HQ Player

[Miska]

1 hour ago, dean70 said:

The output rate is 176400. PCM source works fine with this filter ( both 44.1k and 48k families).

 

That's why it doesn't work.. It doesn't support 1:1 conversion ratios. I have improved the rate check rule for next release. Since DSD64 1/16th rate is 176.4k, that the DSD-to-PCM conversion output already has 176.4k rate..

 

1 hour ago, dean70 said:

On an unrelated issue with pcm out, there is a snippet that plays from a previous track after stopping and starting a new playlist (like the contents of the last buffer) that only occurs with pcm out. This has been occurring over the last few versions.

 

Which filter are you using when this happens? I'll check the buffer flush code... Usually this is left-over in the filter...

 

[dean70]

9 minutes ago, Miska said:

 

That's why it doesn't work.. It doesn't support 1:1 conversion ratios. I have improved the rate check rule for next release. Since DSD64 1/16th rate is 176.4k, that the DSD-to-PCM conversion output already has 176.4k rate..

 

 

Which filter are you using when this happens? I'll check the buffer flush code... Usually this is left-over in the filter...

 

 

So sinc-M needs to be an even multiple of the dad source, eg min 352800 pcm out for dsd64?

 

re buffer: it does not appear to be filter dependant. I have had this occur across a number of the poly-sinc* and closed-form filters.

 

[Miska]

1 hour ago, fgribas said:

Well, with PCM 44.1, the sound is absolutely amazing. My new favorite! It was closed-form-M before (that now is my 3rd favorite). But another odd thing using this filter is that the audio comes after 14 seconds of playing the track.

 

It's not like other heavy filters that "freezes" the system for some seconds before playing the track (only once after I launch HQP). With sinc-M, the playback starts immediately (on Roon and HQP), but the audio comes after 14 seconds, and audio stops when the track time ends, so I can't listen to the last 14 seconds of the track. This combined with the impossibility of playing anything other than 44.1 source (like MQA 48, 88 and 96 from Tidal on Roon) makes this filter impossible to use for me. I'm sad because it sounds amazing.

 

When using PCM output, sinc-M is mostly useful on 16x output rates (705.6/768 kHz), then the delay is like 1.5 seconds, or at least 8x rates (352.8/384 kHz) at about 3 seconds delay.

 

I need to rethink the flush code if people want to use it lower rates too...

[Miska]

11 minutes ago, dean70 said:

So sinc-M needs to be an even multiple of the dad source, eg min 352800 pcm out for dsd64?

 

Either 88.2k or 352.8k for DSD64 source. DSD-to-PCM conversion outputs PCM at 1/16th of the DSD rate and then the output can be further processed to the other PCM output rates. Possible options depend on combination of DAC and filter capabilities, so intersection of the two sets.

 

[fgribas]

1 hour ago, Miska said:

 

I need to rethink the flush code if people want to use it lower rates too...

 

Great! On my main DAC it will be OK (768 kHz max), but the Dragonfly is limited do 96 kHz.

 

1 hour ago, Miska said:

 

That's why it doesn't work.. It doesn't support 1:1 conversion ratios.

 

 

That's it! MQA decoded by Roon to 88.2 or 96 is already the maximum rates supported  by Dragonfly.  Is it possible to make 1:1 possible?

[Account Closed]

I also use ASRC as some others have reported here and love it but I do have a unique situation. I have a Mark Levinson 390s CD/Processor that is limited to 44.1/24 or 48/24 input on SPDIF. I use HQ Player with ASRC to bring everything coming into HQ Player converted (up from 44.1 and down from the rest) to 48/24 out to the ML 390s which then further up-samples to 384/24 using its own internal filter (no doubt FIR). I have also found that for me it works best without any dither. No matter which one I tried, I could here some distortion. There probably needs to be somewhere to stash the the dither noise and at 48/24 I doubt that there is room.

 

This works well while I take my time replacing this aging but still amazing DAC.

[arglebargle]

So I guess sinc-M is optimised for CUDA similar to closed-form? I get about 1/3 the cpu load with CUDA enabled. Nice. 

[arglebargle]

double 

[Miska]

5 hours ago, bobflood said:

I also use ASRC as some others have reported here and love it but I do have a unique situation. I have a Mark Levinson 390s CD/Processor that is limited to 44.1/24 or 48/24 input on SPDIF. I use HQ Player with ASRC to bring everything coming into HQ Player converted (up from 44.1 and down from the rest) to 48/24 out to the ML 390s which then further up-samples to 384/24 using its own internal filter (no doubt FIR). I have also found that for me it works best without any dither. No matter which one I tried, I could here some distortion. There probably needs to be somewhere to stash the the dither noise and at 48/24 I doubt that there is room.

 

This works well while I take my time replacing this aging but still amazing DAC.

 

Using filters from the poly-sinc family that can convert 44.1 to 48 do quite a bit better job on that conversion than ASRC. So I'd recommend those instead.

 

[Whitigir]

@miska , can you please tell me more about closed form M ? What taps length is it, and what will it affect if my LKS004 also has a filter on it own after this.  Thanks

[Miska]

1 hour ago, Whitigir said:

@miska , can you please tell me more about closed form M ? What taps length is it, and what will it affect if my LKS004 also has a filter on it own after this.  Thanks

 

It is documented in the manual... But M is million taps and 16M is 16 million. closed-form-M is mostly useful for 705.6/768k PCM output rates and maybe for 352.8/384k. While closed-form-16M is mostly useful for DSD256 and higher.

 

Having some filter following at least 2x or 4x rate conversion doesn't really matter other than usually limiting possible input rates. Rest depends on how good the DAC's built-in filter is, but much less than at lower input rates.

 

[jimdukey]

Can anyone give a run-down on the sound of the various oversampling Filters?

I pick up bits and pieces here, like the Closed Form-M being for DSD 256 and higher.

I don't have all the choices listed in the manual either, some FIR choices missing.

I'm on IMac, High Sierra, 3.2-3.6 GH Processor, HQ 3.24 Desktop.

I listen to Jazz, Classical and Rock/Blues.

I haven't hit on one that is IT for me, differences seem small.

[Theobetley]

i just tried the latest HQP with the M filters and

I must say I never heard anything better in my system. So congrats! But I still have trouble with initial connectivity. I have a i5, 12 gb memory - microRendu. - Lio dac 2.0  setup but can’t seem to get connectivity.after a shutdown unless I restart the microRendu. Is there an easier way to do this? 

[jimdukey]

What rate do you listen to with the M Filters?

DSD 256 or 512?

Or PCM?

 

Mistakes? 

[Theobetley]

I listen to pcm at 352 and dsd at 128. Didnt try the higher sampling yet.

 

[jimdukey]

Well, that's one of my questions-

The Closed Form M is supposed to be for DSD 256-512.

I also listen at DSD 128.

[Le Concombre Masqué]

On 9/12/2018 at 2:07 PM, AnotherSpin said:

I am trying new filters. Most impressed with ASRC (sinc previously) - PCM, mac mini to iDSD micro (bit-perfect).

how come you output PCM ? With SDM music comes and goes like particles in quantum vacuum. Really, that's the image that came to my mind 

I'm really happy with 3.24,

 

best

[numlog]

You are right, CIC is pretty good. It sounds colder than IIR so it was hard to get into at first but it seems like its fixing some issues with DSD and tightnening everything up, similar to DSD5v2 and the colder but more accurate ASDM7. PCM was typically thinner and colder than DSD but more accurate, now they are pretty similar except DSD doesnt have the same subtle negative traits I get with PCM (thin, shouty).

 

I think native DSD might sound better (or different at least) but using DoP allows you to control buffer size. With native DSD it is fixed 10ms for DSD64 and 20ms for DSD128, not sure if that is the same for everyone but this is what the ASIO control panel indicates with my DAC.

Using a low bitrate of 44.1x 64 allows a very low buffer size of 5ms without stuttering, the sound quality of 5ms DSD64 is very nice and beats DSD128 at its minimum of 15ms, DSD128 is still more resolving but the 5ms buffer has smoothness and cohereny that is not worth sacraficing for that resolution.

 

Also using HQ Player to set buffer size instead of ASIO control panel didnt work for DoP DSD, a 10ms buffer in HQ player with DoP DSD64 was indicating something weird like 312ms in the Asio control panel.

[AnotherSpin]

1 hour ago, Le Concombre Masqué said:

how come you output PCM ? With SDM music comes and goes like particles in quantum vacuum. Really, that's the image that came to my mind 

I'm really happy with 3.24,

 

best

 

Well, In my present system which is much simpler in comparison to what I had before PCM gives more volume, weight, presence, while SDM gives if more refined and fluid sound, the same time too... anemic...)

[Yviena]

@Miska

what do you think about this  https://archimago.blogspot.com/2018/01/audiophile-myth-260-detestable-digital.html

 

especially this part

Quote

The key here is to remember that within a properly bandwidth limited signal where all the frequencies are below Nyquist, a linear phase FIR filter actually does not create ringing regardless of the impulse response appearance. As I have said in the previous weeks, any decent recording will follow this rule. And if it does, then the ideal filter to use is clearly a linear phase, sharp filter that can reconstruct all the frequencies in the audio data with essentially ideal temporal resolution.

And that folks is the "myth" we need to say goodbye to in 2018! Linear phase, steep "brick wall" type antialiasing/anti-imaging digital filters performed with high precision, and with no intersample overloading, do not "ring" with good recordings that only contain "legal" frequencies below Nyquist.

 

 

[Le Concombre Masqué]

1 hour ago, AnotherSpin said:

 

Well, In my present system which is much simpler in comparison to what I had before PCM gives more volume, weight, presence, while SDM gives if more refined and fluid sound, the same time too... anemic...)

actually I meant DSD source played via SDM. I was pretty much agnostic until recently but since I reworked my convolution filters DSD sourced files have the edge in my system ; even more so since I switched to CIC. 

[AnotherSpin]

10 hours ago, Le Concombre Masqué said:

actually I meant DSD source played via SDM. I was pretty much agnostic until recently but since I reworked my convolution filters DSD sourced files have the edge in my system ; even more so since I switched to CIC. 

 

I know you are very persistent in getting the best sound possible, but, I am not getting close to convolution at all...)) In fact, my approach now is less is more. I was playing with wtfplayer for a while and liked its sound very much, only because there is no easy way to use it in my system I didn't opt for it as a main player.

[Le Concombre Masqué]

8 minutes ago, AnotherSpin said:

 

I know you are very persistent in getting the best sound possible, but, I am not getting close to convolution at all...)) In fact, my approach now is less is more. I was playing with wtfplayer for a while and liked its sound very much, only because there is no easy way to use it in my system I didn't opt for it as a main player.

being in the right mindset helps perceiving satisfying SQ and that is what matters in the end

[AnotherSpin]

51 minutes ago, Le Concombre Masqué said:

being in the right mindset helps perceiving satisfying SQ and that is what matters in the end

 

Exactly!!!! All filters located in mind.

[pkane2001]

15 hours ago, AnotherSpin said:

 

Well, In my present system which is much simpler in comparison to what I had before PCM gives more volume, weight, presence, while SDM gives if more refined and fluid sound, the same time too... anemic...)

 

I’m sure you know that there is a real volume difference between PCM and DSD and that the default 6dB compensation in HQPlayer is just an approximation to get the two closer? In many cases the difference will be something other than exactly 0dB or 6dB. To properly compare the two, you’ll need to properly match the levels.