A toast to PGGB, a heady brew of math and magic

[davide256]

Trying to figure out if the Denafrips Pontus/Venus/Terminator with dual AES/EBU inputs are an alternative to the dual BNC Chord solution. Anyone use this?

Struggling to find a description/limitations.

[austinpop]

29 minutes ago, davide256 said:

Trying to figure out if the Denafrips Pontus/Venus/Terminator with dual AES/EBU inputs are an alternative to the dual BNC Chord solution. Anyone use this?

Struggling to find a description/limitations.

 

The reason so many of us Chord users are using a USB to dual-SPDIF bridge is to bypass the Amanero USB controller, in favor of a lower-latency input, dual-SPDIF. But you're lucky your DAC has an XMOS2 controller, that is known to not have the latency issues of Amanero, so I would consider going USB direct to the DAC. I understand the Gaia/Iris DDCs may have other benefits, but at that point you would be weighing between the benefits of:

1. using a DDC with native files (no PGGB)
2. going USB direct using 32FS PGGB files (for Denafrips DACs).

As for DACs with "dual" SPDIF or AES inputs, I'm not aware of many. There's Chord, with the dual BNC SPDIF (16FS), dCS with their dual AES (but only goes to 8FS), and after that I'm drawing a blank. Do the Denafrips (or any other?) DACs support this "dual" mode on their SPDIF or AES inputs? Anyone?

 

[Progisus]

1 hour ago, Fourlegs said:

 

Thanks for experimenting. Perhaps we Antipodes owners need to jointly put pressure on Antipodes (in a nice way) to put other players on their streamers that will support 705/768kHz files.

 

Innuos use Squeezelite on their streamers and they have found a way to get their 2.0 app with Squeezelite to be able to play 705/768kHz files. I will be asking Antipodes if they can do the same but please can others ask the same question! The squeaky wheel is the one they gets oiled!

I just setup lms and squeezlite player on Antipodes and it plays up to 768. Only issue I had is bits needs to be set to auto and not 24 in my case. Tidal and Qobuz also play through the lms app. I can also use ipeng for control. There is no need to run the minimserver or dlna server as the lms server handles the library linking. Sound quality is very good. I am still impressed with roon/hqplayer using the embedded in the Antipodes.

 

I have used almost every player even Pure Music and Amarra and I am a huge fan of roon. 

[sig8]

3 hours ago, davide256 said:

Trying to figure out if the Denafrips Pontus/Venus/Terminator with dual AES/EBU inputs are an alternative to the dual BNC Chord solution. Anyone use this?

Struggling to find a description/limitations.

I am not answering the question you asked as I have not experimented with that, but I have a Gaia and T+. I am going direct USB into T+ at 32fs and it sounds just amazing. Gaia may hit the auction block soon (less clutter). I can praise the SQ all day long, but it is just unbelievable. This is what one can call End Game, truly. I really don't crave for any further improvement in SQ. Time to finally just enjoy the music, which I/we love.

[Fourlegs]

4 hours ago, Progisus said:

I just setup lms and squeezlite player on Antipodes and it plays up to 768. Only issue I had is bits needs to be set to auto and not 24 in my case. Tidal and Qobuz also play through the lms app. I can also use ipeng for control. There is no need to run the minimserver or dlna server as the lms server handles the library linking. Sound quality is very good. I am still impressed with roon/hqplayer using the embedded in the Antipodes.

 

I have used almost every player even Pure Music and Amarra and I am a huge fan of roon. 

Can you tell me which versions of LMS and Squeezelite you are using? The K50 has 7.9.3 LMS and  1.8.7 Squeezelite player and refuses to play any 16fs files whatever their configuration. I have even tried Roon + squeezelite on the k50 and that refuses to play. 

[Progisus]

28 minutes ago, Fourlegs said:

Can you tell me which versions of LMS and Squeezelite you are using? The K50 has 7.9.3 LMS and  1.8.7 Squeezelite player and refuses to play any 16fs files whatever their configuration. I have even tried Roon + squeezelite on the k50 and that refuses to play. 

1.9.9.1386 is squeezelite, 8.2.0 is lms .

 

BTW… I got the tag issues fixed so the MPD controller apps get the albums straight. I used mp3tag and copied each cover to cover or folder.jpg and then removed all the cover art from the tags. There appears to be an issue with the writing of the cover art on some files. Now to figure out the multiple disc tags. 

[Zaphod Beeblebrox]

I have released v1.2.05, this improves memory use and stability on Macs by reducing memory pressure during critical portions of the algorithm where there can be sudden increase in memory demand. With this new version, on M1 Mac, I have found setting max taps to 256M and setting number of workers to 1 is most efficient and stable even for long tracks.

 

As I promised earlier in my response to Rob Watt's questions, I have added an option to optionally remove apodizing option and to guarantee the use of pure Whitaker-Shannon coefficients  for CD audio (affects 44.1kHz tracks only). The full change log can be found here.

 

 

 

[kennyb123]

6 hours ago, Fourlegs said:

Can you tell me which versions of LMS and Squeezelite you are using? The K50 has 7.9.3 LMS and  1.8.7 Squeezelite player and refuses to play any 16fs files whatever their configuration. I have even tried Roon + squeezelite on the k50 and that refuses to play. 

Same with my K30, so I was surprised to hear that it was possible on the CX.  I sure wish Antipodes’ current models weren’t stuck with older applications.  It’s like we’re second class citizens relative to those who own their former products.

[Zaphod Beeblebrox]

16 hours ago, Zaphod Beeblebrox said:

I have released v1.2.05, this improves memory use and stability on Macs by reducing memory pressure during critical portions of the algorithm where there can be sudden increase in memory demand. With this new version, on M1 Mac, I have found setting max taps to 256M and setting number of workers to 1 is most efficient and stable even for long tracks.

 

As I promised earlier in my response to Rob Watt's questions, I have added an option to optionally remove apodizing option and to guarantee the use of pure Whitaker-Shannon coefficients  for CD audio (affects 44.1kHz tracks only). The full change log can be found here.

 

 

 

 

I released a patch just now v1.2.06, there was a bug in v1.2.05 that caused some DSD remastering to fail.

 

 

[zettelsm]

On 6/10/2021 at 7:37 AM, austinpop said:

 

As for DACs with "dual" SPDIF or AES inputs, I'm not aware of many. There's Chord, with the dual BNC SPDIF (16FS), dCS with their dual AES (but only goes to 8FS), and after that I'm drawing a blank. Do the Denafrips (or any other?) DACs support this "dual" mode on their SPDIF or AES inputs? Anyone?

 

 

Hi Rajiv,

 

The dCS Vivaldi DAC offers ethernet, USB and dual AES/EBU inputs. However I am not at all sure that dCS implementation is the same as Chord. dCS dual AES/EBU top sample rates are 384/DSD128, same as their USB input. However utilizing AES/EBU also offers bi-directional control and indication functionality for the dCS "stack" of equipment (DAC, Upsampler) in addition to data transmission.

 

Steve Z

[ASRMichael]

22 hours ago, Zaphod Beeblebrox said:

I have released v1.2.05, this improves memory use and stability on Macs by reducing memory pressure during critical portions of the algorithm where there can be sudden increase in memory demand. With this new version, on M1 Mac, I have found setting max taps to 256M and setting number of workers to 1 is most efficient and stable even for long tracks.

 

As I promised earlier in my response to Rob Watt's questions, I have added an option to optionally remove apodizing option and to guarantee the use of pure Whitaker-Shannon coefficients  for CD audio (affects 44.1kHz tracks only). The full change log can be found here.

 

 

 

Hi 

 

Have you compared using with & without Apodizing? Which do you prefer? 

[Zaphod Beeblebrox]

11 minutes ago, ASRMichael said:

Hi 

 

Have you compared using with & without Apodizing? Which do you prefer? 

Glad you asked! For myself and the  beta testers who tried it earlier on, apodizing is the preferred option and it is also the reason why this checkbox never made it to the public release earlier. 

 

The only reason I enabled this option is for those who may want to try it themselves, and to alleviate any concerns that apodizing by removing aliased components in frequency domain causes timing inaccuracies.

 

Now everyone has a choice to try and come to their own conclusion. If someone likes the non-apodizing, that is good too as what matters at the end is you are liking it and the technical details don't matter at that point. 

 

If any of you try and like  apodizing over non-apodizing or vice-versa, do post your impressions here, it is good to have more data points.

 

This option affects CD audio only (44.1kHz)

[Egill23]

On 6/10/2021 at 4:37 PM, austinpop said:

 

The reason so many of us Chord users are using a USB to dual-SPDIF bridge is to bypass the Amanero USB controller, in favor of a lower-latency input, dual-SPDIF. But you're lucky your DAC has an XMOS2 controller, that is known to not have the latency issues of Amanero, so I would consider going USB direct to the DAC. I understand the Gaia/Iris DDCs may have other benefits, but at that point you would be weighing between the benefits of:

1. using a DDC with native files (no PGGB)
2. going USB direct using 32FS PGGB files (for Denafrips DACs).

As for DACs with "dual" SPDIF or AES inputs, I'm not aware of many. There's Chord, with the dual BNC SPDIF (16FS), dCS with their dual AES (but only goes to 8FS), and after that I'm drawing a blank. Do the Denafrips (or any other?) DACs support this "dual" mode on their SPDIF or AES inputs? Anyone?

 

The Denafrips Terminator has dual AES/EBU input. From the manual:

Dual AES/EBU Input

1. Press the Mute button once to enter configuration mode

2. Press the INPUT+ momentarily, AES 1, AES 2 LED will turn on/off

-AES1 On = Dual AES/EBU Input Enabled

-AES2 Off = Dual AES/EBU Input Disabled

3. Wait for 10s

4. DAC back in operational mode

[austinpop]

1 hour ago, Egill23 said:

The Denafrips Terminator has dual AES/EBU input. From the manual:

Dual AES/EBU Input

1. Press the Mute button once to enter configuration mode

2. Press the INPUT+ momentarily, AES 1, AES 2 LED will turn on/off

-AES1 On = Dual AES/EBU Input Enabled

-AES2 Off = Dual AES/EBU Input Disabled

3. Wait for 10s

4. DAC back in operational mode

 

Interesting. What are some examples of dual-AES sources for the Terminator? And what is the max rate the Terminator can accept in dual-AES mode?

 

I thought these dual data modes were proprietary, like the dCS example Steve Z mentioned above. I'm pretty sure Chord MScaler also only works with Chord DACs in dual-data mode.

[Zaphod Beeblebrox]

8 hours ago, Zaphod Beeblebrox said:

 

I released a patch just now v1.2.06, there was a bug in v1.2.05 that caused some DSD remastering to fail.

 

 

I released another  patch just now v1.2.07, further improvement to stability while remastering DSD.

[WilliamWykeham]

It seems that the most of the contributors on this thread are using Chord dacs.  I wanted to share some impressions from a Holo May owner.

 

Curiously, I’m getting much better results at 16FS, 20 bits than I am at 32FS.  The sound at 16FS is more full bodied.  I also have sense of feeling more relaxed listening at 16FS than at 32FS.  

 

I’m wondering if any other May owners share the same opinion, or if you’re more partial to 32FS.  

 

 

 

 

 

[Fourlegs]

1 hour ago, WilliamWykeham said:

It seems that the most of the contributors on this thread are using Chord dacs.  I wanted to share some impressions from a Holo May owner.

 

Curiously, I’m getting much better results at 16FS, 20 bits than I am at 32FS.  The sound at 16FS is more full bodied.  I also have sense of feeling more relaxed listening at 16FS than at 32FS.  

 

I’m wondering if any other May owners share the same opinion, or if you’re more partial to 32FS.  

 

 

 

 

 

I'm looking forward to seeing contributions regarding the Holo May with PGGB.

 

I wonder though, why are you using 20bit with 16FS? Is this a Holo May input requirement?

 

With the Dave I either use 32bit, 16FS on USB or now (and preferred) I use 24bit, 16FS for dual BNC input to the Dave using the SRC-SX usb to dual bnc convertor.

[happybob]

8 hours ago, Zaphod Beeblebrox said:

I released another  patch just now v1.2.07, further improvement to stability while remastering DSD.

I've been testing with v1.2.07 several "challenging" albums (i.e. 44K/16 classical albums with long tracks) with my M1 MacBook Pro (16GB RAM, 2TB SSD with 1TB free) and I can reliably do 256MTap (1 worker) conversions (16FS, 24 bit) - I'm using a Chord Dave Dac (with SJ DC4 LPS), currently through a Chord MScaler (since I also use the same setup for streaming audio and PGGB can't process streaming tracks, an opticalRendu feeds the MScaler which then feeds the Dave via 2 Blackcat Tron BNCs).

 

So even though the MScaler also upsamples everything to 7xxK, I find the PGGB tracks to sound quite a bit better than what the MScaler does and of course both sound a lot better than the original 44K/16 tracks.

 

Side question in case anyone has a thought: Given that I am still using the MScaler in the chain (eventually plan to not use the MScaler in this chain) - should I use PGGB Adaptive Noise shaping or not? Zaphod thought it's likely best to use it and also use 24 bits (given the MScaler is in the chain and that I'm not Dave direct via USB), but neither of us are sure whether the MScaler does its own Adaptive Noise shaping if it receives a track already at the 7xxK sample rate in which case the MScaler basically does nothing, or maybe it does do something? I of course can do testing to see which I like better - as I also will with the HF noise filter (currently I'm using the default Moderate and also Transparency Natural and Presentation Transparent). 

 

Another question I'm pondering: Does the MScaler essentially do what a SRC•DX usb to dual bnc convertor does in terms of benefitting the sound for Dave by not requiring Dave to use the USB input? Has anyone sonically compared an MScaler to an SRC•DX assuming the MScaler isn't doing anything (both are fed PGGB 16FS 24 bit tracks)?

 

I'm so impressed with the SQ of PGGB tracks, even at "only" 256MTaps! Before PGGB I was at a point in my listening where I really didn't enjoy much classical music because it just didn't sound right (even with the MScaler) but now with PGGB it does!

[Fourlegs]

19 minutes ago, happybob said:

I've been testing with v1.2.07 several "challenging" albums (i.e. 44K/16 classical albums with long tracks) with my M1 MacBook Pro (16GB RAM, 2TB SSD with 1TB free) and I can reliably do 256MTap (1 worker) conversions (16FS, 24 bit) - I'm using a Chord Dave Dac (with SJ DC4 LPS), currently through a Chord MScaler (since I also use the same setup for streaming audio and PGGB can't process streaming tracks, an opticalRendu feeds the MScaler which then feeds the Dave via 2 Blackcat Tron BNCs).

 

So even though the MScaler also upsamples everything to 7xxK, I find the PGGB tracks to sound quite a bit better than what the MScaler does and of course both sound a lot better than the original 44K/16 tracks.

 

Side question in case anyone has a thought: Given that I am still using the MScaler in the chain (eventually plan to not use the MScaler in this chain) - should I use PGGB Adaptive Noise shaping or not? Zaphod thought it's likely best to use it and also use 24 bits (given the MScaler is in the chain and that I'm not Dave direct via USB), but neither of us are sure whether the MScaler does its own Adaptive Noise shaping if it receives a track already at the 7xxK sample rate in which case the MScaler basically does nothing, or maybe it does do something? I of course can do testing to see which I like better - as I also will with the HF noise filter (currently I'm using the default Moderate and also Transparency Natural and Presentation Transparent). 

 

Another question I'm pondering: Does the MScaler essentially do what a SRC•DX usb to dual bnc convertor does in terms of benefitting the sound for Dave? Yes, the MScaler eliminates the need for Dave to use its USB input like the SRC•DX does, but has anyone compared an MScaler to an SRC•DX sonically assuming the MScaler isn't doing anything (i.e. its fed PGGB 16FS tracks)?

 

I'm so impressed with the SQ of PGGB tracks, even at "only" 256MTaps! Before PGGB I was at a point in my listening where I really didn't enjoy much classical music because it just didn't sound right (even with the MScaler) but now with PGGB it does!

I also retain the mscaler (with a dedicated DC4) in my system for streaming but i much prefer 32bit PGGB files to go direct to DC4 Dave by usb rather than to take them through the mscaler. I do not find ‘pass through’ on the mscaler to be transparent.
 

It is only when using the SRC-DX that i then prefer 24bit PGGB files going to the DC4 Dave via dual bnc.

 

 

[happybob]

36 minutes ago, Fourlegs said:

I also retain the mscaler (with a dedicated DC4) in my system for streaming but i much prefer 32bit PGGB files to go direct to DC4 Dave by usb rather than to take them through the mscaler. I do not find ‘pass through’ on the mscaler to be transparent.
 

It is only when using the SRC-DX that i then prefer 24bit PGGB files going to the DC4 Dave via dual bnc.

 

 

Thanks for the perspective! I'm wondering, do you then do 2 sets of PGGB upsamples, one with 16FS/32bits and another 16FS/24bits? Also wondering what BNC cables you use? I can have an entirely different feed to Dave directly for PGGB tracks, but since I've found the opticalRendu to make a big improvement in SQ (I'm not yet at the point of having a Taiko although I'd like to get there, am currently using Roon via a SonicTransporter i9 as source), that means I'd need yet another opticalRendu/LPS to feed Dave or an SRC-DX directly and then another 2 BNC cables. This makes for a lot more $ and also more space taken up (I've got limited space in my current setup and unfortunately not unlimited funds). 

 

Stated differently and trying to put things in perspective: is the improvement achieved with PGGB upsampling greater than the improvement achieved by not having an MScaler in the chain (and so going from source directly to Dave or via SRC-DX)? Or is this comparing apples to oranges?

[Fourlegs]

1 hour ago, happybob said:

Thanks for the perspective! I'm wondering, do you then do 2 sets of PGGB upsamples, one with 16FS/32bits and another 16FS/24bits? Also wondering what BNC cables you use? I can have an entirely different feed to Dave directly for PGGB tracks, but since I've found the opticalRendu to make a big improvement in SQ (I'm not yet at the point of having a Taiko although I'd like to get there, am currently using Roon via a SonicTransporter i9 as source), that means I'd need yet another opticalRendu/LPS to feed Dave or an SRC-DX directly and then another 2 BNC cables. This makes for a lot more $ and also more space taken up (I've got limited space in my current setup and unfortunately not unlimited funds). 

 

Stated differently and trying to put things in perspective: is the improvement achieved with PGGB upsampling greater than the improvement achieved by not having an MScaler in the chain (and so going from source directly to Dave or via SRC-DX)? Or is this comparing apples to oranges?

I used to do two sets of pggb files 32bit and 24bit when i was deciding how to feed the Dave but now all my pggb processing is in 24bit to go though the SRC-DX. 
 

I am the guy who makes WAVE Storm bnc cables so obviously i use those because i think they are best 🤣. (The bnc cables used with the SRC-DX do make a difference so it is worth trying a few different ones).

 

One big sound quality improvement for me was stopping using Roon. For the PGGB files I use MPD with my Antipodes or the Innuos 2.0 beta app with my Innuos. Both sound great. 
 

Answering your final question in a different way, i find the sound quality of the SRC-DX into a Dc4 Dave for normal redbook or streamed files to be sufficiently good that  I am not too bothered about using the MScaler for that. I decided long ago that i would rather have the DC4 than have the mscaler because the dc4 makes a bigger difference to my ears. So for much of the time i just stream direct (no mscaker) to dave through the src-dx then it is all set up anyway to play pggb files. 
 

That is my 2p worth or 2c worth depending on which side of the pond one comes from. 

[Progisus]

For Antipodes users who are experimenting with pggb files here is my equipment for processing and playback. I have tested many playback schemes and found the following to work best for me. Playback tested: squeezelite, mpd direct, mpd upnp/dnla, hqplayer with hqpd control, hqplayer direct, hqplayer with client, Audirvana dlna, roon direct and roon with hqplayer.

Processing: 2020 iMac, 10 core, 40gb ram with pggb (perpetual license), 16fs 24b and mp3tag

roon core: nuc7i3 with 2tb internal ssd for redbook library, 3tb usb drive for pggb

network: fibre from core to Antipodes

Antipodes EX: all software stripped except hqpe. hqpe to alsa/usb to src-dx to tt2

playback: streamed - roon/hqpe, pcm,24b,poly-sinc-guass-long,lns15,-3db
playback: pggb - roon/hqpe, pcm,24b,none,none,0db. I boot to stream settings and change on main screen of embedded app as necessary for pggb.

Of course imho, ymmv

[Zaphod Beeblebrox]

7 hours ago, Fourlegs said:

I'm looking forward to seeing contributions regarding the Holo May with PGGB.

 

I wonder though, why are you using 20bit with 16FS? Is this a Holo May input requirement?

 

With the Dave I either use 32bit, 16FS on USB or now (and preferred) I use 24bit, 16FS for dual BNC input to the Dave using the SRC-SX usb to dual bnc convertor.

The choice of 20 bits is based on linearity measurements that have been published. With R2R DACs, resistors need to be precisely matched and doing so to retain 24 bits of accuracy is close to impossible currently. Resistor variation would cause the DAC output to not be linear in the last few bits, this would in turn result in distortion. Zeroing out the last few bits would help but it will in turn result in increased quantization noise because fewer bits (20 instead of 24 bits) are used to represent the signal. This is where noise shaping comes in. Using a well designed noise shaper, even when only 20 bits are used,  quantization noise can be almost removed from the audible range by taking advantage of the 16FS or higher sampling rate. Zeroing out the last bits and noise shaping not only linearizes the DAC to 20 bits, but also effectively increases the dynamic range by reducing the noise floor further.

 

Even with DAVE, one of the main reasons why  going 24 bits via SRC-DX  is  still effective even though only 24bits are used instead of 32, is because noise shaping is used. Even Mscaler noise shapes to 24 bits and it is not in pass through if the input is 32 bits.

[Fourlegs]

@Zaphod Beeblebrox Thanks. 

[ray-dude]

2 hours ago, Zaphod Beeblebrox said:

The choice of 20 bits is based on linearity measurements that have been published. With R2R DACs, resistors need to be precisely matched and doing so to retain 24 bits of accuracy is close to impossible currently. Resistor variation would cause the DAC output to not be linear in the last few bits, this would in turn result in distortion. Zeroing out the last few bits would help but it will in turn result in increased quantization noise because fewer bits (20 instead of 24 bits) are used to represent the signal. This is where noise shaping comes in. Using a well designed noise shaper, even when only 20 bits are used,  quantization noise can be almost removed from the audible range by taking advantage of the 16FS or higher sampling rate. Zeroing out the last bits and noise shaping not only linearizes the DAC to 20 bits, but also effectively increases the dynamic range by reducing the noise floor further.

 

Even with DAVE, one of the main reasons why  going 24 bits via SRC-DX  is  still effective even though only 24bits are used instead of 32, is because noise shaping is used. Even Mscaler noise shapes to 24 bits and it is not in pass through if the input is 32 bits.

 

Fantastic post and summary.  The magic of PGGB is all these factors coming together, and control being externalized from your DAC so you can optimize digital processing in software to get the absolute most out of the actual digital to analog conversion stage of your DAC.  

 

Each DAC will have its own optimal recipe, and some will be easier to bypass their internal digital processing than others.  Hopefully over time we'll see some world class DACs that are designed from the get go to externalize all their digital processing (for example, I'd love to hear PGGB content directly feeding a Chord-like pulse element array digital to analog conversion stage to a cost no object discrete analog output stage)