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CPU Load and Sound Quality


STC

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My normal usage for playing stereo is usually around 5%. And going full convolution DSP it would hit 100%.  Except for some drop outs at 100% I do not hear any difference to the sound quality. 
 

The original sound was recorded in DAW and the file is very large. It is available on request. 
The sound in the video is from the phone and the changes you hear is due to my movement and not due to the load. 
 

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Firstly, that was a stereo file. The PC was running normal processes of all the ones associated with Windows10 including Firewall. The network is activated. So too the anti virus. And ACDSee is running in the background. For serious listening all the processes will be killed and if I remember correctly it will be 1% with just JRiver. It is probably 5% because the data is linked to DAW via virtual cable. That would have added the load. Thankfully the cheap system somehow not degrading further despite CPU reaches 99%. 
 

Sorry I didn’t know this is what XXhighender is all about. 
 

 

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1 hour ago, sandyk said:

 

S.T.

With previous versions of Windows I obtained a worthwhile improvement by running Windows in Safe Mode, which reduces the number of background processes running, as well as prevents non essential programs from starting.

However, Windows 10 is a P.I.T.A. to do this..

Fidelizer by Windows X does similar  with Windows 10 as well.

If I wanted to create a poorer sounding audio file for comparison purposes I also ran  a full Security scan at the same time.

Even unplugging or switching off the cable to the Broadband Router resulted in a small but worthwhile improvement with Audio.

Alex

 


I need to get a more powerful machine than the current i7-7700. Hopefully, one can handle twice the load so that I could utilize all the IRs. Maybe then I will probably concentrate on other tweaks. Frankly, I don’t see there is a need to limit a PC’s usage. These minor tweaks becomes immaterial for my setup. Not to say they won’t make a difference but those difference are not important when considering other errors. 

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42 minutes ago, sandyk said:

In which case, given the widespread use of programs such as Fidelizer, and others minimising Start up programs etc. , a facility that even Norton Security now provides,  there must be a lot of people worldwide with crappy computers.

 Neither should there be any need to play music from System Memory as many members have found for improved results, and neither would JRiver have been requested to provide this option by so many users if this wasn't a problem with REAL WORLD PCs/Servers, NOT just your theoretical world of Digital Audio.

Incidentally, STC has previously admitted that he hears many things differently to other people, which may, or may not, be a bad thing.


Killing unwanted processes started in early days of computers with limited CPU and RAM. This started with gamers.  Audiophiles thought the same must be true and maybe some entry level PC would have benefited by turning off unnecessary processes although I suspect it got more to do with the reason Mansr stated as audio takes very little CPU load to play music without DSP. 
 

I hear differently than most audiophiles but it doesn’t mean I am incapable of hearing differences. It is just I don’t hear things that audiophiles often make but couldn’t prove. 
 

In my example, I could capture the output digitally and the 24/96 files should show the difference. 

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6 minutes ago, sandyk said:

Then prove it by participating with what you hear different between the various examples that John provides..

 John is doing his research for the benefit of future generations with archival material.

 John's results can only be as good as the actual listening feedback from the various participants, including both yourself and STC.

 


Alex, you can ask John about my contribution. Including about one of the version you praised so highly. 

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45 minutes ago, sandyk said:

Sorry STC, but this IS true and not just Audiophile folklore .


I cannot comment on this because I do not know how the system actually works. Technically, digital data streams serially so a SPDIF output would have a sample delay between the left and right. That would be so small to be easily visible looking at the waveforms. However, this can make a difference in the soundstage. Depending how the DAC was designed whether the delay was compensated or not is not known. 
 

Is it possible that when you stream directly the small delay between the channels either increased or reduced? If it so then it points to a flaw in the modification. 

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17 minutes ago, mansr said:

That is incorrect. Although the left and right channel samples are interleaved, the receiver buffers the data and does the A/D conversion simultaneously.

 

The early Sony CD players did have a half-sample delay between the channels, but that's ancient history. Also, the 12 μs difference is inaudible. Moving your head 4 mm makes a bigger difference.


Thanks for confirming. 12μs can be audible. IME, there is some inconsistencies when playing with delays for crosstalk. The real time cancellation and playing a preprocessed file directly still causes some difference in the soundstage which we are unable to pinpoint where it originates from. I am now thinking this is happening in the DAW which supposed to compensates all delays. 
 

BTW, how did you get half a sample delay?  

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39 minutes ago, mansr said:

Maybe, under some carefully controlled conditions. If one channel has a constant 12 μs delay, it is equivalent to the listener being 4 mm closer to one speaker. That's not a problem unless you are this guy:

head-vise.gif.18c79d6db4b46b2e75ab2538ee92baf1.gif

 

Those early CD players had a single DAC serving both channels.


Humans use ITD from as little as 10μs to 600 or 700μs. If the delay is constant ( and it could be true) then it is possible that the center image to be slightly off centered. That probably explains why every audiophile would have noticed that getting center image is always inconsistent. So what carefully  controlled condition you are talking about?

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9 minutes ago, fas42 said:

 

Say, Dragonfly Red uses

 

 


Even PCM2704 as in the picture is a stereo DAC. Maybe I didn’t phrase the question correctly. The data reaching the DAC chip would be carrying left and right. Is the a delay between this signal due to the fact that SPDIF carries the data in series. 
 

I ask this because theoretical value and actual value in my system differs by about 11us. 

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4 minutes ago, pkane2001 said:

 

I’m not sure why not. One sample at 44.1k can be easily measured if the ADC is running at say 192k, no? 


When I set the project to 192, the difference becomes negligible even with original 44.1. Unfortunately, the bandwidth is limited to 8 channels. So for now I am stuck with 96. 
 

Thanks to all for the answers. Apologies to the OP for going OT. 

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16 minutes ago, mansr said:

Did you not see the illustration I included? A constant 12 μs delay in one channel is equivalent to sitting about 4 mm off centre. You'd need to clamp your head in a vice to maintain your position that accurately.


So? There won’t be other delays?  All signal come with constant delay?  I do this on daily basis and it is audible if you are looking. 

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2 minutes ago, mansr said:

Every modern audio DAC chip has at least two independent channels. The incoming serial data is buffered and released to the conversion stages simultaneously.

 

The only audio devices I'm aware of with the channel offset are the first generation of Sony CD players. Their DAC chip had only a single D/A conversion stage that alternated between the channels. Philips used two TDA1540 chips in their early models, later switching to the TDA1541 which has two channels.


That is where I am still looking for an answer to find out why the delay doesn’t tally with the calculation. 

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46 minutes ago, sandyk said:

 Neither did you see the same  differences in some " colour" images that all the other participants did, and you also said this :

" I also don’t perceive height with binaural recordings with headphones. ;)

From this it is clear that you have different perceptions to many others, both audible and visual. 


Alex, I may have wasted my time giving feedback to the videos you sent. I stated the difference existed due to different frames and included different frames for you to compare and see where they are identical. Instead of looking at it objectively whether the difference existed because of you/we comparing different frames you used it to justify difference existed in the videos .

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3 minutes ago, sandyk said:

 Just as you wasted my time when reporting hearing differences just before the earpiece fell onto the floor ?

" I just finished listening with headphones and about to email. I think I could distinguish both files. However, I have to a proper blind test. This is interesting. I will download the files later before going to bedtime and give it another try. "

 

 This is all completely off topic though  and has nothing to do though with the present thread.

 Perhaps you should as the OP, clean up this thread ?


Maybe you are right. It is getting childish. 

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2 hours ago, sandyk said:

 

 

It's interesting to note that the more vocal Anti Subjective members never seem to show pictures of their systems. ¬¬


What is more interesting is that none of the high end audiophile manufacturers ever posted audio of their system’s playback. Some of them make speakers for studios and have at theIr disposal the best microphones which can reproduce the sound exactly as how they would capture a life performance.

 

Don’t you find it strange that some magazine are publishing the measurements but coy when it can produce accurate recordings of the playback? All high end system will sound more or less the same in the same room. 
 

One reviewer attempted to that and it was unbelievable good sound so much so I knew it wasn’t the sound that what was shown in the video. Someone spotted it and doubted about its authenticity .  The admission came after that that the audio was recorded separately even that I doubt of it’s accuracy. 
 

I have downloaded the video and looking for similar setup to record the same track playing in the system. 

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10 minutes ago, Blackmorec said:

By recognising the type of speakers and the way they are positioned far too close together, too close to the wall and with no toe-in GUILTY

By seeing components on a table with lots of lose, unsecured panels that will vibrate, or on a cabinet that’s essentially just a huge resonant wooden box GUILTY

By seeing components stacked on top of one another with no thought to radiated EMI or vibration control GUILTY

By seeing how power and interconnect cables are tangled with each other and coiled up behind components GUILTY

By noting the use of cheap power strips GUILTY 

By seeing speakers positioned right next to and level with large cabinets that will cause huge amounts of diffraction. GUILTY

By seeing systems installed in highly asymmetrical rooms with walls or windows to one side and complete open space on the other, completely destroying any chance of a sound stage and imaging. NOT GUILTY 

 


 

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5 minutes ago, Blackmorec said:

What are you actually listening to when you re-play a recording? 

 

Exactly!  Your system, in your room.

 

If you have a pair of say Rodgers LS3/5as with a Naim amp and you listen to the recording of a pair of Wilson Chronosonics driven by D’Agostino Relentless Amps, which system do you think the recording is going to sound like?  Its the same reason it doesn’t sound like the Berlin Philharmonic when you play a von Karajan recording 


Just because I need glasses to see things clearer doesn’t mean I couldn’t tell the difference between two pictures. 
 

I thought it is common knowledge that sound evaluation is to heard via headphones to eliminate room signature. You mean there are people out there listen to playback of a system through loudspeakers???  And I was getting frustrated with people listening to binaural recording with loudspeakers and then hearing them complaining they were too bright. 
 

 

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