Berkeley Alpha USB still relevant?

[mansr]

1 hour ago, The Computer Audiophile said:

I must disagree with much of your post. Your confidence knows no bounds and your perception of audiophiles is a bit insulting and uninformed.

I thought he was spot on.

 

11 minutes ago, tmtomh said:

Anyone who uses "audiofools" in an argument immediately drops in credibility in my book.

Anyone who pays hundreds of dollars for a piece of aluminium the size of a shirt button that costs pennies to make, believing it will enhance the sound if glued to a wall, deserves being called an audiophool. There's no other way about it.

[mansr]

1 minute ago, The Computer Audiophile said:

That's an easy way out for you. I could claim the same thing with respect to anything on Earth and anyone who doesn't engage the points fails. 

 

The legitimacy of the points doesn't matter because anyone who fails to address them fails in your book.

Why are you attacking LowMidHigh like this? Did he steal your lunch in high school?

[mansr]

Just now, The Computer Audiophile said:

You're overlooking many of the details that effect purchasing decisions and if the little aluminum does anything good or bad. If the aluminum rings at a certain frequency in the room, like some of the products currently available, there may be a difference heard by the consumer.

No, there may not. There is simply no basis for this in our physical reality.

 

Just now, The Computer Audiophile said:

In addition, if one makes a million dollars per week, is it really foolish to spend budget dust on one's own amusement in a hobby s/he loves?

Yes, it is still foolish even if it is not reckless.

[mansr]

5 minutes ago, The Computer Audiophile said:

S/He rubbed me the wrong way with his/her incredible confidence that glossed over facts.

But you tolerate far worse from others.

 

5 minutes ago, The Computer Audiophile said:

I have several devices about which s/he speaks and I have personal experience listening to them and talking to the people who make them and competitors who talk about all the devices

I don't see a single product or manufacturer mentioned by name in the post you responded to.

 

5 minutes ago, The Computer Audiophile said:

Not every engineer in audio peddles snake oil. Some of them are brilliant and capable of giving me an unbiased opinion about other products in the marketplace.

Nobody is suggesting otherwise. All LowMidHigh said was that many moderately priced products perform equally well to much more expensive ones. I really don't see why that should "rub you the wrong way."

[mansr]

3 minutes ago, The Computer Audiophile said:

You mentioned an entire class of devices. In my book, however thin you may believe it is, that means you're talking about at least two products in that class. 

Yes, the topic was USB to S/PDIF converters and whether a $1900 device is worth the premium compared to the functionally similar $179 Schiit Eitr, for instance. I see now that your glowing praise for the Berkeley Alpha is quoted on their website. Perhaps this has something to do with your unusually defensive posture.

[mansr]

3 minutes ago, The Computer Audiophile said:

There's more to it than that. Much of it I don't see.

Surely you have seen at least one post by GUTB.

 

3 minutes ago, The Computer Audiophile said:

There's no need to mention a product or manufacture when one says "all those converters are pretty much alike." All means all.

Unless proven otherwise, there is every reason to believe they are. The quality of the clock matters, of course, but that's it. Crystek sells excellent oscillators for about $25.

 

6 minutes ago, The Computer Audiophile said:

If that's all s/he said, I would've agreed 100%. What I read in the original post was that a Kia Is equal to a Ferrari if one only uses off the shelf parts and solid engineering (as defined by Kia).

A Kia is equal to a Ferrari for the purpose of driving to the store and buying groceries. The fancier car won't make the food taste any better (though the shopping trip can certainly be more fun).

[mansr]

28 minutes ago, The Computer Audiophile said:

As an engineer, how can you say a Synergistic bell that audibly rings can't be heard?

That's not what SR says they do. Besides, a piece of aluminium that shape and size doesn't ring audibly. If it did, I don't see how those devices could be anything other than annoying.

[mansr]

1 minute ago, The Computer Audiophile said:

Take step back. The reasons for that quote are that the device is spectacular. I have first hand experience with the device over several years. Can you see this through non-colored glasses and understand that my personal experience is contradictory to that of lowmidhigh and that's the reason for my disagreement?

When you wrote that review, the cheaper devices didn't exist. Are you saying that, today, the Berkeley Alpha is worth the 10x higher price over the Schiit Eitr? That would be an opinion you're entitled to, but I don't think it unreasonable to disagree.

[mansr]

8 minutes ago, The Computer Audiophile said:

You are trying to have it both ways. 

 

You say Synergistic is nothing but snake oil, now you want to believe them and tell me, "That's not what SR says." Hmmmm. 

If a device has no clear mechanism for operation, it is not unreasonable to start by examining how the manufacturer claims it works rather than other equally implausible theories.

 

8 minutes ago, The Computer Audiophile said:

Maybe we are talking about different products. I'm talking about this one - http://www.synergisticresearch.com/acoustics/passive/acoustic-art/

I was talking about their HFT devices.

[mansr]

12 minutes ago, Blake said:

I think the "worth" question is personal (subjective) and there is no black or white correct answer.

Right, and that's why I don't see any reason for Chris to get so worked up over it.

[mansr]

9 minutes ago, R1200CL said:

If Schiit was up on the test, I think it should be compared against the UltraDigital or the SU-1. And with LPS-1.2. 

 

Those are probably the best devices available present in that price range. 

The question was what difference, if any, is provided by the more expensive Berkeley device.

[mansr]

1 minute ago, LowMidHigh said:

I did and ABX between the Eitr and Mutec MC3+USB ($179 vs $1,100),

I called it out right 8 out of 15 times...If at all, the Etir sounded more natural.

8/15 means you might as well have flipped a coin. It's the result I would have expected.

[mansr]

20 minutes ago, firedog said:

When he wrote that review, several cheaper devices existed.

The Schiit didn't, though, and that was one of the ones mentioned as an alternative.

[mansr]

4 hours ago, firedog said:

Well, the iFi people basically implied he doesn't know how to use his measuring equipment properly. See their reaction to his measurements of their little PS. 

The iFi people are basically arrogant jerks that throw insults at anyone who dares criticise their products. They even had a post by Archimago taken down on bogus copyright claims.

[mansr]

32 minutes ago, firedog said:

Whether they are arrogant or not (and I agree they sometimes are) has little to do with the validity of what they wrote about the proper way to use the Audio Precision device. Neither does the reaction to a post by Archimago. 

Their history of dealing with criticism means I simply don't trust that they are being truthful when accusing Amir of misusing the equipment. Maybe he is, but I'd like to hear that from someone else with experience using it.

[mansr]

36 minutes ago, barrows said:

Ayre does the following:  

 

1. The USB receiver circuit is powered by an independent, isolated power supply.

2. The USB receiver circuit is isolated from the rest of the DAC by optocouplers.

3. The masterclock(s) are located on the "clean" side of the isolation, close to the input of the DAC/processing stage.

4. 1 & 2 above insure the DAC and analog stage are isolated from noise produced both by the source component (USB feed) and from the USB processor itself.

That's a pretty obvious way to design things, if you ask me. The isolation devices should only add a few dollars to the BOM.

 

36 minutes ago, barrows said:

5. The digital feed to the DAC/processing stage is re-clocked, directly by the masterclock, right before the dAC/processing stage.

This may not even be necessary. For many DAC chips, only the master clock matters for performance. The bit and word (LR) clocks typically need to be synchronised to the master clock, but no particular phase relationship is required. The first thing that happens to the I2S input is deserialising and deinterleaving, and after that the bit/word clock signals are no longer used. One could say the data is "reclocked" by the input stage of the chip.

[mansr]

7 minutes ago, barrows said:

While I agree in a sense, for instance with an ESS chip running in its (normal) asynchronous mode with its internal DPLL active, this is not the case for the same chip(s) if one wants to run them in the synchronous mode.

The resampling filters, sigma-delta modulator, and output stage are clocked by the master clock, not the I2S bit or word clock. As I said, the I2S input needs to be synchronised with the master clock, but no phase relationship is required. Here's a quote from the DSD1793 datasheet:

 

Quote

The DSD1793 requires the synchronization of PLRCK and the system clock, but does not need a specific phase relation between PLRCK and the system clock. If the relationship between PLRCK and the system clock changes more than ±6 PBCK, internal operation is initialized within 1/fS and analog outputs are forced to the bipolar zero level until resynchronization between PLRCK and the system clock is completed.

 

A little jitter on the I2S inputs is completely harmless.

[mansr]

13 hours ago, Miska said:

In DSC-1 the timing is entirely on BCLK as long as it transitions somewhere within stable period of the data lines, because the value goes through a latch driven by the clock. But due to design, the phase between data lines and BCLK need to stay well within stable period of the data lines. Since for this kind of design, there's no higher speed MCLK nor low speed WCLK, one can utilize lower phase-noise of 512x (22.5792/24.576M) clocks without need to frequency dividers or DPLL's.

The clock that matters is the one driving the D/A conversion stage. In your DSC-1 design, this is the DSD bit clock. In PCM DACs it is typically a "system" or "master" clock at 12.288 MHz (256x) or higher. The I2S clocks are not critical. The datasheets even say so explicitly. I quoted BB/TI above. Here's AKM:

 

Quote

The external clocks, which are required to operate the AK4497, are MCLK, BICK and LRCK. MCLK, BICK and LRCK should be synchronized but the phase is not critical. The MCLK is used to operate the digital interpolation filter, the delta-sigma modulator and SCF.

Quote

The AK4497 has a DSD playback function. The external clocks that are required in DSD mode are MCLK and DCLK. MCLK should be synchronized with DCLK but the phase is not critical.

 

[mansr]

26 minutes ago, hopkins said:

I tend to think it is also a question of how the software is generating the USB signal, and more à question of timing (how the signal output deviates from the audio frequency) but I may be wrong...

Software doesn't generate the USB signal, hardware does. The host controller handles the low-level protocol without software intervention. DACs using the asynchronous/adaptive interface (all of them these days) do not depend on the precise timing of USB data.

 

26 minutes ago, hopkins said:

Simple buffer settings in Alsa unfortunately don't do the trick.

No, those settings determine the size of the ring buffer the USB hardware reads from. The individual USB packets are much smaller, and the hardware doesn't care how often the source memory address repeats.

 

26 minutes ago, hopkins said:

There is no guarantee that a USB device attached to a PC (regenerator, reclocker, convertor... Whatever) will improve the USB audio signal.

Of course not, since there is always the possibility that the upstream port is better.

[mansr]

8 minutes ago, hopkins said:

Well in that case something else is going on... With the exact same hardware, how can you explain differences due to the software? How can you explain that with the exact same hardware and software people hear differences between various settings such as buffering?

I'm not convinced they do.

[mansr]

28 minutes ago, Ryan Berry said:

Some have various levels of digital processing that sounded better to the people making it that isn't even an option to turn on and off while others have explicit digital processing. I don't think the software influences noise, but from the time Charley was working with a team to make a software solution, I can say that there's quite likely some influence in what actually is sent to the DAC from the file.

Comparing software, in the context discussed here, without verifying bit accurate output is meaningless.

[mansr]

11 minutes ago, Ryan Berry said:

Certainly so.  I'm just not aware of a list of verified bit-perfect solutions.  For the average user looking at the options out there, it's pretty reasonable to believe that many of them are not listening to such a player.

Anything using exclusive access to the sound device is likely to be bit perfect unless configured otherwise. All the usual players discussed in these pages have the ability.

[mansr]

34 minutes ago, hopkins said:

I seem to remember that the processor's activity causes noise (bursts), and that there are ways to "stabilitze" it, but nothing more specific either.

An audio player uses less than 1% of the CPU time. Even if everything it does is managed to perfection (whatever that means), the remaining 99% of what the computer does is entirely beyond its control.

[mansr]

2 minutes ago, hopkins said:

the idea that the software can have an impact is relatively widespread

A lot of flawed ideas are or have been very widespread, not only in audio.

[mansr]

18 minutes ago, barrows said:

Well, many in the pro audio field (generally considered to be less "insane" than audiophiles) believe in the sound quality differences between pro audio recording/mixing softwares (recall that Amarra comes to us from the developers of the pro audio software: Sound Blade).  Of course pro audio users are not using it in a generally bit perfect fashion most of the time.

I have verified that Cubase is bit perfect with default settings. Obviously, a typical production workflow will involve some amount of processing, and this will for sure vary between software packages.

 

18 minutes ago, barrows said:

Of course I use Audirvana Plus and oversample in it, generally to DSD 128 or 256, so not bit perfect at all anyway.  This also uses a great amount of processor resources with my preferred filter settings (just an I5 here as well).

Upsampling, whether PCM or DSD, obviously alters the data (that's the point), and as such different algorithms might sound different.