Excluding hi-res and HD, why use a DAC with processing capabilities greater than 16 / 44.1?

[Mike Mcsweeney]

The upsampling is done on the fly. It's in what JRiver calls the DSP Studio. You can output any file at any sample rate from PCM 44.1 to DSD128. Even mp3's.

 

You can access it with the little button to the left of the search box with an icon that looks like an equalizer. Or you can access it in the menus with this path:

 

Tools | Options | Audio | Settings | DSP & Output Format.

 

I use DSD upsampling on my Loki and LampizatOr DACs. It works amazingly well.

 

hmmm...i will send you private email...it says not compatible hardware, but i am pretty sure it is...maybe i need to configure the hardware somewhere else in jriver first...

sony uda-1/b

[Mike Mcsweeney]

hmmm...i will send you private email...it says not compatible hardware, but i am pretty sure it is...maybe i need to configure the hardware somewhere else in jriver first...

sony uda-1/b

nvm...i found it in audio device...asio for sony dac device

 

instantly much louder and deeper bass and more space, but some songs sound distorted...must require some more tweeks...but i can see how this can greatly improve things. the hi-res sony media player that came with the dac has VERY LIMITED features, but does seem to work nicely without any tweeks..well, plenty more to play with this weekend than i anticipated....

[pgawan2b]

I'm very surprised not to have seen the best reason for purchasing a DAC that accepts hi res.

 

Virtually every DAC, with only a handful of exceptions, upsamples to 352.8/384kHz, then sends that upsampled output to a delta-sigma modulator, before converting that bitstream to analog. Software these days is capable of doing a better job of this than internal DAC chips. So the closer you can get to 352.8/384kHz or DSD at your DAC input, the more of this upsampling you can do in software, avoiding the inferior job done internally.

 

Jud, I hadn't heard this argument before. By software, are you referring to software incorporated in the newer DACs or the software run by the computer during playback?

[Jud]

I'm very surprised not to have seen the best reason for purchasing a DAC that accepts hi res.

 

Virtually every DAC, with only a handful of exceptions, upsamples to 352.8/384kHz, then sends that upsampled output to a delta-sigma modulator, before converting that bitstream to analog. Software these days is capable of doing a better job of this than internal DAC chips. So the closer you can get to 352.8/384kHz or DSD at your DAC input, the more of this upsampling you can do in software, avoiding the inferior job done internally.

 

using that logic and reasoning, wouldn't it also be beneficial to have a software/dac combo that upsamples to 5.6mhz? not being sarcastic, just wanting your opinion since i believe you were one that was adverse to the higher sampling?

 

Hi Mike. Not for or against higher sampling by itself. I favor two things:

 

- The minimum possible number of conversions in the chain from recording to listening

 

- Where conversions must be done, doing them as well as possible

 

Very often, though not always, the earlier any conversions are done, the better. Thus better at the studio than by the listener; and better in the listener's PC than in the DAC.

[Jud]

I'm very surprised not to have seen the best reason for purchasing a DAC that accepts hi res.

 

Virtually every DAC, with only a handful of exceptions, upsamples to 352.8/384kHz, then sends that upsampled output to a delta-sigma modulator, before converting that bitstream to analog. Software these days is capable of doing a better job of this than internal DAC chips. So the closer you can get to 352.8/384kHz or DSD at your DAC input, the more of this upsampling you can do in software, avoiding the inferior job done internally.

 

Jud, I hadn't heard this argument before. By software, are you referring to software incorporated in the newer DACs or the software run by the computer during playback?

 

Software run by the computer. The closer to a 352.8/384 (or DSD, for capable DACs) sample rate the DAC can accept, the more this can be taken care of by studio equipment or a program running on a PC, instead of a $5 DAC chip.

[wakibaki]

There's only one good reason for having >16/44k1, that's if you use some kind of software volume control, where having 24-bit resolution can mean that you can turn down the volume in software without any loss of true resolution.

 

I never use software volume control, I always output maximum volume from the computer and turn it down using a pot, hence I see no need for a DAC with >16/44k1.

 

Up sampling 16/44k1 to send it to the DAC is crazy (from a scientific POV, that's the engineering position).

 

w

[Paul R]

There's only one good reason for having >16/44k1, that's if you use some kind of software volume control, where having 24-bit resolution can mean that you can turn down the volume in software without any loss of true resolution.

 

I never use software volume control, I always output maximum volume from the computer and turn it down using a pot, hence I see no need for a DAC with >16/44k1.

 

Up sampling 16/44k1 to send it to the DAC is crazy (from a scientific POV, that's the engineering position).

 

w

 

That is your opinion of course, but it is very far from a proven or a generally accepted fact.

 

It is pretty obvious that a lot of people think, for a lot of different reasons, some more sound than others, that Hi Res, up sampled, transcoded, or other wise manipulated signals can sound better. And they require a DAC able to accept and process signals higher than 16/44.1.

 

And a whole lot of those people are very scientific audio engineers. Pretty much all of them making DACs today. That is not even considering the mastering engineers, artists, and the great number of consumers who also disagree with you.

[k6davis]

I'm very surprised not to have seen the best reason for purchasing a DAC that accepts hi res.

 

Virtually every DAC, with only a handful of exceptions, upsamples to 352.8/384kHz, then sends that upsampled output to a delta-sigma modulator, before converting that bitstream to analog. Software these days is capable of doing a better job of this than internal DAC chips. So the closer you can get to 352.8/384kHz or DSD at your DAC input, the more of this upsampling you can do in software, avoiding the inferior job done internally.

 

Your point is correct Jed, but it doesn't apply to the original poster's question.

 

If I understand correctly, he is considering purchasing a DAC that accepts a maximum of 16-44 because he'll only be listening to 16-44 content.

 

My advice was that a more modern, HD DAC would provide better conversion of 16-44 content, and yes, I agree that the software upsampling in the server is usually superior to upsampling in the DAC.

[wakibaki]

That is your opinion of course, but it is very far from a proven or a generally accepted fact.

 

It is pretty obvious that a lot of people think, for a lot of different reasons, some more sound than others, that Hi Res, up sampled, transcoded, or other wise manipulated signals can sound better. And they require a DAC able to accept and process signals higher than 16/44.1.

 

And a whole lot of those people are very scientific audio engineers. Pretty much all of them making DACs today. That is not even considering the mastering engineers, artists, and the great number of consumers who also disagree with you.

 

Yes, Paul, but as I refrained from pointing out in the previous thread where you intervened, suggesting that I was merely expressing opinions, you evidently have no understanding of the meaning of provenance where evidence is concerned.

 

The last substantive study of this subject was by Meyer and Moran, as I am sure you are aware, and showed no evidence of a panel of trained listeners showing any ability to distinguish between original 'hi-res' files and those passed through a16/44k1 throttle.

 

If you have something more to offer on this subject than hearsay or induction, I will be glad to examine your evidence, but thus far you have shown me nothing.

 

w

[k6davis]

There's only one good reason for having >16/44k1, that's if you use some kind of software volume control, where having 24-bit resolution can mean that you can turn down the volume in software without any loss of true resolution.

 

I never use software volume control, I always output maximum volume from the computer and turn it down using a pot, hence I see no need for a DAC with >16/44k1.

 

Up sampling 16/44k1 to send it to the DAC is crazy (from a scientific POV, that's the engineering position).

 

w

 

False.

 

There are certain, highly regarded DAC chips (notably the ESS Sabre) that are optimized for the conversion of high resolution signals, which allow for less harmful digital filtering. There are also DAC's which offer DSD performance that is superior to their PCM performance - and vice versa.

 

In fact, now, most DAC's upsample the signals they receive - particularly PCM signals, which even at 24-192 are too low in resolution to maximize the capabilities of the internal hardware of modern DACs.

 

Whether conversion is done in the computer, in the DAC, or not at all, the point is to provide a signal to your DAC's internal circuitry that will produce the best result for your DAC.

 

Proper upsampling of digital audio does not impact the sound of the music, for better or worse. Properly done, mathematical interpolation of data that would have existed between two points in a file will not alter the sound of that file. It does, however, provide a superior starting point for processing in a DAC designed for high resolution decoding.

 

People who have the opinion that upsampling is impure seem to overlook that representing an audio waveform as a PCM stream of 16 bit or 24 bit data words is already a massive abstraction that requires a large overhead of data manipulation and processing to turn into music.

 

I would have thought the excellent performance of current, highly regarded DACs that upsample everything internally (such as the Auralic Vega) would have settled this debate by now.

 

Keith

[wakibaki]

I agree that the software upsampling in the server is usually superior to upsampling in the DAC.

 

What do you base this assertion on? Upsampling is a simple process of zero stuffing and low pass filtering. It's not as though there's much to go wrong. While may be possible to more closely approach brick-wall filtering in software than most DACs will bother with, it will be incontrovertibly inaudible at these frequencies. The design of these filters is so well understood that it can be done by a program (Matlab), and no DAC will be cost-engineered to the point of audibly affecting performance. What would be the point of that? These techniques are employed to achieve as good or better performance at reduced cost, not to pass off an inferior product.

 

Far too much of what is posted on here is predicated on the premise that engineers are mad, and constantly trying to put one over on an unsuspecting public. It's not the engineers who are crazy.

[wakibaki]

k6davis

 

You have totally missed the point. I am talking about upsampling in software, not upsampling in the DAC.

 

w

[k6davis]

k6davis

 

You have totally missed the point. I am talking about upsampling in software, not upsampling in the DAC.

 

w

 

So upsampling using the software in your server is "crazy" but upsampling using the software in your DAC is good?

 

Keith

[wakibaki]

So upsampling using the software in your server is "crazy" but upsampling using the software in your DAC is good?

 

Keith

 

Exactly. What would you want to store a load of redundant, grossly inflated files for?

 

w

[Paul R]

Yes, Paul, but as I refrained from pointing out in the previous thread where you intervened, suggesting that I was merely expressing opinions, you evidently have no understanding of the meaning of provenance where evidence is concerned.

 

The last substantive study of this subject was by Meyer and Moran, as I am sure you are aware, and showed no evidence of a panel of trained listeners showing any ability to distinguish between original 'hi-res' files and those passed through a16/44k1 throttle.

 

If you have something more to offer on this subject than hearsay or induction, I will be glad to examine your evidence, but thus far you have shown me nothing.

 

w

 

Well, just to be clear, I doubt I would engage you to vet or peer review any study or research of mine. You demonstrate an appalling lack of the objectivity needed for that task.

 

Given that though, you are rather sadly out of date in your knowledge or understanding of the issues involved. Or of the studies that have clearly provided different conclusions that what you have choosen to adopt- as your opinion.

 

Nor am I employed as a researcher paid to do your basic research for you.

 

But mostly your opinion is greatly at odds with people in the industry and hobbyists whose opinions I respect.

 

And even the most arrogant of them are not usually given to making such statements of "fact" as you did in the message I originally politely replied to.

 

Seriously, you should consider doing some more research and losing the attitude. Doesn't bother me, but it is not the way to win friends and influence people around here. Even when one has intereting things to say, as I am assured you do.

 

No skin off my back though- no amount of nastiness will get me to do your research for you or gain you the respect you desire. Or get me to engage with you in a foolish conflict of opinions.

[k6davis]

Exactly. What would you want to store a load of redundant, grossly inflated files for?

 

w

 

The files on my hard drive are unchanged.

 

JRiver Media Center 19 upsamples them to DSDx2 in real-time as I play them.

 

The sound, with my DAC, is fantastic.

 

I find it interesting that you, calling people "crazy" with your "scientific POV" and broad knowledge of the supposedly agreed upon "engineering position", aren't aware of the current developments in the hobby, whether you're for or against them.

 

...Even when one has intereting things to say, as I am assured you do.

 

Though I often agree with Paul, I'm not so sure he's right in this case.

[Jud]

Your point is correct Jed, but it doesn't apply to the original poster's question.

 

If I understand correctly, he is considering purchasing a DAC that accepts a maximum of 16-44 because he'll only be listening to 16-44 content.

 

My advice was that a more modern, HD DAC would provide better conversion of 16-44 content, and yes, I agree that the software upsampling in the server is usually superior to upsampling in the DAC.

 

Yes, Paul, but as I refrained from pointing out in the previous thread where you intervened, suggesting that I was merely expressing opinions, you evidently have no understanding of the meaning of provenance where evidence is concerned.

 

The last substantive study of this subject was by Meyer and Moran, as I am sure you are aware, and showed no evidence of a panel of trained listeners showing any ability to distinguish between original 'hi-res' files and those passed through a16/44k1 throttle.

 

If you have something more to offer on this subject than hearsay or induction, I will be glad to examine your evidence, but thus far you have shown me nothing.

 

w

 

Yep, it applies to the original poster's question.

 

wakibaki, intentionally or not you're moving this thread to a place of subjectivity and controversy, and I would much prefer to keep the responses to the OP on solid math and engineering ground.

 

While trying to be thorough in my research and careful in what I say, I am no expert, so if there is anything in the following that is inaccurate I will appreciate corrections.

 

Let's go back to the dawn of the CD. The very first CD players used what are known as "brick wall" filters to convert the digits to music. There were audible problems with this, including aliasing and high levels of harmonic distortion. Very soon, even before the first separate DACs, what is variously called "upsampling," "oversampling," or more properly "interpolation" was used to avoid these problems. "8x oversampling" quickly became the industry standard, meaning that the DAC chip in the CD player or in separate DACs when they started being made first interpolated the 44.1kHz incoming sample rate to 352.8kHz before doing the digital to analog conversion. (This is why the discussion is relevant to the OP's question, since his DAC is overwhelmingly likely to be doing this internally.) Nearly all DAC chips do this internal oversampling of 44.1 material in three "rounds" of doubling - first to 88.2, then 176.4, and finally 352.8.

 

Now let's talk about the two "sides" of the resolution label. When you see something labeled as 16/44.1 or 24/96, the right side has to do with the sample rate. The left side is what's called the "word length." It's how many "bits" are available to denote the loudness of the signal. So 24/96 material coming from a studio theoretically has 8 more "bits" of potential loudness variation (i.e., dynamic range) available. Now there's controversy about how much of a difference these 8 bits can actually make, and I don't propose to get involved in that controversy here because it is irrelevant to the original question. I bring it up only because wakibaki, through either confusion or an imprecise use of terms, brought up the notion of "zero padding." As it is usually used regarding DACs, "zero padding" means simply appending zeros to the word length for purposes of processing 16 or 24 bit material in a 24 or 32 bit internal process. It makes no earthly difference whatever to the sound. It is the equivalent of writing 1 as 1.00000000 - no difference in quantity at all. That sort of process on the left side of the resolution figure is *not* what is interesting or relevant to the OP's question.

 

So now let's return to talking about the interesting side of the resolution figure, the right side, which is the sample rate. We were discussing the "8x oversampling" industry standard that was put in place even before separate DACs. What's so hard about multiplying something 8 times in a chip? Well, for one thing, it isn't multiplication. Interpolation, to use the more proper term, is done by means of filters that use math called Fourier transforms. (The interpolation does use zeros, though not at all in the same unimportant and completely innocuous way as zero padding of the word length.) The thing about Fourier transforms is that they have what are called "conjugate variables." As one of a pair of conjugate variables is optimized or becomes more exactly defined, the other becomes less optimized or less exactly defined. This is sheer mathematics - it's the way Fourier transforms work. In the case of the filters used to do interpolation, time domain properties of the filter (e.g., impulse response) and frequency domain properties (e.g., how good the filter is at removing frequencies above the cutoff point) are conjugate variables. It is thus mathematically impossible to optimize the filters used in virtually all DACs for both time domain and frequency domain response. This necessarily means all interpolation filters are the designer's idea of a good compromise.

 

Most people have no idea how much variability there is between interpolation filters used in different DAC chips. To give you some idea of this, here are impulse response tests of the same software manufacturer's filter using two different settings:

 

 

 

You wouldn't expect to find this level of relative variation in impulse response between two speakers, let alone in your electronics. (These test graphs are from SRC Comparisons , a very nice source of information about the performance of filtering software on various tests. Specifically, they are from the "steep no alias" and "intermediate phase" settings, respectively, of iZotope 64-bit sample rate conversion software, bundled with the Audirvana Plus music player and acknowledged to be among the best such software available.)

 

Now these graphs are from filtering software, which I've mentioned is considered to do a better job than the filters within DAC chips. There are two reasons for this. One is sheer computing power. Though the chips in the very first external DACs competed somewhat favorably regarding computing power with the CPUs in consumer PCs at the time, now the situation is very much reversed. The chip that runs your computer has a huge amount more computing power to throw at this filtering problem than any DAC chip. More computing power means an ability to run more sophisticated filtering algorithms that can do a better job of the inevitable compromise between time domain and frequency domain performance. The second is that the market for filtering software is very competitive in terms of performance, whereas the chips used in DACs are very much a commodity product, perhaps $5 apiece, and external standalone DACs represent a negligible slice of the market for these chips. One other consideration is that filtering software can do 8x oversampling with a single application of the filtering algorithm, in contrast to the three rounds of doubling used in nearly all internal DAC chips.

 

We therefore have some very simple, very practical choices in deciding where we will do our oversampling. Let's say our DAC is like most these days, and accepts a 176.4/192kHz input. If it's available, we could purchase a 176.4/192kHz hi res file, meaning any oversampling would be done at the recording studio on equipment potentially more sophisticated than PC software, and certainly more so than the filtering in the internal DAC chip. Or we could buy the CD and oversample in PC software to 176.4 or 192kHz resolution. The DAC chip would then use one round of its own oversampling filters to raise the sample rate to 352.8 or 384kHz. So we would have substituted either the studio's or the PC software's oversampling filters for two rounds of the filter in the DAC chip. If you have a DAC capable of accepting 352.8 or 384Khz sample rates at input, then you can avoid the internal DAC chip's interpolation filter entirely.

 

The way this relates to the original poster's question is that he asked whether there is a reason to have a DAC that accepts higher than 44.1kHz sample rates if all he is going to do is play CDs or files from CDs. The answer is yes, because:

 

1) Your DAC is internally oversampling to 352.8 or 384kHz.

 

2) This oversampling is not a process capable of mathematical perfection. It involves necessary compromises between time domain and frequency domain performance. These compromises can be done better in software or at the studio than in the $5 commodity chip in your DAC.

 

3) A newer DAC with available 176.4/192 or 352.8/384kHz input allows you to avoid some or all of the internal oversampling done by your DAC chip.

 

(There's still the sigma-delta modulator that has been in nearly all DAC chips for a long time, though not quite as long as 8x oversampling has been around; but this comment is way past long enough already.)

[pgawan2b]

Jud, thanks for taking the time to explain this. It has been helpful for me. I have not moved yet to computer sourced audio; maybe I need to consider this on the basis of sound quality and not merely convenience (although for me, computers are not always convenient).

[Jud]

Jud, thanks for taking the time to explain this. It has been helpful for me. I have not moved yet to computer sourced audio; maybe I need to consider this on the basis of sound quality and not merely convenience (although for me, computers are not always convenient).

 

Thanks, I appreciate the positive feedback. As for moving to computer sourced audio, I suppose like most folks frequenting a forum called Computer Audiophile I feel sound quality is better, but some people report they like the sound from spinning silver disks better. There will always be at least a temporary inconvenience when you move from something you know to something you don't know as well, but fortunately there's no rush. You can make the change in your own good time, or decide not to, just as you prefer.

[Paul R]

Jud- this is written very well indeed. Might I suggest you put it into one of your blogs if it does not already exist there please?

 

It deserves to have a permanent place that we can reference in the future.

 

-Paul

 

 

Let's go back to the dawn of the CD. The very first CD players used what are known as "brick wall" filters to convert the digits to music. There were audible problems with this, including aliasing and high levels of harmonic distortion. Very soon, even before the first separate DACs, what is variously called "upsampling," "oversampling," or more properly "interpolation" was used to avoid these problems. "8x oversampling" quickly became the industry standard, meaning that the DAC chip in the CD player or in separate DACs when they started being made first interpolated the 44.1kHz incoming sample rate to 352.8kHz before doing the digital to analog conversion. (This is why the discussion is relevant to the OP's question, since his DAC is overwhelmingly likely to be doing this internally.) Nearly all DAC chips do this internal oversampling of 44.1 material in three "rounds" of doubling - first to 88.2, then 176.4, and finally 352.8.

...

...

...

[k6davis]

Jud- this is written very well indeed. Might I suggest you put it into one of your blogs if it does not already exist there please?

 

It deserves to have a permanent place that we can reference in the future.

 

-Paul

 

+1

 

Your detailed post was very helpful Jud.

[The Computer Audiophile]

I use an Oppo 980H as my source, and I play only redbook CDs. Since redbook CDs are recorded at 16 / 44.1, of what value is a DAC whose processing capability is greater than 16 / 44.1? Please exclude characteristics such as build quality, brand recognition, re-sale value, and future applications from the discussion. I'm only interested learning what the value add of the processing capability has on redbook CDs. Thanks,

Don't be fooled by the numbers. They mean nothing. You need to listen to a DAC to make a judgement. Some 16/44.1 DACs may sound much better than 24/192 DACs and vice versa.

 

Please take this in the spirit in which it's intended. Your question has no answer.

[Booster MPS]

Jud your response should be further refined to be published on the front page if CA would consider it. This is a great topic given the DAC arms race that is ongoing.

 

This is just my opinion. Great job!

[Jud]

Don't be fooled by the numbers. They mean nothing. You need to listen to a DAC to make a judgement. Some 16/44.1 DACs may sound much better than 24/192 DACs and vice versa.

 

Please take this in the spirit in which it's intended. Your question has no answer.

 

I agree with Chris, with the proviso that what I explained regarding internal versus external upsampling is true, but is only part of the story. A DAC limited to 16/44.1 input may indeed sound better than one with hi res input, if the 16/44.1 DAC has better parts and construction quality and a really well done internal filter programmed into its DAC chip. (That said, technology has absolutely made some progress over the years - I bought a $450 DAC a couple of years ago that far outperformed a 20 year old DAC by the same designer that cost five times as much.)

[k6davis]

Don't be fooled by the numbers. They mean nothing. You need to listen to a DAC to make a judgement. Some 16/44.1 DACs may sound much better than 24/192 DACs and vice versa.

 

Please take this in the spirit in which it's intended. Your question has no answer.

 

I believe the OP's question does have an answer. Should he consider an HD DAC if he'll only be listening to SD material? Yes.

 

But to your point, technological advances in computing power and DSP over the last 10 years have been dramatic. Can you think of a single example of a DAC limited to 16/44 input that "sounds much better" than the modern DACs you give the C.A.S.H. awards to?