Getting the best out of Usb Benchmark

[theob]

Chris I know you have tried spdif vs usb and preferred usb on your Benchmark. I am on a mission to get real 24/96 and 24/192 out of my Benchmark. I use a dell xp sp2 and run cplay with asio4all (which is limited to 24/48). I use asio4all because it is the only asio I can find on the webb . Now I think that 24/192 capability asio is only available on soundcard packages. Reason I need an asio interface is that cplay requires it (cplay is by far the best playback software I have heard, I tried wmp11, foobar and jriver and cplay beats them handily in my system). So do you know of anywhere I can d/l a sample of asio w/o buying a soundcard? What's a good sound card to get? Many of those available are 7.1 with all kinds of other stuff on them I don't need. I have been recommended to try asus and juli.

 

I know you are running a mac but can you say a little more about the program you use? Are you getting true 24/192 off the Benchmark etc etc? Appreciate your or anybody else's feedback on this.

 

[poop]

Alot of questions here...

 

First, you cannot achieve 24/192 output from the Benchmark via USB. The max the USB can handle is 24/96. You will need to use one of the other digital connections in order to get 24/192.

 

You seem a bit confused about ASIO. There is much debate as to whether or not it is even required, but if you decide to use it, then asio4all is fine.

 

In regards to you comments about cplay - that all seems a bit odd to me. What do you mean by 'cplay is by far the best playback software I have heard'? The software really shouldn't have any impact on the sound at all if configured correctly. Are you using some sort of 'audio enhancements' to make things sound 'better'? You also mention it is limited to 24/48... so I'm not sure why you are using it at all in that case.

 

Refer to the Benchmark wiki (google it) it should answer all of your questions. If you have more after reading it, post them here and we can help out further.

 

 

 

[Innertuber]

Here's a couple cards to give you something to peek at and study, I'm guessing you want stereo?

 

http://www.m-audio.com/products/en_us/Audiophile192-main.html

http://www.lynxstudio.com/product_detail.asp?i=11

 

There are others too, if I could find the right old posts somewhere. As I recall the cards are kind of lumped into $100-200, then $300-450, then a few that are $1000+. That was late last year.

 

I'm not that up on Dells, but can you bypass the soundcard with ASIO and just USB (or whatever) out? I'd think most soundcards would be inferior to your benchmark as well as unrequired? I'll probably learn why that makes little sense.

 

I thought ASIO was a way of handling the data rather than a data file itself? Wouldn't you want to download just say a 24/192 file to test? Don't worry, I ask lots of dumb questions!

 

[theob]

I know that the Benchmark only does 24/96 through usb. I also know that asio4all (which is limited to either 24/48 or 24/96) is free and is required with cplay. Cplay is like wmp11 or foobar. It plays audio files. What is so great about it, among other things, is that it provides ram playback so its like playing files off a flash drive or a sshd -- no moving parts. There are tons of posts on AA pc audio about cplay. Bottom line is that for me it sounds better than jriver (my prior reference) by a long shot. Asio and asio4all are software/hardware interfaces (asfaik) and it provides a clearer path than direct sound in Windows xp. Since I would like to use cplay and my Benchmark at 24/192 I need to use a toslink or spdif input but I also need asio support for 24/192. Asio4all (free) only provides either 24/96 or 24/48 so I need asio capability better than asio4all. I thought the only place I could get that and interface with my Benchmark was a pci based sound card like juli or asus or emu. Some of these provide 24/192 out via spdif and have asio drivers that support 24/192. I would like to listen to hirez stuff from RR without downsampling.

 

Make more sense on what my dilemma is?

 

[all300b]

sampled everything up/down to 110 khz internally. Would it not sample 24/192 material down to go through its converter?

 

 

 

 

 

[audiozorro]

The Benchmark DAC1 USB has 24/192 capability through all of its inputs/outputs except for USB, which is limited to 24/96. It resamples everything internally to 110 kHz for jitter control. Benchmark Media has been highly successful in their implementation and if not the best, is clearly the universally accepted DAC for judging all other DACs. The following is from Elias Gwinn at Benchmark that explains the technical aspects more clearly:

 

"The process of upsampling does not inherently improve sound quality during D/A conversion. However, Benchmark converters re-sample for a very specific reason: jitter immunity. Benchmark converters use a proprietary clocking system (we refer to it as UltraLock). It works like this... The incoming digital signal is immediately re-sampled by an ASRC (asyncronous sample rate converter). The ASRC, as the name implies, is not syncronized to the clock of the incoming digital signal. Therefore, its performance is independant of the quality of that clock. In other words, it doesn't matter if the signal came from a cheap transport with cheap cables, or from a $10,000 signal chain. The large amount of jitter caused by the cheap transport and cheap cable will be moot with respect to the ASRC process. The output of the ASRC is then clocked to an on-board clock with extremely low jitter and strategic sheilding and board traces. The output of the ASRC can be configured to any sample rate that we choose, including the original sample rate. However, we dictated the re-sample rate as 110 kHz because it is the highest sample-rate at which the digital interpolation filter of the D/A chip will operate optimally. The ill-effects of the digital interpolation filter at higher-then-110 kHz include pass-band ripple (non-linearities in frequency response) and inferior attenuation of stop-band frequencies (which results in aliasing). Therefore, the D/A performance is optimized by maintaing 110 kHz. Many converter designers have since employed similar topologies, but use lower re-sampling frequencies, such as 96 kHz. By resampling to 110 kHz, the low-pass filter of the ASRC and D/A are moved as far up as possible as to not infringe on the analog bandwidth of the audio."

 

"On the question of: Why does the DAC1 re-sample to 110 kHz? Here is why: it is the highest frequency to maintain the full oversampling of the D-A chip. EVERY D-to-A chip on the market cuts the oversampling rate in half to accommodate 192 kHz. This will also implement a different type of digital low-pass filtering which is inferior to the filter used at and below 110kHz. This is also why most recording engineers don't use 192 kHz. The higher bandwidth seems appealing, but the stat-of-the-technology is such that 192 kHz conversion is actually inferior to 96 kHz. Also, the DAC1's oversampling ASRC and resulting 110 kHz sample rate reproduces 96 kHz signals much more faithfully then a D-A converting the original 96 kHz signal. This is because the Nyquist frequency is on the slope of the filter (attenuated, but not completely). This is undesirable for two reasons. The first reason is the Nyquist frequency is not faithfully converted to analog (ie, the analog bandwidth of 96 kHz conversion is actually less then 48 kHz). With the DAC1, the full bandwidth of a 96 kHz signal can be faithfully reproduced. The second problem with 96 kHz conversion is the frequencies at and above Nyquist (48 kHz and up) are not completely attenuated, so some aliasing and imaging will occur. With the 110 kHz upsampling and conversion in the DAC1, the frequencies below 55 kHz are not in danger of being aliased."

 

 

[theob]

The limiting factor is asio4all which won't pass 24/192 (neither will the usb). What I am trying to do is get 24/192 to the Benchmark through asio via a soundcard. I choose to use asio because it isrequired with cplay which is a superb player. So I was looking for some soundcard recommendations. What I was also asking is how does the mac do it?

 

[The Computer Audiophile]

Hi theob - The Mac can pass 24/192 to the Benchmark only through a converter or an internal card for a Mac Pro. Right now I can pass 24/192 to the Benchmark from my MacBook Pro laptop through the Weiss Minerva which has a coax output. This is kind of a ridiculous option because nobody with the Minerva would output to the DAC1. I haven't used the Weiss Vesta, but this should pass the digital stream to a DAC1 for half the price of the Minerva.

 

For internal sound cards and connectivity to the DAC1 PRE the limiting factor will be lack of an AES input. On the DAC1 USB there is an AES input which will come in handy. On a PC or Mac you can use the Lynx AES16 or AES16e to pass a bit perfect stream up to 24/192 via AES. This method is rock solid, but is pretty expensive.

 

I don't know for sure about the M-Audio Audiophile 192 or Audiophile 2496, but I believe they are very nice cards that can pass a bit perfect 24/96 or 24/192 stream. They are also much cheaper I believe.

 

is this the kind of info you are looking for theob? let me know and we can discuss this one until you get the answers you're looking for.

 

[poop]

I think we need to discuss your question more, because it is probably an issue for a few of us without our realising it.

 

I didn't realise asio4all was limited in any way, and assumed it would work with 24/192. There aren't many alternative solutions for ASIO. I use an EMU 1212M which has Patchmix software, and you can alter ASIO settings within it. To be honest, I'm not sure if it will output 24/192 because I have not been using it. I was under the impression that the Benchmark could output 24/192 via USB, and only recently found out that is not the case. I haven't tried listening to any 24/192 material yet, so it has not been an issue, but I plan to very soon so have to figure it out. The prospect of fighting to understand the Patchmix software again is not a pleasant one! If I can get it working I'll let you know.

 

In terms of a sound card to use with the Benchmark, a cheap one is fine because you simply need a source which can feed the DAC bitperfect information. Perhaps look at the EMU range as well as others that have been suggested - from your last post you seem to have a good idea of the better brands. My only gripe with the EMU is Patchmix - not the most intuitive software around, but it's fine once you figure it out... There are a number of well priced options in the for sale forum of Head-fi.org so it might be worth a look before you purchase anything.

 

My (limited) understanding of the Mac system is that it does it all natively. The internal soundcards and system are built to be capable of outputting higher resolution audio bitperfedtly.

 

I hope this response is a little more helpful, or at least on track in terms of what you are asking.

 

 

EDIT: So Chris from your post above do I understand correctly that Macs also require additional gear to output 24/192, and that the highest they output natively is 24/96?

 

 

[The Computer Audiophile]

Hey Poo - This is one of the most frustrating things with the Mac. The specs all say the optical output port is capable of 24/192 output. I discussed this with apple and they also say it is possible. But, I don't know any anyone who has been able to even select anything higher than 24/96 in the Audio Midi settings for the built-in optical port. FireWire certainly gives you bit perfect 24/192 out of a Mac and I am using it as I write this post. USB is the same for Mac and Windows and is mainly limited by the USB DACs connected to the port. Nobody is doing above 24/96 via USB so we are at the liberty of the DAC manufacturers.

 

[theob]

The Benchmark usb is a good dac at 24/96 but presents a thorny issue to get and process true 24/192. Maybe its not a big deal because, as you mentioned, it will d/s back to 110 khz anyway. But as an audiophile I am driven to hear it.

 

 

Thanks for your perseverence in this.

 

[EliasGwinn]

Hello all,

 

If there are any questions about the DAC1 / USB / PRE that you'd like me to answer, please don't hesitate to ask. I'll check back here over the next few weeks to see if there is any questions.

 

Thanks,

Elias

 

[The Computer Audiophile]

Hey Elias - There was a comment a while back about the DAC1 downsampling everything to 110kHz. Can you explain a little about this if it's true? If readers want to playback the new HRx files at 24/176.4 is there going to be any sonic effects etc...

 

Also, are you aware of many people using 24/192 output from computers into the DAC1? Without FireWire this is a little harder.

 

[audiozorro]

Since you offered - what's the possibility of adding firewire to a future version of the DAC1? The Apogee MiniDAC, which is a close competitor, offers optional configurations with either USB or FW, so I assume it is not impossible. Given that I read that improvements in the DAC1 PRE could not be backfit to earlier versions of the DAC1, I am not expecting any backfit mods or upgrades for FW, though you may have a lot of takers. Obviously the 24/192 problem rests with Apple, but I assume they would say they offer an excellent 24/192 solution through FW.

 

For me, it makes no sense to use a FW MiniDAC connected to my Mac to feed the DAC1 through either the toslink or coaxial digital connections in order to playback my HRx tracks at 24/176.4 or other files at 24/192. So what configurations do you recommend for maximizing the playback capabilities of the DAC1, i.e. 24/192?

 

[EliasGwinn]

It is true that the DAC1 will resample everything to 110 kHz. There is a good reason for doing this, as I mentioned in the quote posted earlier on this thread. The overall performance of the system is optimized this way.

 

Lets put it this way, there are costs and payoffs associated each different clock-management approach. The DAC's that don't reclock are extremely susceptible to jitter-distortion. The DAC's that do reclock above 110 kHz suffer from filter non-linearities (distortion in the freq response) and aliasing (lack of stop-band attenuation).

 

The approach taken by the DAC1 has two costs: THD+N from the resampling process (< -133 dB, well below the threshold of hearing) and analog bandwidth limited at 55 kHz (due to the 110 kHz re-sampling rate).

 

So, as you can see, the DAC1 has very neglible distortion, including jitter, and up to 55 kHz analog bandwidth. We feel this easily outweighs the non-musical distortion that results from jitter and aliasing, both of which are easily avoided with the DAC1's clock management scheme.

 

We could have designed it to do anything. We could have built the DAC1 to convert at 192 kHz, just like the others. But we've earned a reputation for not sacrificing true performance for gains in unsubstatial popularity, and we will continue along that path.

 

To answer your question, the sonic effects of the the DAC1's clocking scheme on 176 and 192 material is simple: instead of an analog bandwidth of 88.2 kHz or 96 kHz (respectively), the analog bandwidth will be 55 kHz. But, more importantly, the ultra-sonic information will not be aliased into the audible band, and jitter will not be an issue whatsoever.

 

As for 3rd party devices for streaming hi-rez from the computer, we are looking into that at this time. Unfortunately, we haven't found a good solution yet. I will be sure to post here and everywhere if we find a satisfactory, affordable solution.

 

Thanks,

Elias

 

[The Computer Audiophile]

Thanks for the detailed response Elias!

 

[EliasGwinn]

At the risk of seeming aloof, I can only say that we have no plans for a DAC1 with Firewire. We have a policy to not talk about any future plans for products.

 

I understand your position, though. We are looking for a good computer interface to stream 192 kHz into the DAC1. Hopefully we'll have a solution soon.

 

Thanks,

Elias

 

[audiozorro]

Thank you for your honesty. I still feel that Mac users have a distinct advantage over Windows users when it comes to computer audio, though it would be nice if Apple could get its act together to give us 24/192 capability through the Mac built in toslink or if we could get a complete solution from Benchmark without having the uncertainties and significant costs of third party solutions.

 

One more question:

 

I never got into the balanced versus unbalanced debate but is there any advantage in using the balanced XLR analog line outputs over the unbalanced RCA analog outputs in the DAC1? I think I have heard comments all over the place ranging from it makes no difference to only if the rest of your system is balanced to slightly better regardless of whether the rest of your system is balanced.

 

 

[EliasGwinn]

1. Balanced vs. Unbalanced

 

The advantage of balanced vs. unbalanced is a higher signal-to-noise ratio (SNR). Balanced provides twice as much signal (6 dB more) as well as a topology that allows the downstream device to remove common-mode noise.

 

Common-mode noise is any noise that is induced on the signal from electro-magnetic interference (EMI) and ground loops.

 

In environments with very low EMI, short cable runs, and proper grounding, unbalanced may be very comparable to balanced.

 

However, a truly balanced connection is the safest, most bullet-proof connection.

 

A true-balanced connection is not a system-wide connection, but merely the type of connection between two pieces of gear (e.g., DAC1 PRE to amplifier).

 

2. Sample-rate of digital stream to DAC1

 

The USB connection of the DAC1 allows the host to declare the sample rate of the incoming digital stream. In the case of DS and WMM, this is determined by kmixer. Providing there are no conflicts, kmixer will allow the original sample rate of the source proceed unaffected to the DAC1.

 

With other API's, the controlling software will dictate the sample rate. For example, an ASIO driver can dictate the sample rate, even if it means converting from the original sample rate.

 

Thanks,

Elias

 

[theob]

This combo of mine is as good if not better than anything I have heard in my 30+ years as an audiophile. Hi Elias nice to see you here. I have a question on asio4all. The control panel for asio4all says 24/96, it certainly passes a 24/96 signal and it sounds great. However some say that asio4all is only 24/48. Can you confirm that asio4all is either 24/96 or 24/48?

 

[theob]

Since posting last time I have purchased a ESI Juli card so that I can get higher than 24/96 to my Benchie. I use dell athlon dual 64 cplay --> juli--> asio 2.0 interface-->breakout cable with bnc connector--audio alchemy powered I2S cable with bnc connectors-->benchmark. I am currently running 24/176khz with -145 src in cplay. Cplay has also recently been updated to using the ssse features of either intel or athlon chips(look it up on google). I want tell you that this combo on my sys is just unbelievable. I never thought this type of clarity, high frequency extension, imaging and overall dynamics was possible out of any source let alone a pc. I'm reporting this because all this stuff is free (see cplay and cmp in Audio Asylum). I am not able to go -145 src and 192 rate because of limits on my current pc hardware but I have an old computer and have ordered a mobo, Intel 7200 chip and other hw to get me to 196. This cplay thing is the real deal but implementing is not for the faint of heart. Its a real bumpy road to get there. But if you have a Benchie you gotta try.

 

[appleteapot]

Hi Elias,

 

I am a Benchmark DAC 1 fan since 2005 since my first audition of DAC1 in Hong Kong showroom. I have been enjoying the excellent price/performance ratio for the past 3 years and DAC 1 (not USB DAC1) is in the heart of my whole playback setup. The DAC1 manual is a piece of great work from engineer and I think it sets a standard. The 5-year warranty proves DAC1's ruggedness. The device never overheats even under HK Tropical weather and I never need to turn it off. IMO It well deserves the "Class A" component recommendation from Stereophile magazine.

 

We have a local discussion group in Hong Kong on HTPC since 2005 and I have praised the performance of the DAC1 throughout the past 3 years. My current digital path is RME 9632 output AES to Benchmark DAC 1.

 

However, I am quite surprised to learn that the later model, USB DAC 1 comes with 24/96 USB input (whereas the USB DAC1 claims to be 24/192 which is, IMHO, needs clarification). And the latest member, Pre DAC1, has no AES input port. As a professional DAC manufacturer (BTW do you consider yourself professional DAC equipment manufacturer?), I would expect Benchmark in the same league with Apogee, Weiss, Mytek, and your products should at least have an AES input. I put my PC in study room, run a piece of 5 meter long AES cable to the DAC 1 in my living room. This is to isolate any HTPC fan noise. AES is good for long connection and the excellent physical locking mechanism prevents the connection from being accidentally unplugged.

 

cPlay, is by far the best sounding playback software I have used. We used cPlay to replace Foobar. cPlay's core competence, 24/192 upsampling, provides a 145dB dynamic range, which translates into excellent dynamics and musicality. With cplay/9632/DAC1 combination, I enjoy music more than ever. That's why I am concerned about your recent changes to use 24/96 USB and the removal of AES input.

 

I am a fan of Benchmark DAC1 because it is an good sounding DAC with careful engineering and the price performance ratio. If I just go for USB convenience, there are many much cheaper choices.

 

 

[The Computer Audiophile]

Hi appleteapot - Welcome to Computer Audiophile. I may be able to help clarify a little. The USB inputs on the DAC1s are capable of up to 24/06, while the other inputs are capable of up to 24/192. I believe this has been the same since the first USB DAC1s were released.

 

I too would like an AES input on the DAC1 PRE, but I think Benchmark had to cut back somewhere to allow the analog input on the DAC1 PRE. I would really like to connect my Lynx AES16e to the DAC1 PRE via AES, but it just is not going to happen.

 

[EliasGwinn]

I'm glad to hear you've enjoyed your DAC1 so much!

 

As Chris just mentioned, the USB input of the DAC1 USB / PRE is limited to 24/96, while the other inputs can still accept up to 24/192.

 

However, the coaxial inputs of the DAC1 PRE are AES compatible. If you are using an XLR/AES interface, you simply need a passive transformer like this one: http://store.haveinc.com/p-49466-canare-imp-transfrmr-xlrf-bnc-jack-10db.aspx

 

The DAC1 series is built to professional standards. However, its performance has led to a wide appreciation in the Audiophile world. The DAC1 PRE was designed as "the best of both worlds". At the core is the DAC1 professional conversion system, with the best digital and analog circuitry we know of. The inputs, however, are suited for both pro and home users, since the coaxial inputs are AES and SPDIF compatible.

 

Thanks,

Elias

 

[theob]

I was listening via usb to the benchmark for several months and loved it. Originally using j river then I migrated to cplay which I absolutely loved. Could only get 24/96 out of usb so I was compelled to try new approaches to get 24/192. Well since I loved cplay I built a new computer per cmp (see Audio Asylum) which wasn't as hard as I thought. It is basically a gigabyte mother board, intel 7200 dual processor and kingston ram (all per cics specs). Then I got a juli sound card which is capable of 24/192 digital output (just what I needed). All this plus I loaded the cmp software of cics (again see Audio Asylum). This was significant because cmp software gets rid of much of the windows overhead to make for clean 800+ meg cue file playback. I hit the motherload--absolutely beautiful: dynamic, layered, transparent and very undigital. But over the last few days I tried to think what else could I do to identify bottlenecks in my system elsewhere. I had been running a dbx driverack pa electronic crossover into my martin logan monolith III speakers for ultra clean triamping. But it always concerned me that the pa took analogue in and converted it to 24/48 to perform all the crossover functions and then reconverted back to analogue. When I had a sony scd1 I tried bypassing the x-over for the highs by running the sony output directly to my high frequency amp (through the external ML passive crossover) but I didn't like it. For some reason the electronic x-over was better. But since I made so many changes I thought it was time to challenge this prior conclusion. Since I read a comment by Elias that the Benchie could drive xlr y connectors I had my enabler for running highs directly to the panels and bass through the dbx to my bass and subwoofer amps.

 

What a revelation!!! I can honestly say this set up (Benchmark->QSC amp->Electrostatic panels of the martin logans) is so analogue like that it is scary. I know you all heard this before somebody tries a little tweak and all of a sudden instant kharma. But this pc audio is a whole different and unique (per cicplay/cmp hardware/software) approach. The results are stunning in the natural sense of transparency, non fatigue, dynamics and low level detail that I literally listen to music (rewardingly) for 5 - 7 hours a day (I am retired).

 

Wanted to share with all here because this approach is very cost effective. The pc parts cost me $420 and all the hardware/software for cplay/cmp specs are free. I rebuilt an old computer so I saved $ on case, cd drive, dvd drive costs. You all know how value minded the Benchie is, but you should hear it at 24/192 going straight into a high powered amp into electrostats. Finally after 40 years as an audiophile I have the sound I have wanted. Sure I can get better but we all know the feeling when we 'breakthrough' to some new level of performance and wow did I breakthrough. Please don't take this as yet another guy bragging 'I have the best system'. I know I don't in an absolute sense but I do 'for me'. I wanted to share the unbelievable performance levels one can get for very little money (by historic high end standards) so that others who share my love of / need for music can also try this approach. Yes it is difficult if you are not a computer geek but I was not and now I reflash my motherboard bios with no fear. Guys this is the real deal. It took me 6 months to build this but I am slow because I was a newbie to computers. There are a whole bunch of cplay/cmp grads on AA that are willing to help you when you run into trouble. Look I didn't like computers and resisted for a long time but now I have the front end of front ends as well as the knowledge of computer hardware/software that I picked up along the way. Whats not to like?