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In Reply to: RE: Attention High Resolution Proponents. Neil Young/Steve Jobs posted by geezerrocket on October 07, 2015 at 14:00:05
And they could be right.
In the case where a recording master has dynamic range of less than 16 bits and a frequency response less that the maximum covered with a 44.1K sampling rate and where the ADC and the associated DAC at 24/192 creates more noise and high frequency artifact than that of Redbook CD, perhaps yes.
Maybe.
Will a recording played using the 'Worlds Best' 16/44.1 DAC sound better than the same original track up-sampled and played back on the worlds cheapest POS 24/192 DAC?
Maybe.
Am I responding to a TROLL Post?
Could be!
Follow Ups:
Ivan,it was fun seeing you at RMAF.....too bad we were too late for the Magico room.
not true.....44.1 does not sound better. but I would also agree that the native resolution of the recording can limit what resolution sounds best. but assuming some level of higher rez than redbook to start out, or starting out as analog, then the higher rez native resolution should sound better. I prefer almost always to hear any recording in it's native format. which includes tape, direct to disc Lp, and any digital.
I have what is said to be the best PCM dac, the Trinity Dac, and I can say unequivocally that while the redbook is superlative, higher rez.....88, 96, 176, 192 is better.....as you go up the steps better and better.....generally.
the Trinity dac has easily the finest sounding redbook I have yet heard. it overcomes all the inherent artifacts of PCM playback and takes it to another level. as a 15 year very strong proponent of dsd and a person with 3000+ CD's sitting dormant 'for years' until 6 weeks ago, I was blown away by hearing redbook on the Trinity dac. it changed my world view of things digital. but....and this is a big but......it takes a very high effort to overcome PCM's artifacts. dsd is much easier to approach optimization.
I have 7 terabytes of PCM, and another 8 terabytes of dsd, and so I have plenty of data to have a pretty good idea of how this works. I also have a server set-up which is pretty SOTA, the CAPS v4 Pipeline with additional hot-rodded LPS units added.
last month I ripped 1500 of my 3000 CD's.....and after I recover from all that i'll rip the remaining 1500. I'm delighted to be reacquainted with my old CD friends in their shiney new clothes. lots of great music which now sounds so much better.
cheers,
mikel
Edits: 10/07/15 10/07/15 10/07/15 10/07/15
They SHOULD turn me away, You? I'd have let you in, set the room up again and given you a private demo! ;-)
Agree that the BEST Redbook CD could be surpassed by the BEST Hi Rez and yes, DSD from mic to speaker with no PCM conversion in between (Jared at Channel Classics certainly does the 'chicken right') would be the perfect world.
ALMOST as good as full analog chain mic to speaker via vinyl or 15 IPS tape. ;-)
At the end of the day, recording quality makes as much difference as playback technology IMNSHO. I can generally tell a great recording from a mediocre one just by streaming it over the internet.
Mostly. Hopefully I will someday find out for certain.
Envy you your Trinity DAC. Looks like you have four times the PCM1704U-K as I'll have soon (Audio-GD Master 11 on a slow boat from you know where). If it turns out to be even 25% as great at 5% the price, I'll be happy.
Keep your eye on Peter Q at Audio Note. Haven't spoken to him in a while but understand he still feels there is miles to go in Digital to Analog conversion via discrete resistor ladder DACs.
Wish him luck!
Great to to see you again and keep pushing the envelope!
I got my Master 7 and my friend got his in 5 days
Alan
Chinese Holiday I didn't know about even living in San Francisco where Lunar New Year is a bigger deal than January 1.
I understand they are now quite busy clearing the backlog as the word is definitely getting out.
That said, the PCM1704U-K is still available from Mauser at about $60+ in large quantities (I put 150 in my 'cart' just to see if they would take the order and the do) so I don't thing there is any fear that King Wa will run out of chips.
The technical explanation behind the Lianotec design is somewhat fuzzy to me.... I guess I'd have to listen to one in order to find out if it really does something special.
I think the notion that a 384 kHz native signal being oversampled in order to get rid of analog post filtering altogether seems nice on paper (and I think isn't unique to this particular design), but the potential RFI being passed through unfettered at such high sample rates kind of makes me cringe...... Not to mention potential havoc it might cause with some solid state electronics.
nothing 'nice on paper' about it. it is the real deal. it is by far the quietest, cleanest, most explosive redbook i have heard. vocals, pianos, any strings, horns, all sound great. super quiet backgrounds. amazing dynamic energy. yet zero edge or harshness.
i compare it favorably to state of the art dsd, vinyl and RTR tape in my system and it holds it's own like redbook never has before.
i realize that skepticism is the stock in trade of many. guilty till proven innocent when you can't connect the dots of your personal experience or technical understanding. benefit of the doubt is for others...not you.
here are 2 links which might help.....or not.
more detailed explanation....
http://www.whatsbestforum.com/showthread.php?12023-Trinity-DAC&p=238265&viewfull=1#post238265
how Trinity is different....
http://www.whatsbestforum.com/showthread.php?12023-Trinity-DAC&p=244005&viewfull=1#post244005
sorry, don't know how to post links in audioasylem
YMMV, just my 2 cents, and all that.
mikel
Fifty-eight grand..... (According to The Absolute Sound report... Link below.) Didn't realize the price of this was in the ionosphere..... It *should* be the best DAC in the world............
This is a product you could charge admission for people to listen.....
that article references the 'old' Trinity dac which was a 3 chassis affair and is no longer made.the newer one I have is a single chassis and it came out in late 2013.
I really don't know the exact list price as it's not imported into North America. it is around that same price area though.
I don't charge admission. you would be welcome to visit anytime to hear it. it is worth hearing to anyone who has an interest in the best possible redbook or PCM.
cheers,
mikel
Edits: 10/08/15
I wonder if this product was at RMAF........
Todd,
the Trinity dac was not at RMAF.
Trinity is a small German company with limited capacity. they are a huge deal in HK, particularly as it's a redbook/PCM centric market, and that market does absorb much of their output. with the labor intensive process for the dac they can only produce 3 a month.
at this point they sell direct into the North American market. they did have a USA distributor but do not currently.
I sort of stumbled onto them researching the ultimate PCM dac. the more I looked and researched and read, the more I was intrigued. I bought mine without ever hearing it. and it surpassed my expectations.
mikel
What are you listening to DSD with. I know the Trinity Dac does not do DSD. I have the Audio-GD Master 7 which uses the same 1704 chips. I have only heard DSD downloads on my Mytek Stereo 192. PCM on my Master 7 is wbetter than DSD on the Mytek. I have never liked DSD or SACD's
Alan
as of this moment I'm using both the Playback Design MPS-5 and the Lampizator Golden Gate dsd 'only' dac. the MPS-5 is a pure dsd dac which does not convert dsd to PCM. same with the GG.
the MPS-5 and the GG both do 2xdsd; I have 2-3 terabytes of 2xdsd.
the GG also does Quad dsd, although I don't yet have any Quad dsd files. I've only had the GG since Sunday; it is still breaking in.
mikel
Mike,
You'll have to let us know about the Lampi DSD DAC...I'm thinking of going the same route.
Thanks.
Vbr,
Sam
The best 16/44.1 playback I've heard doesn't just sound better than the best 24/192 I've heard, it trounces 24/192..... Same goes for 16/44.1 vs SACD, for that matter..... I could never go more than half an hour listening high rez, without feeling downright sick.......
This is the main reason why I've never migrated to higher resolution digital audio formats. Vinyl and CD are my main formats, and likely to remain that way 'til I die.
you just need a better reference for higher rez PCM.....
mikel
Trolling?
Christ; I just posted a article a friend sent me.
Easy - not everything is, or should be a pissing match.
Meat; It's the right thing to do. Romans 14:2
Where to start?
Bandwidth limited input signal? Of course the digital system is perfect.
But it's not perfect vs. the un-bandwidth limited input signal, only the bandwidth limited signal.
He never even talks about phase shift in the band pass caused by the analog anti-aliasing filter or how those phase shifts would be moved up, further away from what we can hear, if the sampling rate was increased.
"An anti-aliasing filter (AAF) is a filter used BEFORE a signal sampler to restrict the bandwidth of a signal to approximately or completely satisfy the sampling theorem over the band of interest."
He says the only thing a higher bit rate is good for is lowering the noise floor?????
16 bit is good for 65536 different signal levels.
24 bit is good for 16,777,216 different signal levels.
So is he saying that sample amplitude accuracy is only important to the noise floor?
Who is this guy?
Tre'
Have Fun and Enjoy the Music
"Still Working the Problem"
Bandwidth limited input signal? Of course the digital system is perfect.
But it's not perfect vs. the un-bandwidth limited input signal, only the bandwidth limited signal.
Exactly, which introduces the possibility of hearing the effect of the anti-aliasing filter. Of course, he would then tell you that it's impossible to hear a brick wall anti-aliasing filter because the cutoff frequency is above the magic 20 KHz. But yet we hear differences between different converters. And he would say no, you don't, and demand a rigorously conducted DBT that proves it. And so goes the cycle of debate.
He never even talks about phase shift in the band pass caused by the analog anti-aliasing filter or how those phase shifts would be moved up, further away from what we can hear, if the sampling rate was increased.
Those phase shifts don't really exist in modern oversampling ADCs because the anti-aliasing filter is a FIR type. Nevertheless, I think every audiophile who has experienced a lot of different digital audio products over the years would agree that every anti-aliasing and reconstruction filter sounds a little bit different. It may not be the phase shift, but there is something about steep filtering in or near our hearing range that sounds artificial to our ears.
So is he saying that sample amplitude accuracy is only important to the noise floor?
As long as there is dithering, he is correct on that point. Without dithering, the quantization errors are correlated with the signal and become distortion. With dithering, the quantization errors are decorrelated from the signal and become noise.
"Exactly, which introduces the possibility of hearing the effect of the anti-aliasing filter. "
"Those phase shifts don't really exist in modern oversampling ADCs because the anti-aliasing filter is a FIR type."
I didn't even realize that these filters were active filters. (I assume they have to be to be phase accurate)
So I don't take hearing the effects of the filter as a possibility, I take it as a certainty. Not the filter, but the op-amp.
I am amazed that CD's sound as good as they do.
Tre'
Have Fun and Enjoy the Music
"Still Working the Problem"
> "Not the filter, but the op-amp."
Would you explain that some please?
--------------------------------------------------------------------
Big speakers and little amps blew my mind!
Sure, I don't like the sound of op-amps.
I am assuming that the active low pass filter (anti-aliasing filter) is built around op-amps.
Tre'
Have Fun and Enjoy the Music
"Still Working the Problem"
An oversampling ADC really has two anti-aliasing filters. The primary one is a digital filter, not an active analog filter.
A typical modern ADC channel might be capturing data from a mic feed at 24-bit/96 KHz using 8x oversampling, which means the actual sample rate that it's operating at internally is 96x8 = 768 KHz. It needs to have an analog anti-aliasing filter at the input to block anything above fs/2 from going into the modulator, but with such a high sample rate this can be a gentle RF filter since the stop band can start as high as 384 KHz. The modulator incorporates noise shaping, which pushes down the quantization/dither noise within the audio band at the expense of more out of band digital noise. The data is then downsampled from 768 KHz to 96 KHz at the output of the ADC, eliminating that out of band noise. Downsampling requires an anti-aliasing filter, which is implemented entirely in the digital domain using a linear phase (FIR) filter with a transition band between 40-48KHz. This digital filter is the primary anti-aliasing filter for the ADC, not the analog filter at the input for rejecting RF.
The digital data would then go through the whole production process at 24/96 before it gets downsampled to 16/44.1 at the very end. The downsampling process includes another anti-aliasing filter, again a linear phase digital filter, but this one will have a brick wall response between 20-22 KHz. This last filter is going to be the one most likely to have audible artifacts.
I guess I don't understand "quantization".
It seems to me the higher the bit rate the less quantization is needed and less dithering to keep the distortion low so less noise?
Tre'
Have Fun and Enjoy the Music
"Still Working the Problem"
"It seems to me the higher the bit rate the less quantization is needed and less dithering to keep the distortion low so less noise?"The word length also reduces quantization.
What dither noise does is make the least-significant bit (LSB) switch randomly to signal levels or resolutions at amplitudes less than the LSB. When played back, with classical "sinc" filtering, the resultant level of the analog signal is close to the original signal below the LSB. Without the dither, the LSB would not be triggered randomly at all, the quantization error would then increase as a result.
With longer word lengths (ex: 20 bits instead of 16), with the extra bits, the levels "below the LSB" are now handled by those extra bits, the low level signal is tracked more accurately than without the extra bits and using dither.
Edits: 10/08/15
It seems to me the higher the bit rate the less quantization is needed and less dithering to keep the distortion low so less noise?
Right.
With truncation, the magnitude of the quantization error is < 1 LSB, and with rounding, it's < LSB/2. And to linearize the quantization process, the required dither is at the level of the LSB. The smaller the LSB, the lower the dither noise. More bits = lower digital noise floor.
The truth is I don't understand what the quantization process is.
How can the amplitude of each sample be made correct without an infinite number of bits?
How does the system know how much to "fudge" (and in which direction) the amplitude of each sample?
Thanks.
Tre'
Have Fun and Enjoy the Music
"Still Working the Problem"
Let's see if I have this right.To me the quantization process is not a process at all, it's the outcome of a less than perfect system.
When you digitize an analog signal, while using a word length that is less than infinite, quantization takes place and creates distortion.
Dithering will lower the harmonic distortion orders down into an increasing noise floor than has been deemed to be acceptable.
So now back to what I was trying to say.
In absolute terms, the higher the bit rate the less, distortion causing, amplitude inaccuracies there will be.
It begs the question, how high of a bit rate would it take to listen to the stream without dithering while having acceptable HD numbers?
Tre'
Have Fun and Enjoy the Music
"Still Working the Problem"
Edits: 10/08/15 10/08/15
Let me give you the long answer.The analog signal that you are sampling contains the musical signal S(t) plus an analog error signal consisting of distortion D(t) plus noise N(t). Digital sampling without dither turns this into sequence of quantized sample values x(i) which encode the analog signal plus error value (S+D+N) at time t(i) along with the addition of a second error signal Q(i). The quantization error Q is the difference between the value of S+D+N and the nearest digitally representable signal level. So x(i) = S(t(i)) + D(t(i)) + N(t(i)) + Q(i).
Because Q is a residual of rounding the sum of S+D+N to the nearest digital value, it is correlated to the sum of S+D+N. The spectrum of Q will depend on the spectra of those components and on how well it correlates with each of those S,D,N components, which in turn depends on the level of those components relative to Q. For example, if S> D> N> Q or S> N> Q> D, then Q will be mainly correlated with N and will have a noise-like spectrum. Since it's lower in level than N, it's buried in the analog noise floor and you can consider the sampling process to be effectively transparent. On the other hand, if S> Q> D> N or S> Q> N> D, then Q will be mainly correlated to the signal S. In that case, it will be a type of distortion whose spectrum is a function of the signal and the sampling rate. Since it is also above the analog noise floor N, the sampling process cannot be considered transparent.
For 16-bit sampling, Q is at the -96 dB level, which may be above or below the analog noise floor depending on the nature of the input signal (tape, mic feed, MIDI, etc.) So there are cases involving low noise input sources where 16-bit sampling without dithering can produce a spectrum of distortion products that many people agree is audible and everyone agrees is undesirable.
Dithering in analog to digital conversion is the intentional mixing of a pseudo-random analog signal into the input of the sampler at a level slightly above Q. So x(i) = S(t(i)) + D(t(i)) + N(t(i)) + dither(t) + Q(i). A typical dither signal used in audio is at the level of +/- 1 LSB and has a triangle PDF. It has a flat white-noise like spectrum. It's level is 3 dB above Q, so Q will end up being correlated with the dither rather than the signal, and therefore Q will have a white-noise like spectrum. The tradeoff is slightly more noise but no added distortion.
A second benefit of dithering is that periodic signals which are below the quantization noise floor are preserved in the sampled data and can extracted by time averaging. Without dithering, in the cases where Q> N, those signals are turned into distortion by the quantization process and lost.
For 24-bit sampling, Q is at the -144 dB level, which is pretty much guaranteed to be below the analog noise floor of any current recording equipment. So dithering is optional in a 24-bit ADC but it is usually done anyway. However, if you take 24-bit data and reduce it to 16-bits to fit on a CD, you still need to dither (in a digital form) to avoid introducing distortion. More generally, any process that results in a reduction in bit depth should also include dithering. Many operations performed in the digital domain by a mixing or mastering engineer require multiplication of digital values, a process which results in a longer word length than the input (e.g. multiplying two 24-bit numbers can product a result that is up to 48 bits in length). These results have to dithered rather than simply truncated or rounded to fit back into 24 bits. So most digital audio operations involve dithering of intermediate results.
To finally answer your question: With a minimalist production chain, you could conceivably record, produce, and release at 24-bits with no dither used at all. But the more processing you perform, the greater the likelihood of some quantization distortion reaching an audible level. Which is one of the reasons why dithering is a de facto standard in modern digital audio regardless of the end user delivery format.
Edits: 10/09/15
This answer and your other answer plus some more reading on my part and I now have a different understanding of the whole subject.
I am now left with the question, why doesn't digital sound better than it does?
I used to work at a recording studio. We had a Ampex ATR 102. We also had a 24/96 digital system using Apogee I/O's (circa 2005).
Comparing the live mic feed to the ATR (properly aligned using 456 at +3 and 30ips no Dolby) was almost indistinguishable.
The digital never did that. At least not to my ears.
If the blame doesn't belong where I was placing it....where does it belong?
Whatever, I'm going to go play an old, made from a analog master tape, vinyl record. :-)
BTW I just read the link below, circa 1998. 24/96 digital to analog converter, 8x oversampling filter.
Tre'
Have Fun and Enjoy the Music
"Still Working the Problem"
I am now left with the question, why doesn't digital sound better than it does?
...
If the blame doesn't belong where I was placing it....where does it belong?
An issue with the CD format in particular is that the sample rate is too low.
And a general issue with digital audio is timing, i.e. jitter.
Beyond that, it's probably integrated circuit design. The sonic performance of audio op-amps is all over the map, despite having similar specifications. Same for ADCs and DACs.
Some will blame RFI. I'm less convinced of that.
.
Have Fun and Enjoy the Music
"Still Working the Problem"
+1 ! Thanks, Dave K.
That was an excellent summary/description of the main important points/concepts of digitizing a signal.
Much of what you wrote and explained is stuff which I knew, but you pulled it all together into one tidy little package, like few people can do. Great job on your post!
Higher sample rates yield a more "focussed" digital image of the analog signal, and higher bit depths yield a more accurate representation of "soft to loud". Quantization error makes perfect sense to me. But dither has been a question which you've helped to explain.
It's not surprising that pro studios use high bit rates and high bit depths when recording tracks - there's often a lot of processing afterward.
:)
:)
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