My Summary about Computer Audio Music Server

[jabbr]

32 minutes ago, KingRex said:

I don't think he is changing the tempo, just detuning.  I know this is a gimmick with Blue's players.  

 

When I pulled up his server I was interested in his filters and how the server was eliminating ringing, as well as using the partial MQA unfold and completing the process.  

IDK, set off my bull$hit detector. “Detuning” means playing the track slightly slower so 440 Hz becomes 432 Hz ... thank you I’ll play the note the artist intended. Changing the tempo would be playing back the same notes more or less frequently. (Often done w MIDI)

 

Unless there’s a great explanation, lost any credibility. 

 

In any case there are all sorts of synth packages like Reason etc which can do all sorts of alterations to musical tracks.

 

Um right, software like HQPlayer and others have a “bitperfect” mode called ... don’t upsample and just use a NAS ... so you’ve got a $11K NAS? Nice case  

[FredericV]

6 hours ago, jabbr said:

 

Look, that study has nothing to do with anything we are doing here — namely recordings of music. Nor do you describe what your processing is actually doing.

 
We do here:
http://432evo.be/index.php/faq

[jparvio]

2 hours ago, KingRex said:

...Jparvio, is your friend using the Pure Audio Project Trio 15 with horn. I just upgraded from the Trio 10 with Voxativ to 15 with horn.  I get them in about a week.  Super excited. 

 

Indeed that´s the one.. However he heard them at my premises ...And yes, excited is the word, spot-on.

[jabbr]

On 9/8/2018 at 2:30 AM, FredericV said:

 

Not sure if it's the same CEC, but one of my resellers listens to a 28k euro CEC and he finds our 432 EVO MASTER server better:
 

...

In fact in the Aries Cerat + 432 EVO MASTER system that won best of show 2017 in Munich from both Hifi Pig & AVShowrooms, the CEC was taken to the booth as backup but never played. In his home we compared some Bach CD on his CEC and also ripped it on the server. On this system, the difference in favor of the server was not small.

 

It was in response to this that I asked for some detail, and you point to a "study" that demonstrates preference for "Verdi G" tuning (yes its that old) over conventional tuning, and then I ask for more info and you point me to the website... which describes "autotune to Verdi G" ... groan... c'mon man.

 

Look, if I were publishing a CD as a musician and you were retuning it to 432 I'd be pissed. As a consumer, I let the work stand on its own. I upsample because that improves the playback electronics, and room correct because I'm correcting my room, but I want my electronics to play the work as intended by the artist/recording/mastering engineers.

[FredericV]

 

Quote

Look, if I were publishing a CD as a musician and you were retuning it to 432 I'd be pissed. As a consumer, I let the work stand on its own. I upsample because that improves the playback electronics, and room correct because I'm correcting my room, but I want my electronics to play the work as intended by the artist/recording/mastering engineers.

They are actually surprised, one of my customers did the test and played a band's album without telling the conversion was active.

Another example is Jazz Profilactika, who tried several Protools plugins, but found our conversion method to sound better, which is what was used for their album release:
http://432evo.be/index.php/432-hz-en/tick-tock-by-jazzprofilactika

We are NOT forcing you to use the 432 Hz feature. It can do bitperfect but also interesting resamplers such as Archimago's intermediate phase. Anyone should listen without prejudice what sounds best to them.

If you like upsampling, we have a lot of filters to play with:
http://432evo.be/sqi

[look&listen]

23 minutes ago, FredericV said:

Another example is Jazz Profilactika, who tried several Protools plugins, but found our conversion method to sound better,

Can you connect it to effects pedal?

[amir57bs]

20 hours ago, jparvio said:

This might be partially OT, but eventually touches the issue...

 

An interesting debate yet not without it´s problems. Not to take sides on (computer vs. transport) debate I somewhat understand Amir´s agenda. Having said that I can relate to the "modern sound" is worse than "old" -proposition.

 

Having heard a variation of systems I sometimes find myself wondering have we lost the way. I do respect those tour de tech force-products that try to (over)achieve. I am sure the few who can afford these multidollar monuments (big Wilsons, YG´s, Magico´s and so on) will be happy until the next generation or update is available. 

 

My motto is simple; if the sound does not touch You, it is not worth it. As we know, an interaction between room and speakers is the most important factor in creating aural wonders. Yet other speakers create the hard to describe "feeling factor" better than others. What I find intriguing is that these special moments are rarely achieved by the latest technological improvements (ie. better driver-, cone- etc. materials). Admittedly new is almost every time better by definition; resolution, neutrality, less distortion... But involvement, feeling, emotions are better served with something else (YMMV).

 

Once again this became evident a while ago when a friend of mine was looking for a system from ground up. After listening to many setups he faced this same questions as we all do; what is the (personal) goal and how to get there. His process is still ongoing but a milestone has already been found; big 15" paper basses, horn structure for tweeter and over 94dB sensitivity, tubes, r2r-dac... Well You get the point. 

 

I cannot wait until we get to dwell deeper into digital front end... Perhaps he ends up buying a turntable instead

 

 

 

 

 

  

 

 

Thank you

[marce]

On 9/8/2018 at 6:36 AM, Em2016 said:

Eeep, from that link:
 

"CEC sounds smooth and analog like NOT because of the belt thing, but because CEC engineers modified the SPDIF trace to be unrecognizable as square. They filtered it and reshaped it to be without the low square harmonic fundamental. So DACS read it but lack bass. This is the fake analog smoothness which was created to fool people into the belt bullshit story.  And NOBODY EVER discovered this."

 

The problem is that statement is incorrect (in fact its complete and utter twaddle), the shape of the SPDIF wave has nothing to do with the resulting analogue bass output, the analogue signal is buried in the bits, surrounded by extra data such as reed-solomon information etc.

[marce]

On 9/11/2018 at 12:22 PM, marce said:

The problem is that statement is incorrect (in fact its complete and utter twaddle), the shape of the SPDIF wave has nothing to do with the resulting analogue bass output, the analogue signal is buried in the bits, surrounded by extra data such as reed-solomon information etc.

Further to my above post, the square wave looks to have been passed through an integrating circuit which is... a low pass filter so he can't even get his filtering right...

[amir57bs]

18 hours ago, marce said:

Further to my above post, the square wave looks to have been passed through an integrating circuit which is... a low pass filter so he can't even get his filtering right...

If you look at kondo (Audio note Japan) transformer square wave you will think it is not right but kondo gives you the best sound . Do not judge audio by oscyloskop.

[marce]

Most people who are not involved with digital design or signal integrity think a square wave is the text book image of a square wave complete with 90 deg. corners. In reality most are not, DDR memory being a prime example. In SI the worse case situation is a none monotonic rising or falling edge on the wave causing spurious switching, a perfect square wave is also not desired, the rise and fall times should be chosen to be as gentle as possible without introducing other problems such as ISS. The slower the rise times the less high frequency energy being fed into the line and the more the wave edge will slope.

What worry;s me is that someone who improves digital equipment could come out with such a statement, that is wrong, the frequency content of the transmitted digital information has no relevance to the bandwidth of the resulting audio analogue signal after conversion. Also the fundemental frequencies are present, it is the high frequency harmonics that have been filtered from the wave shown, resulting in a very slow rise and fall time, but still leaving a wave that will cause periodic switching, thus the data gets through.

[Speedskater]

Digital signals have nothing to do with square waves.  Even if the signal looks somewhat like a square wave on an O-scope. The digital information is in the leading (and trailing) edges or slopes of the signal. And these slopes are often in some way low-pass limited.

[amir57bs]

none of us have any experience about what different square waves affect the sound.

 

we just know the CEC TL0 3.0 is very good

[marce]

Square waves don't effect the sound, they carry the data, that is converted at the DAC to an analogue signal. What we ndo understand in detail is how that data is delivered, bit perfect 0 and 1's all the way.

[amir57bs]

On 9/17/2018 at 12:58 AM, marce said:

Square waves don't effect the sound, they carry the data, that is converted at the DAC to an analogue signal. What we ndo understand in detail is how that data is delivered, bit perfect 0 and 1's all the way.

 

I think they affect, if there was no difference then all transports should sound identical

[marce]

No they don't, the shape of the square wave as long as the data gets through has ZERO (no) effect of the resultant analogue conversion... The information is carries in the 0's and 1's not in the wave, the shape of the wave as long as it is within the required tolerances is separate from the analogue output.

[amir57bs]

we could not get better sound without designing audiophile grade motherboard.

 

all computers cause error in their data processsing/transfering but the error correction make perfect transfer but in usb audio we have no error correction .

https://darko.audio/2016/05/gordon-rankin-on-why-usb-audio-quality-varies/

 

Via email, I posed the following question to Rankin: when I transfer a file over USB to an external hard drive it doesn’t make transfer errors – the file at the destination is the same as the source – so why should sending digital audio over USB be any different?

What came back was an epic reply. Strap yourself in – we’re going for a ride.

“While all of these interfaces (Firewire, SPDIF, USB, Ethernet, Thunderbolt so forth) have specifications. The % differential from one supplier to another in electrical, cabling, device and host seem to vary quite a bit.“

“All data moving between a host computer and a device over USB is done electrically. There are different speeds and different protocols that determine how a device and the host communicate.”

“Any interface between two points cannot be totally error free. If you use a hard drive over USB, Ethernet or Firewire there are transmission errors. That means the transmitting device is told to resend the packet that has the error in it. Most of the time this is one bit in a packet size of length X.”

“Remember, the carrier is modulated on the data so the larger X, the bigger chance of errors. Also the faster the interface the more chance that there will be an error.”

“The three main USB transmission protocols are Bulk, Interrupt and Isosynchronous. Bulk (used for data transfer to a hard drive) and Interrupt are error correcting. Isosynchronous (used for audio) is not.”

“Bulk and Interrupt are immediately NAK (negative acknowledgement). The receiver is designed to detect a bad packet immediately and the packet is resent.

 

 

[amir57bs]

“For USB audio, the receiving device is basically translating a serial stream of data with a clock interwoven throughout. At the end of the packet sits some sort of block check. If the block check does not match the data then that packet is flagged as an error.”

“With Isosynchronous USB transmission, packets are sent without any error correction / resending. But guess what? This is the USB protocol used for audio frames. The bad news is they are not error free. The good news is these Isosynchronous frames are afforded the highest priority in the system.”

“A couple of years ago, I bought an expensive Tektronix USB setup. I have had protocol analyzers since designing my first USB DACS some twelve years ago. The Tektronix is useful because it allows me to see errors better both in electrical and data packets.”

“The big thing that many people don’t realize is that not all USB ports are created equal. Not all USB cables are created equal and it’s the same for devices and even operating systems. Since getting the Tektronix I have tested probably thirty different USB cables on the fifteen computers in my lab. These computers run a variety of operating systems and the Tektronix results vary between computers even when the cable remains the same. Lets just say it’s not as pretty as I thought it would be.”

“Just a couple of things to think about in regards to USB ports. First, look to see what else is located on that tree. Each USB port can handle 127 devices. Sometimes there are additional ports hidden (inside your computer) and there are internal devices sitting on those ports – this could be the same tree that is hosting your USB DAC”.

[amir57bs]

“Speed plays an important part in all of this too. You may have heard the terms UAC1 and UAC2 – these are USB Audio Class protocols. UAC1 was designed for Full Speed device and host interaction. A data packet is sent every 1ms. In that packet are up to 1023 frames.”

“In high speed or UAC2 those 1024 frames each contain eight micro frames. Therefore, the amount of data we can send over UAC2 is basically eight times greater than that of UAC1. But with more data at faster speeds comes more errors and system configuration becomes harder. I almost never see an error on a UAC1 device, on a UAC2 device I can pretty much count on errors in both directions.”

[amir57bs]

Guys, 

Look I have a ton of computers here. Isosynchronous frames have the highest priority on the USB buss. 

Lets remember how async works first. The computer sends a stream of samples to the DAC. The DAC is setup to send a feedback pipe back to the computer to indicate if it needs more, less or the sample stream is fine. The USB "IN" for the feedback pipe is a select protocol. The computer sends the IN and the DAC sends the feedback frame. The feedback frame is isosynchronous and therefore does not have error correction (big problem). 

The number one problem with asynchronous USB is usually one of the following problems: 

1) Cable is poor and the computer sends the IN and the overhang of the cable effects the feedback pipe sent from the DAC to the computer and therefore the computer sees this as an error and discards it. This actually happens allot. Especially with longer cables. 

2) The software in the DAC does not compensate for the feedback correctly and the buffer is effected by this and an overrun or underrun occurs and pops and clicks. 

3) Poor USB on the computer side, there are many devices off this port including the DAC and the computer device interface effects the performance of the product. Just move the damn cable and it will probably sound better. 

When you have a number of oscillators or PLL in the DAC circuit #2 has to compensate for the this. I have designed some software for companies (not mine) that do this as they have several (more than 2) or have specialized oscillators for some other aspect of the product. In all cases of these designs we were able to tweak the software in the feedback to optimize the product. 

Hey the good thing is most of these have upgradable software so it can be fixed. 

Thanks, 
Gordon Rankin

 

 

https://db.audioasylum.com/mhtml/m.html?forum=pcaudio&n=141913&highlight=Gordon+Rankin&search_url=%2Fcgi%2Fsearch.mpl%3Fsearchtext%3D%26b%3DAND%26topic%3D%26topics_only%3DN%26author%3DGordon%2BRankin%26date1%3D%26date2%3D%26slowmessage%3D%26sort%3Ddate%26sortOrder%3DDESC%26forum%3Dpcaudio

[amir57bs]

On 9/18/2018 at 8:39 PM, marce said:

No they don't, the shape of the square wave as long as the data gets through has ZERO (no) effect of the resultant analogue conversion... The information is carries in the 0's and 1's not in the wave, the shape of the wave as long as it is within the required tolerances is separate from the analogue output.

 

https://db.audioasylum.com/mhtml/m.html?forum=pcaudio&n=126223&highlight=Gordon+Rankin&r=&search_url=%2Fcgi%2Fsearch.mpl%3Fsearchtext%3D%26b%3DAND%26topic%3D%26topics_only%3DN%26author%3DGordon%2BRankin%26date1%3D%26date2%3D%26slowmessage%3D%26sort%3Ddate%26sortOrder%3DDESC%26forum%3Dpcaudio

 

 

https://db.audioasylum.com/mhtml/m.html?forum=pcaudio&n=126223&highlight=Gordon+Rankin&r=&search_url=%2Fcgi%2Fsearch.mpl%3Fsearchtext%3D%26b%3DAND%26topic%3D%26topics_only%3DN%26author%3DGordon%2BRankin%26date1%3D%26date2%3D%26slowmessage%3D%26sort%3Ddate%26sortOrder%3DDESC%26forum%3Dpcaudio

[marce]

It goes on about data errors, hence the signal integrity of the square wave is not up to the task, that should not happen these days, a badly engineered interface...

[amir57bs]

23 hours ago, marce said:

It goes on about data errors, hence the signal integrity of the square wave is not up to the task, that should not happen these days, a badly engineered interface...

 

Check again john post 

[marce]

Read it, if the digital data gets to the DAC bit perfect then everything is hunky dory, the shape of the square wave as long as it allows the data to be read has NO bearing on the resultant analogue output, its not an analogue signal it is a digital waveform transmitting a packet of information as bits. Most digital waveforms are not perfectly square, DDR memory is a perfect example, the data gets through, thats how it works, whether its an audio DAC or a high speed 24 bit DAC working in MHz's. As long as the digital wave is withing the signal integrity limitations for the interface then the data gets through and how that digital wave looks has no bearing on the analogue output.

[audioking87]

On 9/28/2018 at 7:31 AM, amir57bs said:

 

Check again john post 

 

Amir are you still selling your CAD gc3?