Meridian Unplugged - Powered by MrTechGuy.co.uk Forums Discussion Meridian - Systems WAV music into G68

What does the transport straight to G68 sound like?

Is the transport upampling?

The upsampler has no idea what it's upsampling which is part of the reason for the low pass filter. Upsampling is basically educated mathematical guess work where a few conditions are known. Nyquist theorem suggests that the highest frequencies present in the original signal are half the sample rate. When you calculate an intermediate sample you will almost inevitably introduce high frequency spuriae into the data. Therefore you can be pretty sure that everything in your new 88.2/96KHz data stream above 22KHz is noise, so you filter it out with a low pass filter.
The catch-22 is that with a limited amount of DSP power (the operations occur in real time) you can either use an algorithm that produces less noise and a less severe noise filter or a noisier algorithm and a severe noise filter. Or you can hope and pray that the losses from your tradeoff between the two are compensated by the gains from running your Delta-Sigma DAC at a higher rate.
Apologies if I've over-simplified upsampling, I'm aware that the reality is even worse than I've made out.

I understand basically how upsampling works which is why I asked why you said:


By "rolls off the mid-highs", "emphasize the bass" and "don't have a flat frequency response" do you mean that it filters out frequencies above 22.05kHz? If not what do you mean by those phrases? Filtering out content above 22.05kHz occurs whether you upsample or not. I am also curious how you were able to determine that the upsampling algorithm in the G68 falls in the group of "most upsamplers" which don't have a flat frequency response. Which upsamplers fall within the group that do have a flat frequency response?

Mark, I appreciate a lot of the technical background you've give.

But do you really think Meridian's upsampling is that bad?

Granted I barely hear any improvement when I toggle it on, but it certainly doesn't sound any worse. However when I toggle it I'm supposedly trading off G68 vs DSP speakers from upsampling. But I never heard much difference when when running analog speakers either.

If the tradeoffs were as bad as you say I would think Meridian would make an option in the DSP speakers and they do not.

I understand the whole Nyquist thing but I think there is more to it. There are tons of applications that take lower resolution data and with processing can greatly improve things. Perhaps they are trade offs but some tradeoffs are well worth it.

I think many of us have trusted Meridian's Track Record in making things sound good. And will continue to do so until they trip up which they have not done yet. I'm exciteed to try the apodizing filters in the G68 but I'm also realistic not to expect too much.

I do know that hooking transporter directly to DSP6000's sounds detailed but a bit harsh and fatiging. But when going through the G68 it's warmer. At the cost of a slight amount of detail. Perhaps this is a tradeoff like you mention but one well worth it. This is probably more the resolution enhancement (16->24 with dithering) which is also a "mathematical guessing" (i.e. tradeoffs).

I really wish we could have more "518" knobs to play with though. I don't want to deal with a 518 due to it's 48khz processing limit.

Hi There
it might be that my answer is a bit OT, but:

Have some DSP speakers and Sooloos running in some shops ( yes Im sales rep for the German M distributor)`
Sound is nothing short then stunnig. The use of the M upsampling and the apadizing filter in the DSP7200 does a job as good, that no more question remains open.

Best
Armin

It may be OT, but I am very interested. Can you please elaborate on how exactly this is hooked up. Also have you done an A - B comparison with an 808.2?

Thank you and aloha,

Quite simple actually: first the waveform is transformed from the time domain to the frequency domain using a fourier transform ... at least that is how Audio Remaster works ... however, you can transform calculations in the frequency domain back to a filter in the time domain, and that is how most resamplers work (an older version of Audio Remaster worked that way).

Only bad ones do ...

I'm not suggesting that Meridian upsampling is bad, far from it, but it is performed in real time with limited DSP horsepower. Can it be bettered? Probably. Within the processing confines of the G68/861v4? Maybe...
I recall the comment from Meridian about apodizing that they tried many different filters and the one everyone preferred used the most DSP power.
BTW the method evdberg suggests is A) computationally intensive and B) basically what the SRC code does. A much easier way is to make predictions based on the last n real samples, which works much better with integer multiples because there is no weighting factor involved. You still need a low pass filter though to remove noise generated by the upsampling.

A) That depends on the size of your buffer, you could even take the n samples you mentioned ... and it will sound most definitely better than the method you propose.
B) As said, most serious resamplers do and I am sure also Meridian does.

If you are suggesting that a FFT->IFT sounds better then I'm totally in agreement with you. I apologise if I'm being pedantic but the computational requirements for FFT->IFT are always going to be higher than even a weighted average of multiple samples. And when you say serious resamplers do you mean offline PC based or realtime hardware DSP? I'm useless at writing DSP code in part because I can't figure out ways of fitting the operations required into a limited amount of processing power.

Are you saying that M are unlikely to be using FFT? I would be surprised as I thought that FFT operations were built native into many DSP processors so M would probably make use of them if FFT allowed M to make a better sounding upsampler.

Without access to their code, you can only infer certain things from the behaviour. It would appear that Meridian are using a low pass filter. Why this is used we cannot know. In the case of the apodizing filter the use of a minimum phase low pass causes a dramatic drop in pre-echo.