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PCM, DSD - Trying to get my head round some basics

I've recently revived an interest in music/equipment, but am only slowly putting the pieces together regarding the PCM/DSD discussions.

My main query is regarding the value of a sampled bit. There are plenty of references that state things like '24 bits has a higher dynamic range than 16 bits'. Why does a bit sample always have to have the same amplitude value? Why can't you expand (say) 90dB into 24 bits instead of 16 bits, and get higher resolution?

With DSD, is a 0/1 bit switch the same amplitude as one of the 65535 16 bit bits?
DSD is quoted as having a better impulse response than PCM. Surely though (at least in theory) PCM can go from 0-65535 in one sample, whereas DSD would take 65535 samples to do the same? So DSD would have to be at 65535 times the sample rate to get the same response?
Obviously in practice you wouldn't get these extreme values, but what is the reality of the speed of a transient? DSD would only be good with transients of low amplitude.

Or have I got everything completely bass-ackwards?

Re: PCM, DSD - Trying to get my head round some basics

Reply #1
I'm far from one of the experts here, but as I understand bit depth and dB:

I think you are confusing dB and dB SPL. The former is a relative relationship between two values, the latter indicates absolute sound pressure level, as measured in dB relative to the lowest threshold of hearing. So regarding your question, it's just a matter of mathematics. the 144dB theoretical number for 24-bit is just 2 different ways of saying the same thing. You double the dynamic range with each bit, and you double the dynamic range every roughly 6 dB. Hence, 24-bit translates to 144 dB. I might be using dynamic range instead of signal-to-noise ratio interchangeably, and it might be wrong (I can never seem to grasp the subtle difference), but you get the idea.

Re: PCM, DSD - Trying to get my head round some basics

Reply #2
[...] You double the dynamic range with each bit [...]

But why? Isn't it just some arbitrary scale?
Why can't it be 1.6x (or whatever) per bit?

Re: PCM, DSD - Trying to get my head round some basics

Reply #3
You misunderstand DSD.  DSD, like PCM can go from one rail to the other as fast as the reconstruction filter allows.

In (16 bit) PCM you'd go from -32768 to 32767 in one sample and there'd be a reconstruction filter at 1/2 the sample rate after that: e.g. for 44.1k there'd be a 22.05k filter that would limit the speed that that step could happen.

In DSD you'd go from a series of 0's to a series of 1's in one sample and then thru the low pass filter, which for DSD is usually a shallower filter than PCM but starting at, say, 50k or 80k instead of 22.05k for 16/44.1k PCM.  So the transition is faster in DSD.  DSD can transition faster than 24/96 but about the same as 24/192k.  (a maximum value in DSD isn't really all ones, but the slightly more complicated complete picture doesn't affect the above explanation.)

You could invent many different encodings of the bits, PCM and DSD are but two.  They represent the extremes - in PCM each bit of a sample represents twice the value of the previous bit, the maximum value for just two symbols.  In DSD all bits have exactly the same value.

Re: PCM, DSD - Trying to get my head round some basics

Reply #4
It's basic counting in a base-2 system.
1-bit -> 2 possible values: 0, 1
2-bit -> 4 possible values: 00, 01, 10, 11
3-bit -> 8 possible values: 000, 001, 010 etc etc
4-bit -> 16 possible values
etcetera

If you use more bits for the signal, you have more resolution to express smaller, quieter signals. A potentially larger difference between the loudest and quietest signal means a larger dynamic range. But increasing the resolution of an existing signal obviously does not magically increase the resolution of its contents. It's just being stretched across the new space.

Decibel is a relative scale that doubles values every time you add 6. That number comes from the logarithmic math, so it's not a random number someone chose. So if you have an ampitude of x, then adding 6 dB means your amplitude is now 2x. Conversely, increasing the bit-depth by 1 means you make room for small signals that have half the amplitude of the previous smallest signal, and lo and behold, half the amplitude is an extra 6dB range.

My knowledge of the actual mathematics of dB is shallow at best, so I can't go much deeper than that.

> Why does a bit sample always have to have the same amplitude value?
It doesn't. It's all relative to 1, also called fullscale, also called 100% volume. The only absolute sense of amplitude is in the real world with pressure levels in air and voltages in the wire. When we say 16 bits has X dB dynamic range, it means the difference between the largest and smallest storeable signal is X dB. If you want to put that in an amplifier and blast it or play it quietly, sure, that doesn't change the relative amplitudes of what's coming out of your speakers.

Re: PCM, DSD - Trying to get my head round some basics

Reply #5
You misunderstand DSD.  DSD, like PCM can go from one rail to the other as fast as the reconstruction filter allows.
[...]
In DSD you'd go from a series of 0's to a series of 1's in one sample [...]

So one sample can contain many 0's or many 1's? Now my head really hurts.
I thought it was 1 bit x sample rate?

Re: PCM, DSD - Trying to get my head round some basics

Reply #6
No, DSD is really simple: a stream of ones and zeros.  A one means to head for the positive voltage rail as fast as possible and a zero means head for the negative rail as fast as possible.  The speed of movement is restricted by a low pass filter.  - That's it.  All you need to play DSD is a low pass filter on the stream of bits.

For example 0101010101 for ever represents a 0 output (as does 1100110011001100...)  The stream 0001000100010001... represents a value 1/2 of the way from a zero output to the maximum negative output...  Tho these patterns flop around, by the time they go thru the low pass filter that noise is filtered out.

Calculating those bits is very weird, but one method that's very inefficient but really does work is to search thru all streams of bits looking for the one that (after being low pass filtered) gets you closest to the signal you want.  Usually a sigma delta modulator is used: one way of thinking of a SDM is to compare the low pass filtered stream of bits generated so far to the input signal and produce a one if the stream is too low and a zero if the stream is too high - then repeat forever.

Re: PCM, DSD - Trying to get my head round some basics

Reply #7
I've recently revived an interest in music/equipment, but am only slowly putting the pieces together regarding the PCM/DSD discussions.

There is really very little to discuss. It is basically a PCM world, and DSD is a little placebophile side show that you can safely ignore.

If properly implemented, PCM  and DSD sound exactly alike

For about the last 20 years just about all of our DACs have been converting PCM to DSD under the covers prior to turning the data into analog, and nobody but a few technical specialists knew or cared.  The code name for this process is Delta-Signal or Sigma-Delta.

DSD was taken out from under the covers in order to provide IP protection. Some marketing genius decided to mine the same old reliable but fraudulent vein of gold that the placebophiles have been mining for decades, and claim that DSD "Sounded Better".  It doesn't and if you take what I said about almost all of our DACs turning PCM into DSD as a stardard part of our modern way of ding business, to heart, you can see why.

Quote
My main query is regarding the value of a sampled bit. There are plenty of references that state things like '24 bits has a higher dynamic range than 16 bits'.

That is because it is true. But it does not follow that any music sampled with 24 bits has higher resolution than music sampled with 16 bits.  Both processes depend on the same analog data, and the same acoustical event. In general live music has less than 16 bits of resolution.

Quote
Why does a bit sample always have to have the same amplitude value?


It doesn't.  For example the most signfiicant bit of any PCM word has twice the value of the next one.

Quote
Why can't you expand (say) 90dB into 24 bits instead of 16 bits, and get higher resolution?

Because PCM words represent data, they don't define it. On a good day, they don't limit it, either.  

Data is what it is. You can reduce its resolution or preserve it, but you can't increase it without going back to the original physical process that it represents, and changing it.  That's why many of us have a hearty laugh about all those DSD and 24 bit PCM files that are based on analog tapes and legacy PCM files.  Interestingly enough, about half the SACD and DVD-A recordings released through about 2006 were this way, and no golden ear ever noticed it. Many of those recordings are still being sold, and they have been joined by a host of newer recordings that had the same limitations simply because of the usual forces of nature.

Virtually every so-called "Hi Rez" recording has the bits but not the data to be true to its name.

Quote
With DSD, is a 0/1 bit switch the same amplitude as one of the 65535 16 bit bits?

No.

In DSD there is only one bit and it is switched very rapidly on and off to provide the same kind of handling of the various signal levels as we have with 16 or 24 or whatever bits in PCM.

Quote
DSD is quoted as having a better impulse response than PCM.

...As if impulse response was that relevant to sound quality, which in general it is not...

If you had exactly comparable DSD and PCM systems, the impulse response would be the same. But, as implemented, the systems are not the same. So, they have different impulse responses.

Quote
Surely though (at least in theory) PCM can go from 0-65535 in one sample, whereas DSD would take 65535 samples to do the same?

Well, a sample in DSD is not just one bit.  The parity you suggest is real, so a DSD data stream has to have a whole lot more bits in it to do the identical same thing as a comparable PCM data stream.  The PCM data stream weights the bits with binary values, and the DSD stream does not.

Quote
So DSD would have to be at 65535 times the sample rate to get the same response?

True if all things were equal but they are not all equal in the commercial implementations.

Quote
Obviously in practice you wouldn't get these extreme values, but what is the reality of the speed of a transient? DSD would only be good with transients of low amplitude.

Like I said, if all things were equal, but...

Quote
Or have I got everything completely bass-ackwards?

More like comparing apples with oranges than bass ackwards.  If you really are a newbie, you would do well to concentrate on other issues, such as the ones that actually matter for sound quality. Things like acoustics...


Re: PCM, DSD - Trying to get my head round some basics

Reply #8
Ouch, my entire post was somehow lost. I don't feel like typing this all over again... :-(

Re: PCM, DSD - Trying to get my head round some basics

Reply #9
Ouch, my entire post was somehow lost. I don't feel like typing this all over again... :-(

Try My Account -> Account Settings -> Profile Info -> Show Drafts

Re: PCM, DSD - Trying to get my head round some basics

Reply #10
If you use more bits for the signal, you have more resolution to express smaller, quieter signals. A potentially larger difference between the loudest and quietest signal means a larger dynamic range. But increasing the resolution of an existing signal obviously does not magically increase the resolution of its contents. It's just being stretched across the new space.

Decibel is a relative scale that doubles values every time you add 6. That number comes from the logarithmic math, so it's not a random number someone chose. So if you have an ampitude of x, then adding 6 dB means your amplitude is now 2x. Conversely, increasing the bit-depth by 1 means you make room for small signals that have half the amplitude of the previous smallest signal, and lo and behold, half the amplitude is an extra 6dB range.

My knowledge of the actual mathematics of dB is shallow at best, so I can't go much deeper than that.

> Why does a bit sample always have to have the same amplitude value?
It doesn't. It's all relative to 1, also called fullscale, also called 100% volume. The only absolute sense of amplitude is in the real world with pressure levels in air and voltages in the wire. When we say 16 bits has X dB dynamic range, it means the difference between the largest and smallest storeable signal is X dB. If you want to put that in an amplifier and blast it or play it quietly, sure, that doesn't change the relative amplitudes of what's coming out of your speakers.

I'm afraid I'm having problems interpreting this post.  :(

Surely, either the scale is logarithmic (and double sound pressure is double on the scale), and therefore the folks saying that "anything over 20-bits is a waste of time as it is beyond human endurance" are correct, OR we are capturing an amplitude of -1 to 1 in n resolution, regardless of the amplitude max sound level, then we get a higher resolution to "to express smaller, quieter signals".

Are these two things not contradictory?

Re: PCM, DSD - Trying to get my head round some basics

Reply #11
No, DSD is really simple: a stream of ones and zeros.  A one means to head for the positive voltage rail as fast as possible and a zero means head for the negative rail as fast as possible.  The speed of movement is restricted by a low pass filter.  - That's it.  All you need to play DSD is a low pass filter on the stream of bits.

For example 0101010101 for ever represents a 0 output (as does 1100110011001100...)  The stream 0001000100010001... represents a value 1/2 of the way from a zero output to the maximum negative output...  Tho these patterns flop around, by the time they go thru the low pass filter that noise is filtered out.

Calculating those bits is very weird, but one method that's very inefficient but really does work is to search thru all streams of bits looking for the one that (after being low pass filtered) gets you closest to the signal you want.  Usually a sigma delta modulator is used: one way of thinking of a SDM is to compare the low pass filtered stream of bits generated so far to the input signal and produce a one if the stream is too low and a zero if the stream is too high - then repeat forever.

Ah, I see now. That's not how I thought it worked.
Thanks.

Re: PCM, DSD - Trying to get my head round some basics

Reply #12
I've recently revived an interest in music/equipment, but am only slowly putting the pieces together regarding the PCM/DSD discussions.

There is really very little to discuss. It is basically a PCM world, and DSD is a little placebophile side show that you can safely ignore.

If properly implemented, PCM  and DSD sound exactly alike

For about the last 20 years just about all of our DACs have been converting PCM to DSD under the covers prior to turning the data into analog, and nobody but a few technical specialists knew or cared.  The code name for this process is Delta-Signal or Sigma-Delta.

DSD was taken out from under the covers in order to provide IP protection. Some marketing genius decided to mine the same old reliable but fraudulent vein of gold that the placebophiles have been mining for decades, and claim that DSD "Sounded Better".  It doesn't and if you take what I said about almost all of our DACs turning PCM into DSD as a stardard part of our modern way of ding business, to heart, you can see why.

Quote
My main query is regarding the value of a sampled bit. There are plenty of references that state things like '24 bits has a higher dynamic range than 16 bits'.

That is because it is true. But it does not follow that any music sampled with 24 bits has higher resolution than music sampled with 16 bits.  Both processes depend on the same analog data, and the same acoustical event. In general live music has less than 16 bits of resolution.

Quote
Why does a bit sample always have to have the same amplitude value?


It doesn't.  For example the most signfiicant bit of any PCM word has twice the value of the next one.

Quote
Why can't you expand (say) 90dB into 24 bits instead of 16 bits, and get higher resolution?

Because PCM words represent data, they don't define it. On a good day, they don't limit it, either.  

Data is what it is. You can reduce its resolution or preserve it, but you can't increase it without going back to the original physical process that it represents, and changing it.  That's why many of us have a hearty laugh about all those DSD and 24 bit PCM files that are based on analog tapes and legacy PCM files.  Interestingly enough, about half the SACD and DVD-A recordings released through about 2006 were this way, and no golden ear ever noticed it. Many of those recordings are still being sold, and they have been joined by a host of newer recordings that had the same limitations simply because of the usual forces of nature.

Virtually every so-called "Hi Rez" recording has the bits but not the data to be true to its name.

Quote
With DSD, is a 0/1 bit switch the same amplitude as one of the 65535 16 bit bits?

No.

In DSD there is only one bit and it is switched very rapidly on and off to provide the same kind of handling of the various signal levels as we have with 16 or 24 or whatever bits in PCM.

Quote
DSD is quoted as having a better impulse response than PCM.

...As if impulse response was that relevant to sound quality, which in general it is not...

If you had exactly comparable DSD and PCM systems, the impulse response would be the same. But, as implemented, the systems are not the same. So, they have different impulse responses.

Quote
Surely though (at least in theory) PCM can go from 0-65535 in one sample, whereas DSD would take 65535 samples to do the same?

Well, a sample in DSD is not just one bit.  The parity you suggest is real, so a DSD data stream has to have a whole lot more bits in it to do the identical same thing as a comparable PCM data stream.  The PCM data stream weights the bits with binary values, and the DSD stream does not.

Quote
So DSD would have to be at 65535 times the sample rate to get the same response?

True if all things were equal but they are not all equal in the commercial implementations.

Quote
Obviously in practice you wouldn't get these extreme values, but what is the reality of the speed of a transient? DSD would only be good with transients of low amplitude.

Like I said, if all things were equal, but...

Quote
Or have I got everything completely bass-ackwards?

More like comparing apples with oranges than bass ackwards.  If you really are a newbie, you would do well to concentrate on other issues, such as the ones that actually matter for sound quality. Things like acoustics...

There are proponents of both sides.
I like to try to gather information so I can make an informed decision, instead of blindly following one herd or the other - and understanding how these things work hopefully goes a long way to cut through the misinformation.

From the few samples I've downloaded so far, I can understand where the two camps come from, but I'm not sure the difference is worth fighting over.

Re: PCM, DSD - Trying to get my head round some basics

Reply #13
There are proponents of both sides.

Right, but this isn't politics, it is science.  This isn't new controversial science, it is established science. And the confirming or denying experiments aren't rocket science or take a million dollars worth of equipment to do. The science for testing these things is well known, commonly practiced, and can be done for almost no money if you are interested.

Quote
I like to try to gather information so I can make an informed decision, instead of blindly following one herd or the other - and understanding how these things work hopefully goes a long way to cut through the misinformation.

The science says, PCM and DSD is compared on an apples-to-apples basis must sound the same. You'll probably never find a DSD proponent admitting that in public, but you will find a lot of people are hip to audio and are well-informed about science saying so. Go figure.

Quote
From the few samples I've downloaded so far, I can understand where the two camps come from, but I'm not sure the difference is worth fighting over.

Oh, you'll hear or at least you'll think you hear a lot of differences. The DSD proponents have a number of ways to trick you into doing bad listening experiments where there are hidden and not-so-hidden audible variables like differences in mastering.  There is big money on the table or at least some people think so.

Fact is just about every high-rez initiative fails to deliver on its perceived promise. Notice that SACD and DVD-A aren't in what's left of local record stores like Best Buy. Note that Pono's music site went bankrupt and only used an NOS players are all that seems to be available.

Note that there are about 400 different DAC on the market, many featuring DSD or other hi-rez support but they  are languishing in the marketplace and none of them really went mainstream. Hint: The Emperor has no clothes and many people quietly see it and just keep their money where it was or spend it on something else.

All by yourself, as relatively easy as it really is, most people resist doing proper listening tests. I just got called a bunch of dirty names for politely pointing that out on another thread. But know it or not, you signed up to agree to that when you registered here. Please see TOS 8.

Re: PCM, DSD - Trying to get my head round some basics

Reply #14
I'm afraid I'm having problems interpreting this post.  :(

Surely, either the scale is logarithmic (and double sound pressure is double on the scale), and therefore the folks saying that "anything over 20-bits is a waste of time as it is beyond human endurance" are correct, OR we are capturing an amplitude of -1 to 1 in n resolution, regardless of the amplitude max sound level, then we get a higher resolution to "to express smaller, quieter signals".

Are these two things not contradictory?

It relates to the operating voltage range of the recording and playback device, it is on the analog side.

Let's ignore the digital part, now I have an analog tape recorder, mixer or anything which are purely analog. If you look at the manual and find the specs, it will say something like max level: +20dBu, 8V RMS and so on. If you feed a +30dBu, or a 10V signal to these devices, will result in distortion. Anything, unless digitally synthesized and will never be played on speakers or headphones, are limited by the analog part.

Now the bit depth part is simple, it simply refer to how this +20dBu or 8V signal can be divided into 65536 steps, for example, when recorded in 16-bit. Now if you play the recorded file from a consumer device, like a mobile phone, which is less powerful, for example, only 0.5V, and the phone has a 16-bit DAC, then this 65536 steps are just remapped to a less powerful state.

It is just like a 1080p mobile phone vs a 1080p TV.

 

Re: PCM, DSD - Trying to get my head round some basics

Reply #15
All by yourself, as relatively easy as it really is, most people resist doing proper listening tests. I just got called a bunch of dirty names for politely pointing that out on another thread. But know it or not, you signed up to agree to that when you registered here. Please see TOS 8.

I didn't realise that I'd taken sides. I thought I'd deliberately been libertarian about the whole thing; not offending anyone or poking any cages by sitting on the fence. Now I'm being chastised for it?

I once (years ago) brought up the subject of CD burning, and that in an experiment I had done the newly burned CD sounded better than the stamped original. I didn't know why, it just did as far as I was concerned. I was pounded for my efforts, as "science says it isn't so", quoted Reed-Solomon etc, etc.
Recently I found out that Reed-Solomon is only effective if the disc is tracking properly, and mistracking could easily have been the reason for the difference in sound. I have seen many discs where the punched centre hole is completely misaligned with the data layer on the disc.

Because of my own experiences, I don't try to tell anyone which side of the fence to sit on, that's up to them. I am happy for anyone to believe what they like until science changes its mind and proves otherwise.

That's why I'm asking for help in understanding the processes.

Re: PCM, DSD - Trying to get my head round some basics

Reply #16
I'm afraid I'm having problems interpreting this post.  :(

Surely, either the scale is logarithmic (and double sound pressure is double on the scale), and therefore the folks saying that "anything over 20-bits is a waste of time as it is beyond human endurance" are correct, OR we are capturing an amplitude of -1 to 1 in n resolution, regardless of the amplitude max sound level, then we get a higher resolution to "to express smaller, quieter signals".

Are these two things not contradictory?

It relates to the operating voltage range of the recording and playback device, it is on the analog side.

Let's ignore the digital part, now I have an analog tape recorder, mixer or anything which are purely analog. If you look at the manual and find the specs, it will say something like max level: +20dBu, 8V RMS and so on. If you feed a +30dBu, or a 10V signal to these devices, will result in distortion. Anything, unless digitally synthesized and will never be played on speakers or headphones, are limited by the analog part.

Now the bit depth part is simple, it simply refer to how this +20dBu or 8V signal can be divided into 65536 steps, for example, when recorded in 16-bit. Now if you play the recorded file from a consumer device, like a mobile phone, which is less powerful, for example, only 0.5V, and the phone has a 16-bit DAC, then this 65536 steps are just remapped to a less powerful state.

It is just like a 1080p mobile phone vs a 1080p TV.


That's how I thought it should work. Any input is split into whatever resolution you want (16/24/32).
The folks propagating "extra bits = greater dynamic range" as the ONLY reason for greater bit depth are wrong.

Re: PCM, DSD - Trying to get my head round some basics

Reply #17
I once (years ago) brought up the subject of CD burning, and that in an experiment I had done the newly burned CD sounded better than the stamped original. I didn't know why, it just did as far as I was concerned. I was pounded for my efforts, as "science says it isn't so", quoted Reed-Solomon etc, etc.
Recently I found out that Reed-Solomon is only effective if the disc is tracking properly, and mistracking could easily have been the reason for the difference in sound. I have seen many discs where the punched centre hole is completely misaligned with the data layer on the disc.
I fear you imagine things like you will imagine other things relating to the sound of DSD if you only find a written argument you feel that convinces you. Waste of time.
Is troll-adiposity coming from feederism?
With 24bit music you can listen to silence much louder!

Re: PCM, DSD - Trying to get my head round some basics

Reply #18
I once (years ago) brought up the subject of CD burning, and that in an experiment I had done the newly burned CD sounded better than the stamped original. I didn't know why, it just did as far as I was concerned. I was pounded for my efforts, as "science says it isn't so", quoted Reed-Solomon etc, etc.
Recently I found out that Reed-Solomon is only effective if the disc is tracking properly, and mistracking could easily have been the reason for the difference in sound. I have seen many discs where the punched centre hole is completely misaligned with the data layer on the disc.
I fear you imagine things like you will imagine other things relating to the sound of DSD if you only find a written argument you feel that convinces you. Waste of time.

My point exactly.

Re: PCM, DSD - Trying to get my head round some basics

Reply #19
Recently I found out that Reed-Solomon is only effective if the disc is tracking properly, and mistracking could easily have been the reason for the difference in sound.
Hogwash.
Is 24-bit/192kHz good enough for your lo-fi vinyl, or do you need 32/384?

Re: PCM, DSD - Trying to get my head round some basics

Reply #20
All by yourself, as relatively easy as it really is, most people resist doing proper listening tests.

This is not necessarily as easy as you make out. My first post on HAudio was in the ABX section for Foobar, which was making comparisons impossible as ReplayGain wouldn't set the levels between PCM and DSD correctly (possibly due to the noise, but I won't speculate as I get told off for doing that).

So I HAVE tried a proper listening test, and the technology failed me.


Re: PCM, DSD - Trying to get my head round some basics

Reply #22
All by yourself, as relatively easy as it really is, most people resist doing proper listening tests. I just got called a bunch of dirty names for politely pointing that out on another thread. But know it or not, you signed up to agree to that when you registered here. Please see TOS 8.

I didn't realise that I'd taken sides. I thought I'd deliberately been libertarian about the whole thing; not offending anyone or poking any cages by sitting on the fence. Now I'm being chastised for it?

Are you being chastised?  It was my intent to not chastise you just as surely as it was your intent to not take sides. ;-)

Here's a challenge - quote me chastising you. Bear in mind that I consider myself to be a member of the group I sometimes refer to as  "Most people". So, when I talked about "most people" I wasn't singling out you, I was talking about us.  Even though I've done a ton of DBTs, I resist them, probably as much as anybody. Difference is, I get over it some of the time.

Quote
I once (years ago) brought up the subject of CD burning, and that in an experiment I had done the newly burned CD sounded better than the stamped original. I didn't know why, it just did as far as I was concerned. I was pounded for my efforts, as "science says it isn't so", quoted Reed-Solomon etc, etc.

I would put that under the category "a little knowledge is a dangerous thing". You didn't deserve to get to be pounded because what you described is possible, if one understands more practical and theoretical details about CDs. Your correspondents didn't know the whole story.  Let me start it out like this.

What happens when there is a detected error reading a CD on a typical CD player? From a user standpoint there has been an organized, well-funded effort that has been going on for decades to keep you from knowing that there was an error. So as a user you often don't know there was an error  and have no practical way to find out.  Fact is that many CD players have an electrical line that changes state when there is an error reading a CD, and it is designed to be probed with an oscilloscope. I haven't looked for one or probed it lately, so this may not be true of the latest-greatest optical players. Back in the day I spent some time watching this line with my oscilloscopes., while I was playing various CDs. I even simulated damaged CDs by cutting narrow strips of black electrical tape to simulate damage to the disc.

There is a difference between ripping CDs and playing CDs on a regular optical disc player which is that the CD plays every track once and tries to do the best it can on the fly. When you rip the CD on a computer, most good ripping software watches for errors and tries to recover them by any number of means, most of which involve rereading the track until there is no error. Failing that it will reread the track many times and take advantage of the fact that the errors are inconsistent and  try to get a best average of the reads  and put that into the output file.

So knowing that, on several occasions I have taken  CDs that were damaged and played poorly on some CD player, ripped them, burned a CD and it played very nicely on that same player,  thank you!

Quote
Recently I found out that Reed-Solomon is only effective if the disc is tracking properly, and mistracking could easily have been the reason for the difference in sound.

I'll take that to be a concise version of what I just wrote.

Quote
I have seen many discs where the punched centre hole is completely misaligned with the data layer on the disc.

CD players have built-in means to compensate for that precisely and obtain error-free results provided the off-center condition is not too large.  They don't even need to look at the Reed-Solomon, they get a precise read on the eccentricity from the laser head tracking system (part of it is like a speaker voice coil)  and the error  gets compensated for right there.

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Because of my own experiences, I don't try to tell anyone which side of the fence to sit on, that's up to them. I am happy for anyone to believe what they like until science changes its mind and proves otherwise.

The science of digital audio as applied to CDs is pretty cut-and-dried and has been so  for a decade or more.  I can and have backed up everything I write with real world experiments, peer reviewed papers and textbooks, but I've already done so, so many times that getting me to do so today may not be easy. ;-)


Re: PCM, DSD - Trying to get my head round some basics

Reply #23
All by yourself, as relatively easy as it really is, most people resist doing proper listening tests.

This is not necessarily as easy as you make out. My first post on HAudio was in the ABX section for Foobar, which was making comparisons impossible as ReplayGain wouldn't set the levels between PCM and DSD correctly (possibly due to the noise, but I won't speculate as I get told off for doing that).

So I HAVE tried a proper listening test, and the technology failed me.

You may have given up far easier than I would have. I no doubt have many resources, both hardware, software, training and experience-wise, that most people don't have. For example, I  distributed free Windows ABX software that I wrote from scratch towards the end of the last Millenium. It worked in its way but it was bad enough that it had exactly the results I hoped for. Many others wrote better ones.

 I consider the current version of FOOBAR2000 to be good enough to recommend all the time.


Re: PCM, DSD - Trying to get my head round some basics

Reply #24
Recently I found out that Reed-Solomon is only effective if the disc is tracking properly, and mistracking could easily have been the reason for the difference in sound.
Hogwash.

Why do CD ripping programs have authentication databases if they get it right every time?
How does that defend the position that a CD-R will provide better quality audio than the original disc from which it was sourced, catastrophic failures, not withstanding?

Or are you only talking about completely uninteresting trivial cases?
Is 24-bit/192kHz good enough for your lo-fi vinyl, or do you need 32/384?

 
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