"The XO is a 24.576mhz extremely low jitter crystek part which is perfect for doing 96/24(256fs) or 192/24(128fs)."
I am trying to figure out how the ASRC in Metronome works, but somewhat confused.
Could someone explain to me where does the number 256fs and 128fs above come from, and what does it mean?
Thanks!
I am trying to figure out how the ASRC in Metronome works, but somewhat confused.
Could someone explain to me where does the number 256fs and 128fs above come from, and what does it mean?
Thanks!
hbarki said:"The XO is a 24.576mhz extremely low jitter crystek part which is perfect for doing 96/24(256fs) or 192/24(128fs)."
I am trying to figure out how the ASRC in Metronome works, but somewhat confused.
Could someone explain to me where does the number 256fs and 128fs above come from, and what does it mean?
Thanks!
"fs" stands for "sampling frequency". Specifically since we are using the output in master mode to take advantage of the low jitter clock, we will be using that clock as the reference.
256fs = 256 x sampling frequency which at 96khz = 24.576mhz.
Likewise 192khz x 128 = 24.576mhz.
So to get 96khz output you set the metronome for 256fs on the output in master mode.
To get 192khz output you set the metronome for 128fs on the output in master mode.
Hope that helped.
Cheers!
Russ
Thanks Spartacus for your information, I should have pointed to your post in my original email.
So let me get this correct, is the Metronome an unsampler or re-clocker? Or perhaps both?
Also thanks to you analog_sa for your help, but i'm still a little confused.
If I want to use the Metronome with my other Opus stuff should I put it between the USB input via I2s to Metronome to OPUS Dac via 12s? If this is correct? How is the Metronome powered? If it is re-clocker is there a clock? Do I need anything else?
Sorry if I'm being a bit dense, digital electronics are new to me and if I don't ask I won't know.
So let me get this correct, is the Metronome an unsampler or re-clocker? Or perhaps both?
Also thanks to you analog_sa for your help, but i'm still a little confused.
If I want to use the Metronome with my other Opus stuff should I put it between the USB input via I2s to Metronome to OPUS Dac via 12s? If this is correct? How is the Metronome powered? If it is re-clocker is there a clock? Do I need anything else?
Sorry if I'm being a bit dense, digital electronics are new to me and if I don't ask I won't know.
rjbaldwin said:
Both...
Datasheet says:
The input data is passed
through interpolation filters which up-sample the data, which
is then passed on to the re-sampler. The rate estimator
compares the input and output sampling frequencies by
comparing LRCKI, LRCKO, and a reference clock. The
results include an offset for the FIFO pointer and the coefficients
needed for re-sampling function.
I ordered a "metronome" myself, just to see if it's an actual improvement.
I really don't know what to expect, half of the functions of the SRC4192 are already in the WM8804 receiver, the other half of the functions are already performed by the WM8740.
If it's a really audible improvement, we'd simply know that the WM8740 is not the best choice as a DAC.
OneyedK said:
Well not exactly correct...
The way things are done are not the same at all, but you will still get very good results even without a metronome.
What I mean is you can't put 44.1/16 into WM8804 and get out a 192/24 sample. With the metronome you can.
At first the metronome was an experiment purely for my own pleasure/curiosity, but it seems to have grown legs. I don't expect everyone will think its as fun as I do.
I also happen to think it makes things sound a bit better, at least thats my subjective opinion.One Key benefit is that instead of the clock being "generated" by a PLL (ala WM8804/PCM2707) you get a nice clean low jitter master clock. I think that is probably one of the key benefits. Also the whole metronome has its own VREG so its supply is very clean. You can even power it completely separately from the rest of your stack if you so choose.
Cheers!
Russ
Another key point. I wanted to provide the metronome for cases where people only had (easily accessable) the bit clock, word clock, and data PCM signals. No master clock. In this situation the metronome provides the master clock. All is well.
An example is some of the Opus owners who have CDPro transports etc.
It also seems to clean up the USB module I2S signal very very well.
Cheers!
Russ
An example is some of the Opus owners who have CDPro transports etc.
It also seems to clean up the USB module I2S signal very very well.
Cheers!
Russ
It seems to me that the Wolfson is already an upsampling DAC. So you don't need the ASRC for sample rate conversion. To me, logically, it seems that what the ASRC is really giving you is a cleaned up clock. In that case, when using modern upsampling dacs does just the addition of something like this with some buffering instead of an ASRC make sense?
http://www.cirrus.com/en/products/pro/techs/T16.html
http://www.cirrus.com/en/products/pro/techs/T16.html
OneyedK said:
This statement does not make much sense. Altough reading the datasheets is certainly better that asking completely clueless questions.
ASRCs have a primary function to reduce jitter. An ideal receiver, with a perfectly working PLL would extract a jitter-free clock but real life receivers are not perfect. Especially at lower jitter frequencies (under a few kHz). Some high-end dacs use a secondary PLL sometimes capable of reducing jitter down to a few hertz. Unfortunaltely, despite great results, effective PLL circuits are not exactly diy-friendly. Especially if the diyers would rather see a dentist than read a datasheet.
The only simple, readily available solution (without any transport mods) is the ASRC. Both in commercial and diy practice these offer a compromise. On one hand the jitter is quite likely reduced, on the other - all sorts of additional issues arrise. In pracice ASRCs just don't work as cleanly as the datasheets suggest. Deep waters for me and the main reason i was curious of what they do.
With some types of music the ASRC can sound very impressive. But if you really value timbral accuracy and natural ambience then the ASRC can easily be found to add something artificial.
As i find home music reproduction to be artificial anyway, the added "special effects" from the ASRC don't bother me too much. It's a compromise which works for me. I can imagine a NOS fan really hating it.
A further bonus is of course allowing the dac to work at its highest sampling rate.
Ok, Point taken. Sample rate conversion vs. oversampling. Got it.
In any case, would a buffer stage with cleaned up clock using something like the CS2000 type chip not provide an improvement for those averse to sample rate conversion? Possibly a good improvement when using SPDIF with NOS (or any other) converters when the basic properties of the converter are desired and ASRC is not wanted?
If I read the data sheet correctly you could use the CS2000 like this:
SPDIF generated clock used to clock data into a buffer.
SPDIF generated clock fed to the CS2000 to create a new clock that is synchronous with the SPDIF clock and nicely cleaned up with ultra low jitter.
The new CS2000 clock is then used to clock the data out of the buffer and into the converter. It seems to me that one of these chips and a small CPLD would do all you need.
Does this make sense? Would it be worth it?
In any case, would a buffer stage with cleaned up clock using something like the CS2000 type chip not provide an improvement for those averse to sample rate conversion? Possibly a good improvement when using SPDIF with NOS (or any other) converters when the basic properties of the converter are desired and ASRC is not wanted?
If I read the data sheet correctly you could use the CS2000 like this:
SPDIF generated clock used to clock data into a buffer.
SPDIF generated clock fed to the CS2000 to create a new clock that is synchronous with the SPDIF clock and nicely cleaned up with ultra low jitter.
The new CS2000 clock is then used to clock the data out of the buffer and into the converter. It seems to me that one of these chips and a small CPLD would do all you need.
Does this make sense? Would it be worth it?
Russ White said:
I know, I'm a happy user for two months now...Russ White said:
What I mean is you can't put 44.1/16 into WM8804 and get out a 192/24 sample. With the metronome you can.
Yes, one can, but is it really an improvement?
After all, we're creating samples that never existed in the original signal. But then again, a DAC creates an analog signal form "a few samples" only, so why not help it ;-)
I'm sure the Metronome does that all quit well. I was just thinking out loud. I'm more of a thinker than a doer. No time to be a do-er. Having brain farts is easy. Which explains why I have one of your Metronomes on order! BTW, if you ever want to try anything like this, I can do the CPLD coding. Coding is easy but I'm all thumbs when it comes to making boards.
OneyedK said:
Well yes!!!
Especially for the use cases I outlined. For the USB module re-clocking via the metronome has immediate benefits. And those with CDPro units are also very glad. 
Also, while I admit the sonic improvement is likely at best completely subjective, I think both the COD(PCM1794) module and the Opus(WM8740) sound better at 192/24 than they do otherwise. What is better? Well to me details just seem more evident. It is especially notable with headphones. I hate writing in audio reviewer speak, so I won't really go there.
Cheers!
Russ
Just had a quick look at the CS2000. For DAC use it's PLL could be useful. However the Wolfson receiver already has a fairly high performance PLL on board. A possibility for someone who wants lower jitter without sample rate conversion might be to daisy chain the receiver boards, so that the first board would output S/PDIF, and the second I2S.
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