Misleading Measurements

[lucretius]

8 hours ago, firedog said:

I'd prefer to find a unit that measures well and sounds good - and not spend my money on euphonic distortion. 

 

Without euphonic distortion, all DACs and amps sound the same.

[lucretius]

1 hour ago, March Audio said:

I think it simply a case of Stereo Coffee thinking that the "consumer level" convention of -10dBv (0.316v rms) - and I say convention because I dont think its been enshrined in any formal standard - is still being used.

 

Correct me if I'm wrong, but there is a big difference between nominal output voltage (e.g. 0.316v rms) and maximum output voltage. The spec sheets for consumer equipment* these days show the maximum output voltage,  somewhere around 2 Vrms (sometimes a little less and sometimes a little more).

 

*My Mytek Brooklyn DAC+  apparently has a maximum output (at 1kHz) of 9.84V from the balanced outputs.

[lucretius]

3 hours ago, March Audio said:

This will be a typical peak voltage of around 2.8 volts.

 

I know that you know this but for others*:  the RMS voltage (V_RMS) is determined by multiplying the peak voltage value by 0.7071.

 

* I had to look this up :)

[lucretius]

36 minutes ago, bluesman said:

I don’t want to continue beating my head against a stone wall here.  But the most recent posts have injected a new level of misinformation and confusion that has me screaming inside my head - and I have to relieve the pressure.

 

You can not commingle traditional dB measurements and terms with dBFS (or dB FS, since the omission of the space upsets one participant despite the fact that almost everyone omits it). The “full scale” unit of measurement is ONLY used for digital signal levels in digital equipment.  It is a totally different metric from the rest, which are to be used ONLY for analog signals in analog equipment.

 

0 dBFS is the maximum possible level in a digital audio signal.  There is no + side on the scale because it’s a rigidly defined metric whose ascending scale ends at 0.  There is no headroom above 0,  there is only a scale below it.  There is no “line level” standard for it and it is not useful in any way in listening to, calibrating, measuring, describing, or designing analog equipment.

 

The full scale metric is potentially useful to audiophiles ONLY in systems that are fully digital to the output stage.   Th level of a digital signal is not measurable in volts because it’s not an electrical signal - it’s a series of numbers.  It is it a continuous measure - it’s purely binary, and it cannot be amplified.  Level changes are effected only by interchanging 0s and 1s in the strings of them that ARE the signal.  
 

Unlike analog signals, digital signals have no property that can be used directly to generate electrical or physical output.  Electrical power and its components are measurable and controllable properties that define an analog audio signal and are manipulated to turn the low amplitude outputs from source devices into signals identical in nature and configuration but with sufficient power to push them into your ears and brains.  Digital signals have no such properties and require conversion to analog representations so we can hear them.

 

Once you drop a digital device into a signal path, you throw all convention out the window.  No measurement system exists to define signal levels.  And once you go completely digital (to the output stage, since there has to be D-A conversion to make our primitive brains perceive the information as sound, you can no longer use any of the metrics that have some of you so upset.  
 

Analog metrics are only useful in analog systems to measure analog signals.  Once you inject any interconversion to and from the digital domain, there are no applicable metrics or standards.  And the entire debate here over line level is both irrelevant and erroneous except when discussing purely analog signals and systems.  That’s why I used the term analog as a descriptor in every post on this topic in this thread.  

 

I agree with the above.  Nonetheless, even in the pre-digital world, the nominal levels mentioned (i.e.  −10 dBV  and +4 dBu) defined only the zero reference standard (in the pre-digital world) and not the maximum line output voltage that should come from source equipment. Further, you needed sufficient headroom above this zero level to allow transient audio peaks to exceed the nominal level without damaging the system or the audio signal, e.g., via clipping. (You could go above or below zero reference by some amount as dictated by the dynamic range of the equipment.)  AFAIK, the maximum line output of consumer source equipment was always greater than the nominal/zero reference (for the pre-digital world).  Could be as low as 0.5 Vrms -- I've seen a lot of consumer equipment (pre-digital) where it was 1 Vrms.  When CD players came along, that maximum line output for consumer equipment tended towards 2 Vrms (following Sony, I suppose).

 

 

[lucretius]

7 hours ago, bluesman said:

They aren’t “turned into” analog signals - they’re used as a template for the creation of de novo analog signals

 

Seems like a disagreement due to word choice and not due to any substantive disagreement.

[lucretius]

2 hours ago, bluesman said:

A digital signal file is indeed a discrete entity that exists "in isolation",  despite your erroneous insistence a few posts above that "[d]igital signals are turned into analogue signals. They don't exist in isolation".  SInce you don't seem to understand what a digital signal is, here's one you can see in isolation - it's a snippet of an mp3 (shown in hex format):

 

 

 

 

To be clear, the signal is the representation (just the information*, an abstraction), not the physical manifestation.  Then, signals do not exist in the same sense as physical objects.

 

*For example, the same signal can be physically represented in many ways.

[lucretius]

4 hours ago, lucretius said:

*For example, the same signal can be physically represented in many ways.

 

Indeed, the same signal can be physically represented in both a "digital" and in an "analog" way.  A DAC converts one physical manifestation of a signal into yet a different physical manifestation of the same signal. The resulting signal (in its physical analog representation) coming from the DAC is not a "de novo" analog signal (that would suggest it was created from nothing).

[lucretius]

17 hours ago, bluesman said:

You can’t listen directly to a digital signal.

 

The same is true of an "analog" (representation of a) signal coming from a DAC (or from a turntable for that matter) -- it's just a pattern of voltages, not sound waves.

[lucretius]

On 4/30/2021 at 11:31 AM, bluesman said:

And with digital signals, there is neither headroom nor clipping as we understand them in the analog world.

 

Some observations:

1. The AES standard (AES17) defines 0 dBFS as the RMS value of a full-scale sine wave.  Because the definition of full scale is based on a sine wave, it will be possible with square-wave test signals to read as much as + 3.01 dBFS.
    
2. It's true that the value in dBFS does not relate directly to the original absolute sound pressure level of the audio measured in dB. Nonetheless, we do convert between digital and analog levels, although no single standard/convention exists. Thus, a potential for ambiguity exists when assigning a level on the dBFS scale to a waveform; some engineers (and software programs) use a conventional RMS calculation leading to a full scale sine wave reading -3.01 dBFS (which is incorrect per the AES standard), while other engineers choose the reference level so that RMS and peak measurements of a sine wave produce the same result.
    
3. Peak meters in some DAWs and digital mixing systems indicate the peak among the sampling points, not the resulting waveforms. The resulting waveforms represented by the samples may exceed full scale. That is to say, a digital signal that does not contain any samples at 0 dBFS can still clip when converted to analog form due to the signal reconstruction process interpolating between samples.

[lucretius]

6 hours ago, March Audio said:

Just on point 1, its not quite correct, sorry :).  The waveform shape makes no difference as it is the peak voltage level that is 0dBFS.  With a sinewave the RMS level will be - 3.01dB

 

But doesn't the standard say to set this max RMS level as the reference -- 0 dbfs?  That is to say, the 0 dB reference for either peak OR RMS measurement is that of a sinewave at full scale.  (And isn't the case that waveform shape does make a difference when it comes to computing RMS?)  Seems to be a lot of discussion about this on the net, mostly confusing.

 

 

 

 

[lucretius]

3 hours ago, bluesman said:

We need to be honest and clear when discussing things based on a well defined standard.  First, #1 is not your observation - you're quoting definition 3.4 in the very AES standard under discussion.  Here's a link to the actual document, so everyone can read it.

 

Actually, I looked a a bunch of documents but I didn't have access to the AES standard. Thanks for the link.

 

3 hours ago, bluesman said:

Second, while it's true that a square wave contains more energy than a sine wave of the same frequency and amplitude, the standard specifically tells us in the ssentence after the one you offered as your observation that only a sine wave is to be used, specifically:  "Square-wave signals at this level are not recommended because tilt or overshoot introduced by any filtering operations will cause clipping of the signal".

 

Yes.  Only a sine wave is to be used for setting the zero reference.  But music is not sine waves. So the point of bringing up the square waves is to show that the meter could read higher than 0 (using an RMS meter?).

 

3 hours ago, bluesman said:

Third, "full scale amplitude" is defined clearly as the "amplitude of a 997-Hz sine wave whose positive peak value reaches the positive digital full scale, leaving the negative maximum code unused". It isnnot defined as the RMS value of that signal - it's defined on the basis of peak value.

 

Yes.  Then the standard also says this:

 

5.4

Input for full-scale amplitude

 

     NOTE The characteristic to be specified is the analog signal voltage required to reach digital clipping under normal device settings.

 

In systems where the output is accessible in the digital domain, the input for full-scale amplitude shall be the r.m.s. voltage of a 997-Hz sine wave that shall be applied to the input to obtain a digital signal whose positive peak value reaches the positive digital full scale.

 

and this:

 

6.3 Output amplitude at full scale

 

In systems where the input is accessible in the digital domain, the output amplitude at full scale shall be the r.m.s. voltage that results from a sine wave whose positive peak value reaches the positive digital full scale under normal settings of gain controls.

**************************************

 

Color me confused.  Perhaps someone can explain to me what is meant in 5.4 and 6.3 above.

 

[lucretius]

7 hours ago, pkane2001 said:

 

Didn't read the document, but from your quote, it's just saying that analog input/output required to produce a full-scale signal (at input or output) is to be reported as the RMS voltage of a sine wave, rather than its peak value. For example, when 0dBFS is reported to require 2v input, this means 2v RMS, not peak. The actual voltage that produces 0dBFS is then the peak value of about 2.8v.

 

 

That's how I read it.  However, if we set 2v RMS* = 0 dB FS, then the peak value is 2.8v only for the 1k sine wave, Music can actually produce a greater peak value than 2.8v, meaning you could have peak values greater than 0 dB on the meter (which of course is measuring the analog peak).

 

*assuming this is max RMS

 

[lucretius]

4 minutes ago, pkane2001 said:

 

As @March Audio said, the RMS value varies with the waveform. Peak value corresponding to 0dBFS is the maximum output.

 

Actually. it's peak value corresponding to full scale sine wave.

[lucretius]

5 hours ago, March Audio said:

Its better not to think of it in relation to the RMS of a sine wave.  Just accept the peak value cannot go past 0dBFS which is the highest digital level (+32768 in 16 bit system).  There are no more numbers to describe the input value so it simply cant be higher.

 

Yes a different signal to a sine may have a higher or lower crest factor, but this just means you will have a higher or lower RMS value. A sine wave rms value is 0.707 of the peak, but for a triangle wave the rms value is 0.577 of its peak.

Whatever the peak signal level still cant go past 2.8v, it will just clip. 

 

To look at the statement again:

 

the output amplitude at full scale shall be the r.m.s. voltage that results from a sine wave whose positive peak value reaches the positive digital full scale

 

Its saying that the analogue full scale voltage in RMS is defined by the peak level of a sine wave equalling the maximum digital level.

 

Thanks.  I think I am beginning to see it. For 2 vrms (sine wave) which = 2.8 v peak, if we set that to  0 dB FS, then for a square wave, the rms = peak = 2.8 v.  We can never have a peak higher than 2.8v.  However, rms can vary from 2.8 v and downwards for different waveforms. Is this correct?  And do we zero the rms meter? (It would seem that we would need to zero it at 2.8 -- this would mean that both peak meter and rms meter get zeroed at 2.8).

 

 

 

[lucretius]

I think I'm going to order this book:

https://www.amazon.ca/Audio-Dictionary-Glenn-D-White/dp/0295984988/ref=cm_cr_arp_d_pl_foot_top?ie=UTF8