Misleading Measurements

[pkane2001]

6 minutes ago, lucretius said:

 

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

 

No. 0dBFS applies to all waveforms, it cannot be exceeded.

[SoundAndMotion]

It might help to not spend too much effort understandIng AES17-1998 given the Forward of AES17-2015 states:

Quote

This document substantially revises and updates AES17-1998

and the Forward of AES17-2020 states:

Quote

This document clarifies the definition of levels, the units FS and dBFS.

I don't have the new document (nearly twice as long as the 1998 standard), but perhaps all is made clear!?

[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

 

[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.

 

 

 

 

[Miska]

On 5/10/2021 at 4:19 AM, March Audio said:

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

 

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.

 

[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 ;)