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O2 AMP + ODAC - Page 206

post #3076 of 3481
Quote:
Originally Posted by James-uk View Post


Interesting point about dac volume being at 100% . Correct me if I'm wrong but does this mean it's best to turn the digital volume down a few db in bit perfect/ pure music for mac etc for care free listening at best sound quality?

 

Yes, a single single notch below 100% is best. It's rare to hit 0dB in high quality recordings, but if you happen to listen to tracks with significant 0dB content, it's a useful adjustment. I found a great explanatory article last year and will have to dig that up again...

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post #3077 of 3481
Quote:
Originally Posted by jseaber View Post

Yes, a single single notch below 100% is best. It's rare to hit 0dB in high quality recordings, but if you happen to listen to tracks with significant 0dB content, it's a useful adjustment. I found a great explanatory article last year and will have to dig that up again...

Thank you.
post #3078 of 3481
Quote:
Originally Posted by jseaber View Post
 

 

Yes, a single single notch below 100% is best. It's rare to hit 0dB in high quality recordings, but if you happen to listen to tracks with significant 0dB content, it's a useful adjustment. I found a great explanatory article last year and will have to dig that up again...

Please do so.

 

Also, how is progress on an ODA? Just curious because I remember asking a couple months ago and was curious how you guys are doing, unless you're not allowed to say which is perfectly fine.

post #3079 of 3481
Quote:
Originally Posted by schmalgausen View Post
 

In all manuals we can see 2 methods to connect O2 and ODAC according to usage of 3.5 input socket: 1) to keep O2 input on it, 2) to change it to ODAC output. 

However, ODAC has two outputs: four pins in the middle of PCB and 3.5 mm socket. Why nobody uses them both? Pins to connect ODAC with O2 and 3.5mm socket to get direct ODAC output. Does such setup affect the output quality with some negative influence of O2 input?

 

Hi,

 

it could be problematic to have the ODAC drive two powered on amps - this depends on the amps input impedances - O2 is quite high here (which is desirable) but others might not be so kind...

 

Also it's not recommended at all to have anything drive the unpowered input stage of any amp as this might result in funny sound at best and sth breaking if you're not lucky.

 

I plan to solder the 3.5" jack on the ODAC board when I find time and will use an extra 3.5 plug in the O2 line in socket to isolate the O2 from ODAC using the latters line-out - my O2 has the input wired as an input when a plug is in.

 

Joachim

post #3080 of 3481

John,

 

NwAvguy has stated:

 

SINE WAVE TESTING: While I’ve abused the O2 in a variety of ways playing real music into assorted worst case loads, and it has survived nicely, it will eventually overheat the output op amps with sustained high power sine wave (or square wave) testing into low impedance loads. This is true of many high power headphone (and loudspeaker) amps that are designed for music not test signals. It’s important to limit full power sine wave testing to a few seconds below about 150 ohms and let the output ICs cool down between tests. You should also run sine wave tests with at 15 VAC or greater wall transformer.

 

To be fair have you measured the booster board at 150 ohms?

 

Alex

post #3081 of 3481

Hi, something I've noticed is that when I feed O2 LOs other than ODAC, it doesn't output as detailed sound and less definition.  I'm not sure if O2 is transparent.  

 

What I noticed is ODAC is overly bright in itself, and matches ODAC well, as I believe O2 probably cut's off the sharpness.  The pairt is perfect, the best sounding source I've ever heard for my CIEMs.  Believe me, I've tried all the high end DAPs out there, and nothing can match this pair.  So accurate and realistic.  The pair is amazingly clear and detailed.  I don't if you guys ever heard very dynamic imaging, but it pushes my CIEMs to unbelievable dynamism.  I didn't know CIEMs can do such things.

 

Now, if I can get a portable to sound exactly like this pair, I'm done searching for portables.  You hear me JDS Labs?


Edited by SilverEars - 6/11/14 at 8:19pm
post #3082 of 3481
Quote:
Originally Posted by adydula View Post
 

John,

 

NwAvguy has stated:

 

SINE WAVE TESTING: While I’ve abused the O2 in a variety of ways playing real music into assorted worst case loads, and it has survived nicely, it will eventually overheat the output op amps with sustained high power sine wave (or square wave) testing into low impedance loads. This is true of many high power headphone (and loudspeaker) amps that are designed for music not test signals. It’s important to limit full power sine wave testing to a few seconds below about 150 ohms and let the output ICs cool down between tests. You should also run sine wave tests with at 15 VAC or greater wall transformer.

 

To be fair have you measured the booster board at 150 ohms?

 

Alex

 

agdr and I have discussed this and believe NwAvGuy made a mistake here. With a 15VAC transformer, voltage drop across U5 and U6 is higher, resulting in another thermal issue. It's best to use 14VAC with O2 as far as heat is concerned.

 

Yes, I suspect that even the default NJM4556s would overheat with time while driving since waves at full power. The issue is that I was able to complete testing with O2 at 32 ohms. I was not able to complete the same tests with an O2 booster at 32 ohms. Its output power will need to be limited next time testing is conducted.

 

The booster board is thermally stable into 150+ ohm loads. See post #3055 for initial test notes. Output power is unchanged with the booster board into these loads, so it brings no audible advantages aside from the relay's pop-free on/off behavior (which is nice).

 

Quote:
Originally Posted by HybridCore View Post
 

Please do so.

 

Also, how is progress on an ODA? Just curious because I remember asking a couple months ago and was curious how you guys are doing, unless you're not allowed to say which is perfectly fine.

 

I've revealed as much as I can in post #3073  :-)

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post #3083 of 3481
Quote:
Originally Posted by jseaber View Post

Yes, a single single notch below 100% is best. It's rare to hit 0dB in high quality recordings, but if you happen to listen to tracks with significant 0dB content, it's a useful adjustment. I found a great explanatory article last year and will have to dig that up again...

Sorry , another question regarding this. Is it the dac volume I should turn down or the digital volume (dither) ? My dac volume is adjustable from the mac master volume and I can also turn down the digital volume using 64 bit dithered volume control. What are your thoughts on dithered volume control ? Sorry for the questions i just like to know I'm getting the best possible playback etc .
post #3084 of 3481

John,

 

Thanks for the reply.....sounds like your working on a desktop amp of some kind...and if you have tested a LME49600 board my guess its designed around that!!! LOL??

 

Anyway...what voltage is your dScope outputting for this testing???

 

In "his" blog I see a graph of 4.2vrms for 33ohms....so if your running tests above this voltage I would expect this to overheat???

 

Alex


Edited by adydula - 6/11/14 at 12:26pm
post #3085 of 3481
Quote:
Originally Posted by James-uk View Post


Sorry , another question regarding this. Is it the dac volume I should turn down or the digital volume (dither) ? My dac volume is adjustable from the mac master volume and I can also turn down the digital volume using 64 bit dithered volume control. What are your thoughts on dithered volume control ? Sorry for the questions i just like to know I'm getting the best possible playback etc .

 

Either volume control will satisfy the end goal of avoiding a 0dB condition as observed at the DAC. Sorry, I'm not familiar with the specific controls in your system.

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post #3086 of 3481
Quote:
Originally Posted by jseaber View Post

Either volume control will satisfy the end goal of avoiding a 0dB condition as observed at the DAC. Sorry, I'm not familiar with the specific controls in your system.
I thought as much but just checking. Thanks.
post #3087 of 3481
Quote:
Originally Posted by James-uk View Post

I thought as much but just checking. Thanks.

I'm not sure I'm understanding this. I use JRiver. I've set the volume control to change the internal volume but how much should I cut it in dB? 10dB? 20dB?

Sorry for being obtuse.
post #3088 of 3481

zorofox...

 

take a look here:

 

http://www.audioholics.com/audio-technologies/the-case-for-not-going-above-0-dbfs-for-digital-playback-systems

 

I also found this:

 

"Unless you ask Jriver to perform any manipulation of the data it will output exactly what the source bits are and there should be no need to activate additional clipping protection. The default is bit transparent to the sound driver.

Then it depends on the sound driver whether you have any manipulation. ASIO, Kernel streaming, and both WASAPI modes are bit transparent. Jriver's volume control only attenuates."

 

AND THIS:

 

So with just about any DAC, you should be avoiding playback at 0dB on the computer, regardless of whether or not they're clipping this test signal.
JRiver's Volume Leveling feature takes this inter-sample clipping into account, but I don't know of any other players which do. (and JRiver won't prevent inter-sample clipping without Volume Leveling or Peak Level Normalization enabled)
I would suggest reducing the volume by at least 4dB and probably 6dB to be safe, if you are using player which does not account for this.

 

Hope this helps

Alex
 

post #3089 of 3481

The way I understand it is that intersample clipping is basically the result of an unholy alliance between music that's being recorded at 0 dBFS, the absolute highest 'legal' state possible and digital processing in the sound card or DAC that's not properly implemented and under certain conditions (very quickly changing output that's either mounting or descending close to 0) can cause 'overshoots' due to the processing not using the actual 'real' level but a projection based on the 'curve slope'.

 

Most quality DACS/sound cards are actually designed to'permit' a small overshoot of 1-2 dB without clipping the signal.  Don't ask me how as imo it flies in the face of the very definition of 0 dBFS but anyway.

 

Sloppy recording in the loudness wars and too much and badly implemented digital wizardry will get you any time.  I stick to classical, better dynamic range, no such issues, oversampling and filters 'off' on my DAC and everything into my tubes where the distortion at least sounds nice (kidding, I do listen to ss too).

post #3090 of 3481
Thanks for the input guys. I've dropped the max volume by 6dB just in case. To be honest I rarely listen to the kind of recordings likely to cause problems but I guess it's better to err on the side of caution.
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