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In Reply to: Perhaps you might read the linked article posted by E-Stat on January 26, 2015 at 19:40:43:
Thanks. This answered my questions. I note that they used live acoustic music with decent venues, so it is not surprising that they got positive results, even though listeners had not been specifically trained to discriminate sample rates. However, they did not fully comply with the ITU recommendation for blind tests. [ITU-R BS.1116-1, ITU-R BS.1534-1, ITU-R BS.775-2] Had they used specifically trained listeners or used a preliminary stage to select the best scoring listeners they would likely have gotten much greater statistical significance.
The paper didn't give a specific model number for the RME ADC that was used, so I could not investigate further as to the specific ADC chips used. In all likelihood, they used a multi-bit sigma delta modulator that sampled at around 5 Mhz. The converters therefore downsampled to whatever PCM formats were chosen. As far as I know, all modern audio ADCs that output PCM downsample internally. The differences between downsampling in hardware and software will be small and will depend on details of the DSP involved. There is nothing lost in principle in following a two step process downsampling over a one step process except loss of 1 bit of resolution from the original output format. Given that the RME outputs 24 bits, this would mean restricting the precision to 23 bits, which if reduced to 44/16 would be irrelevant. In the case of reducing to 44/24 it might possibly be relevant, however the 88/24 format already has more resolution in the range 0-20 kHz because of the higher sampling rate.
One might have made the 44 kHz playbacks sound better by using different filtering in playback. Apparently, the experimenters just used the filters provided by the RME Fireface DAC.
IMO this was a good paper, great in comparison to horrible examples such as Meyer - Moran. It could have been better. One thing that I did not like is the use of second tier converters. Better results could have been achieved by taking the converters out of the picture and using file level conversions. This would allow extensive experimentation on how the different possible filter designs affect sound. It would also ensure that all of the analog equipment operated identically in all cases and the only differences being the DSP processing. Furthermore, when doing DSP processing file to file it is possible to capture that impulse response of the filters very precisely, providing much better documentation of the experimental procedure.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
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Follow Ups
- RE: Perhaps you might read the linked article - Tony Lauck 01/27/1509:08:03 01/27/15 (1)
- Thanks for the eval - E-Stat 09:28:05 01/27/15 (0)
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