The Chord Hugo DAC is causing quite a stir on the forums & Rob Watts, the designer, attributes it's apparently great sound to the use of 26K of taps in the interpolation filter, implemented in an FPGA. A presentation of the technical details of the DAC are to be found here
One of his interesting slides states:
It has been stated here & elsewhere that the subsample timing accuracy of a 16/44 reconstructed waveform is some tens of picoseconds (or was it hundreds?).
This is for a bandlimited signal & of course band-limiting a signal (at the ADC or before) smears it's timing accuracy. So we have picosecond subsample timing for an a signal that already has had it's timing limited.
But aside from that limitation which we can do nothing about (it's already baked onto the 16/44 file), the above highlighted text is stating that the perfect reconstruction of the waveform at playback requires an infinite number of taps.
Some information on FIR filter design & characteristics are given here particularly from slide 77 on.
Any comments from those more closely involved with this DSP area?
One of his interesting slides states:
- The interpolation filter (an FIR filter that has a line of taps multiplying coefficients to delayed data) recovers the original amplitude and timing information of the recording
- This filter re-creates the missing bits between samples
- If you look at the original Whittaker-Shannon sampling theory, then for a bandwidth limited signal, if you use an infinite tap length FIR filter then the “missing bits” will be perfectly reconstructed
- The FIR filter has a sine(x)/x response – if you use taps that have 16 bit coefficient accuracy, you need about 1,000,000 taps for an 8 times filter!
- Practical filters have limited tap length – a few hundred maximum
- These conventional filters do not properly reconstruct the original timing of transients
It has been stated here & elsewhere that the subsample timing accuracy of a 16/44 reconstructed waveform is some tens of picoseconds (or was it hundreds?).
This is for a bandlimited signal & of course band-limiting a signal (at the ADC or before) smears it's timing accuracy. So we have picosecond subsample timing for an a signal that already has had it's timing limited.
But aside from that limitation which we can do nothing about (it's already baked onto the 16/44 file), the above highlighted text is stating that the perfect reconstruction of the waveform at playback requires an infinite number of taps.
Some information on FIR filter design & characteristics are given here particularly from slide 77 on.
Any comments from those more closely involved with this DSP area?