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A lot of CD players in the early 80s had real flaws that impacted audio quality. It wasn't until the early 90s that digital audio (and CD players) were reliably

A lot of CD players in the early 80s had real flaws that impacted audio quality. It wasn't until the early 90s that digital audio (and CD players) were reliably good. This article describes a lot of the problems with early DACs in the 1980s: https://www.stereophile.com/content/pdm-pwm-delta-sigma-1-bit-dacs https://www.stereophile.com/content/pdm-pwm-delta-sigma-1-bi... > But they didn't go for the Burr-Brown solution of adding an adjustment pin to the converter chip, prescribing the use of an external trim pot, and praying for constant temperature. Yikes! Some early CD players' sound quality was temperature dependent!

mc32 5y ago
Maybe that was so, but this was Neil Young and Steve Jobs introducing uncompressed audio in 2012, not mid '80s, so neither one of them is making sense. As I linked elsewhere this is Monty's response: https://web.archive.org/web/20160304013821/http://people.xiph.org/~xiphmont/demo/neil-young.html https://web.archive.org/web/20160304013821/http://people.xip... And explained via Video: https://www.xiph.org/video/vid2.shtml https://www.xiph.org/video/vid2.shtml
- TedDoesntTalk 5y ago
  All you really need to know is Nyquist's thereom "A bandlimited continuous-time signal can be sampled and perfectly reconstructed from its samples if the waveform is sampled over twice as fast as it's highest frequency component." This is where the magic 44.1 Khz sampling rate comes from. It is twice the limit of human hearing at 22 Khz. Now I don't really understand why 16-bit or 1-bit makes a difference. Maybe someone can explain that to me.
  - cillian64 5y ago
    Nyquist’s theorem only covers sampling (turning a “continuous” signal (which exists at all points in time) into a “discrete” function (which only exists at certain points in time)). Each of the Nyquist samples is a real-valued number which can take an infinite number of values. To get from Nyquist samples to something we can actually represent digitally we need to do a second step: quantisation. This is where we take the real- valued samples and quantise each to one of a fixed set of values. The bit-depth determines the number of values we can quantise to. The higher the bit depth the more values we can quantise to and so the less error introduced by quantisation. You can think of quantisation error as introducing a noise term which gets added onto your signal. Higher bit depth lets you either reduce the amplitude of quantisation error or increase the range of signals you can represent (this is called dynamic range). Once you get to about 16 bits the quantisation noise is inaudible in even the quietest listening environments while at the same time the dynamic range is high enough to cause instant hearing damage. So, in a similar way to Nyquist telling us that more than 44kHz sample rate is pointless, dynamic range tells us that more than 16 bit depth is pointless (for playback. Things get a little more complicated for recording and editing). Edit: as linked by another commenter, Monty from Xiph explaining it better than me: https://web.archive.org/web/20160304013821/http://people.xiph.org/~xiphmont/demo/neil-young.html https://web.archive.org/web/20160304013821/http://people.xip...
    - TedDoesntTalk 5y ago
      Thank you for the excellent explanation. Do you know why CD players went through a phase advertising “1 bit!” after a lengthy time advertising 16-bit? What does or even mean you have a 1-bit depth? (Perhaps marketing)
      - gwbas1c 5y ago
        If you have the time, read through https://www.stereophile.com/content/pdm-pwm-delta-sigma-1-bit-dacs https://www.stereophile.com/content/pdm-pwm-delta-sigma-1-bi... The short answer is that a commodity 1-bit DAC operating highly in excess of 44khz sounds a lot better than the earlier commodity DACs from the 1980s. The above link explains a lot of the problems that those DACs had, like the "zero crossing problem."
    - gwbas1c 5y ago
      16-bit audio's quantization error is audible unless there's good dithering and/or noise shaping. Otherwise, the quantization error becomes audible over ~80db. (Normal speech.) Undithered / unshaped 16-bit audio can sound better than a record or cassette, although some people might prefer the sound of analog hiss to the choppiness of rounding.
  - gwbas1c 5y ago
    Some people can go up to 28khz: https://en.wikipedia.org/wiki/Hearing_range#Humans https://en.wikipedia.org/wiki/Hearing_range#Humans > Under ideal laboratory conditions, humans can hear sound as low as 12 Hz[11] and as high as 28 kHz It's just a fraction of an octave over 22khz, but that probably explains why some people swear that 96khz sources sound better than the CD. Related: When I ripped my DVD-A collection, (I like music in surround,) I downsampled everything to 48khz, 16-bit. Later, if I played the original 96khz version, I did notice a slight crispness to the music. It's anecdotal, and I thought it was placebo effect, until I came across the above link.
  - gwbas1c 5y ago
    > Now I don't really understand why 16-bit or 1-bit makes a difference. Maybe someone can explain that to me. Short answer: It's a means to an end. A 1-bit DAC with a DSP is easier to make than a 16-bit DAC with no DSP. Long answer: If you're using dithering, every time you double the sampling rate, the perceived "bits per sample" for everything else doubles. IE: For sounds below 22khz, a 16-bit, 44khz will sound just as good as 15-bit, 88khz, or 14-bit, 176khz. Add some noise shaping, (basically, tuned hiss,) and the perceived bits per sample also go up. Thus, a 1-bit per sample DAC at a super-high sampling rate, coupled with a well-designed DSP, and an analog low-pass filter, will perfectly preserve all of the information in the original 16-bit, 44khz signal. What the DSP does is move all of the distortion into frequencies well above the cutoff of the analog low-pass filter. That's how a 22khz tone comes out as a sine wave instead of a square wave.
rasz 5y ago
>This article describes a lot of the problems with early DACs in the 1980s: https://www.stereophile.com/content/pdm-pwm-delta-sigm https://www.stereophile.com/content/pdm-pwm-delta-sigm I started reading this and it hurts. I looked author up and he is an Engineer who got EE degree in the fifties and simply doesnt understand digital electronics :( Still thinks signal coming out of Digital player is a staircase, makes comparison to cutting something down to micrometers and how its super difficult while digital chips have been fabbed on smaller processes more than 10 years before this article was even written. Thinks bits in the ADC dont flip all at once etc. Remember, any time you see someone even mention jitter when it comes to CD audio you know they are an audiophool talking nonsense. Blind test studies proved even "golden ears" idiots cant tell the difference up to hundreds of nanoseconds of jitter. For example https://www.jstage.jst.go.jp/article/ast/26/1/26_1_50/_pdf https://www.jstage.jst.go.jp/article/ast/26/1/26_1_50/_pdf Cheapest crystal oscillators have rms jitter measured in picoseconds, and obviously no one ever used RC oscillators in CD players :-) yet you see those morons replace factory crystals with "audio quality" snake oil ones to "fix jitter".

mc32 5y ago

Maybe that was so, but this was Neil Young and Steve Jobs introducing uncompressed audio in 2012, not mid '80s, so neither one of them is making sense. As I linked elsewhere this is Monty's response: https://web.archive.org/web/20160304013821/http://people.xiph.org/~xiphmont/demo/neil-young.html https://web.archive.org/web/20160304013821/http://people.xip... And explained via Video: https://www.xiph.org/video/vid2.shtml https://www.xiph.org/video/vid2.shtml

TedDoesntTalk 5y ago
All you really need to know is Nyquist's thereom "A bandlimited continuous-time signal can be sampled and perfectly reconstructed from its samples if the waveform is sampled over twice as fast as it's highest frequency component." This is where the magic 44.1 Khz sampling rate comes from. It is twice the limit of human hearing at 22 Khz. Now I don't really understand why 16-bit or 1-bit makes a difference. Maybe someone can explain that to me.
- cillian64 5y ago
  Nyquist’s theorem only covers sampling (turning a “continuous” signal (which exists at all points in time) into a “discrete” function (which only exists at certain points in time)). Each of the Nyquist samples is a real-valued number which can take an infinite number of values. To get from Nyquist samples to something we can actually represent digitally we need to do a second step: quantisation. This is where we take the real- valued samples and quantise each to one of a fixed set of values. The bit-depth determines the number of values we can quantise to. The higher the bit depth the more values we can quantise to and so the less error introduced by quantisation. You can think of quantisation error as introducing a noise term which gets added onto your signal. Higher bit depth lets you either reduce the amplitude of quantisation error or increase the range of signals you can represent (this is called dynamic range). Once you get to about 16 bits the quantisation noise is inaudible in even the quietest listening environments while at the same time the dynamic range is high enough to cause instant hearing damage. So, in a similar way to Nyquist telling us that more than 44kHz sample rate is pointless, dynamic range tells us that more than 16 bit depth is pointless (for playback. Things get a little more complicated for recording and editing). Edit: as linked by another commenter, Monty from Xiph explaining it better than me: https://web.archive.org/web/20160304013821/http://people.xiph.org/~xiphmont/demo/neil-young.html https://web.archive.org/web/20160304013821/http://people.xip...
  - TedDoesntTalk 5y ago
    Thank you for the excellent explanation. Do you know why CD players went through a phase advertising “1 bit!” after a lengthy time advertising 16-bit? What does or even mean you have a 1-bit depth? (Perhaps marketing)
    - gwbas1c 5y ago
      If you have the time, read through https://www.stereophile.com/content/pdm-pwm-delta-sigma-1-bit-dacs https://www.stereophile.com/content/pdm-pwm-delta-sigma-1-bi... The short answer is that a commodity 1-bit DAC operating highly in excess of 44khz sounds a lot better than the earlier commodity DACs from the 1980s. The above link explains a lot of the problems that those DACs had, like the "zero crossing problem."
  - gwbas1c 5y ago
    16-bit audio's quantization error is audible unless there's good dithering and/or noise shaping. Otherwise, the quantization error becomes audible over ~80db. (Normal speech.) Undithered / unshaped 16-bit audio can sound better than a record or cassette, although some people might prefer the sound of analog hiss to the choppiness of rounding.
- gwbas1c 5y ago
  Some people can go up to 28khz: https://en.wikipedia.org/wiki/Hearing_range#Humans https://en.wikipedia.org/wiki/Hearing_range#Humans > Under ideal laboratory conditions, humans can hear sound as low as 12 Hz[11] and as high as 28 kHz It's just a fraction of an octave over 22khz, but that probably explains why some people swear that 96khz sources sound better than the CD. Related: When I ripped my DVD-A collection, (I like music in surround,) I downsampled everything to 48khz, 16-bit. Later, if I played the original 96khz version, I did notice a slight crispness to the music. It's anecdotal, and I thought it was placebo effect, until I came across the above link.
- gwbas1c 5y ago
  > Now I don't really understand why 16-bit or 1-bit makes a difference. Maybe someone can explain that to me. Short answer: It's a means to an end. A 1-bit DAC with a DSP is easier to make than a 16-bit DAC with no DSP. Long answer: If you're using dithering, every time you double the sampling rate, the perceived "bits per sample" for everything else doubles. IE: For sounds below 22khz, a 16-bit, 44khz will sound just as good as 15-bit, 88khz, or 14-bit, 176khz. Add some noise shaping, (basically, tuned hiss,) and the perceived bits per sample also go up. Thus, a 1-bit per sample DAC at a super-high sampling rate, coupled with a well-designed DSP, and an analog low-pass filter, will perfectly preserve all of the information in the original 16-bit, 44khz signal. What the DSP does is move all of the distortion into frequencies well above the cutoff of the analog low-pass filter. That's how a 22khz tone comes out as a sine wave instead of a square wave.

rasz 5y ago

>This article describes a lot of the problems with early DACs in the 1980s: https://www.stereophile.com/content/pdm-pwm-delta-sigm https://www.stereophile.com/content/pdm-pwm-delta-sigm I started reading this and it hurts. I looked author up and he is an Engineer who got EE degree in the fifties and simply doesnt understand digital electronics :( Still thinks signal coming out of Digital player is a staircase, makes comparison to cutting something down to micrometers and how its super difficult while digital chips have been fabbed on smaller processes more than 10 years before this article was even written. Thinks bits in the ADC dont flip all at once etc. Remember, any time you see someone even mention jitter when it comes to CD audio you know they are an audiophool talking nonsense. Blind test studies proved even "golden ears" idiots cant tell the difference up to hundreds of nanoseconds of jitter. For example https://www.jstage.jst.go.jp/article/ast/26/1/26_1_50/_pdf https://www.jstage.jst.go.jp/article/ast/26/1/26_1_50/_pdf Cheapest crystal oscillators have rms jitter measured in picoseconds, and obviously no one ever used RC oscillators in CD players :-) yet you see those morons replace factory crystals with "audio quality" snake oil ones to "fix jitter".