Hi,
On 2022-01-10 15:41, Attila Kinali wrote:
On Mon, 10 Jan 2022 12:35:17 +0100
Attila Kinali <att...@kinali.ch> wrote:
That said, are yo sure you need such stringend phase noise requirements?
It's audio. Nobody is going to hear whether the noise is -60dBc or -80dBc @ 1Hz,
much less -120dBc.
To give here a bit more background: psychoacoustic masking, which is the
relevant
metric here, mans that we cannot discern sounds that are close to eachother with
one of them being louder than the other. Depending on who you listen to, it's
usually a sound masking another sound at a distance of 100Hz up to 20dB to 40dB
lower. Even if we account for someone with golden ears and use 60dB, that would
translate to a noise spec of -60dBc @ 100Hz offset. That's a spec that almost
all
XO do fulfill. A good VCXO (40-100MHz) is somewhere around 90-100dBc @ 100Hz.
Any OCXO will fulfill that spec too, even the tiny DIL-14 ones (most are at
-110-140dBc @100Hz @10MHz).
A classical paper on jitter for audio is written by late Julian Dunn:
http://www.nanophon.com/audio/jitter92.pdf
He wrote a range of good papers which is worth digging up.
AES-3 (also known as AES/EBU) is the professional version of the digital
audio interface that in the consumer version is called S/P-DIF. Much of
the reasoning for AES-3 applies directly onto S/P-DIF.
And this doesn't take into account that we are arguing about audio frequency
specs
at HF frequencies. I.e. if we use the 10MHz clock and use it to derive a
sampling
clock for an ADC to sample a 20kHz signal, the noise performance improves by
another
~25dB... at least (if the design is done right, it can be up to 50dB)
What is more important than close in noise, though, is broadband noise
performance
and spurs. For someone with good ears, it's not unheard of to be able to discern
far away noise and spurs down a -100dB-120dB. Especially the spurs can be quite
hard to control, depending on what clock synthesis system is used.
Another important spec, especially for recording, is accuracy of frequency. An
offset of just 1ppm becomes 3.6ms if you record for an hour. That's something
most people can hear already. But whether this actually elevant or not depends
on how the recording is done. The usual way is to have a central master clock
that feeds all clocked devices, such that all of them have the same notion of
time/frequency. In that case, quite high frequency deviations can be tolerated,
way beyond what a simple XO would deliver.
In professional audio we tolerate +/- 10 ppm for larger productions and
for all my experience, the exact frequency have not been as much trouble
as that of different frequencies and the degradation that comes from
slips, resync or Sample Rate Conversion (SRC). It has been interesting
to teach the TV and radio techs about basic synchronization and how
doing it well once save money, time and quality. There is a few war
stories to be told.
Cheers,
Magnus
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