As I’ve stated a couple of times through this series, my reason for writing this stuff was not to prove that high res audio is better or worse than normal res audio (whatever that is…). My reason was to highlight some of the advantages and disadvantages associated with LPCM audio at different bit depths and sampling rates. Just as a bullet-point summary of things-to-remember/consider (with some loose grouping):
- “High resolution audio” could mean
- “more than 16 bits per sample”
- “a sampling rate higher than 44.1 kHz”
- “more than 16 bits per sample”
- These two dimensions of the specifications have different implications on the signal
- Doubling the sampling rate only increases your audio bandwidth by 1 octave.
Yes, it’s twice as much information, but that’s only one octave. If you add an extra octave on top of a piano, you don’t get twice as many notes.
- Just because you have more bits per sample doesn’t mean that you are actually getting more resolution.
There are examples out there where a “24-bit recording” is just a 16-bit recording with 8 zeros stuck on the end.
- Just because you have a higher sampling rate doesn’t mean that you are actually getting a recording that was done at that sampling rate.
There are examples out there where, if you do a spectral analysis of a “high-res” recording, you’ll see the cutoff filter of the original 44.1 kHz recording.
- Just because you have a recording done at a higher sampling rate doesn’t mean that the extra information you get is actually useful.
- There is no such thing as “temporal resolution” or “better timing information” caused by higher sampling rates. It’s not film.
- Staircase drawings of digital audio signals are just there to help you understand the concept – they don’t actually exist in the audio signal.
- If your playback system has sampling rate converters (it probably does), try to make sure that they’re good.
- If they’re bad (which happens often), then it could be that a “high res” signal sounds/performs worse than a “normal res” signal.
- If you are filtering the audio signal at low frequencies, it’s better to have a lower sampling rate.
- If your processing distorts the signal for some reason, it’s better to have a higher sampling rate to keep the aliased distortion artefacts as far away from the audio signal as possible.
- If you are a lazy DSP engineer who thinks that filters give you the expected magnitude response, no matter what the centre frequency, you’d better have a higher sampling rate. (Or you could just stop being lazy and compensate.)
- If you need a lower noise floor for the same audio bandwidth, it’s more efficient to add bits than to increase the sampling rate.
- There are many cases where you want equipment that has higher specifications than your audio signal.
- If you have a volume control after the conversion to analogue, then 93 dB of dynamic range (16 bits, TPDF dithered) might be enough – especially if you listen to music with a limited dynamic range. However, if your volume control is in the digital domain, and you have a speaker that can play loudly, then you’ll probably want more dynamic range, and therefore more bits per sample hitting the DAC.
Like I said, I’m not here to tell you that one thing is better or worse than another thing.
As I said, my intention in writing all of this is to help you to never fall into the trap of assuming that “high resolution audio” is better than “normal resolution audio” in all respects.
More is not necessarily better, sometimes, it’s not even more. Don’t fall victim to misleading advertising.