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A novel way to massively improve the SQ of computer audio streaming


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Most important: please realize this thread is about bleeding edge experimentation and discovery. No one has The Answer™. If you are not into tweaking, just know that you can have a musically satisfying system without doing any of the nutty things we do here.

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7 minutes ago, hols said:

If I can start again from the very beginning I would probably go for PCM upsampling rather than DSD because the demand on computer is much less and there is a high potential to get equal satisfaction on musical feeling as in DSD. YMMV.

Hols, sorry to be so dense here, but are you saying that, with the endgame being PCM 1411.2k, any source files, even DSD files?  IOW, you are saying DSD64 to PCM 1411.2k is better than DSD64 to DSD256 (using EC modulators), not to mention the cpu savings?

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17 hours ago, sig8 said:

I am considering T+ and Gaia, but based on @hols reporting above if higher speeds in PCM and perhaps in DSD sound better, then Gaia will limit you to DSD512, and PCM to 768 kHz. Don't know if T+ at higher speed is better than T+ and Gaia combo at lower speed. Thoughts? Experience?

I don't have personal experience on Gaia. All along my focus is on USB rather than I2S. I have tried Audiobyte, and Singxer SU-1 with Holo Spring but I don't prefer using them but I must say that I have not modified the SU-1 or tried good enough cables. I have not tried I2S input for Terminator Plus. But Terminator plus performance with USB is already very satisfying be it in PCM 705.6 or DSD 256, not to say if you go all the way to PCM1536. I would suggest you go for Terminator Plus first. You can always add a Gaia afterwards. 

I may be able to get hold of a Gaia sometime later. I will report back if I have further personal observation.

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Hols, thanks. I have always preferred PCM files staying PCM (and DSD staying DSD) with my Holo dac, for example. So your conclusion that even higher resolution 1411.2 sounds great with red book fits my expectations. I’d be very surprised, however, to hear that DSD to 1411.2 PCM ( an inherently lossy conversion) upsampling withstands and is superior to DSD remodulation. Looking forward to more experiments. 😀

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18 hours ago, Exocer said:

 

Thanks @ray-dude.

 

@guiltyboxswapper - I am grateful for your nudge to research HQPlayer. It makes for an interesting combination with the Yggy and I look forward to playing with the filters more. The pair sound very good together! There were some hurdles with HQPlayer Embedded initially since some of the default settings did not align with the documentation (on GentooPlayer) but once those issues were resolved I was able to locate the configuration page.

 

Other recent changes:

 

1. Switch from HDPlex 400W DC-ATX to 800W DC-ATX.

  • I think this made for a rather worthwhile improvement even with my current SMPS powering the server.

2. External clock for ER. Thanks to @MartinT's experiment I gave this clock a try and I do feel it improved the soundstage, separation and calmness without a loss of dynamics. This is no Mutec Ref 10 but for a fairly cheap price this experiment has proven to be rewarding. I have since switched my LPS 1.2 to the clock (since I prefer stock SMPS over the LPS 1.2 for the ER).

  • clock1.jpg.756c35925ce362b0995b429b64e6cc7b.jpg
  • clock2.jpg.0a11d5bebf63397df828804957acd3ec.jpg

 

Happy listening all,

-Rob

 

HQPlayer Configs1.JPG

 

@rob

@ray-dude

@hols

@austinpop

 

have you experimented with the hqp buffer time?

i have found that increasing it always improves the SQ.

 

the maximum permitted through the hqp web interface is 250ms (range checking on input**)

with an opticalRendu, i found better SQ with 500ms, which could be set by editing the settings.xml file

also, with a NUC (audio-linux for example, haven't tried w/ euphony) which has a lot more RAM available,

you can increase the buffer time even much more significantly (just note that the buffer is in units of time, not bytes,

so that a larger value is possible with red book than with hi-rez sources)

 

i'd be interested in any conclusions you have with both the extreme and diy servers w.r.t. buffer time settings.

 

**interesting story--------i had a vp who was getting a demo of a new modeling program for laying out the location and design

parameters for mobile cell sites, incorporating terrain, propagation, etc. 

he sat down and immediately entered a value for a cell tower that was 2000 miles tall, the program crashed, and he walked away saying their program design had flaws.  in my case, i set the buffer value to use the maximum amount of ram available, calculated using red book cd parameters.  when i went to play a 24/96 track, i crashed the NUC.  however, the maximum ram buffer did improve SQ.

 

 

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When I was running HQP on my NUC, my point of diminishing returns on buffer was between 2000-3000 msec (set in the settings.xml).  Any larger than that and I wasn't hearing a benefit.

 

With Extreme, things are all different.  Buffer size has a definite impact on dynamics.  Smaller the buffer, the better the dynamics but perhaps thinner?  Larger buffer slower but perhaps fuller.  The combination of DAC bits and buffer are basically tone controls now (on Extreme), and the final tweaking I do after adjusting some other part of the chain.

ATT Fiber -> EdgeRouter X SFP -> Taiko Audio Extreme -> Vinnie Rossi L2i-SE w/ Level 2 DAC -> Voxativ 9.87 speakers w/ 4D drivers

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I think saving in RAM and Buffer size are two different things. Saving in RAM is about separating the music files from the network noise. The music is played from the fast working memory.

Buffer is about fetching the largest possible data packets on a slow network to avoid dropouts. In my experience, however, a large buffer has a very negative effect on the latencies. I have set both the network card adapter and USB to the smallest buffer.

See Denafrips USB ASIO as an example. The output latency is very low 1.50 ms. That makes an audible positive difference in my system.

 

spacer.png

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36 minutes ago, StreamFidelity said:

I think saving in RAM and Buffer size are two different things. Saving in RAM is about separating the music files from the network noise. The music is played from the fast working memory.
 

 

It isn’t exactly the same thing but I run Euphony and Ram root.  All the my music is buffered and played from memory.  I have tried removing the network cable after the music is buffered and I hear no difference in sound quality.  This approach in Euphony does a great job in reducing network noise.

 

Speakers: Vandersteen Model 7s, 4 M&K ST-150Ts, 1 VCC-5; Amplification: 2 Vandersteen M7-HPAs, CI Audio D200 MKII, Ayre V-6xe; Preamp: Doshi Audio Line Stage v3.0; Phono Pre: Doshi Audio Phono Pre; Analog: Wave Kinetics NVS with Durand Telos composite arm; SME 3012R arm, Clearaudio Goldfinger Statement v2; Reel to Reel:  Technics RS-1500; Doshi Tape Pre-Amp; Studer A810, Studer A812, Tascam BR-20; Multi-channel: Bryston SP-3; Digital: Custom PC (Sean Jacobs DC4/Euphony/Stylus)> Lampizator Pacific

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36 minutes ago, StreamFidelity said:

I think saving in RAM and Buffer size are two different things. Saving in RAM is about separating the music files from the network noise. The music is played from the fast working memory.

Buffer is about fetching the largest possible data packets on a slow network to avoid dropouts. In my experience, however, a large buffer has a very negative effect on the latencies. I have set both the network card adapter and USB to the smallest buffer.

See Denafrips USB ASIO as an example. The output latency is very low 1.50 ms. That makes an audible positive difference in my system.

 

spacer.png

when running a one box solution and running in ramroot, the amount of ram limits the buffer size

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24 minutes ago, dminches said:

This approach in Euphony does a great job in reducing network noise.

 

I like to believe that.

 

Still, you could take a look at your network map. There are a multitude of settings in the network adapter that influence the buffer in the data traffic. That's what I mean. There a small buffer ensures low latencies and, in my and other experiences, this improves the audio playback considerably. Look for example to Pink Faun: "During the process of developing the streamer 2.16, we found out that latency is one of the key elements to achieve the really smooth and natural sound; the lower the latency, the better."

 

My contribution should only make it clear that not all buffers are the same. I also think it makes sense to cache it in RAM, and the buffer should be very large there. The buffer in the network traffic, however, should be as small as possible. 😉

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9 minutes ago, StreamFidelity said:

 

I like to believe that.

 

Still, you could take a look at your network map. There are a multitude of settings in the network adapter that influence the buffer in the data traffic. That's what I mean. There a small buffer ensures low latencies and, in my and other experiences, this improves the audio playback considerably. Look for example to Pink Faun: "During the process of developing the streamer 2.16, we found out that latency is one of the key elements to achieve the really smooth and natural sound; the lower the latency, the better."

 

My contribution should only make it clear that not all buffers are the same. I also think it makes sense to cache it in RAM, and the buffer should be very large there. The buffer in the network traffic, however, should be as small as possible. 😉

 

Understood, but if I don’t hear any sound improvement with my server off the network then what improvements could I hear in changing my network?

 

I also don’t know how to change the network adapter setting with Euphony.  I have done that when I run Roon in Windows.

 

Speakers: Vandersteen Model 7s, 4 M&K ST-150Ts, 1 VCC-5; Amplification: 2 Vandersteen M7-HPAs, CI Audio D200 MKII, Ayre V-6xe; Preamp: Doshi Audio Line Stage v3.0; Phono Pre: Doshi Audio Phono Pre; Analog: Wave Kinetics NVS with Durand Telos composite arm; SME 3012R arm, Clearaudio Goldfinger Statement v2; Reel to Reel:  Technics RS-1500; Doshi Tape Pre-Amp; Studer A810, Studer A812, Tascam BR-20; Multi-channel: Bryston SP-3; Digital: Custom PC (Sean Jacobs DC4/Euphony/Stylus)> Lampizator Pacific

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21 hours ago, cat6man said:

 

@rob

@ray-dude

@hols

@austinpop

 

have you experimented with the hqp buffer time?

i have found that increasing it always improves the SQ.

 

the maximum permitted through the hqp web interface is 250ms (range checking on input**)

with an opticalRendu, i found better SQ with 500ms, which could be set by editing the settings.xml file

also, with a NUC (audio-linux for example, haven't tried w/ euphony) which has a lot more RAM available,

you can increase the buffer time even much more significantly (just note that the buffer is in units of time, not bytes,

so that a larger value is possible with red book than with hi-rez sources)

 

i'd be interested in any conclusions you have with both the extreme and diy servers w.r.t. buffer time settings.

 

**interesting story--------i had a vp who was getting a demo of a new modeling program for laying out the location and design

parameters for mobile cell sites, incorporating terrain, propagation, etc. 

he sat down and immediately entered a value for a cell tower that was 2000 miles tall, the program crashed, and he walked away saying their program design had flaws.  in my case, i set the buffer value to use the maximum amount of ram available, calculated using red book cd parameters.  when i went to play a 24/96 track, i crashed the NUC.  however, the maximum ram buffer did improve SQ.

 

 

I've spent several months messing with these settings on the Extreme and agree with Ray Dude that they are basically tone controls when using 10ms - 250ms,  until I tried "Default" on both settings (buffer time and bits).  To me, "Default" ended up being the best parts of all the other settings with none of the weaknesses.  

 

When using any setting between 10-250ms, I would prefer one over the other depending on the album I was listening to or because of some recent system upgrade or change.  "Default" has sounded good across all my music and through all the recent changes and upgrades.  It was a very obvious improvement, not something I had to focus on to hear.  

 

When ever I would increase the buffer, let's say from 20ms to 50ms, the top end sounded rolled off, there was a loss in dynamics, loss in details and speed, but sound was fuller and smoother.  With Default - I am hearing all the details, speed, fullness, dynamics, etc with none of the drawbacks.  

 

Wanted to see if anyone else is hearing the same thing with "default" that I am so I can finally put this dead horse to rest.  

 

I'm running Extreme to Dave to AFC-10 to Susvara.  

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3 hours ago, lmitche said:

Is it just me or do any of you also notice a SQ improvement with the latest version of Roon (build) 610?

 

It looks as though this version is distributing processing across more cores which could explain what I am hearing.

 

Larry

 

Emile noted the same over at WBF.  Maybe after all the bluster they're paying some attention to SQ? (hope so)  I've been holding off upgrading out of being in a happy music place right now (tweaking hiatus, at least for things not easily reversible)

 

Thank you for the project update Larry!  I learn a lot every time you post one of these, very much appreciated.

ATT Fiber -> EdgeRouter X SFP -> Taiko Audio Extreme -> Vinnie Rossi L2i-SE w/ Level 2 DAC -> Voxativ 9.87 speakers w/ 4D drivers

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1 hour ago, lmitche said:

An Adnaco fiber USB rig with the single mode Finisar SFPs described above beats the SQ of the Monoprice solution. 

Do you mind sharing which model Adnaco?  There's quite a few.  As mentioned the JCAT XE + Monoprice aren't a stable option.  

 

And thanks for the memory tip, gskillz 4266mhz b-die running at 3200mhz (and standard! spd profile otherwise) is a good step forward indeed.  

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