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Help understanding audio sampling

Started by Ritual April 14, 2007
"Jerry Avins" <jya@ieee.org> wrote in message

> David Morgan (MAMS) wrote:
> > "Jerry Avins" <jya@ieee.org> wrote in message
> >> David Morgan (MAMS) wrote:
> >>> "Jerry Avins" <jya@ieee.org> wrote in message...
> >>>> David Morgan (MAMS) wrote:
> >>>>> "Jerry Avins" <jya@ieee.org> wrote in message...
> >>>>>> You can reach the limits of audible frequencies at sampling rates far > >>>>>> below 96 KHz. You can exceed the accuracy of tape or vinyl with numbers > >>>>>> smaller that 32 bits.
> >> I did not! Please quote accurately.
> > You did not what? Are you saying that you did not say that? > > > > Please see your message: news:I8-dnZFSkOzv17_bnZ2dnUVZ_hSdnZ2d@rcn.net...
> Sorry! My mistake.
Lends me to believe that you're knee-jerking at me for some reason, Jerry. ;-)
> >>> I wasn't including "processing" math; which I for one, had much rather keep > >>> in the analogue domain.
> >> What are you babbling about?
> > Babbling? Contain yourself dear boy, at look at the REAL recording industry,
> Claiming that mixing and setting levels -- processing to be sure -- > should be by analog means
Babbling, my dear Jerry, is thinking you can read my mind, and thay you have all of the correct answers before any questions are asked... THAT is "babbling." But while we're at it, an analogue mixer will not impart summing problems and CPU issues that one will run into when mixing "in the box". Unfortunately, most people simply don't have the experience to understand that and thus can't hear when they are damaging their material.
> ...even in digital recordings is babbling.
"DIGITAL" is nothing more than another storage medium. Decent software and decent interfacing hardware allows *complete* interfacing to the real- world of analogue for any and all processes. I prefer that world of quality hardware far more than using a handfull of algorithms to mathematically attempt to model said analogue world.
> If you didn't mean that, what did you try to write?
You definitely ARE haveing a knee-jerking afternoon, Jerry. I never "CLAIMED" a damn thing, nor did I ever suggest that anything "SHOULD" be done a specific way. I SAID, "I for one, had much rather keep in the analogue domain." It's right above, in plain sight; read it again. My presonal preference should be no call for a knee-jerking assault of 'incorrect' on your part.
> > not the million homes full of dunces with computers who THINK they actually > > comprehend how real music is recorded and mixed. If I had a dollar for every > > dim-witted, inexperienced, shill who purchased a PC and some software and > > printed a business card claiming to be a recording engineer / producer, etc., > > I'd be happy to buy you and me a *real* studio using all of the "digital tracks" > > you would ever care to store. ;-)
> You seem to have a bone to pick with audio types.
Excuse me? I've spent the last 33 years of my life as an "audio type". If I have a bone to pick, it is with snake-oil salesmen and the continually dropping 'bar' of quality audio recording and reproduction. Of course, DSP freaks who think there is only one way to access, process, and otherwise manipulate a digital audio file don't make life any easier. Digital is a storage medium... and if you treat it as such, a whole new world of potential quality will open up for you.
> DSP involves a lot > more than that, with the same principles applying to all of it. There's > a common thread that you're too eager to particularize.
The common thread between us, thus far, is that you apparently lack any real-world or analogue audio experience. Don't foget... all raw audio (moving air), no matter how captured or how stored, is in fact ANALOGUE in it's creation and in it's perception.
> >> How do you mix and adjust gain with digital tracks?
> > You use a REAL analogue mixing desk. It requires a little investment of > > money for interfacing and a certain level of experience and intelligence > > of practical application. If you haven't seen one lately, check my web site.
> Gee! You could change sample rates that way, too. I wonder why > decimation and interpolation are so widely used.
Yes, you could... but I don't necessarily recommend it -- any more than I would recommend recording using unecessarily high sample rates under the misunderstood guise that it somehow reproduces with more accuracy to the human ear (mathematics aside for a moment). Above 48K and you are basically wasting your time, drive space, accuracy in returning to the standard delivery medium, and recording the unrecordable for the purpose of addressing the hearing of dogs.
> >>>> I try to avoid numbers in threads like this because they often lead to > >>>> unproductive digressions.
> >>> I can see where that would become an issue around here. > >>> > >>> ;-)
> >> It's an issue in any thread started by a request for basic concepts.
> > Basic concepts begin with capturing and processing real audio, not > > mathematics. Sampling rates are the biggest lot of hype and baloney > > to hit the consumer audio marketplace in a long, long time. They're > > called "marketing bits" by anyone with real-world experience in audio > > recording.
> Try to sample 20 KHz music with an 8 KHz sampler, then tell me what > baloney is.
You just completely changed our discussion by entering into a realm that is not only assanine, but plain damned stupid, and a million miles away from the subject of using abnormally high and unecessary sampling rates. Moving to an analogy of comparing to lower sampling rates makes about as much sense as the bozo who relates digital imaging to digital audio. Have a nice day Jerry. -- David Morgan (MAMS) http://www.m-a-m-s DOT com Morgan Audio Media Service Dallas, Texas (214) 662-9901 _______________________________________ http://www.artisan-recordingstudio.com
"David Morgan \(MAMS\)" <findme@m-a-m-s.comC/Odm> writes:
> [...] > But while we're at it, an analogue mixer will not impart summing problems > and CPU issues that one will run into when mixing "in the box".
What summing problems are you referring to? What "CPU issues" are you referring to? I've written a few digital summers in my day and I don't know what you're talking about. In both the analog and digital worlds, summing requires a reduction in dynamic range/SNR, just in case this is the "problem" you are referring to. -- % Randy Yates % "Watching all the days go by... %% Fuquay-Varina, NC % Who are you and who am I?" %%% 919-577-9882 % 'Mission (A World Record)', %%%% <yates@ieee.org> % *A New World Record*, ELO http://home.earthlink.net/~yatescr
"Randy Yates" <yates@ieee.org> wrote in message news:m3zm573fr9.fsf@ieee.org...

> "David Morgan \(MAMS\)" <findme@m-a-m-s.comC/Odm> writes:
> > But while we're at it, an analogue mixer will not impart summing problems > > and CPU issues that one will run into when mixing "in the box".
> What summing problems are you referring to? What "CPU issues" are you > referring to?
You are a person with some audio experience... I don't need to explain this you. I'm wondering why you asked me to come here and then would argue this point.
> I've written a few digital summers in my day and I don't know what > you're talking about.
I think you do. As a matter of fact, I think you have to take into account this fact when doing your designing of any digital audio summing buss.
> In both the analog and digital worlds, summing requires a reduction in > dynamic range/SNR, just in case this is the "problem" you are referring > to.
Randy! Are you really saying this? I think you're misinterpreting dynamic range, which doesn't necessarily ever need to be altered at all in the process of summing multi-channel audio for mixing. Signal to noise also has nothing to do with summing in the analogue world. If it's noisy, you mix with the noise or find a method of reducing it... it has nothing to do with anologue summing either. Perhaps these are "problems" you face in mixing 'in the box' summing (in the world of algorithms and math), but neither play into the analogue mix buss to the best of my knowledge. If you're talking about a (additive) reduction of RMS power as channels are added to the mix buss, this is not "dynamic range" in the literal sense of the word... you're getting into headroom issues, not dynamic range. -- David Morgan (MAMS) http://www.m-a-m-s DOT com Morgan Audio Media Service Dallas, Texas (214) 662-9901 _______________________________________ http://www.artisan-recordingstudio.com
"David Morgan \(MAMS\)" <findme@m-a-m-s.comC/Odm> writes:
> [...]
David, I'm finding that your attitude is offensive, and you're not really bringing any real facts to the discussion. When these things change, we might be able to have a meaningful discussion. -- % Randy Yates % "...the answer lies within your soul %% Fuquay-Varina, NC % 'cause no one knows which side %%% 919-577-9882 % the coin will fall." %%%% <yates@ieee.org> % 'Big Wheels', *Out of the Blue*, ELO http://home.earthlink.net/~yatescr
"Randy Yates" <yates@ieee.org> wrote in message news:m3irbv8zce.fsf@ieee.org...

> I'm finding that your attitude is offensive, and you're not really > bringing any real facts to the discussion. When these things change, > we might be able to have a meaningful discussion.
Thanks for your answer.
"Randy Yates" <yates@ieee.org> wrote in message news:m3irbv8zce.fsf@ieee.org...
> "David Morgan \(MAMS\)" <findme@m-a-m-s.comC/Odm> writes: > > [...] > > David, > > I'm finding that your attitude is offensive, and you're not really > bringing any real facts to the discussion. When these things change, > we might be able to have a meaningful discussion.
I honestly can't believe that you would say what you said while accusing me of telling lies, and then fail to justify your position because *I* have an attitude problem. My problem is two-fold... I'm wasting my time trying to enter into threads with some valid input while I'm here looking for a bit of assistanceand I'm in a hurry; and I have a problem with someone who tells me that I'm telling lies in the meantime. Thus far, aside from a few technical threads which admittedly barely or do not comprehend, this place is -generally speaking- another mass of ego and misdirection like rec.audio.opinion. Sorry brother, but signal to noise and dynamic range are *poor* excuses for accusing me of lying in that CPU and summing buss math issues are prevalent in computer based recording. If you've never seen a CPU choke up with anything from a high track count (including caused by silly high sampling rates) to multiple plug-ins... or, you have never seen the additive effects of track layering on a digital summing buss, you are truly one of a kind. DM
David Morgan (MAMS) wrote:
..
> I honestly can't believe that you would say what you said while accusing > me of telling lies, and then fail to justify your position because *I* have an > attitude problem. My problem is two-fold... I'm wasting my time trying to > enter into threads with some valid input while I'm here looking for a bit of > assistanceand I'm in a hurry; and I have a problem with someone who > tells me that I'm telling lies in the meantime. Thus far, aside from a few > technical threads which admittedly barely or do not comprehend, this > place is -generally speaking- another mass of ego and misdirection like > rec.audio.opinion. > > Sorry brother, but signal to noise and dynamic range are *poor* excuses > for accusing me of lying in that CPU and summing buss math issues > are prevalent in computer based recording. If you've never seen a CPU > choke up with anything from a high track count (including caused by > silly high sampling rates) to multiple plug-ins... or, you have never seen > the additive effects of track layering on a digital summing buss, you are > truly one of a kind. >
It seems to me there is some confusion here over exactly what is being compared with what. Are you comparing an analogue mixing desk (OK, understood to be a top-of-the-range model with zero per-channel noise) with a digital mixing desk, or an analogue desk with a computer-based DAW? It is of course possible to exceed CPU capacity on the latter because it is an open system, whereas a digital mixer is a closed system, designed to manage a fixed set of resources. You can't freely duplicate resources in or extend an analogue desk, it always requires more hardware. Also, I feel bound to point out that skill in audio mixing, and resources with which to mix, are almost orthogonal. A bad engineer is as likely produce bad results with a good analogue desk as with a good digital one; whereas a skilled engineer will likely get good results even out of an "average" digital desk. And nobody accused you of lying, simply of wrapping up opinions as asertions. You wrote, for example: " "DIGITAL" is nothing more than another storage medium." That is not a "fact", but an opinion - almost a political position. It's a good undergraduate essay title with the instruction "discuss". Digital is also a transmission medium. In the UK we will soon have no analogue TV broadcasts at all. We now have HD screens, which would be prohibitive cost-wise with analogue technology (HD CRT anyone?). Digital is not just about audio! So if you will not qualify your assertions, you cannot be surprised if others seek to do so, in what is supposed to be an "informative" thread. Most people will agree (snake-oil notwithstanding) that the 96KHz srate is, in itself, OTT (at least for a delivery medium). Whereas they will also agree that 48Khz is barely adequate, especially if you want genuinely linear behaviour right up to 24KHz. We need srate headroom just as we need dynamic headroom. Many dsp processes gain great advantage by oversampling internally. Technically, it is simplest just to double things, rather than argue about the virtues of 54KHz v 60Khz, etc. It is practical for hardware design - divide down a single clock, etc. There is a difference between just ~using~ 96KHz because it is there, and evangelising about it. Nobody on this list, probably, has any interest in evengelising; people just use what is there that gets the job done. People asking basic informative questions on this list must despair, if responses so easily digress into heated assertion v assertion, that have almost nothing to do with the original question! Richard Dobson
"David Morgan \(MAMS\)" <findme@m-a-m-s.comC/Odm> writes:

> "Randy Yates" <yates@ieee.org> wrote in message news:m3irbv8zce.fsf@ieee.org... >> "David Morgan \(MAMS\)" <findme@m-a-m-s.comC/Odm> writes: >> > [...] >> >> David, >> >> I'm finding that your attitude is offensive, and you're not really >> bringing any real facts to the discussion. When these things change, >> we might be able to have a meaningful discussion. > > > I honestly can't believe that you would say what you said while accusing > me of telling lies,
I did not accuse you of telling lies. I said you're not really bringing any real facts to the discussion. For example, You are a person with some audio experience... I don't need to explain this you. I'm wondering why you asked me to come here and then would argue this point. This statement says nothing about summers. It provided no new information, and its main intent is to attack me personally. > I've written a few digital summers in my day and I don't know what > you're talking about. I think you do. As a matter of fact, I think you have to take into account this fact when doing your designing of any digital audio summing buss. "This fact" has not been defined. What are we even talking about?
> Thus far, aside from a few technical threads which admittedly barely > or do not comprehend, this place is -generally speaking- another > mass of ego and misdirection like rec.audio.opinion.
I disagree. In my opinion, you are the one bringing down the SNR of this group. David, I'm not trying to insult you. I'm simply giving you some feedback that is intended to facilitate more meaningful dialogue. Cool down. Be courteous and kind. Stick with facts and information relevent to the technical side of things rather than hurling insults and innuendo. If you do this, there are many extremely smart and experienced folks here (myself not necessarily included) who would be willing to discuss the fascinating world of DSP with you. -- % Randy Yates % "And all that I can do %% Fuquay-Varina, NC % is say I'm sorry, %%% 919-577-9882 % that's the way it goes..." %%%% <yates@ieee.org> % Getting To The Point', *Balance of Power*, ELO http://home.earthlink.net/~yatescr
David Morgan (MAMS) wrote:

   ...

>>>>> I wasn't including "processing" math; which I for one, had much rather keep >>>>> in the analogue domain. > >>>> What are you babbling about? > >>> Babbling? Contain yourself dear boy, at look at the REAL recording industry, > >> Claiming that mixing and setting levels -- processing to be sure -- >> should be by analog means > > Babbling, my dear Jerry, is thinking you can read my mind, and thay you > have all of the correct answers before any questions are asked... THAT is > "babbling." > > But while we're at it, an analogue mixer will not impart summing problems > and CPU issues that one will run into when mixing "in the box". Unfortunately, > most people simply don't have the experience to understand that and thus can't > hear when they are damaging their material.
You can get pretty good SNR with an analog mixer, but you can do better digitally with enough bits. You ought to know that. I believe you do.
>> ...even in digital recordings is babbling. > > "DIGITAL" is nothing more than another storage medium. Decent software > and decent interfacing hardware allows *complete* interfacing to the real- > world of analogue for any and all processes. I prefer that world of quality > hardware far more than using a handfull of algorithms to mathematically > attempt to model said analogue world.
Recording? That's a case of tunnel vision. How are "digital" and record" inextricably linked? (OK; somebody out there might be recording my digital phone conversations.) There's data logging, but no recording in the digital servos I've designed.
>> If you didn't mean that, what did you try to write? > > You definitely ARE haveing a knee-jerking afternoon, Jerry. I never > "CLAIMED" a damn thing, nor did I ever suggest that anything "SHOULD" > be done a specific way. I SAID, "I for one, had much rather keep in the > analogue domain." It's right above, in plain sight; read it again. > > My presonal preference should be no call for a knee-jerking assault > of 'incorrect' on your part.
OK: you didn'y say *I* should go analog, only that you think *you* should. Accepted.
>>> not the million homes full of dunces with computers who THINK they actually >>> comprehend how real music is recorded and mixed. If I had a dollar for every >>> dim-witted, inexperienced, shill who purchased a PC and some software and >>> printed a business card claiming to be a recording engineer / producer, etc., >>> I'd be happy to buy you and me a *real* studio using all of the "digital tracks" >>> you would ever care to store. ;-)
Those dunces have nothing to do with either of us. Some of them even use half-inch bundles of Litz wire for speaker cabling and swear they can hear the difference.
>> You seem to have a bone to pick with audio types. > > Excuse me? I've spent the last 33 years of my life as an "audio type". > If I have a bone to pick, it is with snake-oil salesmen and the continually > dropping 'bar' of quality audio recording and reproduction.
I was in the analog end of audio far enough back to be running a Scully lathe. You must be aware of the "pinch effect" caused by recording with a cutter that lies essentially in a vertical plane and playing back with a rounded cone. Pinch effect shows up as even-harmonic hill-and-dale motion due to lateral recording, and has very little effect in mono. It comes through loud and clear in stereo. Listen carefully to some early stereo recordings and you'll hear for yourself. We at RCA worked out non-linear compensating electronics to correct that. Sort of. (We also used excursion look-ahead to set variable groove spacing, with finer pitch on soft passages. It gave us up tp 20% more playing time.)
> Of course, DSP freaks who think there is only one way to access, > process, and otherwise manipulate a digital audio file don't make > life any easier. Digital is a storage medium... and if you treat it as > such, a whole new world of potential quality will open up for you.
Most of the signals that get digitally processed get used, not recorded. Get over that hangup. (Saving bandwidth is probably more important than reducing storage space. A single-layer DVD holds an awful lot of Bach.)
>> DSP involves a lot >> more than that, with the same principles applying to all of it. There's >> a common thread that you're too eager to particularize. > > The common thread between us, thus far, is that you apparently lack > any real-world or analogue audio experience. Don't foget... all raw > audio (moving air), no matter how captured or how stored, is in fact > ANALOGUE in it's creation and in it's perception.
I built my first low-distortion sound system from components in 1950. When did you start?
>>>> How do you mix and adjust gain with digital tracks? > >>> You use a REAL analogue mixing desk. It requires a little investment of >>> money for interfacing and a certain level of experience and intelligence >>> of practical application. If you haven't seen one lately, check my web site. > >> Gee! You could change sample rates that way, too. I wonder why >> decimation and interpolation are so widely used. > > Yes, you could... but I don't necessarily recommend it -- any more than > I would recommend recording using unecessarily high sample rates under > the misunderstood guise that it somehow reproduces with more accuracy > to the human ear (mathematics aside for a moment). Above 48K and > you are basically wasting your time, drive space, accuracy in returning > to the standard delivery medium, and recording the unrecordable for the > purpose of addressing the hearing of dogs.
Consider a 48 KHz sample rate for an audio system that needs to be flat to 20 KHz. (Don't ask me why: that's the spec.) To avoid aliasing, there can be no energy above 24 KHz. Clearly, there will need to be a little aliasing because no filter is perfect. How much to allow? 40 dB down? Then we need an a_analog_ filter that has 40 dB attenuation at 28 KHz and doesn't louse up the signal below 20. That's about 250 dB/decade. No ringing, little phase shift. Clearly, commercial designs don't meet that spec. (28, not 24 if we also digitally lowpass after sampling. Double the slope otherwise. You'll have the same constraints with the reconstruction filter after the DAC. Now let's sample higher, 96 KHz because that a standard. Now, we need to be down 40 dB at -- wait: there's so much slop, we can go for 80 dB. 80 dB down at 176 KHz with after-sampling removal aliases above 20 KHz. That's a filter I can actually make. The reconstruction filter will have the same relief. If you want to save storage apace, decimate before storing and maybe interpolate before going analog.
>>>>>> I try to avoid numbers in threads like this because they often lead to >>>>>> unproductive digressions. > >>>>> I can see where that would become an issue around here. >>>>> >>>>> ;-) > >>>> It's an issue in any thread started by a request for basic concepts. > >>> Basic concepts begin with capturing and processing real audio, not >>> mathematics. Sampling rates are the biggest lot of hype and baloney >>> to hit the consumer audio marketplace in a long, long time. They're >>> called "marketing bits" by anyone with real-world experience in audio >>> recording. > >> Try to sample 20 KHz music with an 8 KHz sampler, then tell me what >> baloney is. > > You just completely changed our discussion by entering into a realm > that is not only assanine, but plain damned stupid, and a million miles > away from the subject of using abnormally high and unecessary > sampling rates. Moving to an analogy of comparing to lower sampling > rates makes about as much sense as the bozo who relates digital > imaging to digital audio.
No, /you/ changed the discussion I was having with the OP who wanted to know if the signal between the samples got lost and how the signal between the quantization levels behaved. You barged in like a bull in a china shop and started throwing numbers around. No, not a bull: an old geezer on a hobby horse.
> Have a nice day Jerry.
You too, David. Jerry -- Engineering is the art of making what you want from things you can get. &macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;&macr;
On Apr 16, 6:21 pm, "David Morgan \(MAMS\)" <fin...@m-a-m-s.comC/Odm>
wrote:
> Don't foget... all raw > audio (moving air), no matter how captured or how stored, is in fact > ANALOGUE in it's creation and in it's perception.
All raw audio is the result of individual molecules bouncing off of eardrums and microphone heads (exchanging quantized energy levels at that). Audio only appears to be analog once the measurement error or approximation abstracts sound above the level of thermal noise from the Brownian motion of these collections of air molecules, etc. Not sure how low the dynamic range of hearing goes, but the human eye reacts to countably small integer numbers of light photons. IMHO. YMMV. -- rhn A.T nicholson d.0.t C-o-M