[paul2qute]

I bought a Linn LP12 way back in the 80's...i had a british friend buy one when the dollar was very strong and bring it to me on a trip to the states....I was shocked to see used versions of the same turntable selling for more than i paid for it in the 80's...do not write off the CD...i personally have about 5000 of them

I couldn't agree more mate

 

[Triode User]

I couldn't agree more mate

But you haven't got a working CD player so how much to ship them to me?

 

[paul2qute]

But you haven't got a working CD player so how much to ship them to me?

CD player went to nintronics for parts and I'm sure you won't want my CD collection, John Denver Neil diamond etc lol

 

[ray-dude]

If I won the lottery I would pay for the divorce papers then buy a pair of Grande utopias

By that metric, my B&W 802d3's were the most expensive speaker purchase ever

 

[jlbrach]

 
By that metric, my B&W 802d3's were the most expensive speaker purchase ever

i own the 802 d2's...love them!

 

[paul2qute]

By that metric, my B&W 802d3's were the most expensive speaker purchase ever

Nice speakers them mate,the best speakers I bought were focal utopia speakers first generation, they were 18 and a half grand and my pair were in a hi-fi magazine, I bought them from the bloke who bought them from new, they were just 2 and a half years old when I got them and I paid 8 grand for them, neighbours hated them but I loved the speakers

 

[Sonic77]

I wonder if the Blu MKII will be at SoCal Can Jam?

 

[onsionsi]

  I wonder if the Blu MKII will be at SoCal Can Jam?

Yes as per Jude video in his Youtube Channel it will be there (Chord min 34:00)
 
https://www.youtube.com/watch?v=qf4ilaX37Kw
 
I'm looking to know opinions who will attend and got a chance to listen to DAVE with Blu MKII

 

[EVOLVIST]

Okay, I'm sure this will probably get an elementary answer, but since I haven't asked a stupid question this month, here goes...

768kHz out of the Blu2 + DAVE combo, right?

Now, let's say your amplifier has a frequency response that maxes out at 200kHz. Most max at about 100kHz. Is that then truncating the frequencies, squeezing 768kHz into a 100kHz pipeline? Moreover, once you get to the headphones or speakers, you're generally maxed at 20kHz, not to mention that the ears cannot hear above 20kHz, anyway.

So, we have a TON of out of band frequencies. What good is it to the audio and can it even be reproduced through your cans or speakers?

 

[icebear]

I think you are confusing digital input sample frequency (up to 768 khz) with analog output (20hz to 20khz +/- 0.1dB)

 

[EVOLVIST]

I think you are confusing digital input sample frequency (up to 768 khz) with analog output (20hz to 20khz +/- 0.1dB)

I might be, but in this chain: Blu2 -> DAVE - > Amp -> these are all inputting to one another until it reaches the headphones or speakers. Technically the digital domain ends at the amplifier input. It is this point that I question what happens to all of these out-of-band frequencies.

Now, if I'm not mistaken (and I very well may be) that the increase in frequencies are a byproduct of upsampling 16x, and not necessarily the draw. The tasty parts are that the bit-depth has now been reconstructed to what it was before the analog master hit the ADC.

So, again, unless I missed your point, isn't there some sort of truncation going on with the frequencies in the path? Bit-depth should be retained, but the frequencies?

 

[x RELIC x]

You can't confuse an analogue audio frequency wavelength with a digital sampling rate.

https://en.m.wikipedia.org/wiki/Audio

https://en.m.wikipedia.org/wiki/Sampling_(signal_processing)

In the analogue realm the frequency represents the vibrations per second to create an individual frequency.

When referring to a PCM digital signal (sampled data) the amount of kHz represents the number of samples per second. So, a 44.1kHz digital sampling rate has a sampling at 44100 times per second. A 768kHz sampling rate has a sampling of 768000 per second. Each of those samples represents a sample of the original data in time, and the bit depth represents the voltage amplitude (for 16bit that's 65536 possible voltage amplitudes - possible integer values - and a SNR of 96.33dB) for each sample. Both the sampling rate (time) and bit depth (amplitude) are required for digital audio reproduction.

So when looking at analogue audio frequency think of tone (20Hz-20kHz). When looking at digital data sampling rates think of how many samples of the original data per second, not tone.

I'm no audio engineer but this is the best way I can describe the differences between the two.

 

[Rob Watts]

Okay, I'm sure this will probably get an elementary answer, but since I haven't asked a stupid question this month, here goes...

768kHz out of the Blu2 + DAVE combo, right?

Now, let's say your amplifier has a frequency response that maxes out at 200kHz. Most max at about 100kHz. Is that then truncating the frequencies, squeezing 768kHz into a 100kHz pipeline? Moreover, once you get to the headphones or speakers, you're generally maxed at 20kHz, not to mention that the ears cannot hear above 20kHz, anyway.

So, we have a TON of out of band frequencies. What good is it to the audio and can it even be reproduced through your cans or speakers?

 
Nobody can hear a frequency above 20 kHz - I can't hear above 16 kHz - so does that mean I need to worry about over 20 kHz?
 
Absolutely. You can't hear 20 kHz but you can perceive timing errors. The interaural delay resolution is about 4uS (that's 250 kHz) - so although you can't hear 250 kHz, you can hear a timing error of 4 uS.
 
Imagine also a 100 nS delay circuit. Now nobody can hear 10 MHz (100nS time) - but if you switched in an out the delay at 1 kHz (1mS) rate, you would hear it, and you would be able to measure a host of 1 kHz inter-modulation products that the delay being switched in and out would create. So although we can't hear 10 MHz, we can easily hear timing errors. And it is the same with digital audio - here we have a sampled, or a discontinuous signal, but analogue is a continuous signal - and a DAC has to convert a discontinuous signal back to a continuous signal again without any timing errors. In practice a DAC will introduce timing errors in that the interpolated (or re-created signal) will not have exactly the same time as the original analogue signal in the ADC. In short, instead of a delay being switched in and out we have a delay (too late) or an advance (too early) that is dependent upon the sample rate and the original signal. What we want to do is to have no delay or advance at all - and theory tells us exactly how to do this - and that gets us back to infinite tap length sinc function filters. And theory also states that there are absolutely no short cuts to do this - if we want to perfectly reconstruct the signal, we must use an infinite tap length sinc function filter. Anything less than this will create timing errors.
 
My view - and this is based on listening tests - is that the brain is incredibly sensitive to the smallest possible timing error - and we need to worry about nS and not 4 uS (the interaural delay). And I will keep pushing and improving the DAC's ability to accurately recover the original timing information until I can no longer hear any improvement - and I think I have more to do on this problem.
 
Rob

 

[Triode User]

I might be, but in this chain: Blu2 -> DAVE - > Amp -> these are all inputting to one another until it reaches the headphones or speakers. Technically the digital domain ends at the amplifier input. It is this point that I question what happens to all of these out-of-band frequencies.

Now, if I'm not mistaken (and I very well may be) that the increase in frequencies are a byproduct of upsampling 16x, and not necessarily the draw. The tasty parts are that the bit-depth has now been reconstructed to what it was before the analog master hit the ADC.

So, again, unless I missed your point, isn't there some sort of truncation going on with the frequencies in the path? Bit-depth should be retained, but the frequencies?

Rob has answered you but I don't think he has answered the question you were asking.

As others have said you are still confusing audio frequency with digital sampling rate of the analogue waveform.

For example, you could have a 7 kHz audio tone which is being sampled at 768 kHz. The sampling frequency represents the accuracy of the reconstructed audio waveform and has nothing whatsoever to do with the audio frequency.

 

[EVOLVIST]

I guess this is why when I try to listen for a tone above 16kHz I cannot hear it, but my body can feel it. Because it is so high pitched, if I have an 18kHz tone playing for 20 seconds right up to my ear, I begin to feel agitated, repulsed, that I have to immediately take the tone away. I don't hear a thing, but yeah, something is going on.