-
96khz?
In somewhat simple terms could someone explain to me exactly what the sampling rate is and how it works. I hope I am even asking this question correctly. Thanks once again
-
Quote:
Originally Posted by cashlz
In somewhat simple terms could someone explain to me exactly what the sampling rate is and how it works. I hope I am even asking this question correctly. Thanks once again
I'd be better able to explain this to you if I only knew how to draw diagrams directly into the computer which I don't. My computer literacy is not what I wish it were. Having said that, let me try to make this concept understandable to you.
An analog audio waveform is comprised of a multitude of up and down "squiggles" (for want of a better term). These up and down excursions are constantly varying in both speed (frequency) and amplitude (volume).
In order to convert this analog waveform into the digital format, the waveform is "sampled" by an A to D converter at TWICE the frequency of the highest pitched sound that a given format is capable of. In the conventional "Redbook" CD format, this "sampling rate" is 44.1 thousand times per second, which means that the format can store and reproduce any frquency up to 22KHz (22 thousand "cycles" per second which is higher than any human that I know can hear). This sampling yields a voltage value of the waveform at that instant in time. It is translated into a 16 bit "word" of 1s and 0s which can represent any of a possible 65,536 different values.
In order to convert this digital signal back into analog form (so that it can be heard as sound), a D to A converter is used that takes each digital "word" sample and converts it to a voltage value that is theoretically the exact same value as the original analog signal voltage that the sample represents. Therefore, it's possible to reconstruct the original audio waveform to its original state - recreating the original sound in (almost) perfect fashion.
I hope this made the whole concept a bit clearer for you
-
Ok, I gotcha that makes alot of sense and I suppose thats why its called "true" 96. Thanks alot!
-
Quote:
Originally Posted by cashlz
In somewhat simple terms could someone explain to me exactly what the sampling rate is and how it works. I hope I am even asking this question correctly. Thanks once again
I'll offer perhaps an ever simpler explanation: sampling is analagous to a connect-the-dots picture. For every small sample of time, numbers or "dots" are assigned to the waveform. In context to your reference to a higher than Redbook standard rate, the problem with RB is that under certain conditions, there are not enough dots to look like a smooth line. During very quiet passages, the number of bits firing (again the dots) is greatly reduced, causing the RB standard to go deaf. Similarly, while the RB standard can handle simple waveforms at very high frequencies, complex harmonic overtones produced by multiple instruments overwhelms the fixed budget of dots available. Hence the reason behind the high resolution DVD-A and SACD standards.
rw
-
In context to your reference to a higher than Redbook standard rate, the problem with RB is that under certain conditions, there are not enough dots to look like a smooth line.
Well, your analogy is a non starter. There are no dots to connect at the CD player output, it is a continuous wave. THe Red book is sufficient.
During very quiet passages, the number of bits firing (again the dots) is greatly reduced, causing the RB standard to go deaf.
Nonsense.
Similarly, while the RB standard can handle simple waveforms at very high frequencies, complex harmonic overtones produced by multiple instruments overwhelms the fixed budget of dots available.
More nonsense.
Hence the reason behind the high resolution DVD-A and SACD standards.
rw
No, that is not the reason at all. Maybe a good marketing ploy, yest, to appease the audiophile community. Mastering and archiving and multi channel playback is the reasons for DVD-A.
-
Quote:
Originally Posted by mtrycraft
There are no dots to connect at the CD player output, it is a continuous wave. THe Red book is sufficient.
Ah yes, the ever perspicacious comments from our resident ditch digger. I have no doubt that RB is sufficient for you.
rw
-
Quote:
Originally Posted by E-Stat
, the problem with RB is that under certain conditions, there are not enough dots to look like a smooth line.
rw
This would, in a crude way, be true if the output stage did not include an anti-alias filter. Of course, any properly designed cd player has such a low pass filter. Therfor the result is a smoothed sinewave, as it should be. Their is a trend of some 'exotic' cd players to not include an antialias filter(they claim this, though, still a mild low pass filter(1st order), otherwise the aliased signal could possibly damage electronics and/or speakers--this insufficient slope will not provide adequate filtering of aliased data to prevent audible side-effects), the aliased bi-products can result in audible inter-modular artifacts when combined with the main audible passband. These would most probably be identified in a blind test easily! :-)
-Chris
-
Quote:
Originally Posted by E-Stat
Ah yes, the ever perspicacious comments from our resident ditch digger. I have no doubt that RB is sufficient for you.
rw
If this is true, I am his apprentice-in-training. He digs, I clean his shovel. :-)
Seriously, though, RB standard is adequate for me, except for that it only allows for two channels. For some music, multi-channel is very important, IMO.(New age, pop, etc. that has begun to implement such surreal sound effects in it intended to be played over multichannel as released in it's SCAD and/or DVD-A version(s) .)
-Chris
-
Quote:
Originally Posted by E-Stat
Ah yes, the ever perspicacious comments from our resident ditch digger. I have no doubt that RB is sufficient for you.
rw
Why didn't you address the issues you messed up?
-
Quote:
Originally Posted by WmAx
He digs, I clean his shovel. :-)
-Chris
Great. :) As you can see, it meeds a lot of cleaning :)
-
I have never understood this....
Quote:
Originally Posted by woodman
I'd be better able to explain this to you if I only knew how to draw diagrams directly into the computer which I don't. My computer literacy is not what I wish it were. Having said that, let me try to make this concept understandable to you.
An analog audio waveform is comprised of a multitude of up and down "squiggles" (for want of a better term). These up and down excursions are constantly varying in both speed (frequency) and amplitude (volume).
In order to convert this analog waveform into the digital format, the waveform is "sampled" by an A to D converter at TWICE the frequency of the highest pitched sound that a given format is capable of. In the conventional "Redbook" CD format, this "sampling rate" is 44.1 thousand times per second, which means that the format can store and reproduce any frquency up to 22KHz (22 thousand "cycles" per second which is higher than any human that I know can hear). This sampling yields a voltage value of the waveform at that instant in time. It is translated into a 16 bit "word" of 1s and 0s which can represent any of a possible 65,536 different values.
In order to convert this digital signal back into analog form (so that it can be heard as sound), a D to A converter is used that takes each digital "word" sample and converts it to a voltage value that is theoretically the exact same value as the original analog signal voltage that the sample represents. Therefore, it's possible to reconstruct the original audio waveform to its original state - recreating the original sound in (almost) perfect fashion.
I hope this made the whole concept a bit clearer for you
If I have understood what you are saying there are 65,535 different possible values a given point on a waveform can have within a range of 0(?) - 22,000 Hz on a CD. This implies to me that the accuracy of the representation of that waveform is limited.
This would mean that for 96KHz with a 16 bit word the accuracy of the representation of the waveform would be worse as the range is now 96,000/2 or 48 KHz to be represented with the same 65,535 possible values.
However if we use a 24 bit word (with 16,777,215 possible values) the accuracy of the representation of the waveform is higher.
Correct? Or am I a mile off base again?
-
Quote:
Originally Posted by mtrycraft
Why didn't you address the issues you messed up?
I'll let Sir Terrence enlighten you instead. Good luck.
http://forums.audioreview.com/showpo...6&postcount=71
rw
-
Quote:
Originally Posted by E-Stat
I am curious. Why should the opinion of this person you refer too overturn the proven acceptable bandwidth, such as the limit CD employs? I mean, their is nothing wrong with having suspicions(they may even be correct-heck, aliens may be really probing anuses!), but to go around quoting as fact is a bit silly, IMO, when you are claiming the opposite of what has been peer reviewed and accepted as the standard by AES. No one has yet, demonstrated in repeatable controlled tests, that the bandwidth used on CD format is discernable by subjects as compared to a broader bandwidth in music playback. If they had, then the current bandwidth standards accepted for playback would be rejected and a new standard adopted.
-Chris
-
Quote:
Originally Posted by WmAx
I am curious. Why should the opinion of this person you refer too overturn the proven acceptable bandwidth, such as the limit CD employs?
He is an experienced pro with compelling evidence as to what a large number of people already hear. If mtry should decide to debate STTT, it will be interesting to see the reaction from his sophomoric comments.
Quote:
Originally Posted by WmAx
If they had, then the current bandwidth standards accepted for playback would be rejected and a new standard adopted.
Two new standards have already been widely adopted.
rw
-
Quote:
He is an experienced pro with compelling evidence as to what a large number of people already hear.
You would have to define what you mean by compelling. His word, compared to peer reviewed scientific study, is certainly not 'compelling evidence' too me.
Quote:
Two new standards have already been
Yes, as far as a 'format' standard is concerned. But this is not based on an audible basis. The AES [1]standard I refer to is a base minimum standard concerning bandwidth playback audibility, not a specific medium format.
-Chris
[1] "Which Bandwidth Is Necessary for Optimal Sound Transmission?" G. Plenge, H. Jakubowski and P. Schone
-
Quote:
Originally Posted by E-Stat
I don't need luck at all.
I have never been a big fan of redbook CD, however I am a HUGE fan of multichannel DVD-A AND SACD because they come the closest to recreating a live event.
He is a big fan of multi channel music or movie sound. Me too. 2 channel just cannot reproduce the soundstage. I never claimed otherwise. Actually, 5.1 is not enough either. Tomlinson Holmann has demonstrated this too with his 10.2 system demo.
Two channel vinyl(or CD) fail meserably in this area because by shear fomat design they misplace spatial cues, and have problems with handling the harmonics of cymbal crashes(which have huge amounts of energy to 40khz) and percussive transients of drums, piano's chimes and glocks. Anytime a format rolls off the highest frequencies, it will have a horrible time with the leading edge of transients.
He is just absolutely wrong on the ultrasonic information having any meaning. He just cannot come to grips with that. I have cited very recent papers on it. His problem.
Enjoy your confusion.
Oh, make sure you apply your spelling criticism to all, or you have a hardup for me?
-
Quote:
Originally Posted by WmAx
These would most probably be identified in a blind test easily! :-)
-Chris
But he would have no idea about that. He abhors blind tests and runs from them as if they were the plague. :)
-
Quote:
Originally Posted by WmAx
I am curious. Why should the opinion of this person you refer too overturn the proven acceptable bandwidth, such as the limit CD employs? I mean, their is nothing wrong with having suspicions(they may even be correct-heck, aliens may be really probing anuses!), but to go around quoting as fact is a bit silly, IMO, when you are claiming the opposite of what has been peer reviewed and accepted as the standard by AES. No one has yet, demonstrated in repeatable controlled tests, that the bandwidth used on CD format is discernable by subjects as compared to a broader bandwidth in music playback. If they had, then the current bandwidth standards accepted for playback would be rejected and a new standard adopted.
-Chris
These guys should pick up THe Digital CD by Makus Erne, Amazon. It has some very interesting tests that can be done single blind between tracks:) Such as influence of sampling rates, number of bits, etc. But then, I am deaf with a boombox to boot :D
-
He is an experienced pro with compelling evidence
I must have missed it. Please point this out. He has no compelling evidence. All experienced pros are created equal? They are immune from being mistaken, or plain wrong? Really?
as to what a large number of people already hear.
Really? Like you hear in sighted lisening? That is not evidence, not compelling. It is a good story.
If mtry should decide to debate STTT, it will be interesting to see the reaction from his sophomoric comments.
Maybe he should debate real experts then. Would that impress you?
Two new standards have already been widely adopted.
rw
And? What is that supposed to prove? Oh, aren't they multi channel formats?
I love multi channel formats. Stereo is just incapable of rendering a good soundfield. That was demonstrated by Bell Labs in 1930s, for your information.
-
I was hoping someone technical would answer my question which was:
"If I have understood what you are saying there are 65,535 different possible values a given point on a waveform can have within a range of 0(?) - 22,000 Hz on a CD. This implies to me that the accuracy of the representation of that waveform is limited.
This would mean that for 96KHz with a 16 bit word the accuracy of the representation of the waveform would be worse as the range is now 96,000/2 or 48 KHz to be represented with the same 65,535 possible values.
However if we use a 24 bit word (with 16,777,215 possible values) the accuracy of the representation of the waveform is higher.
Correct? Or am I a mile off base again?"
Anyone???
-
Quote:
Originally Posted by cashlz
In somewhat simple terms could someone explain to me exactly what the sampling rate is and how it works. I hope I am even asking this question correctly. Thanks once again
It's simple. You have the title "96kHz" - which is 96,000 cycles per second....in sampling terms, that means you're going to slice something up into 96,000 pieces every second.
Then, each one of those slices is assigned a number to represent the sample's position(yes, it's referenced to something). So now you have a numeric representation of you're original waveform and you can store it on a digital storage medium, like a CD. (BTW - those "slices" are not sequential on a CD, so if you get a scratch, it doesn't wipe out a large chunk of continuous data, but that's another topic of discussion.)
You do the opposite to reassemble it. Just take those numbers, put the sample back into position according to it's value and glue all of them back together, that's the basics.
-Bruce
-
Quote:
Originally Posted by mtrycraft
Oh, make sure you apply your spelling criticism to all
Everyone has a typo now and again. I don't exclude myself either. Yours, however, are a regular event and downright funny at times. Sorry, I have trouble taking anyone serious who demonstrates the writing ability of a high school sophomore. Articulateness is just not you. Nor apparently are your powers of musical discernment.
http://forums.audioreview.com/showpo...2&postcount=59
Quote:
Originally Posted by mtrycraft
...or you have a hardup for me?
Do you mean hard on?
My senior high school english teacher failed anyone whose paper contained a comma splice. Your sentence referenced above is an example of one.
rw
-
Quote:
Originally Posted by maxg
I was hoping someone technical would answer my question which was:
"If I have understood what you are saying there are 65,535 different possible values a given point on a waveform can have within a range of 0(?) - 22,000 Hz on a CD. This implies to me that the accuracy of the representation of that waveform is limited.
This would mean that for 96KHz with a 16 bit word the accuracy of the representation of the waveform would be worse as the range is now 96,000/2 or 48 KHz to be represented with the same 65,535 possible values.
However if we use a 24 bit word (with 16,777,215 possible values) the accuracy of the representation of the waveform is higher.
Correct? Or am I a mile off base again?"
Anyone???
Not quite. The 65,535 points represent volume, that's all. The sampling rate determines the maximum frequency response, that's all.
In the case of volume, that number represents the 16 bit word, which equates to a dynamic volume range of 96dB. Add 6dB more per bit and 24 bit becomes 144dB theoretically, except that you run into the thermal noise floor first.
-Bruce
-
FLZ,
Thanks for the reply - I am now formally giving up on ever understanding this stuff!! The 65,535 points represent volume? OMG - er....wait a moment...volume = amplitude of the waveform...nope..thought I had it back...gone again. Time to dust off the vinyl again.
-
Quote:
Originally Posted by maxg
The 65,535 points represent volume? OMG - er....wait a moment...volume = amplitude of the waveform...nope..thought I had it back...gone again. Time to dust off the vinyl again.
Moving from the theoretical to the real world, however, we find compromises. While the DACs themselves may be 16 bit, the effective number of bits (ENOB) is always fewer. As the level of a signal drops, it is represented by fewer and fewer bits. The result is called quantization distortion - the point at which the DAC goes deaf. The "solution" is a novel scheme called dither. Although it sounds counterproductive, dither introduces noise intentially to the signal to keep the bottom from falling out and is later removed. While there are many flavors and the process is largely successful, I think that some of the apparent quiet and lack of low level detail from the RB standard is simply uncorrected amounts of this distortion, IMHO. I have high hopes for the high rez standards.
rw
|