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In Reply to: RE: he's not completely wrong ... posted by knewton on March 14, 2012 at 10:35:20
"it [CD] can be far less forgiving than those same predecessors in terms of recording and playback chain implementation."
Well, I guess that's possible but the example I used was an analog recording passing along problems caused by previous digital processing and domain conversions.
Regards, Rick
Follow Ups:
By less forgiving of recording and playback implementation, I meant in terms of the subjective listening qualities I'd listed earlier. I've listened to many relatively noisy, or high distortion, or limited bandwidth analog sources which, none the less, remained 'musical' and 'natural' sounding in ways which too often eludes CD to my ears. Examples are, FM radio, cassette tapes, and well worn LPs. Again, the issue which critics have long had with CD are not related to the traditional objective parameters. Indeed, CD has always excelled at those. The issue has always been certain, yet important subjective qualities. These have been well enumerated over the years.I hope you realize that I might provide an example to illustrate my chosen point without it necessarily having to have something to do with an example you provided to illustrate your chosen point.
_
Ken Newton
Edits: 03/15/12 03/15/12 03/15/12 03/15/12 03/15/12 03/15/12 03/15/12 03/15/12 03/15/12
"I hope you realize that I might provide an example to illustrate my chosen point without it necessarily having to have something to do with an example you provided to illustrate your chosen point."
Brain hurts... But I assume after breakfast I'll agree.
Do you think that there is something specifically wrong with CD's as a medium or is it digital audio in general that is unforgiving?
By the way I suspect the answer is yes, but I am interested in your take on it.
Regards, Rick
The technical problem I have with digital audio in general, and CD in particular, is that the intended signal (music) contains time-domain sensitive information. The 44.1ksps CD channel rate was designed with the frequency-domain requirements of music in mind. As far as I can tell, little to no consideration was given to the time-domain implications of the medium, either for recording or playback.The root of the time-domain problems are two-fold, as I see them. First, the fact that CD's channel bandwidth of 22.05kHz is so close to the recorded information bandwidth creates the requirement for very sharp anti-alias (recording) and anti-image (playback) filter responses. Which necessarily have a poor time-domain response, manifesting as the now fimiliar high-Q filter ringing. As it turns out, this severe ringing is fundamental to producing an accurate frequency-domain reconstruction of the original signal upon playback. Yes, I'm also aware that such ringing occurs at the edge of the ultrasonic range and should be pretty much inaudible, but that is only half the story, to which I'll shortly return. If the information to channel bandwidth ratio were wider, as it can be with high-res. digital, then both the anti-imaging and anti-aliasing filters could be much less sharp, with greatly reduced time-domain distortion. Mike Story of dCS has published a paper concluding that to be one of the reasons why high sample rate digital sounds superior to CD.
My own empirical experiments lead me to suspect that a second, non-obvious mechanism is also at work. I suspect that the time-domain problems extend to the dynamic inter-action, or intermodulation, of the near ultrasonic ringing responses of the multiple SINC filters utilized from recording, to mixing or other re-sampling, and finally to playback. Resulting in artifacts within the audible range.
One of my self designed experimental DACs contains a programmable digital SINC filter. This programmable filter has enabled me to empirically evaluate oversampling, non-oversampling, and apodising digital filters. Here's what I heard. As is well known by now, non-oversampling, aka, NOS - which eliminates the playback SINC filter but does not affect the recording and mixing SINC filters - produces the natural and non-fatigueing sound so typically lacking in CD. Apodising - which retains the playback SINC filter, but removes the affect of the recording and mixing SINC filters - sounds equally natural and non-fatiguing as NOS. Oversampling - with all SINC filter responses in place - on the other hand, produces the typically fatiguing and course CD sound.
My hypothesis is that the primary source for what we have come to know as digititus, or traditional CD sound, is the dynamic intermodulation of the multiple SINC filter time-domain (ringing) responses of the standard CD recording, mixing, and playback chain. Eliminating one or the other of these sharp filter responses greatly restores the natural and non-fatiguing quality otherwise absent. This hypothesis would also explain why high sample rate audio is often disappointingly not completely rid of unpleasant CD type artifacts. The use of SINC filters across a high sample rate chain could still produce time-domain filter response interaction which are audible. I will surmise that high sample rate recording-playback chains which take advantage of the extra channel bandwidth not for increased frequency-domain signal capture, but for utilizing anti-alias and anti-image filters having much less ringing in their time-domain responses, will subjectively provide the best high sample rate audio quality.
I've not yet developed an experiment to test this hypothesis, so it may prove faulty in so far as the exact distortion mechanism responsible is concerned. The empirical results, however, have been consistant and very obvious.
_
Ken Newton
Edits: 03/15/12 03/15/12 03/15/12 03/15/12 03/15/12 03/15/12 03/15/12 03/15/12 03/15/12 03/15/12 03/15/12 03/15/12 03/15/12 03/15/12 03/15/12 03/15/12 03/15/12
So you hold it against CD for forcing us into such a low SR huh?
Well, can't argue with that logic, it's too true.
As to the dynamics, I'm afraid that is also, the BW has to accommodate all of the data which includes the sideband energy in case you want to do something rash like tap a cymbal...
I wish I knew the link, but I saw an interesting video from one of the Stereophile show panels where someone, Keith Johnson I think, was presenting data showing how things went when the upper part of the spectrum was modulated. Poorly. Gross distortion from the largely missing USB.
And yet CD's can sound very good. Sometimes.
Rick
I make 44/16 recordings using an apodizing filter. They will sound best when played back with a SINC filter, and perhaps slightly rolled off but still not unpleasant when played back through an apodizing filter. Older recordings made with a SINC filter will sound smoother when played back with an apodizing filter. Thus for best results one may wish to have a choice of playback filters available.
The problem with apodizing filters is that they will ring unless they are very slow roll-off. There is a tradeoff between loss of high frequency detail (dullness and air), harshness (aliasing) and image blurring (ringing) that are unavoidable mathematically given the 44.1 kHz sampling rate. If the original live music had significant high frequencies, there will be unavoidable loss of fidelity when running at this sample rate, and it will appear in different guises according to the tradeoffs made by the recording engineers or equipment designers. Because of these unavoidable tradeoffs, 44.1 kHz recordings will never reproduce anything close to the sound of a live microphone feed, but results can be very good if things go well.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
If a piece is recorded and mixed at 24/96 digital, wouldn't releasing it in that format avoid all problems introduced by converting among representations?
"If a piece is recorded and mixed at 24/96 digital, wouldn't releasing it in that format avoid all problems introduced by converting among representations?"
One would think so, but then this place is an "asylum." :-)
There are people who believe two files that have identical file content (as evidenced by a MD5 file checksum, for example) can sound different, depending on their history of file creation. (Example: WAV file converted to FLAC, then converted back to WAV. When the two WAV files are played back they are said to sound different.)
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
Hmmm ... said to "sound different?" I just round-tripped a 24-96 WAV file to/from FLAC and got back exactly the same file. (Verified by null test against the original.) What am I missing?
"Hmmm ... said to "sound different?" I just round-tripped a 24-96 WAV file to/from FLAC and got back exactly the same file. (Verified by null test against the original.) What am I missing?"
1. Compare files with MD5 file checksum. (Software below.) You may find that the file contents are different even though the audio samples are the same. This can be due to different metadata or other header information in the file. While this "shouldn't" affect sound quality the altered bits are known to the player software and could do strange things.
2. If the two files coexist simultaneously, they must have different storage locations and file names. While this "shouldn't" affect sound quality it could.
3. Playing the same file (or equivalent files) on different occasions may sound different due to changes in physical environment, component aging, warmup, and most likely, one's different mental state. One may falsely attribute differences for these reasons to some other cause such as switching to an equivalent file.
4. Playback of files from hard drive may involve different electrical activity depending on how these were written, i.e. the timing of bits coming off the spinning rust, the operation of RS error correcting codes, etc. These could somehow affect the sound.
I expect the sound to vary each time I play the music, if only because I've heard it one more time. Before I consider the possibility that a digital copy has somehow become "different" I listen multiple times, even in the case of large differences such as "clicks" and "pops". I will notice, but not attribute until confirmed. If I repeatedly were to hear significant differences between files with identical audio samples I would get rid of my DAC and replace it with one that was more immune from extraneous differences on the input signal or electrical power. I am not concerned about minor differences, rather just those that might affect the realism or musical enjoyment of the playback. Given a choice of two DACs, one that provided consistently good sound or one that provided inconsistent sound that on average was slightly better, I would have absolutely no use for the latter DAC. For me, audio components are tools that are used to make recordings or enjoy music and have no intrinsic interest otherwise. I am not interested in "moody" or "flaky" components.
The only times that I listen hypercritically to music is when I am evaluating a recording that I am making or when I am evaluating or setting up my system. I try to spend as much time as possible just listening and enjoying the music and have no interest in swapping components to get a slightly different sound. If I want to revoice my system I have ample ways of doing so at low cost, including repositioning my listening chair, repositioning my speakers, or readjusting the crossover (11 adjustments total). Usually a simple volume adjustment or polarity reversal suffices to make a recording sound good, unless the recording "needs work" or is "hopeless". Along this line, in over 50 years I have purchased only three amplifiers for my main system. I am not so foolish as to replace components in the vain attempt to make poor recordings sound good. (I prefer to be foolish in other ways. :-) )
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
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