3 ms·

What bit-depth, frequency, etc. is considered to meet the highest standards. Years ago I read 24 or 32 bits and ~96 KHz, but maybe those aren't even the specifi

What bit-depth, frequency, etc. is considered to meet the highest standards. Years ago I read 24 or 32 bits and ~96 KHz, but maybe those aren't even the specifications that matter any more.

masklinn 10y ago
> Years ago I read 24 or 32 bits and ~96 KHz These are useful to record and produce, they're of no use whatsoever to listeners[0]. 16/48 is the highest standard. 24b (let alone 32) is completely pointless and just wasteful, and 192kHz can actually be detrimental due to ultrasonics. [0] https://people.xiph.org/~xiphmont/demo/neil-young.html https://people.xiph.org/~xiphmont/demo/neil-young.html
- Wildgoose 10y ago
  In theory you are correct - but not in practise. The "Loudness Wars" means that we don't actually get the full 16 bits, in which case I would argue that anything up to the 24/96 standard may well be an improvement.
  - pimeys 10y ago
    I was thinking the same, until I read this article about different qualities: http://tweakheadz.com/16-bit-vs-24-bit-audio/ http://tweakheadz.com/16-bit-vs-24-bit-audio/ TLDR: 16/44.1 is enough for listening purposes.
  - masklinn 10y ago
    > The "Loudness Wars" means that we don't actually get the full 16 bits, in which case I would argue that anything up to the 24/96 standard may well be an improvement. You'll get the exact same loudness issue at thrice the storage requirements. Fight for better mastering (and ubiquitous requirement of loudness checker or across-the-board audio normalisation as iTunes Radio does), not for worthless and detrimental gimmicks. Incidentally, loudness has been trending down (very very slowly) on lossless sources since its 2005 peak: http://www.tristancollins.me/computing/dynamic-range-analysis/ http://www.tristancollins.me/computing/dynamic-range-analysi...
    - pimeys 10y ago
      As a music collector, the loudness wars is one of the biggest reasons to not use the streaming services. For some old albums, services like Spotify have only the remastered versions, if available. Especially in the early 2000s remasters were very loud in expense of the dynamics. Luckily it's usually quite cheap to find the original CD versions, and the loudness wars database [0] is very helpful when hunting these. Also for some new recordings the vinyl version has much better mastering because of the physical limitations of vinyl. Good examples are Radiohead or Björk, where the vinyl versions sound absolutely amazing compared to the overly compressed digital versions. Of course there are always nice surprised, like the new Kate Bush live album, which is almost exactly 89 dB and has amazing dynamics. [0]: http://dr.loudness-war.info/ http://dr.loudness-war.info/
  - Dylan16807 10y ago
    Loudness wars mean you need fewer bits to store the resulting audio.
- Veratyr 10y ago
  I've always wondered after reading this: "Sampled signals are often depicted as a rough stairstep (red) that seems a poor approximation of the original signal. However, the representation is mathematically exact and the signal recovers the exact smooth shape of the original (blue) when converted back to analog." The image attached to this looks basically like a sine wave and it's clear that if you do some basic interpolation on the sampled signal you'll get something that looks the same as the original. But what happens when the signal you're sampling isn't as simple? What happens when you've got numerous tones overlapping each other out of phase, making the wave uneven and jagged? Will the signal still be accurately reproduced? Can humans distinguish the difference?
  - xorcist 10y ago
    It will, and this is indeed what Nyquist tells us. Think about the engineering challenges involved in producing an exact staircase output! A stepped signal isn't nearly as beautiful. Most DACs today are also of the delta-sigma variety which produce something much more sinister. But that's all fine, as long as we describe the output signal. There can be overtones high up in the ultrasound for example. The output filters are then what produces the nice signal you listen to. Bad output filters can produce artifacts that sometimes hearable, especially in DACs from the 80s, but if you to stick to the suggested implementation of commercially available DACs today then you should be fine. Quantization errors is more problematic since they can appear anywhere in the post production chain. When you're in the digital domain you can get all sorts of aliasing artifacts just as you get for images. I think this is a common source of when people can hear differences between 44 and 96 kHz signals, where they just downsampled with a linear filter and they hear artifacts of this resampling. A little quantization noise will take care of that, and competent software such as sox and audacity will do this for you.
  - dTal 10y ago
    Yes, the original signal will be accurately reproduced, provided it did not contain any frequencies at or above 1/2 the sampling rate (Nyquist). If you're looking for some intuition, the restriction on input frequencies guarantees that the signal doesn't "change too fast" between samples. You don't lose any information simply because you make sure there's nothing in the signal you can't capture. All this assumes infinite measuring precision, which is of course incorrect - in reality, the limited number of discrete levels a signal can take in a digital system introduces noise, or more accurately distortion since the noise is correlated with the signal. We fix this with dither, and it's absolutely essential for a digital audio system - by adding a small amount of random noise, you decorrelate the noise from the signal (you can do even better with "noise shaping"). Here's an illustration from wiki showing how nasty quantization error is [1]. To understand dither, it's interesting to think of the limit case of infinite sampling rate but only two levels, low and high. You could actually still reproduce sound perfectly with this setup, albeit still with a frequency limit. What you do is add "high frequency" (higher than the signal) random noise before you measure "low" or "high". This makes the probability of measuring "low" or "high" depend on the exact signal level, rather than whether it's above or below 50%. With enough samples you get a "pulse density modulated" signal, which lingers on low for low values and vice versa [2]. This is equivalent to a continuous signal summed with some nasty, but high-frequency noise. Filter the high-frequency noise you added back out and voila (your ear will do the filtering in a pinch, which is how you can get analog sound out of digital stuff, like hard drives and GPIO pins). Basically: finite sampling rate, infinite precision - fine, we can squeeze the information into the precision. Infinite sampling rate, finite precision - still fine, we can squeeze the information into the time domain. Finite sampling rate and finite precision, i.e. the real world - there will be some error. [1] https://upload.wikimedia.org/wikipedia/commons/2/22/Quanterr.png https://upload.wikimedia.org/wikipedia/commons/2/22/Quanterr... [2] https://upload.wikimedia.org/wikipedia/commons/e/e7/Pulse_density_modulation.svg https://upload.wikimedia.org/wikipedia/commons/e/e7/Pulse_de...
shmerl 10y ago
Extra bit depth is useful only if you want to remaster the audio (apply whatever effects and etc.). Otherwise 16 is just fine. Lossless source though is needed if you want to encode in any lossy codec.

What bit-depth, frequency, etc. is considered to meet the highest standards. Years ago I read 24 or 32 bits and ~96 KHz, but maybe those aren't even the specifications that matter any more.

masklinn 10y ago

> Years ago I read 24 or 32 bits and ~96 KHz These are useful to record and produce, they're of no use whatsoever to listeners[0]. 16/48 is the highest standard. 24b (let alone 32) is completely pointless and just wasteful, and 192kHz can actually be detrimental due to ultrasonics. [0] https://people.xiph.org/~xiphmont/demo/neil-young.html https://people.xiph.org/~xiphmont/demo/neil-young.html

Wildgoose 10y ago
In theory you are correct - but not in practise. The "Loudness Wars" means that we don't actually get the full 16 bits, in which case I would argue that anything up to the 24/96 standard may well be an improvement.
- pimeys 10y ago
  I was thinking the same, until I read this article about different qualities: http://tweakheadz.com/16-bit-vs-24-bit-audio/ http://tweakheadz.com/16-bit-vs-24-bit-audio/ TLDR: 16/44.1 is enough for listening purposes.
- masklinn 10y ago
  > The "Loudness Wars" means that we don't actually get the full 16 bits, in which case I would argue that anything up to the 24/96 standard may well be an improvement. You'll get the exact same loudness issue at thrice the storage requirements. Fight for better mastering (and ubiquitous requirement of loudness checker or across-the-board audio normalisation as iTunes Radio does), not for worthless and detrimental gimmicks. Incidentally, loudness has been trending down (very very slowly) on lossless sources since its 2005 peak: http://www.tristancollins.me/computing/dynamic-range-analysis/ http://www.tristancollins.me/computing/dynamic-range-analysi...
  - pimeys 10y ago
    As a music collector, the loudness wars is one of the biggest reasons to not use the streaming services. For some old albums, services like Spotify have only the remastered versions, if available. Especially in the early 2000s remasters were very loud in expense of the dynamics. Luckily it's usually quite cheap to find the original CD versions, and the loudness wars database [0] is very helpful when hunting these. Also for some new recordings the vinyl version has much better mastering because of the physical limitations of vinyl. Good examples are Radiohead or Björk, where the vinyl versions sound absolutely amazing compared to the overly compressed digital versions. Of course there are always nice surprised, like the new Kate Bush live album, which is almost exactly 89 dB and has amazing dynamics. [0]: http://dr.loudness-war.info/ http://dr.loudness-war.info/
- Dylan16807 10y ago
  Loudness wars mean you need fewer bits to store the resulting audio.
Veratyr 10y ago
I've always wondered after reading this: "Sampled signals are often depicted as a rough stairstep (red) that seems a poor approximation of the original signal. However, the representation is mathematically exact and the signal recovers the exact smooth shape of the original (blue) when converted back to analog." The image attached to this looks basically like a sine wave and it's clear that if you do some basic interpolation on the sampled signal you'll get something that looks the same as the original. But what happens when the signal you're sampling isn't as simple? What happens when you've got numerous tones overlapping each other out of phase, making the wave uneven and jagged? Will the signal still be accurately reproduced? Can humans distinguish the difference?
- xorcist 10y ago
  It will, and this is indeed what Nyquist tells us. Think about the engineering challenges involved in producing an exact staircase output! A stepped signal isn't nearly as beautiful. Most DACs today are also of the delta-sigma variety which produce something much more sinister. But that's all fine, as long as we describe the output signal. There can be overtones high up in the ultrasound for example. The output filters are then what produces the nice signal you listen to. Bad output filters can produce artifacts that sometimes hearable, especially in DACs from the 80s, but if you to stick to the suggested implementation of commercially available DACs today then you should be fine. Quantization errors is more problematic since they can appear anywhere in the post production chain. When you're in the digital domain you can get all sorts of aliasing artifacts just as you get for images. I think this is a common source of when people can hear differences between 44 and 96 kHz signals, where they just downsampled with a linear filter and they hear artifacts of this resampling. A little quantization noise will take care of that, and competent software such as sox and audacity will do this for you.
- dTal 10y ago
  Yes, the original signal will be accurately reproduced, provided it did not contain any frequencies at or above 1/2 the sampling rate (Nyquist). If you're looking for some intuition, the restriction on input frequencies guarantees that the signal doesn't "change too fast" between samples. You don't lose any information simply because you make sure there's nothing in the signal you can't capture. All this assumes infinite measuring precision, which is of course incorrect - in reality, the limited number of discrete levels a signal can take in a digital system introduces noise, or more accurately distortion since the noise is correlated with the signal. We fix this with dither, and it's absolutely essential for a digital audio system - by adding a small amount of random noise, you decorrelate the noise from the signal (you can do even better with "noise shaping"). Here's an illustration from wiki showing how nasty quantization error is [1]. To understand dither, it's interesting to think of the limit case of infinite sampling rate but only two levels, low and high. You could actually still reproduce sound perfectly with this setup, albeit still with a frequency limit. What you do is add "high frequency" (higher than the signal) random noise before you measure "low" or "high". This makes the probability of measuring "low" or "high" depend on the exact signal level, rather than whether it's above or below 50%. With enough samples you get a "pulse density modulated" signal, which lingers on low for low values and vice versa [2]. This is equivalent to a continuous signal summed with some nasty, but high-frequency noise. Filter the high-frequency noise you added back out and voila (your ear will do the filtering in a pinch, which is how you can get analog sound out of digital stuff, like hard drives and GPIO pins). Basically: finite sampling rate, infinite precision - fine, we can squeeze the information into the precision. Infinite sampling rate, finite precision - still fine, we can squeeze the information into the time domain. Finite sampling rate and finite precision, i.e. the real world - there will be some error. [1] https://upload.wikimedia.org/wikipedia/commons/2/22/Quanterr.png https://upload.wikimedia.org/wikipedia/commons/2/22/Quanterr... [2] https://upload.wikimedia.org/wikipedia/commons/e/e7/Pulse_density_modulation.svg https://upload.wikimedia.org/wikipedia/commons/e/e7/Pulse_de...

shmerl 10y ago

Extra bit depth is useful only if you want to remaster the audio (apply whatever effects and etc.). Otherwise 16 is just fine. Lossless source though is needed if you want to encode in any lossy codec.