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If Bits are Bits

Galliardist

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solderdude

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Realized that Lindemann has a USB input. I had used it before with a disc drive. Did some listening. CD is only slightly better than the stream. Bryston still rules.
Conclusion.... it is not the bits. There are differences between the devices. This means this is measurable. The most obvious difference would be the output level.
When the Bryston rules it would be a fair bet the output voltage is higher or the converter in the streamer is of that low quality it becomes audible (also very measurable)

When you really want to go to the bottom of this you would have to record the analog outputs of both devices. For that you would need a good ADC and record with at least 96kHz.

To determine the difference in output voltage you can also use an ADC but can be a crappy one (the one in a PC might do).
 

Galliardist

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And now we’re on page six in this thread. I don’t have much hope that he’ll actually try finding the cause, if any, for the differences in sound.
Page 7, and @Takaya and @solderdude have now got the most likely suspect in for questioning, even if we don't have the evidence for the court case just yet.

Well worth persevering!
Conclusion.... it is not the bits. There are differences between the devices. This means this is measurable. The most obvious difference would be the output level.
When the Bryston rules it would be a fair bet the output voltage is higher or the converter in the streamer is of that low quality it becomes audible (also very measurable)

When you really want to go to the bottom of this you would have to record the analog outputs of both devices. For that you would need a good ADC and record with at least 96kHz.

To determine the difference in output voltage you can also use an ADC but can be a crappy one (the one in a PC might do).
I'd put my money on the streamer being low quality. Bryston are generally a good engineering company and if they quote 2v in the manual the output's going to be 2v. Amir measured the Lindemann at 2v, IIRC. But -
@Takaya - it'd still be good experience to do the test.
 

solderdude

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I left out the most obvious cause though and have a sneaky suspicion the comparison is done while knowing which is playing and without statistical relevance.
 

Adi777

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there's some nostalgia or tactile feel to CDs, that enhances your listening experience, I don't see any reason to avoid it.
I've been thinking about it. If our eyes and brain recognize us, why should we not be fooled? At least a bit.
As long as you don't write later that this game is better than that one.
 

Killingbeans

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Absolutely.

If the looks of my audio gear, or the feel of the buttons and knobs, or the smoothness of its graphical user interface makes it sound better to me, I see nothing but good things in using it to my advantage.

But claiming that it alters the actual waveform of the audio would be silly of me.
 

hedrick

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Odd. I haven't noticed this. I'm using a Bluesound Powernode, so most of the electronics are the same for both. For CD I'm using a ripped copy, and for streaming Qobuz.
 

dorakeg

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We hear from folks that bits are bits. Why does a CD redbook player sound better than a Streamer at 44.1/16, or even DSD.

I was googling about more information regarding ring face and found this which I feel is very good explanation.

 

Killingbeans

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It's just common knowledge mixed with nonsense.

4120ed0cefbbc3665984b2d1fb3c7d84b1f87abe.png

a2b965e2d110c6da54050e7a759631d6928bc6de.png

"This diagram shows how an analogue sound wave can be represented with 24-bit 176,400 samples per second PCM encoding. The sample rate being higher than CD audio above allows for a greater representation across the X axis of this graph, whereas the higher bit-depth allows for the exact amplitude of the wave to be more accurately represented with each sample – the Y axis."

"It is a common misconception that 24-bit audio simply records louder and quieter sounds than is possible with 16-bit audio, but this is not the case. Instead, the same range of loudest to quietest is measured, but with 24-bit sampling it is done with considerably more steps than with 16-bit. This means the absolute value of the waveform at any given point can be much better represented."


Absolute bullshit. Showing that last 22.7us of analouge 16 bit output as a straight line is nothing short of malevolent misinformation (I refuse to believe, they are that stupid).

They keep trying to invoke the false intuition most people have about "greater representation". There's no such thing.

"This results in the Ring DAC having class-leading distortion performance, particularly at lower signal levels. This means more fine detail can be resolved and heard in the audio."

Okay, so. A current source ladder solves a number of problems and takes DAC performance to a new level? Then why just keep rambling about the tech, and not just show measurements of the performance? If it's as good as they say, it should be easy as pie.
 

Galliardist

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It's just common knowledge mixed with nonsense.

4120ed0cefbbc3665984b2d1fb3c7d84b1f87abe.png

a2b965e2d110c6da54050e7a759631d6928bc6de.png

"This diagram shows how an analogue sound wave can be represented with 24-bit 176,400 samples per second PCM encoding. The sample rate being higher than CD audio above allows for a greater representation across the X axis of this graph, whereas the higher bit-depth allows for the exact amplitude of the wave to be more accurately represented with each sample – the Y axis."

"It is a common misconception that 24-bit audio simply records louder and quieter sounds than is possible with 16-bit audio, but this is not the case. Instead, the same range of loudest to quietest is measured, but with 24-bit sampling it is done with considerably more steps than with 16-bit. This means the absolute value of the waveform at any given point can be much better represented."


Absolute bullshit. Showing that last 22.7us of analouge 16 bit output as a straight line is nothing short of malevolent misinformation (I refuse to believe, they are that stupid).

They keep trying to invoke the false intuition most people have about "greater representation". There's no such thing.

"This results in the Ring DAC having class-leading distortion performance, particularly at lower signal levels. This means more fine detail can be resolved and heard in the audio."

Okay, so. A current source ladder solves a number of problems and takes DAC performance to a new level? Then why just keep rambling about the tech, and not just show measurements of the performance? If it's as good as they say, it should be easy as pie.
The articles seem to be the responsibility of the Head of Customer Service.
For the company that dCS once was, this seems rather odd. Maybe their engineers have been tied up writing returns policies for the last decade?
 

fpitas

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The articles seem to be the responsibility of the Head of Customer Service.
For the company that dCS once was, this seems rather odd. Maybe their engineers have been tied up writing returns policies for the last decade?
Often the job falls to tech writers, who may or may not have a good grasp of engineering.
 

KSTR

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Absolute bullshit. Showing that last 22.7us of analouge 16 bit output as a straight line is nothing short of malevolent misinformation (I refuse to believe, they are that stupid).
I agree their presentation style is not optimal, for the sample value display they should have used pin plots (not connecting the dots) rather than bins one sample wide. But their meaning is explained fully correct:

"This diagram shows how an analogue sound wave can be represented with 24-bit 176,400 samples per second PCM encoding." (Underline mine).

That's what sample values do, they represent the voltage at the (infinitely short, theoretically) point in time the snapshot was made.
If you were to put out this value constantly for one sample period you have a NOS DAC "reconstruction" filter which is what you comment is probably about, but I don't see that in the context of the graphs and associated text. The sample stream is meant to be read as the abstract data it represents, not a time-domain actual output waveform of the DAC. Those dCS guys are serious engineers...
 

Killingbeans

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The sample stream is meant to be read as the abstract data it represents, not a time-domain actual output waveform of the DAC.

I can accept that. But I guarantee that 99.9% of all people who read those articles don't see it that way. They just see the words "greater representation" and go: "Yes! I knew it! I can definitely hear that when I listen to my DAC!"
 

danadam

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They keep trying to invoke the false intuition most people have about "greater representation". There's no such thing.
My impression too.

Some loose comments:
It is generally accepted that the human ear can perceive equivalent to 20-bits of dynamic range, which equates to around 140dB
140? Not 120? Also it would be nice to mention, that at a single instance, it's more like 60 dB.

Moving to a hi-res format like 24-bit, this dynamic range increases to 144dB – assuming the equipment is actually capable of working in true 24-bit.
Such a good place to mention, that physics caps the capabilities of the equipment below 24-bit.

It is a common misconception that 24-bit audio simply records louder and quieter sounds than is possible with 16-bit audio, but this is not the case. Instead, the same range of loudest to quietest is measured, but with 24-bit sampling it is done with considerably more steps than with 16-bit.
Also a common misconception, that those additional steps make much difference [in distributed files] :)

And I'd say 24-bit actually does allow recording quieter sounds.

Imagine for a moment trying to measure the height of a particular window on a skyscraper. ...
Sure, sure, but let's do something else first. Let's listen to this below 90 dFBS white noise [insert audio here]. You don't hear anything? Well, this is the only difference between 24 and 16 bits. And it will be similar for any recording. Here you have it 30 dB louder, you should hear something now [insert audio here].

And if you don't know how much 30 dB is, here you have it louder by another 30 dB [insert audio here].

Ok, now let's imagine those scary errors :)

Quantisation noise is the audible noise which is generated by the error in the measurement.
How audible? Is it always audible?

[DSD]
The result is near 24-bit performance in the audio band (0 – 20kHz) and a signal bandwidth that extends beyond 100kHz. The price for the 1-bit approach is a very large amount of noise in the ultrasonic region (20kHz – 1.4MHz)
Maybe that's my cynicism showing, but I feel like they hope, that the readers will remember this 100 kHz but forget the noise part :)

Most ADCs will work using a symmetrical filter. What this means is that for any digital recording, there will be (necessary) pre- and post-ringing present on the recording, as a result of the filter which was used.
If I'm not mistaken, only if there was content at the cut-off frequency, which usually there isn't much with music signals.
 
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voodooless

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If we have a 120 m skyscraper and have a 16-bit height resolution, we can determine the height of a window with up to 1.8mm precision. For 24 bits, we can do it with up to 7.5 µm. And you can simulate higher than 16-bit height resolution, by moving the window up and down really fast in a random manner ;) To do this, you'll need two small black wholes spinning very fast around each other, fairly close to said building :cool:.
 

Blumlein 88

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If we have a 120 m skyscraper and have a 16-bit height resolution, we can determine the height of a window with up to 1.8mm precision. For 24 bits, we can do it with up to 7.5 µm. And you can simulate higher than 16-bit height resolution, by moving the window up and down really fast in a random manner ;) To do this, you'll need two small black wholes spinning very fast around each other, fairly close to said building :cool:.
That brings up a new worry. What if two small black holes were spinning very fast around your DAC? Gravitational jitter?
 

Blumlein 88

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Absolutely! Didn’t you know Ligo is secretly sponsored by PS Audio?
Hey, I've been holding back on a method for reducing outside influences. Everyone knows (don't they?) life is only possible on earth because our magnetosphere works like shields on the Star Trek Enterprise to keep radiation away from us on the earth's surface. Why not have a magnetosphere generator for your audio gear. It is science and all that.

 
D

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That brings up a new worry. What if two small black holes were spinning very fast around your DAC? Gravitational jitter?
You'll need some damn good shielded cables for the electrons not to escape. Maybe then it's actually time to buy some of those audiophile ones.
 

dorakeg

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It's just common knowledge mixed with nonsense.

4120ed0cefbbc3665984b2d1fb3c7d84b1f87abe.png

a2b965e2d110c6da54050e7a759631d6928bc6de.png

"This diagram shows how an analogue sound wave can be represented with 24-bit 176,400 samples per second PCM encoding. The sample rate being higher than CD audio above allows for a greater representation across the X axis of this graph, whereas the higher bit-depth allows for the exact amplitude of the wave to be more accurately represented with each sample – the Y axis."

"It is a common misconception that 24-bit audio simply records louder and quieter sounds than is possible with 16-bit audio, but this is not the case. Instead, the same range of loudest to quietest is measured, but with 24-bit sampling it is done with considerably more steps than with 16-bit. This means the absolute value of the waveform at any given point can be much better represented."


Absolute bullshit. Showing that last 22.7us of analouge 16 bit output as a straight line is nothing short of malevolent misinformation (I refuse to believe, they are that stupid).

They keep trying to invoke the false intuition most people have about "greater representation". There's no such thing.

"This results in the Ring DAC having class-leading distortion performance, particularly at lower signal levels. This means more fine detail can be resolved and heard in the audio."

Okay, so. A current source ladder solves a number of problems and takes DAC performance to a new level? Then why just keep rambling about the tech, and not just show measurements of the performance? If it's as good as they say, it should be easy as pie.

They are just plotting sample value over time using bar chart (can also call it histogram). So, the presentation is correct. It is supposed to be a straight line because the value does not change during that period.

If you were to use an equipment/computer to read the output values, and show them on a histogram, that would be exactly what you will get. Values will only change during beginning of each ~22.7 block, then remain constant throughout the block period.
 
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