FAQ – Current models

Do Classé preamp/processors support DSD over HDMI, USB or Ethernet?

We do not currently support DSD nor do we believe it is necessary. All surround processors that say they support DSD over HDMI and virtually all outboard DACs that support DSD over USB or streaming simply convert the DSD stream to PCM before converting it to analog. Some D-to-A converter chips do offer a dual capability (the CP-800 and Sigma SSP use such DACs), but we believe conversion to PCM is better than letting the DAC handle the DSD stream directly.

There are many good reasons for this; among them are 1) the lack of processing tools for DSD (tone, EQ, bass management, etc. are all performed in the PCM domain); 2) the need to have some parts of the signal path be dedicated to DSD while other parts are dedicated to PCM (asynchronous USB and filtering to name two), thereby requiring duplication of circuitry and switching to select the appropriate path. The switches themselves compromise the signal, making it a questionable tradeoff; and 3) the questionable benefit of DSD over PCM. We don’t believe there is one.

It is a fact that DSD signals that are converted to PCM will retain their signature noise characteristics. In other words, you can convert DSD to PCM and it still sounds like DSD. The opposite is not true, however. Converting PCM signals to DSD makes them sound like DSD.

Most DSD recordings began as PCM or have some or all of their tracks converted for processing at some point along the way. There is no way to reliably tell if a recording has been maintained with all parts in the DSD domain throughout its creation (claims are often made about recordings that aren’t strictly true), but the number of recordings that meet that criteria is small.

If you have one of the rare pure DSD recordings and you want to play it back optimally, your best choice will be to use a high-end SACD player that is optimized for decoding DSD and output the signal to a Classé preamp/processor in the analog domain. Or even better, convert the signal to PCM and let us handle it digitally. It will sound great.

Better to do PCM right than just make a box that “does” everything. That’s our philosophy.

Can I use my Android device with the USB inputs on the CP-800 and Sigma SSP?

The USB Host connectors on our preamp/processors are dedicated to Apple iOS devices. Signals handled by this connector benefit from the same asynchronous processing as USB signals from a computer connected to the Device connector on the rear panel.

Are you going to build a music server?

While a music server with a Classé logo could be sold as a matching source component, we don’t believe it is the best product category for us or any other high-end audio company to add value. A music server is made of up essentially two or three basic parts: Memory, computer and interface. You can build a server with or without enough memory to store your music collection, but people often think of music servers as an all-in-one repository for their music. Let’s handle each of the three in turn:

  1. Memory or storage for your music files can be accomplished in numerous ways and the needs of our customers vary considerably. There is no single approach to storage that is best for all or even the majority of customers and there is no way for Classé to store data better or more accurately than any of the many computers and storage devices like NAS drives already on the market. Audio companies don’t’ add value with better data storage.
  2. Music servers rely on what is essentially a computer and operating system to translate the commands you give from the interface to data streaming to your preamp/processor. There is also the connection to your network and probably the Internet. If you think you want a Music server because you don’t want a computer in your listening room—think again. A server is just a computer that can’t do as much as other computers. High-end audio companies can’t build better, more accurate computers than Apple or any of the many other computer manufacturers out there. This is not to say that all computers are the same in their handling/processing of music, just that it’s the computer’s software that is the variable more than its hardware.
  3. The choice of a user interface for accessing your music collection is a personal one. We don’t believe there is a single user interface that is the best because what is best for you depends on your collection, how you think, what types of files you have and how and where your files are stored. Someone with a large classical music collection may want a different interface than someone who listens mostly to jazz or R&B. This is why there are so many different interfaces available. Can high-end audio designers design the best user interface for you? Maybe but is it the best use of their talent and resources and will you ever want to change?

The main point is that we believe the hardware (computer/server/memory and DACs) should be separate from the software or interface so you can be free to choose what you like and put them together. Then, when a new interface comes along next year and you want to try it, you’re free to do so without having to buy new hardware. Likewise, if a new preamp/processor (DAC) comes out that you want, why should you then have to change the way you interface with your music collection?

One additional benefit of using a computer rather than a dedicated server is that it allows you to do all the other things a computer can do: go online to check email or the score of a game; view photos from a photo library or watch YouTube videos or a Netflix movie. It’s a more powerful solution that can be customized to be what you want it to be, and the inexpensive (or free, like iTunes) interface program can be changed whenever you want to change it without having to change your audio hardware.

What is the difference between the CT and Delta series?

Although they have different industrial designs, new Delta and CT products share the same electronics and they sound identical.

AMPLIFIERS (CA/CT-M600/300, CA/CT-2300, CA/CT-5300)

What is special about the new Delta and CT amplifiers?

In a good system, the audio signal from a preamp/processor contains all the subtle nuances required for accurate and enjoyable music reproduction. It’s therefore the job of any amplifier to amplify this signal without adding anything or taking anything away.

Amplifiers typically do this in 3 stages – the input stage, driver stage and the output stage. The input stage buffers the tiny input signal, isolating the power amp stages from the output of the preamp and removing any high frequency parasitic noise that may have been added on its journey from the source. This tiny conditioned signal then controls a driver stage which boosts the voltage for use by the output stage transistors, where the tremendous amount of current required by the loudspeakers is supplied.

Our driver stage uses a completely new design which significantly outperforms its predecessors. The driver stage is exceptionally low distortion and nurtures the signal in preparation for amplification at the output stage.

The root of this performance is our “special sauce”, the way our engineers use parts which are available to all our competitors to create the best components money can buy. This is obviously secret Classé intellectual property but there are certain things that we can explain.

Essentially we have miniaturized the driver stage. This involves very small parts, very small signal paths, so it is very compact. By reducing the size of the circuit, we’re able to protect the delicate input signal from distortion caused by noise. All electronic components create noise as part of their normal operation, almost like a car engine creates noise when you turn on the ignition. This noise is particularly problematic in an amplifier, since it can be picked up by the audio signal as it passes through the amplifier and ultimately gets amplified.

A signal path is like an antenna that picks up noise. The longer the signal path, the greater its capacity to pick up noise. All our designs are extremely low noise but miniaturizing the driver stage further reduces its physical capacity to pick up any noise that we cannot eliminate. As a result, the driver stage outputs a more perfect representation of the original signal.

Output stage distortion is reduced by our stiff power supply. Dynamic music sucks huge amounts of power from the power supply and these sudden bursts typically ripple back into the power supply, like an echo of the original audio signal. The power subsequently flowing into the output stage contains the ripples of this echo, which the transistors add to the music that is actually playing. We call this “the boomerang effect” and it basically means that what you are listening to now is distorted by the audio you have just listened to, blurring the sound.

The CT and new Delta amplifiers have vast, controlled power supplies which cruise at maximum power. This means there is no boomerang effect. No matter how intense the music, our output transistors are always supplied with clean, undistorted power.

CT and New Delta amps are therefore low noise, low distortion and high power. They also sound superb. We started out wanting to design the best amplifiers Classé has ever built. We think we’ve ended up with the best amplifiers in the world.

Should I be concerned about the fan?

We believe we have designed the quietest fan-cooled design available. CT amplifiers are designed to be installed in a rack and have a unique heat management system, the ICTunnel. The original plan for our new Delta series amplifiers was to have a traditional heatsink, such as those on the original Delta series amplifiers. However we realized that the ICTunnel has significant sonic benefits in that it ensures that the amplifiers rapidly reach their ideal optimal operating temperature and then remain at that temperature regardless how hard they are driven. A traditional passive heatsink simply does not have this control. It heats up slowly during operation (the “warm up” period) and then just keeps on heating up to some temperature that is based on the environment and load. Unlike the ICTunnel, passive heatsinks have no way to ensure the electronics are in the perfect operational environment. In fact the passive heatsink’s only real job is to prevent the amplifier overheating and self destructing.

The ICTunnel has 4 temperature sensors that report to a microprocessor monitoring the overall temperature. This microprocessor adjusts the fan speed to keep the electronics at a constant temperature. The design is so efficient that the fan merely stirs the air when it is operational and it is certainly not a concern in terms of listening. Any concerns about the presence of the fan are allayed the moment you actually experience the product, so we encourage you to audition the amplifiers to not hear the fan for yourself.

Why are the amplifiers not specified to 2 ohms?

All amplifiers except the CA/CT-5300 have a virtually linear doubling of output between 8 ohms and 4 ohms but since it can be a misleading figure, we do not supply details of their performance at 2 ohms.

This by no means implies that our amplifiers cannot operate at 2 ohms, which they obviously can. Impedance varies with frequency and most high quality speakers have the capacity dip to 2 ohms. Since our amplifiers are specifically designed to drive the world’s best speakers, they will have no problem driving a 2 ohm load.

Nevertheless, driving 2 ohms in the real world requires an enormous amount of power, which is sometimes technically unavailable from the AC mains power outlet. This means that these types of specifications are vulnerable to external conditions, and can therefore be misleading. Clearly, we do not want our customers to make decisions based on potentially misleading information.

Why is the CT/CA-5300 not specified to 4 ohms?

While we can artificially supply the AC mains power required to create a 4 ohm specification for a 5300 in the factory, a standard domestic power outlet cannot provide the power involved. 3000W are required to drive all 5 channels at 600W at 4 ohms, which, at 60% efficiency, would require drawing over 5,000W from the wall. This is approximately 42 amps at 120V, which is simply not available in real world conditions.

In order to prevent the amplifier from asking more from the wall outlet than it can safely supply, the CT/CA-5300 protection circuitry on individual channels will cut in at 420W into 4 ohms to prevent the channels from drawing too much power. For test conditions to evaluate the actual capability of the channels, the protection can be removed, but that is also an unrealistic and misleading scenario. However, you should have absolutely no concerns about each channel’s ability to drive and deliver considerable power into 4 ohms loads.

Why don’t you build a 7 channel amplifier?

If you consider the physical size of our five-channel designs you can imagine that there are serious challenges to developing a practical seven-channel design of Classé quality. In addition to sheer size and weight, the available power from the AC mains also limits a seven-channel amplifier’s practicality.

Given these realities, we have no plans to develop a seven channel amplifier.

Anyone interested in a high-performance 7.1 system should invest in a stereo amp or mono blocks to drive the front left and right speakers. A five-channel amplifier could then be used to drive the remaining channels.

How much power should I expect Classé amplifiers to consume?

Take the rated power consumption specification, which is 1/8th power at 8 ohms. Halving the impedance doubles the consumption.

Standard circuits are 15 amp but should not be loaded more than 12 amps, which is 1440W (12A x 120V).

You should divide this 1440W appropriately based on your speaker impedance. However this represents much more power than you would expect to draw at normal listening levels.

Can CT and newer Delta series amplifiers be stacked directly on top of each other?

The CT amplifiers are designed to sit one atop the next in an equipment rack. The Delta series amplifiers have special feet with Navcom® inserts which will compress excessively when chassis are stacked. This compromises the performance of the foot but otherwise does not create a problem. The ICTunnel works by expelling the heat generated inside the amp out the back. The front and rear should not be obstructed since this would essentially defeat the thermal management system by preventing cool air from entering and hot air from escaping.

What is the total power capacity in microfarads of the amplifiers?

CA-CT-M300: 67,200 μF
CA/CT-2300/CA/CT-M600: 134,400 μF
CA/CT-5300: 224,000 μF

Please note that microfarad totals can be very misleading since you can have a lot of capacitors but use them inefficiently. Our efficient designs ensure that the capacitors are fully optimized, allowing us to achieve better filtering and power delivery than other designs with similar capacity.

How many rack units (RU) are the CT series products?

4U for everything (including the CT-SSP) except the 5300, which is 5U


Does the SSP-800 have HDMI 2.x or pass 4K video?

The HDMI 2.x spec contains numerous features and bandwidth capabilities, many of which are not yet possible

I sometimes see error codes on the SSP-800/CT-SSP and need to cycle power. Is there a problem with my unit?

Available soon.

Can I get multichannel audio on the HDMI out of the SSPs?

Available soon.

What are the SSP DACs?

The SSP-800 has 5 DACs each of which are dedicated to pairs of channels.

3 x Burr Brown PCM1792 DACs serve the Front Left and Right, the Center and Sub, and the AUX channels.

2 x Burr Brown PCM1796 serve the rears and surrounds.

The SSP-800 accepts incoming signals at sample rates of up to 192 kHz and bit lengths of 32-bit. Our DACs are also capable of handling these bit rates but their optimum performance range is 96/24. We therefore convert all signals to the optimum for the DACs.

Are they Apodizing DACs?

Pre and post ringing of digital filters have been an area of fascination for us for many years and we fully understand the capacity of apodizing filters to reduce pre-ringing. However, they tend to increase post ringing and add additional processing that we prefer to minimize.

Rather than focusing on a single element in the signal path, our system based designs ensure the quality of the complete signal path. The SSP-800/CT-SSP is a fine example of this philosophy and delivers what is widely considered ultimate performance with negligible pre and post ringing.

Should I connect a Blu-ray player via HDMI or the multichannel analog input?

Connect via HDMI. The SSP performs D/A conversion as perfectly as we think it is currently technically possible to achieve. No Blu-ray player’s analog output can compete with it. By using the HDMI path, signals remain digital and available for all of the processing features of the SSP without losses from additional and unnecessary digital/analog conversion stages.

We recommend the multichannel analog inputs be used with multichannel SACD players, if you have one. Converting the SACD to analog from its native form should give the best results.

Does the SSP-800 handle the raw DSD bitstream from SACD players with HDMI out?

No. We took this decision because DSD really is a unique. The best way of understanding this is by considering DSD in terms of bit length and sample frequency.

CD is generally 16-bit 44.1 kHz. DVD is 16-bit 48 kHz and Blu-ray is 24-bit and up to 192 kHz.

DSD is 1-bit at 2.8224 MHz. This is a structurally different bitstream and handling it correctly does not involve a single chip but a complete signal path from the HDMI input to the DSP.

The logical solution is to either convert the signal to PCM or perform the conversion of DSD to analog in a high quality SACD player, such as the type likely to be owned by a potential CTSSP/ SSP-800 owner. These players can then be connected to the SSP via its multichannel analog bypass.

How do I set up a home theater configuration?



This will take you to the CONFIGURATION SET UP screen where you should select a configuration from 1-6. Select


Set all the speakers SIZE to CROSSED OVER and select SUB ENABLED for the Sub.

In this configuration all frequencies below the crossover point will be directed to the sub. In cases where you are playing 5.1/7.1 audio, these frequencies will be added to the discrete .1 track, which is dedicated to the sub.

Note that you can name the configuration so you can easily identify it as your multichannel or home theater configuration.

You should assign this configuration to all your home theater A/V inputs. To assign a configuration to an input press:

MENU/ SYSTEM SET UP/ INPUT (press NEXT INPUT if necessary to advance through inputs) / CONFIGURATION and select the appropriate configuration.

You can also set up a configuration with all 7.1 speakers enabled and set to full range. You could set the subwoofer SUB ENABLED and also select E BASS.

If you have all your speakers set to full range with the sub enabled and have ebass engaged then a full range signal will go to all your speakers but the frequencies below the crossover point will also be duplicated in the sub. Therefore if you have your speakers set to full range, but with the crossover at 80 Hz, the frequencies below 80 Hz will be sent to the speaker and duplicated in the sub.

How do I set up a two-channel configuration?



Again this will take you to the CONFIGURATION SET UP screen where you should select a configuration from 1-6. Select


Set the FRONT LR speakers SIZE to FULL RANGE and disable all other speakers by selecting NONE.

In this configuration a full range signal will be sent to your front left and rights but all other speakers will be disabled, allowing pure two channel playback.

Note that you can name the configuration so you can easily identify it as your two-channel or stereo configuration.

You should assign this configuration to your two channel sources by pressing:

MENU/ SYSTEM SET UP/ INPUT (press NEXT INPUT if necessary to advance through inputs) / CONFIGURATION and select the appropriate configuration.

Should I set my speakers to full range or crossed over?

You should only set your speakers to crossed over if you have a subwoofer. However, even if you have a sub, you do not have to set your speakers to crossed over.

If you have all your speakers set to full range, with the sub enabled then you will only get sub output if you are playing multichannel audio with that contains a discrete channel for the sub.

If the multichannel audio is DTS or Dolby Digital 5.1, or Linear PCM from a Blu-ray disc, then the sub output level will be defined by the incoming material. We are simply converting the contents of the discrete .1 channel to analog and feeding it to the sub.

If post processing, such as Dolby PLIIx, is being used to convert two channels to multichannel, then the sub level should be quite low, if not silent, when the speakers are set to large. This is because the algorithm needs a crossover level to do bass management. Without a crossover, almost all the low frequencies will be distributed to the main speakers, not the sub.

If you have your speakers set to crossed over with sub enabled, all frequencies below the crossover point will be directed to the sub. In cases where you are playing 5.1/7.1 audio, these frequencies will be added to the discrete .1 track.

What is Ebass?

Ebass is a way of directing low frequency information to your sub even when your speakers are set to full range. This can sometimes be useful in achieving smooth bass response at the listening position.

If you have all your speakers set to full range, with the sub enabled and are playing 5.1 or 7.1 audio, then only the frequencies in the .1 discrete channel will be fed to the sub.

If you have all your speakers set to full range with the sub enabled and have ebass engaged then a full range signal will go to all your speakers but the frequencies below the crossover point will also be duplicated in the sub. If you are playing 5.1 material these frequencies will be added to the discrete .1 track.

Therefore if you have your speakers set to full range, but with the crossover at 80Hz, the frequencies below 80Hz will be sent to the speaker and duplicated in the sub.

What is a crossover slope and which slope should I apply?

To understand the crossover slope, imagine a straight line that drops off in a curve. The point when the curve begins would be the crossover point, which is where the frequencies begin to be filtered out. The sharpness of the curve would be the crossover slope, which would represent the speed at which frequencies below the crossover point are filtered. If you set the slope to 24dB it will be steeper than 12dB. A steeper crossover filters the frequencies beyond the crossover point faster.

Unless otherwise defined by a subwoofer manufacturer or suggested by a professional installer, 12 dB is the slope typically used.

What is the normal operating temperature of the unit?

The temperature sensor is located very close to the DSP, which generates a lot of heat during its normal operation. As a result, the normal operating temperature range is anything in the high forties to low fifties Centigrade.

I have a unit with a Dual DSP. Why is Dolby TrueHD/ DTS HD Master Audio not listed on the MODE menu?

Dolby True and DTS HS Master audio are lossless algorithms that carry HD audio, not post processing modes that add channels or manipulate the audio in other ways.

As such you cannot apply Dolby True or DTS HD Master audio to an incoming bitstream (basically since they are incoming bitstreams) and neither appear in the list of play modes.

The names will appear on the touchscreen when connected via HDMI to a Blu-ray player outputting Dolby TrueHD/DTS HD Master audio as an encoded bitstream.

What does Discrete mode mean?

Recordings are made with a discrete number of channels. A stereo recording has two channels. When you watch a movie with Dolby Digital, it is usually recorded with 5.1 discrete channels. Some Blu-ray titles with Dolby TrueHD or DTS HD Master Audio may include 7.1 discrete channels. When the discrete mode is selected, the SSP plays back the number of channels that are actually on the recording without using any post processing to fill unused channels.

If you have a 7.1 channel system and play a 5.1-channel recording with discrete as the playback mode, only 5.1 channels will be reproduced and your rear channels will remain off. To be sure that the rear channels of a 7.1-channel system are always on for 5.1 channel recordings, choose a post processing mode such as Dolby Digital EX or DTS Neo: 6.

Post processing modes can be used to automatically fill the rear channels by assigning them as the default favorite processing mode for multichannel recordings for each source. Press: MENU/system setup/input (advance by pressing next input until you reach the desired input), then press audio/fav. processing/multichannel, then choose Dolby Digital EX or DTS Neo: 6.

Why are some MODES greyed out?

Modes often require post processing and they are not always suitable or valid options for the type of signal being played. The more channels you have in the original material, the fewer post processing modes will be available.

Multichannel modes such as Dolby Pro Logic are designed to generate multichannel audio from stereo fewer. As such, you will notice that the list of favorite processing modes that may be assigned to 2-channel signals is much larger than the corresponding list for multichannel sources.

I have a 7.1 system but am not getting rear output all the time.

Apply a Dolby Digital EX or DTS Neo: 6 post processing mode. Multichannel audio bitstreams are decoded by the SSP into the discrete available channels. This basically “recreates” the channels present in the original recording. If the original material only has 5.1 channels, you need to apply post processing to generate the 2 additional channels in the rear.

Post processing modes can be used to automatically fill the rear channels by assigning them as the default favorite processing mode for multichannel recordings for each source. Press: MENU/system setup/input (advance by pressing next input until you reach the desired input), then press audio/fav. processing/multichannel, then choose Dolby Digital EX or DTS Neo: 6.

Does the CTSSP / SSP-800 have auto calibration?

No. We realized that manual calibration is relatively simple, involving measuring the distance from the listening position and setting uniform levels using an SPL meter. Manual calibration delivers the most consistent results, while auto calibration can be challenged by specific environments.

Does the CT-SSP / SSP-800 have an AES EBU input?

No. We decided against including it in order to dedicate as much as possible of the available back panel real estate to HDMI. We are confident that the performance of the SSP using HDMI or SPDIF digital connections will surpass your expectations.

How does the PEQ work?

What is often erroneously called “room correction” is based on parametric equalization, which is the use of filters to “correct” for the effects of the room itself on the overall amount of each frequency being heard at the listening position. As sound bounces around the room some frequencies add and reinforce each other while others cancel each other out. Moving the loudspeaker and/or the listening position can have a major impact on which frequencies are being reinforced and which are being canceled , as well as the degree to which this is occurring. You can think of the correction that is applied as countering the effects of the room by adding to or reducing the level of different frequencies to counteract the effects of the room. Therefore where the room introduces a peak, the PEQ cancels it out by introducing an inverse and equal trough.

Parametric is defined as the changing of parameters and there are three filter parameters to adjust: center frequency (which is the target frequency, usually the highest or lowest point in the peak or trough), the gain (which is how much you are attenuating or increasing the signal) and the Q which is the width or steepness of the filter shape (which is how sharply the filter affects the frequencies around the target frequency). Higher Q means a steeper, more narrow range of frequencies are being adjusted than with a lower Q setting. Generally, lower Q adjustments are more easily heard because they have an effect over a wider band of frequencies. Similarly, narrow band peaks or troughs in the natural room response are not as audible or troublesome as broad, lower Q peaks and troughs. Therefore, as a rule, adjustments to counter low Q, broad band room effects are the most necessary and desirable.

The standard EQ filter is a peaking filter which is what our PEQ is based on. We use these very flexibly since you have 5 filters or bands which can be combined or used individually for each loudspeaker in the system.

Is the PEQ automatic?

No, we don’t use automatic room correction. Anyone concerned that this decision will result in the SSP sounding worse than a competing processor with automatic room correction may be misunderstanding of the power of equalization to “correct” a room and the ability of automatic systems to achieve the best results.

The sonic performance of any processor is defined by its ability to maintain the resolution of the incoming signal. No equalization system, even a manual one in the hands of a master acoustician, can compensate for a loss of resolution.

In this context, a processor with perfect equalization but resolution loss will sound worse than a processor with no equalization but which maintains perfect signal integrity. This ultimately means that equalization is secondary to fidelity.

There are other limitations to consider as well. A speaker with poor off-axis response will have more distortion or worse frequency response for sounds that reflect off room boundaries compared with the direct (on-axis) sounds arriving at your ears first. Measuring the frequency response at the listening position can only reveal the total amount of sound pressure at each frequency and adjusting the equalizer to achieve a flat or target curve response cannot change how much of the total sound is reflected, where it is reflected from or the quality of that reflected sound. In other words, equalization cannot correct for the fact that your speakers may have poor off axis response or be poorly positioned in the room.

Imagine also a listening room with a very hard, reflective surface on the right side and a very absorptive, non-reflective surface on the left side. Together, the total amount of reflected energy might produce a smooth frequency response without EQ, yet the differences in reflection will detract from the fidelity of the system and be distracting to the listener. You could also imagine a room where the right side wall is very close to the listener compared to the left side wall. The frequency response at the listening position might be good without EQ, but the early reflections from the right may arrive so soon after the direct sound as to impair the clarity and fidelity of the system. There is no way to truly “correct” for these types of acoustic situations, so it is important to understand the limitations of equalization.

A final consideration on “room correction” is what is called time domain response. This refers to the time it takes for sounds to be reflected back to the listening position and the time it takes reflected sounds to decay in the room. Both vary by frequency and it is possible to make adjustments to try to counter time domain problems in the room, but like frequency domain equalization, the same types of acoustic limitations apply.

The conclusion is therefore to understand the importance of starting with high quality components and loudspeakers, position them as best you can in the room and deal with acoustic problems at their source before trying to employ equalization.

The implied promise of Automatic EQ systems is that a layman may set up a microphone, press a button and have a computer algorithm assess and then correct for the response of the room. As discussed above, whether automatic or not, there are real limitations to how effective “correction” can really be. So the question is not whether the room can be corrected, but within the limitations of correction, how is the best way to do it? Classé’s position is that the best results are achieved by employing experienced acousticians. Their knowledge and experience enables human judgment to supplement the measurements, helping them arrive at the final adjustments for best performance. By comparison, we see automatic systems which cannot arrive at the same correction curve twice, and where no two systems ever agree on the adjustments to be made. So which is best? Automatic systems may indeed improve the performance of an otherwise good, un-equalized system, but that is not the same as saying they can actually correct for the room response or optimize the performance of the system.

How do I use the manual PEQ?

There is no easy way to teach yourself to properly use the PEQ, which is why we recommend hiring a professional. If you use a spectrum analyzer and test tones or white noise, you might be able to pick out a few offending peaks and through trial and error, tame them, but to do it right you really need to hire an acoustician who’s got the tools, knowledge and experience. A lot of dealers are interested in the subject and some are self-taught and may be quite capable, but it’s human judgment backed by both knowledge and experience that ultimately will deliver the optimum results.

To access the PEQ press MENU/ SYSTEM SET UP/ ROOM EQ. You then highlight the speakers that you want to equalize and press select. You then have 5 bands to adjust per speaker. There are 3 values. G is the gain, showing how much you want to attenuate or increase the level. F is the frequency that you want to address. The Q value is the ratio of the center frequency to the 3dB bandwidth and is therefore frequency dependent. A table detailing the filter width for each Q value at each frequency is available from Classé.

Each band may be enabled/disabled. Similarly, the “activate group” function allows you to enable and disable the entire group of EQ settings that you have applied on that speaker. If the group is not activated then no EQ is applied to that speaker. Similarly, the “Activate EQ” button allows you to enable and disable the EQ setting for the entire system.

Given the limitations of equalization, it is generally agreed that the PEQ is best employed primarily to reduce the peaks in bass response called room modes. Adjustments below about 350 Hz are the most important and trying to address issues above that frequency may be unsuccessful or counter productive. From time to time a higher frequency rattle or vibration mode may need to be addressed by identifying the frequency and building a filter with very high Q to surgically remove the source of excitation. But for most applications it is recommended that you stay away from high frequency corrections and focus on the room modes.

The height, width and depth dimensions of a room often result in reflected acoustic waves arriving at the listening position and adding to or reinforcing each other at three specific frequencies related to the wavelength of the sounds corresponding to the various room dimensions. The PEQ may be used to attenuate the output of the system at those frequencies to reduce masking effects and improve overall clarity and intelligibility from the system.

The PEQ is used to help compensate for your environment (the acoustic interaction of loudspeakers and the boundaries, furnishings, etc. in your room). A single set of EQ settings is therefore applied equally to all processed sources, post processing modes and configurations. However the PEQ is a feature of the unit’s DSP and therefore does not affect analog inputs that are set to “Bypass Select,” or the multichannel analog inputs, which are hardwired in bypass mode.

Why is the PEQ gain limited to +3dB?

Variations in the frequency response caused by room acoustics are the result of many direct and reflected acoustic waves arriving at the listening position at the same time and either reinforcing or canceling each other. If they reinforce each other, it is a relatively simple matter to attenuate the output of the system at that frequency to achieve a net response which is more flat or closer to a desired target curve. If they cancel each other however, an entirely different problem exists.

Imagine the peak of a wave and the trough of another wave being the exact same frequency and amplitude, arriving at the same time. They would sum together and cancel each other entirely. If you amplified that frequency, you would just have two larger waves cancelling each other. So boosting a frequency is not an effective way to fill troughs in the signal caused by wave cancellations. For this reason, there is not much point in trying to provide a large range of gain for the PEQ on the plus side.

There are also technical considerations which impose a practical limit in the gain one would want to accommodate in a PEQ system. + 3dB is the maximum gain we can introduce without reducing the quality of a 24-bit signal. The only we way to add more gain would be to attenuate the incoming digital signal, effectively reducing the dynamic range. We do not want to do this since it would conflict with our overriding aim of preserving the integrity of the incoming signal.

How do I listen to headphones on an SSP?

You would connect a headphone amp to the AUX channels and set them for stereo downmix.

If the headphone amp does not have a volume control you would set them to Downmix variable. if the headphone amplifier does have a volume control you would set them to Downmix fixed.

To do this press:


Select an appropriate configuration and selecting:


Keep in mind that if you are using the AUX channels for stereo downmix, you need to set them up this way for each configuration you enable. Otherwise, you may inadvertently find yourself using a configuration that has no AUX channels defined and therefore no output to them.

Can I simultaneously connect the balanced and single ended outputs?

The SSP single-ended output is derived from the non inverted (or positive) signal path of the balanced output. Therefore if you connect both the outputs simultaneously, the non-inverted signal will be divided between the RCA and balanced outputs, while the inverted signal will remain unaffected. This may result in a slight quality degradation from the balanced output that could be audible.

What is Center Width and Dimension?

Center Width and Dimension are fixed at 3 and 0 respectively for all flavors of PLIIx except music mode where they can be varied.

Center Width makes the center speaker image smaller or wider by moving the mix of content between the Center and front L&R.

Dimension moves the surround image from front to back by moving the mix of content between the fronts and surrounds/rears.

There is no specific way to set them and it’s basically a personal preference. Some music may sound better with a more enveloping image and others may not.

What does “dial norm offset” mean?

Short for dialog normalization, Dolby offers a means to keep the average dialog levels about the same from program to program. A dialog normalization offset value is encoded in the Dolby bitstream and read by our DSP, which automatically makes the required level adjustment. It is a Dolby requirement that we provide a way to inform customers when they are doing this, which is why the “dialnorm offset” message sometimes appears. The OSD message can be disabled by pressing


Is the SSP THX certified?

No. We recognize THX’s historical role in addressing some of the common problems in early home theater technology, setting minimum performance standards and assuring some consistency between home and movie theater sound reproduction, especially in the days of analog multichannel. The dominance of discrete digital multichannel recordings today makes using proprietary THX algorithms for analog matrix modes unnecessary. Meeting the certification standards is not particularly difficult or expensive, as evidenced by the many low-cost components that carry the logo. Today, we find that customers at the Classé level no longer demand or expect THX certification as a means to assure that a minimum quality standard has been met, so we don’t pursue it.

Does the SSP handle I2S (I squared S)?

The HDMI specification requires that audio be carried as multichannel I2S and SPDIF. Since the SSP is fully HDMI compliant, it absolutely handles both these signals.

Does the SSP handle WAV or FLAC files in their native form?

Our DSP cannot handle media files such as WAV and FLAC in their native form because this type of software is designed to function within the Windows Operating system and the SSP is not Windows compatible. Connected computers with files stored in these formats will therefore convert the FLAC file into Linear PCM before outputting it to us over optical or coax connections.

Do CDs sound better via HDMI?

HDMI has five audio signal paths. Four of these are dedicated to stereo pairs of Linear PCM channels which combine to create the eight channels required for 7.1 audio. These pairs can also handle bitstream HD content, such as encoded Dolby TrueHD or DTS HD Master Audio.

The fifth signal path is dedicated to SPDIF audio, which is same signal that passes through a COAX or Optical cable. It handles CD audio and legacy bitstreams, such as DTS and Dolby Digital 5.1.

CD audio transmitted via HDMI is therefore decoded by the SSP into the same SPDIF signal as CD audio transmitted via COAX or Optical. All things being equal, there should therefore be no perceptible difference between a CD played by COAX/Optical or HDMI. If the player introduces problems on either output, then you might find one or the other better. The only way to be sure which you like better is to listen.

What is the maximum video bit length that the SSP can handle?

The SSP-800 is deep color compatible and can carry video signals of up to 16 bits per color, representing a total color depth of 48bits.

DVD video is 8-bit per color (RGB), representing a total bit length of 24 bits. Blu-ay is up to 12-bit RGB, representing a total bit length of 36 bits.

Please note that 48-bit video technically exceeds the depth of perception of the human eye and to experience this quality you will need a 48-bit player and display, together with 48-bit video material. None of this is commercially available.

Does the SSP have video processing/scaling?

No. High quality video scaling has become a feature of all modern digital displays where the performance of the video processor is tailored to the needs and capabilities of the display. Further, most critical video sources arrive at the SSP at HD resolutions and do not require additional scaling. These factors render an internal video scaler feature redundant to the SSP and when executed at a Classé quality level, would add significant cost and complexity without adding commensurate value.

How can you have an OSD if you don’t have a video processor?

An OSD is normally generated by an internal video processor, which we believe is not required or desirable in a high quality SSP. We provide the OSD feature in the chip (called an FPGA) where digital video signals are mixed and switched. Temporary messages are generated by copying the portion of the touchscreen display that shows the message and overlaying it onto the video signal. This feature is available for video signals of all resolutions. The large OSD is generated in the same way, copying the whole touchscreen rather than a portion of it.

Can the SSP convert HDMI video into Component video?

No. This is forbidden by license agreement because it would allow rendering a non-copy-protected HD video signal from the copy-protected HDMI source.

Can the HDMI 1.4 version of the SSP handle 4K video?

No, but it will be a while before things settle down enough to worry about switching 4K or UHD video. There are some display devices now available that can display 4K video but almost no content to play on them. There are no disc players or even a next-generation Blu-ray spec, no broadcast standards, no game standards, etc. so while little bits of content may appear from one place or another they don’t conform to a standard that you can say will be used a few years from now.

There are some Blu-ray players offering 4K video scaling, which is absolutely the wrong thing to do. There is no reason to burden the entire HDMI signal chain with all that extra made up data when a 4K TV has its own scaler already built in to do that anyway. You should send signals to the display at their native rate and let it do the scaling.

So to summarize, there aren’t really commercial sources of 4K to switch at the moment. Anything you find that does have native 4K content can be connected directly to the display for video and directly to the SSP for audio. The HDMI, coax and optical inputs on the SSP-800/CT-SSP can handle all the audio formats that the unit is able to process anyway, so it will be a while before switching 4K becomes a real issue.

I am having problems with the DC trigger.

If you have set the trigger to STANDBY OPERATE and not activated INVERSE LOGIC then a 12V signal will be sustained at the trigger output from the moment you bring the unit out of standby to the moment it is put into standby.

If you activate INVERSE LOGIC, 12V will be sustained when the unit is in standby and it will revert to 0V whenever the unit is brought out of standby.

The trigger design is very simple and has proven to be solid over time. If you have problems with the trigger on a specific component then you should test it with another component and ensure that it is working properly. If it does you should check the nature of the trigger on the problem component and confirm that it is compatible with our design.

Our triggers have capacitors at the output which are drained when connected to a source (or load). If you measure the output without a load the small voltage from these capacitors will be present.

I am getting a loss of audio and video at resolution changes.

While it may seem logical that content should only be interrupted if the HDCP negotiation fails, the link also has to be reset for other changes, such as clock frequency.

Changes of clock frequency do occur at resolution changes, so audio and video will be interrupted at this time. This performance also affects other HDMI components, since everyone has to work within this architecture.

The only way to eliminate the interrupt is to keep the links alive continuously, so the source always thinks it is connected to a display, even when it’s not. This type of technology will become increasingly mainstream in the future but it is not a feature of the current CTSSP/ SSP-800 hardware.

How does the noise generator work?

Any noise generator needs to supply a uniform output level from all speakers. The SSP outputs pink noise at a uniform reference level. You can then tailor the output so every speaker outputs the same perceived level in the listening position. An inexpensive SPL meter set to C weighting and slow response can be used to set levels. When levels are the same, the SSP knows whether and how much attenuation to apply to each and every channel to achieve the correct playback balance from the recording.