lindsayt said:If we're a tiny bit sensible and limit the maximum volume to 120 dbs, that gives us a range limit of human hearing of 100 to 120 dbs for people with non-damaged hearing. That's more than 16/44.1 can provide full stop. And it's way more than 16/44.1 can provide without noticeable amounts of distortion.
Phileas said:lindsayt said:If we're a tiny bit sensible and limit the maximum volume to 120 dbs, that gives us a range limit of human hearing of 100 to 120 dbs for people with non-damaged hearing. That's more than 16/44.1 can provide full stop. And it's way more than 16/44.1 can provide without noticeable amounts of distortion.
Clearly there's more to digital audio than The Sampling Theorem (which assumes exact sampling) so there's no point trying to debate that here. The important question is whether you can actually hear the distortion.
It seems to me the best way to determine that would be to insert a Red Book ADA "bottleneck" into an analogue replay chain and compare it with the original in a properly controlled ABX test. I suspect the result would be negative.
fr0g said:While your theory is ok, in practice, in a normal listening room, it's irrelevant. I know for a fact that in my house (in a very quiet area) that the noise floor is at 30 to 40 dB when the room is "silent". CD quality is more than enough, and way more than the DR of vinyl.
However I would be quite happy if they upped the standard to 24 bit. It wouldn't hurt and would help with digital volume control issues.
As for Nyquist-Shannon....it has nothing with that to do...44,1 covers any human need...
lindsayt said:Frog, thank you for that. So to clarify, would you agree with me that Steve_1979's post about the limits of human hearing and 16/44.1 format was factually incorrect?
I don't understand why he was talking up the CD format to make it appear to be better than it actually is?
lindsayt said:fr0g said:While your theory is ok, in practice, in a normal listening room, it's irrelevant. I know for a fact that in my house (in a very quiet area) that the noise floor is at 30 to 40 dB when the room is "silent". CD quality is more than enough, and way more than the DR of vinyl.
However I would be quite happy if they upped the standard to 24 bit. It wouldn't hurt and would help with digital volume control issues.
As for Nyquist-Shannon....it has nothing with that to do...44,1 covers any human need...
Frog, thank you for that. So to clarify, would you agree with me that Steve_1979's post about the limits of human hearing and 16/44.1 format was factually incorrect?
I don't understand why he was talking up the CD format to make it appear to be better than it actually is?
andyjm said:manicm said:fr0g said:manicm said:steve_1979 said:A 44.1kHz sample rate can accurately reproduce any wave upto 22.05kHz and 16bit audio can reproduce the full dynamic range that is audible to humans. What other considerations are necessary?
(A genuine question. Not just being awkward. )
Reading from one of the links that CnoEvil provided the bit depth (16 as you refer to here) has got nothing to do with dynamic range, but the amount of data that can be stored at any interval of time, and which allows for smoother wave form over a given time - hence the potential to sound less sharp or 'digital'. And that is quite understandable to the layman.
This shows a complete misunderstanding of the theorum. NOTHING can sound "digital". It is ALL analogue, and so long as the sample rate is more than half the maximum frequency required, then that analogue wave that comes out of your speakers is identical to what went in...
I wasn't talking about the theorem, I was talking about the bit depth - which you conveniently ignore in your response. In which case it's not a 'complete misunderstanding'.
Manic, I would recommend a bit of googling on the subject as your understanding is definitely adrift here.
Bit depth drives dynamic range.
Sample rate drives maximum frequency that can be correctly sampled.
andyjm said:oldric_naubhoff said:in general you've got two processes occuring simultaneously when you digitize an analog signal; sampling and quantizing. sampling is based on N-S theorem and it is true that sampling is error-less up to its limit, ie. sampling half band frequency. sampling in analog to digital conversion is responsible for digitazing the frequency spectrum of the signal. however, quantizing is used to capture level of the signal being digitized and quantizing, by definition, will never be error-less, simply because you've only got a limited amount of volume levels which you can apply to analog signal during quantization. as Lindsayt rightly points out; 16 bit resolution gives you 65 536 (or 2^16) discrete levels, 24 bit resolution givvevs you 16 777 216 discrete volume levels, 32 bit resolution nearly 4.3 bilion discrete volume levels. aliasing is a by-product of quantizing process and the reason why it occurs is exactly the fact that you can't fully quantize analog signal.
but on the other hand one might argue that even for 16 bit resolution the number of discrete volume levels is high enough, provided you only use a limited DR within 16bit full DR range, that you could capture the analog signal very faithfully - in effect indistinguishably from the analog input signal. still, the reality is that more bits of resolution are better for the purpose of faithfully capturing of the analog signal.
Quantisation has nothing to do with aliasing,
andyjm said:Aliasing occurs when an input signal is sampled at less than twice its maximum frequency. Frequencies above (sample rate)/2 appear in the reconstructed signal as lower frequencies or aliases.
The higher frequencies act as if they have been reflected around (sample rate/2). So a 22KHz signal sampled at 40KHz shows up as 18KHz in the reconstructed signal, 30KHz sampled at 40KHz shows up as a10KHz alias. Strange effect, but there it is.
This is a avoided by having an anti aliasing filter before the sampling process to ensure no signal above (sample rate/2) is sampled.
oldric_naubhoff said:good post there. correct me if I'm wrong but doesn't it mean you'll always have problems with aliasing artifacts since you'd always have input signal with frequencies higher than sample rate/2 (doh). in which case an anti-aliasing filter (regardles digital or analog) is an essencial part of construction of any DAC?
it's not a secret that steep roll off filters induce high amounts of pre and/or post ringing into reconstructed signal. therefore it may be useful to use slow roll-off anti-aliasing filters together with higer sampling frequncies to avoid attenuating of high end of audible frequency range and bleeding of aliasing artifacts into audible frequency range. this is nothing "scientificaly proven" but I red in numerous places that slow roll-off filters "sound" better than fast roll-off ones, which IMO could be plausible since slow roll-off filters induce less digital artifacts into converted signal... this is the only reason I could find that justifies using higher sampling frequencies than standard Red Book.
steve_1979 said:... 16/44.1 gives you full dynamic range beyond what anyone would need in any real domestic situation. Unless you are a young child living in an anechoic chamber, 16/44.1 can go from being as quiet as a silent residential room right the way up to ear damaging/borderline painfull volume levels without you needing to touch the volume control. Why would you need anything more than that?
steve_1979 said:steve_1979 said:... 16/44.1 gives you full dynamic range beyond what anyone would need in any real domestic situation. Unless you are a young child living in an anechoic chamber, 16/44.1 can go from being as quiet as a silent residential room right the way up to ear damaging/borderline painfull volume levels without you needing to touch the volume control. Why would you need anything more than that?
Here's an ironic thought that just struck me. :?
How many of the audiophiles who think that they can hear a difference with 24 bit audio are using underpowered 30-100 watt amplifiers together with power sapping passive speakers? This type of hifi system doesn't even have the necessary dynamic range capabilities to make full use of what 16 bit audio can offer never mind 24 bit.
steve_1979 said:.... underpowered 30-100 watt amplifiers together with power sapping passive speakers?
steve_1979 said:.... underpowered 30-100 watt amplifiers together with power sapping passive speakers?
chebby said:(And a little odd from someone using "power sapping passives".)
matt49 said:This has been an interesting discussion. By banging your "all passive systems are rubbish" drum, you will make it less interesting.
Matt
steve_1979 said:The ironic thought that I mentioned in my previous post was that many of the audiophiles to think that 24 bit audio is of benefit to them are using underpowered amplifiers (anything under 100 watts give or take) together with passive speakers (which are less efficient than actives) and are only capible of producing a limited amount of dynamic range anyway.
fr0g said:http://www.linnrecords.com/recording-just-music-exclusive-linn-compilation.aspx
manicm said:steve_1979 said:The ironic thought that I mentioned in my previous post was that many of the audiophiles to think that 24 bit audio is of benefit to them are using underpowered amplifiers (anything under 100 watts give or take) together with passive speakers (which are less efficient than actives) and are only capible of producing a limited amount of dynamic range anyway.
Please provide a scientific explanation of this, that is easy enough to a layman, that a system with a good 100w amp, or even a good 80w amp, with really good passive speakers is unable to reproduce 24 bit audio adequately. And why will only an active system do so.
Methinks you're skating on very thin scientific ice.
steve_1979 said:I never said that only an active system would do. Notice that I said: "the same could be said for any underpowered active systems too".
Buy to have the dynamic range capibilities to go loud enough to make full use of what 16 bit audio can offer require an amplifier with lots of power.
Paul. said:I was learning some things from this thread, please don't make it a passive/active **** measuring thread.
*goes back to lurking*
manicm said:steve_1979 said:I never said that only an active system would do. Notice that I said: "the same could be said for any underpowered active systems too".
Buy to have the dynamic range capibilities to go loud enough to make full use of what 16 bit audio can offer require an amplifier with lots of power.
So, once again, explain why a passive system, with a really good 80w amplifier and quality speakers, is unable to playback high-res audio to a level that is discernible to 16/44, and only a 100w+ system is.