Benchmark DAC1 now available with USB
May 28, 2007 at 5:08 PM Post #601 of 3,058
Quote:

Originally Posted by tonygeno /img/forum/go_quote.gif
I have hooked up the Dac1 USB to a Mac Mini via USB. The device shows up in the Sound System Preference. When listening to iTunes all is well and I can listen to the music through my stereo but when I listen to music over Safari (Rhapsody, for instance), the sound defaults to the computer speakers, even though I have chosen the Benchmark as my sound device. If I switch to the optical connection on the Benchmark, the music then plays on my stereo. Any ideas as to why?


Beside the settings in the System Preferences Audio pane, check your Audio Midi Setup program as well and confirm that audio is going out to the DAC1. I had a problem with web browsers going to neither internal nor the DAC1 and Audio Midi Setup fixed this issue.
 
May 28, 2007 at 5:52 PM Post #602 of 3,058
Quote:

Originally Posted by lowmagnet /img/forum/go_quote.gif
Beside the settings in the System Preferences Audio pane, check your Audio Midi Setup program as well and confirm that audio is going out to the DAC1. I had a problem with web browsers going to neither internal nor the DAC1 and Audio Midi Setup fixed this issue.


I'll give it a go. Thanks.
 
May 29, 2007 at 2:21 AM Post #604 of 3,058
Great thread so far. I think I've seen almost every aspect of the DAC1's design examined, but I don't think I've seen a discussion as to the choice of a volume pot. I'm not for or against, but just wondering as to the choice of what (I assume) is a single volume pot rather than a resistor ladder or some form of digitally controlled analog volume control.

I don't know enough to question one method over another, other than the already mentioned mechanical limitations at the extremes of the volume pot, but was simply wondering if this design choice was made for cost reasons, or if the volume attenuation functions of the DAC1 are on par quality wise with its DAC functions.
 
May 29, 2007 at 1:23 PM Post #606 of 3,058
Tonygeno,

What os level are you on with your mac mini? There was a change in the usb subsystem that was fixed in 10.4.9. This was a kinda goofy thing in that it did not effect most audio usb controllers but did with the TAS1020 that is also used in the DAC1.

You may simply have to upgrade the os. You can also look at the USB chain in the system profiler and see what the DAC1 is coming in as. Sometimes if there is a problem the info here will be garbled.

Thanks
Gordon
 
May 29, 2007 at 6:58 PM Post #607 of 3,058
I figure the reason this thread has run 31 pages--so far--is that Elias takes no offense and keeps answering questions. In the years I've been involved with audio I've never known of of a company so willing to engage in the conversation.

I still use my DAC1s every day and continue to be delighted with them. It's an excellent product. Simple, direct, and sounds great.

I still wonder sometimes, though... if Benchmark wanted to make an ultimate, cost-no-object DAC, what would they change? How much of a difference would the changes make? I have no intention of going into the pursuit of modifications, as the sound quality of the stock unit is stellar, but in the design of any real-world product compromises must be made. I wonder what compromises were made with the DAC1... and how much midnight oil was burned in the discussion of those compromises.
smily_headphones1.gif
 
May 29, 2007 at 7:09 PM Post #608 of 3,058
yeah, i would like to really thank mr. gwinn. i know no other large company that has ever talked to their customers like this.

i'd like everyone to check in that uses their dac1 as their headphone amp. or at least one of their headphone amps. how do you think it compares to your others? i don't mean saying this one or that one is better. i mean how does the "sound" compare. i don't like when people say something is just better. the dac1's amp is not insulted by anything. either you like hearing the truth or you don't.

6moons compared the amp to the cia vhp-1 and found the dac1 "better". that is not so important though. it is what he said about the dac1's amp. exactly what i have been saying. i know of other amp that is as true to the source. that doesnt mean a more natural/neutral amp exists, i just don't know of it
wink.gif


music_man
 
May 29, 2007 at 7:21 PM Post #609 of 3,058
Quote:

Originally Posted by puntloos /img/forum/go_quote.gif

Hah, yeah I guess my brain just connected the dots the wrong way huh, basically both you and Bel Canto are 'extremely proud of the anti jitter' and both were called 'ultra....ock' .. Additionally, I seem to remember some review or some forum post that said the two used the same technology.



I'm not sure how their anti-jitter technology works, but, as I said before, there is no plug-and-play chip that creates jitter immunity. We use an ASRC, as do many other manufacturers. However, the entire anti-jitter technology in the DAC1 (UltraLock) is very much propriety and individual to the DAC1. This is why we named it...because it is a very specific topology. I'm not sure why Bel Canto chose a name so similar to ours (BC's UltraClock vs. Benchmark's UltraLock). It reminds me of the Walkmans I used to see on the streets in NYC with the brand-name Coby, written in a font and style suspiciously similar to Sony.

Quote:

Originally Posted by puntloos /img/forum/go_quote.gif
Well, admittedly this was my conclusion after reading what basically amounts to the 'marketing version' of the specs. Bel Canto claim complete galvanic isolation of the inputs and shielded input transformers. While I think these are 'good things' in general, maybe these are also totally trivial and you simply chose to not mention this in your specs since 'who wouldn't do it that way!'.


An understandable interpretation.,.. Galvanic Isolation is just another way of saying transformer isolated. Both the BNC (coax) and XLR inputs on the DAC1 are transformer (galvanic) isolated.

Quote:

Originally Posted by puntloos /img/forum/go_quote.gif
Done.. thank you for your explaination.. I am afraid I am out of my depth though. While my 'cowpoopie detector' is not ringing with you, I really cant gauge how your design choices would rank against the choices the belcanto, aqvox or lavry engineers made. (aqvox and belcanto sample at 192khz with the Burr Brown 1796 and 1792 DACs respectively.). It would be interesting to hear you guys discuss this subject matter, although it'd probably be like an ant watching giants fight and trying not to get stepped on.


We test most chips to determine if they are something we should use. Just like with any engineering, their are trade-offs (each has pros and cons, none out-shine in all respects). We have tested the 1792 (the 1796 is just a cheaper version of the 1792 with similar topology but less performance). It had advantages and disadvantages vs. the AD1853.

The difference between the BB 1796 vs. the DAC1's AD1853 are: the former has better filtering and signal-to-noise ratio. But it has serious linearity issues. Linearity means 1 dB increase in digital amplitude results in 1 dB increase in analog amplitude (input vs. output). The Burr-Brown chips have serious problems with linearity. Also, intermodulation distortion is an more of an issue with these chips. They are also very sensitive to temperature fluctuations, so that these issues are amplified as the temperature fluctuates.

In other words, take your choice: more distortion and less noise (1796) or more noise and less distortion (1853). Since the S-to-N of the 1853 is -117 dB, we chose this over increased distortion.

Quote:

Originally Posted by puntloos /img/forum/go_quote.gif
I do have one additional question though. I am trying to arrange to get a Bel Canto, an Aqvox and a benchmark in one room. Do you have any suggestions on how, or what to test? For example is there some way you would reccommend someone with limited pro resources to create a bad jitter situation to test a dac's resilience?


Run 500 feet of cable from the source to the DAC. Other then that, there is no way to definitely increase jitter. However, you could start with a cheap transport.

Thanks,
Elias
 
May 29, 2007 at 7:33 PM Post #610 of 3,058
Quote:

Originally Posted by tonygeno /img/forum/go_quote.gif
I have hooked up the Dac1 USB to a Mac Mini via USB. The device shows up in the Sound System Preference. When listening to iTunes all is well and I can listen to the music through my stereo but when I listen to music over Safari (Rhapsody, for instance), the sound defaults to the computer speakers, even though I have chosen the Benchmark as my sound device. If I switch to the optical connection on the Benchmark, the music then plays on my stereo. Any ideas as to why?


I have been in contact (phone and email) with Tony since he posted this I have solved this problem. I will post the result for anyone else who encounters this.

I don't know why, but Rhapsody simply refused to 'talk' to the DAC1 USB via the USB input. So, I tried other web-browser-based audio streams such as MySpace, and it was no problem. But not with Rhapsody. This was the case with both Firefox and Safari. This was also the case with other 3rd party USB devices.

After messin' around for a while, I finally got it to work, and here's how: I changed the sample-rate of the Benchmark in the Mac Audio MIDI Setup. I changed it to 44.1 kHz (48 kHz worked also). After changing the sample-rate, I changed the "Default Ouput" to "Built-In Output". Then I immediately changed it back to "Benchmark 1.0". It worked.

I don't know how Rhapsody works exactly (technically speaking), so I can't comment as to why it happened. But at least we know how to work around it.

I will also add the side note which I offered to Tony, and is very important for anyone who uses Mac for audio:

The sample-rate setting in Audio MIDI Setup should always be set to the sample-rate of the music file being played. In the case of Rhapsody, 44.1 kHz is most likely the original sample rate. This applies to iTunes and everything else, also. If you are playing 44.1 kHz audio, the sample-rate in Audio MIDI should be set as 44.1 kHz. If you're listening to 98 kHz audio, then you should set it to 98 kHz. The reason for this is that the Mac will convert the sample-rate from the original rate to the rate selected in Audio MIDI Setup (if the two rates are different). The sample-rate conversion is very poorly executed by the Mac (poor programming), and significant distortion will result. It is very important, from a fidelity standpoint, to always be sure your sample-rates are the same.

Thanks,
Elias
 
May 29, 2007 at 7:48 PM Post #611 of 3,058
Quote:

Originally Posted by Audio_newb /img/forum/go_quote.gif
Great thread so far. I think I've seen almost every aspect of the DAC1's design examined, but I don't think I've seen a discussion as to the choice of a volume pot. I'm not for or against, but just wondering as to the choice of what (I assume) is a single volume pot rather than a resistor ladder or some form of digitally controlled analog volume control.

I don't know enough to question one method over another, other than the already mentioned mechanical limitations at the extremes of the volume pot, but was simply wondering if this design choice was made for cost reasons, or if the volume attenuation functions of the DAC1 are on par quality wise with its DAC functions.



This is a great question. This is another example of engineering design trade-off. ie, each solution gives a little here, but takes away a little (or a lot!) there.

Specifically, here are the common choices (not including digital volume control, which is pre-D-to-A):

DIGITALLY CONTROLLED ANALOG VOLUME CONTROL:

These are the poorest performing from a distortion point of view. These are simply not satisfactory with high-quality audio.

POTENTIOMETER:

This is good for the following: low noise, low distortion, low cost, tight volume resolution (ie, easy to achieve small changes in volume). Pots are bad for the following reason: the first 15-20% of rotation is not evenly matched from left to right. After this initial inaccuracy, however, it is VERY accurate. Very comparable to any other (analog) solution.

RESISTOR LADDER:

This is good for the following: low distortion, high channel accuracy, low noise (at rest). The last point indicates that it is noisy when switching from one volume setting to another (switching noise), but quiet once at rest. The resistor ladder is bad for the following reasons: expensive, low volume resolution. This last point is because a switching resistor ladder has 24-positions, where the detented-pot of the DAC1 has 41-detents (and can be settled between detents). This means you can make smaller changes in volume with a pot then you can with a resistor ladder.

So, as you can see, there are trade-offs, and there are no right or wrong way. The reason we choose the potentiometer is because it works just as well as the resistor ladder after the first 20% of rotation, and the lower half of rotation should only be used to achieve full-off. So, as long as it is being operated in its upper 80%, it is as good as any other solution, while being significantly less expensive and higher volume resolution.

Thanks,
Elias
 
May 29, 2007 at 8:01 PM Post #612 of 3,058
Quote:

Originally Posted by Lord Chaos /img/forum/go_quote.gif
I figure the reason this thread has run 31 pages--so far--is that Elias takes no offense and keeps answering questions. In the years I've been involved with audio I've never known of of a company so willing to engage in the conversation.

I still use my DAC1s every day and continue to be delighted with them. It's an excellent product. Simple, direct, and sounds great.

I still wonder sometimes, though... if Benchmark wanted to make an ultimate, cost-no-object DAC, what would they change? How much of a difference would the changes make? I have no intention of going into the pursuit of modifications, as the sound quality of the stock unit is stellar, but in the design of any real-world product compromises must be made. I wonder what compromises were made with the DAC1... and how much midnight oil was burned in the discussion of those compromises.
smily_headphones1.gif



Well, I'd like to answer you, but I don't want to give any design secrets away
wink.gif
.

This question is hard to answer without sounding over-prideful, but I will say this, there are no parts in the DAC1 that can be improved with more expensive parts. The most expensive components of the DAC1 (and the only components which are chosen based on price) are the faceplate and chassis etc.; circuit components are not limited by cost. In other words, the sonic-performance of the DAC1 was not compromised based on price.

The only way the DAC1's sonic performance could be improved is with a more elaborate topology - a whole new design technique. This would not cost a whole lot more from a component point of view, but it would from a R&D point of view.

Thanks,
Elias
 
May 30, 2007 at 6:10 AM Post #613 of 3,058
The PCM1796 is not R2R, it's a hybrid multilevel sigma-delta DAC. The operation is the same as shown in Fig.30 in the PCM1794 datasheet (likewise for the PCM1792).

Galvanic isolation is not necessarily transformer based; it can be optical as well
smily_headphones1.gif
 
May 30, 2007 at 8:44 PM Post #615 of 3,058
Quote:

Originally Posted by Elias
...I will say this, there are no parts in the DAC1 that can be improved with more expensive parts.


Very interesting... and pretty much what I thought, given the way the DAC1 sounds. I know you caution people about the imbalance at low volume settings, but you must be using a really good pot in there because I've never noticed balance problems at the low settings required for my sensitive headphones and sensitive ears. Thank you for the response.
 

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