Zp 24/96
A
Ability to play 24bit/96 files (like the competition: slimdevices transporter)
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There are no gaps. There are no steps. The digital samples represent the input waveform at that instant. At any other time the level is in fact undefined. Have a look at the second video at the link I posted.
Oh, and a 256kbps file is that size because of lossy compression, which is a whole different ball game. It's still based on 16-bit and (usually) 44.1kHz sampling.
Oh, and a 256kbps file is that size because of lossy compression, which is a whole different ball game. It's still based on 16-bit and (usually) 44.1kHz sampling.
Have a look at the second video at the link I posted.
I tried. Apple Safari is not supported.
The quote was from this:
http://www.msbtech.com/support/How_DACs_Work.php
Quote:
Digital Filter - Fills in the Blanks
The digital filter is the computer algorithm that looks at the digital audio in the past, and the digital audio in the future and tries to figure out what was going on with the music, and tries to shape the final analog output to match as closely as possible that original waveform, now missing for ever.
Unquote
What I understand from this is that even the 16/44 file will have gaps that will have to filled in by the DAC to give a step less analog output, that matches as closely as possible the analog signal recorded, before it was digitised.
It follows from this that a 256kbps file will have more gaps and something will have to do the work of trying to figure out what these larger gaps are so that a step less analog signal can be produced.
It also follows that more content than a 16/44 file will have smaller gaps to be filled, but the result will still be an approximation of the original wave form.
I also understand that the approximation from the 16/44 information content is good enough to audibly sound the same as the original analog wave form.
Question: whats wrong with the above?
What's wrong? Pretty much everything (which goes to show either how widespread this misconception is even amongst professionals, or how far they go in order to trick their customers into buying snake oil).
If shouting doesn't work, maybe frequent repetition will help to get the point across: There. Are. No. Gaps. A recording sampled with 16/44 will contain everything there was in the analog waveform below 22 kHZ with the same absolute, mathematically unerring preciseness. There will be not the faintest amount of approximativeness. It's not that the difference isn't audible - it does not exist. At all.
Everything you might gain by using a higher sample rate is that higher frequencies will be captured. Your ears are not able to hear them, in the same way that you can't see x-rays with your eyes. On the downside, this can in some cases actually lead to a lower sound quality, as most audio gear wasn't designed to handle these higher frequencies.
To sum it up: If you're lucky, there will be no difference. In any case, you won't gain anything, but your chances of running into other problems are high.
Ceterum censeo: There are no gaps.
PS: Audio compression (as in "256kbps") has nothing to do with that. With any lossy compression method (mp3 etc.), information will be lost. The dichotomies here are i) lossy vs. lossless compression (audible differences can occur, lossy may sound worse) and ii) 16/44 vs. 24/192 (theoretically no audible differences occur, though practically 24/192 may sound worse).
Oh, and a 256kbps file is that size because of lossy compression, which is a whole different ball game. It's still based on 16-bit and (usually) 44.1kHz sampling.
Ok, but the question here is:
Is the way the 256kbps file is handled done differently by different DACs or is this in the codec that did the compression and every DAC will handle it the same way while using the information to produce the step less wave form?
Is the way the 256kbps file is handled done differently by different DACs or is this in the codec that did the compression and every DAC will handle it the same way while using the information to produce the step less wave form?
Decompression is done by the codec. For the DAC, it doesn't make a difference.
Ok, thanks everyone.
I did look up Nyquist on the net, but the minute I saw the math that looked suspiciously like Calculus, I realised I am never going to understand it.
Time now to listen to some very high quality music from my recent 256kbps I tunes purchases and clear my mind:).
Time now to listen to some very high quality music from my recent 256kbps I tunes purchases and clear my mind:).
That's the spirit! 😉 Enjoy it!
http://www.msbtech.com/support/How_DACs_Work.php
I also found:
we know that there is data missing between each sample, and that the original level may have been a little more or less than the level recorded (quantization error). This is where the science ends and the art begins as we try to guess what the errors were and what is missing between the blanks
I'm sorry, I stopped reading at that point and searched in vain for the words 'dither' and 'Nyquist' on the page.
Not unrelated, I'm sure, is the fact that this is all to help sell a $7k DAC....
we know that there is data missing between each sample, and that the original level may have been a little more or less than the level recorded (quantization error). This is where the science ends and the art begins as we try to guess what the errors were and what is missing between the blanks
I'm sorry, I stopped reading at that point and searched in vain for the word 'Nyquist' on the page.
Not unrelated, I'm sure, is the fact that this is all to help sell a $7k DAC....
Articles like that should be the target of lawsuits. I'm dead serious. It is the very definition of consumer fraud.
Indeed. It has been a good day. To quote Rumsfeld, I have progressed in this area from an unknown unknown to a known unknown.
Not unrelated, I'm sure, is the fact that this is all to help sell a $7k DAC....
Ahh..you cynic! I didn't even get to that part of the page:).
🙂. In which case in addition to digital snake oil salesmen, the lawyers will also get on the train.
At least (in my country) lawyers are limited to a 33% cut, unlike the profit margin of the snake oil salesmen which can be 3000%-4000%-10000% and beyond.
Indeed.
The word " real" is interesting. Now, it is one thing to say that someone with an engineering education selling the common snake oil isn't a real Engineer.
Not necessarily. They just have a different requirements to work to. Those requirements may be, at heart, dishonest of course.
But would you extend it to the ones developing newer DAC chipsets?
I am not sure if it is even in their brief to measure their output performance via a double blind AB test with precision level matching, to validate their advance. I am not saying this is right or good, just pointing out the probable situation.
Well there are good reasons to support higher resolutions in certain applications, so the need to develop hi res ADC and DAC chipsets to support those is valid.
Some of those applications, such as certain types of professional recording, are even audio applications. They're just not the ones you need at home to listen to music. And there are reasons (like DSP and volume control) why it makes sense for audio DACs to have a certain amount of "hires" capability, for internal use within audio kit.
And, of course, mostly the Engineers involved in the chipsets are not normally the people setting the product strategy. And, as you say, they are not normally asked to research or test the end-user applications. I have never, for instance, seen a datasheet for a hires codec which includes the words "transparency", "warmth", "soundstage" or any of the terms so often used by audiophiles.
The thing to bear in mind is that hires DAC chips are so cheap and easily available that it makes sense to put them into many products by default, and there are some advantages to using them
The DAC chips in Sonos kit are, strictly speaking, "hires". There's nothing wrong with using a component that is over-specified if there is no significant cost or downside to it. In fact there's often very good reasons for using a higher specification that you need. For instance, the extra resolution is useful to make digital volume control simpler. And many modern receivers include a lot of internal digital signal processing which is often better done in hires.
However, extending this capability to the user usually doesn't make sense: it usually costs more to do, and presents Engineering challenges that need to be solved and with no real technical benefit. That is certainly the case with Sonos where extending hires beyond the DAC chipset would place massive additional load on the processing, buffering, and networking parts of the system.
And there are lots of engineers involved in developing lots of new products that don't fit the description you use of a real Engineer. Actually, I would venture to say that a small percentage of all the engineers in the world do.
In many companies including consumer electronics but especially chip design, it's a higher percentage than you think.
You are right that not everyone who "does electronics" has come from a formal Engineering training background. There are people who are, no doubt, far more practically experienced and knowledgeable in electronic circuit design construction than many formally trained Electronics Engineers (who often don't end up doing Electronics as a career). Many of them work in smaller companies, or are just enthusiastic hobbyists. They are mostly very good and skilled at what they do, but they are lacking much of the formal methodology that is drilled into well trained Engineers and Scientists.
The same applies to many technicians in recording studios. They are very experienced and skilled in things like room acoustics, microphone selection and placement, and running a mixing desk, but they mostly don't seem to have an Engineering background. That's fine because they don't need it to do their job, but it means they usually lack the skills to be rational about things like hires. Many of them simply don't understand it that well.
It doesn't help that their industry is steeped in traditions, skills, practises, knowledge and superstitions that evolved in the analogue days and have been handed down to successive generations.
Cheers,
Keith
I find absolutely nothing cynical in ratty's statement.
I do find it sad that the statement needs to be made, and so frequently, with regards to certain (if not all) sales pitches. Especially (but not limited to) the high-end audio equipment arena.
We Sheeple are so very gullible.... :o
Best of Luck
Kumar,
Think of that stair step representation of digital audio as a stylized diagram introduced for discussion. Unfortunately, at first glimpse, the audiophiles left the room waving their hands above their heads -- yelling "jaggies", "missing parts", and on and on. They should have stayed around and listened to more of the discussion.
In reality, the vertical lines on that stylized staircase cannot be perfectly vertical because this would require more energy than we could physically come up with. This implies that there is some integration, rounding if you like, of the playback waveform. Those vertical edges are not exactly vertical and the corners are not perfectly sharp. At this point the real question is "what is the minimum sample rate" still allows these round offs to smooth things and recreate the original sampled source. This is the problem that Nyquist solved.
In school we were introduced to the difference between a theoretical scientist and an engineer by presenting a problem:
A guy and a girl are separated by a certain distance and the guy is running towards the girl at a speed that halves their separation each second. The question: "How long until the guy reaches the girl?"
The science student replies "never", while the engineer will reply "after nn seconds the guy will be close enough for all practical purposes."
Think of that stair step representation of digital audio as a stylized diagram introduced for discussion. Unfortunately, at first glimpse, the audiophiles left the room waving their hands above their heads -- yelling "jaggies", "missing parts", and on and on. They should have stayed around and listened to more of the discussion.
In reality, the vertical lines on that stylized staircase cannot be perfectly vertical because this would require more energy than we could physically come up with. This implies that there is some integration, rounding if you like, of the playback waveform. Those vertical edges are not exactly vertical and the corners are not perfectly sharp. At this point the real question is "what is the minimum sample rate" still allows these round offs to smooth things and recreate the original sampled source. This is the problem that Nyquist solved.
In school we were introduced to the difference between a theoretical scientist and an engineer by presenting a problem:
A guy and a girl are separated by a certain distance and the guy is running towards the girl at a speed that halves their separation each second. The question: "How long until the guy reaches the girl?"
The science student replies "never", while the engineer will reply "after nn seconds the guy will be close enough for all practical purposes."
In school we were introduced to the difference between a theoretical scientist and an engineer by presenting a problem:
A guy and a girl are separated by a certain distance and the guy is running towards the girl at a speed that halves their separation each second. The question: "How long until the guy reaches the girl?"
The science student replies "never", while the engineer will reply "after nn seconds the guy will be close enough for all practical purposes."
Would that be a theoretical scientist or a philosopher named Zeno?!
I am no engineer, but I am curious about how things work. And even a known unknown - I tend to keep gnawing at it.
Based on more conversation with a knowledgeable friend, I have conceptualised this for myself in this manner:
Sound is nothing but vibrations in the air. The sound vibrations have a width and height characteristic - to use those terms for the time and sound levels involved in the vibrations.
What is said to be an analog wave form is nothing but joining the dots of the two points of the width extremes and the two points of the height extremes to get a wave form representation. There is actually no wave - or maybe there is and perhaps this is also the good old wave particle duality thing in light as it applies to sound. But it doesn't matter for this purpose.
So, if one were to know these four points, joining the dots would produce exactly the same wave form representation of the vibration. Exactly the same waveform, naturally, because one is just joining the same dots. Or perhaps just two points - one of the height points and one of the width points is all that needs to be known, I suspect. And the word sample is part of what causes the misstep because it is commonly understood to be a representative sample. But what is being captured is a 100% sample. And taking samples 40000 times a second gives one a 100% sample for all sound frequencies up to 20khz. Using these for reproduction gives all the information needed to reproduce the sound "wave" completely. There are no gaps because sampling has been done as fast as the vibrating air molecules are able to move.
If I have understood it right then, capturing all the information of these points for a 20khz frequency sound means that one has also captured it for every frequency below that, not just down up to 20hz.
Now I have to do the same thing for the 16bit thing, to get a hold of that in a way I understand it!
PS: It just struck me that the width point information referred above is nothing but the number of peaks in a second...
To state this further:
I am now pretty sure that the representational wave you see on an oscilloscope is drawn by the electronics in the instrument by just these two bits of information - of peak heights and number of peaks in a second, and we end up thinking that there are data points for all of points on the lines shown that connect these information points. There aren't so...NO GAPS:).
For digital to be 100% truthful it has to carry all the information about the peak heights for as many times as they occur in a second. And sampling 40000 times a second allows this to be done for all the data points that exist for sound frequencies up to 20khz. The 44000 times is for a margin of safety.
I think this time around, I have reached the right conclusion...
The times when it does not happen is not because of a flaw in this reasoning, but because of engineering constraints that have to be overcome in converting this state of affairs to just as truthful electrical signals that reach the speakers.
Digressing again, my experience of modern digital audio equipment tells me that these constraints have now been overcome to the point that further progress isn't to be audibly heard in a well constructed listening test. And the solutions overcoming these have by now also found their way to cheap digital components.
I am now pretty sure that the representational wave you see on an oscilloscope is drawn by the electronics in the instrument by just these two bits of information - of peak heights and number of peaks in a second, and we end up thinking that there are data points for all of points on the lines shown that connect these information points. There aren't so...NO GAPS:).
For digital to be 100% truthful it has to carry all the information about the peak heights for as many times as they occur in a second. And sampling 40000 times a second allows this to be done for all the data points that exist for sound frequencies up to 20khz. The 44000 times is for a margin of safety.
I think this time around, I have reached the right conclusion...
The times when it does not happen is not because of a flaw in this reasoning, but because of engineering constraints that have to be overcome in converting this state of affairs to just as truthful electrical signals that reach the speakers.
Digressing again, my experience of modern digital audio equipment tells me that these constraints have now been overcome to the point that further progress isn't to be audibly heard in a well constructed listening test. And the solutions overcoming these have by now also found their way to cheap digital components.
By the way, a link from someone that isn't selling DACs:
http://documentation.apple.com/en/soundtrackpro/usermanual/index.html#chapter=B%26section=2%26tasks=true
And a quote from there:
The sample rate is the number of times an analog signal is measured—or sampled—per second. You can also think of the sample rate as the number of electronic snapshots made of the sound wave per second. Higher sample rates result in higher sound quality because the analog waveform is more closely approximated by the discrete samples.
Unquote...no wonder there is so much confusion around this subject.
Separately, I also think I have found the way to understand the bit and sampling frequency things to think of the first as sample size, and the second as...well, as how often the sample is taken.
And that the sample size determines the dynamic range that is captured.
http://documentation.apple.com/en/soundtrackpro/usermanual/index.html#chapter=B%26section=2%26tasks=true
And a quote from there:
The sample rate is the number of times an analog signal is measured—or sampled—per second. You can also think of the sample rate as the number of electronic snapshots made of the sound wave per second. Higher sample rates result in higher sound quality because the analog waveform is more closely approximated by the discrete samples.
Unquote...no wonder there is so much confusion around this subject.
Separately, I also think I have found the way to understand the bit and sampling frequency things to think of the first as sample size, and the second as...well, as how often the sample is taken.
And that the sample size determines the dynamic range that is captured.
Not to mention their $74,950 model ( :eek: )
A guy and a girl are separated by a certain distance and the guy is running towards the girl at a speed that halves their separation each second. The question: "How long until the guy reaches the girl?"
The science student replies "never", while the engineer will reply "after nn seconds the guy will be close enough for all practical purposes."
In the version I heard, there were three parties: the scientist, the engineer and the salesman. They were told that every N seconds they could halve the distance to the girl.
As noted, the scientist gave up in disgust and the engineer was soon close enough to engage with the girl.
The salesman spent all his time trying to negotiate for a greater reduction of the distance on each cycle, so never even started.
Plus extras, of course, including the $1790 heat sinks.
http://documentation.apple.com/en/soundtrackpro/usermanual/index.html#chapter=B%26section=2%26tasks=true
And a quote from there:
The sample rate is the number of times an analog signal is measured—or sampled—per second. You can also think of the sample rate as the number of electronic snapshots made of the sound wave per second. Higher sample rates result in higher sound quality because the analog waveform is more closely approximated by the discrete samples.
Unquote...
Again, no mention of the Nyquist–Shannon sampling theorem and the limits of human hearing. It's the 'more is better' mantra.
Kumar, find a way to take a look at that video I linked. The practical demonstrations, especially of a 20kHz sinewave being perfectly reconstructed (give or take a bit of inherent low level noise in the analog test equipment), should reassure you.
Kumar,
A lot of the confusion and misrepresentation arises because of the oscilloscope model of the waveforms that is presented. It has it's uses but it actually misrepresents audio because it only loosly approximates to what we hear.
If you spend to much time looking at, and trying to analyse, oscilloscope waveforms then you go off on the wrong track.
A more appropriate view of the waveform would be the spectrogram because our ears and brains tend to resolve frequencies, not instantaneous sound level.
By the way, sound waves in air exist purely as areas of air compression and decompression (rarefaction). It's all relatively straightforward and there isn't the same duality as their is with light: there is no "sound particle".
A lot of the maths involved in audio (Nyquist, etc.) involves transforming various waveforms (such as oscilloscope-style time/amplitude waveforms) into the frequency domain so we can understand how it sounds.
If you take, as an example, a classic square wave:
http://mathworld.wolfram.com/images/eps-gif/SquareWave_700.gif
If you look at the oscilloscope display it doesn't look as if there is much audio information in there at all. Intuition suggests that there wouldn't be much to hear.
Intuition is misleading.
The reality is that square waves produce a huge (theoretically infinte) range of harmonics creating a very full "fat" sound.
http://ptgmedia.pearsoncmg.com/images/chap2_9780132349796/elementLinks/2-6.jpg
Rather than looking at oscilloscope-style displays, start thinking of it in terms of sine-waves. Sine waves are the building block of audio. Every sound can, ultimately, be considered to be constructed from combinations of sine waves of different frequencies and amplitudes.
When you see the jagged stepped sine waveform so often shown when talking about digital audio, your intuition tells you there is something missing, but that is incorrect. Your intuition is misleading you.
In actual fact the opposite is true. Our hearing lives in a world where everything has to be created from (or resolve to) a pure, complete sine-wave these steps simply cannot exist.There simply cannot be bits of the sine wave missing, because the smallest possible representation of sound is a complete sine-wave.
In fact, the stepped sine-wave diagram actually corresponds to a a sine wave with bits added in the form of higher-frequency harmonics.
But it's important to realise that the steps don't exist in the first place. The stepped waveform diagram is a misrepresentation of how sampling actually works. It's not an Engineering diagram, it's a marketing one used, primarily, to deceive people. It is a lie.
Because the liars, crooks, and con-men in the industry know how to manipulate our intuition in order to make us think that their $7k DAC would be a good thing.
If you want to investigate this further, I really recommend Monty's videos:
Digital Media Primer
Digital Show and Tell
Cheers,
Keith
A lot of the confusion and misrepresentation arises because of the oscilloscope model of the waveforms that is presented. It has it's uses but it actually misrepresents audio because it only loosly approximates to what we hear.
If you spend to much time looking at, and trying to analyse, oscilloscope waveforms then you go off on the wrong track.
A more appropriate view of the waveform would be the spectrogram because our ears and brains tend to resolve frequencies, not instantaneous sound level.
By the way, sound waves in air exist purely as areas of air compression and decompression (rarefaction). It's all relatively straightforward and there isn't the same duality as their is with light: there is no "sound particle".
A lot of the maths involved in audio (Nyquist, etc.) involves transforming various waveforms (such as oscilloscope-style time/amplitude waveforms) into the frequency domain so we can understand how it sounds.
If you take, as an example, a classic square wave:
http://mathworld.wolfram.com/images/eps-gif/SquareWave_700.gif
If you look at the oscilloscope display it doesn't look as if there is much audio information in there at all. Intuition suggests that there wouldn't be much to hear.
Intuition is misleading.
The reality is that square waves produce a huge (theoretically infinte) range of harmonics creating a very full "fat" sound.
http://ptgmedia.pearsoncmg.com/images/chap2_9780132349796/elementLinks/2-6.jpg
Rather than looking at oscilloscope-style displays, start thinking of it in terms of sine-waves. Sine waves are the building block of audio. Every sound can, ultimately, be considered to be constructed from combinations of sine waves of different frequencies and amplitudes.
When you see the jagged stepped sine waveform so often shown when talking about digital audio, your intuition tells you there is something missing, but that is incorrect. Your intuition is misleading you.
In actual fact the opposite is true. Our hearing lives in a world where everything has to be created from (or resolve to) a pure, complete sine-wave these steps simply cannot exist.There simply cannot be bits of the sine wave missing, because the smallest possible representation of sound is a complete sine-wave.
In fact, the stepped sine-wave diagram actually corresponds to a a sine wave with bits added in the form of higher-frequency harmonics.
But it's important to realise that the steps don't exist in the first place. The stepped waveform diagram is a misrepresentation of how sampling actually works. It's not an Engineering diagram, it's a marketing one used, primarily, to deceive people. It is a lie.
Because the liars, crooks, and con-men in the industry know how to manipulate our intuition in order to make us think that their $7k DAC would be a good thing.
If you want to investigate this further, I really recommend Monty's videos:
Digital Media Primer
Digital Show and Tell
Cheers,
Keith
Kumar, find a way to take a look at that video I linked. The practical demonstrations, especially of a 20kHz sinewave being perfectly reconstructed (give or take a bit of inherent low level noise in the analog test equipment), should reassure you.
There are plenty of places that subscribe to the more is better thing - places that aren't selling anything.
I will look at the videos too. More for understanding, less for reassurance. I was and remain convinced about the audible sound quality from my digital kit - regardless of the engineering reasons for that.
By the way, sound waves in air exist purely as areas of air compression and decompression (rarefaction).
Understanding just the quoted part, described also as vibrating air, was key for me to get my head around this. The sine wave representation of the sound wave actually misled me into thinking there needs to be information data points about every point on the sine wave line. Whereas this approach led to the realisation that all the information that exists is the amplitude of the vibrations and their frequencies. Sampling 40000 times captures all the information content about the amplitudes of all the vibrations up to 20000 times a second, i.e. 20khz. It is a 100% sample till there.
Because the liars, crooks, and con-men in the industry know how to manipulate our intuition in order to make us think that their $7k DAC would be a good thing.
$7k DAC? That is the bottom end model. I went to the products page only after reading about the $ 70K DAC here.
Who buys this stuff? And it doesn't even seem to have the fancy/heavy/milled out of one ingot - box that is the usual clothing for such gear.
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