24 bit

[jellyfish420]

what is the actual sound difference between 1 bit D/A and 24 bit? what do the bits do?

[audiolife]

transfer of information. 1 bit does it 1 bit at a time 24 bit 24 bits at a time. depending on the recording you can tell the diff. if all you listen too is fat joe or the eastside boys i doubt you could tell the diff.

[Shugarra]

Okay, no way in hell I could explain it correctly, so I'll just send you to a link, you might have to sign up for the message board, but it's worth it:

http://forum.elitecaraudio.com/showt...threadid=95760

1 bit isn't necessarily worse than 24 bits, by the way.

[johnecon2001]

uhh.. I don't think thats correct. But I'll ask around to be sure.

I know 1bit D/A is not the same kind of thing as 24bit.

[audiolife]

it depends on what you do with it after the deck........

[Shugarra]

Found a more direct thread: http://www.elitecaraudio.com/forums/...3&pagenumber=1

I'm going to copy and paste the part about bits in the ADC process but I want to be very clear that the only part I had in this information was the copying/pasting part. All credit goes to Werewolf.

So far, we have only 'sampled' the analog waveform. But we rest assured in the knowledge that, thanks to Mr. Nyquist, we have not yet introduced ANY ERRORS ... because later on, we can completely recover the signal from it's samples. But those samples are still kinda analog ...

So now we do the true analog-to-digital conversion process, and convert these 'kinda analog' voltage samples to DIGITAL WORDS (yes, we really do call 'em words). We will see that this process ... called quantization ... does in fact introduce errors (and believe me when I tell you, a person could EASILY devote an entire career to the study of quantization).

What better way than to proceed with an example? And we'll ask for our very first homework assignment shortly Let's say we have an analog voltage waveform that spans the range from 0.10 volts to 0.90 volts ... can take on ANY value between these two limits (chosen for simplicity). Furthermore, let's say we want to build ourselves a 3-bit ADC to "digitize" this waveform (OK, so 3-bits doesn't sound very high-end ... but I caution you to not judge this book by it's cover ... plus it's alot easier as an example).
So here's how the converter will work : if the analog voltage sample (let's say we've done the sampling described above) is BETWEEN 0.10 and 0.20 volts, we give to that "analog" sample the "digital" word : 000 . If the analog voltage sample is BETWEEN 0.20 and 0.30 volts, we give the analog sample the new digital value of : 001 . So let's construct the followng table :

Analog Voltage Sample Value Corresponding Digital Word

0.10 --> 0.20 000
0.20 --> 0.30 001
0.30 --> 0.40 010
0.40 --> 0.50 011
0.50 --> 0.60 100
0.60 --> 0.70 101
0.70 --> 0.80 110
0.80 --> 0.90 111

So here's an example : analog voltage sample of 0.37 would be assigned a digital value of : 010. But please note, that an analog voltage sample of 0.34 would be assigned the SAME digital value ... so here, for the first time, we introduce an ERROR ... called (cleverly) QUANTIZATION ERROR, or (somewhat incorrectly, but vastly used), QUANTIZATION NOISE.
In the case of my example of 0.37, the actual quantization error would be 0.02 ... because we "expect" the digital word 010 to correspond precisely with the analog voltage halfway in it's region, or 0.35 (and 0.37 - 0.35 = 0.02).

So why would anybody put a precious analog signal through such a "noisy" process? Because, once the signal is digitized into these digital words, it is (virtually) immune to further errors in storage and/or communication (this is the essence of why we like digital). And if we use ENOUGH BITS to quantize the signal (more than 3), the quantization noise will be very small indeed ... much lower than other forms of noise that plague analog storage & communication.

How do I build such a converter out of real circuits? Well, the type of analog-to-digital conversion described here is pretty straightforward to build. I might start with a 1 volt voltage reference, and use a resistor-divider chain to establish all the "boundary" voltages (0.1, 0.2, etc), creating a series, or bank, or array of voltages. Then I use a series of comparators, all of which have one input tied to the analog signal, and the other input tied to one of each of the voltage references. The comparator outputs will tell me what digital word to assign to the analog input. Make sense? The whole key, however, for this process to work is this : I must VERY ACCURATELY establish ALL of these "boundary" voltages, to compare my signal against.
And if you think that this technique, for 16-bit conversion, means I must establish voltage references that are PRECISE to within ONE PART in 2**16=65,536 ... that's where you would be RIGHT. A VERY difficult challenge ... more on this later.


He goes on later to explain how a 1 bit process can be just as good or better than a multibit decoder so long as the proper oversampling is used.

[m()nk3yb()y]

If you are into the nitty gritty parts of audio, especially digital audio, read that post. Werewolf knows his shit for sure.

[desertheat]

Exactly, I currently use 1bit burr browns in most of my very very high end home audio gear and still in my car audio gear. People that bragg about there 24bit d/a converter usually have no idea how a dac works in the first place. They just think because it says 24 on it and 24 is a higher # than 1, that there 24bit dac is WAY better... to funny to me Eclipse owners do this all the time, bragging about there 24bit dacs and how superior they are to everything else, not knowing that the same dac they use is found in many $150-$200 decks. Not saying eclipse does not make a fantastic deck, but that should not be the feature they bragg about the most Ok bring on the fire hahaha.

[desertheat]

I like this section also...

The real benefit of a one-bit converter is this : mismatch can only cause a harmless gain or offset error ... one bit means only two values or "points" on a curve, and two points define a straight line, pure and simple (it's deviations from a staight-line transfer function that cause distortion).

So the analog signal first passes through a GRADUAL analog anti-alias filter, which just needs to provide healthy attenuation by 3Mhz. Next, the signal is sampled at maybe 3Mhz (or higher), and then digitized by an algorithm very similar to the one you guys built. In short, yes it's only one-bit ... and hence very "noisy". But the algorithm is designed to make sure that the noise is "shaped" in frequency so that the noise is VERY LOW in the low frequency band of interest (20kHz). This is why even simple averaging, like what you guys did, "reveals" the high resolution possible ... averaging is a form of low-pass filtering, which in this case "removed" alot of the one-bit quantization noise at high frequencies.


In fact, the one-bit signal is then sent to a DIGITAL low-pass filter, not unlike the averaging process you guys did. This is interesting ... it's really this digital filter that : removes most of that one-bit quantization noise, "revealing" higher precision digital words, and ... get this ... provides the SHARP anti-aliasing needed before the final step of "decimation" ... which is simply lowering the sampling rate (a sampling process itself) back down to 44.1kHz Yes, a digital filter provides the real sharp anti-aliasing needed for 44.1kHz samples !! Why do we like a digital anti-alias instead of analog? No component drift, no chance of power supply noise creeping in, and finally ... can be implemented as FIR with perfectly linear phase By the way, an FIR filter is really NOTHING MORE than a "weighted" averaging filter, that takes a running average of many (much more than 64) one bit samples to produce a higher precision digital word. And as a nice bonus, it turns out that the computation required for the FIR is not bad at all in a decimation environment ... since there's no feedback in this filter structure, you never need to calculate the outputs you're going to ignore after you downsample to 44.1kHz So that's it, the one-bit bitstream is digitally filtered by a long FIR lowpass filter, generating higher precision digital words ... and we only need about every 64th word (after filtering) to supply the 16 bit words at 44.1kHz





Why do ya think alpine never went away from 1bit..?? hmmmmm

[3.5Max6spd]

I like this section also...

Why do ya think alpine never went away from 1bit..?? hmmmmm

Ever notice how extensive Alpine's DSP is? The Eq dept is lacking for a reason in comparison tho say Pio's 20 bit and Eclipse 24bit units- there's a limit to how much processing a 1bit DAC can handle after the lazer pickup without sacrificing the digital source quality- larger DAC's can handle this more efficiently...

[desertheat]

Ah so you are saying that all of the time alignment and crossover functions are handled directly by the adc and dac and the adc and dac are what processes the crossover and time functions..?

Nope they are not. It is a totally seperate processor and a 1bit dac or adc is MORE than capable of taking care of the changes in the audio signal made by the seperate processor.

No matter what, the adc is handeling what comes off the disk and the dac is handeling the audio after the signal is processed for time / crossover functions. Because the sound is processed before the dac, it does not matter that it is still a 1 bit dac.

That would be like saying a cd recorded with alignment built in the recording or frequencies pre crossed over "on the original recording" would be to much for a 1 bit dac to handle, sorry it is simply not the case.

I have talked to alpine engineers on several occasions and the ability to add several more bands of eq, mor eq ability and even steeper slopes is 100% there, but they will not do it for **** stupid reasons and tell me that more will come with later versions... darn deck nazi's grrrr

[desertheat]

transfer of information. 1 bit does it 1 bit at a time 24 bit 24 bits at a time. depending on the recording you can tell the diff. if all you listen too is fat joe or the eastside boys i doubt you could tell the diff.

I would give you my whole system if you can tell the difference on a recording between a well made 1bit dac and a 24 bit dac. Sorry man you just will not hear the difference.

But is there anything wrong with having a 24bit dac..? Nope, anything wrong with a correctly built 1bit dac..? Nope... Both are great if built and deployed correctly and both will sound perfect to 99.9999999999999999999999999999% of everyone on the planet except the self proclaimed "super ear" retards that claim to be able to hear the difference of a cd player when it is tilted or set flat hahaha.

In closing, pine, lips and pio make steller decks and all should be considered "audiophile" grade when it comes to car audio.

[audiolife]

i gave up my 1999 top of the line eclipse for what i have now simply for going all digital and at the time there was a difference plus with what i have now i can do everything i need 1 time in 1 little box not to mention when you run fiber out i bypass my decks dacs anyway. now granted its still bigger than 1 bit but from what i understand 1 bit when it get processed is more likely to shift out of its normal range and allow/not allow changes . if there wasnt any more to it then why did the company that basically fought about the goods of 1 bit go to 24 bit when they went with the onboard digital processing? i have read in sooooooo many factions that fight one way or the other but to this day i have yet to hear a high quality 1 bit process of dsp. when you have that much control literially a 1 bit wont do it as well.

[desertheat]

You guys are getting onboard processing confused with your output dac, kinda like getting your processor confused with your nothbridge on your pc With your fiber optic, it is only as good as your adc. Both 24 and 1 bit are fine. Again you will not be able to tell the difference as much as any of you want to be able to

[audiolife]

ok then tell me why they dont use 1 bit dac in a serious digital processor. eclipse has 1 bit down pat i wont argue that but on their high end decks they are 24 bit, the alpine f1 24 bit, sony es 24 bit even the p1 or was it the p9? pioneer 24 bit pioneer odr was 20 bit and 24......why did a dry sony es c -90 sound better than my eclipse? no processing (i wasnt running balance out but thats about as big of a deal as 1 bit vs 24 bit)