Two or more Benchmark DAC1 or DAC2 converters can be used together in phase-coherent multichannel audio systems even though their internal clocks are not synchronized. This seems to defy logic, but an examination of the system details reveals why this is possible.
Our bench tests show that if identical signals are fed to two separate DAC1 boxes, the outputs will be precisely in phase with each other, at sample rates up to about 110 kHz. Below this sample rate, the L to R differential phase response of a single DAC1 is almost identical to the differential phase between the outputs of two separate DAC1 boxes.
Similar tests show that two or more DAC2 converters can be used to create phase-coherent multi channel systems at any sample rate. Please note that the time delay through a DAC1 is not the same as a DAC2. This means that the two generations of Benchmark D/A converters should not be mixed when phase-coherence is required.
Benchmark converters achieve nearly perfect box-to-box phase matching without using a separate clock input. This note discusses some of the technical details that contribute to this box-to-box phase matching.
The DAC1 uses an AD1896 asynchronous sample rate converter (ASRC) to isolate the incoming digital audio clock from a free-running low-jitter conversion clock. Every DAC1 has its own free-running master clock. These free-running clocks do not need to be synchronized in order to have a predictable delay through the system. The DAC1 uses the AD1896 to upconvert the audio to a sample rate of about 111 kHz. When upconverting, the AD1896 has a fixed delay that is a function of the input sample rate only. The phase relationship between the input and outputs clock will not change the delay through the SRC. The SRC filter coefficients are always selected to maintain a constant delay from the average phase of the input clock signal. The averaging function prevents jitter artifacts, and is one of 3 significant advantages provided by an upsampling SRC topology.
In a multi-channel system, the digital signals feeding multiple DAC1 converters are all derived from the same clock, and are in phase. However, the internal conversion clocks (inside the DAC1 boxes) are all running independently and yet phase accuracy is maintained (see the plots in the manual). The phase accuracy between two DAC1 boxes is only a function of the frequency matching of the two independent fixed-frequency conversion oscillators. These are crystal oscillators and are actually much more closely matched than they need to be. The clock frequencies are matched to +/- about 50 PPM (+/- 0.005%), and if we do the math, the worst-case miss-match will produce a phase error of only +/- 0.004 degrees at 20 kHz! The analog circuits (and the measurement equipment) have orders of magnitude more phase error than the SRC & D/A conversion systems in the DAC1!
Multiple DAC1 boxes are phase accurate up to an input frequency of 110 kHz. We do not claim phase accuracy between DAC1 boxes running at 176.4 or 192 kHz.
The delay on the 110 kHz side of the SRC is 1.01 ms +/- 50 ppm. The +/- 50 ppm variation is due to the +/- 50 ppm variation in the oscillator used for the 110 kHz D/A conversion clock.
50 ppm is 0.00005
0.00005*1.01 ms = 50.5 ns
50.5 ns is equivalent to moving the position of a microphone by 0.00005 feet, and is equivalent to the electrical delay through a 25 foot cable.
At 20 kHz, +/- 50.5 ns is:
50.5E-9*20000*360=0.36 degrees (at 20 kHz)
This +/- 0.36 degree variation is insignificant, even when the channels are summed.
The amplitude error after summing is: 20*Log((cos(0.36*2)+cos(0))/2)=-0.00035 dB at 20 kHz
What about comb filtering? The first null in the comb filter will occur no lower than: 1/(2*50.5 ns/2)= 19.8 MHz
The bandwidth of 96 kHz digital audio is limited to 48 kHz, so a null at 19.8 MHz is of no consequence.
The DAC2 system is also asynchronous, but due to improved technology, it maintains box-to-box phase accuracy at all input sample rates.
If you look at the back of any Benchmark product, you will find balanced XLR analog-audio connectors. As a convenience, we also provide unbalanced RCA connectors on many of our products. In all cases, the balanced interfaces will provide better performance.
We build our unbalanced interfaces to the same high standards as our balanced interfaces, but the laws of physics dictate that the balanced interfaces will provide better noise performance.
This application note explains the advantages of balanced interfaces.
Benchmark has introduced a new analog-to-analog volume control circuit that features a 256-step relay-controlled attenuator and a 16-step relay-controlled boost amplifier. The volume control has a +15 dB to -122 dB range in 0.5 dB steps and is a key component in the HPA4 Headphone / Line Amplifier.
Our goal was to produce an analog-to-analog volume control with the highest achievable transparency. We wanted to be able to place this volume control in front of our AHB2 power amplifier or in front of our THX-888 headphone amplifier board without diminishing the performance of either device. Our volume control would need to have lower distortion and lower noise than either of these amplifiers. Given the extraordinary performance of these THX-AAA amplifiers, this would not be an easy task!
This application note discusses the engineering decisions that went into the development of this new analog volume control circuit. The end result is a fully buffered volume control with a signal-to-noise ratio that exceeds 135 dB. THD measures better than the -125 dB (0.00006%) limits of our test equipment.
SEAS, a well-known manufacturer of high-quality loudspeakers, selected the Benchmark AHB2 as a key component for use in testing loudspeakers. They created an innovative test system that measures loudspeaker motor strength and moving mass with higher accuracy than previous methods. This new measurement system was documented in the December 2017 Journal of the Audio Engineering Society.
According to the AES paper, the SEAS team selected the Benchmark AHB2 for the following reasons:
"A Benchmark AHB2 amplifier is used, which has excellent signal-to-noise ratio and bandwidth, low output impedance, and is suitable for laboratory use (with advanced overload protection)."
The AHB2 was designed to outperform all competing power amplifiers in terms of noise and distortion. The result is an amplifier with unrivaled transparency.
Our goal was to create the ultimate amplifier for the enjoyment of music. It is nice to know that the AHB2 is also being used to test new and improved loudspeakers!