Are all 24-bit DAC’s up-sampling??
Dec 2, 2003 at 9:05 PM Thread Starter Post #1 of 9

boead

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I can’t seem to get a straight answer on this.

Since Redbook CD’s are 16-bit, isn’t anything over that just being up-sampled?

Some manufacturers call their DAC’s up-sampling, and some do not.

So are all 24-bit DAC (specifically Burr Brown’s) up-sampling from a 16-bit Redbook source?
Either standalone players or separate DAC’s.
 
Dec 2, 2003 at 9:13 PM Post #2 of 9
No they are not.
 
Dec 2, 2003 at 9:57 PM Post #3 of 9
Quote:

Originally posted by tom hankins
No they are not.


Please explain.
 
Dec 2, 2003 at 9:58 PM Post #4 of 9
A 24 bit dac means the dac handles, at maximum, 24 bit word lengths. A 16 bit dac handles 16 bit word lengths at maximum.16 bit word lenghts can be increased up to 24 bit word lenghts with dither or resolution enhancement in some processores from DCS or Perpetual Technologies for instance. Upsampling is increasing the speed of the normal redbook sampling rate of 44.1 khz to multiples of that rate. These rates are usually 88.2/96/176.4/192. Both of these processes are usually done together with todays technology and have ended up being looped together; but upsampling originally referred only to increasing the sampling rate. DVD audio is encoded at a true 24 bit word length. No processes are needed to increase the word length because its already at the 24 bit maximum.
 
Dec 2, 2003 at 10:10 PM Post #5 of 9
boead: There are tons of possibilities and capabilities on and with digital audio devices. It all depends on individual device capabilities, settings and signal routing.* For the possibilities of differents dacs, s/p-dif transceivers, filter chips et cetera you'd best look at the data books on manufurer web-sites like Burr-Brown, Crystal, Wolfson... That should also give you a basic impression how these chips could work together and what features would be involved.

* Example: For 24/96 upsampled redbook, you could either use an upsampling cd/dvd player, or the player with an external upsampling dac, or (if the input receiver of the dac also accepts 24/96 - which goes for most modern upsampling dacs and, of course, 24/96 or 24/192 dacs that are meant to be combined with a separate upsampler) the player the dac and a separate upsampler in between - or a 24/96 or even 24/192 capable soundcard plus software (be that separate software or soudcard driver) upsampling, or... whatever.
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But if your question means: "If I hear sound playing from a redbook audio cd rendered in 24/96, is there upsampling involved?", then the answer is: "Yes, there is."
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Greetings from Munich!

Manfred / lini

P.S.: Definition disclaimer: I'm using the term upsampling for both sampling rate and word length increase together.
 
Dec 2, 2003 at 10:13 PM Post #6 of 9
Any 24-bit DAC that includes a digital filter (i.e. virtually all of them) will generate 24-bit values internally, post digital-filter, regardless of the input word size.

For a really good discussion of this (how, why, etc.) see this application note from Analog Devices:
http://www.analog.com/UploadedFiles/...3938AN-327.pdf
 
Dec 2, 2003 at 10:50 PM Post #7 of 9
Quote:

Originally posted by lini
But if your question means: "If I hear sound playing from a redbook audio cd rendered in 24/96, is there upsampling involved?", then the answer is: "Yes, there is."
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Thanks, and yes.

I just read this:

Oversampling-
The oversampling technique was developed to get away from a "brickwall filter". A digital system interpolates new points between the different original samples to obtain an artificially higher sampling rate. This allows the use of a less aggressive filter because it doesn't have to eliminate frequencies as close to the frequencies it must not affect. It first began as four times (4x) oversampling (i.e. 176.4kHz), then later eight times (8x) oversampling (i.e. 352.8kHz). The digital filter must perform many mathematical calculations to determine the value of the point it must add to the original digital signal. Often, this calculated value may fall between two discrete values, so the oversampling system must round off the value to the closest discrete value. To increase the precision of the resulting calculated value, DACs and digital filters with more than 16-bits of resolution were therefore introduced. We have seen 18-bit, 20-bit and 24-bit digital filters and DACs. It is important to note that oversampling creates an artificially higher sampling frequency, which does not extend the real frequency response of the original media or the system, but simply extends the frequencies that need to be filtered out, allowing for a simpler and better sounding analog filter.

Upsampling & Upconversion-
One of the latest storage mediums is the popular the Digital Versatile Disc (DVD). When developing this new standard, a higher-than-CD resolution PCM format was adopted with a maximum resolution of 24-bits/96kHz. For the professional market, this new format had to be compatible with the CD's 16-bit/44.1kHz resolution. This would allow the conversion of original recordings to the new standard. So a sample-rate converter chip, which is nothing more than an oversampling digital filter, was created to actually convert any digital signal from one standard format to another format. For example, a 16-bit/32kHz signal can then be converted to 24-bit/96kHz and 24-bit/96kHz can also be converted to 16-bit/48kHz. This gave rise to the marketing hype with the concepts of upsampling and upconversion, which claims could upsample or upconvert your 16-bit/44.1kHz CD to a 24-bit/96kHz resolution digital signal prior to the digital to analog conversion, resulting in DVD-audio like quality from CD. While this statement is a great idea for marketing purposes and is surely impressive to most consumers, it is technically only half true, and is not the best way to improve the audio quality that can be derived from CDs.
Why?
Digital filtering is digital filtering regardless of name assigned to it, and how the interpolation is made still relies solely on the arithmetic calculations implanted in dedicated hardware or software. The main difference is how well the "mechanics" of the mathematics will assist in the signal's reconstruction. When changing the sampling rate, it is better to maintain an integer multiple of the original signal's sample rate, so the processing is kept simple. More importantly, the end result is more accurate, thus enabling a higher fidelity of sound reproduction. A two times (2x) oversampling system will double the sampling rate, by adding one easy to find numerical value in between each actual sample. For example, when a 44.1kHz digital signal is processed, a 88.2kHz digital signal is obtained. It is simple, effective and precise because it is a direct multiple of the original digital signal. For an upsampler to make a 96kHz digital signal from a 44.1kHz signal, it will have to perform awkward mathematical operations to obtain a 96kHz signal. (96kHz / 44.1KHz equals 2.1768707...). This results in a less accurate output from the digital filter, with everything else following (i.e. digital-to-analog conversion and analog filtering) also being less accurate. As well, exactly like oversampling, the artificially higher sampling frequency created by an upsampler doesn't increase the actual frequency response of the system, but simply increases the lower limit of the frequencies that need to be eliminated.
 
Dec 3, 2003 at 2:09 AM Post #8 of 9
Quote:

Originally posted by boead
For an upsampler to make a 96kHz digital signal from a 44.1kHz signal, it will have to perform awkward mathematical operations to obtain a 96kHz signal. (96kHz / 44.1KHz equals 2.1768707...). This results in a less accurate output from the digital filter, with everything else following (i.e. digital-to-analog conversion and analog filtering) also being less accurate.


This used to be true (and still is if you use the brute-force approach to 44.1->96 kHz). However, the current generation of asynchronous upsamplers (the Analog Devices AD1896 and Burr-Brown SRC4192) use very sophisticated table-driven techniques to perform the conversion (check out the AD1896 datasheet for details) and are extraordinarily precise. The SRC4192 even does all internal math at 28-bit precision. Nice.
 
Dec 3, 2003 at 8:24 PM Post #9 of 9
great post boead, but Wodgy is right.. ASRC chips and SSRC programs tries to reconstruct the waveform (ie. find the function which describes it or get it from their stored tables) and decimate it again, digital filters are inserting the values in between the neighbour samples using plain mathematic.. I hope I'm right on this
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