FAQ – Discontinued models


Can original Delta Series amplifiers be stacked on top of each other?

No. They generate a lot of heat in normal operation which they are designed to manage but they do require certain minimum conditions to do so.

Units must always be properly ventilated and placed in accordance with the unit placement instructions. You should allow 6″ clearance above each amplifier and 3″ on their sides.

Why are they not specified into 2 ohms?

While we can artificially generate the power required to create a 2 ohm specification in the factory, a standard domestic power supply cannot provide the power involved. If we were to quote this specification, we would therefore be detailing a performance that the customer could not expect to achieve when actually using the component. This is potentially misleading and we clearly do not intend to mislead our customers.

However, you should have absolutely no concerns about our amplifiers ability to properly drive speakers that dip to very low impedances. Impedance varies with frequency and most high quality speakers have the capacity to dip to 2 ohms or even lower. Since our amplifiers are specifically designed to drive the world’s best speakers they can easily meet the demands involved.

What is the total capacitance in micro farads (µF) of the original Delta series?

CA-M400 has 24 x 5,600 μF capacitors with a total of 67,200μF

CA-2200, CA-3200 and CA-5200 have 8 x 5,600 μF per channel

CA-2100 has 6 x 4,700 μF per channel and CAP-2100 has 4 x 4700 μF per channel

CA-5100 has 3 banks of capacitors each of which have 5,600 μF for a total of 16,800 μF.

How many output bipolar output transistors per channel?

The CA-M400 has 32.

All other amps have 16 per channel, except the CA-5200 center channel which has 12.

What is the Slew rate in volts per micro second (V/µs)?

CA-M400 rise time is 1.82 microseconds, and it reaches 140V in this time. This equals 74.7V per microsecond.

CA-2200/CA-5200 rise time is 3.3 microseconds and it reaches 130V in this period. This equals approximately 39V per micro second.

CA-2100 rise time is 2.1 microseconds and it reaches 90V in this period. This equals approximately 42.5V per micro second.

What is their standby power consumption?

All original Delta series amplifiers consume 4W in standby


What is the difference between the CP-500 and CP-700?

While the CP-500 and CP-700 are both extremely high quality components, they are also profoundly different. The CP-500 design places an emphasis on integrated circuits (ICs), which are mirco-chips. The ICs deployed in the CP-500 are very high quality and designed for audio applications but they have specific functions, which cannot be modified by our engineers. In the CP-700, many of these ICs are replaced by discrete circuits which have been specially designed by our engineers. This approach is more cost intensive but it gives our engineers more freedom to influence the quality of the sound. The CP-700 amplification section is an example of this. It is a pair of boards that replace two ICs in the CP-500 and which are considered to be among the highest performing boards we have ever built.

In addition to its external power supply, the CP-700 also makes more sophisticated use of balanced circuitry, and has three independent paths per channel (one for single-ended, one for balanced inverting and another for balanced non-inverting) thus keeping the single-ended and balanced paths entirely independent. Each of these three paths is itself handled as a balanced path requiring inverted/non-inverted paths for each, so there are actually six signal paths per channel in the CP-700. The CP-500 has only two signal paths per channel (balanced inverting and non-inverting), so single-ended signals share the same circuitry as balanced signals.

These fundamental differences result in different sounding products, with the CP-700 is the more costly and higher performance of the two.

How can I connect the CAP-2100 or CP-500 or CP-700 to a sub?

On the CAP-2100 you should connect the pre-amp outputs to your sub. Use the subwoofer’s internal level and crossover adjustments to tailor its output for the best mix.

The CP-500 has 2 pairs of outputs, one balanced and one single-ended. However they are not designed to be used simultaneously, so if you connect a sub to the single ended output while using the balanced output to connect to your amplifiers, you may notice a loss in performance from the balanced outputs. The CP-500 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 will result in a quality degradation that you may find audible.

The CP-500 does have a tape output that you can connect to an active sub via RCA but this will mean that you have to manually adjust the Sub volume control every time you change the source volume.

Certain active subs can also be connected direct to the speaker inputs, without degrading the system performance. However, you should confirm that the sub is suitable for this configuration before investing in it.

Finally you can use Y cables to connect the CP-500 to both an amplifier and a sub with stereo inputs. If you are using RCA Y cables they have to be signal to signal and ground to ground. If you are using XLR cables then you need inverted to inverted, non-inverted to non-inverted, and ground to ground. Essentially you are duplicating the signal path on both ends of the Y cable.

The CP-700 has independently buffered outputs. This means that each output is completely independent and both single-ended and balanced outputs can all be used simultaneously without degradation.

How should I use the Input Offset?

Adjusting the input offset has no effect on the overall imaging or sonic performance. It basically boosts or reduces the level on the input compared to other inputs. So if you adjust the offset by +6 on a specific input and play it at 72, the level would be the same as if you left the offset at 0 and played it at 78. Choosing an offset adds nothing to the signal path. Internally, the master volume control is simply changing by whatever the offset is to help keep all your sources at the same relative level without you having to manually adjust volume every time you switch inputs.x

How can I integrate a Classé Delta series preamp or integrated product into my Home Theater system?

Connect the AVR or SSP Front Left and Right outputs as if it were a source to any input the Classe Delta series CP or

CAP. You need to rename the preamp or integrated amp input that’s connected to the AVR or SSP with the letters ssp. You would connect the preamp outputs to the amps which serve your front Left and Right amplifier channels (or speakers if using an integrated amp), while the other channels would be connected to the amps which serve exclusively home theater channels.

Switching to the SSP input on the CP or CAP will automatically disable the volume control and set the input to unity gain, which essentially means that what comes in goes out. This means that when you need multichannel audio you switch to the SSP input on your preamp/integrated and it acts as a bypass. Switching to another input on the preamp/integrated will mean the front Left and Right channels will revert to handling the selected 2-channel audio source.

When listening to 2-channel audio you would power up the CAP-2100 and sources. When listening to Home Theater you would power up your entire system.


Is the SSP-300/600 compatible with Dolby TrueHD or DTS HD Master Audio?

No these codecs can only be transmitted via HDMI and require different digital signal processing capabilities. The SSP-300/600 was designed before HDMI was developed and is therefore not HDMI compatible. You can use an Optical or COAX digital connection but the Blu-ray player will only output standard Dolby Digital or DTS 5.1 audio.

If your Blu-ray player has multichannel analog outputs you can connect the player to the SSP-300 analog 7.1 input. The player will then decode the HD codecs, convert them into analog and send them to the SSP-300. However, since this is a bypass input, you will not be able to apply any additional processing to the audio.

What are Configurations and how do I set them up?

Configurations are an enhancement to what was originally called the Listening Positions feature. Configurations allow the creation of several completely different speaker set ups to accommodate either different listening positions/locations or different speaker arrangements. For instance, you could set up two different speaker configurations, one for stereo music playback with only the front Left and Right set to large and the sub and other speakers deactivated, the other for pure home theater, where all the speakers are active and set to small so that low frequencies are sent you your sub.

The configurations were an element of our ultimate SSP-600 code 1.5.3 and regrettably their operation is not a feature of the SSP-600 manual. To check the software your unit is running, press MENU/STATUS/VERSION INFO and then check the FIRMWARE version. If it is less than 1.5.3 you will need to update your unit using the software on the website.

If your unit is running 1.5.3 then you can set up a configuration by pressing:


You will see four configurations, each of which can be set up to have a unique combination of speakers, levels and/or crossover settings. Logically you should select CONFIGURATION 1 then calibrate as required. You should also note the MORE button in the top right corner of the CONFIGURATION 1 screen. Pressing MORE will take you the “speaker config” which will allow you to set the system crossover and speaker sizes. Large means a full range signal is being sent to the speaker. Small means that only those frequencies above the crossover point reach the speaker, with the lower frequencies being blended with the content of the .1 channel and being sent to your sub.

In addition to being available for any input at any time, configurations may be assigned to be engaged automatically whenever an input is selected. Once you have created the configurations you require simply apply them to the desired input. Press MENU/SYSTEM SET UP/INPUT then advance to the appropriate input. Then select CONFIGURATION and apply the desired configuration. Whenever this input is selected, that configuration will be automatically applied.

Can I apply a Configuration to the 7.1 input?

Bass management, configurations, speaker settings and any type of post processing are a feature of the DSP and are therefore unavailable on the 7.1 bypassed input.

Level settings are a feature of the volume controls, not the DSP. You can therefore make level adjustments to a configuration and apply them to your 7.1 input. You would apply this configuration to the 7.1 input by programming an Fkey to switch to this configuration whenever required.

To program an Fkey press MENU/ REMOTE FKEYS/ select Fkey/ assign CONFIGURATION

What is the meaning of the MULTICHANNEL mode?

The Multichannel mode is designed to select the multichannel processing that the disc authors have applied as their default processing. Therefore if a DVD disc is authored with Dolby Digital 5.1 as its default processing, the Multichannel mode will apply Dolby Digital 5.1 when playing the disc.

What DACS are used in the SSP-300/600?

The SSP-300/600 DACs are 4 x AK4393 for multichannel sources and 1 x AK 4524 which deals with 2-channel sources.

Can I get Zone or REC Output for digital sources?

No. The zone/ REC is restricted to the analog inputs.

I am getting distortion on my component video output.

The first thing you should do is set the video input to bypass by pressing MENU/ SYSTEM SET UP/ INPUT (advance to DVD input using NEXT INPUT if necessary) VIDEO/ BYPASS/ then select the component input you want bypassed.

The SSP-300/600 video path is designed to handle standard definition video (480i/576i). The Component Bypass feature bypasses the normal video path (which includes the touchscreen and OSD generator) and allows the SSP-300/600 to handle the added bandwidth of High Definition video. For component video this is generally up to 1080i.

However the component bypass is a pure bypass and it does have certain limitations. OSD is not available, since it is generated by the video chip, and it also doesn’t have Automatic Gain Control (AGC).

I sometimes get flashes of brightness or “video blooming”. Should I send the unit for repair?

You are probably using the SSP-300/600 component bypass feature, since the performance is only occurring in certain scenes.

If you are not, you should activate it by pressing MENU/ SYSTEM SET UP/ INPUT (advance to DVD input using NEXT INPUT if necessary) VIDEO/ BYPASS/ then select the component input you want bypassed.

However the component bypass is a pure bypass and it does have certain limitations. OSD is not available, since it is generated by the video chip, and it also doesn’t have Automatic Gain Control (AGC).

Like all video processing chips, the SSP video chip is equipped with AGC, which adjusts the gain on signals that are beyond its acceptable amplitude. Since the component bypass does not go through the video chip, video signals that pass through it are not subject to AGC. Therefore if the signal is beyond the circuit’s acceptable amplitude, they will simply clip. This clipped signal will then be sent to the display.

Since the display’s video processing chip will have AGC, it will adjust the gain on any input, which is why the issue does not occur when the DVD is connected direct to the display. However, there is nothing it can do when it is receiving an already clipped signal from the component bypass.

This performance is therefore a byproduct of the bypass feature and the unit does not need repairing.

I am not getting any audio from an RCA analog input

RCA analog inputs are hardwired so if you connect an analog component to RCA input 1, it will be hard wired to serve input 1. However, digital inputs take precedence over analog inputs, so if you have assigned inputs 1-4 to be digital, you have effectively disabled RCA inputs 1-4.



My amplifier has gone into protection. What should I do?

The amplifier is in protection If the LED is green but this is not necessarily a sign of a defect on the amplifier.

Completely disconnect the unit from all power sources and other components and leave the unit off for at least an hour. This will allow the capacitors to drain and the unit to reset if it has gone into protection.

If the amp has a problem it will generally go into protection immediately on power up with nothing connected except the line cord. If the unit is a CA-200/201 try this a few times since the reset is sometimes not immediate.

If you are able to power up the unit successfully then the protection circuitry is probably engaging due to a connected component with an electrical issue which is outputting DC.

You should systematically rebuild the system and identify which element in the system is making the amp go into protection. This element should then be tested.

One amplifier or one side of my amplifier is hotter than another. What should I do?

Ensure that the amplifier is equally ventilated on all sides (or mono blocks are located in similar environments)

If the difference in temperature is not due to environmental factors, disconnect the amplifier from everything and leave it to cool. Connect the AC line cord only then power up. If one side of the amplifier is hotter than the other the bias may have shifted on one channel. The bias sets the amount of current passing through the channel output transistors and it can shift over time. If it has shifted to allow more current to pass then the channel will run hotter as a result. If it shifts to allow less current to pass the amp will run cooler. If the bias has shifted you can download the service manual for the amplifier (from this site) and you can take it to a local technician. Alternatively you can send the unit to us to have the bias adjusted.

If both channels (or amps) run at the same temperature when connected via the AC line cord only, then the amp is heating up due to an external factor and the amplifier has nothing wrong with it. Systematically reconnect the system until you identify the component or cable that is making the unit heat up.

One channel is quieter than another. What should I do?

At the moment you have a level drop in one channel but you do not know where the problem lies.

In order to ascertain whether the problem is on your preamp/processor or amplifier, simply swap channels connections between the two. Connect preamp/processor outputs for the channel with reduced output to a channel which is currently working fine.

If the problem moves to the previously working channel, then you know it relates to the preamp/processor output, the cable or the source component. In order to investigate the possibility that the issue is related to the source component, you should confirm whether the problem is consistent regardless of the selected source. If the problem occurs on one source only, swap channel connections between source and the preamp/processor. If the problem switched channels, then you know the source or its cable is the problem. To check the cables, swap them from the opposite ends to see if the problem moves with the cable or stays with one channel of the source.

If after swapping connections to the amplifier the previously working channel continues to operate normally then you know the issue is related to the amplifier, speaker or its cabling. You should connect a working speaker and cable to the problem channel. If the problem remains you should send the unit for repair.

I am getting hum from my speakers. What should I do?

If the hum is due to the amplifier, it will be present even if all source components are disconnected.

Disconnect everything except the line cord and the speakers from the amp and see if the hum is at the same level. Please note that the nature of our Legacy design means that there should be a small hum when no inputs are connected. If the problematic hum is not present when the amplifier is only connected to the speakers then systematically rebuild your system until the hum appears. You will then be able isolate the source of the problem.

I am getting hum from inside the unit. What should I do?

This is probably a sign of DC or noise on the AC line. The transformer is designed to operate at 50 – 60 Hz, which means that the current serving the transformer switches from positive to negative between 100 and 120 times a second.

Interference on the AC line into the amplifier can create a higher frequency signal, which massively increases the number of times the current in the transformer switches from positive to negative. This vibrates the transformer, causing it to hum.

DC has 0 Hz, which is significantly below the transformer’s performance range, and can place high loads on the transformer, causing it to hum.

Both of these issues relate to the external power supply and are not a sign of a fault on the amp.


Should I leave my Classé products on all the time?

Generally we don’t recommend leaving amps on all the time, which is why we design them with easily accessible power buttons. The major reason for this is because it is not environmentally friendly.

Capacitors do age and the rate of aging is affected by the way they are used. Turning units on and off stresses capacitors, which logically means you should leave amplifiers on all the time to extend capacitor lifespan. However, the heat generated when the amplifier is on also stresses capacitors, which logically means you should switch them off when not in use.

What this essentially means is that there is no absolutely correct approach to extending capacitor lifespan.

To be environmentally friendly, you should turn your amp on about half an hour before you intend to listen and turn it off when you’ve finished. This reduces energy consumption and allows the amplifier to reach a temperature that delivers both optimum sonic performance and reliability.

Do you recommend any cables, power conditioners or surge protectors?

We only provide information and recommendations for products that are designed by Classé and do not recommend specific brands or types of accessory product. If you are interested in investing in these types of products, your dealer is ideally placed to help and you should discuss it with them.

As with any audio product, we highly recommend that you audition it in your system first and trust your own ears.

Do you recommend XLR cables (also known as Balanced cables) over RCA cables?

Yes, assuming otherwise equal quality levels. XLR cables use differential designs and, as their name implies, they only let differences pass. They use two signal paths, inverted (negative) and non-inverted (positive), plus a ground. The inverted and non-inverted signals are exact reflections of each other, so where one has a peak, the other has a trough.

While the non-inverted and inverted signals are exact opposites, any noise picked up as the signals pass through the electronics may be common to both. The noise will be added to both the inverted and the non-inverted signal paths, affecting both almost identically.

Our differential designs are highly effective at only letting the exact opposite signals to pass. Since any noise will be the common to both signals, it is rejected (or subtracted away—remember “differential”), producing a very low-noise design. Technically, this is called Common Mode Rejection since the noise, which is common to both signal paths, is subtracted away and rejected.

Single-ended signals use a single conductor in the cable which typically also uses a shield wire to make a ground connection. The shield cannot be 100% effective, so noise picked up by the signal conductor becomes forever part of the audio signal is amplified through the signal path.

Most Classé designs offer both balanced and single-ended signal paths. The balanced path has 6dB higher level relative to the same signal using a single-ended connection. That’s because the balanced (or differential) signal is really the difference between two signals the same size as the single-ended signal. A +2V single-ended signal would become a +4V balanced signal (two volts minus a negative two volts = plus four volts). That equates to a 6dB higher level, or 6DB higher signal-to-noise ratio. All things being equal, this and the general advantage of common mode noise rejection tip the balance in favor of balanced connections.

I want to create an RCA to XLR convertor cable

RCA has only 1 signal path and ground, so if you are creating an RCA to XLR cable, you should connect Pin 1 as ground, Pin 2 as the signal and leave Pin 3 disconnected.

The primary benefit of XLR designs is noise reduction through common mode rejection. Since the original RCA signal has only a single signal path and common mode rejection needs two, exactly opposite signal paths, you will not be benefitting from this if you use an RCA to XLR connector.

Adapting from one to the other is useful only as a matter of convenience for using cables on hand or for connecting two components where one has only RCA and the other only XLR connections. There is no sonic advantage to using either the RCA or the XLR connector if a proper conversion stage is not employed to convert between the two signal types.

I want to create an XLR to RCA convertor cable

XLR has 2 signal paths and ground. The pin configuration convention on an XLR connector is Pin 1 ground, Pin 2 positive, Pin 3 negative. The two signals carried are literally the inverse or opposite polarity of each other.

RCA has only 1 signal path and ground, so if you are creating an XLR to RCA cable, you should connect Pin 1 as ground, Pin 2 as the signal and leave Pin 3 disconnected. If you connect both XLR Pin 2 and Pin3 signals the audio will cancel itself out.

Adapting from one to the other is useful only as a matter of convenience for using cables on hand or for connecting two components where one has only RCA and the other only XLR connections. There is no sonic advantage to using either the RCA or the XLR connector if a proper conversion stage is not employed to convert between the two signal types.

What is the longest HDMI cable I can use?

The HDMI specification does not dictate cable length requirements but their compliance testing expects the ability to drive a maximum of 10m. Although numerous SSP systems are driving longer cable lengths, we cannot guarantee this will be possible in any specific installation. This is because the SSP-800 is not the only factor defining how long a cable can successfully carry an HDMI signal, the receiver chip inside the TV or projector also plays a major role. Receiver chips that include a feature called “cable equalization” are able to compensate for weaker signals thereby extending the potential length of any cable that is used with that device.

Maximum cable distances also vary depending on the types of signals carried. A 1080p signal with HD audio requires the most bandwidth and therefore taxes the maximum distance most. HDMI signal extenders are available for systems where unusually long runs are required. Your dealer can help select the right cables and equipment for your application.

HDMI recommends that systems be tested before installing.

What are the important factors in selecting digital cables?

An important factor on digital cables is the impedance control. For all digital cables except AES EBU, the impedance across the length of the cable should be 75 ohms. For AES EBU cables the impedance control should be 110 ohms. There is a limit to the value of impedance matching in this context, however, because the RCA and XLR connectors used do not (in fact they cannot) conform to the 75/110 ohm specification.

Should I worry about impedance matching?

Precise ‘impedance matching’, where specific impedances (often 50 or 75 ohms) must be adhered to, is correct for radio frequencies, where cables above a meter or so act as a transmission line. But at the highest audible frequencies (20 kHz) even a 200m long input cable doesn’t behave as transmission line.

You should therefore not be concerned about impedance matching for analog signals for your system.

What is a monaural amplifier?

Mono recordings have just one channel of audio. If you connect two speakers and have a mono signal, the same audio will come out of both speakers.

Stereo is two-channel audio. This means that different audio signals can be sent to the left speaker than the right speaker. When these two channels are played simultaneously they can create the illusion of a stage between the speakers that you are doubtless familiar with.

Home theater, such as Dolby 5.1, is multichannel audio. This means that different audio is sent to five or more speakers, creating an enveloping soundscape that is designed to make media true to life.

Monaural amplifiers are single-channel amplifiers. Therefore a stereo system would require two monaural amplifiers, one for each channel. For multichannel home theater, you would need additional amplifiers depending on the nature of the system you wish to create.

What are the advantages of monaural amplifiers?

All things being equal, monaural amplifiers are generally accepted to deliver ultimate performance. There are three advantages available from monaural amplifiers.

  1. Only one channel of audio is present in the chassis, so signals from other channels cannot bleed into and mix with its signal. In a stereo or multichannel amplifier, channel separation is the specification which measures the degree to which adjacent channels may interfere and mix with one another. In a monaural amp, the separation is maximized.
  2. Only one channel is connected to the power supply, so all of the power is available for the exclusive use of the channel and the demands of adjacent channels for power can have no effect.
  3. Since monaural amplifiers have only one channel, they may be located near the speaker to which they are connected. It is generally accepted that longer speaker wires (delivering current) compromise sound quality more than longer interconnect cables (delivering voltage), so monaural amplifiers allow the minimum length for speaker cables.

Is Classé designing a Blu-ray player or another type of next generation source?

No. We take pride in building products that offer both high quality and value. It is tempting to offer high-end products in some of the latest categories. We know we could deliver on our promise of quality, but are less certain of the value proposition. The honest answer is that there is not much to be done to a basic Blu-ray player or hard drive storage system to improve its performance to the point that would justify a 10x or 20x price premium.

There are certain places where high-end designers can apply innovative technologies and more expensive parts to deliver justifiably higher performance. For the most part, these are areas where data are processed, converted to analog and amplified. In other words, high-end designers can add real value in the preamp/processor and power amplifier categories and that is where our product development efforts are currently focused.

Will Classé make a music server?

No. Like the Blu-ray player, we believe the opportunity cost of developing a server would detract from our efforts in areas where we can add more value.

A music server has three components: interface, data storage and signal delivery. Large software corporations are in the best position to develop compelling user interfaces because they can amortize the cost over millions of customers. Cheap, reliable data storage solutions are available from the consumer electronics giants. We don’t add value in these areas and are not going to become a computer company or sell to the masses.

Classé is all about sound. What we want to do is pull your audio off a network, or from wherever it is stored, and make it sound better than anyone else. We want to add value by acquiring and then rendering audio with our own preamps/processors, performing signal processing, D-to-A conversion, adjust the volume and amplify it. These are the things that we do best and it’s where our efforts are being directed.

All this means that it is not logical for us to develop sources. We’re focusing on developing products like the SSP-800/ CP-800 which are able to extract the full value from the media that people are using today and convert it into the analog signals that amplifiers and speakers need to work. Then we have the new amplifiers that are able to accurately amplify this signal, making the world’s best speakers sound even better.

What are typical bit rates / sample frequencies?

A bit is a one or a zero. 64 bits are therefore a string 64 ones or zeros.

Each bit gives you 6 dB of dynamic range.

CD is typically 16-bit, giving 96 dB of dynamic range. When representing a sine wave this is a total of 65,535 possible values in each sample.

Blu-ray is 24-bit and DVD-A can also have this bit length. This provides 144 dB of dynamic range, which is way beyond the capacity of human hearing and current technology. This represents 16 million possible values in each sample.

32-bit = 192 dB (4.3 billion values per sample)
64-bit = 384 dB (about 10 trillion times more than 32-bit per sample!)

The sample rate is the number of samples recorded or stored or reproduced per second.

Red Book CD is 44.1 kHz, which 44,100 samples recorded/played back per second.

DVD-A is up to 192 kHz in stereo and 96 kHz in 6 channels (HDMI only).

What s the maximum sample rate/ bit length possible via HDMI?

8 channels of 24-bit / 192 kHz audio

What are the maximum sample rates bit lengths possible via Sony Philips Digital Interconnect Format (SPDIF)?

The maximum multichannel audio via SPDIF is 24-bit/48 kHz which is suitable for DVD video.

The maximum 2-channel audio via SPDIF is 24-bit/96 kHz (PCM).

SPDIF is a single-ended version of the AES/EBU professional standard and the SPDIF specification is derived from the AES/EBU specification. AES/EBU is specified up to 96 kHz and the performance at 192 kHz is “to be confirmed”.

What is the effect of changing bit lengths / sample frequency?

We have found that increasing bit depth has more of an impact than sample frequency, and have not generally been able to perceive an improvement in sample rates above 96 kHz. In fact many parts that are capable of the higher sample rates do not perform as well at their highest rates.

What is sample rate conversion?

An analog audio signal is a smooth line. Digitizing that signal involves dividing it into a specific number of samples, each of which is a snapshot of the analog audio signal at a specific instant. This converts the smooth analog audio signal into a staircase, where each sample is a step.

Increasing the sample rate means the staircase is composed of more, smaller steps which would start to look more like the original analog signal.

Sample rate conversion occurs purely in the digital domain and it involves changing the number of samples in a signal by either fabricating new samples to go between the samples it’s given, or reducing the number of samples by calculating sample values that should appear if the same signal were recorded at a lower sample frequency. This is a hugely complex process but if the material is native 192 kHz then it can be down-converted to 96 kHz by taking the average from every two samples. If the material is native 96 kHz it can be upsampled by extrapolating additional samples from data contained in the original signal.

I hear small pops from my speakers when my amplifier powers up or down.

This is not something to worry about.

Our amps are direct drive amplifiers and there is no buffer between their outputs and the speakers. This is a deliberate design decision because we believe that the highest audio quality is delivered when the amp is separated from the speakers by the speaker cable only.

The reason the sound is occurring is due to the power supply capacitors. When the unit is powered up/down, the capacitors begin to charge or discharge and there can be minute differences between the speed of this process from one capacitor to another. This minor difference creates a voltage which results in the sound. There may be differences in how loud a turn-on pop is depending how long the amp has been off. When the amp is off for a long time, the capacitors completely discharge and the voltage is greater.

Technically the voltages are called DC offsets and they are minimal, about a 1,000th of the incoming voltage. This is certainly less than the voltages involved in actually reproducing music, so it’s not a risk to the speakers.

It doesn’t happen on all amplifiers, but amplifiers that do produce this performance are within spec.