Setting up a HQ Player (w/ convolution) 2.2 system

[Miska]

24 minutes ago, blue2 said:

I have thought about this too. If you use HQP to upsample and do the convolution then you'll need a 3 or 4 channel DAC. I don't think there's a technical reason that HQP couldn't output to your stereo DAC and also the line out on your computer (for the subs) but it's not supported. Maybe @Miska can clarify this point?

 

They are not clocked from the same source. And most OS don't support launching multiple unrelated devices in sync.

 

There are multichannel DACs on the market, so that's not an issue.

 

27 minutes ago, blue2 said:

Also note that if you're a fan of DSD you lose that by going with convolution

 

Why would you?

 

[Miska]

1 minute ago, semente said:

So the best (only?) option is to get a multi-channel DAC?

 

Yes, if you'd like to run up to DSD256 then the options are pretty much exaSound and Merging...

 

[Miska]

16 hours ago, blue2 said:

Can HQP apply convolution filters to DSD natively? I guess it's a bit academic if HQP can do DSD in DSD out with convolution and there is no loss of SQ, but as you know especially on this site folk are quite pernickety, and I'm curious.

 

Let's say you play DSD64 upsampled to DSD256 with convolution. The path looks like this:

DSD64 -> convolution at 2.8 MHz sampling rate -> 4x SDM conversion to 11.3 MHz sampling rate -> DSD256

 

And for 44.1/24 PCM upsampled to 352.8/24 with convolution the path looks like this:

44.1/24 -> convolution at 44.1 kHz sampling rate -> 8x PCM conversion to 352.8 kHz sampling rate -> 352.8/24

 

HQPlayer has two separate DSP engines, one for PCM and one for SDM. Improvements in sound quality are similar both DSD and PCM.

 

16 hours ago, blue2 said:

(A) that's a pity. In a multichannel system this would be a showstopper, but for subs the frequency range is probably < 200Hz, even < 80Hz. Would synchronisation really be noticeable on subs?

 

I rather not enter such gray areas...

 

[Miska]

6 hours ago, bogi said:

I am curious, what's the path please, when you listen to 44.1/16 content, want to upsample to DSD128 and your IR WAVs are 192/24 ?

 

44.1/16 -> convolution at 44.1 kHz sampling rate -> 128x PCM-to-SDM converstion to 11.3 MHz sampling rate -> DSD128

 

[Miska]

27 minutes ago, bogi said:

Thanks. So when the the IR WAV sample rate does not match the source content rate it is always first resampled to the source sample rate to perform convolution at that sample rate.

 

Yes, although "resampled" is wrong term, it is scaled using algorithm designed for that particular purpose.

 

28 minutes ago, bogi said:

are the IR WAVs at first upsampled to DSD64 to perform convolution at 2.8 MHz?

 

Yes, although here too "upsampled" is wrong term. So they are scaled and converted to suitable form, and optionally with the "HF Expansion" that extends the frequency response flat beyond Nyquist frequency of the filter sampling rate.

 

 

Overall, the filters always go through bunch of adaptation steps to make them suitable for actual processing.

 

[Miska]

32 minutes ago, semente said:

I'm using a UD-501 at DSD128. Should I export the convolution WAV filter at 44.1kHz or at 384kHz?

 

Most of my content is 44.1 but I also have 48kHz, 88.2 and 96kHz.

 

If the tool can create a proper filter at higher rate, it is better to create it for a rate same or higher than highest rate content you have. So if possible at least 192k or 352.8k. 352.8k should cover all PCM content there is available (since I haven't seen any 384k) and also bandwidth wise cover all possible content frequencies.

 

[Miska]

15 minutes ago, jamesg11 said:

/& if highest rate content is dsf256?

 

The same, 352.8k filter has frequency response spanning from 0 to 176.4 kHz which should be enough for any musical content.