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Everything posted by pkane2001

  1. John, you keep saying static EQ doesn't make sense with dynamic processing, which is true. Nevertheless, the decoder produces a consistent static frequency change across all the processed tracks. This can't be the effect of dynamic processing, since that would change based on the signal and wouldn't be the same across two different recordings. For whatever reason, there's a static frequency curve that appears to be applied to all decoded content. That's what Klaus has been reporting, and that's what I've been saying to you for a while, even in our previous private conversations. Although the c
  2. Yes, linear phase correction in DW if phase is not selected, so you're right, if minimum phase filters were used this will not undo them fully. Engaging phase correction will, though, but I think this will just negate all the main effects of the decoder :) I did listen to a couple of tracks, RAW, Decoded, and de-EQed by DW. I think the latest decoder does a better job with frequency balance, although to me, there's an overemphasis on upper-mid frequencies. Vocals sound brighter than they should, a bit more sibilance. Not something unexpected if you look at the frequency bump from
  3. Klaus, not sure if you've tried it, but this is what I do to de-EQ John's files. I simply use DeltaWave to match the RAW and Decoded file, and use non-linear level EQ correction only (uncheck phase). The result is to undo the large-level EQ in processed files. You can then play or export the corrected files. Here's an example: Uncorrected EQ (RAW vs. DEC): After DeltaWave frequency correction: But, I'm thinking this will make the decoded files sound a lot more like their RAW versions. At least it does to me when listening to these.
  4. I've interacted with Klaus on forums and privately for a while now, and have a good sense of his technical ability and skills. He's not a stalker or a troll, he has deep background in audio and understands signal processing and analysis. While we don't always agree on all the technical points, he's never been disrespectful or acted in any way other than being completely professional in our interactions.
  5. That is a mischaracterization of @KSTR and it is really unfortunate that his technical and subjective reports are being actively erased.
  6. Try a similar thread on ASR. I’ve missed the posts here, but @KSTR did post his listening notes and analysis there ... which also didn’t end well.
  7. Frank must be glueing the optical cable in, to make sure it makes the best possible contact. Right Frank? That would make it difficult to swap ;)
  8. The New2Dec sounds slightly better, it's moving in the right direction. Just to be clear: I didn't create the EQ by ear -- these were done to bring the decoded frequency response more in-line with what was on the vinyl clip. I agree that average spectrum isn't the greatest way to do it, but it can reveal systematic frequency errors and so can help pinpoint issues. Here's the comparison between the three clips (red=New2Dec, blue=NewDec, green=vinyl): What I hear on both, NewDec and New2Dec is not enough clarity around the vocals. The voice sounds a bit muffled and
  9. John, out of curiosity I applied an EQ to JustOneLook-NewDEC to shape the spectrum more like the vinyl version. I like it better this way, but maybe it's just me. If interested give this parametric EQ a try: Peak F=60Hz, Gain=-5dB, Q=2.0 Peak F=400Hz, Gain=-2.5dB, Q=0.5 Peak F=2kHz. Gain= 3dB, Q=0.6 HS F=7KHz, Gain=-12dB Green line is the spectrum of the vinyl clip, blue is JustOneLook-NewDEC with the above EQ:
  10. Here's the spectrum difference between old EQ and new:
  11. RIP, Alex! You'll be missed. I may have to start marking my own posts with a 'D', now that you are not going to do it.
  12. I know... it's yet another release... v1.0.64 The two notable changes are: Change: Added a View->Chart Options->Linearity index to 0 menu option to turn on/off linearity plot normalization (on by default) Added: FFT Scrubber processing and plot window with audio scrubbing The second change is a way to view spectrum details using a scrubber control and short FFTs over time. This is almost like an unrolled spectrogram, except in 2-D, where you control which time interval you want to view. You can see only one interval at a time:
  13. The tiny/noise-swamped sound of the music after a null, below -100dB, is usually due to the least-significant bit differences introduced by dither or noise shaping. These are absolutely inaudible with the main program material, because they are the same frequency and phase as the main recording. It's not that this signal is correlated -- it is the same signal, just at a very low level with some intentional noise mixed in. Oh, and I agree, don't use the Non-linear EQ if you are trying to listen to very fine differences, as this corrects for phase and frequency errors, including some
  14. Updated version 1.0.63 addresses a few issues reported in .62: Fix: Settings window toolbar can overlap text below under certain DPI settings Fix: Exception and stop processing when linearity plot contains too few samples in the lower few bits Fix: When subsample correction is enabled but drift correction is turned off, small delays below 1/1000000 of a sample might still be processed and applied, unnecessarily. These were properly ignored when drift correction was enabled
  15. I actually use an Apogee Element not Antelope for measurements as an ADC. Should have an RME ADI-2 Pro here by the end of the week :) Let me see what I can do in the meantime. Apogee has a Thunderbolt interface and doesn't work with Windows, but I can use another USB DAC that does and record with Apogee.
  16. I don't recall your explanation for what a blinking cursor does to cause audible changes at the output. If it's not CPU activity, then it must be, what? Graphics card or monitor generating EMF/RFI, like @yamamoto2002 suggested?
  17. Maybe I misunderstand something... don't you believe in dither and analog noise? These can't be removed from an analog signal, so 100% the same isn't possible with analog, regardless of what you do. But I can tell the difference between random analog noise and a pattern signal, such as might be generated by a blinking cursor or a CPU stress utility turning on and off.
  18. I'm trying to understand and duplicate your results, Peter. The only claim I'm making is that it is not hard to see the level of detail you posted with modern tools. For example, I've run a stress-test utility while playing audio through a DAC and compared the result to a capture where the PC is not so stressed. The result? No measurable or observable difference, very similar to the null I posted earlier in both cases. What does that prove? Nothing, really, except that my DAC, PC, USB whatever, isn't as susceptible to PC activity. I can't generalize this to other systems and DACs,
  19. Peter, I'd love to see the same type of a recording, with a modern DAC with a decent USB implementation. Nothing fancy, just a decent implementation. I see a lot of examples where the PC activity is reflected at the output of a DAC in case of a ground loop or just a poor differential USB implementation or a bad cable. I've yet to see or measure anything like what you're showing. (And yes, DeltaWave allows you to zoom in to a level well below the resolution that you show, and also display in dB or linear units). Here's an actual null from a DAC, zoomed in:
  20. I tried various kinds and amounts of jitter added to music tracks by DISTORT. With music I usually can't tell the difference until something in the neighborhood of 50ns+ peak-to-peak. For a sinewave modulated sample clock, that turns out to create -70dB sidebands (around 3kHz, where my hearing is most sensitive). With simple test signals, jitter is easier to hear at lower levels, but I mostly tested with music. But it's not very interesting to hear what I can or can't hear. The point of DISTORT was that others can test it for themselves. Julian Dunn published a graph of audibility
  21. They seem to frequently plagiarize my posts on AS. What's up with that? ;)
  22. Modulating signal magnitude needed to alter zero crossing of a 44.1k sine wave by 1ns? Doing this mostly in my head, as I'm typing. Hopefully not too far off! The error in magnitude should be something like: Δv = 2 * π * F * Δt Assuming Δt = 1ns and F = 44100: Δv = 2 * π * 44100 * 1e-9 = 0.00028v (assuming a maximum amplitude of 1V, peak-to-peak)
  23. Phase is not something mysterious, Jud. It's easy to measure using FFT. Any software or filter that alters it to a significant degree might create audible issues. DeltaWave computes phase differences between two recordings, so you can examine it and even correct for it. Phase noise and modulation of the DAC clock results in jitter in the time domain. This is also measurable, and can become audible when excessive.
  24. Mani, that's pretty much what I said to Peter on the other thread when he suggested a blind test using microphone recordings. The variability of recordings with a microphone is large enough, so that the detection of one minor difference becomes near impossible. More than that, only you know what this difference sounds like on your system.
  25. That's been done with equipment. A test where a randomly selected component (of two) was put into the circuit each time the system was turned on. The listener didn't know which one it was. The test circuit also tracked how much time the listener spent with each component over a period of a few months. The expectation was that a better sounding component would make it easier to listen to, and the listener would spend more time with it. In the end, the result wasn't very conclusive. I posted on this test before, but will need to locate the description for more of the details. It wasn't a true ex
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