1: ALL CD players have error correction. ALL have time base correction which is inherent in the PLLs and de-shuffling RAM.
2: True but I haven't seen a player with analog filters since the early '80s. The square wave response is what is should be for a bandwidth limited system.
Try graphing a square wave from the math when you only use the first 5 terms. You'll basically see the 1 KHz CD square wave. What this means is the system is working correctly.
G²
An interesting article here, about a CD player that does actually correct the waveform, instead of fuzzing in with stuff that sounds plausible but is wrong.
IMO the non-correctable errors are far far more common than people like to think.
Clipping is the modern mastering process of chopping the tops of the music waveform off.
An externally hosted image should be here but it was not working when we last tested it.
Look at the negative point on this waveform, the corrected or 'declipped' section is also shown as a reconstructed waveform. The horizontal line at the bottom is the original clip.
All (99.9%) modern pop CDs are clipped. You will not believe me, so rip a CD and look at the waveform with Audacity or another waveform viewer to see for yourself. The CD music I listen to is not like this, as I use software to fill in the chopped off sections, the music sounds more dynamic and the treble far sweeter.
The longest clip on a CD is about 256 samples long on a Black Eyed Peas track. Clipping is a serious distortion that also wreaks havoc with DACs that are unprepared for a clip rate that exceeds 200Hz on many tracks.
The LP has a big advantage here: LPs are rarely clipped 😉
So in your view a CD player that tries hard to fill in the missing information based on what it does have is simply plausible but wrong....
... but software which does pretty much the same thing (adding information back which is impossible to know) gives an improvement.
So it seems that a guessing process done in software gets your approval, but hardware guessing does not. Jolly funny 😀
"The CD music I listen to is not like this, as I use software to fill in the chopped off sections, the music sounds more dynamic and the treble far sweeter. - Globulator"
I use SeeDeClip Duo Pro (Cute Studios) for processing some of the CDs and Music Videos that the "Loudness Brush" has caused to sound damn annoying.
I agree with the comments by Globulator. I often notice an improvement in the sound stage as well. I also normalise the .wav file (Sound Forge 9) to reduce the need to play around with the volume control when playing selections from various albums.
SandyK
I use SeeDeClip Duo Pro (Cute Studios) for processing some of the CDs and Music Videos that the "Loudness Brush" has caused to sound damn annoying.
I agree with the comments by Globulator. I often notice an improvement in the sound stage as well. I also normalise the .wav file (Sound Forge 9) to reduce the need to play around with the volume control when playing selections from various albums.
SandyK
Yes.
The lack of data quality error correction on a CD means that all parts of the waveform are suspect and I find the multiple reading done by computer 'rippers' reduces this error considerably.
No CD hardware deals with a clipped waveform, the same clipping found in all modern pop music. The clipped signal was deliberately put on the glass master: it is not an error.
I find listening to declipped music far better than clipped music, that's just my preference.. ... and when record companies stop clipping their music, I'll listen to that too 😉
So how have you compared the errors you get with the two methods? I'd like to learn how to do this on a typical CD player - how to get access to the uncorrected error count?
Ah, there we disagree.😀 Its an error all right, just not one caused by mistracking or scratches on the disk.
I don't knowingly buy clipped music these days - the last CD I bought which I found heavily clipped was Macy Gray 'Big'. I won't waste time on software which guesses what might have been chopped off. Just don't go there any more - eventually the record companies will get the message from customers not buying. Plenty of other stuff is available which hasn't been destroyed by being over modulated. But sure, its going to take a very loooooong time as they're very poor listeners.😛
Me too - I'd like to understand the difference between upsampling and oversampling. Up until now, I had assumed the 'difference' was mere marketing...
And I thought Google was everywhere 😉
Upsampling
Oversampling
Actually I tend to convert from 44.1/16 to 88.2/24 most of the time, so maybe I could have been more specific.
I don't need to upsample my LPs BTW..
HTH.
I want you to explain, and in the context of digital audio replay. I have no need for generalising Wiki pages.
I second Werner's request (again). Those two Wiki pages don't explain at all why you prefer upsampling to oversampling. Indeed, according to the explanations Wikipedia gives, there's considerable overlap between the two terms. So why do you employ one, and not the other and what's the importance to you of the distinction between them?
If speaking of CD, you can't.
Or rather, you can, but are not allowed to.
I agree wholeheartedly. You need to read up.
But Shannon, not Nyquist. Harry really didn't have to do that much with this, apart from having his name attached to Fs/2.
Hi,
you are requested to re read my post.
I didn't say analog filters,I said brickwall filters. Imagine a waterfall in a river.
The bloody signal is subjected to a fall in order to be released from the quantization distortion,that is ,to put it simply,non linear harmonics,that are generated by the whole A to D processing.Those filters introduce phase and transient intermodulation .Unfortunatelly, AUDIBLE,because it modulates the 20 hz to 20000hz audio band,tha we supposedly hear.
Jitter is the time distortion,that occures,from the restructuring of the signal.
It is simply the time difference that the processor,the cables,the input receiver introduce,and the correction circuits fail to correct.
If you are interested in a scientific explanation please google Digital Audio.
B.L.
As far as I understood it jitter is essentially caused by inaccuracies of the clock.
That is a digital sample is converted into an analog voltage at not exactly the right moment in time but rather a little bit to soon or too late.
Which would mean that if one were to have a perfect clock for playback one would still have/hear the inaccuracies of the AD convertor clock used during recording.
Presumably this error is random and therefore cannot be removed by any means even if one were to know which convertor/clock was used.
That is a digital sample is converted into an analog voltage at not exactly the right moment in time but rather a little bit to soon or too late.
Which would mean that if one were to have a perfect clock for playback one would still have/hear the inaccuracies of the AD convertor clock used during recording.
Presumably this error is random and therefore cannot be removed by any means even if one were to know which convertor/clock was used.
Hi,
I will try to explain, as simply as, it gets.
Upsampling is the reading of a signal with a bit rata higher than the standard 16 bit sampling,in order to read all the 16 original recorded bits.There ,in the low level signal portions of the recording,signal gets squashed by quantization distortion.The famous "most significant bit".
The first and generally bad recordings,were either a transfer to D,or were recorded directly D,but with the known analogue techniques.Low level,and peak limited,to avoid analogue tape saturation in the high frequences.
Therefore the signal was lost to quantization. Digital recording requires the highest ,possible level of signal with no clipping.By upsampling you are digging further into the signal,just to read and preserve the low level details,a significant part of the dynamics of the signal,and avoid to record at 0 db level.
A thing,that recording or mastering engineers use,just to get the kind of ssound that is needed,for the boombox and low to mid -fi fraternity.
There is the market,not the audiophile lunatic fringe.(I am a looney)
Oversampling is the introduction of phantom signal into the original at rates multiple 48 khz,96khz,192khz and so on ,therefore fooling the processor to think that the signal goes up to there,and halve the sampling to the wanted frequency.By doing so the brickwall filtering required, is moved much higher in the signal range,and by doing so the levels of intermodulation are minimized.
Certainly not a marketing trick,but a marketing tool to boot.
Hope this helps
B.L.
Hi
Exactly.No perfect tools exist.That's the reason why, we do make compromises.
People with lots of money,are able to buy less compromises.Aka High end.That's the name of the game.Fortunately our Creator, gave us the greatest gift he could. Adaptability.That goes directly to the myth of "system burning" We simply get used to the sound of a system,by filtering out what we perceive as distortion,and enjoy the music which is the destination.
Don't tell me that your music is less satisfying with a cheap monophonic transistor radio,than with a super hi end system.Simply we adapt and our brain reconstructs.
B.L.
I sincerely don't want to be rude, but after explaining things like this on the 'net for over 15 years now, you may want to appreciate that I cannot afford
any more time lavished on diplomacy, so ...
you are wrong on all accounts.
This said ... trying to beat down the misinformation on the internet that is related to the sampling theorem, signal theory, and digital audio truly
is like ... oh, never mind ...
This is pretty much the opposite of the analogue and digital recording practices IME.
When recording to analogue tape a level as high as possible is very much preferred (My mentor always said about analogue signals "If it ain't red it's dead!"). Tape saturation is a good thing in so far as it sounds really nice and is a lot more subtle than a compressor. Many engineers had their VU meters recalibrated to read zero when the signal was in fact +3 or +6dB.
There is practically no need for limiters in analogue recording and one can get away with minimal use of compressors (usually bass and vocals) due to the benign overload characteristics of tape.
Digital recording on the other hand demands comparatively low average levels to avoid clipping the signal which is very bad as a clipped digital signal basically injects bursts of white noise into the audio. Because of this most digital recording chains include a hard limiter before the convertor.
Originally it was recommended to record at -18dBFS on average, this was later upped to -12dBFS. These days with the loudness wars going on it is more like -6dBFS or even higher. This requires a lot of signal compression/limiting.
Sorry SY,
my use of scientific English language is below sea level,and I will make a mess of it,if I try to explain it further.I know the meaning of the subject,I am using it ,with the Wavelab editor, but I cannot write down the math side of it.My experience goes, as far as reading.
B.L.
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