Thank you for a very interesting and valuable work!
I have read this and the great articles from benchmarkmedia to better understand the problem with intersample overs.
@John_Siau at your webpage (in Q&A section) you are addressing the problem with intersampling overs by not recommending using a software approach:
Q: Many folks use software (HQPlayer, Roon, etc.) based sample rate and/or format (PCM<->DSD) conversion upstream of the DAC. Do you see any benefit to this, and if so, is there a preferred sample rate and/or format for your DACs?
We do not recommend upsampling fixed-point digital audio as this will clip intersample peaks. We do this upsampling inside of the DAC2 and DAC3, but we do it with adequate headroom so that clipping cannot occur.
If I have understood this correct, does not this test show that interpolating upstream like CamillaDSP does it, is indeed a very good way of doing it instead of the dac? To quote
@mdsimon2:
To me even more reason to use CamillaDSP as you can capture with something like the UR23 and then drop the level by a few dB in CamillaDSP to safely handle intersample overs.
This problem is new to me, so please bear with me if I have misunderstood something.
We use Roon to apply a 3 dB reduction prior to the miniDSP SHD processor whenever we use that processor in our listening rooms. We do not apply this 3 dB reduction when feeding the Benchmark DAC3 directly from Roon. We have done the same 3 dB reduction using JRiver and several other high-end players. We have found that the DSP headroom functions perform as advertised.
DSD does avoid the intersample clipping problem if the DSD is converted natively by the D/A converter. Conversion from PMC to DSD is not the solution to the PCM intersample over problem. This PCM to DSD conversion process may introduce clipping of the intersample peaks if the conversion processor lacks sufficient headroom. The correct solution is to apply a 3 dB reduction before the first DSP process (including the interpolation in an oversampled sigma-delta D/A converter chip). The output of the 3 dB reduction should be 24 bits and not 16 bits.
I would never recommend converting PCM to DSD. This is a lossy process that
always adds noise and distortion to the audio. This loss in quality is easy to predict mathematically and it is easy to measure. But, the reduction in performance is small enough that it may not be noticeable. The performance of DSD64 is very good, but it is not as good as 44.1/24.
DSD is a terrible format for studio production, and it is problematic for the end-user who would like to add a DSP process such as a volume control, a crossover, or a room correction system.
I would recommend converting DSD to PCM when you wish to add a DSP process such as a crossover, EQ, or room correction. The DSD to PCM conversion process is mathematically transparent except for the reduction in bandwidth. However, the reduction in bandwidth may be advantageous because it removes the high-amplitude ultrasonic noise from the DSD source. With DSD64, any musical content above 22 kHz is obscured by the DSD noise shaping, so the DSD bandwidth above 22 kHz is not useful because of the high noise levels. Some power amplifiers and tweeters can fold this ultrasonic noise into the audible band, so conversion to PCM may actually improve the performance at the output of the loudspeakers.
Convert DSD64 to 44.1/24. Convert DSD128 and higher to 88.2/24. Reduce the PCM level by 3 dB, then apply your DSP processing.
we.tl
