I bet this was answered already somwhere on this forum but I just had to ask. How do DACs like AD1955 or the AKM and Wolfson parts handle 384Khz or more when these are specified to 192 Khz?
Anagram offers an example for 384Khz
http://www.anagramtech.com/products/hardware-for-hifi/s2m/
From what I see the upsampling is done in the dsp, however that doesn't explain how pcm is getting played through AD1955
http://www.anagramtech.com/products/hardware-for-hifi/s2m/
From what I see the upsampling is done in the dsp, however that doesn't explain how pcm is getting played through AD1955
Hi,
the DAC-chips contain a digitalfilter, that does over-/upsampling too. So the overall upsampling factor is basically split between the Input receiver / Samplerateconverter (4fs) and the digital filter (2fs). When the DAC-chip receives a signal on 192kHz (4fs) it upsamples to 384kHz (8fs) and this signal is fed its internal DACs.
Anagram now bypasses the DAC-chip´s internal digital filter and omits with the Input receiver and puts these stages all together into one module. The signal of the Anagram module is fed imediately to the DACs at 8fs (384kHz). Advantages over the conventional way could only stem from the way the signal is recovered from the incoming data stream (improved jitter behaviour) and different digitial filter algorithms. The clock rate is nothing special though as most modern DAC-chips allow to bypass their digital filter sections and drive their DACs directly with 384kHz. But it is a nice feature for marketing.
jauu
Calvin
the DAC-chips contain a digitalfilter, that does over-/upsampling too. So the overall upsampling factor is basically split between the Input receiver / Samplerateconverter (4fs) and the digital filter (2fs). When the DAC-chip receives a signal on 192kHz (4fs) it upsamples to 384kHz (8fs) and this signal is fed its internal DACs.
Anagram now bypasses the DAC-chip´s internal digital filter and omits with the Input receiver and puts these stages all together into one module. The signal of the Anagram module is fed imediately to the DACs at 8fs (384kHz). Advantages over the conventional way could only stem from the way the signal is recovered from the incoming data stream (improved jitter behaviour) and different digitial filter algorithms. The clock rate is nothing special though as most modern DAC-chips allow to bypass their digital filter sections and drive their DACs directly with 384kHz. But it is a nice feature for marketing.
jauu
Calvin
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Thank you Calvin, much appreciated!
I understand now why I've seen Cambridge players with anagram dacs going up to 768 Khz.
The main ideea is that this can be done, but with some DSP/FPGA programming know-how.
I'll wait for the day when USB will be able to asynchronously stream 32bit/384Khz 🙂
I understand now why I've seen Cambridge players with anagram dacs going up to 768 Khz.
The main ideea is that this can be done, but with some DSP/FPGA programming know-how.
I'll wait for the day when USB will be able to asynchronously stream 32bit/384Khz 🙂
on the block diagram of ad1955, you see something marked "s&h", that is how they generate the rest of the samples that feed the modulator,a "hold" function.
32bit 384khz, sounds like journalist talk. And how old is that dCS ring dac ? 10-13?
That must be fed 5bit 3mhz... hehe...
32bit 384khz, sounds like journalist talk. And how old is that dCS ring dac ? 10-13?
That must be fed 5bit 3mhz... hehe...
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"journalist talk" ? 🙂 Yeah, probably, but it's still an interesting exercise... and as long as some studios are recording at this sample rate, it would be an interesting exercise in the sense that you would avoid re-sampling the audio data..
Of course, that if you are able to send flawlessly 384Khz to the dac in the first place..
Of course, that if you are able to send flawlessly 384Khz to the dac in the first place..
"journalist talk" , exactly. 96khz is all what we have , and studio gear on its knees, converters and dsp resources alike.
http://www.esstech.com/products/digitalaudio/Sabre PF 080221.pdf
32-bit, up to 192KHz. Although many of the bits seem to go to waste and noise if the dynamic range is 135 dB.
Ideal DR:
16-bits -> 96 dB
24-bits -> 144.5 dB
32-bits -> 192.7 dB
...although human hearing probably only approaches 100dB of dynamic range. Seems like 24-bits is plenty already (although more bits may help inside the DSP for math purposes).
32-bit, up to 192KHz. Although many of the bits seem to go to waste and noise if the dynamic range is 135 dB.
Ideal DR:
16-bits -> 96 dB
24-bits -> 144.5 dB
32-bits -> 192.7 dB
...although human hearing probably only approaches 100dB of dynamic range. Seems like 24-bits is plenty already (although more bits may help inside the DSP for math purposes).
ok this ESS thing starts to look like comic book edutainment at its best so I wont comment anymore.
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