bluegaspode;694580 Wrote: > > I obviously never cared about how DA converters work, based on the > paper I now think of thousands of of sinc-producing circuits which all > add up to the final waveform. > Probably this is not how it works in practice but this a missing link > for me now to agree that Nyquist theorem and good circuits is all that > we need.
We have to do away with a huge misunderstanding here, I believe. This article is NOT about how a DAC should work, which technology is best and what kind of limitations practical recording and playing equipment may have. All of this has nothing, really NOTHING to do with the topic at hand. It just doesn't matter how good or bad the DAC or the ADC or the speaker or the microphone is. All of this has absolutely NOTHING to do with the storage format for the music. All that Nyquist/Shannon says is: if you have a frequency X, which is the maximum frequency you are interested in (here: the highest frequency you could probably ever hear), then if you use a sampling frequency of 2*X to store your sampled data, then ALL information contained in the signals below X will be in included in the information you store. There is NO additional information you get by using a higher sampling frequency. Nothing. All you get is additional information about frequencies ABOVE X but not below, the information on the frequencies below X is already there and it's complete. What we have to understand here, is that this has nothing to do with any imperfections in the recording or playback process, these will of course be there, but if your recording is crap, i won't get better just because you STORE more of it and at a higher sample rate. And if your DAC is distorting, then it will not get any better just because you throw higher frequencies at it - to the contrary, the article implies it's getting actually WORSE because of effects letting distortions from the higher frequencies leak into the lower frequency spectrum. To be clear: this is NOT "missing information" that was just not recorded due to the low sample rate, it's DISTORTED information due to the bad reproduction process in the DAC. What the article does NOT say is that it doesn't make sense to use different sample rates or sample sizes for processing. It can make sense to use something different while processing your data, for example because of limitations of the technology you use. A good example is the 24 bit sample size used in the Squeezebox internally. This makes perfect sense because what the Squeezebox does is it does digital processing to change the volume. If you stick to 16 bit data, you would get rounding errors and information losses due to this processing that you can avoid if you go to 24 bit in processing. But it does NOT mean, that anything gets better if the data you throw at it is already 24 bit. To use a somewhat different analogy: When a bank calculates interest, it will use 4 additional digits behind the cent (so 1$ is 1.00 0000 for them). Why? Because if you get, for example, 1% of interest for your dollar per year and that interest is paid monthly, the monthly interest you get would amount to 0.00 0833 ct. If you round that, you just get 0 so you would never get any interest which would be plain wrong because to calculate things right, they will have to pay you 1 ct per year. HOWEVER, they will never actually PAY you 0.0833 ct because there is no such thing. and your Dollar doesn't get any different just because you write it as 1.00 0000 $, it's still exactly the same thing as 1$ and actually everything behind the last cent digit has no meaning at all (or it would already be a rounding error). Likewise, nobody says that there will be no way to invent some fancy technology that does a more accurate recording of the analog audio signal at 2 MHz sample rate and this can be superior to a 16 bit 44.1 kHz microphone. HOWEVER: If you then take the digital output of that hypothetical processor and down-convert it to 44.1 kHz samplerate audio, then for all frequencies below 22.05 kHz, there will be NO, not even the slightest loss of information. So it doesn't make sense to STORE and TRANSMIT the data at higher frequencies. There are a few good arguments for 48 kHz, most notably that since it's common to use 96kHz or 192 kHz equipment in processing (remember: it can STILL make a lot of sense to do PROCESSING at higher frequencies, especially in the digital domain) you get pretty much simplified up/downsampling logic. I can't argue about 24 vs. 16 bit, that does indeed depend on the actual dynamic range you can record and reproduce and I don't know where technology is here, purely from an information theory standpoint, 24 bit word size DOES contain more information than 16 bit word size, that's different from the sample rate thing. Now there is a third thing, and that's the "trust your ears" thing. 1. Yes, you should, because in the end it's all that matters 2. Normal people do that but as of my experience, audiophile's don't. They just trust the money or some other rationale, otherwise they would not be so opposed to double-blind tests. The problem with "trust your ears" is also two-fold: 1. It has nothing, really nothing, to do with all we've discussed above. 2. It can lead to unexpected results. What you PERCEIVE as superior sound does not have to be the sound that has the higher similarity to the original signal. All of what we discussed about above only spoke about how similar the signal you are reproducing is to the original sound as it was mastered. It says nothing about how GOOD that actually sounds. One extreme example: if you compare 128kbps mp3 compressed audio (lame codec at high quality) to the original file and you do that across a large sample of songs and on good equipment, your chances are somewhat high that you are able to discriminate between the two signals. What that means is: when you do an ABX test and X is either A or B and you don't know what either is you have a pretty high chance to say whether X is actually A or B. Now if you ask an ENTIRELY DIFFERENT question: you ask whether A or B sounds BETTER, you have about a 50% chance to find mp3 to sound better than the original file. If you have good hearing, your chances to prefer mp3 are actually a bit HIGHER (because most hearing defects destroy the psychoacoustic masking mp3 uses to one extent or the other. The same can of course hold true of a 192kHz sample rate with reproduction artifacts coming from ultrasonic frequencies just that if you KNOW what you are listening to you will say "yes, that mp3 degraded the quality" while with the HD stuff you say "oh yes, that additional samplerate added more detail". Chances are very high that if you hear more "detail" due to inaudible processing difference it's because that "additional detail" just wasn't there in the original recording and made up by your equipment. To sum up: 1. It CAN make a lot of sense to use whatever technology gives you the bet result in creating a digital representation of an analog signal, if this involves high frequencies, so be it, but you always have to be aware that the opposite can be true as well. 2. For STORAGE and TRANSPORTATION of data, it's perfectly fine to downconvert this to for example 48kHz or 44.1 hKz samplerate, since you di this in the digital domain you don't even have losses due to bad equipment, inaccuracies or whatever, you will lose no, no theoretical and no practical information about the audible frequency range. 3. When you are reproducing (playing) the audio then, again, what is the best technology available to you can depend on a lot of technical details. Evidence from the article above seems to indicate that using ultrasonic frequencies in your samples makes things worse, not better, but in the end all of this will be up to the engineer developing the DA conversion system. But again: for all of this it makes NO difference whether the material you throw at it is 48 or 96 kHz, except maybe for some practicality reasons (not having to transcode data at some source) but that's pure handling and has no impact on quality, if it does, the DAC designer has made a really bad job. -- pippin --- see iPeng, the Squeezebox iPhone remote and *New: iPeng for iPad*, at penguinlovesmusic.com ------------------------------------------------------------------------ pippin's Profile: http://forums.slimdevices.com/member.php?userid=13777 View this thread: http://forums.slimdevices.com/showthread.php?t=93990 _______________________________________________ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
