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

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.

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 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

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.

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.

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

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.

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

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

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 :=)

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?

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

"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

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).

Very cool Bob. Thanks for the follow-up.

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.

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.

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

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

enjoy,
clay

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

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 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.

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."

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

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.

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?

That's really great to hear Steve!

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.

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

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).

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 :~)

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.

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...