9 ms·

If you're thinking "The highest rate I need my signal to be able to replicate is X, so I should set my sampling rate to 2X," then you're wrong and this article

by f5ve 3y ago

If you're thinking "The highest rate I need my signal to be able to replicate is X, so I should set my sampling rate to 2X," then you're wrong and this article gives several reasons why. As far as I can tell, though, it doesn't mention what may be the most important reason (especially to the folks here at hackernews): resampling and processing. This is why professional grade audio processing operates at a sample rate many multiples higher than human hearing. It's not because of the quality difference between, say, 192 and 96 kHz, but rather if you're resampling or iterating a process dozens of times at those rates, eventually artifacts will form and make their way into the range of human hearing (20 kHz).

gumby 3y ago
Yeah, a lot of people think “Nyquist” is a synonym for 2 and stop thinking further.
- Espressosaurus 3y ago
  Heh. Then they don't actually understand what it implies. Which makes sense I suppose.
rcxdude 3y ago
Not just eventually: many effects, such as basically any non-linear mapping like a distortion, will create overtones that will immediately alias down if you are not oversampling. You either need to use some DSP tricks or oversample (usually a mix of both) to avoid this happening, which often happens in just one step of an effects chain.
- markkitti 3y ago
  Even the term "oversampling" implies that sampling beyond Nyquist rate is excessive. I think you would agree that one is not being excessive. It is necessary to sample well beyond accepted "Nyquist rate" in order to reconstruct the signal.
  - rcxdude 3y ago
    That is the term, however. You see it in many contexts where a higher sample rate is traded for some other desirable attribute. (For example, it's often desirable for an ADC to sample faster than the higher frequency content you care about in an analog signal, for the reasons detailed in the paper as well as because it can give you a lower noise ADC. delta-sigma converters being an extreme case of this, helped by a seperate trick of noise shaping). It's worth noting it's a tradeoff, even in pure processing: almost all non-linear transfer functions will create an infinite number of overtones, so it's impossible to avoid aliasing completely: you can only reduce them to some threshold which is acceptable to the application.
  - KeplerBoy 3y ago
    I'd phrase it differently. Your signal contains all kinds of frequencies: Those you care about and those you don't want in your recording. You can't just sample at the Nyquist rate of the interesting frequency and expect all the other frequencies to vanish. They will mess with the frequencies you are actually interested in.
- scns 3y ago
  A great explanation by Dan Worrall: https://www.youtube.com/watch?v=-jCwIsT0X8M https://www.youtube.com/watch?v=-jCwIsT0X8M
shampto3 3y ago
You’re right, but I fear this idea has become prevalent in audiophile communities where they only want to listen to files that are 96kHz or higher. In my opinion, having a high sample rate only really matters during the production phase and does not have a noticeable effect on the final form factor. If the producer uses high sample rate during the creation process, I see no reason why the listener would care if the file they’re listening to is higher than even 44.1kHz unless they are planning on using it for their own production.
- deleted 3y ago
  [deleted]
- hunter2_ 3y ago
  The same could be said about bit depth: 24 bits offers far less quantization artifacts than 16 bits, and those artifacts can readily show up during production processes such as dynamic range compression, but they are extremely well hidden by dithering with noise shaping which gets applied during mastering so ultimately listeners are fine either way. However, any type of subsequent processing in the digital domain, even just a volume change by the listener if it's applied digitally in the 16 bit realm (i.e., without first upscaling to 24 bits), completely destroys the benefit of dithering. For that reason, we might say that additional processing isn't confined to the recording studio and can happen at the end user level. I'm unsure whether this same logic applies to sampling frequency, but probably? I guess post-mastering processing of amplitude is far more common than time-based changes, but maybe DJs doing beat matching?
  - hamstergene 3y ago
    I detect some fallacy here. The real benefit is not using 6x network bandwidth, storage, memory, processing power and more battery of the mobile device. That benefit is not going anywhere, no matter what. Post-processing is applied to the signal which is physically impossible to distinguish from the source. It is true that it often needs higher resolution, and DSPs will upsample internally and then back and operate on floats. But to claim without evidence, that post-processing may give human listener back the ability to tell apart whether 192/24 medium was used instead of 48/16, would be to reintroduce the same quality-loss paranoia, just with an extra step. If one couldn't hear the difference before an effect was applied...they won't hear it after. As for DJs, they do use high-res assets when producing mixes. That's still mastering stage, technically.
charcircuit 3y ago
Another factor which I don't see mentioned is that the tech audio signal is not always directly going to your ear. Is it possible for the sound bouncing around the room to break such assumptions?
- throwaway0665 3y ago
  This is all about representing signals inside a computer. Audio played from a speaker (or as it exists in the physical domain) is continuous and your ear doesn't have a sample rate. So there's no concept of a Nyquist limit or aliasing with physical sound.
deleted 3y ago
[deleted]
flyinghamster 3y ago
Also, when the sampling rates get extreme (software-defined radio), it is well worth moving to complex samples. Doing so allows you to use a sampling rate equal to your theoretical maximum bandwidth, instead of 2x. That's not such a big deal at audio bandwidth, but when your Airspy is slinging a 6 MHz chunk of spectrum, it becomes a huge deal.
jmsgwd 3y ago
I think you're mixing up the effects of _sample rate_ and _bit depth_ here! Everything you said about sample rate applies more to bit depth. Higher bit depth (bits per sample) results in a lower noise floor. When audio is digitally processed or resampled, a small amount of noise ("quantization distortion") is added, which accumulates with further processing. This can be mitigated by working at higher bit depths - which is why professional grade audio processing routinely uses 24 bit formats (for storage) and 32-bit or 64-bit floating point internally (for processing), even if the final delivery format is only 16 bit. Sample rate, on the other hand, affects bandwidth. A higher sample rate recording will contain higher frequencies. It doesn't have any direct effect on the noise floor or level of distortion introduced by resampling, as I understand. (It could have an indirect effect - for example, if certain hardware or plugins work better at particular sample rates.) A survey of ~2,000 professional audio engineers done in May 2023 showed that 75% of those working in music use 41.1 kHz or 48 kHz, while 93% of those working in post production use 41.1 kHz or 48 kHz.[1] These are the basic CD-derived and video-derived sample rate standards. From this it's clear that even in professional audio, higher sample rates are a minority pursuit. Furthermore, the differences are extremely subjective. Some audio engineers swear by higher sample rates, while others say it's a waste of time unless you're recording for bats. It's very rare (and practically, quite difficult) to do proper tests to eliminate confirmation bias. [1] https://www.production-expert.com/production-expert-1/sample-rate-survey-the-results https://www.production-expert.com/production-expert-1/sample... EDIT: add link to survey.
rightbyte 3y ago
Higher sampling rate makes it easier to identify non-sound disturbances. Like vibrations or electrical, that can show up in multiple orders of some frequency.

by f5ve 3y ago

gumby 3y ago

Yeah, a lot of people think “Nyquist” is a synonym for 2 and stop thinking further.

Espressosaurus 3y ago
Heh. Then they don't actually understand what it implies. Which makes sense I suppose.

rcxdude 3y ago

Not just eventually: many effects, such as basically any non-linear mapping like a distortion, will create overtones that will immediately alias down if you are not oversampling. You either need to use some DSP tricks or oversample (usually a mix of both) to avoid this happening, which often happens in just one step of an effects chain.

markkitti 3y ago
Even the term "oversampling" implies that sampling beyond Nyquist rate is excessive. I think you would agree that one is not being excessive. It is necessary to sample well beyond accepted "Nyquist rate" in order to reconstruct the signal.
- rcxdude 3y ago
  That is the term, however. You see it in many contexts where a higher sample rate is traded for some other desirable attribute. (For example, it's often desirable for an ADC to sample faster than the higher frequency content you care about in an analog signal, for the reasons detailed in the paper as well as because it can give you a lower noise ADC. delta-sigma converters being an extreme case of this, helped by a seperate trick of noise shaping). It's worth noting it's a tradeoff, even in pure processing: almost all non-linear transfer functions will create an infinite number of overtones, so it's impossible to avoid aliasing completely: you can only reduce them to some threshold which is acceptable to the application.
- KeplerBoy 3y ago
  I'd phrase it differently. Your signal contains all kinds of frequencies: Those you care about and those you don't want in your recording. You can't just sample at the Nyquist rate of the interesting frequency and expect all the other frequencies to vanish. They will mess with the frequencies you are actually interested in.
scns 3y ago
A great explanation by Dan Worrall: https://www.youtube.com/watch?v=-jCwIsT0X8M https://www.youtube.com/watch?v=-jCwIsT0X8M

shampto3 3y ago

You’re right, but I fear this idea has become prevalent in audiophile communities where they only want to listen to files that are 96kHz or higher. In my opinion, having a high sample rate only really matters during the production phase and does not have a noticeable effect on the final form factor. If the producer uses high sample rate during the creation process, I see no reason why the listener would care if the file they’re listening to is higher than even 44.1kHz unless they are planning on using it for their own production.

deleted 3y ago
[deleted]
hunter2_ 3y ago
The same could be said about bit depth: 24 bits offers far less quantization artifacts than 16 bits, and those artifacts can readily show up during production processes such as dynamic range compression, but they are extremely well hidden by dithering with noise shaping which gets applied during mastering so ultimately listeners are fine either way. However, any type of subsequent processing in the digital domain, even just a volume change by the listener if it's applied digitally in the 16 bit realm (i.e., without first upscaling to 24 bits), completely destroys the benefit of dithering. For that reason, we might say that additional processing isn't confined to the recording studio and can happen at the end user level. I'm unsure whether this same logic applies to sampling frequency, but probably? I guess post-mastering processing of amplitude is far more common than time-based changes, but maybe DJs doing beat matching?
- hamstergene 3y ago
  I detect some fallacy here. The real benefit is not using 6x network bandwidth, storage, memory, processing power and more battery of the mobile device. That benefit is not going anywhere, no matter what. Post-processing is applied to the signal which is physically impossible to distinguish from the source. It is true that it often needs higher resolution, and DSPs will upsample internally and then back and operate on floats. But to claim without evidence, that post-processing may give human listener back the ability to tell apart whether 192/24 medium was used instead of 48/16, would be to reintroduce the same quality-loss paranoia, just with an extra step. If one couldn't hear the difference before an effect was applied...they won't hear it after. As for DJs, they do use high-res assets when producing mixes. That's still mastering stage, technically.

charcircuit 3y ago

Another factor which I don't see mentioned is that the tech audio signal is not always directly going to your ear. Is it possible for the sound bouncing around the room to break such assumptions?

throwaway0665 3y ago
This is all about representing signals inside a computer. Audio played from a speaker (or as it exists in the physical domain) is continuous and your ear doesn't have a sample rate. So there's no concept of a Nyquist limit or aliasing with physical sound.

deleted 3y ago

[deleted]

flyinghamster 3y ago

Also, when the sampling rates get extreme (software-defined radio), it is well worth moving to complex samples. Doing so allows you to use a sampling rate equal to your theoretical maximum bandwidth, instead of 2x. That's not such a big deal at audio bandwidth, but when your Airspy is slinging a 6 MHz chunk of spectrum, it becomes a huge deal.

jmsgwd 3y ago

I think you're mixing up the effects of _sample rate_ and _bit depth_ here! Everything you said about sample rate applies more to bit depth. Higher bit depth (bits per sample) results in a lower noise floor. When audio is digitally processed or resampled, a small amount of noise ("quantization distortion") is added, which accumulates with further processing. This can be mitigated by working at higher bit depths - which is why professional grade audio processing routinely uses 24 bit formats (for storage) and 32-bit or 64-bit floating point internally (for processing), even if the final delivery format is only 16 bit. Sample rate, on the other hand, affects bandwidth. A higher sample rate recording will contain higher frequencies. It doesn't have any direct effect on the noise floor or level of distortion introduced by resampling, as I understand. (It could have an indirect effect - for example, if certain hardware or plugins work better at particular sample rates.) A survey of ~2,000 professional audio engineers done in May 2023 showed that 75% of those working in music use 41.1 kHz or 48 kHz, while 93% of those working in post production use 41.1 kHz or 48 kHz.[1] These are the basic CD-derived and video-derived sample rate standards. From this it's clear that even in professional audio, higher sample rates are a minority pursuit. Furthermore, the differences are extremely subjective. Some audio engineers swear by higher sample rates, while others say it's a waste of time unless you're recording for bats. It's very rare (and practically, quite difficult) to do proper tests to eliminate confirmation bias. [1] https://www.production-expert.com/production-expert-1/sample-rate-survey-the-results https://www.production-expert.com/production-expert-1/sample... EDIT: add link to survey.

rightbyte 3y ago

Higher sampling rate makes it easier to identify non-sound disturbances. Like vibrations or electrical, that can show up in multiple orders of some frequency.