[Shadorne]

Quote:

Originally Posted by little-endian /img/forum/go_quote.gif The question is if the jitter really increases with cable length or just the attenuation.

Yes. Cable length is a factor.

 

[wavoman]

I did a bunch of experiments tonight, and the Benchmark UltraLock is amazing.

My SPDIF source was a Wadia iTransport. I ran the SPDIF to a 75-ohm RCA patch panel, and then into either a Wadia DAC or the Benchmark. Both locked on at 44.1 flawlessly.

Then I patched other units in the middle of the chain. First a Sonifex (pro gear) Redbox SPDIF repeater. I only plugged one thing into it, not multiple. The Benchmark locked on, no problem. The DAC would lock on, then lose it, etc. -- lots of drop outs.

Then I took the Sonifex out, and put in a SPDIF upsampler, the Monarch DIP. This is also supposed to be a jitter-reduction device. Again, Benchmark had no problem. But the Wadia would not sync. I am testing these one-at-a-time, not splitting the signal.

Very impressive performance for the Benchmark. Cable length, jitter injection, etc. are all real-world problems if you have a complex set-up. I am now a huge UltraLock fan!

I came out of the Benchmark balanced in to a Stax 717, and listened with Lambda Sigs, and also Baby Orpheus with a moon-audio adapter. Wonderful wonderful SQ (the iPod in the Wadia iTransport had all uncompressed WAV files).

 

[EliasGwinn]

Quote:

Originally Posted by hpz /img/forum/go_quote.gif Hi Elias, I have just purchased a DAC1 USB and i am using them with IEMs via the headphone outputs. Is there anyway to lower the gain even further than level -10db? I'm only using about 4-5 ticks up (8 o clock) and its already loud enough, but the sound is imbalanced and unclear. Anyway to reduce the gain even more? Any help would be appreciated Thanks

hpz,

The HPA2 headphone amplifier in the DAC1 USB has two gain ranges...normal and '-10 dB'. In the DAC1 PRE and DAC1 HDR, the HPA2 has an additional lower gain range (-20 dB).

IEM's often have very low impedance and very high sensitivity. The HPA2 is a very powerful headphone amplifier, so it doesn't surprise me that your headphones are too sensitive for the HPA2. You may find the additional gain ranges of the DAC1 PRE and DAC1 HDR more suitable for your headphones.

All the best,
Elias

 

[G-U-E-S-T]

Well, it wasn't easy - but I finally located a great solid-state stereo amp (not class-D or tube) with excellent specifications, and that produces less than 30 watts with an input sensitivity greater than 1.5 volts.

As always, Elias is correct - this has in fact made a very noticeable and quite nice improvement to our overall sound. Since we only listen at low/medium levels through terrific bookshelf speakers with a powered subwoofer, we might not even normally be using half the power of this much less powerful amp.

Thanks Elias for your advice about maintaining a good signal-to-noise ratio, by not using an amp with "too much" power for normal listening levels. I'm thinking that like myself, many people probably buy a way-overpowered amp for their needs (which seems to be the common sales pitch these days), thus actually reducing the quality of their sound.

Also, since it's not that easy to actually locate such an amp as described above (class A/AB, great specs, less than 30w, input sensitivity > 1.5v), perhaps Benchmark would consider actually adding such an item to their lineup?

 

[lamikeith]

Quote:

Originally Posted by G-U-E-S-T /img/forum/go_quote.gif Well, it wasn't easy - but I finally located a great solid-state stereo amp (not class-D or tube) with excellent specifications, and that produces less than 30 watts with an input sensitivity greater than 1.5 volts.

Please, tell us the brand and model of the amp so that we all may benefit from your research.

 

[Shadorne]

Quote:

Originally Posted by G-U-E-S-T /img/forum/go_quote.gif Well, it wasn't easy - but I finally located a great solid-state stereo amp (not class-D or tube) with excellent specifications, and that produces less than 30 watts with an input sensitivity greater than 1.5 volts. As always, Elias is correct - this has in fact made a very noticeable and quite nice improvement to our overall sound. Since we only listen at low/medium levels through terrific bookshelf speakers with a powered subwoofer, we might not even normally be using half the power of this much less powerful amp. Thanks Elias for your advice about maintaining a good signal-to-noise ratio, by not using an amp with "too much" power for normal listening levels. I'm thinking that like myself, many people probably buy a way-overpowered amp for their needs (which seems to be the common sales pitch these days), thus actually reducing the quality of their sound. Also, since it's not that easy to actually locate such an amp as described above (class A/AB, great specs, less than 30w, input sensitivity > 1.5v), perhaps Benchmark would consider actually adding such an item to their lineup?

FANTASTIC - you must be delighted.

What "terrific bookshelves" and what amplifier?

Also "good signal to noise" is not that hard to achieve - you make it sound like it was some nigh impossible quest. Use good balanced XLR equipment with ordinary "Mogami" interconnects and plug everything into the same outlet with no cheater plugs and you should be done in most cases. I wonder if you did not have a ground loop issue (all too common with RCA where negative signal wire and ground are one and the same)? Or perhaps the simple answer might be that something else was wrong with your previous amplifier? All I am suggesting is that there might be more to it before jumping to conclusions (other than this fixed your problem which is great news).

For example, do not confuse "way-overpowered" amplifier (and assume that most people are making a mistake by having a powerful amplifier) with input sensitivity and gain - the difference between 30 watts and 300 watts is not that significant....it will only be perceived as twice as loud - we are talking a mere 10 decibels here (it is the sensitivities/gains that count not the overall power capability). The most "common" error in amplifier selection is a poor match to speakers: either impedance match issues or the amp is underpowered or overpowered for the speaker. I simply don't think you can jump to the conclusion that you have stumbled on something that is a problem for most people with powerful amplifiers...

 

[EliasGwinn]

Quote:

Originally Posted by little-endian /img/forum/go_quote.gif 1. DAC1 inside view Weren't you going to investigate why the inside views disappeared on Benchmark's web site?

Here is the internal view of the DAC1 USB:

http://www.benchmarkmedia.com/system...-rm_inside.jpg

Here is the internal view of the DAC1 HDR:

http://www.benchmarkmedia.com/system...hdr-inside.jpg

Quote:

Originally Posted by little-endian /img/forum/go_quote.gif 2. Volume control Here and in the feedback newsletter you explained how and why DSPs reduce the SNR aka dynamic range compared to volume pots. I don't doubt that a second however it is still not totally clear to me why this gives any drawback in real-life practice: You said a DSP controls the values before the D/A conversion. Lower values mean a lower SNR by definition. A volume pot might lower the noise as well, keeping the SNR by definition, isn't it? But if one listens at lower levels for comfortable listening, reaching ~ 70 dB(A) at maximum for instance, don't my ears limit the SNR / dynamic range as well? Aren't the ears comparable to the DAC - lower "values" (or sound pressures) increase the "wasted" headroom? From hearing threshold level to maximum there might be just 40-50 dB left (due to environment noise). Lowering the volume, wouldn't the dynamic range decrease due to hearing capability anyway - regardless of the technique used to lower the volume? Asked another way: Assuming I have music playing with an approximate dynamic range of 30 dB (this should be realistic for recordings beyond the loudness war; early 80s for instance) - one time with the DAC1, the other with an AVR using a DSP for volume control for instance. In both cases I set the volume so the loudest parts reach 80 dB(A) (and the quietest 50 dB (A) respectively). What exactly will I gain in regard to SNR or dynamic range when listening to the DAC1 compared to the AVR? Of course - by definition, the noise will be slightly higher with the AVR since the DAC's full scale isn't entirely used anymore but wouldn't that introduced error below the ~ 50 dB of the quietest parts in the music anyway - being masked so to speak?

Don't confuse dynamic range of the music to the dynamic range of the audio electronics. Even if the dynamic range of the music is 10 dB, you can still hear noise well below that.

Noise can only be masked by more noise, not a collection of tones. If there is enough tones in enough different frequencies at sufficient amplitudes, it may approach noise and create a masking affect. But a violin recording at -2 dBFS will not mask noise at -80 dBFS.

With that being said, other factors may limit the signal-to-noise of your listening experience (HVAC noise, your ears ringing, etc). However, many people have very quiet listening environments that benefit from a maximized electronic signal-to-noise ratio.

Quote:

Originally Posted by little-endian /img/forum/go_quote.gif 3. AVRs I know you avoid taking position when it comes to competitive products for good reason but what do you think in general about most AV-receivers. As far as I know almost all use DSPs to adjust the volume, aren't they? Are 32-bit DSPs common or something else?

I don't have enough information to say with authority what DSP chips most AVR's use. However, it doesn't change the fact that the D/A chip will still have a fixed noise floor that will be 'closer' to the signal when a lower digital signal is presented.

All the best,
Elias

 

[EliasGwinn]

Quote:

Originally Posted by JamieGreen /img/forum/go_quote.gif First I have to say that I like the Benchmark DAC1 pre with my IEM (Wum3x) very much with 20 dB gain reduction. The question is if I can set one headphone jack -20dB and second left with -10dB for my another 300Ohm headphones? Thanks

Hello JamieGreen,

I'm sorry I haven't addressed your question sooner.

The short answer is... no. Both headphone jacks use the same HPA2 headphone amplifier. The gain setting on that HPA2 affects both jacks equally.

All the best,
Elias

 

[EliasGwinn]

Quote:

Originally Posted by little-endian /img/forum/go_quote.gif Yeah, I think that would be my forth point for Elias. It is interesting - I recognized that my Harman/Kardon AVR for example outputs a distored audio signal if the S/PDIF is too bad while the Benchmark DAC1 plays the same source flawlessly and never distorts but mutes when in doubt. The question is if the jitter really increases with cable length or just the attenuation. It would be interesting to know what causes the different input sesitivity of different devices.

Cable lenght absolutely affects jitter levels.

The thing that affects input jitter sensitivity is the clock-recovery mechanism. We have demonstrated that the DAC1 can maintain its fully-rated performance even with severe levels of jitter on the line.

All the best,
Elias

 

[rmh1]

Hey Elias,

Looks like the manual for the HDR needs an update. The section on changing the jumpers is the old one from the PRE manual, pages 15 and 16. You need to change the header descriptions and photos on pages 18 and 19 of the HDR manual to indicate the new header numbers.

Here is the internal view of the DAC1 HDR:

http://www.benchmarkmedia.com/system...hdr-inside.jpg

 

[mahesh]

After reading 180 pages,i orderd dac 1 usb and got it yesterday,
its still burning,i am using adam a5 and sennheiser hd 25 ii. Till now iam satisfied but I will try to give my impression later,thank you Elias and everyone for helping to choose dac1.
Sorry for my bad english

 

[mahesh]

 

[EliasGwinn]

@ mahesh:

 

[pekingduck]

Hi Elias,

Any chance of changing the HPA2's gain jumpers to a front panel switch in the next generation DAC1?

As a potential buyer I find it quite troublesome having to open the chassis everytime to change the gain, especially for people having headphones of widely varying sensitivities.

Thanks

 

[little-endian]

Quote:

Originally Posted by EliasGwinn /img/forum/go_quote.gif Here is the internal view of the DAC1 USB: http://www.benchmarkmedia.com/system...-rm_inside.jpg Here is the internal view of the DAC1 HDR: http://www.benchmarkmedia.com/system...hdr-inside.jpg

Thanks for the update Elias. I knew everything was fine, hehe.

Quote:

Originally Posted by EliasGwinn /img/forum/go_quote.gif Don't confuse dynamic range of the music to the dynamic range of the audio electronics. Even if the dynamic range of the music is 10 dB, you can still hear noise well below that. Noise can only be masked by more noise, not a collection of tones. If there is enough tones in enough different frequencies at sufficient amplitudes, it may approach noise and create a masking affect. But a violin recording at -2 dBFS will not mask noise at -80 dBFS.

Yes I understand what you mean. This is one part, lossy codecs make use of, isn't it? Similarly, dithering still allows details to be resolved despite its introduced noise floor, right?

Quote:

Originally Posted by EliasGwinn /img/forum/go_quote.gif With that being said, other factors may limit the signal-to-noise of your listening experience (HVAC noise, your ears ringing, etc). However, many people have very quiet listening environments that benefit from a maximized electronic signal-to-noise ratio.

Yeah, okay let's assume one has optimal listening conditions - no background noise, a high SNR of the electronic devices, etc ...

Quote:

Originally Posted by EliasGwinn /img/forum/go_quote.gif I don't have enough information to say with authority what DSP chips most AVR's use. However, it doesn't change the fact that the D/A chip will still have a fixed noise floor that will be 'closer' to the signal when a lower digital signal is presented.

... won't my hearing capability limit the SNR / dynamic range as well I can enjoy when listening to music at low levels? So lower values given to a DAC (either because the music contains quite passages or the overall volume is lowered by a DSP) mean a lower SNR because the DAC's maximum output will be lower as well while its noise floor will be unchanged. That's clear so far. But a level of 0dBSPL is the absolute minimum to hear something, correct? When I set my volume so that a maximum of let's say 60dBSPL is archived - wouldn't be my SNR / dynamic range 60dB at best anyway due to my ears?

So I think I understand your explanation why the poti solution the DAC1 uses is superior - I just wonder if the benefit isn't consumed by the limitations of hearing per se.

Quote:

Originally Posted by EliasGwinn /img/forum/go_quote.gif The thing that affects input jitter sensitivity is the clock-recovery mechanism. We have demonstrated that the DAC1 can maintain its fully-rated performance even with severe levels of jitter on the line.

Yeah, indeed. The performance of the DAC1 on that regard is undisputable and just great. I little hope arises: Does Benchmark plan to develop a device for multichannel PCM by chance? You mentioned that some use several DAC1s to built a multichannel environment but since it's quite a fumble and still involves electronics (the power amplifier), it would be great to have a solution including a DAC and the amplifier with all the Benchmark philosophy.