4 ms·

> you're being severely shortsighted Do you mean the royal you, or me specifically? I think mastering is far more important. And I agree other audio formats

> you're being severely shortsighted Do you mean the royal you, or me specifically? I think mastering is far more important. And I agree other audio formats aren't all great. But uncompressed CD quality audio is perfectly fine, in terms of resolution, humans can't detect beyond that. Neil's Pono stuff was a waste.... it was MASSIVE datawise for no good reason.

hunter2_ 5y ago
Keep in mind there are 2 axes. In the time domain, you're absolutely right, Nyquist–Shannon proves that we don't benefit from sampling higher than twice the upper bound of human hearing. Rates like 96kHz or 192kHz are useless for quality (although in specific live situations, they have utility in that the latency would be half or quarter that of typical rates, for a given buffer size in samples). In the amplitude domain, CD quality (16 bits/sample) is pretty damn great for typical mastered audio (~96dB undithered, ~120dB dithered) but if the audio is unusually dynamic (i.e., loudness is not maximized) like with a symphony orchestra, you might actually turn up your sound system to the point where quantization noise becomes evident in the quietest parts, at which point you absolutely would benefit from more bits per sample (say, 24, yielding ~144dB of range undithered). So in practice, you're right, CDs are fine; but "humans can't detect beyond that" only truly applies in the frequency domain. We can detect latency (not applicable to playback of recordings) and we can detect quantization noise when cranking up the volume knob.
- robbrown451 5y ago
  Right, but of course on most recordings (including Neil's own), that seems very unlikely to be an issue. More significantly, though, the Pono format took about 30 times the amount of data as lossless CD quality. While you are talking more like 1.5 times. https://www.mouser.com/applications/high-resolution-audio/ https://www.mouser.com/applications/high-resolution-audio/
- javajosh 5y ago
  Meta: I have no idea why someone downvoted you, and gave you an upvote because your explanation is accurate, and unusually well-written, IMHO. Although I'm curious what you mean about "detecting latency". You mean between two sources, for positioning? I'm not sure the relevance of that here.
  - hunter2_ 5y ago
    Thanks. I love the stuff! > You mean between two sources, for positioning? I'm not sure the relevance of that here. It's only really meaningful in scenarios like a recording studio where the performer is listening to themselves live, in headphones, if the signal undergoing A/D and D/A conversion with some processing in between. It could mean 2ms instead of 8ms, which can be the difference between distracting and not. Not relevant here (I admitted "not applicable") but just an example of where some utility does exist, but the utility is definitely not audible frequency response.
- Thorrez 5y ago
  >latency would be half or quarter that of typical rates, for a given buffer size in samples Would reducing the buffer size be another way to solve this?
  - hunter2_ 5y ago
    Yes, but buffer of some particular (possibly noticable) size is a necessary evil to avoid underrun glitches.
    - Thorrez 5y ago
      Is it the the size of the buffer in samples, or in milliseconds that matters?
      - hunter2_ 5y ago
        In my experience, samples, although I don't know exactly why. But it's why you can get lower latency glitch-free audio with higher sample rates.

hunter2_ 5y ago

Keep in mind there are 2 axes. In the time domain, you're absolutely right, Nyquist–Shannon proves that we don't benefit from sampling higher than twice the upper bound of human hearing. Rates like 96kHz or 192kHz are useless for quality (although in specific live situations, they have utility in that the latency would be half or quarter that of typical rates, for a given buffer size in samples). In the amplitude domain, CD quality (16 bits/sample) is pretty damn great for typical mastered audio (~96dB undithered, ~120dB dithered) but if the audio is unusually dynamic (i.e., loudness is not maximized) like with a symphony orchestra, you might actually turn up your sound system to the point where quantization noise becomes evident in the quietest parts, at which point you absolutely would benefit from more bits per sample (say, 24, yielding ~144dB of range undithered). So in practice, you're right, CDs are fine; but "humans can't detect beyond that" only truly applies in the frequency domain. We can detect latency (not applicable to playback of recordings) and we can detect quantization noise when cranking up the volume knob.

robbrown451 5y ago
Right, but of course on most recordings (including Neil's own), that seems very unlikely to be an issue. More significantly, though, the Pono format took about 30 times the amount of data as lossless CD quality. While you are talking more like 1.5 times. https://www.mouser.com/applications/high-resolution-audio/ https://www.mouser.com/applications/high-resolution-audio/
javajosh 5y ago
Meta: I have no idea why someone downvoted you, and gave you an upvote because your explanation is accurate, and unusually well-written, IMHO. Although I'm curious what you mean about "detecting latency". You mean between two sources, for positioning? I'm not sure the relevance of that here.
- hunter2_ 5y ago
  Thanks. I love the stuff! > You mean between two sources, for positioning? I'm not sure the relevance of that here. It's only really meaningful in scenarios like a recording studio where the performer is listening to themselves live, in headphones, if the signal undergoing A/D and D/A conversion with some processing in between. It could mean 2ms instead of 8ms, which can be the difference between distracting and not. Not relevant here (I admitted "not applicable") but just an example of where some utility does exist, but the utility is definitely not audible frequency response.
Thorrez 5y ago
>latency would be half or quarter that of typical rates, for a given buffer size in samples Would reducing the buffer size be another way to solve this?
- hunter2_ 5y ago
  Yes, but buffer of some particular (possibly noticable) size is a necessary evil to avoid underrun glitches.
  - Thorrez 5y ago
    Is it the the size of the buffer in samples, or in milliseconds that matters?
    - hunter2_ 5y ago
      In my experience, samples, although I don't know exactly why. But it's why you can get lower latency glitch-free audio with higher sample rates.