Thoughts on the future of HiFi

[theaudiohobby]

If you read what I said clearly, there was a exhibit that went along with the paper. You were allowed into to dedicated listen rooms(just as I described), and the test was administered just as the white paper test was. The conclusions done in the booth were exactly the same as the white paper described, and you could see the measurements taken of the brain activity right after the test was done. It correlated exactly to the measurements outlined in the white paper. It was peer reviewed, which establishes its legitimacy. There has been nothing to dispute this since it was submitted. So it was probably forgotten is nothing more than your theory. It hasn't been challenged is more accurate.

Without the paper, all this is simply handwaving, Accepted theory that establishes a connection between digital encoding and agitation in test subjects would not be hidden away in a single AES paper.

Let's start with the waveform. Once a analog signal is encoded to 16bit resolution, it appears as a stair stepped waveform, unlike a recorded analog which is more rounded

You got this wrong from get-go, a DAC does not see a waveform in the sense you use the term, it recovers a digital datastream of some sort PCM, DSM or whatever from the carrier signal at it's input. The reconstruction filter reconstructs the orignal analog waveform from the datastream recovered at DAC input.

Remember, there are two chief distinctions between an analog and a digital signal. The first is that the analog signal is continuous in time, meaning that it varies smoothly over time no matter how short a time period you consider(the rounded waveform), whereas the digital signal, in contrast, is discrete in time, meaning it has distinct parts that follow one after another with definite, unambiguous division points (called signal transitions) between them.(the square tooth waveform).

As stated previously, a DAC recovers a datastream at it's input, You've conflated the quantization process and the reconstruction process. A DAC uses a filter to reconstruct an analog signal from a quantized datastream. The recovered analog signal is continuous in time by definition whereas the digital datastream received at it's input is quantized.

Jitter is a problem with digital that is not a problem with analog. Aliasing is a problem with digital that is not a problem with analog. Quantization errors (noise) occurs with digital audio, that does not occur with analog, and overload characteristics of digital are profoundly different than that of analog

You've thrown a scatter load of unrelated stuff out there. Aliasing is non-issue, a well designed will reject out-of-band images with no problem, Quantisation noise occurs at the noise floor, dither lowers it even further. Analog has similar issues with noise, tape noise and groove noise are two well known examples.

Jitter is an issue, but it's minor compared to the gross distortions that occur in the analog. And it related cousins in the analog world are wow and flutter and then there is stuff like rumble.

That is a theory argument, not a real world one. If all things were perfect, you would be right. But in the real world all things are not perfect, and that is a fact. No signal chain is perfect whether it is digital or analog.

Ha! Perfection is a non-issue, a DAC always outputs analog waveform, if it doesn't it's broken. Analog waveforms ( at least, audio ones) are continuous in time by definition.

[Sir Terrence the Terrible]

Without the paper, all this is simply handwaving, Accepted theory that establishes a connection between digital encoding and agitation in test subjects would not be hidden away in a single AES paper.

I never said there was only one paper on the subject, I said I read it a couple of years ago, and I saw the test being performed at AES. I am only aware of one study, but I am pretty sure more has been done on the subject, just not for the same reasons.

You got this wrong from get-go, a DAC does not see a waveform in the sense you use the term, it recovers a digital datastream of some sort PCM, DSM or whatever from the carrier signal at it's input. The reconstruction filter reconstructs the orignal analog waveform from the datastream recovered at DAC input.

The DAC turns the binary data in a analog signal, and the reconstruction filter is used to construct a smooth analog signal from the output of a DAC. The question is does the waveform after reconstruction look exactly like the waveform from an all analog signal that never get's digitally processed. The answer to that is no.

As stated previously, a DAC recovers a datastream at it's input, You've conflated the quantization process and the reconstruction process. A DAC uses a filter to reconstruct an analog signal from a quantized datastream. The recovered analog signal is continuous in time by definition whereas the digital datastream received at it's input is quantized.

Agreed.

You've thrown a scatter load of unrelated stuff out there. Aliasing is non-issue, a well designed will reject out-of-band images with no problem, Quantisation noise occurs at the noise floor, dither lowers it even further. Analog has similar issues with noise, tape noise and groove noise are two well known examples.

Actually the issues are not so similar. Quantization noise is more audibly disturbing than the noise-floor in analog system. Dither helps can make that noise less audible, but dither can add grain to the sound as well.

Aliasing is indeed an issue, and if it wasn't an issue, the techniques such as oversampling and upsampling would not be necessary. Reconstruction filters that do not effect the frequency response in the higher frequencies have not been made. You have two choices, begin rolling off the signal before Nyquist frequency is reached, or create a brick wall filter which stands a good chance of ringing. Neither is a perfect solution, which is where oversampling comes into play.

Jitter is an issue, but it's minor compared to the gross distortions that occur in the analog. And it related cousins in the analog world are wow and flutter and then there is stuff like rumble.

Jitter can be just as big a problem as wow and flutter. It depends on how much is in the signal.

Ha! Perfection is a non-issue, a DAC always outputs analog waveform, if it doesn't it's broken. Analog waveforms ( at least, audio ones) are continuous in time by definition.

We can argue digital audio 101 till the cows come home but there is this one basic point. My point is simply this, I can record a live concert using split feeds from my board to two separate paths. One can be sent via a clean all analog path to an analog recorder, the other to a digital recorder or hard drive at 16bits. In a perfect world they should sound exactly alike during playback, but in reality they don't. In the end they both end up as analog, but the sonic character is clearly different. Neither is better(except to the listener), but they sure sound different coming from the same mikes and mixing board. Increasing the bit and sample rate decreases the difference between the digital sound and analog sound.

[theaudiohobby]

I never said there was only one paper on the subject, I said I read it a couple of years ago, and I saw the test being performed at AES. I am only aware of one study, but I am pretty sure more has been done on the subject, just not for the same reasons.

In otherwords, you have precisely zero evidence to support of what is supposedly a well-accepted theory, or proven fact as you put it. Digital encoding schemes are at the heart of telecommunications the world over, if digital induced agitation were an issue, it would be widely known and published.

The DAC turns the binary data in a analog signal, and the reconstruction filter is used to construct a smooth analog signal from the output of a DAC. The question is does the waveform after reconstruction look exactly like the waveform from an all analog signal that never get's digitally processed. The answer to that is no.

My original proposition was the output of a DAC is an analog signal, a continuous waveform by definition. Your claim that the analog signal at the DAC output is not exactly identical to the original analog signal is a strawman.

Actually the issues are not so similar. Quantization noise is more audibly disturbing than the noise-floor in analog system. Dither helps can make that noise less audible, but dither can add grain to the sound as well.

At -96dB in an undithered 16-bit system across the entire passband and greater than -100dB in a dithered one, I think not. tape noise in analog tape was so bad, it required equalisation (i.e. Dolby) to mitigate it's effects.

Aliasing is indeed an issue, and if it wasn't an issue, the techniques such as oversampling and upsampling would not be necessary. Reconstruction filters that do not effect the frequency response in the higher frequencies have not been made. You have two choices, begin rolling off the signal before Nyquist frequency is reached, or create a brick wall filter which stands a good chance of ringing. Neither is a perfect solution, which is where oversampling comes into play.

not sure what your point is here, anti-aliasing is an integral part of DAC design, and like any other engineering endeavour there are trade-offs.

We can argue digital audio 101 till the cows come home but there is this one basic point. My point is simply this, I can record a live concert using split feeds from my board to two separate paths. One can be sent via a clean all analog path to an analog recorder, the other to a digital recorder or hard drive at 16bits. In a perfect world they should sound exactly alike during playback, but in reality they don't. In the end they both end up as analog, but the sonic character is clearly different.

There are any number of reasons why the scenario you describe above can happen, however none of these reasons invalidate the basic fact that the output of a DAC is an analog signal that closely resembles the encoded analog signal within the limits of practical digital design and theory. Secondly, an analog audio signal irrespective of it's source, be it a DAC or a pure analog component is always a continuous signal.

[Sir Terrence the Terrible]

In otherwords, you have precisely zero evidence to support of what is supposedly a well-accepted theory, or proven fact as you put it. Digital encoding schemes are at the heart of telecommunications the world over, if digital induced agitation were an issue, it would be widely known and published.

We are not talking about speech here, we are talking music only. If you choose not to beleive what I have stated, that's your business. I am not here to convince anyone of anything, I gave up that battle years ago, and it will not start again with you.

Who said anything about agitation? I said less relaxed. There is a difference.

Are you asking me to produce the paper? Hey, I know it exists, but I didn't purchase it, and I have no interest in doing so just to appease you. So if you think the evidence is not there, bask in your own ignorance, I don't care. Even if I did purchase it, I could not post it as it would be against AES rules.

My original proposition was the output of a DAC is an analog signal, a continuous waveform by definition. Your claim that the analog signal at the DAC output is not exactly identical to the original analog signal is a strawman.

Once again, that is your opinion, and you are welcomed to keep it. My claim is the fact they are not identical goes along way in explaining why each sounds different even when using the same microphones mixer, amps and speakers(and with zero processing).

At -96dB in an undithered 16-bit system across the entire passband and greater than -100dB in a dithered one, I think not. tape noise in analog tape was so bad, it required equalisation (i.e. Dolby) to mitigate it's effects.

Hey, both examples require bandaids, which is why I prefer high resolution over all of them.

not sure what your point is here, anti-aliasing is an integral part of DAC design, and like any other engineering endeavour there are trade-offs.

Exactly my point.

There are any number of reasons why the scenario you describe above can happen, however none of these reasons invalidate the basic fact that the output of a DAC is an analog signal that closely resembles the encoded analog signal within the limits of practical digital design and theory. Secondly, an analog audio signal irrespective of it's source, be it a DAC or a pure analog component is always a continuous signal.

I never said that what leaves the DAC is not a analog signal, my point is that its waveform is different from an all analog signal, and that can account to why our ears respond differently to each's output.

[theaudiohobby]

We are not talking about speech here, we are talking music only. If you choose not to beleive what I have stated, that's your business. I am not here to convince anyone of anything, I gave up that battle years ago, and it will not start again with you.

Differentiating between music and speech for the purposes of telecommunications is a false distinction.

Who said anything about agitation? I said less relaxed. There is a difference.

Ok

Are you asking me to produce the paper? Hey, I know it exists, but I didn't purchase it, and I have no interest in doing so just to appease you. So if you think the evidence is not there, bask in your own ignorance, I don't care. Even if I did purchase it, I could not post it as it would be against AES rules.

I did not ask you post the paper, I asked you to state the name of the paper or any other pertinent paper for that matter. By definition, an accepted theory would be documented in multiple places.

Once again, that is your opinion, and you are welcomed to keep it. My claim is the fact they are not identical goes along way in explaining why each sounds different even when using the same microphones mixer, amps and speakers(and with zero processing).

Here you are simply wrong, I have posted a link to an academic digital theory primer to make the point. You previously said "Let's start with the waveform. Once a analog signal is encoded to 16bit resolution, it appears as a stair stepped waveform, unlike a recorded analog which is more rounded. A DAC see's that waveform, and attempts to transcode that waveform exactly as it is represented - a series of snapshots that represent a stair stepped pattern" . However the primer says

"Diagram 2a shows an acoustic sine wave with sample points at regular intervals in time from right to left on the x-axis. The sample values shown in Diagram 2b create the digital representation of the wave file shown in Diagram 2c. A low pass filter is used upon output to smooth the edges of the new digital wave, which represent added high frequencies, with the resultant wave shown in Diagram 2d."

In otherwords, the DAC produces a pretty decent analog waveform, not a stair stepped pattern, once anti-aliasing is applied, which is as one would expect given sampling theory.

I never said that what leaves the DAC is not a analog signal, my point is that its waveform is different from an all analog signal, and that can account to why our ears respond differently to each's output.

Well, as the primer clearly illustrates your original contention is simply wrong, the output of the DAC is not a stair-stepped waveform, after anti-aliasing, the resultant analog waveform is smooth and similar to (or in your words, looks like) the original waveform . and by extension your deduction about sound differences due to differences in waveform pattern is also invalid.

[Feanor]

...
Well, as the primer clearly illustrates your original contention is simply wrong, the output of the DAC is not a stair-stepped waveform, after anti-aliasing, the resultant analog waveform is smooth and similar to (or in your words, looks like) the original waveform . and by extension your deduction about sound differences due to differences in waveform pattern is also invalid.

To me it's all very confusing. As I understood the Nyquist Theorem, (granted I'll never understand it perfectly), a sound wave can in principle be perfectly reproduce up to 1/2 the sampling frequency. E.g. if the sampling rate is 44 kHz, then 22 kHz can be perfectly reproduced. But there are various demons which a technically unsophisticated person like myself can see affecting the source signal to the ADC => DAC result ...

• Equipment problem #1: incorrect measurement of the amplitude of the source signal, i.e. getting the bits right
• Equipment problem #2: getting the ADC and subsequently DAC timing right, i.e. sampling at precisely the right rate, i.e. preventing "jitter"
• Technical problem #1, 2, ... etc.: eliminating the spurious sound of samples about 1/2 the sampling frequency, i.e. how do you dump the frequencies above 22 kHz (in case of CD) which are noise? I suppose this might be dealt with more or less well. The techniques of anti-aliasing, dithering, and filtering are very obscure to me, Except I think dithering has to do with randomizing the digital noise -- signal above 22 kHz -- which would otherwise variy directly with the "good" signal below 22 kHz). Also, I understand that filtering can (or must?) affect the phase of the signal below the 1/2 sample rate frequency.
• Psycho-accoustic problem: do sounds above concious audibilty, say 20+ kHz, really have no effect on sound perceptions?

In any case it's obvious that the source sound, (analog), will include frequencies above 1/2 the sample rate, so will not be identical to the ADC => DAC's signal however prefectly the sound below 1/2 are reproduced.

I'd appreciate further clarification of these issues though I can't guarantee I'd understand what you're saying.

[theaudiohobby]

Equipment problem #1: incorrect measurement of the amplitude of the source signal, i.e. getting the bits right

This is a quality assurance issue, If the ADC is unable measure amplitude correctly, it’s broken. That said, there are overload conditions, where the ADC input is overloaded and under this conditions, bits recorded would be wrong, herefore care is typically taken overloading the ADC. Sir Terence is a professional engineer, so he will be in a better position to discuss this one.

Equipment problem #2: getting the ADC and subsequently DAC timing right, i.e. sampling at precisely the right rate, i.e. preventing "jitter"

This is an issue in professional recording and the traditional solution has been to slave all converters to a single clock. See some John Atkinson’s recordings notes

Technical problem #1, 2, ... etc.: eliminating the spurious sound of samples about 1/2 the sampling frequency, i.e. how do you dump the frequencies above 22 kHz (in case of CD) which are noise? I suppose this might be dealt with more or less well. The techniques of anti-aliasing, dithering, and filtering are very obscure to me, Except I think dithering has to do with randomizing the digital noise -- signal above 22 kHz -- which would otherwise vary directly with the "good" signal below 22 kHz). Also, I understand that filtering can (or must?) affect the phase of the signal below the 1/2 sample rate frequency.

Dithering is a process that decorrelates quantisation noise from the digital signal across the entire bandwidth, it’s separate and different from anti-aliasing where noise, specifically, alias images above 22kHz are removed by the anti-aliasing filter i.e. low-pass filter. At this junction a new concept comes into play the passband and the transition band. The passband of CD is typically DC-20kHz, above that you have the transition band, the anti-aliasing filter generally does most of its filtering in the transition band. Filtering affects phase below ½ sample rate frequency, however if the transition band is suffiently large, phase distortion in the passband is negligible. IMO, this is one of the area where a higher sampling rate or oversampling has some benefits.

Psycho-accoustic problem: do sounds above concious audibilty, say 20+ kHz, really have no effect on sound perceptions?

Not sure, there is only way to answer that question, run controlled listening tests.

[E-Stat]

Well, as the primer clearly illustrates your original contention is simply wrong, the output of the DAC is not a stair-stepped waveform, after anti-aliasing, the resultant analog waveform is smooth and similar to (or in your words, looks like) the original waveform

How can anyone possibly disagree with a cute little picture like that?

rw

[Sir Terrence the Terrible]

Differentiating between music and speech for the purposes of telecommunications is a false distinction.

Ok,

I did not ask you post the paper, I asked you to state the name of the paper or any other pertinent paper for that matter. By definition, an accepted theory would be documented in multiple places.

Since I read the paper while passing the exibit, I didn't pay attention to the name of the paper or who did it. I just ask what the long line was for while attending the convention, and the person I was with put that white paper in my face. I asked was it credible, and the guy I was with(who actually WAS interested in it) told me it passed peer review, nobody is challenging it. I read it, I know it exists, but it was not my interest. It is still not my interest, and I am not going to spend my time chasing it.

Here you are simply wrong, I have posted a link to an academic digital theory primer to make the point. You previously said "Let's start with the waveform. Once a analog signal is encoded to 16bit resolution, it appears as a stair stepped waveform, unlike a recorded analog which is more rounded. A DAC see's that waveform, and attempts to transcode that waveform exactly as it is represented - a series of snapshots that represent a stair stepped pattern" . However the primer says

"Diagram 2a shows an acoustic sine wave with sample points at regular intervals in time from right to left on the x-axis. The sample values shown in Diagram 2b create the digital representation of the wave file shown in Diagram 2c. A low pass filter is used upon output to smooth the edges of the new digital wave, which represent added high frequencies, with the resultant wave shown in Diagram 2d."

In otherwords, the DAC produces a pretty decent analog waveform, not a stair stepped pattern, once anti-aliasing is applied, which is as one would expect given sampling theory.

I am not debating on whether you see a analog waveform after reconstruction(anti aliasing is a front loaded process not a rear loaded one, reconstruction is done after conversion) we both agreed what is done is an analog waveform. My contention(and what I have seen) is the digital waveform does NOT look like an all analog waveform, even though they are both analog. I said it sawtoothed BEFORE conversion, and that is what the DAC is reconstructing, snapshots of the signals, not the signal in its entirety like analog represents(at least that was what I was trying to say.)

Well, as the primer clearly illustrates your original contention is simply wrong, the output of the DAC is not a stair-stepped waveform, after anti-aliasing, the resultant analog waveform is smooth and similar to (or in your words, looks like) the original waveform . and by extension your deduction about sound differences due to differences in waveform pattern is also invalid.

Your dependency on theory shows that you have zero experience in real life. If all things were perfect, you would be correct. In the real world, all things are not perfect. When a signal passes through a real life DAC, it is not a perfect process that mirrors you visual example. Theory often fails in the presence of real life ADC and DAC chips. I have not heard a perfect digital system(or analog for that matter).

If all is perfect as you describe, then why oversampling? You shouldn't need it all is perfect right?

Once again, if the two waveforms are exactly alike as you say, then why do they sound so different even when the same equipment is used?

[theaudiohobby]

Since I read the paper while passing the exibit, I didn't pay attention to the name of the paper or who did it. I just ask what the long line was for while attending the convention, and the person I was with put that white paper in my face. I asked was it credible, and the guy I was with(who actually WAS interested in it) told me it passed peer review, nobody is challenging it. I read it, I know it exists, but it was not my interest. It is still not my interest, and I am not going to spend my time chasing it.

I hear you, so much for the widely accepted theory.

I am not debating on whether you see a analog waveform after reconstruction(anti aliasing is a front loaded process not a rear loaded one, reconstruction is done after conversion) we both agreed what is done is an analog waveform. My contention(and what I have seen) is the digital waveform does NOT look like an all analog waveform, even though they are both analog. I said it sawtoothed BEFORE conversion, and that is what the DAC is reconstructing, snapshots of the signals, not the signal in its entirety like analog represents(at least that was what I was trying to say.)

I am baffled by your commentary, why would anyone care about the digital waveform that feeds the DAC? It's an intermidiate process and the DAC is the intended consumer not human ears. The DAC can read the data stream from the modulated carrier signal without issue and recreate the original analog signal, so why the contention?

Your dependency on theory shows that you have zero experience in real life. If all things were perfect, you would be correct. In the real world, all things are not perfect. When a signal passes through a real life DAC, it is not a perfect process that mirrors you visual example. Theory often fails in the presence of real life ADC and DAC chips. I have not heard a perfect digital system(or analog for that matter).

The theory is the basis for the real-life implementation. Without it, the real-life would not exist.

If all is perfect as you describe, then why oversampling? You shouldn't need it all is perfect right?

That's a separate discussion, right? By the way, this is the second time you raised the perfection strawman.

Once again, if the two waveforms are exactly alike as you say, then why do they sound so different even when the same equipment is used?

I disagree, but that's another separate discussion.