[pinnahertz]

 
About "speed": it may be speed of digital stream. It defined by band of discrete signal in analog form.
Analog form here it is physical realization (in electricity) of digital signal.
But this meaning of "speed" term is not applicable to analog audio signal.

It's a wrong usage of the term. He used the wrong term to describe something subjective.

 

[Yuri Korzunov]

   
The issue of analogue filters was effectively solved with oversampling...

 
Absolutelly..
 
In music production used ADC. There high sample rates of capturing also need for solving analog filter issue.
 
If there used downsampling, higher sample rates allow use digital filters with lesser ringing.
 
Therefore why need decreasing of sample rate to distribute to end-user?

 

[gregorio]

 
[1] In music production used ADC. There high sample rates of capturing also need for solving analog filter issue.
 
[2] If there used downsampling, higher sample rates allow use digital filters with lesser ringing.
 
[3] Therefore why need decreasing of sample rate to distribute to end-user?

 
1. Yes, I was referring to the analogue filters in both DACs and ADCs.
2. Theoretically yes but that's not the point of oversampling in ADCs. Oversampling solves the issue of analogue anti-alias filters and also improves SNR.
3. In a sense your question is actually backwards! The typical oversampling at very high (multi mHz) rates in ADCs can only be achieved with very limited bit depth resolution. The process within an ADC is not one of simply down-sampling but of (in effect) exchanging sample frequency for bit depth. We can't simply distribute this high sample rate/low bit depth to consumers because we can't process high sample rate/low bit depth, so we have to convert to a standard PCM rate in order to mix/produce and master a recording. Then the question becomes the reverse of your question; Why up-sample (and bit depth reduce) the master to distribute to the end-user?
 
G

 

[Yuri Korzunov]

   
Why up-sample (and bit depth reduce) the master to distribute to the end-user?

 
 
In my opinion, better way is distributing master as is (better in float point format - if DAW project was done so).
 
DAC may have different distortions for different resolutions.
 
For using better resolution mode of the DAC, we may adjust resolution of master audio file. Also PCM to DSD  or back may be changed.

 

[Argyris]

I found the "Hi-Res" badge on my HD 600 box a bit amusing, since it's on a headphone from 20 years ago, not some new, mondo expensive four figure flagship. It'd almost be better if they didn't certify older designs from a business perspective. Make people think that the badge actually means something and that they have to spend megabux in order to obtain equipment that carries it. Personally I'm a lot more concerned with what the response between 20 Hz to about 18 kHz looks like, since that's what I can actually hear, and above about 22 kHz is where all my music taps out, anyway.
 
A much more useful certification would be for headphones whose deviation from a standard target curve is less than +/- 3 dB, with some grace given for bass and upper treble extension (open dynamics all have trouble with the former, and the latter isn't easy to measure). Of course that would require having a standard curve everybody agrees on, which won't happen anytime soon (if ever), but some variation of the so-called Harman curve (it has a proper designation now, but damned if my Google-fu can turn it up at the moment) would be a good place to start.

 

[StanD]

  I found the "Hi-Res" badge on my HD 600 box a bit amusing, since it's on a headphone from 20 years ago, not some new, mondo expensive four figure flagship. It'd almost be better if they didn't certify older designs from a business perspective. Make people think that the badge actually means something and that they have to spend megabux in order to obtain equipment that carries it. Personally I'm a lot more concerned with what the response between 20 Hz to about 18 kHz looks like, since that's what I can actually hear, and above about 22 kHz is where all my music taps out, anyway.
 
A much more useful certification would be for headphones whose deviation from a standard target curve is less than +/- 3 dB, with some grace given for bass and upper treble extension (open dynamics all have trouble with the former, and the latter isn't easy to measure). Of course that would require having a standard curve everybody agrees on, which won't happen anytime soon (if ever), but some variation of the so-called Harman curve (it has a proper designation now, but damned if my Google-fu can turn it up at the moment) would be a good place to start.

Don't worry so much, in a few years you won't be able to hear much above 12 kHz, so enjoy it while you can. Besides, generally headphones are wonky (FR) if not rolled off above 12 or15 kHz or so. So one might ask how important is above that to really enjoy listening to music and how much of that spectrum is just plain tizzy fizzy sound. Back in the day of CRT TVs, when I walked into a TV store I would nuts from the sound of the chorus of horizontal sweep oscillators and deflection circuits. If I remember correctly the HD600 takes a nosedive before12 KHz and yet they sound pretty darned good.
http://www.innerfidelity.com/images/SennheiserHD600.pdf

 

[U-3C]

Speaking of which, I remember someone mentioned something about headphone frequency range specs being vague or misleading in the HD600 thread a while ago. I can't remember it clearly nor can I find it.

However, I am curious about how it works though. Do you guys know how the frequency range that are often put on the boxes of headphones are measured and what they mean? As well as how much it can or cannot be used as a reference for anything?

The post I mentioned was something about how in any practical real world scenario, the headphones can't reach anywhere near the published specs, but since everyone posts these numbers, Sennheiser fire the same to not look bad. Someone was kind of upset about Sennheiser "lying." Don't know anything about this stuff so I can't really find the words for a proper sounding question, but I am curious about how it works.

 

[Yuri Korzunov]

Do you guys know how the frequency range that are often put on the boxes of headphones are measured and what they mean? As well as how much it can or cannot be used as a reference for anything?

 
I suppose, primarily with panoramic auto SPL meter (acoustic pressure method) in anechoic room.
 
As test signal used sweep tone.
 
May be somebody use other methods.
 
Deviation 1 dB about average value in full band is good for electro-mechanical system.

 

[gregorio]

  [1] In my opinion, better way is distributing master as is (better in float point format - if DAW project was done so).
 
[2] DAC may have different distortions for different resolutions.

 
1. The question would then become what is "as is"? At a certain level, this question is unanswerable, even by the mix and mastering engineers who made the recording! The other issue is, what's the point? If the recording ends up using say 12bits of dynamic range and has no useful information above 22kHz, what would be the point in distributing it in a 384kS/s 64bit container?
 
2. I would expect those distortions to be inaudible, otherwise I would consider that DAC to be faulty/very poorly designed.
 
G

 

[StanD]

Speaking of which, I remember someone mentioned something about headphone frequency range specs being vague or misleading in the HD600 thread a while ago. I can't remember it clearly nor can I find it.

However, I am curious about how it works though. Do you guys know how the frequency range that are often put on the boxes of headphones are measured and what they mean? As well as how much it can or cannot be used as a reference for anything?

The post I mentioned was something about how in any practical real world scenario, the headphones can't reach anywhere near the published specs, but since everyone posts these numbers, Sennheiser fire the same to not look bad. Someone was kind of upset about Sennheiser "lying." Don't know anything about this stuff so I can't really find the words for a proper sounding question, but I am curious about how it works.

You can look at http://www.innerfidelity.com/ for articles. Measuring a headphone's FR is tricky due to each human being's head's varying acoustic properties, sound absorption, etc. They use fake heads, even the placement of headphones on a head causes variances. Then there's compensation curves.
 
http://www.innerfidelity.com/content/approaching-head-measurements#QpcMbwiP8CL3f35P.97  Includes the HD600
 
http://www.innerfidelity.com/content/headphone-measurements-explained-frequency-response-part-one#s82iihwrxudtWWTD.97
 
http://www.innerfidelity.com/content/first-crunched-data-harman-head-measurement-session#rO7ojy3y0vXww99T.97

 

[gregorio]

  Make people think that the badge actually means something and that they have to spend megabux in order to obtain equipment that carries it.

 
That's the whole point about HD/HiRes audio, it doesn't actually mean anything, it's ONLY about making people think it does. The typical studio tools and workflow mixes up all the bit depths and sample rates and that's even without considering if all those bits and sample frequency are being used in the first place!
 
G

 

[Yuri Korzunov]

   
1. The question would then become what is "as is"? At a certain level, this question is unanswerable, even by the mix and mastering engineers who made the recording! The other issue is, what's the point? If the recording ends up using say 12bits of dynamic range and has no useful information above 22kHz, what would be the point in distributing it in a 384kS/s 64bit container?
 
2. I would expect those distortions to be inaudible, otherwise I would consider that DAC to be faulty/very poorly designed.
 
G

> The question would then become what is "as is"?
 
In format od DAW project.
 
> what would be the point in distributing it in a 384kS/s 64bit container?
 
384kS/s it is 384 kHz?
 
Implemetation conversion high resolution to low sample rates is too complex and have more ringing due lack inaudible band, that may be used as transient band.
 
Better way purchase high resolution files and convert it to DAC's optimal (minimum distortions) resolution.
 
 
 
> I would expect those distortions to be inaudible, otherwise I would consider that DAC to be faulty/very poorly designed.
 
Achieving of absolutelly identical features for 2 instances of product or its modes is technically impossible.
 
But probably ideal products and developers exists somewhere

 

[gregorio]

  [1] In format od DAW project.
 
[2] 384kS/s it is 384 kHz?  
[3] Implemetation conversion high resolution to low sample rates is too complex and have more ringing due lack inaudible band, that may be used as transient band.
 
[4] Achieving of absolutelly identical features for 2 instances of product or its modes is technically impossible. But probably ideal products and developers exists somewhere

 
1. There isn't really a format of the DAW (or DAWs). The mix only exists virtually, in RAM, when it's time to print the mix, then a format is chosen and the mix is truncated to 24bit or dithered to 16bit. The only way of having the mix in the DAW format is to own that DAW, all the software plugins employed in that mix and of course the DAW session. That would likely be very expensive and the producer/mix engineer will not generally allow a copy of the DAW session to leave the studio.
2. 384,000 samples per second.
3. I don't understand what you mean.
4. Not sure what you mean here either. Absolutely identical or ideal isn't required.
 
G

 

[Yuri Korzunov]

   
1. There isn't really a format of the DAW (or DAWs). The mix only exists virtually, in RAM, when it's time to print the mix, then a format is chosen and the mix is truncated to 24bit or dithered to 16bit. The only way of having the mix in the DAW format is to own that DAW, all the software plugins employed in that mix and of course the DAW session. That would likely be very expensive and the producer/mix engineer will not generally allow a copy of the DAW session to leave the studio.
2. 384,000 samples per second.
3. I don't understand what you mean.
4. Not sure what you mean here either. Absolutely identical or ideal isn't required.
 
G

1. Yes. DAW is big complex of software and hardware. There may be different interfaces between modules.
 
3. Downsamplig filter to 44 kHz have 2 kHz transient band for -200 dB digital filter. For minimization ringing need so tricks for design such filter. If sample rate 192 kHz, filter may have transient band 20 .. 96 kHz. That significantly simplify design for minimization of ringing.
 
 
4. > I would expect those distortions to be inaudible, otherwise I would consider that DAC to be faulty/very poorly designed
 
What there border in numbers between "faulty/very poorly designed" and "good designed"?

 

[LazyListener]

  First need define: what is "detail"/"resolution" in common terms that defined mathematically.
 
Are you agree that unchanged signal passed thru audio system keep 100% input information (100% of details) at output?

I'll leave it to you "objectivists" to define what detail or resolution translates to in terms of math or objective measurements.  For me, it means hearing everything that can be heard based on the original recording.  Some headphones are better at revealing the detail that is there, while others are less able to do so.  Many factors at play.  Some have to do with the headphone.  Others with the listener.  So just focus on the headphone and the equipment.
 
I'm not sure what you mean by "unchanged" signal?  I do not agree that 100% input information (details) is always revealed at output.  Each output device (headphone) varies in its ability to reproduce the original signal.  Each headphone "distorts" the original signal in some way.  A headphone only provides a representation of the input signal, not an identical reproduction.