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Old 3rd January 2012, 01:32 AM   #81
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You do "pulse height control" in a way that is equal to analog amplifier. You will have the finals open only partially and disipate heat like any AB class.

Also I thought that you have solved the PCM-DSD part. Wasn't that what you where after at first?
And yes, straight DSD is not easy to control in volume.

Last edited by SoNic_real_one; 3rd January 2012 at 01:41 AM.
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Old 3rd January 2012, 02:27 AM   #82
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Pulse height control is..
DSD(1/0)*(V1*C1+V2*C2+V3*C3+V4*C4+V5*C5+V6*C6)
C1-C6 is 1/0 control from PIC(ADC), does not changes while volume is not touched.
V1=1/2Vcc, V2=1/4Vcc, V3=1/8Vcc...
So this is not related to audio frequency, only effects DSD frequency.

DSD amplifier can be un-linear (but to be controllable), for example, +-0.1V pulse in -> +-0.5V final out and +-0.15V pulse in -> +-0.77V final out, is terrible distortion in analog amplifier. but no problem in DSD amplifier.

Before LPF, there are no analog signal, only DSD pulse. attached LPF output from single DSD pulse - it lasts some msecs.
Then thousands of this impulse response will be summarized to final analog(audio range) signal.
So DSD amplifier don't have to work like analog amplifier, just should be fast enough to process DSD.
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Old 16th January 2012, 11:21 AM   #83
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http://www.essex.ac.uk/csee/research...0amplifier.pdf
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Old 16th January 2012, 02:45 PM   #84
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Quote:
Originally Posted by KOON3876 View Post
Before LPF, there are no analog signal, only DSD pulse.
I will try one more time: If you modulate the amplitude of the pulses, that is not digital signal anymore. The transistors will "open" partially, therefore will not be just "switching" ON-OFF. Disipation will be several orders higher.
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Old 16th January 2012, 08:59 PM   #85
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I don't care partial or full...
This amplifier does not require Amplitude Linearity.
If this amplifier has terrible 2nd distortion like below (actually not)
input +-0.5V -> output +-2.5V
input +-0.8V -> output +-3.6V
Still signal result (after LPF) is clean.
This amplifier is built as 'high-speed only' analog amplifier, don't care about NFB / current mirror / constant current source-sink etc.
but this amp is not working in analog area.
Please make your one and look on Oscilloscope, what's happening.
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Old 16th January 2012, 09:53 PM   #86
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I don't need, I know what happens
Sure, you don't have to worry about the zero-crossing distortions, but you will still have variable time-switching due to the fact that you don't saturate the transistors. And the heat issues. All that will result in some level of distortion IMO.
Did you measure it at lower levels of volume (like 10%)?
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Old 1st May 2012, 11:18 AM   #87
Bunpei is offline Bunpei  Japan
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Quote:
Originally Posted by SoNic_real_one View Post
Hey, I have one ideea - why stick to 2.8224 MHz? Let the 48 and 96 generate the higher signal (x64 or x32 give 3.072 MHz) - you are not using a regular DAC anyway after that, so it will not be a problem!
Recently I have been completely absorbed in playing DSD256 sources.
I felt quite a "break through" in the DSD256 sounds obtained by ES9018 DAC.
(In the case of DSD128, the resulting sounds are not so much impressive. However, DSD256 has a great leap! Koon, why don't you try DSD256 in your system?)

Of course, native recordings in DSD256 are very rare. I have just only one source. I convert and upsample PCM and DSD of lower sampling frequencies into DSD256 using Korg AudioGate program. Surprisingly all the sources sound better on DSD256.

While converting 48kHz, 96kHz, 192kHz PCM sources into DSD128(5.6MHz), I remembered SoNic_real_one's comment quoted above.
It must be quite natural to have the 48kHz series DSD when converting original PCM sources of 48kHz series.

I use Chiaki's SDTrans384 for DSD256 play. I asked him whether implementing DSD of 48 kHz series was possible or not. He said "Yes".
I think DSD64,128,256 for 48kHz is feasible in the combination of SDTrans384 and ES9018 and I'm going to try them under Chikaki's help.

Bunpei
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Old 1st May 2012, 10:02 PM   #88
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If ES9018 can lock on those signals 48k-multiple, why not?
ES9018 upsamples too the PCM and DSD inside (like any other sigma-delta DAC). Final internal result is probably 6 bit DSD128 applied to actual sigma-delta converters.
I think that improved quality comes from the fact that you are doing the upsampling/interpolation outside the DAC, with a better algorithm that the one inside ES9018.
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Old 3rd May 2012, 07:40 AM   #89
Bunpei is offline Bunpei  Japan
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Quote:
Originally Posted by SoNic_real_one View Post
ES9018 upsamples too the PCM and DSD inside (like any other sigma-delta DAC). Final internal result is probably 6 bit DSD128 applied to actual sigma-delta converters.
I think that improved quality comes from the fact that you are doing the upsampling/interpolation outside the DAC, with a better algorithm that the one inside ES9018.
I agree with you. In the DAC chip, the final DAC stage is a 6 bit DAC for delta-sigma modulation. (The conversion frequency is not disclosed, I think)
All the digital input signals, regardless PCM or DSD, are internally to be converted into the same 6 bit delta-sigma modulation format.
However, according to my experience, a pre-processed DSD256 input brings the best result.
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Old 9th May 2012, 02:27 PM   #90
Bunpei is offline Bunpei  Japan
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In addition to the usual 44.1 kHz series DSD sources; DSD64(2.8MHz), DSD128(5.6MHz), DSD256(11.2MHz) & DSD512(22.5MHz), I could play DSD sources of 48 kHz series; DSD64(3.1MHz), DSD128(6.1MHz) & DSD256(12.3MHz) with the combination of ExaU2I and Fidelix CAPRICE (a Japanese ES9018-based DAC) using "synchronous master clocking method".

Bunpei
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