Meridian Unplugged - Powered by MrTechGuy.co.uk Forums Discussion Meridian - Systems Could DSP 7200 DACs be upgraded to handle 192khz/24bit material?

"You have X time to do twice the samples. So 2x processing power is needed?"

Nope, you have twice as many samples and each sample occurs in 1/2 the amount of time.

Shawn

Still scratching head

+1

At 96kHz sampling rate you have filters that cover an audio band from roughly 0-48,000hz.

At 192kHz sampling rate you have filters that cover an audio band from roughly 0-96,000hz.

The workload on the filter has doubled.

At 96kHz you have X number of samples in Y amount of time.
At 192kHz you have X number of samples in Y/2 amount of time. Or 2X in Y time.

So twice the workload in 1/2 the time.

Shawn

Still .

For a given order of filter it seems to me that the workload is a linear function of the sample rate only. Are you saying that to get a similar amplitude/phase response in the frequency domain you need to double the order of the filter or have more than one filter of the same order if you double the sample rate?

OK... found what I think is the first post where I heard the 4x DSP requirement for doubling the sampling rate. It was by John Kotches after talking to Bob Stuart on AVS:

"Shawn,

In talking with JRS, he was saying at least 4x DSP horsepower. You have 2x the operations and 1/2 the window for the operations. Buffering might remain constant as you don't have to buffer as long, so maybe there's a break there and in the DACs There has to be a break somewhere in the process, otherwise it's just a vast undertaking.

It's going to be interesting to see what happens next year with HD optical media, I'm not expecting much to happen in 4Q2005. HD-DVD might launch but I think the real fireworks will be in early 2006.

Cheers,"

Good reply expanding on this from Andre Yew:

The computational power depends on the order of complexity of the algorithms used. For some things, there will be a linear increase in computational power, for other things, like convolution algorithms, it will be more like O(n^2) like JRS's example (he may be thinking of room EQ since Gerzon uses that exact analogy in his early room EQ papers). And some will require even more.

Precision requirements will go up by lots, too --- the number of bits in order to accurately compute something --- because the more operations you do, the more temporary accumulation space you need. For example, addition of 2 n-bit numbers requires n+1 bits to store the result. Multiplication requires 2*n. Since you don't want to truncate to the DAC's output resolution until all your computations are done in order to maintain highest accuracy and lowest noise, you need to be able to to hold the internal result accurately until truncation. If you have a 10-tap FIR filter (which is rather small by audio standards), each sample undergoes 10 additions and 10 multplications. For a naive algorithm, that requires a little more than 2*10*n, or 20*n bits. For 32-bit integers, that's over 640 bits per sample. No one does it this way, as there is always truncation at each stage, but some algorithms are highly sensitive to small precision changes, so you have to maintain some amount of extra precision. Meridian seems to do it with 48 bits, and I don't know what Lexicon does. I know audio research people, like the AAC and MP3 people, have found that sometimes many dozens of bits are necessary.

This is the secret cost of high sample rates that no one talks about. Some things become significantly more difficult to do correctly with high sample rates. Any company claiming indefinite future compatibility had better be prepared to show you how they'll significantly increase their computational power.

--Andre

Shawn

Gotcha. Thanks for that.

But it's worth remembering what's relevant here and what isn't: we're looking at levels (zero order), delays (first order), crossovers (who knows, but potentially high order convolution FIR to take advantage of being in the digital domain). We're not talking room eq or upsampling/apodising.

Also Meridian have chosen their precision so if lots of computation is required then there's going to have to be interim truncation - precision doesn't enter into this question.

So the question boils to whether the filters that need to be implemented get longer as the sample rate goes up, which is what I think you're saying here. And that depends on what sort of filters (convolution v recursive etc) M implements in the DSPs.

Anyway, I agree with others that there is and should be no real call for playing 192K material.

"But it's worth remembering what's relevant here and what isn't: we're looking at levels (zero order), delays (first order), crossovers (who knows, but potentially high order convolution FIR to take advantage of being in the digital domain). We're not talking room eq or upsampling/apodising. "


According the Meridian the DSP7200 has both EQ ('sound-shaping') as well as upsampling. Though the upsampling may only be for lower sampling rates.

Shawn

This is not correct. Let's say your samples are 16 bits and the filter coefficients are 32 bits. Then a multiplication results in 48 bits. Additions only add 1 bit, so 10 additions add 2log10=3.32 bits.
The total number of bits without rounding for the 10-tap FIR filter is 52.