Misleading Measurements

[March Audio]

55 minutes ago, lucretius said:

 

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

It's more than that, Bluesman has stated that you can't relate a digital signal level to an analogue voltage level.

 

Clearly this is incorrect.  You just need to know what the relationship is.

 

In the case of modern dacs the pretty much universal convention is that 0dBFS will equate to 2.8 v.  That will be 2 volts rms if replaying a sine wave.

 

There is nothing misleading about this as he erroneously claimed.

 

Ideallyca formal international technical standard would define this relationship but we don't have that.  Instead we have an industry adopted convention.

[March Audio]

43 minutes ago, bluesman said:

With all due respect, I never made such a statement - if you think I did, please quote it so we can all see and understand what you think you saw.

 

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):

 

 

If that digital signal doesn't "exist in isolation", what is it - a mirage? an imaginary construct?  fairy dust?  Files like this are instructions that are read by a computer in a stream, just as a phono needle reads a groove, a tape head reads the magnetic patterns on the tape, and a CD player's laser reads a disc. The digital words in that file tell a voltage modulation stage how to shape the voltage drop across its load so that the AC signal it creates is a model of the input signal coded in the digital stream.  But digital audio signals are not "turned into analog signals" - the digital signal lives in its folder in your computer and will be there for the life of the storage medium, even if you never play it.

 

I thought that perhaps you simply meant that the analog signal is a mirror of the digital data, until you said that digital signals don't exist in isolation.  They do - your statement of the opposite is just plain wrong.  Of course, digital signals "relate" (your choice of verb, and a poor choice of one in this case) to analog signals - the digital signal is a discretely sampled model of both the continuous function that was the input it modeled and the continuous function that is the output of the DAC stage reading its content.  But your statement and belief that "[d]igital signals are turned into analogue signals" is simply wrong.  They are totally separate entities.  It's actually the analog signal that doesn't "exist in isolation" because it only exists when the digital file instructs a DAC to create it.

 

I fear you may not completely understand all this, and I'm clearly not communicating well enough to help you do so.  Your unwillingness to consider the facts contributes to your continued resistance, and I don't want this to deteriorate further.  So I'm sorry but I'm not inclined to continue.

 

No problem, you said the following:

 

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.

 

I havent "co-mingled" digital signal levels with analogue ones at any point - I have only related one to the other.  You have conflated/misunderstood/misconstrued what has been said to imply that dBFS cant be used to reference an analogue voltage, which of course it can.  ie  At 0dBFS the analogue voltage output is 2.8 volts.

 

You also said:

 

"Digital signals are pages of words that can be read, duplicated, edited, printed on paper, archived, etc.  They aren’t turned into” analog signals"

 

So what does a DAC do if it doesnt turn digital word values into analogue voltages?

 

and you claim that I dont understand???? 😜

 

Hence, as I said earlier, I have no idea what point you have been trying to make, other than using semantics to be argumentative.  I think thats unfortunate, so yes lets please desist.

 

 

[March Audio]

52 minutes ago, stereo coffee said:

 

There are two realities and one fantasy here, as I see it. 

: 

Reality 1 is what is at the output of your RCA plugs on your CD player DVD player and is arranged to provide the D/a converter

in your player from a level that is already arranged that way  on the CD's you buy from your record store or online. Yes you can choose to use DAC's that internally amplify that further - but that is your choice, departing from what the industry sensibly has provided which is -10 dbv . the fact you can do it, does not mean everyone else is doing that too.  

 

Here are some articles discussing what is at the output of CD players not when being tested with sine waves, but when playing actual CD's.  https://www.sweetwater.com/insync/understanding-signal-levels-audio-gear/

           https://en.wikipedia.org/wiki/Line_level

           

 

 

 

It infers when a CD, lets say Fleetwood Mac Rumours is manufactured, the CD is held back in level so digitally the information only allows at the analog output stage of the player conversion to  0.316v RMS, 11.7825dBu because the CD manufacturer complies with rules concerning consumer line level. This is imminently sensible, as the CD label wants consumers to enjoy the CD in many forms of consumer equipment, none of which they have control over directly, 

 

The scope image level provided earlier by March Audio, which differed so much to my old school scope image  I suspect  shows his DAC is being switched for higher output,  which is a function usually of adding different resistance settings allowing gain to a inverting opamp expressed as R2/R1, Although his is the gustard X16 we can see the X22 model has feature of adjusting level.   In my test I ran

the output from my sony CD player via a RCA to BNC lead of very short length. 

 

But normal CD players do not have such a feature to increase level.   

 

The question to ask is,  can you presently buy Rumours on CD with 0DbFs   level available anywhere, and the answer is No because

the record labels are sensible with providing their product to match to consumer equipment. In my promised separate thread I have begun explanation of each type of equipment, but it is a work in progress for the moment.

 

Reality 2 ( A fantasy world, enticing some &  playing with mainly lots of visual treats, just look at those colours !  ) 

This is where forums like ASR are at presently, the reality see's pretension that when you play your CD at home via its RCA socket  it somehow gets close to the players 0DbFs level - reread Reality 1 , it is just not happening. At ASR review after review presenting sine waves,  as justifying music, and using Odbfs. Yes for hopefully the last time, players can reach such level, but its not being provided via a normal CD players RCA outputs, for very good reason, so as to reserve the player both digitally and via its analog output to have ability for dynamics in music which is the peak level. For consumer line level its 0.894V peak to peak   

 

Reality 3  ( very Real ) 

This is the levels used in recording studios, the levels here are very impressive but totally unrelated to consumer line level. The future of recorded sounds is in their hands, many fine engineers , and professionals getting it generally right, every day.   

 

Most of us live  with consumer equipment in Reality 1, this is for very good reason, as escaping or trying to escape  ( bypassing the fantasy of reality 2 )  to Reality 3 is full of pitfalls namely eternal frustration not being able to change the master tapes. A quick diversion to MQA - it will never be right as the original companding is not being matched.

 

Reality 1 is eminently sensible, and we can be thankful the CD industry chose this level as it enable the standard player we use to have reserve ability for dynamics, more on this in the promised thread. 

 

Hopefully that clears things up.       

 

None of this is correct.  Im not sure I have the energy to go through it in detail.

 

This is a rip of a Rumours CD ( a very old one, I think I bought it in the late 80's).  I looked at one track, Go Your Own Way and the peak value is -2.35dBFS.  So I dont know where you got -11dB from.

 

Most recordings are normalised to peak close to 0dBFS to maximise SNR.

 

You have been shown data that confirms literally hundreds of DACs conform to the industry standard of 2 v rms out at 0dBFS.  The X16 conforms to this standard.  There are no modern dacs or cd players that conform to -10dBv.

 

Again can you tell us which model of Sony CD player you have and we can look up the specs and check what output voltage it has?

 

Also could you please provide a list of albums and I will post the the amplitude stats for them like above.

[March Audio]

Just for giggles this is the 24/96 remaster of the same track.

 

Rammed up to -0.1dBFS!  So @stereo coffee, not 0dBFS but near as damn it. So it will be peaking 2.8 volts. The old Rumours cd at -2.35dB will peak 2.1 volts.

 

[March Audio]

2 hours ago, fas42 said:

Just bumped into this, on ASR ... https://www.audio “science” review/forum/index.php?threads/things-that-cannot-be-measured.20808/post-690068

 

By member Sawdust123 there, who just happens to be this chap,

 

 

Considering he's in the AP world, I suspect he knows a thing or two about the subject ... bits of a couple of recent posts,

 

 

and

 

 

 

 

Its a good video and raises some great points.  A few comments though.

 

He says towards the beginning that measurements are used to infer a subjective performance level.  Well yes, in simplistic marketing maybe, however thats not what measurements are for.  They are there to objectively assess various technical performance parameters.  The correlation of that data to subjective performance is very much in the domain of psychoacoustics.  As he pointed out, different types of distortion have different levels of subjective impact.

 

Secondly, again whilst in simplistic product specs you may only see a small number of test results, the tests any good designer will be making are much wider than the "single sine wave".  So I think thats a bit of a mis-characterisation.  Some of the more complex test he mentions are often used - you will even see them on ASR, multitone for example.

 

On the other side of things we have significant problems with casual subjective comparisons.  Audiophiles will perform comparisons without any controls in place.  They will see the products which will induce inevitable bias.  They wont accurately match volume which will lead to erroneous conclusions.  Every time I have tested audiophiles and put basic controls in place the previously heard significant differences in sound either evaporate or at least become much, much less significant.  Now dont take that to mean I think everything sounds the same - it absolutely does not, however I take peoples subjective opinions with a very large pinch of salt.

 

Finally one other thing to note is that we really can measure way beyond levels of audibility.  The video is really about types of tests and their interpretation rather than measurements being insensitive of ineffective.

[March Audio]

14 hours ago, lucretius said:

 

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.

Hi

 

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

 

You can see in the statistics how the peak sample value is +- 32768, the maximum possible 16 bit value. This is 0dBFS.  You cant exceed this.

 

 

 

A square wave will peak at 0dB also. In the picture below you can see ripples because we dont have the infinite bandwidth necessary to correctly reproduce a square wave. The ripples are hard clipped at 0dB.

 

 

 

When signal processing/upsampling yes the interpolation can create inter-sample overs.  However these are still clipped, they wont in practice be reproduced over 0dB.

 

So as you never know what the replay might end up doing in that processing you should never really record/normalise to 0dBFS.  With 24 bit recording there is never any real need to either.

[March Audio]

27 minutes ago, danadam said:

Are you saying that @lucretius is not correct or that the AES17 standard is not correct? :-)

The standard clearly says:

 

Its the interpretation of the words in the standard.  The standard is correct in what it says.

 

What its saying is that the analogue output is "full scale" (maximum output voltage measured in volts RMS) when the peak level of the sine wave reaches the maximum digital level (+ 32768 for 16 bit).  This is 0dBFS.

 

For a sinewave the relationship between the peak value and the RMS value is 1.414.  This is equal to 3.01dB.

 

The values cant go above 0dBFS

 

You can see this on the pictures I posted above.

[March Audio]

38 minutes ago, Bill Brown said:

This is getting bizarre, heck, borderline ridiculous.  @March Audio and @bluesman are absolutely correct.

 

There is a conversion of an analog signal to digital (ADC).  You CANNOT encode a music signal with one and zeroes above 0 dbFS, it will all be (terrible) noise, there is no headroom recording digitally.  In fact, you should not approach 0 dbFS, as they said, to avoid intersample overs when the ones and zeroes are converted back to analog (DAC).

 

When a DAC chip converts the digital signal back to analog (in a DAC designed for pros or consumers), it produces either a signal that could be described as current or voltage.  If the former, a current to voltage converter is the next stage (traditional), in the latter, this is done on the chip (modern).  This voltage then goes through additional gain stages in the DAC to either produce -10dbV from consumer DACs (typically, but not always), or +4dbU in professional DACs (more closely adhered to).  These levels are important to interface with subsequent/corresponding consumer or professional gear that is designed to accept the different levels for proper gainstaging (enough to drive to full output levels in preamps, then amps, but not overload them).  It really isn't complicated.

 

Bill

Yes, but (sorry) as discussed at length earlier in the thread there are no domestic dacs that output at -10dBV.  Most are close to 2 v rms at 0dBFS. This is +6dBV. (RCA output)

[March Audio]

4 hours ago, stereo coffee said:

Correct, and incorrect ,  the players certainly have that capability,  but the media is held back deliberately presently to sensibly allow dynamics in music we buy. This has nothing to do with how much level you CAN then add at your DAC or whatever, but is simply the level chosen by the music industry so that music we buy is music and not distorted audio. 

 

It is described here:

https://en.wikipedia.org/wiki/Loudness_war

 

The loudness war (or loudness race) is a trend of increasing audio levels in recorded music, which reduces audio fidelity and — according to many critics — listener enjoyment.

 

Companding is the future 

The larger picture is if you want to have greater dynamics without distortion , you need to use companding to do so, just as recording studios have done to preserve ... let me repeat that ... preserve  dynamics since 1965 - Dolby A - and always one step ahead DBX.  The DBX type 4 White paper gives good overview of the history of companding  + improvements that can be made to digital recording.   https://warehousesound.com/dbxtiv.htm

 

Media we buy would in theory contain the ORIGINAL companding used during recording, and would play back expanded the same audio - companding being a two part process.  It is though possible to enjoy companding real time, to experience this connect a late 1980's early 1990's DBX 150x to a 16 bit  CD player and compare it to a 20 bit player - you should find the 16 bit player is every part ( i was going to say bit )  as good as any given 20 bit CD player.  

 

 

 

 

 

 

 

 

 

Sorry  but again this is all incorrect.

 

In an earlier post I asked if you could provide a list of albums for which I would post the amplitude statistics.  I wanted you to specify the albums so I couldn't be accused of being selective.

 

Most albums are normalised close to 0dBFS to maximise the signal to noise ratio.

 

I will post some examples later.

[March Audio]

5 hours ago, lucretius said:

 

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.

 

 

 

 

Yes 2 volts rms.  However the peak voltage value of 2 volts rms is 2.8 volts

 

It is the peak voltage we have to accommodate.

[March Audio]

1 hour ago, bluesman said:

The entire document and its intent are confusing.  It refers to an analog input driving a digital device, which is already a bizarre and conflicting circumstance for a document that purports to be for the "measurement of digital audio equipment".  And it describes the input signal as "... the maximum analog signal that may be applied to the device for correct operation".   Pure digital devices do not have analog inputs, but the document specifies analog signal generators.  So there are ADCs and DACs in the mix to provide a digital input and an output signal with measurable voltage.  Remember that this document is 23 years old and was last revised in 2004.  Also remember that it is not a true standrad - it's neither enforceable nor used consistently throughout the industry.

 

Pure digital domains exist in which to generate, manipulate, and measure digital audio signals, e.g. MATLAB.  I don't know the current status of digital audio standards regarding pure digital platforms and devices, but MATLAB and similar schema really should be the platform on which future digital audio standards are developed and measured.  In this setting, 0 DB FS would truly be the maximum amplitude digital signal handled by the equipment under test (EUT).  It would not be measured in volts - it would be measured in digital code.

 

For example, in a 16 bit integer domain, digital signal value in dB FS = 20*log10(abs(value)/32768) because 16bit signed has values between -32768 and +32767, e.g. 2^15=32768 expressed in MATLAB code.

This whole conversation is about how the digital data interfaces with the analogue world so ADC and DAC are an intrinsic part of the conversation.

[March Audio]

28 minutes ago, danadam said:

My take on this: the -10 dBV is a line level, which is a different thing than max level. If wikipedia is to be believed, line level is:

and following to nominal level:

If we assume a headroom of 16 dB, then -10 dBV line level (i.e. nominal level) + 16 dB headroom = +6 dBV = 2 V max level. So in my understanding domestic DACs have both -10 dBV line level and 2 V max level.

 

But to be clear, I'm not saying that they should be tested at -10 dBV instead of 2 V or that the music content never reaches near those 2 V :)

 

There is no nominal level, it has to be related to a specified digital level and specified signal.

[March Audio]

6 hours ago, stereo coffee said:

Correct, and incorrect ,  the players certainly have that capability,  but the media is held back deliberately presently to sensibly allow dynamics in music we buy. This has nothing to do with how much level you CAN then add at your DAC or whatever, but is simply the level chosen by the music industry so that music we buy is music and not distorted audio. 

 

It is described here:

https://en.wikipedia.org/wiki/Loudness_war

 

The loudness war (or loudness race) is a trend of increasing audio levels in recorded music, which reduces audio fidelity and — according to many critics — listener enjoyment.

 

Companding is the future 

The larger picture is if you want to have greater dynamics without distortion , you need to use companding to do so, just as recording studios have done to preserve ... let me repeat that ... preserve  dynamics since 1965 - Dolby A - and always one step ahead DBX.  The DBX type 4 White paper gives good overview of the history of companding  + improvements that can be made to digital recording.   https://warehousesound.com/dbxtiv.htm

 

Media we buy would in theory contain the ORIGINAL companding used during recording, and would play back expanded the same audio - companding being a two part process.  It is though possible to enjoy companding real time, to experience this connect a late 1980's early 1990's DBX 150x to a 16 bit  CD player and compare it to a 20 bit player - you should find the 16 bit player is every part ( i was going to say bit )  as good as any given 20 bit CD player.  

 

 

As I mentioned here are a few examples of the signal levels on various albums.  All are at or coming very close to 0dBFS.  This means on a typical dac the RCA peak output voltage will be getting close to 2.8 Volts.

 

Bob Marley - Baylon By Bus

 

The audio data is usually normalised to come close to 0dB to maximise signal to noise ratio.

 

As previously mentioned this should really be looking at the true peak level which accounts for inter sample overs and can be higher than the actual sample value.  Unfortunately it seems that many mastering engineers dont understand this and ram it right up to 0dB.  To be safe you really should back off to maybe around -3dB to ensure there are very few clipped samples.

 

Bill Evans Live at the Montreau jazz Festival

 

Daft Punk Random Access Memories

 

Copland fanfare Fort The Common Man

 

 

[March Audio]

1 hour ago, stereo coffee said:

It is explained here in terms of streaming services, that levels are backed off to allow for dynamics in music. 

 

 

 

Sorry this is wrong.  As I keep demonstrating to you the signal levels are normalised to peak close to 0dB

 

You have got to stop thinking about this in terms of music.  Music cannot be used to quantify signal levels.  You have to use consistent signals of known value.

 

A quiet piano concerto will have very different RMS level to a highly compressed death metal track.  They will boith peak close to 0dBFS.

 

 

Just to add to this you may choose to adjust signal levels to make different tracks/albums etc have a similar perceived volume level, but this is not relevant to the discussion here.

[March Audio]

21 minutes ago, bluesman said:

...which is exactly why AES 17-1998 is confusing many in this thread. Although entitled “measurement of digital audio equipment”, it is based entirely on measuring analog parameters at the ADC input and DAC output.  The amplitude of a pure digital signal is determined by its coded content, which is read rather than measured.  A 0 dB FS digital signal in a digital domain is defined as the highest level achievable by the EUT.  For a 16 bit signal that would be the coded word 32768.

This will be translated into an analog signal by a DAC in the course of making sound from it. But the voltage of that signal will be determined by the design of the DAC, not by an AES standard.  It is usually in the range bandied about in this portion of the thread.  But it differs from device to device by design, and the range of usual values is higher for commercial equipment than for most consumer devices.

No it's not, its relating digital levels to analogue levels.

 

Also its not a standard.  It's a method that has no formal implementation or AFAIK in recognised formal international standards bodies.

 

As a method it is correct. The dac or adc is actually irrelevant.  The method is applicable to all.  It doesn't specify analogue voltage levels.

 

It's implicit that an audio has to be converted to and from the digital domain so I'm really not sure what your issue with it is.

 

 

[March Audio]

11 minutes ago, bluesman said:

You really don’t understand this.   0dBFS (Full Scale) is the clipping point for a signal in a digital audio product. Rather than measuring from the noise floor up, digital signals are measured (or referenced) from full scale down. A 0dB FS (Full Scale) signal contains the maximum amount of digital information that can be used to represent the signal being defined.  
 

The output of a DAC driven with a 0dB FS signal should be at the full electrical potential of the device - this is the voltage drop across the load presented by the next stage’s input.  There is no standard for this value - it is determined by the design of the DAC and it is independent of the digital signal, whose “amplitude” is determined by the digital code it contains.  The same 0 dB FS signal from a digital mixer will drive different DACs to different voltage levels.

 

I’ll say that again - the same 0 dB FS digital signal will produce different output signal levels from different DACs. The digital signal level is proportional to but otherwise entirely independent of the amplitude of the analog signal generated by a DAC.  It is the design of the DAC that determines the voltage of the output it creates from the instructions coded in the digital signal at its input.

We have been round this already.  It is your misunderstanding.

 

You seem unable to separate the concepts of relating one measurand to another, and one being the other.

 

 

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.

 

The design or actual output voltages of the DAC are irrelevent to the statement above.

 

 

[March Audio]

2 hours ago, bluesman said:

But that output amplitude is NOT the same value for every DAC that reads the same 0 dB FS digital signal - there is no specific "equivalent" full scale analog output voltage for a 0 dB FS digital signal because there is no consistent mathematical relationship of any kind between the digital signal being read by the DAC and the analog signal it creates from those digital instructions excepet for the spectral content. Every DAC design outputs the same analog waveform (+/- whatever distortions it intriduces) but at a level determined ONLY by its own design.

 

Whether you think that the analog output signal's peak amplitude is mathematically related to the digital signal or that the analog output signal is the digital signal in a different form (and I can't figure out which of those you actually do believe) doesn't matter because neither is correct.

 

 

 

The AES document *does not* say there is.  It is just relating one to the other.  It relates the analogue voltage peak (whatever that might be) to the digital full scale level.

 

It says nothing about what voltage levels.  It doesnt need to.  The statement is applicable to any system and any voltage.  It is just defining the relationship.

 

You need to re-read what is says.

 

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.

 

 

[March Audio]

1 hour ago, lucretius said:

 

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

 

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.

[March Audio]

1 hour ago, lucretius said:

 

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).

 

 

 

Yep this is pretty much it 😀

 

Im not quite clear on the "zeroing" (bit been a long day my fault).

 

Found this, might help someone

 

https://goodcalculators.com/rms-calculator/

 

 

 

[March Audio]

Well this is one misleading measurement.  Apparently the Sennheiser HD600 and HD650 sound the same.

 

Check out the ASR thread.

 

 

 

 

[March Audio]

19 minutes ago, Miska said:

 

Which is one big source of problems (inter-sample overs), since this is typically done at the sampling rate. When such data is oversampled and/or converted to analog correctly (not case with all oversampling DAC chips) such will commonly exceed 0 dBFS of the source sampling. Because peak sample values at low sample rate (signal is close to Nyquist) rarely coincide actual peaks of the signal...

 

This difference can be up to +3 dBFS without source signal being clipped.

 

 

DSD is not bound to 0 dB scale, but instead the specification allows short term peaks of +3.15 dB. How this is handles varies by DAC chip. For example ESS Sabre scales 0 dBFS PCM = 0 dB DSD meaning that with DSD sources output level could be higher than with PCM sources. AKM chips is DSD Direct mode have output level of 0 dB DSD = -3.5 dBFS PCM. With TI chips the exact DSD output level depends on the selected analog filter.

 

Yep, mentioned this issue earlier ;)