Not to prempt Chord, but the major achievement here is proper capture of transients via the Watts Transient aligned filter with the 10,240 taps. Chord builds their own Dac (delta-sigma type Pulse Array) and their own filter whose coefficients they protect from extraction and duplication by pirates.
The Qute's 2 biggest compromises are the cheap but decent SMPS and the mediocre USB implementation. These are causualties of cost cutting, but perhaps the USB can be uncrippled with a firmware update. One could also say that the lack of a Ram buffer is a big compromise, but actual listening says otherwise. It is by no means bad, but just not as curring edge as the rest of the DAC. Any decent afte rmarket Linear PSU will make the Qute really sing. A firmware update to DSD128 would make this cookie fly even faster off the shelves.
The bass is huge and powerful and when tweaked up with LPSU and quality USB/SPDIF converter, the Qute makes RBCD sound like HiRez!
Read about their Dac technology here: http://www.chordelectronics.co.uk/chord-dac-technology.asp
"WTA Filter: It solves the question as to why higher sampling rates sound better. It is well known that 96 kHz (DVD Audio) recordings sound better than 44.1 kHz (CD) recordings. Most people believe that this is due to the presence of ultrasonic information being audible even though the best human hearing is limited to 20kHz. What is not well known is that 768 kHz recordings sound better than 384 kHz and that the sound quality limit for sampling lies in the MHz region.
768 kHz recordings cannot sound better because of information above 200 kHz being important - simply because musical instruments, microphones, amplifiers and loudspeakers do not work at these frequencies nor can we hear them. So if it is not the extra bandwidth that is important, why do higher sampling rates sound better?
The answer is not being able to hear inaudible supersonic information, but the ability to hear the timing of transients more clearly. It has long been known that the human ear and brain can detect differences in the phase of sound between the ears to the order of microseconds This timing difference between the ears is used for localising high frequency sound. Since transients can be detected down to microseconds, the recording system needs to be able to resolve timing of one microsecond. A sampling rate of 1 MHz is needed to achieve this!
However, 44.1 kHz sampling can be capable of accurately resolving transients by the use of digital filtering. Digital filtering can go some way towards improving resolution without the need for higher sampling rates........
All of the above innovations are implemented in Xilinx Virtex series FPGA's. These FPGA's offer 200,000 gates per device, and merely updating the EPROM memory chip can easily change the design, thus future proofing is assured
Edited by wisnon - 7/8/13 at 4:03am