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Misleading Measurements


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43 minutes ago, stereo coffee said:

The fuss is IMO about levels we find available on CD's DVD's blurays etc . The levels being measured with equipment bear no relationship at all,  being absurdly 6.33 times higher. 

 

Mike Oldfields Amarok is the loudest CD I own, but it still has peaks that accommodate consumer line level.    Hence the effort I consider is to start seeing reviews of equipment that match the variety of media we enjoy. We are hopefully collectively now able to see the fabrication of measuring equipment at 0DBfs that is NOT consumer line level we enjoy from our CD's DVD's etc which is 0.316V RMS. Yes sure the equipment can reproduce such level with test CD's , but few of us would enjoy or have in our collections sine waves at 0DBfs, rather what we have is CD's DVD's etc with music,  that rigidly sticks to consumer line level.  

 

Below is Mike Oldfields Amarok, we can see the level gets to extend to the absolute peak to peak value of consumer line level, which is  0.894V, Vpp , and as can be seen explains visually why it is the loudest CD I own    

 

The secondary fuss is about sensitivity of power amps also matching to consumer line level. If this is done we can then just use resistance attenuation in between, and get in theory as close as possible to what our source has with capability at its output RCA.   We should instead be talking about ideal shunt and series resistances - as  Ike Willis said in Frank Zappa's Thing fish ... moving the project forward 

Screenshot from 2021-04-30 12-12-37.png

 

Can you tell us what CD player you are playing that back on and how you have calibrated Audacity to read volts?

 

Have you just loaded that track into audacity without actually playing it back from the CD player and calibrating to read volts?

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5 minutes ago, March Audio said:

It sounded just fine.

 

Here is Amorok, I didnt search out the loudest segment just grabbed a small section around 6 minutes in.

 

767mV rms, 5.11 mV pk to pk

 

I could easily find louder recordings than this.

 

20210430_090433.thumb.jpg.23434dbcb828ffe6d8661d8fd15fbf55.jpg20210430_090439.thumb.jpg.70f21f53cb149a6e187981f08977f192.jpg

 

 

But was that from the CD player output, or from your preamps output ?  You should find direct from your CD player it remains within consumer line level.  

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On 4/21/2021 at 7:18 PM, stereo coffee said:

Perhaps the most misleading thing is not measuring equipment with respect to recognised standards that the majority of home audio equipment complies with.  If we observe for example consumer line level which is nominal 310mv RMS . .... until they get that right I would suggest ignoring all measurements that fail to understand consumer line level, as anything higher is meaningless with respect to equipment you use every day.  

 

Going back to your original post here ... could you state precisely what you hear, in the music replay you listen to, that provoked you to make that point?

Frank

 

http://artofaudioconjuring.blogspot.com/

 

 

Over and out.

.

 

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23 minutes ago, stereo coffee said:

But was that from the CD player output, or from your preamps output ?  You should find direct from your CD player it remains within consumer line level.  

As the picture shows it was directly from the rca socket on the dac.

 

Please take on board all the evidence shown. As I mentioned please go to ASR and look at the hundreds of dac tests that show they nearly all have a 2 v rms output (or very close to) at 0dBfs and that none have 316mV output at this level.

 

This means that with music, outputs will peak up to about 2.8volts or 5.6 volts pk to pk.  The RMS level will be dictated by the music style, content and recording techniques (compression)

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My scope is old school, but I have it selected to display 0.5v per division @ 0.5ms we can see playing Amarok it occupies close to 2 divisions on peaks  meaning audio signal from my Sony Cd player is outputting as expected at 0.316v RMS and 0.894 Vpp

IMG_1712.JPG

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6 hours ago, March Audio said:

As the picture shows it was directly from the rca socket on the dac.

 

Please take on board all the evidence shown. As I mentioned please go to ASR and look at the hundreds of dac tests that show they nearly all have a 2 v rms output (or very close to) at 0dBfs and that none have 316mV output at this level.

 

This means that with music, outputs will peak up to about 2.8volts or 5.6 volts pk to pk.  The RMS level will be dictated by the music style, content and recording techniques (compression)

The DAC tests are as I see, nothing more than  1khz sine waves, hence alien & unrelated to consumer line level music signals.  

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28 minutes ago, stereo coffee said:

The DAC tests are as I see, nothing more than  1khz sine waves, hence alien & unrelated to consumer line level music signals.  

They are at 0dBFS.  Most recordings are normalised so they peak close to 0dBFS as this provides the highest signal to noise ratio.

 

Therefore music recordings will peak close to 2.8 volts or 5.6 volts peak to peak.  The rms level will be dictated by the music style, content and recording techniques (compression).

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2 hours ago, stereo coffee said:

My scope is old school, but I have it selected to display 0.5v per division @ 0.5ms we can see playing Amarok it occupies close to 2 divisions on peaks  meaning audio signal from my Sony Cd player is outputting as expected at 0.316v RMS and 0.894 Vpp

IMG_1712.JPG

As already explained you can't use music to accurately measure the line level output voltage.  Was that even a loud section of the track? 😉

 

What model cd player is it?

 

Even the very first cd player, the Sony cdp101 had a line output of 2 volts.

 

Screenshot_20210430-162608_Chrome.thumb.jpg.9fd7ef1e42459aa0b8d3c5347b109364.jpg

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23 minutes ago, March Audio said:

They are at 0dBFS.  Most recordings are normalised so they peak close to 0dBFS as this provides the highest signal to noise ratio.

 

Therefore music recordings will peak close to 2.8 volts or 5.6 volts peak to peak.  The rms level will be dictated by the music style, content and recording techniques (compression).

And no dynamics, and sounding squashed will be the direct result. Why ASR is pushing this agenda is the question to ask, as it has nothing to do with enjoyment of music. 

 

I will start a new thread, as its a fascinating subject, there i will provide answer to what is needed.  

 

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3 hours ago, stereo coffee said:

And no dynamics, and sounding squashed will be the direct result. Why ASR is pushing this agenda is the question to ask, as it has nothing to do with enjoyment of music. 

 

I will start a new thread, as its a fascinating subject, there i will provide answer to what is needed.  

 

On what basis do you make this claim?  Can you explain in more detail?

 

There is no agenda.  I am at a loss as to why you wont accept the fact that the signal voltages are higher than you believed.  I am also at a loss as to why you think that directly leads to "no dynamics"  Its just a higher voltage.  The dynamics are the same.

 

Please do start a new thread as I think we have probably exhausted discussion here and your beliefs are not relevant to the topic of "misleading measurements".

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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.   The 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 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.  

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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).

 

 

mQa is dead!

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1 hour 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.   The 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 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.  

Sorry but I have to disagree.  You appear to have a misunderstanding.

 

Digital signals are turned into analogue signals. They don't exist in isolation. This cannot be a totally arbitrary process because you would not be able to have different pieces of equipment correctly operating with each other.

 

The perfect example of this was when CD came out its output levels were significantly higher which caused issues for many amps overloading them and caused volume controls to operate over a very limited range.

 

Yes dBFS indicates the highest digital signal level but this is directly relateable to the highest analogue signal level that exits a dac or cd player.  So it's absolutely relevant. You must have standards, or at least conventions on what this level will be.

 

Knowing this level ( 2 volts with current convention) is essential information that allows you to correctly design the following pre / power amplifier.  To have the correct gain level

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34 minutes ago, lucretius said:

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.

And with digital signals, there is neither headroom nor clipping as we understand them in the analog world.  Analog devices are generally tolerant of minor level excursions over whatever 0 dB reference is used.  Distortion, compression etc are progressive in analog circuitry and increase proportionately (but not linearly) as the level goes further above 0.  But when a digital signal level hits 0 dBFS, the maximum # of 1s has been reached for the word size being used and further increases just discard the additional information.  There is no soft clipping in digital audio, and there are no innocuous or potentially pleasurable effects of digital clipping.

 

DAWs using 32 bit floating point resolution can keep adding more, so they’re theoretically immune to overload during the recording process.  But when you convert back to fixed point to edit, mix, master and/or otherwise turn the capture into a playable and archivable file, every 1 in the code to the right of the truncation point (16 or 24 bits with few exceptions) is simply lost as is the signal information encoded in it.

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2 hours ago, March Audio said:

Digital signals are turned into analogue signals. They don't exist in isolation.

Of course they do, just as this post exists in isolation.  Digital signals are pages of words that can be read, duplicated, edited, printed on paper, archived, etc.  They aren’t “turned into” analog signals - they’re used as a template for the creation of de novo analog signals whose vacillations reproduce encoded patterns of change to a level of accuracy determined by sampling rate, bit depth etc.

 

The process of DA conversion is conceptually like the process of amplification.  The signal on a record, CD or other source is not turned into the signal that powers your speakers.  It’s used as a template to modulate a totally different current flow to change as the source signal changes.  But both current flows exist as independent entities - and the characteristics of the source determine characteristics of the output.  Both sources exist as independent entities, the digital signal as a series of 0s and 1s and the analog signal as undulating grooves pressed into vinyl, etched by lasers into pits on the undersurface of a plastic platter, etc.  You can’t hear a digital file and you can’t hear record or CD - but you can hear their content as above.

 

A DAC uses a digital word stream to modulate a DC source into a replica of the waveform it encodes.  Analog audio devices use tubes, transistors etc to modulate DC current into a replica of the input waveform.  Digital amplifiers push the conversion process to the power stage.  But it’s all the same concept - an input signal is used to shape a separate and distinct output signal.  A digital file is as real as a vinyl disc - and it’s as useful in isolation.

 

You can’t listen directly to a digital signal.  You have to transform its content into an energy form you can sense and understand.  This is what your audio system does.  You can’t read this post with your eyes closed.  You have to transform its content into an energy form you can sense and understand.  This is what your visual system does.  The blind have to do it another way, eg using Braille to convert content into tactile input.

 

This ain’t rocket science.

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13 hours ago, stereo coffee said:

And no dynamics, and sounding squashed will be the direct result. Why ASR is pushing this agenda is the question to ask, as it has nothing to do with enjoyment of music. 

 

I will start a new thread, as its a fascinating subject, there i will provide answer to what is needed.  

 

 

Why playback can have no dynamics and sound squashed is a combination of the way the particular recording was mastered, and using a flawed playback chain - it has nothing to do with signal levels, in the sense that you're talking of them.

Frank

 

http://artofaudioconjuring.blogspot.com/

 

 

Over and out.

.

 

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7 hours ago, bluesman said:

Of course they do, just as this post exists in isolation.  Digital signals are pages of words that can be read, duplicated, edited, printed on paper, archived, etc.  They aren’t “turned into” analog signals - they’re used as a template for the creation of de novo analog signals whose vacillations reproduce encoded patterns of change to a level of accuracy determined by sampling rate, bit depth etc.

 

The process of DA conversion is conceptually like the process of amplification.  The signal on a record, CD or other source is not turned into the signal that powers your speakers.  It’s used as a template to modulate a totally different current flow to change as the source signal changes.  But both current flows exist as independent entities - and the characteristics of the source determine characteristics of the output.  Both sources exist as independent entities, the digital signal as a series of 0s and 1s and the analog signal as undulating grooves pressed into vinyl, etched by lasers into pits on the undersurface of a plastic platter, etc.  You can’t hear a digital file and you can’t hear record or CD - but you can hear their content as above.

 

A DAC uses a digital word stream to modulate a DC source into a replica of the waveform it encodes.  Analog audio devices use tubes, transistors etc to modulate DC current into a replica of the input waveform.  Digital amplifiers push the conversion process to the power stage.  But it’s all the same concept - an input signal is used to shape a separate and distinct output signal.  A digital file is as real as a vinyl disc - and it’s as useful in isolation.

 

You can’t listen directly to a digital signal.  You have to transform its content into an energy form you can sense and understand.  This is what your audio system does.  You can’t read this post with your eyes closed.  You have to transform its content into an energy form you can sense and understand.  This is what your visual system does.  The blind have to do it another way, eg using Braille to convert content into tactile input.

 

This ain’t rocket science.

Sorry but Im really lost as to what point you are trying to make here.  No-one has said you can listen to digital data.  The whole conversation is about how these digital word values get turned into analogue voltages - specifically what voltage level.

 

With respect you have a massive misunderstanding if you think that DACs (digital to analogue converters) dont turn digital word values into analogue voltages.

 

Digital word values are turned into voltages.  Each word value has a discrete voltage level.  As has been shown with modern dacs and CD players if playing a 16bit word the highest value 0dBFS (111111111111111 in a16 bit signed system) gets turned into 2.8 V with the current convention for RCA domestic connections. 

 

So of course these digital word values are directly relatable to an analogue voltage.

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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.

mQa is dead!

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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.

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1 hour ago, March Audio said:

Bluesman has stated that you can't relate a digital signal level to an analogue voltage level.

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):

 

image.png.bb9df6c8b218ba6bb42b068ffa799553.png

 

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.

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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):

 

image.png.bb9df6c8b218ba6bb42b068ffa799553.png

 

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.

 

 

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