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Old 15th February 2012, 06:21 AM   #1121
1audio is offline 1audio  United States
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Will the driver work with other UAC2 devices like the Wavelength Wavelink? What info needs to be provided to make this possible?
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Old 15th February 2012, 06:27 AM   #1122
nikkov is offline nikkov  Russian Federation
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Quote:
Originally Posted by 1audio View Post
Will the driver work with other UAC2 devices like the Wavelength Wavelink? What info needs to be provided to make this possible?
As far I know Wavelength Wavelink works with Thesycon driver. This driver already includes ASIO support.
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Old 15th February 2012, 04:38 PM   #1123
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re: pcm5102, I'm seeing some very annoying and nasty clipping going on when the spdif (translated to i2s input of course) hits 0db. sadly, with 'internet radio' running full at clipping on many stations, you'll see the red clip indicator going off very often, sometimes even regularly on any given song.

some dacs seem to tolerate this better than others.

sadly, I'm seeing the 5102 NOT being very graceful in this manner ;(

I first thought there was a bass clip problem or psu issue but I'm more and more convniced this dac chip does not tolerate '0xff' (so to speak) as its max binary input.

send in normal spdif, from a normal (old style) cd and things sound fine. but as you get the normalized signals from modern cd's and 'radio', this harsh clipping in the dac is going to be an issue.

anyone else witness this on the 5102 chip?

I have a dcx2496 and its known that its lights come up (yellow and red) when its close to 0, not even at 0. and when I see the red or even yellow come on on my dcx, the pcm5102 is clipping *badly* at that point. my other dac (the dc2496, itself) does not sound harsh and gracefully deals a lot better with this near-zero and at-zero value problem.

there's a higher end wolfson dac chip that also has a switch to deal with this (anti-clipping switch). I never really understood why it was there, but maybe I'm getting some insight, now, on this problem.

yes, the stream is at fault, but I'm not sure I'm into using a dac that only works on 'spare headroom' streams.

can anyone else check to see if your 5102 also has this behavior?
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Old 15th February 2012, 06:51 PM   #1124
borges is offline borges  Norway
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Hi,

what you are probably seeing is Gibbs phenomenon - Wikipedia, the free encyclopedia. The peaks of the impulse response add up so that the output of the digital filter inside the DAC chip is larger than what the analog part of the DAC can handle.

With the ES9023 this is particularly bad. Give it a 0dB square input, and even the internal counters will overflow. I.e. 0x7E + 0x02 = 0x81. 0x7E being a valid signal near full-scale positive, 0x02 being a contribution from the impulse response, and 0x81 being near full-scale negative.

Are you seeing phase-reversal like this or mere clipping? Try a software signal generator for a 0dB square wave or the .wav file export in Octave.


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Old 15th February 2012, 07:58 PM   #1125
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interesting! that sounds like it could be it.

this is all by ear, not on a scope yet; but I hear a popping that I thought was in the source; but turns out on my other dacs it plays fine. when I get a chance I'll try to put a software wave into it and see what comes out.

but at this point, I'm a bit sour on this dac chip ;( I just can't 'demand' that people put so-called proper numbers into its digital inputs. the more they compress and normalize (on radio or cd or even file!) the more we are going to see this, I guess.

if this is an internal numbers issue, then no amount of power supply 'strength' is going to help me here, is it? I was thinking maybe the charge pump was being overtaxed but you think its internal value overflows? yikes. that's unsolvable then, isn't it? at least at the chip-user level.
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Old 15th February 2012, 08:03 PM   #1126
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oh, now it all makes sense, of course.

the clipping was extra bad when I played my Bee Gees CD.

(grin)
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Old 15th February 2012, 09:31 PM   #1127
rsdio is offline rsdio  United States
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Since digital audio is a serial stream, is it possible that Bryan is seeing the results of the audio samples being shifted too far left? It would certainly caused nasty sounds if the MSB were being shifted into the bit bucket and the next bits used as the sample.
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Old 15th February 2012, 09:37 PM   #1128
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are you thinking I have the 'AIF' interface between the spdif receiver and the dac set wrong? I picked 24bit i2s which I thought would be correct. if 16bits comes in, 24 bits are still sent (?) but just padded with zero, right?

if things were shifted even 1 bit, I'd hear overflows on everything, wouldn't I? ie, half volume would be max volume and every single cd today has averages well over half-scale sample values.
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Old 15th February 2012, 09:44 PM   #1129
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close-up of how I wired things (not sure if this is enough detail or not, I can reshoot again if needed):

Click the image to open in full size.

the settings on the wolfson are: MS=1 (master mode), txsrc=0, aifconf0=1, aifconf1=0. that should set it to 24bit i2s mode if I read the specs correctly.

note that I did leave master clock off and only wired 3 i2s wires. I doubt that giving that master clock would affect this (?) but I could add that 4th wire if its worth doing and retesting.
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Old 15th February 2012, 09:46 PM   #1130
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Quote:
Originally Posted by rsdio View Post
Since digital audio is a serial stream, is it possible that Bryan is seeing the results of the audio samples being shifted too far left? It would certainly caused nasty sounds if the MSB were being shifted into the bit bucket and the next bits used as the sample.
hang on - just to clarify, I'm NOT driving the pcm5102 from the usb dac and hearing this problem (per se). I have a standalone dac with wolfson 8804 receiver driving the pcm dac chip. the 8804 is in hardware mode (foolproof for me, lol) and I know its i2s is not funky or due to any software or driver issues.

sorry if that was not clear. this has nothing to do with the usb i2s stuff; its a pcm5102 problem.
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