Berkeley Audio Design Alpha DAC Review

Well done, Chris. Thanks for your hard work.

I've placed the order few weeks ago, but the dealer replied me that the delivery is postphoned as the team is upgrading some of the parts, esp. the power supply. Have you heard about it?

Best wishes,
av

Very good thanks.

One point though..at $5000 its no bargain, relative or not.

$500 or less...maybe.

That seems to be a very nice product.

dCS does all that too, and even more : digital outputs, word sync I/O, dsd, ...
And quite interestingly, the power supply is also having a little hum, when you come close.
Ever done a true comparison (I mean outside a show) ?

Elp.

Thank you Chris for this very interesting review of Berkeley Alpha.

I have to admit it would be interesting to read your review of a much cheaper component that is making some hype in computer audio community, I got in mind ASUS Xonar Essence STX sound card.

http://en.expreview.com/2008/10/20/asus-xonar-essence-stx-nice-sound-car...

I am looking forward to reading your opinion on it, especially when using it directly in connection with some good headphones like Grado.

thank you,
limesinferior

It would be preferable to start a new thread than to hijack the Berkeley Alpha DAC review for a sound card.

My personal opinion is that the ASUS is a lot of hype. Not that it isn’t a good soundcard but there seems to be a ton of postings on the Internet, but no one has heard it or has posted any listening evaluations.

ASUS makes several other decent soundcards but I wouldn’t know what makes this one so much better. In addition there are several other good soundcards including the Lynx L22. In general, people have found internal soundcards to be a cheap and decent alternative to doing what CA is advocating here. In fact, I would say that the audio out from my MacBook Pro connected directly to my amplifier is pretty darn good.

Anyway the Berkeley Alpha DAC is a proven winner with substantive reviews and evaluations and that’s no hype.

Hi Elprior,

Some questions concerning your dCS, I hope you're willing to share.

Are you using the Delius or Elgar Plus ?

Which interface are you using ? And is that able to play 192 khz.

And last, do you upsample, and if so, what is the impact of it, and did you try softwarebased upsampling ?

Looking forward to your reply.

Jeroen

Hi Chris

Thank you for the fabulous review. You are 100 % on time and target. dCS I had, I like the alpha Dac better ... and your site anyway. Keep up the great work. Thank you again for the educational results your site has to offer.

Greatings Stephan

I found the review of Berkeley's Audio Design Alpha DAC very useful, especially the part which compares pros of alpha and minerva. It was not my intention to "hijack" ... anything.

Hi Jeroen,

I'm certainly willing to share :).

I'm using the Delius along with the Purcell (upsampler).
Everything is being played at 192khz (or 176khz depending on my state of mind), although there is an annoying limitation here.
The Delius handles 192khz through dual aes or dsd.
The Purcell can only handle 96khz as an input (whatever the input may be).

So the upsampling so far is handled by the Purcell.
The benefits I can hear are numerous :
1) There is much more air between the performers, while everything remains connected.
2) The decays are much smoother.
3) The sound is coming to you (wider soundstage).

Now, I can think of drawbacks too :
1) The pace of the music is somehow quieter.
That is a conclusion I have come to when you are using components that tends to the analogue feeling. You are having a much smoother sound, much more information, air, ... but the rythm is a bit slower.
2) Can't use the Purcell as long as I need a 192khz capable input (as stated above).

As for using my computer as a drive, I'm still stuck with my loosy components, that ain't compatible with ASIO.
So I'm being SRC everywhere in my windows xp based system. I'm willing to give the Lynx card a try, but a forum here is showing someone that can play easily through the dCS, and another one who can't (PLL issue it seems).
Hopefully, I'm about to test the Mac solution with a friend of mine.
I'll let you know where the thing goes with it.

Elp.

Hi Chris,

Thanks for the review although I didn't really learn anything I didn't already know from my own audition, albeit brief, of the Alpha DAC in a friend's system.

I wonder if you compared the Minerva and Alpha DACs using a digital interface that is common to both, i.e, the AES output from the Lynx AES16 PCI card in your PC (or Lynx AES16e in the case of your Mac)? If not how do you know that the sonic differences you noted were not, at least in part, attributed to the Lynx AES versus Firewire interfaces rather than the DAC chip, filtering or analog output stage?

Best Regards,
Aaron

It is a good piece of writing, but like any review it can only tell you so much, it is important to hear every component in your own system, and if possible compare to the other 'possibles' keep up the good work Chris.

Thanks a lot Elp.

So far I didn't buy an dCS, but it is on my shortlist.

When you did some testing with your friends Mac it would be great to hear about it.

Jeroen

Great to hear about this fine sounding equipment.

First of all, technically- I known I don't need a preamp to play from my computer server when I use the DAC. Can I plug my Tom Evans Phono Preamp into the Berkeley and still hear great vinyl sound?

And what sound difference am I likely the experience when I eliminate the BAT tubes in the chain? My amp is solid state.

Thank you

The signal from your phono is analogue.

Coops is exactly right. Your phono signal is analog and the Alpha DAC does not have an analog input.

If you remove the BAT tubes you will remove the sonic signature of the BAT tubes. All equipment has a sonic signature and acts like a filter down stream. Removing the BAT tubes will remove the filter.

So, in order to incorporate a Berkeley and keep my phono- I have to utilize a preamp to connect my phono pre to. Thus I will inevitably run the DAC into the preamp as well- all to go to the amp?

I cannot run the Berkeley directly into the amp without giving up phono.

Am I misunderstanding something here?

Correct. Unless you convert analog to digital before the Alpha DAC.

Trying not to be a doofus here, but ...

If I convert my phono to digital so that I can utilize the DAC and eliminate my preamp, I will be defeating the very analogue sound I enjoy, correct?

Practically speaking, running the Berkeley straight into my amp eliminates my phono option. I have to stick with my preamp if I want to maintain my phono as best as possible. ???

thank you

Exactly right. I just wanted to through the option out there for you to consider.

Sorry for the dumb question, but wouldn't using the digital volume control on the Alpha essentially eliminate the "bit-perfect" signal going to the DAC inside? I understand that it would be bit-perfect when it hits the unit, but wouldn't the digital volume be applied before the DAC stage? Is there a type of digital volume control that hits the analog stage after the DAC got the bit-perfect signal?

Hi Jay - Very good question and one that is best addressed by the people at Berkeley Audio Design. But, I will give you my opinion on this one for what it's worth. A digital volume control, as well as many other things, will most likely change the audio stream so it's not bit perfect. However, delivering a bit perfect stream to the unit itself is probably OK with the Alpha DAC. I'm guessing most other DACs change the audio stream sometime between when it hits the unit and when it's delivered to the actual DAC chip. Upsampling and downsampling are very popular whether a consumer is aware of it or not. For example some DACs allow you to enable upsampling while others upsample without you even knowing. I guess the signal is likely to be changed at some point before the actual DAC chip with most DACs. I think if implemented correctly a digital volume control can be great as it allows you to remove the filter of a preamp and additional cables.

This is just my guess on the situation and it could be totally off base. Hopefully someone can jump in and lend a hand with the subject :=)

Hello,

very good question indeed.
I think the answer is up to the implementation. You can't really know what's going on in your dac, unless you designed it or asked explicitely the constructor, and obtained a clear answer.

Nevertheless, with such units as the Alpha Dac, you can bet Chris' explanation is the expected result.
They must have designed it as good as possible, so as to bypass the use of a preamp.

Still, I know vendors that used to think the same, and changed their mind with high-end setups, where the use of a preamp seems to add more than it removes...
Again, this is all a question of taste (and money a bit).

I can't wait to get an explanation from Berkeley Audio Design.

Elp.

The Berkeley Audio Design explanation (if they want to reveal their secret!) will certainly be interesting. I suspect your answer is right, btw, Chris. I have only a very basic understanding of digital volume control, but have always assumed it to be a killer of "bit-perfection," based on my very limited reading and understanding. Regardless...if it sounds great, I don't personally care at all, and would use the feature. Just morbidly curious more than anything.

This review has thoroughly peaked my interest in this DAC, so thank you. As a guy who's been questioning the pre-amp in my all-digital system since the beginning (yet it's still there after numerous attempts to remove it), I applaud the digital volume feature and I'll assume at this level of technical design (and cost), they would not have wasted their time with a digital volume that wasn't extremely well-done. Can't wait to try it myself. Attenuation, in my experience, is the most tricky part of this whole "all-digital" equation and one that holds the most "you must just try this or that in your system before you know what sounds best (or worse, in many cases)." We've all heard it done "not well" I suspect and quickly spent gobs of frustrating time and money getting that part of the chain "out," or more likely "replaced with something less objectionable." And Elp, I also think you're dead-on in that for whatever reason, there are occasions where the pre-amp seems to add more than it takes away (again, that's with only the most well-done ones I've tried). Another head-scratcher for me, but I long-ago gave up trying to think of that step as just invisible attenuation--my passive pre is in line because the system just sounds better with it for some reason. I hope the Alpha gives me cause to remove it--only because that would then mean my system, for the first time, sounds even better without it!

The soundstage comments alone have me very excited to try this DAC. Everyone else experiencing that same soundstage breadth and depth improvement relative to their current DACs?

Please allow me to share my thoughts regarding several questions I've been reading regarding DAC implimentation:

1. Several of today's DAC take the "bit perfect" data and clock to a different frequency.
2. Controlling volume control for many is an excellent solution - DACs with analogue attenuation may have issues as well with the parts used in the analogue attenuation. Higher-end preamp manufactures always have issues finding transparent volume controls and higher quality volume pots may cost more than the DAC.
3. For those desiring to combine analogue source material, whether it be phone of L-R from a surround system, may consider a single-ended or balance switch that feeds the amps directly.
4. Some users will prefer to drive the DAC, (w/volume), directly into the amps, others may prefer using them with a preamp.
5. There is much more to come but 1-4 may help some of the readers.

Regards.....

Chris-

It seems to me that some damping material on the bottom of the DAC could solve the hum issue.

Let me know the dimensions of the bottom, and I'll send you some.

Aloha,

Dan

He,

I don't think so, that's the power supply circuitry that's audible, not the case.
Am I wrong ?

Elp.

"I'm guessing most other DACs change the audio stream sometime between when it hits the unit and when it's delivered to the actual DAC chip. Upsampling and downsampling are very popular whether a consumer is aware of it or not. For example some DACs allow you to enable upsampling while others upsample without you even knowing. I guess the signal is likely to be changed at some point before the actual DAC chip with most DACs."

Chris,
I believe you may even be understating the likelihood of unannounced digital signal processing that happens... e.g., Amarra are proposing a dedicated device for hardware based DSP as a required component of their Sound Server, are they not?

clay

If the hum is vibration of the power transformer, a PS Audio Hum Buster might fix it.

The Hum Buster cures hum caused by DC offset in your AC power line. However, this usually happens only in the large transformers used in power amps.

Hi Bob - I believe the PS Audio Hum Buster is for a different type of hum altogether. My hum issue does not produce sound out through the audio system. It is the case itself vibrating which is because of the design of the power supply. Thus power supply produces better quality audio but not without a small price (vibration).

Chris,

The mechanical vibration you experience is the *only* type of hum the PS Audio Hum Buster is designed to ameliorate. (Specifically, mechanical vibration of the power transformer due to eddy current in the transformer caused by DC offset in the AC power mains.)

The Hum Buster has no effect on electrical hum audible through the loudspeakers.

Very cool Bob. Thanks for the follow-up.

A digital volume control does not specifically enable you to bypass a preamp, as you know the Benchmark Dac1 allows the user to connect directly to an amplifier and it uses a post coversion analog volume control. The difference between a digital volume control and an analog volume control is the analog volume control does not reduce resolution where a digital volume control does. If the Berkeley is not truncating some bits in its manipulation of volume it would be interesting to learn how the volume is controlled digitally.

Here is a link to a discussion of how digital volume control works, www.audiocircle.com/circles/index.php?topic=36767.20
Of course this may not be the only way to do it but am interested in how Berkeley is able to do it without reducing resolution.

Thanks
Sean

Hi Sean - I talked to the designers of the Alpha DAC about your comment, looking for an accurate answer for you. I'll paraphrase my discussion with Berkeley Audio Design below:

You are correct to be concerned about truncating the digital data. This operation not only reduces the resolution, it also creates distortion products that are frequently audible. Berkeley Audio Design does not truncate the data. They scale it using 32-bit math and re-quantize it to the resolution of the DAC (24-bit) using the proper dither. This process does not generate distortion – it maintains the linearity. In addition, Berkeley Audio Design has carefully chosen the gain structure of the DAC to match the gain of typical power amplifiers and sensitivity of typical loudspeakers. This insures that the DAC itself is operating in an optimal fashion with minimal distortion The noise floor of the DAC in a system is not audible from the listening position, and the noise floor of the DAC is much lower than the noise floor of the recordings being played. There is no loss of resolution.

Thanks for tracking down the answer. Certainly sounds like a more sophisticated method of implementing digital volume control

Sean

Bless them for dithering! I would love to hear the Berkeley because the same folks who designed it also made the Pacific Microsonics HDCD Model 2 DAC, a favorite amongst recording studios until Microsoft bought the company for software patents I believe and axed the hardware side completely. Truncating is why I can't stand using Itunes for playback software. It truncates to 16-bit while my DAC processes using 32-bit float to 24-bit using dithering just as the Berkeley does; and compared to Pro Tools Itunes sounds as if everything is straining through the speakers and fatigues my ears. There is a piece of software called Audio Hijack Pro which allows you to use a variety of dithering plugins for use with Itunes which would better suit my DAC, but I haven't coughed up the $$ for it because I use Pro Tools instead. I like having the option though.

Hi David - What DAC are you using now?

Thanks David, for sharing this idea about dithering. I have Audio Hijack Pro, but was unaware of it's potential for applying plugins.

I found a thread on Apple Support forums that offers more details WRT dithering as a possible explanation for quality loss via iTunes playback as compared to 'pro tools', e.g. Logic, Pro Tools, etc.

http://tinyurl.com/9s92uz

interesting...more ghosts to chase.

In addition to Izotopes 64-bit SRC, Wave Editor also licenses iZotope's Mbit+ Dithering, BOTH of which have been licensed for use in Soundblade (the precursor to Amarra's software).

Wave Editor costs $79, free trial is available. :)

enjoy
clay

Hi Guys - Thanks a ton for the Apple forums link. I'm doing similar research right now and noticing some of the very same things others have noticed between iTunes and other applications like Mastering tools etc...

Lately I've been using Samplitude / Sequoia Digital to test playback sound versus other applications. Now I'll have to check out Wave Editor.

David indicated he uses a Digi Design 003 interface in a previous post.

enjoy,
clay

(I deleted and completely rewrote the analysis I posted yesterday.)

Referring to http://en.wikipedia.org/wiki/Quantization_noise, the unweighted signal-to-noise ratio (SN) for a sine wave digitized with a resolution of B bits is SN = 6B + 1.8 dB. Conversely, the effective resolution B of a sine wave test signal as a function of the signal-to-noise ratio is B = (SN-1.8)/6.

Bob Harley's review of the Berkeley DAC says it uses an Analog Devices DAC chip. Their best chip is the AD1955.

Graph "TPC 13" in the AD1955 spec sheet indicates that the "typical" distortion+noise relative to a full scale 1 KHz sine wave is -110 dB, which is (110-2)/6 = 18-bit resolution. For a 1 KHz sine wave attenuated by 40dB or more, the same graph shows the distortion+noise relative to full scale (NOT relative to the attenuated signal level) is -117dB, which is (117-2)/6 = 19 bits below full scale. This means the DAC chip has only 19 bits of precision. Assuming the 20th bit adds a half-bit of resolution via dithering, and assuming Berkeley achieves an additional half-bit of resolution by using two chips in differential mode, the DAC still has only 20 bits of resolution.

Each 6 dB of attenuation reduces the resolution by 1 bit. Therefore, if you use the digital volume control to attenuate the signal 24 dB, you are losing 4 bits of the theoretical 24-bit resolution. Since the DAC has only 20 bits of resolution anyway, this is OK. However, if you apply more than 24 dB attenuation, you will lose 1 bit of resolution for each 6 dB attenuation below 24 dB.

If you use a high efficiency loudspeaker with a high power amp, you probably need to attenuate the signal more than 24 dB, which would result in loss of resolution.

Hi Bob - Thanks for the interesting analysis. I'm going to run this by the guys at Berkeley Audio Design for their opinions.

Yes, I'm using a Digi Design 003 with a Black Lion Audio Micro Clock and am quite happy with the reproduction.

Hi Bob - I did a little research into your analysis and found it's over-simplifying things a bit. I am not qualified to get into the very technical discussions but I do know this is a much more complex issue that can be explained in a few paragraphs by either of us. If it was simple everyone would produce outstanding DACs.

Anyway, thanks again for your analysis. It's really great to read intelligent opinions from readers around the globe, even if we don't all agree.

Hi, Chris

I finally got my alpha DAC few days ago, but only able to hook it up last night. It's a amazing DAC, in addition to the soundstaging, the vocal and the musical instruments are so alive, just like in my room. Thanks for your recommendation.

I also copy the file from HRx sampler in WAV and AIFF format (using Max) in my iMac, connecting to the DAC by a optical cable (Wireworld SupraNova), and to K&H O300 directly. However, from the screen of the DAC, the iMac is outputting 96kHz signal to the DAC. Is there anything wrong in my iMac setting?

With regards,

WF

Hi avreality - Glad to hear you like the DAC. The iMac optical output is limited to 24/96 for OS X. If you use bootcamp and install Windows Vista the optical output will support 24/192.

Thanks for your reply, Chris.

It's really frustrating, I switched my OS from Window to Mac , partly because of the better performance of itune in mac than that in Window. Is the AFI1 or Vesta from Weiss the only solution for 24/192 output ?

By the way, will the light of HDCD on when the output from my iMac is bit-perfect? What can I do if the light is off when playing music in iTune?

Thanks in advance for your help.

av

Bob Stern - I agree with your analysis. It's independent of the DAC actually. The resolution is ultimately impacted in order to achieve proper listening levels.

You can get some relief with some USB and WiFi interfaces that change the data from 16/44.1 to 24/44.1. the extra 8 bits gives you about 10dB. After this you start whittling away LSB's. Sonos and Empirical Audio USB are two examples that change 16/44.1 into 24/44.1.

The optimum scenerio is to have a DAC that can be reduced in volume about 10-12 dB in the analog domain, or at least without affecting the data. If one could do this, then 24/44.1 data could be reduced in volume digitally an additional 10dB without impacting the data. The 8 bits of additional "padding" gives you about 10dB. This way remote controls such as Touch and iPhone can be used from the listening position and you dont need a remote for the DAC.

One would start by calibrating with a quiet track by adjusting the non-destructive volume control until a good listening level is achieved. Then, when a louder track comes on, one can adjust the volume down by as much as 10dB digitally without impacting the resolution. The difference between the loudest and quietest tracks I have found is less than 10dB.

This is exactly how my new Overdrive USB DAC works BTW.

A 3:1 or 4:1 transformer would probably do the trick for the Berkeley. Just enough to reduce it about 12-14dB.

Steve N.
Empirical Audio

Please confirm if the following is correct or if I am misunderstanding something:

Converting by 16/44.1 files to 24/44.1 will not increase the volume of the files during playback. But it will allow me to digitally reduce the playback volume by about 10dB without losing any information.

Or said another way, any digital volume control on the playback of a 16/44.1 file will degrade the sound.

Thus, I should have 24-bit audio files if I want to use the volume control on my Apple remote or iPhone and do not wish to degrade the sound.

"Please confirm if the following is correct or if I am misunderstanding something:
Converting by 16/44.1 files to 24/44.1 will not increase the volume of the files during playback. But it will allow me to digitally reduce the playback volume by about 10dB without losing any information."

Correct, once you have a comfortable listening volume, an additional -10dB comes at no degradation to the data.

"Or said another way, any digital volume control on the playback of a 16/44.1 file will degrade the sound."

Yes, but you may not be able to hear it depending on the resolution of the system, the software and the amount you are reducing the volume. Sometimes the software makes adjusments as well. I recommend that you do your own experiments.

"Thus, I should have 24-bit audio files if I want to use the volume control on my Apple remote or iPhone and do not wish to degrade the sound."

24/44.1 files are definitely superior, not only for the DAC, but allow for some volume control. Expect better sound quality too. BTW, 24/44.1 files are also bit-perfect with Vista, and 16/44.1 are not. If your player S/W does not support this, you can get R8brain or Adobe Audition and convert the files from 16/44.1 to 24/44.1. Give it a try. I think you will be pleasantly surprised.

Steve N.
Empirical Audio

If your 24-bit DAC has a sample rate converter (SRC), the SRC inherently converts 16-bit data to 24-bit data before it undergoes D/A conversion, so it would be pointless to convert it to 24-bit manually.

I agree with Steve Nugent's recommendation if your DAC does not employ SRC.

Incidentally, SRC is condemned by some audiophiles, but it is used in the superb DAC's by Bryston and Berkeley Audio Design. Perhaps it helps that both use the SRC to perform synchronous upsampling of CD data (i.e., upsampling to 88.2 or 176.4 KHz, which is an integer multiple of the original 44.1 KHz sample rate) instead of upsampling CD's asynchronously to 96 or 192 KHz.

I don't think I agree with "if your 24-bit DAC has a sample rate converter (SRC), the SRC inherently converts 16-bit data to 24-bit data before it undergoes D/A conversion, so it would be pointless to convert it to 24-bit manually".

The recommendation is to use a digital volume control on a 24-bit file instead of a 16-bit file. This happens before any DAC, such as the Berkeley Audio DAC. As discussed previously, using the digital volume control on a 16-bit file will degrade the file before it reaches the DAC, regardless of the DAC used.

I was thinking of a DAC that had its own digital volume control.

You're right that the best way to implement digital volume control in computer software is to first convert 16-bit to 24-bit before applying the digital attenuation.

Sorry for my confusion.

can't seem to find this info anywhere, and have not yet seen an openup / teardown of the Alpha DAC to know precisely which components are used in its conversion architecture. Can anyone confirm:

1/ if the Alpha uses a commercial D/A IC, and if so, which one (it's also possible that there may be several of the same IC used in a parallel configuration)? If not, any info on what hardware is performing the D/A conversion function, ie a module with any visible identification?

2/ does the architecture include any obviously-visible device which could perform signal processing (FPGA, digital filter chip, DSP devices ie ADI SHARC or Blackfin, or TI or other) as a preprocessor to the D/A IC/module? Most probably this is the case, as (at mimimum) vol control and HDCD decode functionality must be performed

3/ a DAC "box" can *accept* up to and including 192 kHz inputs, but the internal D/A conversion ICs do not necessarily operate at this sampling frequency. One example is the Benchmark DAC1, which *accepts* 32-192 kHz, but resamples any input sample rate to approx 110 kHz before conversion as part of their jitter-minimization strategy and to run the ADI D/A chip at an optimal performance-vs-sampling-frequency operating point (which was an ingenious system solution for the time when the DAC1 was designed, using D/A chips which are now of course many years old in design). Does anyone have info confirming that the Alpha DAC will actually operate at 176.4 and 192 without resampling? Does the user guide contain any info in this regard, and any performance-vs-operating-frequency characteristics? The website has only some leading- but definately marketing-speak information.

Any pics of internals to be posted?

Chris and Computer Audiophiles,

Just a brief note to say that my Berkeley Audio Alpha DAC arrived yesterday -- the missing piece to put my high-rez music server in operation. I had a wait of a bit less than six weeks, due in part to Berkeley revising production for EIC certification.

The Alpha has been thoroughly discussed here and in other reviews, so I won't go on and on. Suffice to say that it completely lives up to its billing. High resolution files via the Alpha are stunning -- easily the best digital playback I've had the pleasure to hear.

There was some concern about the power supply noise of the early versions of the Alpha DAC. Evidently that has been successfully addressed in the latest DACs now shipping; my Alpha is completely silent and I've noticed no external hum from the unit.

There is nothing original in my music server system -- all of the ideas pretty much originate from Computer Audiophile postings, and I'm very appreciative of have such a great forum to get this information out here. Keep up the good work!

Steve Z

Mac G4 Powebook/Thecus N5200BR Pro/iTunes > Weiss AFI1 firewire to AES > Berkeley Alpha DAC

That's really great to hear Steve!

Hi Chris,
what's the power amp used in this review? BTW, do you a think the Berkeley AlphaDAC will work fine as preamp for a Mc275?

Thanks

Just the end of August, after reviewing this any several other reviews, all giving the Alpha high marks, I bought one for myself, unheard- and have no regrets at all- the performance is impeccable, and has raised CD playback to levels I once thought not possible- a typical example being the Sheffield CD of the Harry James "King James Version", taken from the 1976 recording; for the first time, this disk starts to approach what the vinyl delivered, which is no mean achievement. In this regard one can say it truly demolishes anything else I've heard at $6K or less. Imaging, transient envelopes, midrange textural trueness, are essentially in a class of it's own at this price. Obviously, the recordings that benefit the most are good ones, but there are other recordings that have not been perceived in the past as all that good because they are so demanding or punishing in some respect, but are now reproduced much more realistically and satisfyingly.

The Alpha DAC does deserve the appellation of "closer to the mic feed" and I find it money well very well spent. Mine is currently being used with a Cambridge Audio CD player as transport and with AES/EBU output from a Fireface 800 connnected to Mac Mini, at various data rates and source types. Next to investigate is a card which converts DSD to 24/192 for SACD, to see how the Alpha works with that material.

Thanks for your review Chris.

however what are you trying to accomplish when you say you need "to investigate is a card which converts DSD to 24/192 for SACD, to see how the Alpha works with that material"?

What is your source for DSD material? Are you looking to rip or digitize SACDs?

I had been contemplating the Korg MR-2000S to digitize vinyl at either:

-PCM audio format: 88.2 kHz @ 24 bit, 96 kHz @ 24 bit,176.4 kHz @ 24 bit, 192 kHz @ 24 bit

or

-1-bit audio format: 2.8 MHz @ 1-bit DSD, 5.6 MHz @ 1-bit,DSDIFF format (extension .dff), DSF (extension .dsf),WSD (extension .wsd)

The Korg AudioGate software can convert the DSD formats to a lower PCM audio format.

I assume if you are looking to tap into the SACD layer this would be the legal and relatively inexpensive way to go barring the expense of needing an excellent SACD player. From the reviews and user feedback of the Korg MR-2000S, the DSDIFF result is indistinguishable from the original SACD played back on an excellent SACD player. Of course you lose a little something if you convert to PCM.

I guess I have similar questions as Audiozorro -- what is your source for DSD material? I've been doing a fair bit of research and haven't found a cost-effective way of extracting the DSD information from a commercial SACD. I own an Esoteric X-03 player which can be set to output the encoded DSD stream from an SACD via iLink (firewire), however short of pro gear to decode and/or do sample rate conversion, I haven't seen anything that will let me capture the stream as a file, let alone do decoding and SRC.

I have a Korg MR2000S, the latest firmware and software (which, BTW allows burning a DSD DVD that can be played) but AudioGate will not capture streaming PCM or DSD. 2L Recordings has about 25 minutes DSDIFF music files at their website that can be downloaded and experimented with (and AudioGate will recognize and manipulate them).
I'd love to find a source of DSD music files. Live music recording with the Korg isn't very feasible for me here in rural Montana -- there just aren't many concerts around here.

For archiving vinyl I highly recommend using DSDIFF, either sampling frequency. DSD can be converted to high resolution PCM without the decimation issues and degradation that can occur with PCM to PCM SRC. I haven't compared the Korg/AudioGate "DSD disk" against DSD converted to 24/192 or 24/176.4 yet, but vinyl recorded to DSD and then converted to 24/192 and 24/176.4 and played via my Berkeley Alpha DAC is very, very good. I haven't developed a firm preference yet for 24/192 or 24/176.4 when converting from DSD, though I have read some theoretical and one or two practical comments that express a preference for 24/176.4 when doing SRC from DSD.

I'd like to learn about others' experiences with DSD and DSD SRC to PCM.

Best regards,

Steve Z
Montana, USA

Hi Steve et al.
I was facing similar problems since the only method of getting music off a SACD is to convert to PCM. I found the solution below and it works great for me. Output is 24/96 and with Lynx, Amarra and alpha DAC the result is more than acceptable. For recording I use Amadeus Pro.

http://freerider.dyndns.org/anlage/HiresAudio_E.htm

Greetings Stephan

Yes, my interest is possible methods of SACD capture for the music server (Mac Mini). I do have a high quality SACD player already, and with Pure Vinyl it's possible to do a good capture at 24/192 and create music tracks in AIFF. Pure vinyl also has the ability to create fairly high quality 44.1 from the high rez capture and has an upsampling playback engine that can integrate with iTunes (Mac), though I haven't tried that out yet. Many things on the to-do list at this point, as I'm also finishing up a major speaker project.

But this weekend I'm buying a used Pioneer DV-79 with the intention to install a board from these people:

http://audiopraise.com/vanity/overview.php

which would do a DSD to PCM conversion and output at 24/192- through SPDIF- which I will try with the Berkeley Alpha DAC. I suspect that may be the way to go, and as mentioned earlier, would use Amadeus Pro or another of the audio stream capture utilities available for the Mac to capture the incoming bit stream. Possibly a dicey concept, but I'd really like to get my SACD's on to the music system, especially the single layer ones with no hybrid RB layer which I can't back up in any form currently and can only play on the SACD player (Marantz SA-11).

Chris,
I'm confused again. Ugh. In rereading the Alpha DAC review, you state (I think) that the DAC rolled off transients when used with the MacPro and Lynx AES16e card compared to Media Monkey and the legacy card in the Dell PC. If so, why do you still use the mac as your server? Have the newer iTunes or Amarra made the issue better? I bought a Berkeley, and need to decide on server type. Currently have a MacPro dual core 2Ghz with all my music under iTunes. Should I use this or boot under Windows 7 and use Media monkey or J.River?

Thanks in advance.

Hi rlodad - Good observation and question. Currently I use a Mac Pro OS X Snow Leopard & Win 7, Mac G5 OS X Tiger, MacBook Pro OS X Snow Leopard, Dell Windows XP, and two custom built Linux servers I am working on. They all have different characters to the sound. I need to use all of them to keep up on everything that's out there :~)

Thanks, Chris, but let's cut to the chase. Do you prefer one over the others with your current system? Amarra vs. "virgin" iTunes vs. Win 7? Can you give any subjective listening qualities of each/ We know you think the transients are rolled off some in whatever version of iTunes you were using for the berkeley review, any change with new iTunes. I know you are avoiding the subject, how about just a hint?

:^)

Ha! I'm really not trying to avoid the subject :~) I just don't want people to think there is one "best" solution for all systems. Right now my go-to server for sound quality is the Mac G5 running OS X Tiger, iTunes & Amarra, and outputting audio via a Lynx AES16 PCI card.

Each of my systems has weak points. I think the G5 has fewer weak points or maybe just weak points that aren't as apparent in my system.

Chris,
Thanks for the information. Any idea why the G5 might sound better in your system than the Intel MacPro? We are all assuming it is not because you're a dummy and built a poorly matched system, oh great leader... I have a 2gHz Intel Mac Pro (32bit, bummer...) and am wondering if you have any subjective evaluation of the difference in sound b/t the two and why the G5 might sound "better" or different.

Thanks again.

Hi rlodad - The following (on-going) thread has more insight into this topic than I could ever write myself. Check it out if you haven't already. The Mac Pro is definitely capable of wonderful sound that's better than physical disc spinners, but the G5 has a little something extra. The Mac Pro seems to sound a little dull compared to the G5 and my WIndows XP music server. I stress that this is in my system because a big factor may be the Lynx card. The AES16e is the only card to fit in the Mac Pro whereas the AES16 PCI cad fits into the G5. I'll have to do much more listening with non-AES DACs to better understand the differences between Mac Pro and G5, but that takes a ton of time.

http://www.computeraudiophile.com/content/Magic-Words-Power-Mac-G5-versu...

I have Berkeley Alpha DAC & LYNX AES16 computer interface card.

The Alpha DAC, even by itself with inexpensive CD player digital out, sounds much better than my Wadia DAC - a surprise to me.

High res files, like 24/96, etc., thru computer LYNX AES16 is best I've heard so far - price no object.

Hi Bill,

Which Wadia are/were you using ?

OK, The Linx card seems to be the way to go for desktop applications but how are people attaching to the Berkeley if you have a laptop? Firewire through ? to what? USB to ? to what? S/PDIF if you are lucky enough? I'm trying to get some ideas as I should be able to get my hands on a DAC next week to try in my home system but I have an older laptop with Firewire and USB only. Thanks in advance for any replies. I'm just trying to get some indication as to what most people who are lucky enough to have this DAC are driving it with.

Hi Wattage - Patience will help you on this one. Great Asynchronous USB 24/194 to S/PDIF converters are on the way in a month or two from top notch manufacturers. I know quite a few people using optical TosLink out of MacBooks waiting for these converters.

any update on the Berkeley USB to S/PDIF converter? And have you ever tried the M2tech hiFace, that seems a bargain at that price?

Have a nice day,
Fabrizio

I've always wondered why no one makes a software HDCD converter i.e. after you rip an HDCD disc it would trans-code the data and create a 20-bit (or 24-bit) file. That way you would not have to worry about the LSB being lost in the playback chain. In my case, since I use digital room correction and have tons of HDCD discs, it would be really helpful!

The HDCD indicator light on my Berkeley Alpha lights on 44.1 HDCD recordings from Reference Recordings, but it does not light when playing back the Hrx Sampler which came with the Alpha. I am playing the original Wav files off the DVD. The sound of the Hrx Sampler is fantastic, but no HDCD indicator light.

Does this mean my system is not bit perfect on Hrx? Also does anyone know if the 96/24 and 88/24 Reference recordings use HDCD

Mac Mini (March 2010 model), firewire to Weiss AFI1, AES/EBU to Berkeley alpha. Dartzeel pre and main amp, Wilson Maxx2
TNT HRX

No, it means the high res HRx recordings don't use HDCD encoding; his was a process developed for CD's.

Hi Jeffrey - All RR tracks use High Definition Compatable Digital (HDCD). No matter what the sample rate the HDCD indicator should be illuminating on your Alpha DAC.

Are you closing iTunes, changing the sample rate in Audio Midi, them reopening iTunes? If not I can see why only the 44.1 tracks are showing as HDCD. Have you tried Amarra or Pure Music with iTunes? Either one will take care of the sample rate changes for you automatically.

Hi Chris,

Thanks for the reply.

I tried shutting down itunes, changing the sample rate to 176/24 in Audio Midi, then relaunching itunes. Still no HDCD indicator light on the Alpha when playing Hrx wav files from the Hrx Sampler.

As an aside, changing the sample rate back to 44.1 in Audio Midi while itunes was still running. still resulted in an HDCD light on, when playing 44.1 HDCD tracks, contrary to some of experiences reported.

This weekend I'll try Pure Music to see if that helps.

Berkeley Alpha DAC: How can I tell if the the stereo audio ('48' kHz indicated on the display) digital stream is being decoded from DVDs @ 48kHz = only 16 bit word length, or full 24 bits? I can't tell if SPDIF stereo audio is 16 or 24 bits.

I understand that typical movie DVDs output PCM stereo audio at 48 kHz (my Berkeley Alpha DAC display indicates '48 kHz' for DVDs, and perhaps most if not all DVD players truncate the digital word lengths to only 16 bits ? - I'm not sure though).

How can I determine if the Berkely DAC is receiving 16 bits, or the full 24 bits from the digital ouput of the DVD player? - SPDIF, or Toslink)

I have a Hyperion movie DVD: Angela Hewitt's "Bach Performance on the Piano" -- with her instructional lecture on how the piano student should play Bach. I connected SPDIF cable from a Pioneer DVD players digital output connector. As I recall, The Berkely DAC indicated 48 kHz when either PCM stereo, Dolby 5.1, or DTS-audio was selected in the DVD player's menu. Can anyone tell me if all DVD players, without exception, reduce the digital word length to only 16 bits for 48 kHz digital sample rate?

Digital 24 bit encoding is an official format option for Dolby and DTS, but it seems that only 16 bits (truncated) digital audio word lengths is all we get from DVD players via SPDIF out?
The corporations that produce DVDs mandate reduced 16 bit digital for DVD player's SPDIF stereo audio output?

The usual way to get 24 bit stereo audio is high-res computer files?

I also have a Lumagen-Radiance-xd video processor that has HDMI inputs, and outputs SPDIF stereo digital 48 kHz audio (from the HDMI). Maybe can get full 24 bits audio @ 48kHz from DVDs this way? I tried going this HDMI route, and the Berkeley Alpha DAC indicated 48 kHz -- but still couldn't determine if the audio decoding was 16 or 24 bits.

I have not yet found definitive info on the internet about SPDIF PCM 16 bit Vs. 24 bit audio from DVD players.

Bill

jeroen020,

Sorry that I didn't reply sooner.

Comparison was to my modified Wadia 2000, x64 Digimaster algorithm.

I don't recall the mod details.

Perhaps many will disagree: I think that the newer models aren't 'all that much better' than my modified Wadia -- just an opinion, of course.
However, for some older 1980's era CDs the Wadia can sound better with it's time-domain processing (may help with old CD 'digititus') with the inherent high-frequency rolloff -- again, just my opinion (I often like the treble rolloff) .
Also, some people actually prefer typical Wadia midrange presentation (so I understand).

Bill

Hi Bill,

With the Berkeley Alpha DAC, it is easy to check if the SPDIF is at 16 or 24 bits. Simply press the Mode button till the Filter light is lighted. The display will either show "1.16" or "1.24"

Geoff

Geoff,

Thanks for the info about how to display SPDIF 16 or 24 bits (using mode button).

I will see if can get 24 bits audio from a DVD player, or via HDMI to Radiance xd video processor & connecting to the processor's SPDIF audio out.

Appreciate your help.

Bill

Sharing some experiences with trying to get bit perfect output from my Mac, which readers may find useful

1. Itunes 9.1/ OSX 10.6.3 outputs bit perfect at 44.1/16, but NOT higher sample rates, unless additional software (such as Pure Music or Amarra is used). I have verified this doing all the prescribed steps: adjust Audio Midi sample rate and bit depth, then relaunch itunes, set volume at maximum. Verification of bit perfection was via playback of HDCD/HRx files, and using the HDCD indicator on the Berkeley Alpha.

2. With Pure Music installed, the HDCD indicator lights up on HRx 176/24 and 96/24 Reference Recording files.

Mac Mini Firewire to Wiess AFI1, Berkeley Alpha, Dartzeel Pre/Main, Wilson Maxx2, TNT HRX, Krell 505

Hi Jeffrey - Don't jump to conclusions just yet :~)

Amarra and Pure Music do nothing more than automatically do what your Mac is already capable of doing with manual adjustments. Please don't take offense but I've been doing this for years and it has always worked this way.

I suspect you are closing the iTunes window by clicking the X in the upper left corner? This only hides the iTunes window. Please go to the iTunes menu at the top and select Quit iTunes.

Let me know what you find :~)

Hi Chris,

No offence taken at all. Appreciate the dialogue.

Not only did I try quitting iTunes from the menu by selecting Quit iTunes, I also tried the following:

1. Cold boot the Mac
2. Select Audio Midi Setup and configure the Weiss AFI1 to 96/24
3. Launch iTunes, and play RR's Exotic Dances from the Opera download which is in 96/24.

No HDCD light on the Berkeley Alpha.

I repeated 1-3, but this time for step 2, chose to output to the built in optical output, selecting 96/24 again. Got the same no HDCD result.

When I launch Pure Music, I do get the HDCD light on this recording.

Come to think of it, probably something to do with the bit depth rather than the sampling frequency.

Chris, do you have a similar Mac-Berkeley setup and RR samples, and are you able to get the HDCD light on without Amarra/Pure Music? If so, then clearly my conclusions are wrong.

Hi Chris,

No offence taken at all. Appreciate the dialogue.

Not only did I try quitting iTunes from the menu by selecting Quit iTunes, I also tried the following:

1. Cold boot the Mac
2. Select Audio Midi Setup and configure the Weiss AFI1 to 96/24
3. Launch iTunes, and play RR's Exotic Dances from the Opera download which is in 96/24.

No HDCD light on the Berkeley Alpha.

I repeated 1-3, but this time for step 2, chose to output to the built in optical output, selecting 96/24 again. Got the same no HDCD result.

When I launch Pure Music, I do get the HDCD light on this recording.

Come to think of it, probably something to do with the bit depth rather than the sampling frequency.

Chris, do you have a similar Mac-Berkeley setup and RR samples, and are you able to get the HDCD light on without Amarra/Pure Music? If so, then clearly my conclusions are wrong.

Hi Jeffrey - I have a few Macs that I connect to my Berkeley DAC. All are capable of bit perfect output without Amarra or Pure Music. I had the Weiss Minerva here and I used it to convert FireWire to AES for my Alpha as a test of bit transparency. No problems.

The other thing I was thinking of this afternoon is this. When you adjust the sample rate in Audio Midi do you make sure to select the correct audio output device on the left side?

When you change the sample rates in Audio Midi manually does your Alpha DAC show the new sample rate immediately? Have you changed the filter on the Alpha? Have you tried rebooting your Alpha?

Just so we are on the same page. You still get the HDCD indicator at 16/44.1 correct?

Hi Chris,

Yes, the correct audio output device is showing on the left side.
Yes, when I change the sample rate in AMS, the Alpha display registers a change in sample rate display immediately.
No, I did not change the filter. The Alpha is correctly choosing 1.16 and 1.24 as appropriate.
Bit-correct HDCD at 16/44.1 is no problem, I get that without PM, and I get that whether I use the built in optical out or the AFI1.

I rebooted the alpha and the AFI1. No change in behaviour.

Have you tested bit correctness at 24/96 or 24/176 in Mac/iTunes without Amarra, PM or other software?

Chris,
when you change the volume in Amarra or at the Alpha DAC is there a loss of resolution? I am wondering if the Alpha DAC is doing this after the D/A stage because the HDCD light stays on. And what happens if you do this with the volume control of Amarra? Is there a loss of bit depth?

So what would be the optimal setting?

Thank you for clarification.

Stephan

In response to VandyMan's question. I found a small program a while ago, that converts ripped HDCDs to 24 bit files. Works like a charm.
http://forum.doom9.org/showthread.php?t=129136
Rudi

Folks,

After roaming this site for quite some time and getting up to speed with CA (NAS, MacMini, feeding my Wadia850i via toslink) I spotted a used BADA around the corner last week. That’s a rarity, since I live in the Netherlands and BADA’s don’t come by here often. Especially 6month old used ones with 1 year of warranty by a trusted dealer (of some other respected brands, no Berkeley dealerships in the EU to my knowledge) for 300 euro over a new QB-9. Being a used kit buff, that caught my interest...

Listening session at dealer happened using redbook only, comparing the BADA to my Wadia850i (CD playing on Wadia versus CD in Wadia feeding BADA via SPDIF). Major difference was size of the sonic picture (I guess that is ‘soundstage’). BADA is much larger, more air, and most importantly, less ‘sticky on the speakers’. The highs on the BADA had a bit more sparkle (Wadia digimaster roll-off?) and in general there was more detail and texture to everything, including the bass. The Wadia however had more bass-pressure and a little more ‘bass induced drama’ because of that. I can understand that there are people out there that prefer the Wadia house sound because of that.. I liked the BADA a lot (over the Wadia) on redbook.

So I took the plunge and the BADA is now sitting in my rack! I also upgraded the interface from the build-in toslink to HiFace (BNC) and installed Pure Music and I was playing HRx in 30 minutes.. wouwza.. And that auto sample-rate switching is nice!

So now, need to get to grips with the new (&improved) sound and do the tons of tweaking & setup. Initial thoughts & questions for this forums BADA users:

1. The hum issue
My BADA is NOT silent (far from it). In fact I think it is noisy. This thread so far mentions two hypothesis:
(a) It is an issue on earlier models and was fixed in later versions or
(b) It is a DC-offset issue. My Nemo’s do have a very minor hum also, whereas their previous owner stated they were dead-silent at his place..so I hope this is the issue.

Does any of you folks have some final insight in this? Suggestions?

2. HDCD light on HRx recordings.
Chris seems to be very sure that HDCD light must be lit, also on HRx recordings. It did not happen here playing HRx using PureMusic. Samplerate displayed ok. I used the demo tracks recently (06/24) posted here on CA by Chris. I downloaded in AIFF to circumvent potential MAX conversion issues. HDCD light on 44.1 is not a problem. Does that imply a bit perfect mess-up on HRx playback someplace? Word-lenght perhaps? The HiFace seems to know only 32bit...

3. Rack mounting options: coupling vs. de-coupling of a BADA.
This thing weighs just about nothing! I am trying to figure out proper placement & mounting, currently using 3 magnetic floaters under, with 4 doorstops (4 x 1.1kg) op top of the BADA for added weight. Experiences please!

Gosh, such a nice hobby this is. Hate to think that I was done....

Hans