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by John Siau April 24, 2010
By Elias Gwinn and John Siau
With the introduction of Benchmark’s HDR-VC™ (High Dynamic Range Volume Control), many audio enthusiasts are taking another look at volume control methods.
‘Dynamic range’ and ‘signal-to-noise ratio’ (SNR) specifications describe the same quality. They both indicate the ratio of the highest possible signal level to the level of the noise floor produced by the component. This noise may be dominated by thermally produced white noise, or it may be dominated by discrete frequencies such as those that that are related to hum and buzz produced by AC power.
SNR measurements are conducted by measuring the noise floor without any signal present. However, this method proved to be inadequate for devices that contained auto-mute technology. These auto-mute circuits would engage when no signal was present, and the ‘noise floor’ measurements would be deceivingly low. Dynamic-range measurements were introduced as an alternative to SNR measurements. Dynamic-range measurements are conducted by measuring the noise floor with a low-level test signal that prevents auto-mute from engaging. For the purpose of this article, ‘dynamic range’ will refer to the noise floor of an audio device relative to the maximum undistorted output level.
A volume control’s effect on dynamic range can be explained via the concept of ‘gain-staging’. Gain-staging is the practice of optimizing head-room and dynamic range by properly aligning the levels of each active gain stage and/or attenuation stage (active and/or passive). A simplified description of proper gain-staging is: coordinating all devices to reach maximum input/output at the same time, operating in the upper regions of those input/output levels, and, when needed, attenuating near the end of the signal chain.
A usable analogy to dynamic range is the height of a room, floor-to-ceiling, where the ‘floor’ represents the noise floor of an electronic device, the ‘ceiling’ represents the highest signal amplitude possible, and the ‘height’ represents the dynamic range. A tall person use more of the available headroom than a short person. An active audio device with analog signal paths will have an intrinsic noise floor due to thermal noise and other electronic factors. The ‘height’ between the ‘floor’ and ‘ceiling’ of a signal stage will remain consistent, but this ‘floor-to-ceiling height’ will only be utilized if the signal path has the proper gain-staging and SNR. This analogy will be used later, when discussing specific volume-control implementations.
All audio devices add errors to a signal. These errors are categorized as ‘noise’ and ‘distortion’. A device’s THD+N measurement quantifies the total errors added by the device.
A THD+N measurement will detect many different phenomena: harmonic distortion, intermodulation distortion, jitter-induced distortion, crosstalk, random white noise, power supply hum, external noise, etc. For a given THD+N ratio, these errors may have very different thresholds of audibility.
A device’s frequency response is a measurement of deviation from linear response. A perfectly linear device would reproduce all frequencies equally. The amount of gain or attenuation would be the same for all frequencies of an input signal in a device with linear frequency response.
In reality, every device has a limited bandwidth. In addition, the accuracy of the frequency response within the pass-band varies among components. An example of an intentional non-linear response is a device with a ‘bass boost’. In that case, the low frequencies will have more gain than other frequencies within the signal.
However, many components have unintentional irregularities in their frequency response. For example, some volume control circuits can change the frequency response as the volume is changed.
‘Inter-channel gain matching’ describes a device’s ability to achieve precise matching of separate channel output levels (i.e. between left and right in a stereo device).
DSP-based volume control of digital audio can be implemented via hardware (DSP in a chip) or software (DAW, media player). It performs multiplications on the digital audio data before the data is delivered to the D/A converter.
In most cases, a DSP-based volume control will limit the dynamic range of the playback system. This is because it doesn’t utilize the entire dynamic range of the D/A converter. Going back to the ‘height of a room’ analogy, the maximum level that a D/A converter can produce is the ‘ceiling’ (or ‘headroom’). The converter reaches the ceiling when a ‘full-scale’ (the highest possible digital amplitude) signal is present at the input.
If the digital signal is attenuated before it reaches the D/A, its peak amplitude will be further removed from full-scale. In other words, the peak output from the D/A converter will be further below the ‘ceiling’, but the ‘floor’ (inherent noise) will remain. In our ‘room analogy’, digital attenuation added a drop-ceiling to our converter, and the new ‘height of the room’ (dynamic range) has been reduced.
Also, when digital attenuation occurs, dither (noise) should be applied to avoid quantization distortion (distortion from quantization errors is beyond the scope of this article, but worth investigating). This dither can lower the SNR of the audio signal because new noise is added to a lowered signal. This is a serious noise contribution because, if the output of the digital volume control is 16 bits, the dither noise is at -96 dBFS (96 dB below full scale). However, if the output is 24 bits, the dither noise is at -144 dBFS. At 144 dB below full scale, this will not add significant noise to the system.
These dynamic-range limitations become less of a problem in D/A converters with significantly high dynamic range, since a playback system will only be as quiet as its noisiest component. In other words, if a system has a power amp with a 100-dB dynamic range, it won’t matter if the dynamic range of the D/A is reduced from 125 dB to 110 dB. The dynamic range of that system will still only be 100 dB. Generally, the higher the dynamic range of the D/A, the more digital attenuation can be applied without affecting the dynamic range of the entire playback system.
In addition to dynamic-range limitations, one should also be concerned about distortion induced by an inferior DSP algorithm. If the designer does not implement proper dithering, severe non-harmonic distortion will occur. Many computer playback systems lack dither. 16-bit systems have noticeable distortion when dither is omitted. 24-bit and 32-bit systems are much more forgiving when dither is omitted.
Other errors in DSP implementation can also result in distortion. However, properly-designed digital volume controls are becoming more common. A well-designed digital volume control will add virtually no distortion to the audio.
Many products use an integrated circuit (IC) to manipulate volume in the analog domain. These analog volume-control IC’s suffer from noise and distortion. These defects are due, in part, to the complications of implementing many precision elements within a single IC package. Specifically, the size, materials, proximity, and voltage limitations of these elements (within the IC) severely limit the overall quality of this solution.
A passive attenuator is simply a resistor network or potentiometer creating a voltage divider in the signal path. The output of a voltage divider is a scaled version of the input signal. A passive attenuator uses only ‘passive’ components, which are components that do not require a power source.
Many people assume that passive attenuators are completely benign. The reality is that passive attenuators can degrade the audio quality. Distortion will be produced when poor-quality resistors are used. Noise will be produced if the impedance of the attenuator is too high. Passive attenuators with high impedance (greater then 500 ohms) are particularly problematic.
It is very difficult to build a high-quality continuously-variable passive attenuator. Potentiometers can contribute distortion, and can cause impedance variations. These variations can cause changes in the frequency response.
In contrast, it is relatively easy to build fixed or stepped attenuators using fixed resistors. High-precision metal-film resistors are readily available and these can be used to build high-quality fixed voltage dividers.
Active gain circuits are simply line-level amplifiers. They work by sending a signal into an active amplifying component (tube/transistor/opamp/etc), whose gain is often adjustable via a potentiometer.
While active gain circuits may suffer from noise and distortion, a properly designed active gain circuit will add very little noise or distortion to the audio. Active gain controls can outperform other controls if the control range is limited to a reasonable range. High performance can be achieved over a +/- 10 dB control range.
Frequency response is constant at all gain settings. Output impedance is very low and is constant at all settings.
The HDR-VC™ system combines an active gain circuit with a low-impedance passive attenuator. The potentiometer in the active gain circuit is motor-driven by the remote-control system. This combination provides low noise and low distortion over a wide range of output levels. Remote volume control can be achieved using DSP or an analog volume control IC, but at the expense of audio performance. Benchmark has chosen to use a motor-driven pot in the DAC1 HDR in order to eliminate the compromises DSP and IC solutions. The other DAC1 models use the same volume control structure, but without the motor drive.
In all DAC1 models, passive attenuators are located after all of the active devices. The SNR at the output of the passive attenuators is identical to the SNR at the input to the passive attenuators. The passive attenuators can be set in 10-dB increments from 0 dB to -30 dB. The passive output attenuators optimize the control range of the active volume control.
The HDR-VC™ system delivers very high performance over a wide range of signal levels while providing remote control.
by Benchmark Media Systems November 20, 2024
Most digital playback devices include digital interpolators. These interpolators increase the sample rate of the incoming audio to improve the performance of the playback system. Interpolators are essential in oversampled sigma-delta D/A converters, and in sample rate converters. In general, interpolators have vastly improved the performance of audio D/A converters by eliminating the need for analog brick wall filters. Nevertheless, digital interpolators have brick wall digital filters that can produce unique distortion signatures when they are overloaded.
An interpolator that performs wonderfully when tested with standard test tones, may overload severely when playing the inter-sample musical peaks that are captured on a typical CD. In our tests, we observed THD+N levels exceeding 10% while interpolator overloads were occurring. The highest levels were produced by devices that included ASRC sample rate converters.
by John Siau April 05, 2024
Audiophiles live in the wild west. $495 will buy an "audiophile fuse" to replace the $1 generic fuse that came in your audio amplifier. $10,000 will buy a set of "audiophile speaker cables" to replace the $20 wires you purchased at the local hardware store. We are told that these $10,000 cables can be improved if we add a set of $300 "cable elevators" to dampen vibrations. You didn't even know that you needed elevators! And let's not forget to budget at least $200 for each of the "isolation platforms" we will need under our electronic components. Furthermore, it seems that any so-called "audiophile power cord" that costs less than $100, does not belong in a high-end system. And, if cost is no object, there are premium versions of each that can be purchased by the most discerning customers. A top-of-the line power cord could run $5000. One magazine claims that "the majority of listeners were able to hear the difference between a $5 power cable and a $5,000 power cord". Can you hear the difference? If not, are you really an audiophile?
by John Siau June 06, 2023
At the 2023 AXPONA show in Chicago, I had the opportunity to see and hear the Hill Plasmatronics tweeter. I also had the great pleasure of meeting Dr. Alan Hill, the physicist who invented this unique device.
The plasma driver has no moving parts and no diaphragm. Sound is emitted directly from the thermal expansion and contraction of an electrically sustained plasma. The plasma is generated within a stream of helium gas. In the demonstration, there was a large helium tank on the floor with a sufficient supply for several hours of listening.
While a tank of helium, tubing, high voltage power supplies, and the smell of smoke may not be appropriate for every living room, this was absolutely the best thing I experienced at the show!
- John Siau