There has been quite a bit discussion on upsampling in the context of NOS dacs, especially R2R.
Yep, bizarre isn’t it? They buy a Non-OverSampling DAC and then oversample it.
Often for oversampling dacs the highest sample rate and bitrate that the dac can accept sounds best
Nope, that’s pretty much never the case. In fact technically, the highest sample rates are worse.
Examples of questions that have been on the table several times:
- [1] What bit depth should one choose when using PCM
- [2] Should one go with the lowest bit depth with perfect linearity or is there something to be gained if some non-linearity is tolerated to get higher bit depth
- [3] What sample rate sounds best
- [4] PCM vs DSD and why
1. When choosing bit depth it doesn’t matter. Each bit represents about 6dB of dynamic range (6.02dB to be precise), so 16bit has about 96dB dynamic range. Music recordings typically have less than 50dB dynamic range, only about 30dB dynamic range with the highly compressed recordings, up to about 60dB with some uncompressed orchestral recordings and a very few classical recordings that go to about 70dB dynamic range. Studio/Music microphones only typically have 70dB or less dynamic range and the most dynamic go up to just over 80dB. In all cases, this is a lot (or a massive amount) less than the dynamic range offered by 16bit, even the highly dynamic (60dB dynamic range) orchestral recordings are effectively using just 10bits of the 16bits, the remaining 6bits are just random values (noise). So, what does 20bit or 24bit get you except ever quieter levels of noise? 24bit is useful when recording because it allows a huge amount of headroom but for playback there is literally nothing (other than noise) to be gained.
2. Not quite sure where this one came from. The lowest bit depth has less linearity but once you get beyond a few bits then dither eliminates any non-linearity, hence why it’s a requirement of digital audio.
3. 44.1kHz or 48kHz. If you go above that, then you run the risk of ultrasonic content, content that is inaudible to humans but may cause an audible non-linear response (inter-modulation distortion) downstream in your amp or transducers. This is not an issue as far as oversampling 44.1 or 48kHz on the DAC end is concerned, as an anti-image filter is applied to remove everything above the Nyquist frequency (half the sample rate).
4. Technically PCM, as DSD inherently has a distortion issue due to only having 1bit and therefore cannot be dithered appropriately. However, in practice (with music) this is pretty much never an audible issue, so it doesn’t really matter which you choose. 16/44 PCM uses less bandwidth than DSD but historically, SACD (DSD) versions were sometimes mastered better than the 16/44 version, as it couldn’t be ripped and could only be played in good listening environments. So, there’s no golden rule, it depends on the mastering of a particular track/album.
That's the problem, according to science nothing matters below ~-110db and above ~15khz depending on age, yet people can reliably pass ABX tests between dac filters that only affect 20khz and things like noise floor modulation are very audible despite being below these limits.
That is not “
according to science” and yes, noise floor modulation is very audible below these limits, if in fact they’re not below those limits, for example if you choose a quiet section of music and raise the playback level significantly (as science indicates). At a reasonable/loudish playback level, say 85dBSPL peak with HPs for instance, then -110dB would be -25dBSPL, which cannot even exist as a sound pressure level and therefore obviously has to be inaudible.
I suggest moving this discussion over to the Sound Science forum if the OP or anyone else wants the actual facts, as discussing science isn’t allowed in this or any other forum on Head-Fi except the sound science forum. If the answer you’re looking for is just marketing misinformation or audiophile myths based on marketing misinformation, then it’s fine to just leave it here.
G