand.

No longer will you need to search through the hundreds, perhaps thousands of CD’s in your collection to find what you want.  With a music server, everything is easily available including all the cover art, song titles and all of it categorized to Genre, tempo, artist etc.

A music server is one of the ways you can really start to appreciate the wealth of music in your system, perhaps as never before.  Let’s face it, how many of us remember every song title in our collection well enough to go find it?  With a music server, you simply scroll through all the available music and select what you want or let the server itself surprise you for hours on end.

Modern music servers are growing in popularity and what’s available today run the gamut from low cost and lacking or high cost and cool without much in between.  We’ll show you how to build a low cost system that lacks very little, is easy to setup and will bring you hours of entertainment.   Is it the ultimate music server?  Heck no, but it’s a fun project and an easy entry into this category for those of you willing to spend a short weekend setting it up.

A music server consists of three main elements: storage, control and playback.

In this system we’ll need only two pieces of equipment to provide these required functions and a third to make the whole setup a high-end performer.  You will need to purchase an iPod Touch for $229, a 160gB Apple TV for $329 and a PS Audio DLIII DAC for $995.

Your first task is one you would have to do regardless of how you built or purchased your music server.  You will need to transfer all your CD’s to a hard drive.  We recommend using iTunes for this purpose.  iTunes is a free software program from Apple and works with Windows or MAC’s.

Open iTunes and set it up to RIP (transfer) your CD’s in the resolution you want.  The choices for resolution vary from uncompressed (identical to the original) or compressed.  If you plan on storing all your music on the Apple TV’s internal hard drive you are limited to about 140gB (although several companies offer 250gB upgrades to the ATV).  This means you can store approximately 450 uncompressed CD’s using Apple lossless, or thousands at a compressed rate.  Choose how you wish to proceed, then transfer your library to your computer’s hard drive or external storage.

The next task will be to setup the Apple TV (ATV).  This is very simple to setup and the included instructions are easily followed.  You will need a TV with either a component video input or HDMI input to setup the ATV.  You can also use a modern computer monitor if it has HDMI for its input.  You will not require this monitor after you setup the ATV.

The ATV communicates over your home’s network either through it’s built in wifi that wirelessly connects to your computer’s wireless router or by simply plugging in an Ethernet cable connected to your network.  It is best if the ATV has access to the internet through either of these means.  Setup is straightforward and the system walks you through the steps right on screen.  Couldn’t be simpler.

Once your ATV is setup and synchronized to iTunes, it will transfer all the music you ripped into iTunes to the ATV’s hard drive.  Once the transfer has taken place, you will not require your computer anymore after the setup is completed.

Now you need to prepare your iPod Touch to be the remote control.  Go here http://www.apple.com/itunes/remote/ and download the free remote application.  Follow the instructions to add this software to your iPod Touch.  Once this has been loaded, all you need to do is connect the iPod to the ATV.

This step is easy.  With both the iPod Touch running and the ATV running, you will be able to see the iPod Touch in the menu of the ATV labeled “remotes”.  Choose the iPod Touch and tell the ATV it’s ok to connect.  At this point, you are basically done.

The last step is to take an optical digital interconnect cable and connect the ATV’s optical digital output to the PS Audio DLIII’s optical digital input.  The audio outputs of the DLIII will be connected to your system’s preamplifier inputs and the system is complete.

At this point you can disconnect the video screen from the ATV and you can turn off your computer.

Touch the remote icon on the iPod, and it will find the ATV and instantly you will be able to see all the cover art, playlists, anything you loaded onto the ATV will be available to you on the iPod Touch.

If you want internet radio available, create a playlist in iTunes labeled “Radio”.  Open the radio tab in iTunes and drag as many stations as you want into the Radio playlist you created.  Synchronize iTunes with the ATV and the radio playlist will be added and available to you on the iPod Touch in playlists.

Enjoy!

Letting the servers listed here do the work of converting lossless and lossy files (FLAC, ALAC, WMA, MP3) to WAV before sending them to the Bridge, improves Bridge performance dramatically and eliminates any chance for dropouts on high resolution audio, up to 192kHz, 32 bit files.  This is the recommended performance setting for best Bridge performance.

dBpoweramp ASSET

To change settings in Asset, first go to your ‘start’ button, select “all programs” and then “Asset UPnP configuration”

A new window appears.

Press the “Edit” button to go to the advanced settings page:

Go to the “Audio Format Streaming” and change FLAC from “as is” to “as WAV.”  Repeat for all your other high-resolution CODECS.  Click “OK” when done.

Asset will now stream all FLAC files form the server to the Bridge in WAV.  You will see the PWD front panel display “WAV” as the format even though the file was stored as a FLAC (or other CODEC)  This is normal.

James River Media Center 15

Start JRiver 15 normally.  From the ‘Tools’ menu select “Options”  or use Control-O

The Options window appears.

To Transcode “on-the-fly” (I.e., only WAV gets sent to the Bridge)

Go to JRiver under tools,>options> Media Network> client options> Audio conversion and set conversion to “Always convert” and encoder to “Uncompressed.”

To send native files from server to Bridge without transcoding (I.e., files stored as FLAC gets sent as FLAC, AIFF as AIFF, etc.) :

Go to JRiver under tools,>options> Media Network> client options> Audio conversion and set conversion to “Never convert” and encoder to “Uncompressed.”

As with Asset, above, JRiver will stream all files to the Bridge in WAV format.  The PWD front screen will show “WAV” instead of “FLAC or your other CODEC.  This is normal.

A user submitted a fault condition where:

The problem is that these problem albums will only play a few of their tracks on the PWD/Bridge—SOME but not ALL.  This is confusing to me.  The tracks show up on my iPhone with the correct track length in metadata but after they begin to play (< 1 sec.) the correct track length, displayed below the track-progress bar, drops to 0.00 and the track skips, sometimes several tracks, one after the next, until it gets to a track that the PWD “likes” and it plays as normal.  This is frustrating b/c, while a minority of the albums, some of my favorites are so affected.

This was fixed by going into the Twonky server utilities and activating the “Rebuild Twonky Database” command.

All is copacetic after the rebuild.

If you are going to store your music on an external hard drive (which is what we recommend), make sure it is either eSATA or USB 3.0.  Our preference is USB 3.0

While most older computers do not support USB 3.0, all new ones will and you should always plan for the future.  USB 3.0 is no more expensive than USB 2.0 but it is 10 times faster in its read/write speeds.  In fact, USB 3.0 is faster than the hard drives and computers you connect them to.  USB 3.0 is also backwards compatible with 2.0 so there’s no worries about compatibility.

Our best experiences have been with Western Digital.

The upcoming PS Audio Silent Server will be fully USB 3.0 compatible for best speed options.

Our recommendation is a USB hard drive, not a NAS.  NAS (network attached storage) are typically slower than USB hard drives (3.0), more expensive, restricted by their operating systems and have built in media servers that are not always the best.

Connecting a USB 3.0 external hard drive to your computer or PS Audio’s upcoming Silent Server is by far the preferred method of storing and backing up your treasured music collection.

As of this writing at 2tB external USB 3.0 hard drive is less than $150 USD delivered.

Once you download and install PS Audio’s eLyric Music Manager (a free program) you can manage your music collection and stream music from it to any UPnP player on your home network, like the PS Audio PerfectWave DAC and Bridge , or any UPnP compatible player.

eLyric must always be running on your computer for this to happen.  This tutorial walks you through how to auto load eLyric when you boot your computer.  Once eLyric is running, you can simply click the eLyric logo (or minimize button) to minimize it and keep it running in the background.

On a MAC

By far the easiest, simply go to the MAC system preferences (Apple logo top left hand corner of your screen).  Select “Accounts” in the System section. Once the accounts window opens up, click on “Login Items”.  Click the “+” to add a program.  The Finder window will appear.  Go to Applications, select eLyric and you are done.

On Windows

Right click on the eLyric icon, select copy, then click on the Start button, select programs, startup, right click on Open, and paste in a shortcut to eLyric. Note that this will only load eLyric once a user logs in. If you want to make eLyric autoload every time the machine boots (recommended), do as follows if you are running Windows 7:

1. Click on Start and then enter the following command in the search box:
netplwiz
Hit ENTER. This will load the Advanced User Accounts control panel applet.
2. In the Users tab, uncheck the box next to Users must enter a user name and password to use this computer.
3. Click on the Apply button at the bottom of the User Accounts window.
4. When the Automatically Log On dialog box appears, enter the user name you wish to automatically login to Windows 7 with. Then enter your account password in the two fields where it’s asked. Click the OK button.
5. Click OK on the User Accounts window to complete the process.
From now on, when your PC starts up, the account you have configured will log on automatically, and eLyric will autoload.
If you are running other versions of Windows, see http://pcsupport.about.com/od/tipstricks/f/auto-login-windows.htm for how to configure autologon.

First step is to purchase a low cost server like the Habey EPC-6568 or simply find an old Windows based computer you have laying around to install everything on.

Then, install PS Audio’s free server software eLryric available for download athttp://www.elyric.com and install, then follow the remaining instructions in this How To.

Installing eLyric is a breeze – just run the executable file downloaded from PS Audio, run eLyric, select the directory where your music files are (which can be stored on the computer or on the network or via a USB hard drive attached) and you’re good to go. For those who want to run a “headless” Windows server, meaning a dedicated server without a keyboard or monitor, here are a few useful tips.

The advantages of such a machine as the Habey are: small, cheap, can fit right in or near your stereo rack if you want, and by running a dedicated box you don’t need to worry about other overhead on your personal laptop or desktop.  There are many options out there, – think of it as a NAS but more flexible.
A. Enable Windows Remote Desktop so you can manage your server over the network from another PC:

http://www.howtogeek.com/howto/windows-vista/turn-on-remote-desktop-in-windows-vista/

Note that While all editions of Windows 7 and Vista (and XP) can be a remote desktop client, only the Windows 7 Professional,Ultimate, and Enterprise OR Vista Business, Ultimate, and Enterprise editions can host a remote desktop connection.

To set up remote desktop in Win XP seehttp://www.microsoft.com/windowsxp/using/mobility/getstarted/remoteintro.mspx

If the version of Windows that you are running in your server does not support being a remote desktop host, you can still get the necessary functionality with any one of the freeware VNC versions .

B. Set up eLyric to autoload:

Right click on the eLyric icon, select copy, then click on the Start button, select programs, startup, right click on Open, and paste in a shortcut to eLyric. Note that this will only load eLyric once a user logs in. If you want to make eLyric autoload every time the machine boots (recommended), do as follows if you are running Windows 7:

1. Click on Start and then enter the following command in the search box:
netplwiz
Hit ENTER. This will load the Advanced User Accounts control panel applet.
2. In the Users tab, uncheck the box next to Users must enter a user name and password to use this computer.
3. Click on the Apply button at the bottom of the User Accounts window.
4. When the Automatically Log On dialog box appears, enter the user name you wish to automatically login to Windows 7 with. Then enter your account password in the two fields where it’s asked. Click the OK button.
5. Click OK on the User Accounts window to complete the process.
From now on, when your PC starts up, the account you have configured will log on automatically, and eLyric will autoload.
If you are running other versions of Windows, see http://pcsupport.about.com/od/tipstricks/f/auto-login-windows.htm for how to configure autologon.
C. Make it possible to remotely shut down and reboot your server:
The Windows Remote Desktop interface does not allow you access to the “shutdown” and “reboot” commands via the start menu. There are several ways around this – one is to set up two icons on the server desktop. Do this as follows:
1. Right-click on the desktop and select “New”, then “Shortcut.” In the field that says “Type the Location of the Item”, enter c:\windows\system32\shutdown.exe /r and click “Next.” As a name for the Shortcut type Reboot.
2. Do the same to create a second shortcut but instead of c:\windows\system32\shutdown.exe /r enter c:\windows\system32\shutdown.exe /s and name it Shutdown.

Voila, you now have a headless server that you can manage remotely from any PC on your network, will autoload eLyric, and which can be rebooted and shut down remotely as needed. Of course you can always add a monitor, mouse and keyboard if you want to, but there really is no need.

Two last tips if you are building a machine to use for this purpose:
1. We recommend using two separate hard drives if possible, so that one can run the OS and its swap file and one can hold the music files. (I am using 7200 RPM drives but the math says that 5400’s should be OK.) If you can only fit one drive in your machine you will probably be OK, or you can also use an external SATA or USB drive.
2. Consider using Panda Cloud Antivirus , as it seems to have rather less CPU overhead than most.
I will post an addendum to this message about recommended power management settings.

Our thanks to Bob Caroll for this DIY project

Not if you don’t want it to.  In the Preferences menu on eLyric Music Manager, you can turn on the program’s ability to embed the cover art and tag information into your music files if you wish, or not.

The default is to not embed the data.

The reason you would choose to embed and make permanenent the meta data and cover art in your music files is if you want a permanent record that stays with your music so whenever you decide to make the music files portable or use another music management program you do not have to repeat the work of cover art and tags.

On the other hand, the opposite may apply.  That is, if you have already tagged your media and added cover art and are using eLyric simply as a server and not a music manager, then make sure the embed feature is turned off (which it is by default).

All the information about the music library eLyric Music Manager uses to display and control your library is stored on a database on the computer you are running the eLyric Music Manager server.  It is possible to edit that database if you want to make large scale changes, such as tagging all albums gapless or tagging individually.

These changes can be made through the user interface by right clicking on any album or track and using the built in tag editor.  However, this process can be slow for large libraries and this is an alternative.

This How To is not recommended for the non-computer savvy person.

The database is called a SQL database (Sequential Query LOgic).  You can download an easy to use SQL editor and make change.  We recommend first making a backup copy of the database just in case you need to go back.

1) If not installed yet install Firefox, any 3.6 or higher version should

work.

2) Install the Firefox AddOn

https://addons.mozilla.org/firefox/addon/sqlite-manager/

3) If needed end using the taskmanager the eLyric, eLyricServer and the

eLyricTransformer

4) Go to the c:userusernameeLyric directory (if on a MAC, open Finder->Your user name area->eLyric Folder)

5) For backup/restore make a copy of the eLyric.sqlite file.

6) Start Firefix and select on the actionbar Tools -> SQLite Manager

7) At the SQLite Manager actionbar select Database -> Connect Database and

select the c:userusernameeLyriceLyric.sqlite database

Select the Execute SQL tab, enter SELECT * FROM Albums and

press the Run SQL button. All your albums will be shown, the last

column is the gapless column withe “false” as values.

If you want to change all albums to gapless:

Enter UPDATE albums SET “gapless” = “true” WHERE “id” > 0 and

press the Run SQL button. You’ll see that all gapless values have

changed to “true”.

If you want to change selective albums to gapless:

Double-click on the album you like to change and a “Edit Record” popup

will appear. Enter true at the gapless field and press the OK

button.

It’s possible to restore the original situation by again stopping the three

eLyric tasks and restore the original eLyric.sqlite file as backuped in step

5.

Adding a library or a folder of music to eLyric is rather simple and straightforward.  Open eLyric, and click on the folder icon on the far left side of the page as shown in the photo.  This will bring up your computer’s Finder window (on a MAC) or directory window (on Windows).  Choose a folder or directory where your music is stored.

Select “Choose” and the library information is added as a library to eLyric.  The actual music files themselves are not added, just their information.  You can then edit and manage this new library any way you wish or add multiple libraries.

You can click on the library name once it has been imported and change the name of the library to anything you wish.

Deleting tracks in eLyric is easy.  Simply highlight and hit the delete key or drag them to the small trash can icon at the bottom of the eLyric window.

Deleting tracks does not delet the actual music track you have stored in your library, rather it just deletes them from the eLyric library database.

Your library or folder where your music is stored may contain multiple track info’s that do not represent the actual tracks themselves.  The actual tracks will typically have the sample rate and bit depth information available for the program to read and list for you contained in the metadata header file of the track.

To delete these multiple entries from the eLyric database and cleanup the library, simply highlight the tracks you wish to delete (hold down the control/command key to select multiple tracks) and then drag the ones to be deleted to the small trashcan icon at the bottom of the eLyric window.

This read error on eLyric user interface means you are not connecting with the library on the server.  If you get an error like this, check to make sure you are connected to the server, which could be eLyric, Twonky, J River for example.  You access the library by opening the Devices tab on the eLyric app and selecting the library you wish to connect with.

Yes.  Go to your Preferences area in eLyric and you’ll see a list of sorting categories with checkmarks that you can turn on/off.  When you do this, they will appear both in the main window and on your phone.  If they show in the main window but not the phone, then you simple need to click on the “Restart UPnP Server” button below the list of sorting categories on the left.

eLyric Music Manager and server is available by going to www.elyric.com which is a page located on this website under the Resources dtab.

eLyric Music Manager and server program is free to download and use and works with any UPnP based program or controller.  The twmost popular controllers being used with eLyric Music Manager are PS Audio’s eLyric Controller, Linn’s Kinsky controller and Plug Player, all available on the Apple app store for download and installation on your Apple mobile device.

In the eLyric Manager you can create and manage playlists from your library that will then appear in the eLyric Controller to play whenever you wish.

To add or delete tracks within a playlist, make sure you have created a playlist (using the + button) and then simply drag the track or tracks you wish into the playlist to be added, or to the trash can to be deleted.

No, it is currently not possible to import or export playlists in eLyric Music Manager.

If your eLyric controller crashes, it is very helpful to PS Audio engineering to receive a crash report so we can make the product better.

The “crash” can best be described as when you attempt to perform a function in eLyric and all of a sudden the program stops and the iDevice (iPhone, iTouch, or iPad) reverts back to the main screen.  When this happens, if you can, please remember what you were doing exactly in the time that led up to the crash and include the notes in the email with the crash log.

To get to the crash log, plug your iDevice into your computer and let it synch to the computer.  (You must have iTunes installed on your computer do do this.)

These crash reports are stored on your iPhone or iPod touch at the time of the crash. When you sync your device with iTunes, the reports are copied to your computer to the following folder:

If you are using a Mac:
<YOUR HOME FOLDER>/Library/Logs/CrashReporter/MobileDevice/<DEVICE NAME>
If you are running Windows XP:
C:Documents and Settings<USERNAME>Application DataApple computerLogsCrashReporter<DEVICE NAME>
If you are running Windows Vista:
C:Users[USERNAME]AppDataRoamingApple computerLogsCrashReporterMobileDevice<DEVICE NAME>
The files will start with the name of the application and contain a date stamp.

All crash reports starting with “PSAudioController<date_of_crash><sequence_number><USERNAME><iPhone(Touch/Pad).crash” might be helpful for us.

On some VISTA machines the C:Users[USERNAME]AppData is hidden and you can’t see the folder from the [USERNAME] window.  If you can’t see it, do a search for “.crash” and it will guide you to the correct folder.

Please email your crash reports (ONLY the files ending in .crash) to support@psaudio.com

along with the notes you made of what you were doing when the crash occurred; this will help the engineering staff track down the problem so it can be fixed.

Thanks!

This procedure is a bit time intensive.

NOTE: PS Engineering has discovered that some wireless Ethernet bridges (not to be confused with the Bridge inside the PWD, but act as a wireless link between the DAC and the router) exhibit the exact same problem.  So for anyone using a wireless bridge that has the problem of not seeing the Bridge, please try and hardwire the PWD DAC Bridge to the router first to see if this corrects the problem.

First, download http://updates.psaudio.com/bridge/StatusReport.zip

2) Type the link (or copy/paste) into your web browser.  Tell the system to ‘save’ the file (to a place you remember for later on).  When the file downloads, ‘open’ the file to see where it is.  You’ll see a file called “StatusReport.zip”  Click on it to unzip the file.  In that file you’ll see another file called “TestBridge”

3) Copy the “TestBridge” file to a USB thumbdrive to the highest (root) directory.

4) Insert the USB drive into the Bridge USB port.

5) Power-cycle the PWD using the rear power switch.  Let the system reboot.  The USB drive will blink its access light.  (with some users the file didn’t write immediately, so wait 30 minutes or overnight)

6) The Bridge will create a file on the USB drive called “Psaudio.log”

7) Send this file to Dennis@psaudio.com

Dennis will send you another zip file, or, if you prefer, the static IP address settings.

9a) If you opt for the zip file, Open the zip file and transfer that file to the USB thumb drive you used before (delete the old “testBridge” file first)  Take the USB drive to the Bridge and insert it into the USB port.  Power cycle the Bridge using the rear power switch.  The static IP address settings will be set into the Bridge.

10a) Start tagNplay and see if it can ‘find’ the DAC.

If it does not work, go to the System setup screen (power the DAC off with the rear power switch, then turn it back on.  When the initialization screen is displayed on the front panel, touch the screen.)  You’ll see the system setup page.  There’s a green button there to reset the Bridge (NOTE: it may take 20 seconds for the system to recognize the Bridge is present.  Don’t panic, it will show up)

You can also set the IP addresses manually.

9b) Starting at Bridge main screen, touch “home.”  Then touch the green button on the right just next to the words “media Bridge” to get to the Bridge setup page.  Touch the green button under the hardwire ethernet icon (middle of page, left side) You’ll see all the IP addresses.  At the bottom of the page is a green DHCP button.  Press it, and it will turn red to indicate you’re now on ‘static’ IP addresses.

Green boxes will appear to the right of each line.  Press the box to get to the editor that will allow you to manually set the IP addresses.

Hit the “DEL”to get to where the correction is to be made, and type in the new address.

Hit “submit” to exit from the line editor.  Use the addresses Dennis recommends.

When you’re done, exit from all screens (press the green check boxes in the lower right corner of all screens) and go back to the main Bridge playing screen.

At this time it is not possible to repeat a track using the eLyric controller app.  Future versions of this controller will allow the repeat function.

Unfortunately you cannot currently repeat a track with the user interface we provide.  In later version of the eLyric user interface and server program this will be possible.

eLyric Controller keeps a record of all the servers and libraries you have connected to in the past in order to help you remember what they looked like.  If they are active, you can easily touch the choice and select it on the touch screen.  If they are inactive, they are gray in color and cannot be selected.

If you wish to delete the inactive servers or libraries from the list, simple swipe your finger from left to right on the library and a delete button appears.  Press delete and the selection is deleted.

If your Bridge “loses” its DHCP operations (e.g., all zeros shown on the IP addresses on the IP setup page) you can restore it with the following procedure.

You Will Need:  A USB drive and a computer with internet connection.

First: download the file from the link below:
http://updates.psaudio.com/rescue/NetworkFix-dhcp.zip

Then unzip it to the USB drive and safely remove the USB drive from the computer using the ‘eject’ procedure.
Power the DAC off using the switch on rear of DAC.
Insert USB device into the USB port on the DAC.
Power on witht he rear switch.
Wait until DAC is compleatly booted (this will take a bit more than 30 seconds)
Power DAC off with the rear switch.
Remove the USB stick.
Power on DAC and it will now be set for DHCP.

You will need a 2 GB USB stick.

Make sure USB is a 2GB USB, smaller one might have issues and larger ones can not be formatted FAT on some windows machines.

First: Make sure the USB stick is formatted FAT.

Windows XP and Windows 7:

Method 1:
Insert your flash drive in other usb port. Go to control panel “System and Security” then “administrative tools”. Double click ‘computer management’. From left panel click ‘disk management’. You will see your flash drive here. Right click on it. Click format option. From file system choose ‘‘FAT”. Choose FAT32 if FAT is not available but never NTFS. From allocation unit size option choose ‘default’. Now do a full format not quick.

Method 2 (Windows 7)

Right-click on the USB drive in the left pane under where it says “Computer”  It will give you the option to click “Format”  Do so and select FAT from the pull-down menu.

Be sure and UNCHECK THE “QUICK FORMAT” Box

Unzip Rescue.zip to USB drive.

Get the Rescue Image from this URL:  (copy and paste into your browser)

http://updates.psaudio.com/rescue/Rescue.zip

Save the file to the USB drive and unzip it.

Make sure Rescue.ub is on the USB drive.
Power the DAC off with the switch on the back.
Insert USB into DAC.
Power DAC on with the switch on the back.
Wait 20 minutes or more as there is nothing on the display showing progress.  The display will look normal not blank.  Please be patient as this is working but doesn’t look like it.  Just wait the time.
Power off DAC
Remove USB
Power on DAC.

The Bridge should be back up and running.

You can now upgrade to the latest version.

Should you run into problems that require you to reset the factory settings for the Bridge, here are the simple steps to doing that.

1.  Power the PWD/Bridge off from the rear panel switch.  Then use the same rear panel switch to power the bridge on.

2.  Touch the front panel when it says initializing. The version screen should come up.

3.  When bridge has booted, a set defaults button will appear. (It may take 20 seconds for it to appear)

4.  Press the button to reset the bridge.

You will need: 1) A computer with ethernet connection to download the file and a USB thumbdrive or other means to store the file, and 2) A computer connected to the router network where the PWD and Bridge are connected.

First: Download the file.

On the computer where you can get internet access,  copy and paste this into your web browser:

http://updates.psaudio.com/bridge/upgrade-0.2.12.ub

Tell it to save the file

It will ask you where you want to save the file.  Give it a path (remember it for later on)  Once the download is complete you’ll see this screen:

When you’re done, open the folder.  If you haven’t already done so, transfer the .ub file to your USB thumb drive or other storage media.

OK, now you’re done with the internet.  Take the USB drive to the computer that is connected to the PWD/Bridge router.  Turn on the PWD and navigate to the Bridge Setup screen to get he IP address.  If you don’t remember, here’s how:

Press the “Home” button on the Bridge front panel.

Press the green button to the right of the “Media Bridge”

Press the green button under the network to find the IP address.

Locate the IP address and write the number down.

Hit the green check boxes in the lower right corner to get back to the main Bridge Screen

Exit from the next screen

Touch “Media Bridge” (blue button) to get back to the main Bridge screen.  Note: This is important for the firmware to load properly

Go to the computer and type the IP address into the web browser

You’ll get the PS Audio page

Press “Web Update”  You’ll get to this page:

Type in the pathway to the .ub file that’s on the USB drive or use the ‘Browse’ feature to find the file.  Point to the correct .ub file that was downloaded in the first step.

Hit ‘open’ and the file name will load into the box on the screen.

Hit “upgrade Firmware” and the system will load.  Do not touch anything while the upload is going on.

When done, a “Successful!” dialog box will appear.  Click the ‘OK’ and the PWD will reboot.

Hit “Back to home” and you’ll be back at the main screen

Press “Network Status” to verify the load was done properly.

And you’re done!  Congratulations!  If not being used, the computer can now be disconnected from the network used by the Bridge.

Edit:  The latest Bridge firmware is at  http://updates.psaudio.com/bridge/upgrade-0.2.12.ub .  These instructions were written for an older version but work with the newest updates.

As PS engineers add new features, functions or bug fixes for the Bridge, new firmware becomes available.   You can go from any firmware version to the latest with the touch of a button on the front panel touch screen of the PerfectWave DAC.   Here are the step-by-step instructions you need to know.

You will need : A uninterrupted ethernet link to the Bridge while the update takes place.

To see the firmware update button, when you’re in the Bridge screen on the PWD press the ‘Home’ button.  You’ll see the various inputs for the PWD.  You should see “Media Bridge” (or the name you gave the Bridge) in the middle of the three options.  On the right side of that you will see a green button.  Press it.  A new page appears, called “Bridge Setup”.  In the center of the setup page at the bottom you should see a green icon immediately to the right of the words “Firmware update”  Press the green button.

A green button will appear in place of the “N/A” shown here.  If you see the green button, press it.

Do not touch anything after you’ve activated the update .  Wait until the PWD reboots (it will take 10 to 20 minutes)

Alternate method (this is more intricate.  Please be careful):

You will need: A computer with uninterrupted internet access that’s connected to the same router as the PWD.  The PWDmust remain connected to the router and ethernet at all times.

1) As above, navigate to the “Bridge Setup” screen.  In the middle of the screen you’ll see a section called “Network Settings”  There are two, one for wireless (on the right) and the other for hardwire Ethernet (on the left)  Under the hardwire Ethernet symbol is a green button.  Press it.

2) A new screen appears, showing the Ethernet communications addresses. At the top of the screen you’ll see the words “IP Address” with some numbers in this format:  XXX.XXX.XXX.XXX

3) Write this number down.

4) Press the little green box in the lower right of the screen with the check mark in it.  You’ll go back to the “Bridge setup” screen.  There’s another one there. Push its green check mark button.  You’ll be back at the PWD Bridge input select screen.  Select the ‘Media Bridge’ and you’ll be at the Bridge operational screen where the album art is.  (This is important, if you don’t go back to the Bridge operational screen the update won’t happen)

5) Go to your computer and type in the IP address into the web browser.  Hit ‘enter.’  You’ll see a PS Audio Bridge web page pop up.

Click the link that says “Network Upgrade”  A new page appears called “Firmware Upgrade- Network.”

Copy the following and paste it into the box on the screen, NOT into the browser.

http://updates.psaudio.com/bridge/upgrade-0.2.12.ub

6) Press the “upgrade firmware” link.  Say ‘yes’ to all the questions.  Let it load.  This may take 10 to 20 minutes.  Do nothing with the browser until it completes.

7)  When everything works a small pop-up box will appear that says “Successful!”  Click on the ‘OK’ button.

The PWD will then reboot.  Wait for the reinitialization.  Go back to the Bridge operational screen and readjust the volume as desired.

9)  Press the “Back to Home” link on the screen or the ‘back’ arrow.  The browser will go back to the PS Audio Bridge Page.  Click on the link that says “Network Status”  You’ll get another page:

NOTE:  The Bridge Version Number WILL update to 0.2.12, the picture below is for a previous update, but you will see “0.2.12” when the update completes successfully.

Congratulations!  You’re done!

For first-time users:

The PSAudio Bridge, once installed in your PerfectWave DAC, is a wonderful system with many benefits.  Before you even buy the Bridge, you can set up your home wireless network and rip your music files if you haven’t already done so.  Please see http://www.psaudio.com/uploads/files/PerfectWave_Bridge_-_Quick_Start_Guide_v1.pdf

1) Rip your music.

You can use the program EAC (Exact Audio Copy) to rip your music.  We agree, it is somewhat complicated to initially set up.  However, once you do, it will faithfully read and re-read the discs over and over again until it makes sure that all the read-errors that crop up occasionally are eliminated.  Instructions for installing EAC can be found here:  http://www.psaudio.com/ps/how-to/how-to-set-up-eac-on-your-computer/

Some users have had good luck with using dbpoweramp for their rips.  See http://www.dbpoweramp.com/  There are different versions available to purchase, and you can select the functionality that you need.

Other users have been successful with using Media Monkey.  See http://www.mediamonkey.com/

You can also buy high resolution music from a number of sources.  See http://www.psaudio.com/ps/how-to/where-to-get-high-resolution-music/

Some users have DVD audio that can be ripped using the following instructions:  http://www.psaudio.com/ps/how-to/how-to-extract-2-channel-audio-files-from-dvds/

2) Install a wireless network router between the computer/NAS that will act as your server and the Bridge.  Please remember that the Bridge can not be linked with a CAT5 cable directly to the server, it must be connected via a router.  There are many different types of network routers available on the market today.  Please choose one and follow its setup instructions correctly.

If you want to have a wireless connection between your server and the Bridge, PSAudio likes the Linksys 610 Ethernet Bridge.  You can use these instructions on how to install it:  http://www.psaudio.com/ps/how-to/how-to-setup-a-linksys-610-ethernet-bridge/

3) Select a music server that you’d like to use and transfer your ripped music to the server.

There are many options.  You can use PSAudio’s eLyric, available at http://www.psaudio.com/ps/media_manager/downloads/    It can run on a PC or a MAC.  If you like, you can have eLyric autoload every time you turn your computer on.  See the instructions at http://www.psaudio.com/ps/how-to/how-to-have-elyric-autoload-when-you-start-your-computer/

James River Media Center is another useful program that runs on a PC.  See http://www.psaudio.com/ps/how-to/how-to-use-j-river-media-center-for-the-bridge/  It has the advantage that it can stream music to the Bridge directly without the need for an iDevice (if the latter happens to get lost at an inopportune time)   You can make playlsits in JRMC.  See http://www.psaudio.com/ps/how-to/how-to-create-playlists-in-james-river-media-center-15/

Asset by dbPoweramp is another program that can run on PC or MAC.  See http://www.psaudio.com/ps/how-to/how-to-set-up-a-dlna-server-asset-by-dbpoweramp/

Twonky is another program that runs on a PC.  Some have stated it is a bit more user-hostile than most.  See http://www.psaudio.com/ps/how-to/how-to-set-up-twonky-as-a-server/ and http://www.psaudio.com/ps/how-to/how-to-add-music-files-to-twonky-music-manager/

If you like to have a dedicated server without having to keep your PC turned on, you can install eLyric onto a $200 Habey fanless computer.  See http://www.psaudio.com/ps/how-to/how-to-build-a-low-cost-dedicated-elyric-server/

4) Set up TagNplay (if you decided to use an iDevice as your controller)  See the instructions in the Bridge Quick Start Guide at http://www.psaudio.com/uploads/files/PerfectWave_Bridge_-_Quick_Start_Guide_v1.pdf

For advanced users:

Your Bridge should be up and running by now.    Occasionally there will be firmware updates for the Bridge and PWD.  You can check which are the latest versions by going to http://www.psaudio.com/ps/forum/viewthread/1798/

How to update the PWD firmware are explained at:

http://www.psaudio.com/ps/how-to/how-to-update-perfectwave-firmware/

The Bridge firmware (separate from the PWD firmware) can be updated with the following methods:

http://www.psaudio.com/ps/how-to/how-to-update-your-bridge-firmware/

http://www.psaudio.com/ps/how-to/how-to-update-bridge-firmware-when-theres-no-internet-access-at-the-dac/

If the Bridge doesn’t seem to be working right, you can reset it with the following procedure:  http://www.psaudio.com/ps/how-to/how-to-reset-perfectwave-dac-bridge-factory-defaults/

Sometimes, if the power goes out during a Bridge update, the firmware will not have loaded properly and the Bridge won’t work at all.  If this happens, use the following procedure to revive the Bridge, then perform the Bridge firmware update again. http://www.psaudio.com/ps/how-to/how-to-recover-a-crashed-bridge-from-a-failed-firmware-upgrade/

Some users have experienced a dynamic IP address problem where the router assigns the same IP address to the Bridge and another device on the network.  If this happens to you, try and set a static IP address by following these instructions:  http://www.psaudio.com/ps/how-to/how-to-get-static-ip-address-from-ps-audio-when-tagnplay-doesnt-see-the-bri/ or http://www.psaudio.com/ps/how-to/alternate-method-how-to-get-data-to-find-a-static-ip-address/

Some users have experienced a problem where the router connectivity is lost and the Dynamic Host Configuration Protocol lost the connectivity to the router.  This can be fixed using this procedure:  http://www.psaudio.com/ps/how-to/how-to-restore-bridge-to-dynamic-host-configuration-protocol-dhcp-when-rout/

No, not using the Twonky server built into the NAS.  Gapless must be initiated on the server and Twonky does not support or play gapless.  To play gapless on the Bridge you will need to install eLyric server which can be downloaded for free at http://www.elyric.com .  You can then point the eLyric server at the QNAP NAS where all your music is stored.

DO NOT: Use a non-UPnP/DLNA music playback program like iTunes or MediaMonkey for the Bridge as it will not work.  They will work fine for ripping your CDs and getting the files onto the hard drive, but they WILL NOT stream music to the Bridge.

DO NOT: attach a standard  CAT 5 ethernet cable directly from the server to the Bridge.  Both the Bridge and the server need to be linked together via a router.  (Apparently there’s a way to do this with a cross-wired CAT 5 cable but I’ve not heard of anyone who’s actually got this to work, so unless you’re an expert at ethernet connections, please don’t try until someone can give us specific detailed instructions on how to make it work.)

DO: use a DLNA/UPnP server for your PC/MAC or a DLNA/UPnP NAS (But it needs a server program like Twonky installed int he NAS) Only these will stream the music to the Bridge properly

Installing the PS Audio Network Bridge is a snap.  Here’s a video demonstrating it.

httpv://youtu.be/mKLq8aFHekk

You can use either program or Twonky as a third.  The only requirement is you must use a UPnP based server (of which all three programs are).

The only downside of using program,s that are no eLyric Server is gapless.  The only program that will support gapless play on ther Bride is eLyric Music Manager which is available free at http://www.elyric.com

Yes, but you do need some type of controller.  J River media center, Twonky and Foobar all can act as a controller for the Bridge.  As for wireless, you will need to get a wireless Bridge to connect the PS Network Bridge to your network (your home’s router).

If the bridge does not appear in the eLyric Controller under devices, try the following:

Touch the refresh button on the controller above the list of possible players.

Reboot the Bridge by power cycling the PWD through the rear panel power switch.  After everything stabilizes and you verify the PWD touch screen show you connected to the network, wait 5 minutes and try again.

Make sure you are on the same wireless network as the bridge.  To do this, go to the iPod Settings area and check WIFI to see if you are on the same wireless network you expected.  Sometimes the Apple mobile device will switch automatically to a neighbor’s WIFI network unexpectedly.

These techniques should wake up the controller to see the Bridge.  If you still are having a problem and you can’t solve them, contact us for help.

Many times skips and small dropouts in the streaming process to the Bridge are caused by the computer that is serving the music.  It is rarely ever the Bridge hardware itself.  There are many ways to solve this, by making sure you are using a good server software program like eLyric  or switching to a faster computer running eLyric or J River server program.

NAS are particularly suspect as their internal CPU devices are notoriously slow and sometimes cannot keep up with the demands of streaming high resolution music like 192kHz 24 bit WAV files.

If you are running a Windows computer, here is a trick the more advanced computer savvy people can do to improve the speed and reduce or eliminate skipping.  Submitted from the PS Audio Community Forums.

“The skip problem can be solved by stopping the windows Demigrator service. I have two batch file to operate stop the service. Problem was the Demigrator service on my PC auto starts about every 15 minutes, so I have used scheduler to stop the service every 5 minutes if it happens to start up. I do let the service run for 4 hours in the early morning thanks to scheduler.

I have had no stability or other issues since Christmas after doing this.

File name one: DemigratorOff.bat

@echo off
sc stop “DriveExtenderMigrator”
sc config “DriveExtenderMigrator” start= disabled
exit

The Linksys WET610N is a recommended Ethernet Bridge that can wirelessly connect your home router to a PS Audio Bridge.

Chances are this device will plug in and simply work for you if you follow the instructions in the quick setup guide and wizard.

However, you may encounter issues with Linksys’ setup wizard -  even with the WET610N plugged directly into the router, a sister Linksys product (WRT610N), Linksys’ setup routine running on a Win 7 laptop did not detect the WET610N. If you experience this, try rebooting and if it still does not work, temporarily drop your firewall (make sure to re-enable it afterwards. J) This appears to be a known issue with Windows 7; by the time you read this a new version of the setup wizard may be available at http://homesupport.cisco.com/

If all else fails, the setup wizard should run without any problems from an XP or Vista machine or a MAC

It is also possible to set the unit up manually without running the wizard. To do so, take the following steps:

1. Ensure     that your DHCP server (this is usually your router but it may be a PC or     NAS on your network) has at least two available addresses to assign, as     you will need one for the WET610N and one for the PS Audio Bridge itself.)
2. Plug     the WET610N into a spare port on your router or switch
3. Check the     active address list in your router to find the IP address assigned to the     WET610N – each router is a little different with regard to where it shows     this but it is usually under “status.” Note the address. If your router     allows this, we recommend reserving this address as the permanent address     of the WET610N.
4. If     your router does not allow permanent address reservations, then make note     of the ‘scope”  -the range of     IP addresses your router can assign – so you can manually “hardcode” the     WET610N with an address outside this range but on the same subnet. For     example, if your scope is from 192.168.1.2 to 192.168.1.117, you can     assign the WET610N an address of 192.168.1.118. (Do not go past 255 in the     last number, and 1 is usually your router and therefore unavailable.)
5. To     continue, log in as http://xxx.xxx.xxx.xxx where the xxx’s are the IP address currently assigned to the WET610N by     your router (not the new number you want to assign, but the one the router     assigned it through DHCP.) Leave the user name blank and use ‘admin’ as     the password.
6. Once     you are logged in, immediately create a new user name/password from the Administration>Management     screen. Don’t forget to hit “Save Changes.” You will then be asked to     re-log in with your new password
7. To     manually add a hard-coded IP address, go to the Setup>Basic Setup     screen, change the address type from DHCP to Static IP address, and input     the information required. The subnet mask will be the same as that which     is used by other devices on your network, as will the gateway. (If you     don’t know this info, go to a command prompt on any Windows PC and type     “ipconfig /all” and hit Enter.)
8. Once     again, save the changes.
9. Before     attempting to connect to the WET610N over a wi-fi link, make sure you have     configured the WET610N to work with whatever kind of encryption your     router is using – do this at the Wireless>Basic Wireless Settings     screen. You will need the encryption type, password, and SSID of the     wireless network you want to connect to.
10. Log     out of the WET610N, unplug it from the router, and reboot it. You should     now be able to log into it over the wi-fi link (wait for a few minutes for     the link light on the WET610N to come on.) If you can’t, you will need to     connect it to your router again with a cable until you find what’s     mis-configured and correct it.
11. Continue     on to the general points below.

GENERAL SUGGESTIONS/INFO

1. Note     that the 5ghz band has less range than the 2.4ghz band. However, it is     much more likely to be free of traffic on adjacent channels from     neighbors, etc. Also, if you run your other network traffic on one of the     2.4ghz channels and reserve the 5ghz channel for only the WET610N, that     has the potential benefit of ensuring that your streaming over that link     won’t have to compete with other traffic. So, if your router supports the     5 ghz band, we recommend trying this. This can be configured in your     router; have the WET610N connect to the appropriate SSID of your 5 ghz     network.

2. It has     been our experience that WPA encryption has a slight impact on bandwidth     but a pretty significant impact on range. For this reason we have disabled     WPA on the SSID that we use for the WET610N and just use a simple MAC     address filter. You can see how strong your connection is and how much     bandwidth you have by going to the Status>Wireless Network screen. This     is a very helpful screen because it enables one to see the impact of     configuration changes, channel, and physical location of the     WET610N/router on available bandwidth. For example:  We have found that radio frequency     propagation in small enclosed spaces can be a funny thing. Moving either     the router or WET610N by even a few inches can sometimes make a huge     difference in the quality of the link and thus available bandwidth. We have     found that my bandwidth increases if the router is just a little bit off     the floor, and the WET610N, though ostensibly it has an omnidirectional     antenna, seems to work a little better if the front of the unit is tipped     up at a bit of an angle. Your mileage may vary  – experiment!

3. Under     the Adminstration screen, select the “Backup Configuration” button and     make a note of where you have saved the config file. This will save you in     case you subsequently screw something up and need to get back to the last     known good configuration (do this with your router too.)

4. In the     Wireless>WMM screen, we currently have “Best Effort” selected, whicourh in  network seems to provide the best bandwidth

5. In     your router, if Channel Width is configurable, we recommend leaving it set     to “wide” for the 5 ghz band, but again, your mileage may vary, so     experiment, and check the Status>Wireless Network screen in the WET610N     to see which setting seems to work best. Same with channel selection for     both the wide and narrow bands.

6. If you     happen to be using Audioengine send/receive units for remote speakers,    it’s not a bad idea to plug in the sender unit near the router for a few     seconds so it can detect what channel your wi-fi is running on and     self-configure to avoid interference.

7. Lastly     – you are now ready to connect the PS Audio Bridge to the WET610N. As with     the WET610N itself, I recommend reserving a permanent address in your     router for the PS Audio Bridge. If this cannot be done, if necessary you     can log into the Bridge by the IP address the router has assigned, and     click on the “Network Configuration” link to assign one manually. If you     do this, make sure it doesn’t conflict with the DHCP scope in your router!

You will need: A computer with internet access that’s connected to the same router as the PWD.

First, let’s get to the IP status page.  Press the ‘home’ button on the Bridge screen.

Next, touch the green button to the right of the words “Media Bridge”

You’ll get to the Bridge Setup screen:

In the middle of the screen you’ll see a section called “Network Settings”  There are two, one for wireless (on the right) and the other for hardwire Ethernet (on the left)  Under the hardwire Ethernet symbol is a green button.  Press it.

A new screen appears, showing the Ethernet communications addresses. At the top of the screen you’ll see the words “IP Address” with some numbers in this format:  XXX.XXX.XXX.XXX

Write this number down.

Press the little green box in the lower right of the screen with the check mark in it.  You’ll exit to the “Bridge setup” screen.  There’s another one there. Push its green check mark button.  You’ll be back at the PWD Bridge input select screen.  Select the ‘Media Bridge’ and you’ll be at the Bridge operational screen where the album art is.

Go to your computer and type in the IP address into the web browser.  Hit ‘enter.’  You’ll see a PS Audio Bridge web page pop up.

Click on the link that says “Network Status”  You’ll get another page:

Copy all this information down (or do a screen capture, or take a picture of it) and send it to Dennis at Dennis@psaudio.com.  This will get him started on you static IP address.  He’ll contact you on trying to find a static IP address that is outside the dynamic range of your router.  Because there are so many routers on the market, there’s not enough space on this page to show you how to work every one.  But please contact Dennis and he can work something out for you.

Best of luck!

Connecting your new PerfectWave Transport  to the internet is how you download song titles and cover art when you play a CD.  Accessing the internet is easy is you have a home wired for network connections: just simply plug the PWT into any internet capable Ethernet connection and the PWT (or the PowerPlay unit) does the rest automatically.  But some homes use only wireless to connect to the internet.  When this is the case, you need a wireless Ethernet Bridge (access point) to connect the PWT to your wireless router.

A wireless network access point (wireless Ethernet bridge) are all about the same thing: they can connect any device that has an ethernet connection, using wireless technology, to a wireless router.  You can read about our favorite bridge here:http://www.linksysbycisco.com/US/en/products/WET610n.  This one is about $100 retail.

To make this work, you need to have a wireless router in the first place.  Wireless routers are inexpensive and easy to setup and allow wireless access to your internet connnection throughout the home.  These routers are available at Best Buy, Staples, Off ice depot or over the internet.  Once connected and configured, you have access to the internet from any device that has WiFi capability (such as an iPod Touch, iPhone, laptop etc.)

If you wish to connect an ethernet capable device, such as the PerfectWave Transport to the internet, you will need to connect its Ethernet port to an internet capable connection, either through a CAT5 cable or through a wireless access point.

Using a wireless access point, or Ethernet Bridge (like the one pictured by Linksys) is easy.  Use a short CAT5 cable to connect the PWT to the Bridge.  Connect power to the Bridge and follow the setup instructions provided with the Linksys.  An overview of the setup process would show that once the Linksys device is powered up, it will immediately communicate with your wireless router automatically.

All you need to do at this point is help the two communicate.  To do this, you’ll insert the setup CD into your computer (the Linksys comes with this setup CD) and follow the few simple steps for setup using a wizard.

Once configured, the PWT will connect to the internet and you will see a “connected to the internet” affirmation on the top of the PWT touch screen.

If you have any problems, call us.

First, go to your “Start” button in the lower left corner and select the “Control Panel”

Select “System & Security”

Then select “System”

Then select the “Device Manager” in the upper left corner.

Select the network adapter that you use to transmit data to your Bridge. The small button on the left of the “Network Adapters” line will expand it for all the available networks. Shown here is a wired network. Select the wireless card if you use that.

Select the “Advanced” tab. There, select “Jumbo Frames” from the “Property” box, then on the right in the “value” box make sure it’s set to “Disabled”

Press “OK” and you’re done!

While EAC has agreeably got to be one of the more user-challenging installations, the benefits are a very accurate RIP.  The program actually takes into account the read characteristics of the optical drive, among other things, to ensure the bits stored onto the hard drive match the original recording as closely as possible.

First go to http://blowfish.be/eac/

All the following links are there on the main page, but here they are just in case you needed a quick link:

Installation:  http://blowfish.be/eac/Install/install1.html

Setup: http://blowfish.be/eac/Setup/setup1.html

Lossy Setup guide (for those that are interested): http://blowfish.be/eac/Lossy/lossy.html

Rip guide: http://blowfish.be/eac/Rip/rip1.html

CD burning guide: http://blowfish.be/eac/Burn/burn.html

FLAC .exe downloads if you want to have EAC compress the RIPped files immediately after the RIP so you don’t have to launch this on your own:  http://flac.sourceforge.net/download.html  Choose the platform format as needed.

Yes, it takes a lot of time to load this program, but the results are worth it!

Best of luck.

When sound pressure comes out from your loudspeaker, it is typically aimed directly at your listening position. When it arrives at your listening position, it is referred to as “direct sound”. Along the way towards that listening position, the sound pressure wave will hit the side walls, pieces of furniture, and so on, causing multiple reflections. The reflections are delayed slightly in their arrival to your listening position. These multiple delayed signals are referred to as reflections. Therefore, when the direct sound arrives at your listening position, it is quickly followed by the same sound, slightly delayed in time, causing a confusion to the music. If we were to play our speakers outside, this would not be a problem (unless it rained). Unfortunately, this is not the case, and we in fact listen inside. Therefore, we must deal with these reflections.

Since we cannot eliminate all reflections, we should concentrate our efforts on the biggest problems first. The biggest problems are known as “first reflections” because they are the first and most predominant reflections to arrive at our listening position. First reflections are almost always created when the sound coming from the loudspeaker hits the side wall (the walls closest to the left and right sides of the speakers). They are relatively easy to correct if you can identify where they occur. To minimize their effect on the sound of your system, you need to place either an object in their path to diffuse their energy (such as a piece of furniture), or an object in their path to absorb their energy (like a tapestry, curtain, or foam rubber).

Locating the point of first reflection is simple, but you will need the aid of a colleague and a small mirror. First, remove the grille cloth from your loudspeaker. Second, have your colleague hold the mirror against the approximate area of the first reflection on the side wall, holding the mirror at the same height as your tweeter is from the floor. Have the assistant move the mirror along the wall, while maintaining the correct height for the tweeter, until you can see the tweeter in the mirror. Mark that spot, and repeat the procedure on the opposite wall. The mirror has visually exposed the location of the first reflection

There’s no small amount of controversy surrounding this issue. That’s partly because some listeners aren’t particularly sensitive to the effect of inverted signal polarity, while for others, it’s very irritating. Since a number of manufacturers include a polarity reversal switch on their components, they obviously think that it’s worth addressing.

First, I should admit that I am in the latter camp. It always bothered me…

It was 1982. I was trying to set up a pair of Magneplanar T-1D loudspeakers. I had actually written the set-up instructions on these in 1976, back in my ARC/Magnepan days. So I was reasonably knowledgeable about how they should sound.

The LP I was listening to had a famous male performer (singer, writer, guitar player) on it. In fact, I happened to know at the time that this performer owned the very same model of speakers!

It was only his voice and his guitar on the recording. But I just couldn’t get the sound right. I could either put him in the room while he played a slightly muffled guitar, or, by swapping the ‘positive’ speaker connectors for the ‘negative’ on both speaker leads, I could make his guitar clear, but then it sounded as if he was singing with rocks in his mouth!

By this time, I had been wrestling with the occasional effects of inverted polarity for several years. I had become aware of it through my work doing freelance recordings for the local NPR affiliate and others.

But this LP recording was especially maddening. Apparently, the microphone

for the performer’s voice was wired with one polarity convention, but the pickup for the acoustic guitar was wired in reverse.

Or perhaps the microphones used different pin-outs for hot and return signal. Since some mics used pin 1 ground, pin 2 hot, pin 3 negative, and some mics swapped pins 2 and 3, it could have been “correct wires to incorrect mics.”

Before I say anything more about that situation, I should probably touch briefly on absolute polarity.

Since some advocates of the audibility of the concept can’t even agree on how best to describe it, I’ll go for my own simplistic description.

First, you should know that there are two places where the original event’s acoustic polarity could be inverted:

(1) Somewhere in the recording chain (microphones, cables, mixer boards, microphone preamps, etc).

(2) Somewhere in the playback system (a component may invert signal polarity, or it could be an incorrect speaker cable hook-up).

Playback of proper acoustic polarity in a recording results in a recorded compression waveform when reproduced in a playback (home) system that maintains that acoustic polarity.

Imagine saying ‘punch’ and projecting the word forcefully from your mouth. Exhaling. That’s compression, similar in a way to positive ‘absolute’ polarity.

Now, imagine saying the same word by pulling in the air as you say it. Inhaling. That’s rarefaction, similar in a way to inverted absolute polarity.

Unfortunately, systems that use loudspeakers that aren’t particularly phase/time coherent may not show much of a difference. And more often than not, this is the source for comments such as “I may hear a difference, but it’s no big deal” or even “I can’t hear any difference. It’s just an illusion.”

Not only is the effect of inverted absolute polarity audible, I actually ran a test in 1984 to verify that I could measure it on a home audio system I had installed. It was easily and repeatedly measurable at the listening seat.

In general, a system that is non-polarity inverting (a positive waveform on the recording produces a positive waveform into the room) will sound clearer with recordings that have positive absolute polarity. Bass will have more attack on the leading edge. It’ll be more defined. Vocals will be more ‘present’ on the soundstage. And treble will sound more ‘correct,’ not splashy and ill-defined.

And then, sometime after my “good singer, bad guitar” incident, there was great news. Someone finally wrote a book about absolute polarity!

In 1988, Modern Audio Association in Boston, Massachusetts, published Clark Johnsen’s book, The Wood Effect, about the effects of absolute polarity. There is still a limited quantity of the second printing available. You can get his book by e-mailing Clark at clarkjohnsen@gmail.com .

OK, why did I spend so much time on a phenomenon that many audiophiles and some reviewers ignore, as if it didn’t exist?

You need to know about the phenomenon in the event that you should inadvertently select two components to compare that happen to have opposite

absolute polarities.

Because the unfortunate result could be that you would make a mistake when evaluating a component, purely because it inverted absolute polarity when compared to another that did not.

Additionally, more than a few power amplifiers and preamplifiers have an odd number of gain stages, which will result in inverted polarity (in both channels of course).

I’m always amused when I read a review (especially if it’s a rave review) about a product that I KNOW inverts absolute polarity. Did it mean that the product is even better than the reviewer thought, or did it mean the reviewer is slightly incompetent? Or did he catch it, but not write about it?

Honestly speaking, this is an instance when purchasing an acoustic polarity indicator might be worth the expense even for a consumer who is considering an upgrade.

Jim Smith

This observation comes from over three decades of experience and hundreds (if not thousands) of successful critical installations.

You definitely want to address the first horizontal reflection from each front speaker. I’m assuming that in most installations that there is a nearby wall or other object(s) that can reflect sound from the side of your speaker. It’s not so much for correction of tonal balance (although it may be required if your speakers have uneven frequency response off axis), but it’s mostly to prevent smearing of the sound. The slightly late arrival of reflected sound will muddy your overall sound and affect your imaging.

Think of a stone dropped into a pool. The waves are like sound waves. If you drop a stone simultaneously near the edge of the pool, those waves will merge with and affect the original waves.

Some audiophiles are surprised to find that with proper room treatments, including absorption, recorded reverberation will be increased, not diminished! For example, the sound of a choir singing a cappella in a large space will sound more spacious when you absorb the unwanted speaker/room reflections than it would if played in a live room without treatment. That’s because unwanted room reflections are smearing and even covering the sound of the subtleties of recorded ambience, spaciousness, and acoustic delay.

(1) The side walls where the sound reflects from the speaker and then arrives at your ear.

(2) The area behind the listening seat.

(3) As many corners as possible.

If you determine that a secondary reflection from the other speaker is capable of reflecting at your seat, I’d consider addressing it as well.

There are two other places where room treatments can help, but they may not be WAF friendly:

(1) The first reflection on the ceiling from each speaker to your ear.

(2) If you have bare floors, the first reflection on the floor from each speaker to your ear. Of course, carpet or area rugs can take care of the floor bounce issue in the mid and high frequencies.

The first step is to locate the best position for your seating and for your subwoofers. The main issue, after finding where the subs work best, is getting them to blend seamlessly with your main speakers. This tip assumes that you’ve already located the best position for your main speakers and that you are voicing the subs for the overall combination of best blend and best bass.

Apart from voicing issues that have already been covered, there are two other difficulties that you are likely to encounter:

(1) The first difficulty can occur when the subs are simply too far behind the main speakers. That will introduce a time arrival/phase problem, especially through the crossover region. It manifests as thinness through that area, due to phase cancellation of the frequencies that are shared in the crossover region between the closer main speakers and the more distant subwoofers.

If, after going through the voicing procedures, there is still a lack of cohesion, you may need to modify the optimal position a bit so that the sub isn’t too far away from the main speakers. You definitely should experiment with polarity and crossover frequencies on the subs as was mentioned in Tip #35 . Many good subs have a 0-180-degree phase/polarity switch. The most useful of these types of controls actually have a continuously variable phase control.

(2) The second difficulty can occur when the crossover frequency is too low for vocals to be effective voicing tools. This also depends on the crossover slope (how steep the roll-off is for the subs and mains). For example, a third-order, 18 dB/octave slope is more abrupt than a 6 dB slope. With steeper crossover slopes at lower frequencies, vocals can simply be too high to be totally reliable, because the steeper (sharper) slope attenuates the vocal region too quickly.

Although I expressly mentioned using vocals for subwoofer voicing, in this case, if you’ve done all you can with phase/polarity, sub location distance (when compared to the main speakers), and vocals, but there’s a lack of palpability of instruments that have certain fundamental notes in the crossover region, I’d select a few recordings that featured solo instruments that play in that specific frequency region. Usually, I find cello pieces work the best. This issue is most often encountered with lower crossover frequencies (below 100Hz) and faster crossover slopes.

Play the passage you’ve selected, and, using the voicing info you’ve received

here, work on fine-tuning the transition between subs and main speakers. You want to see if you can get a bit more body on the cello.

When this is accomplished, you’ll need to go back to your vocal references to make sure that you didn’t go too far and now the sound is too thick.

To be sure, there’s a blend of art and science in this area. It’s sometimes a bit frustrating as you try to satisfy all conditions. Your room can play a MAJOR role in this area. Occasionally, you’ll have to give up the best location for the subwoofers, because, although they are playing great bass, the sound simply isn’t sufficiently seamless through the crossover region. In that case, you’ll have to reduce the subwoofer/main speaker distance differential to obtain the most seamless balance and “they are here” sonic presentation.

Once you get it right, the rewards are immense. Plus, you’ll have the satisfaction of knowing that YOU did it.

Jim Smith

If you have your speakers on the short wall, it’s more convenient to get up and take a few steps to the sidewall than to make the trek all the way over to components located between your speakers (unless your speakers are on the long wall and your components are on the short side wall). Plus, there are at least three significant performance advantages to locating your components on the sidewall, about half way down the length of the wall, if possible:

(1) Your components won’t interfere with the soundstage as much.

(2) There’ll be less visual distraction. Therefore, it’s easier to “suspend your disbelief” that there’s a musical event happening over there between the speakers.

(3) Most importantly, the point in the room where the bass is least strong is probably somewhere in this area. So sound waves will impinge less on the sound of your components, especially if they are vacuum tube, or if you have a turntable to play LPs.

Some will ask, “But what about the detrimental effect of longer speaker or interconnect cables?”

The overall advantage of locating your system where it will have less effect on the audio soundstage, where it will be less distracting visually, and locating it where airborne feedback cannot affect performance will ALWAYS offset any advantage to be gained from shorter cables.

Jim Smith

I don’t know any wives who want their husband’s music listening area to look like an over-damped hi-fi shop. By the way, WAF in audiophile-speak is an acronym for Wife Acceptance Factor.

Sometime ago, when working in more acoustically neutral living/listening spaces, I went away from simply damping the first sound wave reflection points in favor of absorption and diffusion. In my case, the treatment was strategically placed ficus trees (leafy trees). Silk trees seem to work very well. Of course they could be real, but some rooms have little available sunlight.

It’s important that they be leafy, especially at your listening height. They cannot be sparsely populated with leaves.

Using trees (real or silk) makes the room much more inviting. And the effect on the sound may be more satisfying than simply using deadening panels.

It may be more satisfying than using absorption panels because, all too often,too much absorption is introduced into the room with acoustic panels, even by installers who should know better. I’d still prefer acoustic panels in a dedicated room, IF they were the correct panels for the job, correctly placed. Here I’m suggesting a tree for those listening spaces that are also living spaces, serving multiple duties.

In a highly reflective room, leafy trees won’t work. I’m referring to rooms that already have a reasonably natural acoustic. Not too much echo, and not too dead.

I also employ them in corners wherever I can, as well as any other points that need treatment that would benefit from a nicer feeling listening environment. A tree will in no way take the place of a corner bass trap. But it can help to reduce corner-generated slap echo.

They work like a combination of diffusion and absorption. I also use them at shows, where you need instant flexibility and you can’t hang anything on hotel room walls.

Jim Smith

If your listening room is on a concrete slab with wall-to-wall carpeting, before spending lots more money on new components, the chances are that you can transform the sound of your system for a modest investment.

I first discovered this phenomenon when making master concert recordings. When we recorded a concert in a venue with concrete or other artificial flooring, the sound of instruments took on a colder sound.

Those same musicians, when playing at a venue with a wooden surface, made a wholly more musical sound. And they would comment on the live sound themselves. The difference is a slightly colder, less involving sound versus a warmer, more compelling sound.

All you need to do is put down some 2 x 4s on their sides and fire the nails that hold them in place into the concrete. Consider putting some insulation in the void between the concrete and the bottom of the wooden floor that gets screwed to the 2 x 4s.

You want to keep the solidity of the foundational concrete surface, but change its timbre. The floor doesn’t have to be exposed hardwood planks. It can be plywood sub flooring on which you reinstall your wall-to-wall carpeting.

Note: I’ve not found any preamp, amp, or cable that can make such a difference in tone!

Here is one caveat regarding wooden floors-if you already have a suspended wood floor, you may need to brace it from underneath. This is especially important if you can easily feel footsteps in the room.

If not supported firmly, the floor may act like a tympanic resonator, smearing the leading edge of notes and producing boomy bass. That’s why I strongly suggest that the 2 x 4 floor supports on the concrete surface be only 12 inches apart to support the new wood floor solidly.

Listeners are always surprised to hear greater dynamic range when the wood floor is suspended/supported properly. Greater dynamic range and warmer tone will yield greater musical involvement.

Jim Smith

Here’s a way to make both a sand AND air isolation platform for audio gear that costs less than $4.00, contributed by PS customer Steve Fellows.

Sand and air isolation platforms dampen vibrations in different ways. An air platform (Townsend Seismic Sink, Bright Star Air Mass, Vibraplane, etc.) dampens floor and airborne vibrations. A sand isolation platform neutralizes vibes that originate WITHIN the gear, such as from power transformers.

To make a sand isolation platform with “sand dampers” I purchased a quart-bag of white silica sand ($2.40) and a bag of eight, 12-inch balloons ($1.00). Black in color. Use a funnel to fill 4 balloons with sand until they are completely full, each being about half the volume of a tennis ball, then tie them shut. Slightly flatten each of the sand-filled balloons then place them under your component’s feet. With the sand dampers under my preamp the music is cleaner & less blurred than when the preamp sits directly on the rack’s shelf. Placing 4 sand dampers under my cd player made only a slight increase in sound quality. But I experienced a very noticeable benefit when air dampers were placed under my cd player. To make an “air damper,” hold open a balloon so that it fills with air on its own, that is, without blowing any air into it. Just hold it open & tie it shut. It may at first seem there’s not enough air in the balloon but the key is to not overfill it. Place 4 balloons, filled & tied, on an equipment shelf near the corners. Then place another shelf on top of the balloons, making sure the corners don’t touch any part of the rack. You now have a shelf suspended by air. (I haven’t yet done this using my 35-lb power amp as the 4 balloons probably wouldn’t support it. Perhaps having several of the balloons scattered in between the shelves may work. I’ll eventually try it.)

If you don’t have a second shelf the cd player (or preamp) can be placed directly on the 4 balloons. However, damping a component this way will require the gear to be used with a remote, otherwise using push buttons on the faceplate will cause the component to “jiggle” annoyingly. Not a good way to impress your audiophile buddies. But using a second shelf with the balloons sandwiched in between looks great & adds stability. I have about .33 inch of space between my two shelves with the 4 balloons in the middle.

Interestingly, when my preamp (Melos SHA-1) rests on the air dampers the music sounded worse than when placed directly on an undamped shelf: dull & uninvolving. Using the sand dampers described above greatly benefited the Melos. But the cd player’s performance improved significantly when placed on the air dampers than when sitting directly on the shelf or the sand dampers.

One final way to reduce vibrations on a component’s top plate is to fill a small zip-lock plastic bag with 3 or 4 tablespoons of sand. Place the bag on the component’s top plate (assuming there are no heat vents) and spread the sand evenly across within the flattened bag. Then cover the bag with a dark cloth to hide it.

So there you have it. Sand AND air isolation platforms for less than $4.00.

Placing a video subwoofer is sometimes a bit more challenging than an audio subwoofer because in a home theater setting, the system is typically required to be both a music as well as a video setup.

Where you place your subwoofer has as much bearing on how it will perform than just about anything else.

Never place a subwoofer in the middle of any wall or boundry. Certainly not in the middle of the room! This is a common mistake when you only have one subwoofer to add. Logically, if you only have one subwoofer to share between the left and right speakers, it be intuitive to place it between the left and right speakers so as to properly share the low frequency output of the sub.

Unfortunately, this is a big mistake because placed in the middle, room interactions will dramatically reduce the output level of the sub. It is far better to place the subwoofer to one side or the other. Typically, placement next to the right hand speaker is preferable if possible. This is because the bass section of an orchestra is traditionally on the right hand side as the audience faces the orchestra.

Very few large home theater systems can get by with a single subwoofer and two are preferred.

A common mistake made when adding a subwoofer to a video system is to duplicate the low pass filter, thus dramatically changing (for the worse) the sound and transient response of the sub.

Most surround processor have a separate subwoofer output that includes a crossover slope. That is, the sub output on a surround processor is already rolled off, containing no high frequency information. If you place this already rolled off subwoofer information into a subwoofer input that is not specifically designed for this type of output (most are not), you will then wind up with two roll-offs, giving you results that are less than optimal.

The easiest way to make sure this is not a problem, is to turn the subwoofer’s low pass filter control up as high as it will go. By turning the control to its highest frequency setting, only the roll-off built into the surround processor’s subwoofer output will be active and the two will not interfere with each other.

Some video processors allow you to turn off the internal roll-off of the processor allowing you to use the built in one on the subwoffer itself.

In most cases, however, it is easier and preferable to simply turn the roll-off of the subwoofer up as high as it will go, which is usually around 120 Hz.

Placement is critical as well

Here’s how to find out where the best place to put your subwoofer.  You can do this be swapping your listening position with your sub on a temporary basis.

Center your subwoofer in the location that you usually sit to listen, on top of a piece of furniture at your normal listening ear height. Then get down on your hands and knees around the sides and behind your main speakers. Keep your ear vertically the same height as your subwoofer would be on the floor.

Crawl around while listening to some test tones or bass-heavy music. When you find the location were the bass sounds the smoothest, that’s were you should put your sub. Hey! You just found the sweet spot for your sub!

You may need to play around with the direction that the sub is firing in the room to fine tune the sound after the subwoofer is placed in its sweet spot. And this will save your back from consistently moving the sub in 2inch increments…. you can just move your ear!

Adding a subwoofer to your audio system can be a great improvement because the vast majority of loudspeakers cannot reproduce low frequencies properly.  In fact, despite the measurement data supplied by most loudspeaker manufacturers, very few full range loudspeaker are truly full range unless they have either multiple large area woofers or a built in powered subwoofer.  The reason the measurements don’t match up with the speaker’s performance in a room has to do with proper coupling of the air to the woofer.

When you read a specification of a loudspeaker’s bass performance, they are typically measured at 1 meter (3 feet).  This close range measures the output of the speaker but not how well it couples the air in the room and brings bass to your listening position.  To properly couple the air in the room at bass frequencies, you need either a large woofer area or lots of power moving a smaller woofer a long way or both.  In short, you need a subwoofer if you want to hear everything that was recorded on the disc.

There are, however, a few rules to follow regarding subwoofer placement that can make a huge difference in the subwoofer’s performance.

Never place a subwoofer in the middle of any wall or boundry. Certainly not in the middle of the room! This is a common mistake when you only have one subwoofer to add. Logically, if you only have one subwoofer to share between the left and right speakers, it might make sense to place it between the left and right speakers so as to properly share the low frequency output of the sub. Unfortunately, this is a big mistake because placed in the middle, room interactions will dramatically reduce the output level of the sub. It is far better to place the subwoofer to one side or the other. Typically, placement next to the right hand speaker is preferable if possible. This is because the bass section of an orchestra is traditionally on the right hand side as the audience faces the orchestra.

If the subwoofer’s volume control is turned all the way up, and you still do not have enough output to satisfy your craving for bass, try placing it in a corner. This trick works because the corner of your room can act as a horn to acoustically amplify the output of the subwoofer. In fact, this is what we will be doing in the PS Audio listening room. You can find more information about the PS listening room and our subwoofer placement by going here.

Here’s how to find out where the best place to put your subwoofer.  You can do this be swapping your listening position with your sub on a temporary basis.

Center your subwoofer in the location that you usually sit to listen, on top of a piece of furniture at your normal listening ear height. Then get down on your hands and knees around the sides and behind your main speakers. Keep your ear vertically the same height as your subwoofer would be on the floor.

Crawl around while listening to some test tones or bass-heavy music. When you find the location were the bass sounds the smoothest, that’s were you should put your sub. Hey! You just found the sweet spot for your sub!

You may need to play around with the direction that the sub is firing in the room to fine tune the sound after the subwoofer is placed in its sweet spot. And this will save your back from consistently moving the sub in 2inch increments…. you can just move your ear!

This is a modular design that can hold as many pieces of equipment as you have. In this particular project, I built the rack to accommodate five of my Ultimate Outlet power conditioners, but you can adapt this for anything you’d like to isolate.

You can use any type of wood and tubing to coordinate with your existing furniture (which will make the wife happy). I used birch and stained it a Medium Cherry color.

To Assemble:

Cut your wood into 6 1/2″ x 7″ pieces

1. Clamp the cut pieces together
2. Mark the corners to be drilled (I used a 1/2″ drill bit)
3. After the pieces are drilled, take 4 pieces of all-thread and slide one washer and one nut onto the all-thread. Make sure there are only about three threads showing from the bottom of the all-thread to the nut.
4. Take the four pieces of all thread and slide it through each of the four holes in the first piece of wood and set so the all-thread rods are facing upwards.
5. Cut your tubing into pieces 3 1/2″ long, four per shelf.
6. Slide four pieces of the tubing onto the all-thread.
7. Take your next piece of wood and slide it onto the all-thread. Repeat steps 7 & 8 for however many shelves you need.
8. After the last piece of wood is on the all-thread cut the excess all-thread off so you only have about three threads showing. Be careful not to damage the threads when you cut them.
9. Take a washer and a decorative nut (like an acorn nut) and semi-tighten.
10. Align the pieces of tubing so they are straight and then tighten a bit more. When you are happy with the alignment completely tighten all the nuts.

If you choose to label the Ultimate Outlets a P-Touch type labler works very well.

What you will need:

* Enough of your chosen wood to make as many 6 1/2 x 7 inch shelves as needed.     * Four 24″ x 1/8″ pieces of all-thread Eight washers (4 for the top, 4 for the bottom)
* Four 24″ pieces of 3/8″ tubing of your choice
* Four nuts to fit the all-thread (for the bottom)
* Four Acorn nuts (for the top)
* Wood stain
* 2 1/2″ wood screws for each Ultimate Outlet

Submitted by:

Jeff Whitlock

You may not have this problem, but you might know someone who does! As the guy who wrote the owner’s manual for the ARC/Magnepan Tympani 1Ds, I thought I pretty much knew it all when it came to installing these speakers and getting the most from them in any room.

That’s why I’ll never forget the humbling lesson I learned in my lofty role as an audio “guru.” Here’s what happened:

As a high-end dealer in the early ’80s, I had sold a pair of Magneplanar Tympani 1D loudspeakers for use in an Audio Research system. I went out to install the system (which I always insisted on-and your dealer should do it for you as well).

I always carried along a 1/3-octave real time analyzer so that I could quickly see where the bass standing wave problems were located in the listening area of the room.

In less than an hour, I had found the best place to locate the speakers (somewhat tricky due to the negative bass waves emanating from the rear of the speaker), and the best place for the listening seat.

Using my basic 3-step installation technique in Tips #74-77 , plus the information outlined in Tips #78-89 , I worked to get the sound to where I’d be proud to send a prospect over to my client’s house to hear what the Maggies sounded like in a home.

As I left, my client was effusively thanking me for getting him better sound than he thought was possible. I was a hero.

A week later, he called me to complain about a “thickness in the midbass.” I don’t know if you are old enough to remember the Tympani 1Ds, but bass definition and timbre were their best qualities (Harry Pearson, writing in The Absolute Sound , adopted the T-1D as the bass unit for his soon-to-become-famous no-holds-barred hybrid Infinity QRS/Tympani 1D system).

There was NO WAY we could have a thickness in the bass! My client must have changed some component or something (in other words, I was convinced it couldn’t be my installation…).

Well, I finally stopped by later that day, expecting to point out the offending component. But nothing was changed in the system. And boy, was the upper bass thick! I got out my trusty RTA.

Sure enough, there was at least a 6 dB peak at about 125 Hz. Where did this come from? Well, I couldn’t figure it out, but as I moved the RTA about two or three feet in front of the listening position I’d selected (and even marked!), the bass peak gradually disappeared.

So we moved the seat forward and, just to be sure, listened to hear what

the guru (me) had fixed. Oh, no, the peak was back!

I measured the response behind the listening seat where we had originally determined was the best seat in the house (literally). Now the peak was almost gone!

Anyway, I started to think I was on Candid Camera . I was looking so foolish. Then I noticed it. My client had a new sofa. When I had set the system up, we had used an occasional chair for the listening/voicing sessions.

This sofa had a tightly stretched back panel (leather/leatherette). It was stretched so tightly, it produced its own tympanic sympathetic resonances at 125 Hz. Removing the sofa solved the mystery.

So check out any system where the seating could cause a similar effect. I’m still surprised at how many systems can be affected.

Believe it or not, your system is very sensitive to vibrations caused by your loudspeakers at moderate to high levels.

How can this be? Some of it is obvious: vibrations can be easily transmitted to our source mediums because they are mechanical devices, turntables, and CD players. But some elements of our audio/video systems are not so obvious, yet they are affected as well: tubes, capacitors, and even transistors.

The easiest way to isolate your equipment from these vibrations comes in two forms: adding weights so as to damp the vibrations, and isolation so as to remove the mechanical conductivity of the vibration itself.

In this section, we will suggest a few ways you can lower the vibrational levels affecting your equipment.

Adding weight to your units can be as simple as placing a brick on top of the unit. While this isn’t very elegant, it sure works. Try it and see what difference you hear.

Isolation can be achieved for very little money by using a thick rubber mat beneath your equipment, or a little fancier version is available from a number of vendors and uses materials like Sorbathane. These are very effective. Tip Toes, or pointed machined feet can also help a great deal.

If you would like to purchase your vibration reducing solution, a good source is the Audio Advisor. http://www.audioadvisor.com

The biggest problem in any room is getting the bass right.  Most speakers are capable of producing reasonable bass but there is no passive loudspeaker built that can claim true full range sound.  To reproduce full range audio in your room you will need a subwoofer; but placement of that subwoofer can be daunting.  Room modes, standing waves all get in the way of great bass.  Here’s a trick that the pros use that can make your life easy.  Instead of guessing where the sub should go, take a different tack.

Place the subwoofer in the listening position (even on the couch or chair if you can) and then move yourself around the area where the loudspeakers are to find the best sounding bass position.  You can simply walk around the area where your speakers are positioned or even crawl on the floor until you hear the sweet spot.   Then all you need to do is place the subwoofer in the new position and when you sit in your listening area, it will sound perfect.

You may not have this problem, but you might know someone who does! As the guy who wrote the owner’s manual for the ARC/Magnepan Tympani 1Ds, I thought I pretty much knew it all when it came to installing these speakers and getting the most from them in any room.

That’s why I’ll never forget the humbling lesson I learned in my lofty role as an audio “guru.” Here’s what happened:

As a high-end dealer in the early ’80s, I had sold a pair of Magneplanar Tympani 1D loudspeakers for use in an Audio Research system. I went out to install the system (which I always insisted on-and your dealer should do it for you as well).

I always carried along a 1/3-octave real time analyzer so that I could quickly see where the bass standing wave problems were located in the listening area of the room.

In less than an hour, I had found the best place to locate the speakers (somewhat tricky due to the negative bass waves emanating from the rear of the speaker), and the best place for the listening seat.

Using my basic 3-step installation technique in Tips #74-77 , plus the information outlined in Tips #78-89 , I worked to get the sound to where I’d be proud to send a prospect over to my client’s house to hear what the Maggies sounded like in a home.

As I left, my client was effusively thanking me for getting him better sound than he thought was possible. I was a hero.

A week later, he called me to complain about a “thickness in the midbass.” I don’t know if you are old enough to remember the Tympani 1Ds, but bass definition and timbre were their best qualities (Harry Pearson, writing in The Absolute Sound , adopted the T-1D as the bass unit for his soon-to-become-famous no-holds-barred hybrid Infinity QRS/Tympani 1D system).

There was NO WAY we could have a thickness in the bass! My client must have changed some component or something (in other words, I was convinced it couldn’t be my installation…).

Well, I finally stopped by later that day, expecting to point out the offending component. But nothing was changed in the system. And boy, was the upper bass thick! I got out my trusty RTA.

Sure enough, there was at least a 6 dB peak at about 125 Hz. Where did this come from? Well, I couldn’t figure it out, but as I moved the RTA about two or three feet in front of the listening position I’d selected (and even marked!), the bass peak gradually disappeared.

So we moved the seat forward and, just to be sure, listened to hear what

the guru (me) had fixed. Oh, no, the peak was back!

I measured the response behind the listening seat where we had originally determined was the best seat in the house (literally). Now the peak was almost gone!

Anyway, I started to think I was on Candid Camera . I was looking so foolish. Then I noticed it. My client had a new sofa. When I had set the system up, we had used an occasional chair for the listening/voicing sessions.

This sofa had a tightly stretched back panel (leather/leatherette). It was stretched so tightly, it produced its own tympanic sympathetic resonances at 125 Hz. Removing the sofa solved the mystery.

So check out any system where the seating could cause a similar effect. I’m still surprised at how many systems can be affected.

Jim Smith

There are some basic listening room sizes that work well. You can find them on the Internet. But it’s also possible to build rooms that have been designed on some mythological dimensions that have issues in the bass.

Here’s a dimension for a medium-to-small listening room that has relatively

smooth bass for a room of its size: 19′ L x 14′ W x 10′ H. These dimensions, when entered into a room acoustics spreadsheet, work out amazingly well. Fortunately, I’d heard somewhere about these dimensions and built several listening rooms this size before I knew about any of the spreadsheet programs. Without exception, they worked well for rooms in this size category.

You want rooms that have the fewest adjacent standing wave issues as possible. I’m referring to the “boundary effect” region (roughly from 300 Hz down to the lowest audible frequency in your room).

If you’re thinking of building your own listening room, I recommend that you locate a sound contractor who has a track record of success. It’s a good idea to arrange to listen in a room that he has designed, if possible. I do especially recommend Rives Audio.http://www.rivesaudio.com/

Of course, there are many other variables to consider, including such things as air handling and electrical service. If you have no luck finding a contractor, you can contact PS Audio, and, within certain time constraints, we can make some useful suggestions.

Jim Smith

Why you should be sour on a wide “Sweet Spot” for two channel playback by Jim Smith.

A wide sweet spot is almost like having your own harmonic distortion generator! There’s simply no way a serious listener should be satisfied to sit more than a foot away from the ‘equal path length intersection’ (center point) of sound from a pair of loudspeakers. Inter-channel phase and timing information has just been badly compromised, destroying instrumental timbres.

How is it that Audiophiles will accept only phase and time-aligned loudspeakers and then expect to sit off the acoustic center-point, totally destroying the inter-channel phase/time information? Look at it this way…..First, since you probably know this stuff, please forgive the simplified averaged wavelengths, but for purposes of illustration, let’s assume that a 1100 Hz tone (or harmonic) has a length of about 12 inches. Then 550 Hz is almost 2 feet in length (from the top of the sound-wave crest to the top of the next). And 2 kHz is almost 6 inches in length, 4 kHz is 3 inches, etc.

Now imagine that a female vocalist is recorded with her image centrally located in the stereo stage. If you sit two feet off center, that means that any fundamental notes and their harmonics from at least 500 Hz and above have been altered, some dramatically, some slightly.

This is audible, and it’s depressingly measurable! Before we examine the disastrous effects, let’s look at what’s happened to cause the problem…

But wait a minute! What about imaging?
OK, let’s say that now you’re about a foot closer to the left speaker than you are to the right one. Imagine a centrally recorded image that is reproduced at equal volume (amplitude) from both speakers in order to give the illusion of a precise center image.

Without going too far into recording techniques or speaker dispersion patterns, a panned mono center image (such as is produced in a studio) may appear to have shifted left somewhat. While a center image recorded from a stereo pair of mics seems to ‘stay put’ a little better. But these are phantom images at best, lacking in the ultimate richness of tone and body. Here’s why…

It’s not the potential ‘image wander’ that’s troublesome. It’s the harmonic distortion! (Technically, it’s not distortion, but the alteration of harmonic relationships.) The positive cycle (top of the wave crest) of a 1000 Hz overtone arrives at your ear from the (closer) left speaker before it does from the right one. There’ll be an audible – and very measurable – change at that frequency (or harmonic overtone).

Should the distance be equivalent to a half-wavelength further (6 inches), then that particular overtone (harmonic) will arrive exactly out of phase. And you know how your stereo plays less bass when the speakers are out of phase? Well, the effect is exactly the same – a reduction in level at that particular frequency.

Avoid a large sweet spot

Why is this important? You’ve heard of voiceprints. That’s where a recording of your voice can be used to positively identify you, no matter how hard you try to shift the sound of your voice.

How does it work? The unique relationships of vocal overtones are different for each voice. For example, the first harmonic may be 87.3% of the fundamental, the second just 48.1%, the third 54.7%, etc.

The exact relationship of these overtones (their relative strength, compared to the fundamental) is the identifying ‘genetic code’ of your voice. Well it turns out that all instruments and voices have their own particular set of harmonic ratios.

That’s how we know to differentiate two different instruments that are playing exactly the same note – say A (440 Hz). And that’s how an original Guarneri will be chosen over a ‘replica’ – it’s all in the ‘tone’- which is actually the harmonic – or overtone structure.

So, if you’re sitting where the path lengths are significantly unequal from the left and right speakers, you are absolutely guaranteed to hear wild shifts in the harmonics, meaning that an instrument or voice will not sound exactly as it should. This is not just some subjective acoustic theory. It’s not only audible; it’s also measurable in your room at your listening seat!

The sad fact is, you’ve just altered your system’s harmonic relationships. So why did you buy all that stuff with ‘vanishingly low distortion’ if you’re going to introduce a far worse version by not sitting in the center point where the path lengths are equal? Incidentally, this is an incontrovertible law of physics that is part of the good – and the bad – of stereophony.

From a perfectionist’s standpoint, it doesn’t matter if your loudspeakers produce a smooth response off-axis. The varying wavelengths at a listening position off the acoustic center will always produce uneven response on centrally recorded images (actually all images, but it’s easier to think about the centrally recorded image for the purposes of illustration).

Here’s a simple test for you. Put on a Sheffield or other disc that contains pink noise in both channels (pink noise is best, because it contains equal energy per octave, just like music). If you can, put your preamp in mono.

Stereo or mono, what you want is equal amplitude in each channel. Now, from the center position, slowly move your head to the left or right. That huge change in mid/treble tonal balance is exactly what happens if you sit off axis.

And because the wavelengths vary according to frequency, the varying time arrivals of harmonics also produce an unpredictable cancellation effect (well, it is predictable in that it’s never a good thing). And a ‘wide sweet spot’ isn’t really so sweet…

Now that we’ve told it like it is, let’s also admit to having absolutely wonderful experiences listening to music while others have occupied the best seat. If a system has dynamics, if it’s effortless, if it at least starts out being pretty accurate timbrally, then it can be quite listenable off-axis. Just remember that the phantom image produced off-axis in stereo is only an approximation.

Sweet, it ain’t!

Most audiophiles know that – by aiming a loudspeaker a bit off-axis (perhaps to crossfire behind you a foot or two) – they can take the ‘edge’ off the sound.  Especially when compared to aiming the speakers directly at the primary listening seat.  Here’s a method of fine tuning your system’s tonal balance with a few minutes of speaker setup, by Jim Smith.

For some loudspeakers, aiming straight ahead results in the best overall frequency balance.  Generally, when this is the case, the manufacturer or their dealer will make a point of advising you of this.

But did you know that stereo separation – often a matter of a few inches – can make a significant difference in perceived ‘warmth?’ Most audiophiles would suggest moving the speakers a bit further apart (to get closer to sidewalls for bass reinforcement) as a way of warming up the sound.

But 30 years of experience contradicts this idea. Getting a bit closer to the sidewalls may add more bass (and more unpleasant reflections), but the overall sound often gets thinner.

Actually, if your sound is a bit thin, and you’d like a bit more fullness, mid-range body, or warmth,
the best way is often to bring your speakers a few inches closer together. I’ve encountered situations where only an inch or so toward the center gave me the balance I was looking for.

Of course, when you do this, it changes your speakers’ toe-in slightly.  Now they’ll be aimed more to the center, so if you’ve already picked the best angle of your speakers for toe-in, you’ll need to toe them out just slightly to accommodate for the move. And you may just find that you don’t need the speakers toed as far off-axis as you thought when you originally settled on the toe-in

After doing this for hundreds (maybe thousands) of people, I was still baffled as to why this subtle adjustment in separation should do what it did.

About 18 years ago, I observed the effect when experimenting with spaced omni microphones. This was when I was engineering on-location symphonic recordings (where an inch or two difference in separation could yield a warmer or cooler sound).

From that experience I at least developed a theory about what’s happening. We perceive warm or cool sound to some extent by the amount of energy present in the lower mid-range/upper bass. Well, the wavelengths of these frequencies are fairly long, say 2′-6′ in length. If we bring our speakers a bit closer together, the reproduced sound ‘couples’ ever so slightly better, slightly shifting the sonic temperature to ‘warmer.’

Whether the theory is correct or not (and I sure don’t know), I can guarantee that you definitely can change the balance of your system with subtle changes in loudspeaker separation and toe-in.  And the same change often renders more presence as well.

Next is another related observation, that some may not agree with at all.  In this case, it definitely comes down to taste.  But hey, we’re talking about mine… 

In the past 30+ years, I’ve visited countless manufacturers, reviewers, knowledgeable audiophiles, musicans, etc.  I’ve also listened to hundred of systems in dealer showrooms and at various Hi-Fi shows.

On more than one occasion I’ve felt it necessary to ever-so-politely point out an alternative loudspeaker set-up.  I don’t mean overall seating and loudspeaker placement (we’ll get to that soon in this series).  No, I’m still staying with this month’s topic of tuning or voicing with stereo separation.

Personally, I find that a lot of pretty darn smart audiophiles go for pin-point stereo imaging. You know what kind of sound I mean.  The stereo image is displayed precisely across the room, almost in tiny little pin-points of sound…

Instruments take up their own definite little space on the soundstage.  You may very well have your system set up that way right now. And that’s fine, if that effect is your ultimate goal.

The thing is, I find that the very wide separation required to get that sort of imaging is usually too great, in the sense that it thins out the tonal palette of the music.  I know you’ve experienced or at least read this, but we NEVER get that pin-point imaging in any normal seats at a live concert.  So what’s most important to render a musical event so compelling that you are touched by the music?

For me, and for a growing number of music lovers, after dynamics, it’s presence and TONE.  And you can make an orchestra (or band or any vocals and/or instruments) sound bleached out, thin, and totally uninteresting by going too far in your quest for pin-point imaging.

Sometimes a very subtle adjustment back from pin-point can get you the best of both worlds.  But I’d rather have great dynamics, presence and tone from a mono system that tiny little pin-points of sound spread across my room that are anything but rich and engaging.  There, I said it!

Stereophony depends on precise time arrival from each channel to a centrally located listening position. So, as you move your speakers around in the room, you’ll need a temporary floor grid to keep from arriving at erroneous conclusions. Here’s an accurate technique from Jim Smith:

Find the position (from side to side) where you’ll be sitting (we’ll discover the front-to-back position in Part 4). If your room is symmetrical, and you plan to sit in the middle, measure how wide the room is; divide that measurement by two, and that’ll be your centerline.

I recommend you use tape that can be lifted easily from carpet or flooring. Lay down your tape on a line between the general speaker placement area to the general listening area. You’ll have to move the seating furniture out of the way while you’re installing your grid.

If your room isn’t symmetrical, decide on how far you think you’ll want to sit from the nearest sidewall. Mark it on the floor (assuming your seating has already been removed). Measure the distance.
Now, at the speaker end of the room, mark the floor at the same distance from the sidewall. This is your centerline, from the listening area to the speaker area.

At the speaker end of the room, mark the floor every six inches or so away from the centerline, beginning in the area where you generally suspect the speakers may be placed.

Now, swing your tape measure or laser measurement device from the general listening position on the centerline so that it crosses the speaker spacing marks you’ve just laid down.  Place some marks on the floor that represent about six inch intervals – varying distances from the proposed sitting area.

It doesn’t matter if you don’t end up sitting there; we only need an equal path length from the centerline to aid us in moving the speakers repeatedly and equally.

At the end of the next part, you’ll be making some fine adjustments in speaker position, with considerably less than six-inch increments. At this final point, you’ll need to recheck your measurements from your centerline for final positioning.

But for now, we want to establish a starting reference to build our foundation.

This three-step technique will get you to a satisfying sound faster than any other system we’ve seen.  It is written by Jim Smith.

The three steps must be followed in this order.

1. Bass

2. Image

3. Frequency response/tonal balance

OK, what do we do at each step?
First, you’ve got to get the bass generally pretty good. This means that if you have a full-range speaker, it should reproduce the deepest bass with the greatest smoothness.

Why does the bass come first?
Until you know how far away you’ll be sitting (speaker position and listening position), how can you proceed to step 2, getting the best stereo image? And we’ve seen that we can make some adjustments in the overall frequency balance with subtle changes in position (separation and toe-in). But first we’ve got to at least establish the distance to the speakers from the listening seat before we can begin to decide how far apart we want our speakers.

(1) The best bass—a throwback to early TVs.

Here’s how long I’ve been teaching this technique for getting the best bass…

I started out using a TV analogy that asked the installer to compare this step to the tuning methods from TVs of the ‘60s and early ‘70s! Those TVs had a “fine tuning” knob and “channel selector” switch. Here’s the analogy: Finding the best placement for the speaker in the room is a bit like fine tuning for best reception. But finding the best place to locate the listening seat is a bit like using the channel selector!

In other words, the most important consideration (whenever possible) is to discover where in the room you should sit to take advantage of the least negative room interactions (obvious peaks and dips in the bass), and the most positive room interactions (the most extension and attack without annoying overhang).

This is because your room will have obvious standing waves developing in the bass region (we’ll call this region 25 Hz to 250 Hz). These standing waves are very measurable and they are quite audible as resonances or ‘suck-outs.‘ They exist due to your room’s particular geometry.

Moving a speaker forward and back in the room can make a noticeable difference in the bass. But moving the seat forward and back the same distance in an average room will result in much more dramatic differences in bass performance.

Although these resonant room frequencies can be considered axially, tangentially, and obliquely, our primary concern here is with axial. These consist of destructive (to varying degrees) waves and constructive (to varying degrees) waves. A destructive standing wave is produced when two or more wavelengths meet at a point in a room, and—due to the time arrival of these waves—some will arrive slightly (or even directly) out of phase with others.

The varying time arrival is based primarily on room geometry (for example—length of the room vs. height). This results in a cancellation of frequencies at that particular point in the room. Destructive standing waves produce dips in frequency response. Conversely, constructive standing waves can produce peaks in response.

The following system assumes you’ve placed the speakers in a generally acceptable position in the room:
A simple way to prove this theory is to put on a CD recording with a repetitive bass line (preferably the upright acoustic bass—maybe Ray Brown). You’ll need to move your listening seat out of the way, perhaps to the side of the room.

A note on the recording – select a piece that plays bass notes up and down the scale.  While it’s playing on repeat (that’s why I chose CD as the source), walk back and forth slowly through the larger proposed listening area. You’ll notice dramatic differences in bass quality and quantity in a space of +/– 2-3 feet. Listen closer and you’ll find the smaller ‘window’ of acceptability for that particular bass line.

Once you’ve found the best spot to locate the seat (again, only for that particular series of bass notes), you’ll notice that moving the speaker forward and back an equivalent difference makes much less of a difference. This is all to say that your room resonances are going to be pretty much the same for most likely speaker placements, so find out where in the listening end of the room these resonances are least objectionable, and that’s where you’ll sit.

A quick note on finding the overall best bass listening position in a room—the quickest way to do it is with a real-time analyzer, preferably 1/3 octave. You’ll need to use pink noise as your source, set on the slowest filter, using flat or C-weighting. You’ll need to run its SPL level at least 20dB over the room noise floor, so as to avoid any unrecognized interference with your measurements.

You’re only looking at the region from around 25 Hz up to about 250 Hz. You’ll notice immediately that fairly small movements forward and back in the room are very obvious on the display as the various peaks and dips become quite easy to see.

Don’t have a RTA sitting around? Try to borrow or rent one for a few hours. By the way, I do NOT recommend using a Radio Shack SPL meter and test tones for this procedure. Actually, if the tones were 1/3 octave pink noise bands, it could perhaps work, but unfortunately the Radio Shack meter simply isn’t very accurate in its frequency response.  And the RS mic suffers from proximity effect, unlike an omni (which is the preferred pick up pattern for a microphone in this application).

You could use the test disc and your ears, though! Can’t find a RTA, or you’re uncomfortable technically with the idea of looking at how your room behaves in the bass resonance region?

Then find several recordings of music representative of the stuff you like to hear, and adjust for the best bass by listening while moving back and forth in the general listening position. What you’re listening for is the majority of the bass reproduced with the notes neither emphasized nor diminished. You’re also listening for the deepest bass. But sometimes the price for getting the deepest bass in an average room is an uneven bass response in the region where most bass notes occur. This is where you’ll have to pick the best compromise to your ears.

At this point, you’ll find that a difference of six inches or less forward or back will usually present you with a choice of the bass compromise you prefer. Once you’ve discovered this listening position that is least affected by room resonances, mark this spot (or at least measure how far it is to the wall behind you and write it down).

Now you can play with fine-tuning where the speakers go to make the bass better. Once you have that position to the point that is best to your ears, you’ll need to recheck the listening position a bit to make sure that a slight distance forward or back isn’t necessary now.

(2) Imaging and the X-files
Finally you’ve established—pretty closely—where you’ll be, and approximately how far away the speakers will be. Once you know ‘X,‘ you can start to work on ‘Y.‘ X is the distance from your ear to the plane of the tweeter (should be equidistant from the listening seat to each speaker). Let’s say X is 10 feet. A general guideline is to start Y at about 80% of X. Y is the distance from the center of the left tweeter to the center of the right. We use the tweeter because it’s the primary source for directional cues (imaging).

A note on separation—this is to your taste. I personally like Y to be about 83% of X for most speakers. For planar speakers, Y may be smaller, maybe as small as 70-75% of X. Some companies want you to use an equilateral triangle (X and Y are equal in distance), or greater. I suggest playing a mono source like female vocals and keep pulling the speakers apart until the voice becomes a fairly precise point between the speakers. Pulling it apart any further results in a too small voice or one that now begins to come from each speaker. Bring them back to the point where it worked, switch from mono to stereo and check out the image. This technique assumes you’ve established a “grid” on the floor so that movements are the same for both channels.

The other test is simply to notice when the female voice starts to sound unacceptably thin.  Then voice the separation by tonal balance, as well as image precision.  Once again, it’s a compromise.  You have to decide what means the most to you.

(3) Frequency response/tonal balance

Remember how we said that changing the separation could yield a cooler or warmer sound? And how toe-in can also dramatically affect balance (particularly high frequency balance)? Well, now you’re at that point (this assumes that your speakers are either nonadjustable, i.e. tweeter lever control, bass level control, etc., or that you’ve selected the nominal ‘flat’ position as a starting point).

Here’s an example—and listen, it’s just an example—you might feel quite differently. I find that if I set up most direct radiating speakers on an equilateral triangle, the sound (for my taste) is usually too lean. I can hear all the tiny sounds in the soundstage, but it’s become a precise, almost mechanical sound. It’s not ‘relaxed’ for want of a better word. It makes great Audiophile stuff, but the sound just doesn’t have the body and warmth that I hear with live music.

And yet, I know highly respected reviewers and manufacturers who prefer to listen with Y being greater than X. That’s why it’s your taste—remember, it’s not about some notion of ‘accuracy,‘ it’s about the music and you.

Step 3 is the final fine-tuning that will make the difference for you. One final note—some Audiophiles adjust toe-in to make the speaker seem to ‘disappear.‘ This is usually not on axis, but aimed to crossfire somewhere behind your head, or even aimed straight ahead. This is your call as well. I recommend going for the musical balance before going for an audiophile sound effect, but sometimes you can get both, so go for it if you like…

This should get you pretty close to a satisfactory set-up.

There are even smaller increments of adjustments that can pay significant dividends, but that’s a topic for another time (I call it ‘Playing the room’).

Don’t forget to listen at different seating heights and or speaker tilt back (which achieves the same thing).

If you have the ability to experiment with different seating heights or tilting the speaker forward or backwars, do so. Sometimes raising or lowering your listening position slightly can produce a more neutral and more alive sound.

There are generally two or three reasons why this may be true for your system.

First, it’s not uncommon for standing waves to affect your sound vertically.

Note that if you do plan to change the listening position’s height, be sure that it’s standing waves you’re hearing and not different time arrival and related crossover/phase relationships from the drivers!

A way to check that aspect is to slightly tilt the speaker backward or forward to see if what you’re hearing is predominantly an audible effect from the relationship of the speaker’s vertical height/angle to the listening seat.

You should do this anyway, whether or not you change your seat height.  With planar speakers, it’s equally important.  That’s because, apart from standing waves, there’s another issue to be considered called “beaming” which has to do with the tweeter.

There’s usually a quite obvious ‘best’ speaker elevation for speakers on stands.  Going by the manufacturer’s recommended stand height is only a good starting point.  That’s because the manufacturer has no idea what your exact listening height (seat height plus your body height) will be.

About the only speakers where this might not need to be considered would be any concentric array (separate tweeter located within the bass/mid’s voice coil, ala KEF and some others).  We’ve found that it’s rare you’ll be at the correct height for just about any speaker – on a stand or not – concentric drivers or not.

The only other speaker design where it may not be as noticeable is when the tweeters are beside the mids (horizontal relationship vs. vertical).  But that one is a whole ‘nother topic… And side-by-side line arrays for tweeters and mids may need to be tilted to find their optimum angle, just as planars.  Seat height may be critical here as well.

Since we only have a narrow range of adjustment that we can make vertically, it’s nice to at least know what is possible, if you so desire to address it.

Consider finding the right height and then building a platform (if necessary) to raise your sofa or chair height to the perfect position.

Please, do remember to try speaker tilt before trying seat height!  It’s not uncommon for both to make an improvement.

The reason why the effect may be noticeable enough for you to have a preference is that ‘floor bounce’ may be introducing a cancellation at some point.  The slight movement simply shifts the frequency slightly so that you may prefer (or dislike) the sound at a different seating heights.

Don’t consider speaker placement final until you’ve discovered the correct AC polarity for all components.

Incorrect AC polarity from just one component can make your system sound harsher than it should because the components change of reference ground point of the internal power supply transformer.

Usually it’s on the shield side of connectors and interconnects. Since this shield is common to both channels, it can present a slightly ‘grungy’ center fill that you may not notice until it’s been eradicated.

If you have carefully set your speakers up for a precise stereo image, and then you correct the AC polarity, depending on the polarity’s interaction with the other components, you may get a ‘hole in the middle’ when the artificial center fill (mono) information is removed.

With some systems, we’ve found the need to bring the speakers an inch or two closer together to correct for what had been a false center fill. Try it and see what you think…

PS Audio’s How to Section offers an in-depth step-by-step guide to help you get absolute polarity correct.  You can click here to visit our How To Section.

Don’t place your components, especially source components, in areas where there are bass reinforcement modes.

Try to avoid placing components (especially source components) in areas where the bass is very strong or exaggerated due to standing waves.

CD transports and turntables lose their life and dynamics when they’re being bombarded by resonance inducing bass frequencies.

Most vacuum tube electronics are especially prone to adverse effects with such placement.

In fact, it’s even possible to induce acoustic feedback with turntables that are improperly located when the system is played at higher levels (assuming loudspeakers with reasonably extended low frequency performance).

The problem is that we almost never reach that obvious level of deterioration (acoustic feedback), but we approach it. Because it’s gradual as we increase volume, it’s easy to miss the creeping degradation of the life and dynamics of the sound. Not to mention the bass becoming less articulate.

The absolute worst two places to locate source components, and for that matter, most vacuum tube electronics:

1. In a corner

2. Anywhere on the wall behind you

That’s not to say there aren’t other areas where unwanted bass energy may collect, but these two areas are almost certainly among the worst in any room.

Move your source components and any vacuum tube electronics out of these areas and listen to how much more effortless the presentation seems to be (and with better bass).

There is one place that-depending on where you sit in your room-you need to absorb or diffuse sound, no matter what kind of loudspeakers you own.

If you’re sitting fairly close to the wall behind you (if it’s within 4-6 feet), it’s good practice to diminish the first reflections from the main speakers that return from the rear wall. These reflections tend to smear articulation, add unwanted colorations, and mess with your musical enjoyment.

The best way is to absorb them, since you don’t want to add additional reflections to those already recorded, but there are cases to be made for diffusing as well. The best absorptive materials (producing the least sound of their own) are natural materials like cotton or wool.

But if you can’t come up with something soft enough or thick enough, you’re better off to resort to some absorbent panels than leaving the rear wall reflective and undamped.  And if you have a significant other, don’t forget about WAF!

Make sure there are no reflective surfaces before you
This isn’t always practical or possible, but if you have a coffee table or other reflective surface in front of you-at least when you’re listening seriously-if you can’t move it away from in front of you, try to drape a towel or blanket over it while listening.

And if you have a significant other, you’ve already stretched the boundaries with moving the table or covering it, so be sure to PUT THINGS BACK…

Naturally, whenever possible, no table in front of you (between you and your speakers) is best of all.  Getting rid of early and unwanted reflections will allow you to hear more of the music.

If you want to maximize your musical enjoyment, you need a preamp, receiver, or perhaps a CD player or DAC with a continuous volume control, or hopefully one with small volume increments (preferably1 dB or less).

‘Good enough’ simply isn’t-at least when you’re setting the volume for a piece of music. Learn to recognize the point where the music speaks to you, and make sure you play it there.

You can’t ‘set it and forget it’, because different recordings of the same music will likely have been recorded with a different perspective and recording level.

The next time you play the same CD or LP, don’t assume that the volume position you used last time will work this time. Your mood, your attentiveness, the noise/distractions around you, even the temperature and humidity-all contribute to the ‘ambience’ in your room at that time.

When you get the volume right for that music at that time in your room, the result can be capable of lifting you out of your daily life experience and causing your spirits to soar-just like live music does.

Most audiophiles have noticed this phenomenon. There are many conjectures of why this is so, here are a few.

1. The power line feed is more pure-there’s less machinery and motors to mess with it (for those of you with Premier Power Plants, no worries).

2. There may be more power available, especially in times of peak demand (weekdays in the summer).

3. It’s quieter.

4. You’re winding down and you can be more attentive to the music.

5. There are fewer distractions.

6. It’s generally darker in the room which enhances hearing.  When one of our 5 senses has less to do (visual for example) our other senses sharpen to compensate.

7. You can make it very dark, which provides the ultimate home listening situation, as you remove the distracting visual cues that prevent a suspension of disbelief.

The environment is very important.  Remember, much of what we hear and “see” in our imaginary soundstage is all phychological and helping improve the setting helps make a convincing mental picture of the music or the video experience.

If you have a window on the wall at the end of the room where your speakers are located, when you’re listening in the daytime, use drapes, shades or whatever it takes to reduce the light streaming in as you face the speakers.

Try it and you’ll find your enjoyment increases, as psychoacoustic ‘glare’ decreases.

The reason for this is rather simple.  When one or more of our five senses has been deprived of information, such as listening in the dark, the other senses tend to sharpen thier information gathering to compensate.  Thus, by removing visual distractions our hearing acuity improves and our listening experience does as well.

We all understand that the center image produced by our stereo pair of loudspeakers is an illusion. While it may sound like we have three speakers, in a stereo setup we only have two. The center image is sometimes referred to as the phantom image. It is created when the left and the right speakers contain identical program information.

A properly executed center image will be nearly palpable as if you could reach out and touch the image. To achieve this lifelike palpability, it may be necessary to slightly reposition your speakers.

The first step will involve the use of a tape measure. Because center image palpability is somewhat dependant on high frequency information as created by the speaker’s tweeter, making sure that both the left and the right speakers are equidistant from the listener is critical. Due to the short wavelengths involved, accuracy of the placement of the speaker pair is important to within ¼ inch.

First step is to measure the front to back distance from the rear wall. Make sure that it is exactly the same for both the left and the right speaker.

Second step is to then listen to the results of the positioning change in step number one. I recommend a single center positioned vocalist for this exercise. Whatever piece of music you select, use the same one over and over, noting the differences in center image as you go along.

Third step is side to side placement. In small increments, move the speakers closer together, or farther apart, until the center image begins to achieve the level of palpability you desire. A good starting point is to measure approximately six to seven feet between the left and right tweeters.

Fourth step is toe in. Once the side to side distance has been established, the angle of the speakers, relative to the listener must be adjusted. In most loudspeaker pairs, the proper angle of toe in can easily be roughed in visually. Remove the grille cloth of the loudspeaker and, sitting in your listening position, have a friend or colleague angle the loudspeakers in so that the tweeter is pointing at you equally on both the left and right sides. Once achieved, begin the listening process again, equally angling the speakers more or less towards your ears with respect to the tweeter until a perfect center image is achieved.

A note of caution: too much toe in can give you a great center image at the expense of outside information. That is, information present on the recording should go beyond the left and right sides of the speaker. A good compromise between the two is important.

Lastly, place an object dead center on the wall between the speakers.  A picture, a tapestry, anything pleasant to the eye.  Because the center image is a phantom image, it helps reinforce the image if your eyes have somethin in the center to focus on as well.

There are two truths in today’s world of digital audio: at some point the digital audio must be converted to analog so you can hear it, and digital and analog devices should be kept apart from each other.

Their separation is recommended from both a physical and an electrical standpoint because their very different nature of operation can cause audible interference.

Physical separation can be rather simple but quite effective with respect to performance. If space permits, group all of your digital based equipment, CD players, DAC’s, Digital Lens, transports, AC3 processors, together. Group all of your analog equipment such as, preamps, power amps, tuners, together and separated from the digital group by at least one foot, two feet if possible.

Electrical isolation can also be simple. If you can identify the fact that you have separate AC plug circuits easily accessible to your equipment, plug all digital devices into one circuit, and all analog into another. Barring access to separate circuits, group digital and analog equipment into two separate high quality plug strips.

The ultimate solution is the PS Audio Power Plant…..but that’s not free.

rom Newsletter reader William Erdman MD comes this tip that really is correct!

Did you ever notice how after listening to music for a while you find that increasing the volume improves the quality? On the other hand, starting out at that same high volume can be downright unpleasant.”Too loud too fast” is a prescription for the auditory equivalent of pulling a muscle. After that it seems that the system is “just not right”. However, give your ears 5 or 10 minutes of moderate volume and then crank ‘er up a notch or two and “viola” you are having a great time! ( Wait a half hour and you can probably crank it more ….and it gets even better!)

We don’t often think about it but our ears are the most critical piece of equipment between the glorious sounds emanating from that megabuck sound system and our brains. The ear is a loudspeaker in reverse, a transducer that converts sound waves into electrical impulses.(Incidentally it is actually possible for your ears to generate sounds audible to others- but that is a whole different story.)

There are probably a number of reasons for this “warm up” phenomenon. Your ears make mechanical and chemical adjustments to better optimize hearing based on the amount of input they are getting. This is similar to what your eyes do when you enter or leave a darkened room. Just as with your eyes, these adjustments take some minutes to become fully effective.( I suspect that there are some adjustments made in the auditory portion of your brain as well.)

What ever the mechanism, I can think of no tweak ( free or expensive) that can improve your systems performance more, than simply “letting your ears warm up”!

From Newsletter reader Steven Braude comes this tip that really does work!

No need to purchase expensive demagnetizers for CDs. Spend $40 or $50 for a good old Zerostat. These devices don’t just work on LPs.

Zerostats are available through your local dealer, or on-line through theAudio Advisor . Click on the picture of the Zerostat for the link.

Zap your CDs with the Zerostat before playing; it works. And don’t stop there. Periodically zap your speaker cables and interconnects along their length, and AC cords as well. Also the cabinets of your equipment.

An occasional Zerostat sweep of the whole system can clean things up very nicely. It helps tame digital nasties, and also some of the harshness that simply develops over time (you may not realize it until it’s gone).

Here is another vibration do it yourself tip from one of our newsletter readers, Steve Fellows.

Here’s a way to make both a sand AND air isolation platform for audio gear that costs less than $4.00.

Sand and air isolation platforms dampen vibrations in different ways. An air platform (Townsend Seismic Sink, Bright Star Air Mass, Vibraplane, etc.) dampens floor and airborne vibrations. A sand isolation platform neutralizes vibes that originate WITHIN the gear, such as from power transformers.

To make a sand isolation platform with “sand dampers” I purchased a quart-bag of white silica sand ($2.40) and a bag of eight, 12-inch balloons ($1.00). Black in color. Use a funnel to fill 4 balloons with sand until they are completely full, each being about half the volume of a tennis ball, then tie them shut. Slightly flatten each of the sand-filled balloons then place them under your component’s feet. With the sand dampers under my preamp the music is cleaner & less blurred than when the preamp sits directly on the rack’s shelf. Placing 4 sand dampers under my cd player made only a slight increase in sound quality. But I experienced a very noticeable benefit when air dampers were placed under my cd player. To make an “air damper,” hold open a balloon so that it fills with air on its own, that is, without blowing any air into it. Just hold it open & tie it shut. It may at first seem there’s not enough air in the balloon but the key is to not overfill it. Place 4 balloons, filled & tied, on an equipment shelf near the corners. Then place another shelf on top of the balloons, making sure the corners don’t touch any part of the rack. You now have a shelf suspended by air. (I haven’t yet done this using my 35-lb power amp as the 4 balloons probably wouldn’t support it. Perhaps having several of the balloons scattered in between the shelves may work. I’ll eventually try it.)

If you don’t have a second shelf the cd player (or preamp) can be placed directly on the 4 balloons. However, damping a component this way will require the gear to be used with a remote, otherwise using push buttons on the faceplate will cause the component to “jiggle” annoyingly. Not a good way to impress your audiophile buddies. But using a second shelf with the balloons sandwiched in between looks great & adds stability. I have about .33 inch of space between my two shelves with the 4 balloons in the middle.

Interestingly, when my preamp (Melos SHA-1) rests on the air dampers the music sounded worse than when placed directly on an undamped shelf: dull & uninvolving. Using the sand dampers described above greatly benefited the Melos. But the cd player’s performance improved significantly when placed on the air dampers than when sitting directly on the shelf or the sand dampers.

One final way to reduce vibrations on a component’s top plate is to fill a small zip-lock plastic bag with 3 or 4 tablespoons of sand. Place the bag on the component’s top plate (assuming there are no heat vents) and spread the sand evenly across within the flattened bag. Then cover the bag with a dark cloth to hide it.

So there you have it. Sand AND air isolation platforms for less than $4.00.

Here is a vibration platform you can build. It was designed by our customers, Bob mathews.

“I have found a very effective and fairly cheap solution that costs less than spending $100 on one of the fancy isolation platforms from some of the mail order catalogs. I buy racket balls from K-mart. Then I use a piece of marble that you can get at a hardware store in the baking section. Usually at Christmas Time, these marble bakery rolling slabs are on sale. You can also use an old wooden in-out office box. You put the 4 racket balls in each corner of the in-out box, then place the marble slab on top of the 4 racket balls (or you could use tennis balls). Then you place your amp on top of the marble slab.

You could also simply glue the tennis balls onto the slab and ignore the in/out box.

You get the air isolation effect from the racket balls, and other vibrational relief from the marble slab. The only purpose the wooden in-out box provides is to help stabilize the balls from moving around. You can actually tell a difference in the sound. In my system, the bass is much tighter and more dynamic, less mushy sounding, etc.”

Reader Darrell Tunning has submitted an ingenious scheme to magnetically levitate the CD player. Our thoughts on this are good ones and we don’t think the magnetic fields from the permanent magnets should bother the CD player laser mechanism. Remember that a magnetic field decreases in strength exponentially with distance.

Here’s Darrell’s plan:

I enjoyed the tip on the vibration reduction, but if you want a REAL increase in isolation try this: get 24 bar magnets fromRadio Shack 2″x3/4″x3/8″, 4 baseball card plastic cases, 1/8″ rubber sheet large enough to cut 8 rectangle pieces to cover the top & underside of the plastic cases, and contact cement. The magnets are 1-7/8″x7/8″x3/8″ versions, Radio Shack catalog number 64-1877

Glue 3 magnets to the top of each of the baseball card cases (make sure that all magnets are facing the same way so that each of the three magnets are opposite in polarity from the 3 you will glue to the bottom case).

Install the rubber sheets on each outside end of the cases (Top & Bottom). Make sure you can slide each top cover into the bottom cover (Larger one) without scraping the magnets. The magnets will repel each other with about 4-6 lbs. of force per case.

Install each case under the 4 corners of the cd or dvd player, use three if 4 has too much lift. You should see 1/4 to 1/2″ float distance between the top magnets and the bottom magnets. You are now totally isolating the outside low frequency & High frequency from your player!! You will notice: bass is unbelievable, there is much more “Air” around the music, and the soundstage is much more believable.

It is not perfect isolation since the cables and power cord are still attached but it still makes a huge difference!!!

Adding a subwoofer to your audio/video system can be a great improvement because the vast majority of loudspeakers cannot reproduce low frequencies properly. They need a bit of help, and a subwoofer can provide that help.

There are, however, a few rules to follow regarding subwoofer placement that can make a huge difference in the subwoofer’s performance.

Never place a subwoofer in the middle of any wall or boundry. Certainly not in the middle of the room! This is a common mistake when you only have one subwoofer to add. Logically, if you only have one subwoofer to share between the left and right speakers, it might make sense to place it between the left and right speakers so as to properly share the low frequency output of the sub. Unfortunately, this is a big mistake because placed in the middle, room interactions will dramatically reduce the output level of the sub. It is far better to place the subwoofer to one side or the other. Typically, placement next to the right hand speaker is preferable if possible. This is because the bass section of an orchestra is traditionally on the right hand side as the audience faces the orchestra.

If the subwoofer’s volume control is turned all the way up, and you still do not have enough output to satisfy your craving for bass, try placing it in a corner. This trick works because the corner of your room can act as a horn to acoustically amplify the output of the subwoofer. In fact, this is what we will be doing in the PS Audio listening room. You can find more information about the PS listening room and our subwoofer placement by going here.

Here’s how to “get it right” on the fly!

Craig Burns has yet another suggestion for us, but this time it is on how to specifically find the best spot in the room for your subwoofer.

Center your subwoofer in the location that you usually sit to listen, on top of a piece of furniture at your normal listening ear height. Then get down on your hands and knees around the sides and behind your main speakers. Keep your ear vertically the same height as your subwoofer would be on the floor.

Crawl around while listening to some test tones or bass-heavy music. When you find the location were the bass sounds the smoothest, that’s were you should put your sub. Hey! You just found the sweet spot for your sub!

You may need to play around with the direction that the sub is firing in the room to fine tune the sound after the subwoofer is placed in its sweet spot. And this will save your back from consistently moving the sub in 2inch increments…. you can just move your ear!

A common mistake made when adding a subwoofer to a video system is to duplicate the low pass filter, thus dramatically changing (for the worse) the sound and transient response of the sub.

Most surround processor have a separate subwoofer output that includes a crossover slope. That is, the sub output on a surround processor is already rolled off, containing no high frequency information. If you place this already rolled off subwoofer information into a subwoofer input that is not specifically designed for this type of output (most are not), you will then wind up with two roll-offs, giving you results that are less than optimal.

The easiest way to make sure this is not a problem, is to turn the subwoofer’s low pass filter control up as high as it will go. By turning the control to its highest frequency setting, only the roll-off built into the surround processor’s subwoofer output will be active and the two will not interfere with each other.

Some video processors allow you to turn off the internal roll-off of the processor allowing you to use the built in one on the subwoffer itself.

In most cases, however, it is easier and preferable to simply turn the roll-off of the subwoofer up as high as it will go, which is usually around 120 Hz.

Getting more depth out of any loudspeaker system isn’t all that difficult. It is a matter of positioning. In fact, changing the position of your loudspeaker pair by only a few inches can, sometimes, net you a great deal of depth increase.

You must first setup your loudspeaker pair with a solid center image. Once achieved, depth can be enhanced by the relative front to back distance of your loudspeaker pair.

Using a familiar piece of music, preferably a single vocalist, pull the loudspeaker pair away from the rear wall in small increments (perhaps 3 to 4 inches at a time). After you pull the speakers away from the rear wall, return to your listening position and start the musical selection over, noting the increase or decrease in front to back depth.

In some cases, it may be necessary to pull the speakers out from the rear wall by as much as several feet in order to achieve lifelike depth. When properly placed, the vocalist should appear to come from behind the loudspeakers (when proper microphone recording techniques are employed on the recording) and appear to be detached from the speakers themselves.

A note of caution: if you pull the speakers too far from the rear wall, you may start to lose bass coupling. You will have to find an appropriate compromise of position if your speakers do not have controls on them to increase or decrease the bass. Alternatively, you can add a subwoofer to augment the bass.

A second note of caution: before starting the procedure outlined above, it is a good idea to use some masking tape to mark the current placement of your loudspeakers so as to maintain a reference.

This tip from Craig Burns, a happy PS Audio customer and Power Plant owner, works on most every loudspeakers. We would add that there are certainly varying opinions of the subject by the manufacturers themselves. Most loudspeakers do seem, however, to benfit from this upgrade. A nice feature to this tip is the ease of which you can try it and remove it if it doesn’t work for your system.

Here is Craig’s tip:

There are a lot of speaker manufactures that put a fuzzy cloth material around the tweeters of their speakers. It’s in a donut fashion around the tweeter to improve imaging. Why other manufactures do not utilize this I’m not sure? I think this helps with diffraction of the tweeter on the face of your speaker cabinet itself. I have found that if you go to any arts and craft store you can buy for about $5, sheets of felt with a sticky back in assorted colors. Just cut the felt into a donut size shape to fit perfectly around your tweeters. The imaging is greatly improved! It has a nice stock or OEM look to it, and the felt is easily removable. The sticky adhesive stays on the felt, not your speaker if you ever decide to remove them for whatever reason.

Paul’s comment:

This works because reflections from the tweeter, bouncing off of the speaker’s baffle have been dramatically reduced. You can even layer several thin sheets of felt to achieve the appropriate amount of damping, but one sheet is usually sufficient.

Having been involved with speaker manufaturing I can tell you that there are a variety of reasons why this is not universally applied to a speaker’s design. Perhaps the single largest reason this is not used would be appearance; it does look a little cheap on the outside baffle of the speaker. Another reason would be that the designer did his best to compensate for the baffles reflection in the first place, but my experience (limited as it is) is that this addition helps most any design.

One note of caution: if the felt sheet gets too thick it can hamper the radiation pattern of the tweeter’s side lobes so it is important to keep the damping material as thin as possible.

When sound pressure comes out from your loudspeaker, it is typically aimed directly at your listening position. When it arrives at your listening position, it is referred to as “direct sound”. Along the way towards that listening position, the sound pressure wave will hit the side walls, pieces of furniture, and so on, causing multiple reflections. The reflections are delayed slightly in their arrival to your listening position. These multiple delayed signals are referred to as reflections. Therefore, when the direct sound arrives at your listening position, it is quickly followed by the same sound, slightly delayed in time, causing a confusion to the music. If we were to play our speakers outside, this would not be a problem (unless it rained). Unfortunately, this is not the case, and we in fact listen inside. Therefore, we must deal with these reflections.

Since we cannot eliminate all reflections, we should concentrate our efforts on the biggest problems first. The biggest problems are known as “first reflections” because they are the first and most predominant reflections to arrive at our listening position. First reflections are almost always created when the sound coming from the loudspeaker hits the side wall (the walls closest to the left and right sides of the speakers). They are relatively easy to correct if you can identify where they occur. To minimize their effect on the sound of your system, you need to place either an object in their path to diffuse their energy (such as a piece of furniture), or an object in their path to absorb their energy (like a tapestry, curtain, or foam rubber).

Locating the point of first reflection is simple, but you will need the aid of a colleague and a small mirror. First, remove the grille cloth from your loudspeaker. Second, have your colleague hold the mirror against the approximate area of the first reflection on the side wall, holding the mirror at the same height as your tweeter is from the floor. Have the assistant move the mirror along the wall, while maintaining the correct height for the tweeter, until you can see the tweeter in the mirror. Mark that spot, and repeat the procedure on the opposite wall. The mirror has visually exposed the location of the first reflection.

If you are interested in learning more, check out our series on how to build a listening room by clicking on the link.

This tip comes from PS customer Bill Erdman. We thought so much of the simplicity and effectiveness of this tip that we have added the little noise sniffer to our permanent arsenal of test equipment we use when going into the field. Bill Mentions you can go to the hardware store and purchase one. We had a little trouble finding it, but succeeded in the end. This tip is so cool that we’re thinking of having some of these made for us. Perhaps in the future we can even offer them to our customers. Ours is a different brand than Bill’s but they are essentially the same.

The unit can be purchased at Home Depot and many other retailers. You can click here for the datasheet.

Here’s Bill’s tip (for which he won an Ultimate Outlet for:)

I love my PS300 and I love the Multiwave upgrade but this fifteen dollar tweak has improved the sound of my system as much as both upgrades. It is based on science and it is extremely easy to verify the significant beneficial sonic effects. In fact I was able to accomplish this wonder in about 90 seconds.

You will need an AC line voltage sniffer. I got mine from the hardware store for around ten bucks. It is made by AW Sperry and it looks like a fat pencil. It has a sensitivity control, a red LED and an audible squeal which increases as the device is brought in proximity to any source of alternating voltage.

Set the sensitivity so that it squeals and glows when it gets within 4 to 6 inches of an electrical outlet. Then run the tester along the length of your speaker cables while the amplifier is muted. I was amazed to find there were several feet of cable that were picking up AC voltage even though the nearest source of power was a lamp cord 3 feet away. Even more amazing was the tester squealed like a stuck pig when placed near the speaker cones themselves (All my efforts at keeping grunge out of the system were defeated by a lamp cord!!!)

Nonetheless, the solution was easy. Unplug the lamp and “bingo” the tester goes silent. Apparently the lamp cord acts like a radio transmitter antenna and my unshielded speaker cables act like a radio receiver antenna.

Although I could not hear any hum in the speaker with the lamp plugged in, there was a clear difference in listening to music with lamp plugged Vs unplugged. The bass was fuller and richer and the highs were more crisp and had more air about them (especially cymbals). I performed a blinded plugged Vs unplugged comparison and was easily able to get it right 6 out of 6 times. So until I can afford a long (39 ft) set of your shielded lab cables I will be listening (happily) to music in the dark.

Obviously you can use the tester to check shielding on interconnects and power cords. You can even “watch” the music in your speaker cables by laying the tester on top of the cable (lamp unplugged of course) and turning up the volume a bit. The led glows brightly with each drum beat or crescendo. At high volume the tester will glow to the music even if held 2 inches away. That will give you an idea of just how far apart cables need to be separated

This is a modular design that can hold as many Ultimate Outlets as you have. When you have multiple outlets scattered around it can be messy and costly when you take all the long power cords into consideration. Therefore, by setting them up this way I use mostly 3 and 4 foot power cords. This will allow me to purchase the sought after Lab Cables much easier because of the shorter lengths I now need.

You can use any type of wood and tubing to coordinate with your existing furniture (which will make the wife happy). I used birch and stained it a Medium Cherry color.

To Assemble:

1. Cut your wood into 6 1/2″ x 7″ pieces
2. Clamp the cut pieces together
3. Mark the corners to be drilled (I used a 1/2″ drill bit)
4. After the pieces are drilled, take 4 pieces of all-thread and slide one washer and one nut onto the all-thread. Make sure there are only about three threads showing from the bottom of the all-thread to the nut.
5. Take the four pieces of all thread and slide it through each of the four holes in the first piece of wood and set so the all-thread rods are facing upwards.
6. Cut your tubing into pieces 3 1/2″ long, four per shelf.
7. Slide four pieces of the tubing onto the all-thread.
8. Take your next piece of wood and slide it onto the all-thread. Repeat steps 7 & 8 for however many shelves you need.
9. After the last piece of wood is on the all-thread cut the excess all-thread off so you only have about three threads showing. Be careful not to damage the threads when you cut them.
10. Take a washer and a decorative nut (like an acorn nut) and semi-tighten.
11. Align the pieces of tubing so they are straight and then tighten a bit more. When you are happy with the alignment completely tighten all the nuts.

If you choose to label the Ultimate Outlets a P-Touch type labler works very weel.

And there you have the Ultimate Outrack.

What you will need:

• Enough of your chosen wood to make as many 6 1/2 x 7 inch shelves as needed. Four 24″ x 1/8″ pieces of all-thread Eight washers (4 for the top, 4 for the bottom)
• Four 24″ pieces of 3/8″ tubing of your choice
• Four nuts to fit the all-thread (for the bottom)
• Four Acorn nuts (for the top)
• Wood stain
• 2 1/2″ wood screws for each Ultimate Outlet

Submitted by:

Jeff Whitlock

Most audiophiles have noticed this phenomenon. There are many conjectures.

(1) The power line feed is more pure-there’s less machinery and fewer motors to corrupt the power lines. Remember, your sound comes from modulated AC power.  This is a great argument in favor of adding regenerated power like that from a Power Plant as opposed to trying to simply filter the power.  The problems we see on the AC line at night can only be fixed by building new power, not filtering the noise on the line.

(2) There may be more power available in the evening, especially in times of peak demand (weekdays in the summer). 

(3) It’s quieter.

(4) You’re winding down and you can be more attentive to the music.

(5) There are fewer distractions.

(6) It’s generally darker in the room, which leads to #7.

(7) You can make it very dark, which provides the ultimate home listening situation, as you remove the distracting visual cues that prevent a suspension of disbelief.

Balanced (XLR) inputs and outputs on source equipment provide superior performance relative to unbalanced (single ended) inputs and outputs.  Balanced signals have lower noise, twice the output voltage and are more immune to external noise sources.

In a balanced circuit, there are two equal halves of a circuit operating on a waveform to amplify it. To view this easily, picture two independent amplification stages, tied together in parallel. One half of this two-part circuit operates on the positive half of the waveform and the other on the negative half.

But, having said that, there is one key difference that (strictly speaking) has to happen before we can truly call ourselves ‘balanced’. That is an engineering term known as common mode rejection. When a circuit displays common mode rejection it means that any signal presented to it that is COMMON to both inputs is rejected.

Common Mode Rejection (CMR) means that if you have a balanced input on a preamp (for example), and if you put the same signal in both the + and the – inputs at the same time, no signal will appear at the output (balanced or XLR connectors have 3 wires inside them, a ground and a + wire ((just like an RCA connector)) with the addition of a – signal wire). Why is this important? Because, on an input, this feature will reject  noise. Any hum or high frequency noise that gets into an interconnect cable (for example) will be on both wires (or all three wires in the case of a balanced cable). When the signal goes into the preamp, anything common (like the noise) will be eliminated.

However, if we do the opposite, put in a signal that appears (out of phase) on both the + and – inputs simultaneously, the output of the amplifier or preamplifier will double (6 dB rise).

Further, when a circuit inside an amplifier is balanced, distortion components that are common to both halves of a circuit will be reduced or eliminated by this same phenomenon (CMR).

Not all products that suggest they have a balanced input actually have a proper one.

Most Audiophiles would most likely suggest that it is preferable to keep the speaker cables as short as possible and the interconnects long enough to make up the difference.  The truth is, the answer is more specific to the setup you are working with.

For instance, this philosophy works well as long as the interconnects aren’t longer than six feet (2 meters). Once the interconnects exceed this length, their capacitance becomes a factor (it increases with cable length), and will start to make some preamplifiers sound worse.  This is not true of all preamplifiers.  PS Audio products, for instance, use power amp-like output stages designed spefically to properly drive long interconnects.  So the answer really depends a great deal on the type of equipment you have and the lengths of cabling involved.

Here are a few general “rule of thumb” suggestions for narrowing down your decision.

Tube preamplifiers are poor choices to drive long interconnects and should be avoided.  Their high output impedance really negates the use of anything over two meters.  Solid state preamplifiers have a better chance of driving long interconnects but you should check with the manufacturer to find out the preamp’s output impedance.  If the preamp’s output impedance is below 50 Ohms, then you can succesfully drive lengths of interconnects 15 to 20 feet or more.

Balanced (XLR) interconnects are preferred when using long interconnects over single ended (RCA) cables.  This is due to the superior noise rejection of a balanced cable.

Power amps are generally more capable of driving long lengths of wire than preamplifiers are.

An AC power line dedicated to one AC receptacle that optimally powers one piece of equipment.  Most AC power lines in our home have multiple AC receptacles connected to them.  Each piece of equipment we connect to the AC power can add back contaminating noise into the power line.  Using a dedicated line to feed your equipment can reduce the noise contribution of the equipment.

The dedicated line can be installed by the homeowner or an electrician and is really nothing more than a single 12 gauge wire running from the AC outlet back to the home’s power panel.

Dedicated lines can make a tremendous improvement in system noise levels and performance.

Because THD is a telltale sign of far greater problems affecting the power your equipment receives.

The PS Audio Power Plant Premier, as well as other PS Power products, provide a built in THD (Total Harmonic Distortion) analyzer on their front panels to display both the incoming and outgoing THD on the AC line.

Typically, we see THD levels of between 10% and 2% on the AC line.  These levels are reduced by a factor of 10 by the Power Plant’s AC regenerator, providing near perfect power to your home’s AV system.  But why is THD important to reduce?

THD is added harmonics (higher added frequencies).  What comes out of the home’s AC socket is what we refer to as the “fundamental frequency” which is either 50Hz or 60Hz.  A perfect fundamental frequency sine wave will have no harmonics.  The added harmonics we are trying to eliminate are not, in themselves, bad.  What is detremental to our system’s performance are the root causes of these harmonics: flat topping waves.

When the peak of the sine wave gets clipped off (as it does in all our homes) we lose critical energy to feed our equipment properly.  This falt topping is easily measured by the increase in harmonics it produces, and this is why we added a THD analyzer on the front panel of every Power Plant Premier AC regenerator.

Other manufacturers do not include such a feature because few other manufacturer’s products can repair the flat top damage to the AC power in your home.

No power conditioner, either series or parallel can remove these harmonics.  Only be regenerating the AC itself with new energy – as is done with a Power Plant – can restore the missing pieces of the sine wave.

Some power amplifier manufacturers write in their owner’s manuals not to use a power conditioner for their amplifiers.  So, does this mean that  all power products should be avoided with a power amp?

No.  While it is true that the vast majority of power conditioning products should not be used on power amplifiers because they restrict the sound, there are some that work well with a power amp.  The first thing to understand is that there are actually three types of power conditioning products: series, parallel and regenerated.

Most power conditioners fall into the first category, series.  These are, for the most part, to be avoided with power amplifiers.  With but few exceptions (PS Audio’s being one of them), a typical series power conditioner will restrict the dynamics of a power amplifier.  None can improve dynamics.

The next category, parallel power conditioners, do not restrict the AC power to an amplifier.  Their only downside is they have very little benefit to the AC power line in terms of cleaning the AC.  None can improve dynamics.

The last category, regenerated power, is the best of all categories and actually provides improved dynamics over simply plugging into the wall socket.  Regenerated power is superior because it uses active energy storage to provide improved dynamic range, voltage regulation and rock solid AC to a power amplifier.  Products like the PS Audio Power Plant Premier are perfect examples of an AC regenerator technology.

Apple products such as their Apple TV and iPods do not support internet radio unless you know a few tricks.  Here’s how to add internet radio to the Apple TV or Apple iPod Touch remote control software.

The secret with adding internet radio to these products is found in the way Apple products use play lists to manage available content.  If you simply select a radio station you wish to listen to in iTunes and drag that station to a playlist, the station becomes available in any Apple product once you synch the product’s playlist to iTunes.

Here’s a step-by-step procedure.  First, open iTunes.  Next, create a new playlist and label it ‘Radio’.  Then, go to the upper tab in iTunes labeled ‘Radio’ and open the tab.  A list of hundreds of radio stations appears.  Find the station or stations you wish to listen to.  Drag these choices into the Playlist you created called ‘Radio’.

Once you’ve completed this, simply synch the Apple TV or iPod Touch to iTunes so both have this new Playlist ‘Radio’ added to their memories.

When you wish to play the radio on your Apple TV, use either the Apple remote control or the free Apple remote control software available for iPod Touch products.

If you are using the Apple remote and the Apple TV, simply navigate to the ATV’s music section, go to playlists, you will see your newly added ‘Radio’ playlist.

Open the Radio Playlist, you will see your selected radio stations.  Simply select the one you want and it will play.

If you are using the iTouch remote control software to control the Apple TV, use the same procedure as above.

The Bridge does not currently support 24 bit m4a (Apple lossless) files.  It will be supported on a later release.

Ryan successfully converted the files to FLAC using Max.

Also, From Steve Wooten:

HDCD’s decode as redbook CD’s (16/44.1) if you do not have a HDCD player. If you have a HDCD player they decode at 20/44.1.

If you rip w/dBpowerramp set to decode HDCD your Bridge can play HDCD’s natively instead of 16/44.1. That’s good news!

As for how do I know – the PWD shows the resolution as 24/44.1.

For those who don’t know where to get them, one source of 96 kHz and 88 kHz high resolution music is available athttps://www.hdtracks.com/index.php?file=9624albums
http://bluecoastrecords.com/
http://artistconnex.com/
http://www.computeraudiophile.com/content/Two-More-Complimentary-241764-HRx-Downloads-Courtesy-Reference-Recordings

http://www.designwsound.com/dwsblog/?page_id=318
http://www.bowers-wilkins.com/display.aspx?infid=3550

http://www.linnrecords.com/

http://www.highdeftapetransfers.com/

http://www.gimell.com/

http://www.gubemusic.com/?grid=553

I’ll post other sources as I get them.

This tip is from Jim Smith’s tech tips.

I’ll never forget these intertwined events. It was in the early 1980s and I was recording the Alabama Symphony for the Birmingham National Public Radio affiliate.

Several leading union musicians (from the Symphony tape committee) and the conductor would visit my shop one night each month. I’d play back the master recordings that I had recently made of concerts that were to be broadcast. Their job was to pick the best performance. Then I would prepare the broadcast master from my 30 I.P.S. analog master.

I came to know and to spend some time around the conductor. I’d say he was in his middle ’70s at that time. In our informal meetings, he’d have a problem withdiscrimination , which is common for older folks. In this case, discrimination is the term for being able to listen to someone and understand him while others are talking at the same time.

Before I tell you my main point, you need to know one more thing:

It’s REALLY LOUD on stage with a full orchestra when it’s playing power music . I once went out onstage while doing a test recording during a dress rehearsal for Gustav Holst’s The Planets . As luck would have it, it was during “Uranus,” one of the loudest sections.

I had measured the sound pressure levels in the audience. I knew they were about 95 dB. But up on stage, standing next to the conductor with all the brass and percussion wailing away, I was shocked at how loud it was! It felt like my hair was being blown back like that old Maxell tape ad!

Of course, someone who has been exposed to these incredible sound levels for 50 years is bound to have significant hearing damage. Add to that the age factor, and this man should have been lucky to hear an ambulance siren!

During our listening sessions at my shop, he always picked up on problems, often before the much younger musicians. Furthermore, in the middle of a Beethoven Symphony #6 rehearsal, I saw him tell a second violinist several rows back to retune. On more than one occasion, he’d have to call somebody on a blown entrance when there was a full orchestra wailing away!

It always amazed me that he could hear so precisely during the playback sessions and during the rehearsals. My point is that age, and even exposure to lifelong loud levels, seems not to be the only indicator as to whether a trained listener can still hear.

For example, I often sit with younger men and women, teaching them about sound, or just kicking back listening to music. I’m 63, but I can reliably hear things that they miss entirely.

Don’t worry if you’re past 50. It just means that you’re experienced!

It probably sounds pretty goofy, but actually a lot has been written about this in the legitimate audio press, and best of all, it works!

Even though CD’s and DVD’s are not magnetic in any way, using an inexpensive bulk tape eraser, easily purchased at Radio Shack for less than $20, you can make a significant improvement in the way your CD’s and DVD’s sound.

Purchase the bulk tape eraser. Then use it on the CD or DVD as you would a tape. Follow the instruction that are included with the bulk tape eraser. There is no potential for damage that I am aware of by doing this.

There is a commercial version of the bulk eraser made by Bedini. Click on the picture for the link.

In order to benefit from the results, you’ll have to do this each time you play the disc, but for critical listening sessions, it’s a must.

Try it yourself. Listen to a familiar track on a CD. Use the demagnetizer, then insert the disc and listen to the same track once again.

I continue to be amazed at how much better this procedure makes the disc sound.

Our thanks to Bob Harley of Stereophile, Fi, and now the Perfect Vision/TAS for demonstrating this amazing phenomena to us.

If you are using a turntable for phonograph records as many still are, make sure your VTA is adjusted correctly, it can make a huge difference in the way your records sound.

VTA stands for Vertical Tracking Angle and is the angle your needle sits at relative to the surface of the record. We recognize that the left to right angle, as you face the front of the phono cartridge, must be straight up and down. Many of us do not realize, however, that the front to back angle, as we face the front of the phono cartridge, is also critical.

Conventional wisdom would suggest that you want the bottom of the phono cartridge perpendicular with the surface of the record, as viewed from the side. This is certainly a good starting point, but we have found that a slight amount of rear angle can improve depth rather dramatically. That is, as you view the phono cartridge from the side, the rear of the phono cartridge (the rear is the end closest to the turntable arm pivot point) should be slightly closer to the record surface than the front. The angle is generally controlled by the pivot arm height.

Only careful experimentation and listening will reveal the best VTA for your particular arm/cartridge setup, but it can be well worth the effort.

However, in our Hot To Section we offer a complete instrucion on how to setup VTA on the turntable.  Click here to go to our How to on VTA,

Not all houses are wired equally. It’s unfortunate but true. The wiring in your home may be different from AC plug to AC plug. That is, there is a hot and a neutral side to each AC outlet. Upon examination of the AC outlet in your home, you’ll notice that there are typically three holes, in a triangular pattern. Two of the holes are actually more accurately characterized as slots, these are across from each other, and the third is below the two, and is a hole rather than a slot. The two slots across from each other have the actual AC energy between them, the third hole, below these two is a ground. Of the two slots, one is referred to as the “hot” side and the other is referred to as the “neutral” side. It is important that the hot and neutral sides are always the same.

You can determine if they were properly wired by purchasing an under $10 polarity checker at any hardware store. The polarity checker is a simple device that plugs into the wall socket and indicates proper or improper wiring by means of two LED lights.

Purchase one of these inexpensive devices and make sure that each AC wall socket has the correct polarity in your listening room or video room. If you should find an improperly wired AC wall socket, you can reverse the wiring yourself after first turning of that wall socket’s circuit breaker and then making sure it is no longer active using the same polarity checker. Alternatively, you can hire an electrician to do the work.

A common mistake we see concerns where your equipment is plugged in. Keeping in mind that receivers and power amplifiers draw a lot of current to operate (lots of watts), it is critical that you follow these three steps:

Never use an extension cord or plug strip to connect your receiver or power amplifier, if at all possible. The results of using an extension cord or plug strip can be loss of bass, impact, and dimensionality. It is much better to plug a receiver or power amplifier directly into the wall socket. If you must use an extension cord or plug strip, make sure it is wired with the same thickness of wire that is found inside the walls of your home, typically 14 gauge or 12 gauge.

Find a separate plug for your higher-powered items such as power amps or receivers. Try not to plug them in into the same plug as your lower level items such as, preamps, CD players, and so on.

Find a separate circuit for your higher-powered pieces. Many AC plugs in your home are wired together. Typically in a room with (for instance) six AC outlets, three will be wired together on one circuit and the other three will be wired together on yet another circuit. By “circuit” we mean that a separate run of wire will go from the plug to the circuit breaker box in your home. To find the separate run of wire, or circuit, you’ll need some help. With the aid of a colleague positioned within earshot of yourself and near the circuit breaker box, have him selectively turn on and off circuit breakers until you have determined which ones are separate. I recommend using a lamp, plugged into the AC outlets, to perform your test.

Most of you know what balanced power is: when both the hot and neutral of the AC line has power on them equal to 1/2 the total AC voltage, each being out of phase with the other. So, if your balanced voltage is 110 volts, you would measure (as referenced to ground) 55 volts on each leg of the AC, and because the two voltages are out of phase, they add together to make 110 volts. Normally, only one of the prongs in our home’s wall socket has AC power on it, the other being the same potential as ground.

The chief advantages of balanced power is common mode rejection, which provides a greater rejection of noise on the AC line. Your preamplifier or power amplifier’s own input power transformer can reject a great deal in the way of noise on the AC line if it is fed a balanced input.

So, how can we provide balanced power for our equipment? If we live in a country that already has 220 volts, then we must use options 1 or 2 below. This is because homes in countries with 200 volts are themselves single ended. In the United States, Canada and countries where 110 volts is the norm, then we can use any one of the three options below.

There are several methods:

Purchase and install a balanced transformer like and Equi-Tech, Tice or Cinepro.

Purchase and install a PS Audio Power Plant or Accuphase AC power generator.

Add a 220 volt outlet in your listening room.

The third option is the least expensive option of any we know of. To make 220 volts available in your home (if you live in a country whose standard voltage is 120 volts), the electrician will have to provide you balanced power, where each of the two legs (hot and neutral) have 117 volts on them, one out of phase with the other. This is balanced power.

What’s it cost to install? We phoned an electrician and he gave us an estimate of $350 to run a 220 volt service. This pricing, of course, depends on a number of factors: length of run and how you have them install it being the biggest factors. Nearly every home in the US has 220 volts available, it’s what your electric dryer or electric stove/oven operate from.

There are drawbacks to this however: the question of whether or not your equipment will accept 220 volts being the biggest. If you own or are contemplating purchasing a Power Plant AC generator, then there’s no problem at all. Inside the Power Plant, there is a voltage change switch that is easily accessible. If you take this route, you’ll then have balanced power coming into the Power Plant and balanced power coming out of the Power Plant. The same would be true for a properly designed isolation transformer as well.

Another drawback is the outlet. The wall socket of a 220 volt service is quite different than a 120 volt outlet, and you will have to devise a safe adapter for your male US plugs, or change the male end of your power cord to the appropriate 220 volt style. Whatever you decide, make sure others in your household are not fooled into believing the new outlet is a 120 volts and mistakenly plug the wrong thing in!

So, if you have a Power Plant, or if you have a piece of equipment that is able to be run from a 220 volt source, then this is a relatively painless, rather dramatic improvement you can make. Used in conjunction with the Power Plant, or a properly designed isolation transformer, a rather significant decrease in noise can be taken advantage of, if you live in the United States, Canada, Taiwan and Japan.

If you have money invested in AV equipment, you should be concerned about protecting it from surges and spikes generated both outside and inside your home.  No one is immune, even if you have whole house surge protection.

A common misconception is that we need surge and spike protection for our equipment to protect us from events that happen outside our home, like a lightning strike.  Those of us that live in areas of the country that do not experience lightning strikes are probably safe.  Nothing could be further from the truth.

In fact, the biggest generator of surges and spikes is right inside your home and no one is safe from it.

This is one of the biggest reasons why whole house surge protection isn’t effective for protecting our equipment.

Here are the facts.

• A small percentage of surges originate outside a house from nearby lightning strikes, which couple surges into nearby power wires.  This is the rarest of events
• Normal utility operations can cause electrical disturbances
• Perhaps the most common external surge source is when power is interrupted for any reason – a tree falling on wires, a car hitting a pole, wind damage, utility repairs, etc. Wires conducting electricity create a magnetic field. When power is interrupted, the magnetic field collapses, inducing large voltages in the wires. A 12-volt spark coil relies on this principle to generate many thousands of volts to fire spark plugs.
• 80% of surges come from within the home are generated every time equipment cycles on and off.  This is by far the biggest culprit.

Internal surge levels are related to the magnitude of current being interrupted and the length of wire from the power coming into our homes.

The longer the wire and the higher the current, the bigger the surge generated when the power is interrupted.

A classic example is a coffee pot located far from where the power enters your home. Every time the heater kicks on and off to maintain the coffee temperature, significant surges are generated that can affect connected equipment.

It should be obvious that a coffee pot cycling on and off several times an hour is a much more frequent event than a tree falling on the power wires, or a lightning storm.

These are the reasons where specific surge and spike protection are necessary.  Products that protect our equipment, such as the PS Soloist in-wall device , the Duet and Quintet in-line power devices as well as the Power Plant Premier, are important elements in our AV chain.

It probably does not make sense to leave thousands of dollars of AV equipment unprotected when the biggest source of trouble is right in your home.

Transformers hum because their laminations and windings rattle mechanically.  This is best known as mechanical hum.  This can be caused by several things, including poor construction and poor AC power conditions or both.

If your equipment has a noisy transformer, you’ve probably also noticed that it’s intensity varies depending on the time of day, sometimes even the time of month. The reason it varies is due, in large part, to the quality of the AC line voltage and how much DC is on it.

Why do transformers hum?
The short and simple answer is that many times, transformers hum because of an effect known as ‘lamination rattle’ or ‘winding rattle’ caused by DC voltage on the line.

‘Lam’ or ‘winding’ rattle occurs in all transformers to some degree, that degree being related to the quality of the transformer and the quality of the line voltage.

One of the problems we find on the AC line is when there’s an unwanted DC component.  DC (like battery voltage) leaks into most AC lines but its level is low enough not to matter.  In some cases, the DC level is high enough to wreak havoc with a power transformer.  When there’s DC on the line, it creates an asymmetrical magnetic field in the transformer which causes greater vibrations of the laminations and windings.  The laminations are ‘pushed’ together in one direction because of the DC.

To reduce these noises, transformer manufacturer have several tricks up their sleeves: they can varnish, or use super glue to stick the laminations together so they rattle less, and they can make bigger transformers that don’t have to work so hard, even in the presence of DC. The harder a transformer has to work, the more stress and strain is placed on the laminations.

But these measures don’t entirely solve the problem because you need to do that at the source of the problem, the DC on the line.

Why this is bad ?
When a transformer hums, it is actually physically vibrating or shaking inside of the chassis. This, in turn, shakes and vibrates everything else inside the chassis.

Many components in the chassis are sensitive to vibrations, including tubes and capacitors. In an even moderate case, this vibration can effect sound and picture quality as many of these internal components are microphonic and reproduce the humming into the audio or video signal.

Certainly the worst problem is the humming noise your equipment makes.  It’s downright irritating, especially if you’re sitting in close proximity to the humming equipment.

So, what can we do?
It’s best to fix the problem at the source.  Regardless of how well a transformer is made it’s best to keep the DC out of it.  Transformers with DC on them have core saturation problems, some amount of mechanical noise and lowered efficiency.

The Humbuster III lowers DC on the line and in most cases eliminates hum in transformers and, at the same time, helps the system to sound better.

It really depends on the internet radio station. Unfortunately, many of these stations use audio leveling software that removes dynamic range from music in an effort to make a more “consistent” and louder signal that can be enjoyed in the background without dropping out. It’s the same thing any background station does to keep a steady volume. Another reason they do this is so you don’t have one song loud, the next song so low you can’t hear it causing you to reach for the volume control constantly.

PS Audio builds all the audio systems for a restaurant chain in the States called Chipotle Mexican grills and we’re well aware of this problem. In fact, we’re working now on a way to slowly turn the music levels up and down without destroying the dynamic range – much the same way you would turn the level control up or down manually. Currently, these stores use leveling software that is sucking the life and dynamics out of the music and we’re trying to fix that.

There are stations, like Kitchen radio, that don’t use this leveling software and should be the same level as the CD played for that particular song.

In 2004, PS Audio invented the Gain Cell™.  The Gain Cell is a unique variable gain amplification stage that eliminates the need for volume controls, attenuators or potentiometers of any kind to adjust the volume in a preamplifier or integrated amplifier.

Gain Cells are superior to the use of a fixed gain amplifier or preamplifier stage with an associated attenuator.  Since PS Audio’s introduction of the Gain Cells, other high-end manufacturers have followed suit with their own versions of the invention: most notably Ayre and Revox.

To understand how a Gain Cell works and its benefits, it is first necessary to understand the basics of preamplification and attenuation in an audio product.

The basics

The heart of any audio product is a gain stage.  In a preamplifier, for example, the gain stage is needed to amplify a selected audio source such as a CD player, DVD player, tuner, or outboard phono stage.

Gain stages can be built with tubes, transistors, integrated circuits or any number of combinations of electronic components that have been used in countless ways over the years people have been enjoying audio and video in their homes.

Another example might be a power amplifier where there is also a gain stage.  In a CD player or DVD player; yet again there is a gain stage, amplifying the output of the disc.  In fact, every audio product from a surround processor to a simple receiver has at least one gain stage built in. Many designs require multiple gain stages.  The gain stage is the heart of all audio products.

For over 25 years, PS founder Paul McGowan has been working on the perfect gain stage.  A stage with Pure Resolution™ and no compromises.  Along the way this research has manifested itself in a multitude of high-end audio products, each with an improved version of the ubiquitous gain stage; unfortunately none perfect.

The problem with creating the perfect gain stage is the diversity of applications it must fit into.  One gain stage might work best on a DVD player, another could work in a preamplifier and yet another design is required for a power amplifier.  The perfect gain stage should be capable of being programmed to meet all these requirements in any design.

The perfect gain stage would have little to no distortion under any operating condition.   It would have no audible noise, it would be fully balanced from input to output, 100% analog, have exceedingly high common mode rejection, frequency response flat from 1Hz to over 50kHz, handle every input signal from the loudest to the quietest, high input impedance so you can drive it with a tube circuit, and its gain would be adjustable over at least a 100dB range.  That’s a pretty tall order.  But if you could create such a stage, then you could literally drop it into any audio product and create the perfect piece of equipment.

The perfect gain stage

We call it the PS Audio Gain Cell™.  The Gain Cell™ is the culmination of Paul’s 25 years of research and work, turned over to PS Engineering to bring the Gain Cell to life in its modular form.

Gain Cells are not based on the traditional gain block that uses varying amounts of feedback to control the gain as is typical in solid state and tube designs.  In fact, the Gain Cells are entirely different in their concept and operation and perform almost entirely in the current mode rather than the voltage mode.

The PS Audio Gain Cell is a single block of analog gain in a potted module.  You can place any size signal into the Gain Cell’s input, from any source (balanced or not), and the Gain Cell will handle that input with near perfection.  In fact, you can place up to 10 volts rms into the Gain Cell with no problem whatsoever.  For reference, a typical CD or DVD player outputs perhaps 2 volts rms, some Audiophile DAC’s can produce a whopping 5 volts rms, but the Gain Cell won’t care.

Once the Gain Cell has received the input, you can adjust the gain of the Cell anywhere from -100dB (essentially no gain) to +30dB instantly; and because the Gain Cell is 100% analog there are no steps in the gain.  This means that you can adjust the gain in increments smaller than a thousandth of a dB if you wanted to (a gain step this small is undetectable by the human ear).  Further, the performance of the Gain Cell does not vary with gain.  Wherever the Gain Cell is set it will perform identically, so from the loudest to the softest gain, the Gain Cell will sound identical.

The output of the Gain Cell can drive even the most inefficient of preceding stages with its fully balanced output of up to 5 volts rms and its low output impedance of less than 100 Ohms.

How the Gain Cell is used

Gain Cells can be used in virtually any piece of audio equipment as an input gain stage, an output gain stage, a volume control, an electronic input isolation transformer, a phono stage, and the list goes on endlessly.

Perhaps their greatest contribution in an audio system is replacing the volume control or attenuator.  With a Gain Cell, you no longer need listen through these degrading mechanical devices.

In a typical preamplifier once the input has been selected, the preamp’s internal gain stage amplifies the signal at a constant level.  You then need to add a volume control to throw away whatever signal you don’t want.  Even the finest volume controls add distortion and sonic grunge to the signal.  The GCP series using our Gain Cell is the world’s first preamplifier with no volume control, no balance control and no need to ever throw away any of the signal.

The Gain Cell can be controlled over a 130dB range which means that any input from any source, whether it’s a CD player, a DVD player, a phono stage or a tuner can play through the preamp at the very lowest level to the very highest level with no additional circuitry and no degradation.

What’s inside the Gain Cell?

Each Gain Cell is built around a Gilbert Cell, which is a similar to a voltage controlled amplifier where the audio signals all remain in the current domain until the very last point on the Gain Cell’s output.  Keeping the audio in the current domain, rather than the voltage domain, means that the Gain Cells maintain perfect linearity through the entire 100dB range the Cell enjoys.

Each Gain Cell must be hand tuned before it is encased in the epoxy.  After encasement, there are three trimpots used to tweak the Gain Cell’s performance on a per-unit basis.  These adjustments mean we can set each Gain Cell’s performance to within 1/10th of a dB between channels within any particular unit we choose to build.

Each Gain Cell is 100% analog.  That means there are no digital parts inside the Gain Cell.  Each Gain Cell is a fully balanced design from input to output with no coupling capacitors anywhere near the signal.  Gain Cells have one of the cleanest, purest audio paths in the world.

Gain Cells have impressive performance to say the least:

• >80dB input common mode rejection
• >0.005% THD+N @1 volt rms
• >-100dB SN A weighted
• Gain adjustable from -100dB to +30dB
• Step resolution >0.001dB
• Input impedance 47K for both +/- inputs
• Output impedance 100 Ohms for both +/- outputs
• Frequency response <-0.1dB 1Hz to 50kHz
• Direct coupled input to output

Gain Cells will be gracing most every audio product PS Audio manufactures.  Pure Resolution™ has finally been achieved in this ground breaking new technology, from the leader in Pure Audio and Pure Power™ products, PS Audio.

Sonically, it’s better to leave equipment on. The problem is tubes. While tube gear certainly sounds better after it’s been on for a while, tube life will be shortened if it is left on all the time.

With solid state gear (transistor), it is best to leave it on all the time, but a lot of people are afraid to leave their power amps on for fear of something happening that might blow up their speakers or amp or both.

We recommend leaving everything BUT your power amp on.

Also, bear in mind, anything with a remote control is probably not off anyway. It certainly isn’t COMPLETELY off; if it were, there would be no way for the signal from the remote control to be read.

Tubes: leave it off and warm up about 30 minutes before use. Transistors: leave it on always, unless it’s a power amp.

There are a number of options. You can get great results using iTunes Apple LossLess.

See also

http://www.hitsquad.com/smm/mac/CD_RIPPERS/