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…
Yeah, a lot of people think “Nyquist” is a synonym for 2 and stop thinking further.
Heh. Then they don't actually understand what it implies.
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 f…
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, ev…
Bit depth is only useful at reducing the noise floor, the lower the bit rate the higher the noise.
That’s why producers (mixing many tracks in a session) want to use high bit rate stems, because they are summing the noise from n tracks.
It’s a pointless exercise for DJs or anyone listening to a single source to use a higher bit depth.
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 f…
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, ev…
With music, in particular, if you use any analog sources while recording, the signal will contain so much noise that any dithering signal will be far below the floor and will most likely be completely redundant. I know that people claim to hear a difference, but they also claim to hear a difference between gold and copper contacts.
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 f…
People should prefer 48k over 44.1 but not for fidelity. It would just make the world a better place if 44.1k audio files died out. The reasons it was chosen are invalid today and we're stuck with it, and now every audio stack needs to be able to convert between 44.1/88.2 and 48/96 which is a solved problem, but has a tradeoff between fidelity and performance that makes resampling algorithms a critical design feature…
What _is_ the reason people should prefer 48k over 44.1k though?
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…
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.
You’re talking about profiling a normal running program. If you’re profiling a benchmark that’s always executing the hot path, where exactly the hot path contribution lies becomes less clear. This is because you run into aliasing with the samples being collected at unhelpful points. Maybe if you run long enough you start to see a picture but at 1khz you’re going to have to run for a very long time. The other way alia…
I don't think what you're talking about is aliasing: it's more to do with the statistics of sampling. Though even then I still don't quite get what you mean: if a bounds check is 20% of your runtime then you're going to see it in your samples pretty quickly. If it's a small enough fraction of your runtime that you don't expect to see it in millions of samples, then why is it relevant to your performance? Now, if you'…
If the 20% is a hotspot yes. If the 20% is because it’s been inlined and split across 100 different call sites each contributing 0.2%, I don’t think it’s so easy to spot.
I don't think what you're talking about is aliasing: it's more to do with the statistics of sampling. Though even then I still don't quite get what you mean: if a bounds check is 20% of your runtime then you're going to see it in your samples pretty quickly. If it's a small enough fraction of your runtime that you don't expect to see it in millions of samples, then why is it relevant to your performance? Now, if you'…
If the 20% is a hotspot yes. If the 20% is because it’s been inlined and split across 100 different call sites each contributing 0.2%, I don’t think it’s so easy to spot.
Sure, but for basically the same reason that it's difficult to spot in a trace: you need to classify all those segments of code as the same thing.
People should prefer 48k over 44.1 but not for fidelity. It would just make the world a better place if 44.1k audio files died out. The reasons it was chosen are invalid today and we're stuck with it, and now every audio stack needs to be able to convert between 44.1/88.2 and 48/96 which is a solved problem, but has a tradeoff between fidelity and performance that makes resampling algorithms a critical design feature…
What _is_ the reason people should prefer 48k over 44.1k though?
All TV and computer audio runs at it, raise for TV/Film purposes 48000 is a very nice round number.
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 f…
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, ev…
> even just a volume change by the listener if it's applied digitally in the 16 bit realm ...
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…
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?
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.