CHORD ELECTRONICS DAVE

[gnomen]

Those of us in the UK can still listen to a number of last summer's Prom concerts on the BBC iPlayer. A good example is Maher's 7th, Simon Rattle & BPO. It has a full range of orchestral timbres, cowbells in the distance, even the occasional cough from the audience (which you would normally hear only with headphones). My TT reproduces it all beautifully. When I close my eyes, I could be there - albeit in the best seat.

The ability of Rob's DACs to get the best from modest inputs - the iPlayer audio stream, the BT YouView set top box, or 16/44 red book CD is something reviewers should praise more. Not only do the DACs perform well in the absolute sense, under ideal conditions, they deliver the best from a wide range of variable sources and materials.

 

[hieukm]

Hi guys,

Is this correct that i cant add external clock to my Dave?

 

[lovethatsound]

Hi guys,

Is this correct that i cant add external clock to my Dave?

Yes you are right.

 

[x RELIC x]

Hi guys,

Is this correct that i cant add external clock to my Dave?

Yes you are right.

Curious why you would want to. Rob has stated in this thread that it's not necessary at all with his designs, and that his approach eliminates the need.

 

[hieukm]

Curious why you would want to. Rob has stated in this thread that it's not necessary at all with his designs, and that his approach eliminates the need.

 
I am hearing some improvements going from
 
- IFI (power by JS-2) to Dave's USB.
- IFI (power by JS-2) to Hydra-Z(power by JS-2) to Dave's AES.
- Adding FMC to isolate my Ethernet connection to PC.
(All components were plugged into a Torus AVR 20. The DAC/preamp even go through a Furutech FTP 615(plugged into the Torus) for futher elimination of EMI/RF)
 
Some of the above may contradict with some of Rob's statement
 
- Source jitters dont matter.
- Dave's USB is galvanic isolated.
 
Eventually, it led me to wonder if the Dave can benefit from additional external clock. 

 

[Rob Watts]

 

Hi guys,

Is this correct that i cant add external clock to my Dave?

Yes you are right.

Curious why you would want to. Rob has stated in this thread that it's not necessary at all with his designs, and that his approach eliminates the need.

 
In the past, (DAC64 and QBD76) I had a buffer system, so the data could be clocked out using the local crystal master clock. If you wanted to use video though, then you had to use the analogue PLL system, which gave much poorer SQ and measurable problems. I had spent many years improving PLL's but there are some issues you just can't solve, such as low frequency jitter - and low frequency jitter is very audible, and measurable as it creates fringing on FFT's (fringing is where if you play back 1 kHz, you should just see a sharp 1 kHz component - but the FFT will show a spreading and not a single 1 kHz line). The fringing is due to the PLL drifting in low frequency terms as it maintains lock.
 
Because video is very important to me, I wanted a system that would eliminate the PLL SQ problems, and so I developed the DPLL system that you can find on all my DAC's - and it took many years to perfect. But what it means is that when you do a listening test between the DPLL and the buffer system, it sounded almost identical - I struggled to hear a difference. Given the problems of transmitting clocks back (ground and noise issues), I felt that it was better now not to use word clocks from Dave to the source.
 
Now using an external clock to run Dave's master clock would be a completely crazy thing to do, as you cannot transmit 104 MHz with no added jitter. So you have to use local PLL's, and although these can be made to have fS of cycle to cycle jitter, thus giving you wonderful marketing copy, they are awful for low frequency jitter. Now the benefits of pulse array is that because the array is internally cancelled (rising edges is matched by a falling edge pulse) it is innately much less sensitive to master clock jitter. But like all DAC's there is nothing I can do about LF drift as the PLL jumps up and down in maintaining lock. So an external clock would make the measurements worse (more fringing) and the sound quality bright (reduce fringing and it sounds warmer and more natural). I think this is why people "like" external clocks - because when they get added things sound brighter, and its very easy to confuse a brighter sound due to more distortion with a truly more transparent sound.
 
I hope that explains why I don't use external master clocks or transmit word clocks.
 
Rob

 

[analogmusic]

The ability of Rob's DACs to get the best from modest inputs - the iPlayer audio stream, the BT YouView set top box, or 16/44 red book CD is something reviewers should praise more. Not only do the DACs perform well in the absolute sense, under ideal conditions, they deliver the best from a wide range of variable sources and materials.

 
personally I think most reviewers miss the whole point of the Rob's DACs - the musical aspect of them, and how they get the emotional part of music right.
 
There isn't much point listening to music if the emotional aspect of the instruments and voices are not rendered properly.

 

[halloweenman]

Rob Watts or anyone else that can help! I have a bass frequency bump issue in my listening room around 45Hz. Do you know of any good quality low shelf filter software (apple audio unit plug in would be ideal) that could help me without degrading sound quality? I use audirvana software player. Thanks.

 

[Arpiben]

   
In the past, (DAC64 and QBD76) I had a buffer system, so the data could be clocked out using the local crystal master clock. If you wanted to use video though, then you had to use the analogue PLL system, which gave much poorer SQ and measurable problems. I had spent many years improving PLL's but there are some issues you just can't solve, such as low frequency jitter - and low frequency jitter is very audible, and measurable as it creates fringing on FFT's (fringing is where if you play back 1 kHz, you should just see a sharp 1 kHz component - but the FFT will show a spreading and not a single 1 kHz line). The fringing is due to the PLL drifting in low frequency terms as it maintains lock.
 
Because video is very important to me, I wanted a system that would eliminate the PLL SQ problems, and so I developed the DPLL system that you can find on all my DAC's - and it took many years to perfect. But what it means is that when you do a listening test between the DPLL and the buffer system, it sounded almost identical - I struggled to hear a difference. Given the problems of transmitting clocks back (ground and noise issues), I felt that it was better now not to use word clocks from Dave to the source.
 
Now using an external clock to run Dave's master clock would be a completely crazy thing to do, as you cannot transmit 104 MHz with no added jitter. So you have to use local PLL's, and although these can be made to have fS of cycle to cycle jitter, thus giving you wonderful marketing copy, they are awful for low frequency jitter. Now the benefits of pulse array is that because the array is internally cancelled (rising edges is matched by a falling edge pulse) it is innately much less sensitive to master clock jitter. But like all DAC's there is nothing I can do about LF drift as the PLL jumps up and down in maintaining lock. So an external clock would make the measurements worse (more fringing) and the sound quality bright (reduce fringing and it sounds warmer and more natural). I think this is why people "like" external clocks - because when they get added things sound brighter, and its very easy to confuse a brighter sound due to more distortion with a truly more transparent sound.
 
I hope that explains why I don't use external master clocks or transmit word clocks.
 
Rob

 Please at which value in ns do you estimate that LF drifts or LF jitter are audible ?
Just curious since I am used to deal with them in non audio applications (Telecom).
Thanks.
Rgds.

 

[rkt31]

@halloweenman, the most elegant solution of eliminating bass peaks is toeing in of speakers. you must have kept your speakers straight firing. toe in until you get best balance between bass and treble. more toe in than necessary will make vocal sibilant. less toe in will cause boomy bass. it's something like if you can't have non parallel side walls to avoid standing waves , make the speakers non parallel.

 

[halloweenman]

thanks rkt31, the speakers are pretty much already fully toed in, firing to just behind the listening position. the eq really is a last resort as i have tried pretty much everything now.

 

[Crgreen]

personally I think most reviewers miss the whole point of the Rob's DACs - the musical aspect of them, and how they get the emotional part of music right.

There isn't much point listening to music if the emotional aspect of the instruments and voices are not rendered properly.

In fairness, the reviews I've read do draw attention to this, though not perhaps using the same words, but stress the ease and naturalness of replay. I've got extraordinary results from MP3s@320 and lower on the Hugo and the Dave - utterly engrossing, and of older recordings too, such as Furtwangler Brahms.

 

[halloweenman]

anyone using jriver room correction or eq to reduce bass boom?

 

[ecwl]

  anyone using jriver room correction or eq to reduce bass boom?

I do. I run Windows PCs. I optimized my room setup but still couldn't get rid of a bass peak from my left speaker at around 50Hz. I bought an XTZ Room Analyzer Pro (not compatible with Mac) to measure the bass peak and it calculated that I need to run a parametric EQ at 53Hz with Q of 18.5 and -11.8dB to remove the bass peak. I set that in JRiver and that virtually fixed the problem. I always presumed Audirvana would have a similar feature. I wish I didn't have to use EQ with Chord DAVE but I've really optimized my room, seat and speaker setup to the best of my abilities without turning my living room into a hi-fi shop.
 
My take is that to do effective parametric EQ, you need to measure precisely. Any guesswork would just make the situation worse. I think you can go to miniDSP website and order a USB microphone UMIK-1
https://www.minidsp.com/products/acoustic-measurement/umik-1
And then you can download the freeware Room EQ Wizard
http://www.roomeqwizard.com/
Learning how to use Room EQ Wizard is the biggest challenge but once you get the hang of it, you should be able to get it to recommend what parametric EQ you should apply for that 45Hz peak. I thought I wouldn't be able to figure it out so I bought the XTZ Room Analyzer but I have some friends who needed help so I ended up learning how to use Room EQ Wizard for the same and more advanced functionalities. In fact, I now personally believe that all high-end audiophiles with speaker setups should own their own real-time analyzer.

 

[watermad]

I'm a very happy Chord DAVE DAC owner of approximately 4 months and was wondering how fellow DAVE owners have it connected to their speakers.
 
I have been primarily connecting it in DAC mode running through my Music First Audio Classic MKII Passive Transformer pre-amplifier as I still listen to vinyl and during a short A-B of the DAVE in DAC v Pre-amplifier found I lost some dynamics with the DAVE using its pre-amplifier.
 
I understand this is counter intuitive as the DAVE in pre-amplifier mode should be more direct, however my MFA pre-amplifier is as neutral as you can get, I will do more exhaustive A-B listening in the immediate future, however was wondering how other DAVE owners had configured it in their speaker setup?