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Let's develop an ASR inter-sample test procedure for DACs!

Nice signal, and it is waiting here long time. But we need a signal which Amir can easily integrate to his testsuite.
 
Should not you ask him first if he is interested in such kind of testing? ;)

Topping DX5 clipping:
DX5_clipping.png


Clipping disappears when digital volume control knob is set to -2dBFS.

But, when oversampling (48kHz==>96Hz) is used to the original signal, then the digital attenuation does not help.
oversampled.png
 
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Would be nice to also see the FFT for each of the clipped signals, showing the distortion products in a more readable way
 
But, when oversampling (48kHz==>96Hz) is used to the original signal, then the digital attenuation does not help.
Software resampling should be used with software volume control. This means the 2dB (if enough) reduction should also be done with software before sending to the DAC instead of using the hardware volume control on the DAC.

For example the TI PCM1794 chip on my soundcard does not support hardware volume control, I always do it within the playback software to prevent clipping.
 
Software resampling should be used with software volume control.

This is certainly the "gold standard", however, not everybody uses a PC for music playback. I try to avoid it if possible, as I find it much more convenient to use hardware volume control than to use a volume slider in jRiver MC...
 
This is certainly the "gold standard", however, not everybody uses a PC for music playback. I try to avoid it if possible, as I find it much more convenient to use hardware volume control than to use a volume slider in jRiver MC...

The software volume control can be assigned to a physical device (e.g. a physical knob) if you don't like to use a mouse to drag a slider:
I am currently using two physical volume controls with my JRiver-based system. One is USB connected to the host PC and the other is a BT knob. Both will adjust the 64bit VC in Jriver or the VC in my exaSound e38 DAC. Both send Volume up/down signals when rotated so the physical position of the knobs is irrelevant as is what you do with them when the system is not up and running. As a result, after powering down intentionally or consequent to power failures, the system starts up where it left off.

When software resampling is used without also using suitable amount of software attenuation, the signal is already clipped in software domain before entering external hardware. This article explained it.

Here are some illustrations showing software domain clipping on the software resampler output using integer output format:
 
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@Kal Rubinson: could you provide more background infos about the volume knobs you are using with jRiver MC? Like manufacturer, part number. Is any specific software required to get it working with jRiver Mac? See previous post.

Thank you in advance!

Edit: Just saw that you are using the Microsoft Surface Dial, which can be configured to send keyboard shortcuts as required by jRiver MC. Will consider adding such a device to my system.

Edit2: In another context the Contour Multimedia Controller Pro v2 was mentioned (used by Bob Katz for controlling Acourate Convolver). There is also a smaller version, the Contour Multimedia Controller XpressMusic. Both use USB cable connection.
 
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Some additional thought about this topic beforehand:

Intersample peaks do not only occur in the audio domain, but actually should be something to be considered also for video, which at least theoretically may be described as the same phenomenon, just in two dimensions.

In the same way, reconstructed audio may have a higher level then it's "surrounding" sample points, so may pixels (which are the equivalent) call for e.g. brighter levels than their digital values suggest. Especially when images are scaled, transcoded, etc. I could imagine it to happen and I better don't ask if that is regularity accounted for, fearing to already know the answer.

For some strange reason, what is entirely understood and practically handled such as filtering for aliasing, isn't followed when it comes to images or video. How many tools have crappy downscaling algorithms full of aliasing? Yes!

However, many DACs (even excellent and/or competently designed ones) either have no digital volume control or have analogue volume controls that operate at the end of the signal chain.
Like Benchmark's most previous baby "DAC1" back in the days.

Ok, since many people make the argument that "ISP overs are simply a result of bad mixing/mastering"[...]
I would formulate it as "ISP overs are unnecessarily provoked by questionable mixing/masterings" (but should be rendered correctly nonetheless).

ISPs are an entirely digital phenomenon.
Are they? My understanding is that one - at least in test scenarios - may have samples at 0dBFS with peaks in between, calling for analog reconstruction at levels above the 0dBFS equivalent, without using any oversampling as well. One could argue that it's the reconstruction of that analog signal calling for the additional headroom*.

* A devil's advocate argument which comes to mind could be that arbitrarily high intersample peaks may be "artificially" constructed raising the question whether any amount of headroom will ever be enough. In that sense, there can be no "perfect" DAC.


You're arguing for DACs to fix a problem that is caused by mastering.
I think that can be righteously argued from both sides. On one hand, especially nowadays' mixing and mastering on average sadly is as bad as virtually never before, where the irony of technical capabilities and actual use and interest from the average music joe is staggering to say the least. On the other, one could imagine a perfectly fine highly dynamic master still containing intersample peaks for whatever reason. Since it is technical a valid input for a DAC, it should be rendered correctly.

Technically, a DAC doesn't "fix" such masterings but instead doesn't make them even worse. Again, one could argue here that in many cases the source it already so bad that it doesn't matter anymore anyway.

Valid PCM data that represents a waveform that exceeds the limits specified in the format is not valid. PCM is not the end-goal, the actual waveform reproduction is.
That's indeed a matter of standardization and definition. One pro argument to allow intersample peaks however would be that this way, one makes use of the full SNR a certain bit depth allows without the need to have it lower by constraining yourself in the "digital domain".

In other words, if the headroom is permanently built into the DAC, then you lose that dynamic range for all uses, not just in the case of intersample overs.
However, doesn't the SNR of most DACs nowadays exceed the one of the sources anyway?

We eagerly read reviews comparing -98 to -112dB SINAD but yet another test of inaudible technical flaws is dumb and bad for some reason?
Excellent point. What is practically required has been long exceeded even in this forum, nominally not being exactly voodoo-affine.

I have activated the optional 3dB attenuation prior to SRC in my RME ADI-2, and in my opinion this should be a standard feature.
What is really likeable about RME is that they care about the details, such as dynamic loudness or manually activatable deemphasis.

I'm all for expanding testing where it is important, but this is a non-issue and is solely caused by the content being poorly recorded in the first place.
That point may be seen from two different angles and neither I am sure which one to be preferred. One can argue that such inputs containing intersample peaks still contain valid data and of course they do as nominally, all bit combinations are allowed.

But even under the premise that it is a non-issue - so are SINAD differences in the region of dB-fractions which are also lively discussed here. At the end, I find a DAC's behavior to those intersample peaks even more interesting than a few dB of noise or distortion here and there.

It's not up to D/A converters to "fix" mastering issues, especially at the expense of ultimate performance. What's next? 6/9/12dB headroom? Should the D/A converters fix channel imbalances too?
On the other hand, even if one reserves 12dB of headroom, the SNR of most products would only go from "totally insane" to "still insane". I mean, most figures are below what we will ever be able to hear anyway so how much damage is caused by turning the wheel academically even further? One might sassily ask what is worse - having mostly inaudible intersample clipping or mostly inaudible noise?

  1. This is not a mistake in production, it is a mistake in the playback hardware.
  2. We didn't have this problem until oversampled DACs were invented.
Correct me if I'm wrong, but neither the Benchmark DAC1 had such an overhead implemented and since it's also over/undersampling (to something like ~ 110 kHz if I halfway remember that correctly), it should run into similar issues.

Without intending to sound provoking or too cocky, but out of interest: why didn't Benchmark already implement such headroom in the DAC1 already?

My impression is that this whole topic was even a more niche one back in the days of the DAC1 and "discovered" by the scene later on and now of course also for Benchmark is a welcome marketing point.


Intersample overs may occur many times per second, even on a well recorded track. For example, we found 1129 in the 5-minitue long Steely Dan Gaslighting Abbie track. This is about 4 per second and this is not unusual, nor is this track an extreme example. This track is fairly typical.
Yep, that one seems to be yours/Benchmark's "favorite". :)

Mind you, the DVD audio to me looks and sounds even better. If you take the 5.1 and create a downmix to stereo, you will have an even more dynamic result without intersample clipping, showing that as usual, despite the otherwise very good production quality, the official CD release has been unnecessarily limited compared to the multi-channel DVD.


A digital volume control on a DAC may or may not eliminate the intersample clipping problem when turned down. If an ASRC, an SRC, or an interpolator precede the digital volume control, then the distortion will not be eliminated when the volume is turned down.
Maybe impertinent to ask for your opinion on a competitior's product, but from a technical/scientifical standpoint, given your enviable technical expertise, how would you judge RME's implementation of volume control in their ADI DACs, especially in regard to the prevention of intersample clipping?

ASR ranks DACs and power amplifiers by SINAD to levels that are purely academic. At the upper end of the chart, the distortion and noise will be lower than 0 dB SPL in the listening room. In other words, the distortion and noise is absolutely inaudible because it is below the threshold of hearing. I couldn't be heard even if the music was not playing. This means that a significant portion of the ASR rankings may represent measurable but inaudible differences, but as readers, we appreciate this information.
This for me is the core argument why the behavior of intersample clipping is something which should be tested for.

It's just inconvenient yet as it's something new and it's very human to question it at first, but once a testing routine has been established, it should cause much more effort than testing for the usual SINAD fuss, we all got used to by now and also for sure has been something exotic back then.

Same goes for testing against the deemphasis support, which I've raised very recently.

The intersample peak clipping problem may be the single most audible defect in modern PCM DACs and DSP devices. I would not write it off as an "inaudible defect that only impacts poorly mastered recordings".
And yet ... the DAC1 also got away with(out) it, critically acclaimed nonetheless.
 
Interesting measurements of Benchmark DAC3

Archimago on IS overs:
 
Yes, especially Archimago seems to be a decent and "down to earth" source whereas GoldenSound for me is a somewhat confusing case and mixed bag.

On one hand, that guy on Youtube with his very pleasant voice and eloquence seems to know his field, having tremendous knowledge on the subject, including the intersample peak niche, one the other, his videos are also full of nothing but voodoo nonesense where he seriously compares the sound of DACs (which is questionable to begin with as in most cases only possible in borderline cases while crunching your teeth during comparison, if at all) and best of all, presented in principle here through the viewer's DAC while watching it, which doesn't make any sense to even begin with and is pure satire.

Guess the latter is a simple result of "bumping the click rate" as - I have to admit it - that voodoo stuff is somewhat entertaining despite knowing better.
 
Like Benchmark's most previous baby "DAC1" back in the days.
Which is where they must have learned the hard way about these things - that model also featured an ASRC, and boy do those hate overs. Fortunately this fad dissipated rather quickly after the early 2000s.
 
Yes, especially Archimago seems to be a decent and "down to earth" source whereas GoldenSound for me is a somewhat confusing case and mixed bag.

On one hand, that guy on Youtube with his very pleasant voice and eloquence seems to know his field, having tremendous knowledge on the subject, including the intersample peak niche, one the other, his videos are also full of nothing but voodoo nonesense where he seriously compares the sound of DACs (which is questionable to begin with as in most cases only possible in borderline cases while crunching your teeth during comparison, if at all) and best of all, presented in principle here through the viewer's DAC while watching it, which doesn't make any sense to even begin with and is pure satire.

Guess the latter is a simple result of "bumping the click rate" as - I have to admit it - that voodoo stuff is somewhat entertaining despite knowing better.
I think GoldenSound has more or less stopped doing subjective reviews of DACs now. He seems to have had a change of heart on the matter and now considers measurements to be king.
 
On one hand, that guy on Youtube with his very pleasant voice and eloquence seems to know his field, having tremendous knowledge on the subject, including the intersample peak niche, one the other, his videos are also full of nothing but voodoo nonesense where he seriously compares the sound of DACs (which is questionable to begin with as in most cases only possible in borderline cases while crunching your teeth during comparison
I am not trying to judge him and it is really not my intention. In the specific article of him that I have linked above I can see good technical approach and that is the reason why I posted the link. I do not prefer black and white approach.
 
Neither do I still fully understand why Benchmark opted for using a potentiometer (prone to wear and eventually noise or even distortion) for the analog volume control in the DAC1. Even at that time, there were already DSP- and IC-based ones available - as commonly used by AVRs. If I remember correctly, they didn't want to use ICs due to concerns about their quality (fair enough), but neither DSPs due to their SNR reduction (academically comprehensible but worth being discussed).

I am very well aware of John Siau's explanation which goes like "analog volume reduction preserves the SNR while a DSP based one before a DAC chip by principle doesn't". While that is technically true, I still fail to see the practical relevance as I as the listener will lose that part of SNR either way as my ear by definition also will only achieve the full dynamic range when utilizing the threshold of unpleasantness (which probably isn't healthy anyway). As soon as I lower the volume, I'm also left with only my fraction of dynamic range and SNR so I don't see why the technical loss should matter that much. I think RME argued in a similar way once, seeing DSP-based volume controls despite the technically reduced SNR as the best solution of all.

Well, meanwhile Benchmark also uses a DSP-based solution for the volume or actually "gain" control (although in a bit more traditional way with a fixed max and min and no "rotary" control as Amir put it), at least for digital sources and as mentioned in the article, besides taking any mechanical wear out of the equation, they also have the advantage to elegantly circumvent the intersample peak issue alongside as one rarely will use the DAC at its full volume setting but way further down than -3dBFS so the headroom for intersample peaks will be huge in practise.

Still, it would be interesting to know how big the headroom is at the 0dB setting of today's DACs (when used in conjunction with an amplifier and not the maybe built in headphone output) and also how they implement any SRC which I guess is the crucial part if happening before* the volume control.

* if I understand it correctly, usually a sample rate conversation is done before the volume control where e.g. RME offers an optional fixed headroom of 3dB. What in general speaks against doing such SRCs after the volume control? Guess nominally a loss in precision for the math, but how relevant is that in practise if properly dithered in between, I wonder.
 
Neither to I, hence my shared "mixed feelings" as he and his channel definitely also has very good aspects.

Benchmark is in the business of selling D/A converters and other audio products- don't forget that.

It's not unexpected that any of their commentary, "discoveries", educational white papers and design choices would support whatever direction they are taking at this point in time. Of course, all that is subject to change going forward.

The construction quality of their products is outstanding in my opinion. Totally fit for purpose. Not inexpensive, but not unreasonable either.

Just don't believe everything they say is gospel.
 
The story had a happy ending - I emailed the Australian record label about the issue and they generously shared with me their WAV files of all the tracks on the CD, which did not have this problem. They were otherwise bit-identical to the CD - peaks at 0.0, identical waveform.

So the original .wav files they sent you were 16/44 or something else (16/48)? Just wondering if it was the effects of a SRC at the mastering for CD/disc manufacturing end.
 
Just to make sure - in my comment about the "mixed feelings" I was entirely referring to GoldenSound, not Benchmark.

I get that. Benchmark were of course on message, prosecuting their narrative in the comments/responses to GS's review linked.

They have a pattern. It probably works for them and likely for many potential customers of theirs.
 
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