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6th March 2015, 05:32 AM  #311 
diyAudio Member

That isn't a measurement.
That is an extremely subjective opinion dressed with copious amounts of expectation bias and seasoned to taste with tube induced third harmonics... Last edited by spzzzzkt; 6th March 2015 at 05:35 AM. 
6th March 2015, 10:03 AM  #312 
diyAudio Member
Join Date: Oct 2010

Could you provide me wav of those signals? I can measure in oscilloscope or Tektronix am700. Or describe these signals so that I make with Audition.
Square wave clipping is third harmonic distorsión. I have seen different third harmonic distorsión in different filters. I has made measurements with fuzzmeasure all filters. I can put frequency response, phase, impulse, impulse in dB and distortion of each filter with an analog measure on my audio card. I think interesting to see distortion measurements. Varies with each filter, Some filters saturated because they increase about 1020 db distortion of third harmonic in certain frequency bands. Also interesting impulse responses. More interesting impulse response in dB that shows the pre and pos ringing duration in milliseconds. There are many graphs that occupy much space on the thread. Probably 80 graphs. I dont know if you interested in these graphics in thread or would be better to make a pdf document Web hosted downloadable. Can anyone give me an opinion. I could put it all this weekend, it takes me time to capture all those screens. 
6th March 2015, 10:13 AM  #313  
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Join Date: Oct 2010

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6th March 2015, 04:07 PM  #314  
diyAudio Member
Join Date: Dec 2014

Quote:
But there is no real need for measurement, I think. The done measurements (of the worst case) shows that the signal looks as predicted by the theory (within the limits of my equipment), especiallay there is no hardware induced clipping and the only hardware effect is the final analog filter (so we can calculate shape of the output signal sample per sample if needed). ... That is if we assume there are no software impementaion faluts and that we stay in the computaional savety overhead of the "34 bits" with the intermediate signal levels. In this case we do not even to simuate the output, but can simulate the filter in the digital domeain only, to decide if there is clipping. This can perhaps be done with sox (I think) as you did. Or "by hand", especially for simple test signals. Assume the clipping free signal level is 1 to 1. Let s[t] be the samples of your test signal, f[i] your filter taps (already multiplied by the multiplier or gain), then the output o[t] is the convolution of s and f. As example the square wave First without oversampling (zero insertion) 1.) A square wave of amplitude 1, longer than the filter: So for entering the wave we are in the situation s[t]=0 for t<=0; s[t]=1, t>0 The output at time t is the sum of the first t filter taps f[i], i.e o[t]= Sum_{i=1}^t f[i] For exiting the square wave we are in the situation s[t]=1 for t<=0; s[t]=0, t>0 The output at time t is the sum of the all but the first t filter taps f[i], i.e o[t]= Sum_{i=t+1}^l f[i]. Conclusion the filter does not clip with a square wave of amplitude 1, longer than the filter, if neither the absolute value of the sum of the first k taps nor the last k taps exceeds 1 (for all k from 1 to the filter length). For shorter square waves (with long runs of zero levels betweens the squares), the condition would be that no sum of any consecutive filter taps is greater 1 or smaller 1. With 8x oversampling we have as new signal s[]= z[0],0,0,0,0,0,0,0,z[1],0,0,0,0,0,0,0,z[2].... where z[] is the original signal. Doing analog to the above reasoning, we see we get now conditions, each on filter taps with offset 8. i.e. no sum f[i]+ f[i+8]+ ... + f[i+8j] for some i,j, is greater 1 or smaller 1. We can handele FIR1 + FIR 2 similarly. I have not yet given a thought what happens with an intermediate IIR filter. As I mentioned elsewhere, the condition for that a filter can not clip with ANY signal is that the sum of the absolute values of the filter taps is at most 1. With 8x oversampling the condition that it can not clip with ANY signal is that non of the eight sums abs(f[0])+ abs(f[8])+ ... abs(f[1])+ abs(f[1+8])+ ... ... abs(f[7])+ abs(f[7+8])+ ... is greater than 1. Triangle waves should also be no problem. I can write a little script doing these computations and giving the maximum clipfree gain for these waveforms, in the next days. The ultimate goal would be a DAMsimulator for the digital output, taking also in account the fixed point arithmetic in the FPGA. To an certain extend this could be done, but at some point we would need implementation details of Soren. 

6th March 2015, 06:23 PM  #315  
diyAudio Member
Join Date: Jan 2002
Location: Cheltenham

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6th March 2015, 06:54 PM  #316  
diyAudio Member
Join Date: Dec 2014

Quote:
My first calculation was based on a wrong capacity of the capacitor which I had found in some early post of this thread. This calculation indicated that the resistance was higher as the resistance of the DAC ladder alone. This would have meant, that the power supply (i.e. the ensamble of the final voltage regulator and the according bypasses and the shift register), feeding the DAC ladder, had a non neglectable internal resistance. If that would have been the case then it could have been useful to reduce this extra resistance to improve the impulse response of the DAC. Measuring also the capacity showed that the assumed value was wrong and the internal resistance of the DAC is essentially only the value coming from the DAC ladder, so there is nothing to improve (resistance wise) with the power supply. Last edited by zfe; 6th March 2015 at 07:20 PM. 

6th March 2015, 08:45 PM  #317 
diyAudio Member
Join Date: Oct 2010

DAM1021 Filters
Measurements made to DAM1021 DAC with Motu Ultralite MK3. Fuzzmeasure program.
Last edited by oneoclock; 6th March 2015 at 09:00 PM. 
6th March 2015, 08:48 PM  #318  
diyAudio Member

Quote:
You can hardly describe something as a perfect filter when it does no filtering. 

6th March 2015, 08:49 PM  #319 
diyAudio Member

These are done using the impulse test function of Fuzzmeasure?
What are the traces in the HD charts? Frequency response is fairly obvious but what about the two around 100dB range? Last edited by spzzzzkt; 6th March 2015 at 08:57 PM. 
6th March 2015, 08:54 PM  #320 
diyAudio Member
Join Date: Oct 2010

I refer to the different practical measures I have place in the PDF. NOS concentrating more power in less milliseconds.

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