[SlimDevices: Audiophiles] Re: Upsampling
Patrick Dixon;131267 Wrote: Correct - although you still need the analogue filter, it's much a much less demanding spec. OK, just to make sure I've understood this correctly. Say we have a 44.1kHz signal. Putting it through a non-oversampling DAS, the repeated spectra start at 44.1kHz. If we upsample it to 88.2kHz and then feed it through a NOS DAC running at 88.2kHz, then the repeated spectra start at 88.2kHz, which of course allows for a less destructive (in the audio band) post-DAC analogue filter. But in practice how is this any different than using simple 2x oversampling? That would move the repeated spectra out to the same place. I'm still struggling to see the point of upsampling rather than oversampling. Indeed, I seem to recall that the theory behind oversampling says that it doesn't matter what the values of the extra samples you stuff in are. So in that respect, upsampling seems to be just a specific case of oversampling. The only purpose I can see for upsampling (with interpolation) is that it allows you to use non-integral sample rate multiplications (eg. 44.1 - 96). But what's the practical benefit of that? -- cliveb Performers - dozens of mixers and effects - clipped/hypercompressed mastering - you think a few extra ps of jitter matters? cliveb's Profile: http://forums.slimdevices.com/member.php?userid=348 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
I think it's really a matter of semantics. Oversampling is performed solely for the purpose of reducing the harmful effects of brickwall filters. Upsampling is really a form of sample rate conversion, and is necessary for systems that have asynchronous timing (i.e. input and output clocks are different). I think it is the marketers who have created the mystique of upsampling, and to be honest, clive, I agree with your premise. Upsampling could be considered a form of oversampling, and in fact, most upsampling systems also employ oversampling! -- ezkcdude SB3-Derek Shek TDA1543/CS8412 NOS DAC-MIT Terminator 2 interconnects-Endler Audio 24-step Attenuators (RCA-direct)-Parasound Halo A23 125W/ch amplifier-Speltz anti-cables-DIY 2-ways + Dayton Titanic 10 subwoofer He's not hi-fi, he's my stereo. ezkcdude's Profile: http://forums.slimdevices.com/member.php?userid=2545 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
Patrick Dixon;130951 Wrote: This is completely wrong - Upsampling/oversampling doesn't invent any data! Interpolation is actually a filtering process which removes the repeat spectra that are created by the upsampling/oversampling process. This thread seems to have developed in all kinds of directions since I last dropped in, but I wanted to try and clarify the point I was trying to make about the difference between oversampling and upsampling, and which Patrick has disagreed with. 1. Oversampling, as the term is generally used, is the process of adding extra data points (usually zeros) between the real data points. This has the effect of moving the repeat spectra higher up the frequency range, so they can be removed with a much gentler post-DAC filter. I think nobody is in dispute over this. 2. Upsampling, as the term is generally used, similarly involves adding extra data points. But instead of the normal zeros used in oversampling, interpolation is used. If that upsampled signal is then subsequently filtered in the same way as an oversampled one, the outcome is the same. But it is my understanding that this is *not* what is done. From what I've read about upsamplers, the whole purpose is to use the results to feed a DAC running at the higher sampling rate, and to *retain* the additional high frequencies. If this is not the intention, then I have to ask what is the point of upsampling rather than oversampling? If you filter the results in the same way, it doesn't matter what the extra data points are. But if upsampled signals are indeed used to feed higher speed DACs, and the higher frequencies are retained, then I stand by my original point that the high frequency content (over and above that which would have been produced from a normal D/A process) is bogus. -- cliveb Performers - dozens of mixers and effects - clipped/hypercompressed mastering - you think a few extra ps of jitter matters? cliveb's Profile: http://forums.slimdevices.com/member.php?userid=348 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
I give up! -- Patrick Dixon www.at-tunes.co.uk Patrick Dixon's Profile: http://forums.slimdevices.com/member.php?userid=90 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
dwc;131160 Wrote: I think all of the above hulabaloo builds a strong case for those of us on the fringe with non-oversampling filterless DACs. The second benefit is there is no dog poop in the yard because all the neighborhood dogs can't handle the super-high frequency noise. :) non-os dacs, keeping it real. I think it all boils down to how it sounds, not how it works... ;-) -- PhilNYC Sonic Spirits Inc. http://www.sonicspirits.com PhilNYC's Profile: http://forums.slimdevices.com/member.php?userid=837 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
Pat Farrell;131224 Wrote: Ah, well, I hate to break this to you, but according to the Neumann site, under their entries for Historic Microphones http://www.neumann.com/?lang=enid=hist_microphonescid=km83_publications they say that the frequency response of the KM83 is 40 - 16K hz. Sorry but I beg to differ ... ... look at the plotted frequency response rather than than the tabulated specs. Pat Farrell;131224 Wrote: The key is that the 20-20kHz bandwidth was well accepted long before Sony and Philips picked 44.1kHz for their sampling rate. Yes, and I was exagerating for effect, but all the same I would argue that a lot of that acceptance resulted from historic technological limits and the limits (at that time) of psychoacoustics (not that we are so much further now, although Functional NMR is beginning to give greater insights into our response to sonic events) ... I think that we are in serious agreement [even if we nit pick on a few details :-)]! -- reeve_mike reeve_mike's Profile: http://forums.slimdevices.com/member.php?userid=995 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
reeve_mike wrote: Pat Farrell;131224 Wrote: Ah, well, I hate to break this to you, but according to the Neumann site, under their entries for Historic Microphones http://www.neumann.com/?lang=enid=hist_microphonescid=km83_publications they say that the frequency response of the KM83 is 40 - 16K hz. Sorry but I beg to differ ... ... look at the plotted frequency response rather than than the tabulated specs. The frequency response plots all show a tail off at high frequencies. Not quite at 16KHz, but certainly below 20KHz. R. ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
Patrick Dixon;131182 Wrote: I give up! I take it that this is a response to my second post in this thread. Saying I give up doesn't contribute much. It would seem that you believe that my understanding of what upsampling does is wrong. I'm genuinely interested in finding out whether I've misinterpreted what upsampling is all about. Further thought suggests to me that it's possible that the interpolation may not actually generate any higher frequencies, and that upsampling prior to the DAC is simply a digital alternative to the analogue post-DAC filter used in a traditional D/A conversion process. If that is the case, then I'm happy to accept that I have misunderstood what upsampling is all about. But all the published material I've seen (which admittedly is on the marketing side of things) from the likes of dCS seems to imply that the purpose of upsampling is to fill in the gaps and allow a higher sampling rate DAC to be used, with the clear implication that this produces extra high frequencies and gives the eventual signal more resolution. Which is it? -- cliveb Performers - dozens of mixers and effects - clipped/hypercompressed mastering - you think a few extra ps of jitter matters? cliveb's Profile: http://forums.slimdevices.com/member.php?userid=348 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
cliveb;131259 Wrote: I take it that this is a response to my second post in this thread.Sorry, it wasn't aimed you particularly. cliveb;131259 Wrote: Further thought suggests to me that it's possible that the interpolation may not actually generate any higher frequencies, and that upsampling prior to the DAC is simply a digital alternative to the analogue post-DAC filter used in a traditional D/A conversion process. If that is the case, then I'm happy to accept that I have misunderstood what upsampling is all about.Correct - although you still need the analogue filter, it's much a much less demanding spec. cliveb;131259 Wrote: But all the published material I've seen (which admittedly is on the marketing side of things) from the likes of dCS seems to imply that the purpose of upsampling is to fill in the gaps and allow a higher sampling rate DAC to be used, with the clear implication that this produces extra high frequencies and gives the eventual signal more resolution. That's also correct on a simplistic level. However, the extra HF resolution that they're 'claiming' doesn't come from the interpolation process, it comes from being able to specify a simpler analogue reconstruction (post DAC) filter. So any improved HF response is as a result of not (analogue) filtering as sharply as you might have done with a NOS DAC. However, some NOS DACs don't bother with any post-DAC filters at all (instead relying on downstream bandwidth limitations in the speakers/human hearing). There's really no 'magic' to Digital Signal Processing - it's all maths. -- Patrick Dixon www.at-tunes.co.uk Patrick Dixon's Profile: http://forums.slimdevices.com/member.php?userid=90 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
seanadams;130863 Wrote: No, it interpolates. So you get something maybe like: 0, 0.5, 1, 1.5, 2 and so on. What you're talking about is oversampling - just another name for upsampling, but usually used in reference to what modern DACs do internally. It is fundamental to how they work and yes, the smaller steps require less filtering (and yield better linearity, lower noise etc). The DAC in transporter oversamples by 128x, so a 44.1 signal is actually converted to analogue at a sample rate of 5.6 MHz... a high resolution indeed. Now, what's stupid is taking 44.1 CD rips, resampling them to 96KHz and then re-saving to disk, thinking you've given it more breathing room or opened up the high end or whatever. It's total nonsense, exactly like on CSI where they zoom in on a single pixel, click ENHANCE and then read a license plate from a mile away. It don't work that way. I agree. The problem is, however, that a lot of audiophiles report all sorts of differences depending on upsampling etc. My guess is that it has to do with noise (HF) being downconverted in different ways depending on the DACs operating frequency. It is just a hypothesis, but would be interesting to investigate. Another possibility is that it has to do with the jitter spectrum. -- P Floding P Floding's Profile: http://forums.slimdevices.com/member.php?userid=2932 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
Pat Farrell;130865 Wrote: seanadams wrote:[color=blue][color=green]So it is more than twice as good as the SACD single bit rate of 2.82 MHz, eh? Any chance that the DAC in the Transport actually is 5.64 mHz? Comparing sample rates for 1-bit DSD/SACD vs. redbook/upsampled/oversampled sample rates isn't really an apples to apples comparison. Here's a nice 1-pager that describes upsampling/oversampling benefits/challenges in some pretty simple terms. http://www.resolutionaudio.com/Up-Oversampling.pdf -- PhilNYC Sonic Spirits Inc. http://www.sonicspirits.com PhilNYC's Profile: http://forums.slimdevices.com/member.php?userid=837 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
ezkcdude;130930 Wrote: Namely, upsampling shifts aliasing artifacts (so-called ghost images) to a much higher (inaudible) frequency range.Actually this is not quite correct. Alias artifacts are 'fixed' in the signal at Analogue to Digital conversion. Once there they can't be removed. What upsampling/oversampling does do, is to allow the use of digital filters (interpolators) which 'push' the repeat spectra higher and make the job of the (analogue) post-DAC reconstruction filter easier. Many of the current NOS designs don't use a post-DAC filter anyway. -- Patrick Dixon www.at-tunes.co.uk Patrick Dixon's Profile: http://forums.slimdevices.com/member.php?userid=90 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
cliveb;130937 Wrote: As the terms are typically used, it's OVERSAMPLING rather than UPSAMPLING that shifts aliasing higher up the frequency range and makes life easier for the filters. Pretty much every DAC on the planet does oversampling these days (with the exception of the niche NOS ones, which some might consider to be out on the fringe). It's mathematically sound. In contrast, upsampling (as the term is generally used) involves INVENTING additional data (usually by interpolation) in the expectation that it will deliver improved high frequency resolution. But this extra data that's invented can't ever be known to be correct. Quite a lot of the time, it'll be wrong. It doesn't recreate the high frequencies that were discarded when the recording was sampled at whatever lower frequency was used - that's impossible (as Sean pointed out earlier). In other words, the additional high frequencies generated are NOISE and/or DISTORTION. So how come people think upsampled digital audio sounds better? Maybe for the same reason that they think that analogue sounds better: that noise/distortion might actually be euphonic (if it's audible at all). I think you are inventing a distinction that doesn't really exist. Oversampling, as has been done for ages internally in CD players, also invents data, since the oversampled word stream is run through a digital FIR filter whose task it is to filter away, in the digital domain, mirror images above 20 kHz. Doing so allows the filter to be phase-linear, which is not possible in the analogue domain. (In the olden days an extremely steep analogue filter tried to remove anything above 20kHz, resulting in massive phase shift in the treble region.) -- P Floding P Floding's Profile: http://forums.slimdevices.com/member.php?userid=2932 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
cliveb;130937 Wrote: In contrast, upsampling (as the term is generally used) involves INVENTING additional data (usually by interpolation) in the expectation that it will deliver improved high frequency resolution.This is completely wrong - Upsampling/oversampling doesn't invent any data! Interpolation is actually a filtering process which removes the repeat spectra that are created by the upsampling/oversampling process. The actual interpolation process doesn't improve the high frequency resolution at all - if anything the interpolation filter will remove some of the original high-frequency information. -- Patrick Dixon www.at-tunes.co.uk Patrick Dixon's Profile: http://forums.slimdevices.com/member.php?userid=90 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
Patrick Dixon;130940 Wrote: Actually this is not quite correct. Alias artifacts are 'fixed' in the signal at Analogue to Digital conversion. Once there they can't be removed. They can be shifted to a higher frequency range, so that a more gentle reconstruction (anti-aliasing) filter can be used. That is my understanding. -- ezkcdude SB3-Derek Shek TDA1543/CS8412 NOS DAC-MIT Terminator 2 interconnects-Endler Audio 24-step Attenuators (RCA-direct)-Parasound Halo A23 125W/ch amplifier-Speltz anti-cables-DIY 2-ways + Dayton Titanic 10 subwoofer He's not hi-fi, he's my stereo. ezkcdude's Profile: http://forums.slimdevices.com/member.php?userid=2545 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
Patrick Dixon;130953 Wrote: Actually it is, it's just not very easy or practical! Well, OK, it is almost impossible then.. -- P Floding P Floding's Profile: http://forums.slimdevices.com/member.php?userid=2932 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
Patrick Dixon;130951 Wrote: This is completely wrong - Upsampling/oversampling doesn't invent any data! Interpolation is actually a filtering process which removes the repeat spectra that are created by the upsampling/oversampling process. The actual interpolation process doesn't improve the high frequency resolution at all - if anything the interpolation filter will remove some of the original high-frequency information. I agree with your point with regards to overampling. However, with upsampling, if you upsample a 44.1khz data sample to 96khz, how do you not invent new data? The only common data points you will have between these two sample rates will occur every 1 second...all intermediate data points will be completely different... -- PhilNYC Sonic Spirits Inc. http://www.sonicspirits.com PhilNYC's Profile: http://forums.slimdevices.com/member.php?userid=837 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
ezkcdude;130954 Wrote: They can be shifted to a higher frequency range, so that a more gentle reconstruction (anti-aliasing) filter can be used. That is my understanding.Your understanding is wrong I'm afraid. Once the alias is in the signal, it looks just like part of the original signal so if you remove it you'll remove part of the original signal too. I think you're probably confusing 'aliasing' with 'repeat spectra'. A good example of aliasing is that of waggon wheels in western movies appearing to speed up in a forward direction before slowing down and reversing as the waggon builds up speed. This is an example of temporal aliasing caused by the frame rate (temporal sampling frequency) of the film being too slow to capture the speed of movement. Once it's there, no amount of unsampling will remove it; the best you can do is to remove all the temporal high frequencies, in which case you'll end up with a very smeary picture :-( -- Patrick Dixon www.at-tunes.co.uk Patrick Dixon's Profile: http://forums.slimdevices.com/member.php?userid=90 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
PhilNYC;130961 Wrote: if you upsample a 44.1khz data sample to 96khz, how do you not invent new data?You're not inventing data because you're using the information that already exists within the digital signal to create the intermediate points. There's no more information in the signal - you're not creating anything, you're just filtering the signal. You might like to think of it in relation to what happens at the DAC; the digital signal is converted to an instantaneous analogue level at the sampled points, and then held (and filtered) to give a continuous analogue signal. But the signal between the precise sampled points (in the analogue domain) is not 'invented'. Maybe in concept it's even akin to FLAC compression; when you decompress from FLAC to WAV you get more data samples, but you're not 'inventing' data - the information is all in the original compressed FLAC file. -- Patrick Dixon www.at-tunes.co.uk Patrick Dixon's Profile: http://forums.slimdevices.com/member.php?userid=90 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
P Floding;130956 Wrote: Well, OK, it is almost impossible then.. :-) You'll find a simple example of an analogue FIR filter in old PAL TV sets (PAL is the analogue colour TV system in use in much of Europe (the French of course had to be different) - it's NTSC in N America Japan). The colour signal processing uses an analogue delay line (which used to be glass) in a simple two line vertical filter which reduces hue errors. Digital delays are much easier to make and use - so FIR filters are much more common (and can be much more complex) in the digital domain. -- Patrick Dixon www.at-tunes.co.uk Patrick Dixon's Profile: http://forums.slimdevices.com/member.php?userid=90 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
cliveb;130937 Wrote: In contrast, upsampling (as the term is generally used) involves INVENTING additional data (usually by interpolation) in the expectation that it will deliver improved high frequency resolution. But this extra data that's invented can't ever be known to be correct. Quite a lot of the time, it'll be wrong. It doesn't recreate the high frequencies that were discarded when the recording was sampled at whatever lower frequency was used Absolutely, I forget who said it but I think the following sums up well the use of asynchronous sample rate conversion (it was said in the context of its use for jitter reduction but the comment is more generally applicable): it shifts the problem from being the right data at the wrong time to being the wrong data at the right time ... Pages 18 onwards of the following present a nice discussion on the frequency domain errors introduced: http://www.analog.com/UploadedFiles/Data_Sheets/71654447AD1896_a.pdf Although, the audibility of such errors (or their side-effects) is of course another debate ... :-O -- reeve_mike reeve_mike's Profile: http://forums.slimdevices.com/member.php?userid=995 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
reeve_mike;130976 Wrote: Absolutely, I forget who said it but I think the following sums up well the use of asynchronous sample rate conversion (it was said in the context of its use for jitter reduction but the comment is more generally applicable): it shifts the problem from being the right data at the wrong time to being the wrong data at the right time ... Pages 18 onwards of the following present a nice discussion on the frequency domain errors introduced: http://www.analog.com/UploadedFiles/Data_Sheets/71654447AD1896_a.pdf Although, the audibility of such errors (or their side-effects) is of course another debate ... :-O Upsampling has nothing to do with asynchronous sample rate conversion. -- P Floding P Floding's Profile: http://forums.slimdevices.com/member.php?userid=2932 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
P Floding;130981 Wrote: Upsampling has nothing to do with asynchroneous sample rate conversion. Upsampling from 44.1K to 96K is asynchronous sample rate conversion by definition (the input output clocks are different) ... BTW most so called upsampling DACs (the boxes not the chips) use an asynchronous sample rate converter (like the AD1896) to do the upsampling ... [Only a very few companies, such as dcs Wadia, implement their own upsampler ...] -- reeve_mike reeve_mike's Profile: http://forums.slimdevices.com/member.php?userid=995 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
P Floding;130992 Wrote: However, there is no jitter introduced when doing a mathematical upsampling. Agreed! [And I never said that there was - apologies if my tangential reference to ASRC for jitter reduction, as in some add-on boxes marketed, caused confusion, I guess I should have left out the side remark in parenthesis ...] P Floding;130992 Wrote: IMHO any reasonable upsampling should be done in exact multiples of the original sample rate. Again, agreed! -- reeve_mike reeve_mike's Profile: http://forums.slimdevices.com/member.php?userid=995 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
Patrick Dixon;130969 Wrote: You're not inventing data because you're using the information that already exists within the digital signal to create the intermediate points. There's no more information in the signal - you're not creating anything, you're just filtering the signal. This is true for oversampling at full-integer multiples (eg 2x/4x/8x oversampling). But if you upsample from 44.1khz to 96khz, the interpolated data points only match up to the original sample every 96khz points (once per second). Upsampling from 44.1khz to 96khz means that you are creating 2.17687 data points for every original data point from the original 44.1khz sample...there is no way that all original 44,100 datapoints stay intact in that upsampling process. This is why, when you upsample, you need to extend the word length from 16-bit to 24-bit...because the ratio between 44.1khz/96khz will require greater precision to calculate all of those new datapoints. -- PhilNYC Sonic Spirits Inc. http://www.sonicspirits.com PhilNYC's Profile: http://forums.slimdevices.com/member.php?userid=837 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
You guys should read the data sheet for AD1896, which is Analog's ASRC. I'm using it right now for a DAC I am building. It explains very well the theory and implementation. AD1896 is used in practically all upsampling DACs these days. -- ezkcdude SB3-Derek Shek TDA1543/CS8412 NOS DAC-MIT Terminator 2 interconnects-Endler Audio 24-step Attenuators (RCA-direct)-Parasound Halo A23 125W/ch amplifier-Speltz anti-cables-DIY 2-ways + Dayton Titanic 10 subwoofer He's not hi-fi, he's my stereo. ezkcdude's Profile: http://forums.slimdevices.com/member.php?userid=2545 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
Patrick Dixon;130969 Wrote: You might like to think of it in relation to what happens at the DAC; the digital signal is converted to an instantaneous analogue level at the sampled points, and then held (and filtered) to give a continuous analogue signal. But the signal between the precise sampled points (in the analogue domain) is not 'invented'. If you upsample from 44.1khz to 96khz, there are now 2.17687x more data points than in the original sample, and only one of those data points per second is identical to a single data point in the original sample. Maybe in concept it's even akin to FLAC compression; when you decompress from FLAC to WAV you get more data samples, but you're not 'inventing' data - the information is all in the original compressed FLAC file. It's not the same at all. FLAC is decompressed at play-time to the exact original data set from the original sample. Upsampling increases the data set to 2.17687x the number of data points at a mathematical precision of 24-bits, and every one of those new data points is fed to the DAC for analog conversion. -- PhilNYC Sonic Spirits Inc. http://www.sonicspirits.com PhilNYC's Profile: http://forums.slimdevices.com/member.php?userid=837 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
PhilNYC;130999 Wrote: I think this is what is called oversampling Is it? I'm not sure, since I never really was a believer of upsampling. As I understood it oversampling is done as part of the reconstructions process in the DAC, wheras oversampling is a sort of lets do something to the data before we sent it over PSDIF type of thing..? -- P Floding P Floding's Profile: http://forums.slimdevices.com/member.php?userid=2932 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
P Floding;131008 Wrote: Is it? I'm not sure, since I never really was a believer in upsampling (I got turned off by all the hype). As I understood it oversampling is done as part of the reconstructions process in the DAC, wheras upsampling is a sort of lets do something to the data before we sent it over SPDIF type of thing..? Sort of. The definitions I've heard are that Oversampling is synchronous and Upsampling is asynchronous. If you read the Resolution Audio whitepaper I posted earlier in this thread, it also seems to take on those definitions (that oversampling is based on integer multiples of the original sample, whereas upsampling does not). Upsampling can happen after the SPDIF interface, and oversampling happens in a digital filter prior to the DAC chip (my Dodson DAC does both upsampling and oversampling after receving the 44.1/24 signal...upsamples to 96khz, then oversamples 8x, for a final sample speed of 768khz, which is then fed into the DAC chip.) -- PhilNYC Sonic Spirits Inc. http://www.sonicspirits.com PhilNYC's Profile: http://forums.slimdevices.com/member.php?userid=837 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
P Floding;131013 Wrote: Surely, asynchronous means the clocks aren't running in synchrony, which would not have anything to do with sample rate conversion per se? (I have read a fair bit about sample rate conversion.) I don't really see why a one-clock system should perform asynchronous operations? I'm going to make a guess here, so don't hold me to it...:-) ...but if you are changing the sample rate of the data, you will need two clocks, no? One clock runs at the original sample-rate (44.1khz) and the second runs at the new sample rate (96khz). This would perhaps not be true/necessary for oversampling at integer-multiples, but it would absolutely need to be true for upsampling at non-integer multiples, right? -- PhilNYC Sonic Spirits Inc. http://www.sonicspirits.com PhilNYC's Profile: http://forums.slimdevices.com/member.php?userid=837 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Re: Upsampling
PhilNYC wrote: ...but if you are changing the sample rate of the data, you will need two clocks, no? One clock runs at the original sample-rate (44.1khz) and the second runs at the new sample rate (96khz). or one that runs at 44.1*48 and select the proper signal samples off a common clock. There was a time when 44.1kHz was a challenge. By the time SACD came out, 2.82 MHz was not a challenge. At least if you are not trying to use a Tube clock :-) So having one clock is not a big deal. But changing from 44.1 to 48 or 96 is a really bad idea, IMHO. -- Pat http://www.pfarrell.com/music/slimserver/slimsoftware.html ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
Pat Farrell;131023 Wrote: PhilNYC wrote:[color=blue] or one that runs at 44.1*48 and select the proper signal samples off a common clock. There was a time when 44.1kHz was a challenge. By the time SACD came out, 2.82 MHz was not a challenge. At least if you are not trying to use a Tube clock :-) So having one clock is not a big deal. But changing from 44.1 to 48 or 96 is a really bad idea, IMHO. I agree that converting 44.1 to 48 is a bad idea. But at least going from 44.1 to 96 allows you to achieve some of the moving artifacts to a higher frequency so that filtering those artifacts out. But I agree that it appears to be much smarter to simply do a full-integer oversample. -- PhilNYC Sonic Spirits Inc. http://www.sonicspirits.com PhilNYC's Profile: http://forums.slimdevices.com/member.php?userid=837 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
PhilNYC;130998 Wrote: If you upsample from 44.1khz to 96khz, there are now 2.17687x more data points than in the original sample, and only one of those data points per second is identical to a single data point in the original sample.300 per second, surely: 147 periods of one stream are precisely 320 of the other. -- tom permutt tom permutt's Profile: http://forums.slimdevices.com/member.php?userid=1893 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
PhilNYC;130998 Wrote: If you upsample from 44.1khz to 96khz, there are now 2.17687x more data points than in the original sample, and only one of those data points per second is identical to a single data point in the original sample.It makes no difference, you are still not inventing data. I've spent the best part of 25yrs designing equipment that sample rate converts, filters and interpolates so I know a little about it! PhilNYC;130998 Wrote: It's not the same at all. FLAC is decompressed at play-time to the exact original data set from the original sample. Upsampling increases the data set to 2.17687x the number of data points at a mathematical precision of 24-bits, and every one of those new data points is fed to the DAC for analog conversion.I didn't say it was the same I said it was akin - in that the flac file doesn't contain all the original samples at the original precsion of the wav file, but it does contains all the information never-the-less. The sampled digital signal is a representation of what the signal was in the analogue domain, and the mathematics define the limits of the 'accuracy' of that representation. Interpolation, Upsampling and Oversampling don't invent any data, they simply represent the digital signal in another form. -- Patrick Dixon www.at-tunes.co.uk Patrick Dixon's Profile: http://forums.slimdevices.com/member.php?userid=90 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
Patrick Dixon;131047 Wrote: It makes no difference, you are still not inventing data. Within the limit that any filtering in D/D or D/A conversion is 'guessing' the original analog signal between two adjacent sample points - who is to say that it indeed was the smooth transition that the filter yields ... But are such errors audible (either directly or via side effects) ...? -- reeve_mike reeve_mike's Profile: http://forums.slimdevices.com/member.php?userid=995 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
I think there needs to be made a distinction between inventing data and creation of artifactual data. Clearly, the first one is not part of the design of upsamplers or oversamplers. Maybe marketers make it seem that way, but we know better, right? As for the second, it is inevitable that any interpolation technique of a real signal introduces artifacts. Only if we know the actual function, can we achieve perfect interpolation. With music, this is impossible. Therefore, there are artifacts. That's a fact. The presence of artifacts is not a good thing, but modern upsampling or oversampling architectures try to minimize their presence. As I said in an earlier post, read the data sheets for upsampling chips. AD1896 can interpolate with S/N ratio of up to 142 dB! The THD+noise is around -120 dB. -- ezkcdude SB3-Derek Shek TDA1543/CS8412 NOS DAC-MIT Terminator 2 interconnects-Endler Audio 24-step Attenuators (RCA-direct)-Parasound Halo A23 125W/ch amplifier-Speltz anti-cables-DIY 2-ways + Dayton Titanic 10 subwoofer He's not hi-fi, he's my stereo. ezkcdude's Profile: http://forums.slimdevices.com/member.php?userid=2545 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
Patrick Dixon;131047 Wrote: It makes no difference, you are still not inventing data. I've spent the best part of 25yrs designing equipment that sample rate converts, filters and interpolates so I know a little about it! Then can you explain it to me? :-) How do you go from 44,100 data points to 96,000 data points and not create data that did not exist before? I didn't say it was the same I said it was akin - in that the flac file doesn't contain all the original samples at the original precsion of the wav file, but it does contains all the information never-the-less. The sampled digital signal is a representation of what the signal was in the analogue domain, and the mathematics define the limits of the 'accuracy' of that representation. Interpolation, Upsampling and Oversampling don't invent any data, they simply represent the digital signal in another form. I understand that. However, if a process is using mathematics to guess at what that digital signal looks like, then it is still making up data that was not in the original sample. The reason why FLAC is not a good analogy IMHO is because FLAC starts with original data, and then contains header/additional information used to decode the compressed file to restore the original data. In the case of upsampling, there is no embedded information in the original 44.1khz data that can be used to make the upsampling a mere decoding to the original waveform. Companies like dCS have upsampling components that let you select from a variety of upsampling calculations...essentially letting you try out 5-6 of their best guesses and seeing which one you like. btw - here's an interesting whitepaper by dCS on why high sample rates sound better than low sample rates http://www.dcsltd.co.uk/technical_papers/aes97ny.pdf -- PhilNYC Sonic Spirits Inc. http://www.sonicspirits.com PhilNYC's Profile: http://forums.slimdevices.com/member.php?userid=837 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
tom permutt;131046 Wrote: 300 per second, surely: 147 periods of one stream are precisely 320 of the other. Still...out of 96000 data points, that's not a lot... -- PhilNYC Sonic Spirits Inc. http://www.sonicspirits.com PhilNYC's Profile: http://forums.slimdevices.com/member.php?userid=837 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
PhilNYC;131061 Wrote: Then can you explain it to me? :-) How do you go from 44,100 data points to 96,000 data points and not create data that did not exist before?OK, so how do you go from 44,100 data points to an infinite number - which is what you do when you D to A Convert - and not 'create' data? You are not 'creating' data you are just representing the data that's already there in another way. It's like converting a dollar bill into 100 pennies (is that what you call them?) and saying you're creating wealth. -- Patrick Dixon www.at-tunes.co.uk Patrick Dixon's Profile: http://forums.slimdevices.com/member.php?userid=90 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
reeve_mike;131056 Wrote: Within the limit that any filtering in D/D or D/A conversion is 'guessing' the original analog signal between two adjacent sample points - who is to say that it indeed was the smooth transition that the filter yields ... You're missing the point, the characteristics of the signal are determined when you sample the analogue signal: if the original signal is correctly bandlimited prior to A to D Conversion (or the sampling frequency is sufficently in excess of twice the bandwidth), then you know where the 'intermediate' points should be. -- Patrick Dixon www.at-tunes.co.uk Patrick Dixon's Profile: http://forums.slimdevices.com/member.php?userid=90 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
Patrick Dixon;131085 Wrote: OK, so how do you go from 44,100 data points to an infinite number - which is what you do when you D to A Convert - and not 'create' data? That's very different. In the case of D-to-A conversion, you are essentially decoding something that was encoded using the same process in the first place. And I'll also argue that an analog waveform is not data... You are not 'creating' data you are just representing the data that's already there in another way. I can understand what you're trying to say, but I disagree with it. It's still just an estimate of the original wave, with data points guestimated based on the original sample. It's like converting a dollar bill into 100 pennies (is that what you call them?) and saying you're creating wealth. Well, no...that wound be 100x synchronous oversampling. ;-) -- PhilNYC Sonic Spirits Inc. http://www.sonicspirits.com PhilNYC's Profile: http://forums.slimdevices.com/member.php?userid=837 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
Pat Farrell;131091 Wrote: Patrick Dixon wrote:[color=blue]Right. Mike doesn't understand (or appears to not understand) the work of Shannon and Nyquist. All of the digital sampling work is based on their theories. Nyquist showed that sampling at twice the bandwidth allows reconstruction. That is why the RedBook spec uses 44.1 kHz. For decades, the hfi world used a bandwidth of 20 hz to 20kHz as the limits of human hearing. Sampling at 44.1kHz allows a little over. My understanding of the benefit of oversampling/upsampling is primarily to get the digital artifacts resulting from imprecisions in the DAC process to a higher frequency so that they can be more easily filtered in a frequency range that won't impact the audible range. -- PhilNYC Sonic Spirits Inc. http://www.sonicspirits.com PhilNYC's Profile: http://forums.slimdevices.com/member.php?userid=837 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Re: Upsampling
PhilNYC wrote: Pat Farrell;131091 Wrote: Nyquist showed that sampling at twice the bandwidth allows reconstruction. That is why the RedBook spec uses 44.1 kHz. For decades, the hfi world used a bandwidth of 20 hz to 20kHz as the limits of human hearing. Sampling at 44.1kHz allows a little over. My understanding of the benefit of oversampling/upsampling is primarily to get the digital artifacts resulting from imprecisions in the DAC process to a higher frequency so that they can be more easily filtered in a frequency range that won't impact the audible range. I'm not an audio design engineer, so I could be wrong. But I understand it exactly the opposite of this. Digital processing has to have analog filters to cut out unwanted signals and noise. If you use a 44.1kHz sample, you need a radical filter to cut off signals about 20kHz. The standard implementation uses 12dB/octave or even 18dB / octave filters. These do evil things to phase. So if you over/re/up-sample at 96kHz or 192kHz, you can use digital filters (IIR, etc.) for the worst parts, and then use gentle single order analog filters down in the 20-20kHz range. One technical problem with the SACD spec was that it used noise shaping to move the inevitable noise into relatively low frequencies (50kHz, and up). which had potentential to have audible interactions. Someone smarter than me can probably shed some light. -- Pat http://www.pfarrell.com/music/slimserver/slimsoftware.html ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
Sorry if it disappoints but I know well the work of Claude Shannon and Harry Nyquist ... What I was trying to contribute was that the samples on the CD do not faithfully represent the musical waveform, they only represent a version of it filtered at 20.5KHz, which seemed to be relevant at the time but now I can't remember why ... As an aside, just because 20Hz-20KHz has been used for years doesn't make it 'right', there is increasing psycho-acoustic experimental data that suggests that even though pure tones above approx. 20KHz cannot be heard directly their presence in music signals can be 'detected' in some way - as pointed out by Pat ... Arbitrary technical limits are often post-rationalizations of the technology limits of the time (e.g. 44.1/16), and one might point to the historic limits of valve-amp output transformers, speaker drive unit technology etc. (and the immature state of psycho-acoustics) for leading us to 20Hz-20KHz ... -- reeve_mike reeve_mike's Profile: http://forums.slimdevices.com/member.php?userid=995 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
Chapter 3 of Analog Device's Data Conversion Handbook (by Walt Kester) discusses this: Kester Wrote: The basic concept of an oversampling/interpolating DAC is shown in Figure 3.30. The Nbits of input data are received at a rate of fs. The digital interpolation filter is clocked at an oversampling frequency of Kfs, and inserts the extra data points. The effects on the output frequency spectrum are shown in Figure 3.30. In the Nyquist case (A), the requirements on the analog anti-imaging filter can be quite severe. By oversampling and interpolating, the requirements on the filter are greatly relaxed as shown in (B). Also, since the quantization noise is spread over a wider region with respect to the original signal bandwidth, an improvement in the signal-to-noise ratio is also achieved. By doubling the original sampling rate (K = 2), an improvement of 3 dB is obtained, and by making K = 4, an improvement of 6 dB is obtained. Early CD players took advantage of this, and generally carried the arithmetic in the digital filter to more than N-bits. Today, most DACs in CD players are sigma-delta types. The take-home message is that you want to separate the images or artifacts from the Nyquist-limited bandwidth. That way, you're anti-imaging filter doesn't mess with the real data (as much). I have a hardcopy version of this book, but it is also available online (for free) as a series of lecture notes in PDF format. Very good reading if you're interested in how modern (and ancient) DACs really work. -- ezkcdude SB3-Derek Shek TDA1543/CS8412 NOS DAC-MIT Terminator 2 interconnects-Endler Audio 24-step Attenuators (RCA-direct)-Parasound Halo A23 125W/ch amplifier-Speltz anti-cables-DIY 2-ways + Dayton Titanic 10 subwoofer He's not hi-fi, he's my stereo. ezkcdude's Profile: http://forums.slimdevices.com/member.php?userid=2545 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Re: Upsampling
reeve_mike wrote: Sorry if it disappoints but I know well the work of Claude Shannon and Harry Nyquist ... Opps, sorry. What I was trying to contribute was that the samples on the CD do not faithfully represent the musical waveform, they only represent a version of it filtered at 20.5KHz, which seemed to be relevant at the time but now I can't remember why ... I would not expect it to be 20.5kHz. The sample rate is 44.1, so in theory, you could have signals as high as 22.05kHz. Realistically, there is nearly nothing above 20kHz to start. Not only is 20-20kHz the standard spec but most microphones have serious roll off above 17kHz or so. And all of the preamps used for the microphones, especially the 'vintage' ones that people swear sound best. Neve, SSL, etc. And analog signals don't stop at clean numbers like 20.5, they just roll off at X dB per octave. As an aside, just because 20Hz-20KHz has been used for years doesn't make it 'right', there is increasing psycho-acoustic experimental data that suggests that even though pure tones above approx. 20KHz cannot be heard directly their presence in music signals can be 'detected' in some way - as pointed out by Pat ... Right is an interesting concept here. I believe that the idea of a brick wall fall off at 20kHz is dumb, I believe that there are harmonics and interactions. I don't know when the 20-20kHz idea became popular, but by the early post-War days, when Hi-Fi was invented by signal corpmen going to engineering school on the GI-bill (same guys who made ham radio be real), it was established. It is next to impossible to get a brick wall filter using coils and capacitors. I think 24dB/octave is about it, but there might be more. So you would expect that a good filter can only cut the signal by 24 dB going from 20kHz, up an octave to 40kHz. But the mics, preamps, Neumann cutting lathes, and all the anti-feedback controls all combine to each throw away a couple more dB per octave. There just isn't much up there. Arbitrary technical limits are often post-rationalizations of the technology limits of the time (e.g. 44.1/16), and one might point to the historic limits of valve-amp output transformers, speaker drive unit technology etc. (and the immature state of psycho-acoustics) for leading us to 20Hz-20KHz ... All the early HiFi stuff was tubes and transformers. Getting even plus or minus 3dB 20-20kHz was hard and expensive. And no speakers before the early 80s tried for 20-20kHz. One of the better speakers of the era was the Quad 63, which has a writeup in Stereophile on their website http://stereophile.com/floorloudspeakers/416/index11.html The scale is not detailed enough to quote where the -3dB points are, but my eyeballs estimate it as 50-15kHz. Things like Bozak systems went lower, and getting into the 30 hz zone was not all that hard (but took a lot of power/space). All in all, I think that 44.1/16 was a good engineering choice at the time. The key was that CDs were designed to replace casettes, and be better quality, longer living, and harder to replicate. That is why the labels wanted them. They did replace cassettes. And in all but a few cases, replaced vinyl. The current Stereophile (ro maybe it was TAS) has a quote from Boothroyd/Stewart big wig who said that if they had chosen 20 bit and 50kHz (or maybe 55kHz) that we would have enough to have perfect sound forever. :-) -- Pat Farrell PRC recording studio http://www.pfarrell.com/PRC ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
WOWnow thats what I call a response. Thanks for all the info. -- Walleyefisher Walleyefisher's Profile: http://forums.slimdevices.com/member.php?userid=7122 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
Pat Farrell;131153 Wrote: reeve_mike wrote: What I was trying to contribute was that the samples on the CD do not faithfully represent the musical waveform, they only represent a version of it filtered at 20.5KHz, which seemed to be relevant at the time but now I can't remember why ... I would not expect it to be 20.5kHz. The sample rate is 44.1, so in theory, you could have signals as high as 22.05kHz. Surely I can be forgiven a typo ... :-) Pat Farrell;131153 Wrote: Realistically, there is nearly nothing above 20kHz to start. Not only is 20-20kHz the standard spec but most microphones have serious roll off above 17kHz or so. And all of the preamps used for the microphones, especially the 'vintage' ones that people swear sound best. Neve, SSL, etc. And analog signals don't stop at clean numbers like 20.5, they just roll off at X dB per octave. Agreed in general, but I've used some nice vintage Neumann mics that go way up high ... The fact that they don't stop dead but roll off was an implicit part of my point, the ultrasonics are there, even if they are increasingly down, and we don't yet fully understand their effects - I think that we are in serious agreement here! Pat Farrell;131153 Wrote: One of the better speakers of the era was the Quad 63 As an aside, I would claim that it is still a good speaker in this era ... [BTW I am biased because the pair that I bought back in 1984 still serve me well in a secondary system ... :-)] Pat Farrell;131153 Wrote: All in all, I think that 44.1/16 was a good engineering choice at the time. I think that it was a good (and at the time the only available) product engineering choice but I don't think that it was a good sound/music engineering choice ... :-O Mike -- reeve_mike reeve_mike's Profile: http://forums.slimdevices.com/member.php?userid=995 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
I think all of the above hulabaloo builds a strong case for those of us on the fringe with non-oversampling filterless DACs. The second benefit is there is no dog poop in the yard because all the neighborhood dogs can't handle the super-high frequency noise. :) non-os dacs, keeping it real. -- dwc dwc's Profile: http://forums.slimdevices.com/member.php?userid=1892 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Re: Upsampling
reeve_mike wrote: Agreed in general, but I've used some nice vintage Neumann mics that go way up high ... This is way off topic, but which ones? And how vintage? Most of the classic Neumann's like the U87 or M50 fall off pretty seriously. Now my KM184's go up high, but they aren't vintage. Pat Farrell;131153 Wrote: One of the better speakers of the era was the Quad 63 As an aside, I would claim that it is still a good speaker in this era No argument from me. I just don't have a room suitable for a pair. Pat Farrell;131153 Wrote: All in all, I think that 44.1/16 was a good engineering choice at the time. I think that it was a good (and at the time the only available) product engineering choice but I don't think that it was a good sound/music engineering choice ... :-O Sound? Who cares about sound? Labels care about money. Warner Bros just closed their Classical label: http://www.stereophile.com/news/082106classical/ Soon it will be all Britney and hip-hop. -- Pat Farrell PRC recording studio http://www.pfarrell.com/PRC ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Upsampling
I don't understand the premise of your question. Take a signal that looks like 0, 2, 3 then upsample it to 0,0,0,0, 2,2,2,2, 3,3,3,3 at four times the rate. How does this allow the DAC to do anything differently? No, it interpolates. So you get something maybe like: 0, 0.5, 1, 1.5, 2 and so on. What you're talking about is oversampling - just another name for upsampling, but usually used in reference to what modern DACs do internally. It is fundamental to how they work and yes, the smaller steps require less filtering (and yield better linearity, lower noise etc). The DAC in transporter oversamples by 128x, so a 44.1 signal is actually converted to analogue at a sample rate of 5.6 MHz... a high resolution indeed. Now, what's stupid is taking 44.1 CD rips, resampling them to 96KHz and then re-saving to disk, thinking you've given it more breathing room or opened up the high end or whatever. It's total nonsense, exactly like on CSI where they zoom in on a single pixel, click ENHANCE and then read a license plate from a mile away. It don't work that way. -- seanadams seanadams's Profile: http://forums.slimdevices.com/member.php?userid=3 View this thread: http://forums.slimdevices.com/showthread.php?t=26685 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Re: Upsampling
seanadams wrote: then upsample it to 0,0,0,0, 2,2,2,2, 3,3,3,3 at four times the rate. No, it interpolates. So you get something maybe like: 0, 0.5, 1, 1.5, 2 and so on. Thanks for the clarification. What you're talking about is oversampling - just another name for upsampling, but usually used in reference to what modern DACs do internally. It is fundamental to how they work and yes, the smaller steps require less filtering (and yield better linearity, lower noise etc). The DAC in transporter oversamples by 128x, so a 44.1 signal is actually converted to analogue at a sample rate of 5.6 MHz... a high resolution indeed. So it is more than twice as good as the SACD single bit rate of 2.82 MHz, eh? Any chance that the DAC in the Transport actually is 5.64 mHz? Now, what's stupid is taking 44.1 CD rips, resampling them to 96KHz and then re-saving to disk, thinking you've given it more breathing room or opened up the high end or whatever. It's total nonsense, exactly like on CSI where they zoom in on a single pixel, click ENHANCE and then read a license plate from a mile away. It don't work that way. Next you are going to start claiming that little wooden feet that hold your cables off the floor don't improve the bloom and remove a veil. More seriously, I don't understand why anyone thinks 96kHz is a good thing to do to RedBook. For ADAT sources, sure. But taking it to a non-integer multiple makes no sense. If nothing else, it will screw up the dither. -- Pat http://www.pfarrell.com/music/slimserver/slimsoftware.html ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
