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Old 29th September 2011, 07:36 PM   #11
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Quote:
Originally Posted by abraxalito View Post
Sure, comparison with the sound of real instruments is a dead giveaway there's some distortion happening. I prefer voice myself because my own musical experience covers that. But the changes you've been getting probably parallel my own.
Sounds like we are on similar paths. Are you building a DAC based on these?

Quote:
Originally Posted by abraxalito View Post
Whilst I'm not yet sure about time inaccuracy, I am convinced that intermodulation effects are crucial and don't show up on repetitive sinewave testing as you seem to say.
Completely agree on intermoduation distortion. At one point I included a capacitor on the output as an RFI shunt but there was something about the sound that bugged me. Like a slight fuzz in female vocals at moderate listening levels. Almost like a Laryngeal leak or buzz. When I took the time to simulate it in circuit the intermoduation distortion products in the few kilohertz range were increased and easy to find on the real hardware. Removing the capacitor proved a positive, audible and measureable improvement.

What leads me down the path with time accuracy are two personal observations about high frequency information outside of our audible range but that make noticeable difference in the perceived sound quality of music reproduction for me. I noticed at some point that I strongly prefer audio amplifiers and preamps with wider bandwidths over systems limited to the audible range. By comparison it sounds as if the music is truer to being in a room with the instruments. Similarly one of the last changes I made with the 9023 design was comparing an oscillator with decent performance to one that was quite a bit better. The only differences I could measure objectively were increased non-harmonic spurs and slightly decreased normal harmonic content above the audible range with the lesser oscillator. It had a sound signature that made it more difficult to pinpoint where the instruments I use for location cues actually were in the sound field. With the better oscillator, the focus is much better. The objective measurements show an increase in the normal harmonic distortion products but the low level spurious stuff was mostly eliminated.All of this is above 20KHz. The normal harmonics are also related in phase with the fundamental where the spurious stuff is not.

My conclusion from all of this is that since the ears are incapable of "hearing" frequencies above somewhere in the 17-20KHz range but capable to perceive arrival time differences between the two ears outside of this audible bandwidth, time accuracy matters and bandwidth above 20KHz is necessary for reproduction accuracy in stereo or multi channel audio.

Quote:
Originally Posted by abraxalito View Post
Yeah I agree for radiated magnetics they're amongst the best. However on another measure - conducted noise - they're probably the worst owing to their relatively high primary-secondary capacitance. They can easily be an order of magnitude worse than an EI type on this measure..
Agree. This is in part why I use the inlet filter. Im not yet ready to share everything I did for normal mode rejection from earth ground, but it seems to work. I also went to great lengths with the power supply designs to maximize rejection into the hundreds of MHz range. From a few Hz and up the supplies' rejection are > -100dB and increase to two notches targeted at the 12MHz and 50MHz oscillator frequencies. I actually missed the 12MHz notch. It came in at a little better than 9MHz. Close enough though...

It's all overkill but the output noise floor speaks for itself.

Quote:
Originally Posted by abraxalito View Post
I hope the mains earth in your set up is jolly clean then, not polluted by PCs or LCDs with noisy SMPSUs. Myself I prefer not to rely on the earth quality (it sucks where I am) so much in filtering mains common-
mode noise.
Nope, quite noisy here too. I took a different approach to addressing this that appears to work well. It needs more testing in other situations to ensure it does not cause ground loop hum, but it has been flawless so far.

Dave
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Old 29th September 2011, 07:49 PM   #12
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Quote:
Originally Posted by qusp View Post
Great Job Dave!! what did you end up with in the filter pps? or np0? i cant see close enough to pick out of those 1210 size are panasonic pps or perhaps rubicon hybrid polymer/film. I think you'll find you take a pretty different tack with the 9018, the demo board is not a good indication of this chips sonic abilities.
Thanks qusp.

They are currently the 0805 Panasonics. The larger caps you see are for an experimental shelving circuit that is falling off in the next revision. I will be changing the footprint to accommodate up to 1210 sized parts in the higher voltages. Since this is an LPF (the caps are in parallel) I'm not sure if it matters but I want to try. It would also be easier to solder leaded caps to the larger pads for experimentation purposes.

I'm with you on the 9018. I look forward to building something unique to support that chip and see what it is really capable of. I am slowly collecting parts for the BII/D1 and hope to complete it this year. It will be nice to have comparative measurements and time to listen to it as well.

I don't know that the 9022/23 could be beat for the money though!

Dave
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Old 30th September 2011, 01:26 AM   #13
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Quote:
Originally Posted by fitzfish View Post
Sounds like we are on similar paths. Are you building a DAC based on these?
No - I heard one a couple of months back and it had at least one very endearing quality - huge depth and coherence of sound stage. But on voice it introduced a veiling effect which I wasn't partial to. I've pretty much abandoned low-bit DACs these days for a couple of reasons. One, we're already talking about - intermodulation. They generate very strong out of band signals and its tough to keep those signals from interfering with the audio. The other issue which all are subject to ISTM is noise modulation (ESS acknowledges this and claims theirs is better but still not perfect), and its this effect which I hypothesize is responsible for the voice veiling I heard.

Quote:
Completely agree on intermoduation distortion. At one point I included a capacitor on the output as an RFI shunt but there was something about the sound that bugged me.
Caps as RF shunts recently came up on John Curl's thread. I mentioned my own experience of them sounding bad (on current out DACs though). Nobody seemed to share that - Charles Hansen said his experience differed but was unwilling to share details. Since the ESS9023 is voltage out its too different a case to make comparisons.

Quote:
What leads me down the path with time accuracy are two personal observations about high frequency information outside of our audible range but that make noticeable difference in the perceived sound quality of music reproduction for me. I noticed at some point that I strongly prefer audio amplifiers and preamps with wider bandwidths over systems limited to the audible range. By comparison it sounds as if the music is truer to being in a room with the instruments.
My hypothesis for this would be the wider bandwidth amp has lower susceptibility to out of band interfering signals so generates fewer in-band intermod products. IME correct grounding helps improve soundstaging for the same reasons.

Quote:
Similarly one of the last changes I made with the 9023 design was comparing an oscillator with decent performance to one that was quite a bit better. The only differences I could measure objectively were increased non-harmonic spurs and slightly decreased normal harmonic content above the audible range with the lesser oscillator. It had a sound signature that made it more difficult to pinpoint where the instruments I use for location cues actually were in the sound field.
I've noticed some DACs do give 'pin point locations' for e.g. voices. This is not like real sound - listening to a real singer I have noticed no 'pin point' location effects. With real instruments, real sound sources they do sound a bit diffuse, definitely no sharp edges which is what sometimes happens with DACs.

Quote:
Agree. This is in part why I use the inlet filter. Im not yet ready to share everything I did for normal mode rejection from earth ground, but it seems to work. I also went to great lengths with the power supply designs to maximize rejection into the hundreds of MHz range.
When you are ready to share details, such will be very interesting I've offered some ideas about wide band mains filtering on my blog. I haven't found anyone else so far talking about mains filtering up to 100's of MHz - you're the first.
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Old 30th September 2011, 04:27 PM   #14
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Quote:
Originally Posted by abraxalito View Post
One, we're already talking about - intermodulation. They generate very strong out of band signals and its tough to keep those signals from interfering with the audio
One reason I chose a 2nd order LPF. I did not have space for a 4th order.
Quote:
Originally Posted by abraxalito View Post
My hypothesis for this would be the wider bandwidth amp has lower susceptibility to out of band interfering signals so generates fewer in-band intermod products. IME correct grounding helps improve soundstaging for the same reasons
Good point. I also agree on grounding. It took a while to get it right for the DAC board alone.

My context has been the source for a while and I have been thinking about analog audio in the time domain. In retrospect it makes good sense that mid frequency jitter modulation in the digital to analog conversion would cause distortion similar to (or just like...) IMD in the same band in and above our hearing range, although not necessarily harmonically related. Perhaps this and the time accuracy I described are the same. If the instrument sound is not accurate from transient attack through to the end of its decay the brain may not be able to put the image together cohesively. This is certainly true if the left and right channel are not adding theses artifacts identically.

Quote:
Originally Posted by abraxalito View Post
I've noticed some DACs do give 'pin point locations' for e.g. voices. This is not like real sound - listening to a real singer I have noticed no 'pin point' location effects. With real instruments, real sound sources they do sound a bit diffuse, definitely no sharp edges which is what sometimes happens with DACs.
"Pin point" was a poor choice of words from me. The intention was more about the perceptive act of locating the instrument in the sound field rather than a concrete descriptive of the sound. They should not sound like they are coming from everywhere and unable to be localized. They also should not move around with the other sounds. Cymbals again are a good measure of this performance to my ears as the ring decay will tend to attach itself to other sounds if there are issues like those we have been discussing.

I use instruments with percussive attacks for judging imaging and the time accuracy metric I suggested. Percussion, plucked strings, and piano. I agree - voice 'pin point' location is not realistic to me either. In good recordings it should be localized to its relative source location in the sound field (not seem like it is everywhere) and not move around as the music dynamics and mix of other instruments change.

I like to think of the percieved sound field in three dimensions as well. The depth dimension may well be closer related to time perception in the brain than to stereo imaging processing of the audio in it. Maybe not but an interesting topic for discussion elsewhere.

Quote:
Originally Posted by abraxalito View Post
When you are ready to share details, such will be very interesting I've offered some ideas about wide band mains filtering on my blog. I haven't found anyone else so far talking about mains filtering up to 100's of MHz - you're the first.
I am interested in your thought on mains filtering and will take a look.

My purposeful vagueness my have confused two details in the previous posts - The power supplies themselves have been designed to reject into the hundreds of MHz range so that noise passed through the transformers is rejected. The ground noise approach is a little unconventional. Take the two separately.

If it proves to work as well as it seems to, I'll share some details.
Dave
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Old 30th September 2011, 05:35 PM   #15
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Hi Dave,

about your DJitter measurement...

the base signal is usual 16 bit (no dither) -6db with a LSB toggle (square wave)... and then please zoom in to get the narrow modulated symmetric waves (if any). May look at my web about my measurement using a RME BabyFace..

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Old 1st October 2011, 06:27 PM   #16
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Hi Hp,
Quote:
Originally Posted by HpW View Post
about your DJitter measurement...

the base signal is usual 16 bit (no dither) -6db with a LSB toggle (square wave)... and then please zoom in to get the narrow modulated symmetric waves (if any). May look at my web about my measurement using a RME BabyFace..
I used a 48KHz 24 bit version of the JTEST signal for the graph in the original post. This ensures the harmonics from JTEST are under the noise floor. It looks like the fundamental is -3dB.

Here are three additional graphs - first is a 48/24 JTEST capture zoomed in. Second is the 44.1/16 JTEST. Last is a zoom of a 44.1/16 JTEST capture.

Dave
Attached Images
File Type: jpg 48_24JtestZoom.jpg (122.2 KB, 676 views)
File Type: jpg 441_16Jtest.jpg (113.6 KB, 588 views)
File Type: jpg 441_16JtestZoom.jpg (122.0 KB, 574 views)
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Old 2nd October 2011, 07:03 AM   #17
HpW is offline HpW  Switzerland
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Quote:
Originally Posted by fitzfish View Post
Hi Hp,


I used a 48KHz 24 bit version of the JTEST signal for the graph in the original post. This ensures the harmonics from JTEST are under the noise floor. It looks like the fundamental is -3dB.

Here are three additional graphs - first is a 48/24 JTEST capture zoomed in. Second is the 44.1/16 JTEST. Last is a zoom of a 44.1/16 JTEST capture.

Dave
What I still missing is a complete zoom in into the +/- 10...100Hz region.. while your bin freq. is about 0.2Hz. And use also some averaging (may 8 times). Compare also with my measurement (currently presented on my web) where I used a 2^25 FFT size. Be also careful not to overdraft with the level. The basic measurement is by J. Dunn - 6.02 dBFS.

Second on your IM measurement: the level is here also very high near dBFS and this caused some IM products 2KHz -78 dBFS and 16K / 22K -92 dBFS.

You where also writing about some output filter caps to get ride of RF. the better design would be to not have/feed any RF on the analog part. This requires a complete redesign. I did this with my DAC project where the DAC & analog part where in complete different & closed copper boxes O

Additional if the DAc or the analog part is sensitive to IM product, may consider to bench it using an IM cluster 16..18Khz.

Hp
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Old 2nd October 2011, 04:10 PM   #18
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Hi HP,
Quote:
Originally Posted by HpW View Post
What I still missing is a complete zoom in into the +/- 10...100Hz region.. while your bin freq. is about 0.2Hz.
Pictures of captures at +/-400Hz and +/- 100Hz are attached below. What are you looking for?


Quote:
Originally Posted by HpW View Post
And use also some averaging (may 8 times). Compare also with my measurement (currently presented on my web) where I used a 2^25 FFT size.
The R&S HPV analizer I have has some limitations. I cannot do greater than 256K points and averaging in the FFT window is not implemented. It is what I have so I use it... At some point I will probably upgrade.


Quote:
Originally Posted by HpW View Post
Be also careful not to overdraft with the level. The basic measurement is by J. Dunn - 6.02 dBFS.
The JTEST files I have were not generated by me. Can you provide JTEST wave files with the appropriate level? 16 and 24 bit files at 44.1 and 48KHz would be appreciated. 60 seconds in length is fine.


Quote:
Originally Posted by HpW View Post
Second on your IM measurement: the level is here also very high near dBFS and this caused some IM products 2KHz -78 dBFS and 16K / 22K -92 dBFS.
Each tone is set at -6dBFS for that graph. What are you advocating for levels of each tone?


Quote:
Originally Posted by HpW View Post
You where also writing about some output filter caps to get ride of RF.
The purpose of the caps in that experiment was not RF from the DAC but from external sources and the interconnects. It was directly at the output after the JFET LPF and current limiting resistor. Implementing faraday shields and other design changes to mitigate RF at the design level are extreme overkill for a design based on a $2US DAC chip. Just my opinion here.

Quote:
Originally Posted by HpW View Post
Additional if the DAc or the analog part is sensitive to IM product, may consider to bench it using an IM cluster 16..18Khz.
I am unaware of this testing method. Can you describe it further?


Dave
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File Type: jpg 441_16JtestZoom_400.jpg (137.9 KB, 541 views)
File Type: jpg 441_16JtestZoom_100.jpg (134.9 KB, 534 views)
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Old 6th October 2011, 01:21 PM   #19
HpW is offline HpW  Switzerland
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Hi,

>> I cannot do greater than 256K points and averaging in the FFT window is not implemented. It is what I have so I use it... At some point I will probably upgrade.

>> The JTEST files I have were not generated by me.


>> I am unaware of this testing method. Can you describe it further?

IM Cluster : 15kHz ...16kHz range using 6 frequencies some with 100Hz and some with 200Hz separated

All are invited to evaluate my SW while all noted items above, are known implemented features ...

Hp
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Old 6th October 2011, 03:47 PM   #20
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Hi HP,
Quote:
Originally Posted by HpW View Post
What I still missing is a complete zoom in into the +/- 10...100Hz region..
What were you looking to see in this picture?

Dave
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