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pkane2001

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About pkane2001

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  1. A pre-release of v1.0.19. This should address a lot of the issues that @fas42 and @esldude have reported, as well as adds a few new features and enhancements. Let me know what you think: https://drive.google.com/open?id=1STW_HC903Lq8x0JKhDfUBkEFCay8fQ1L Should get better nulls with Audacity-processed 16-bit files Should read 24- and 32-bit integer WAV format correctly Should display correct offset value Adds silence trimming option (on by default) Adds scale display and other enhancements in the spectrogram windows Changes the waveform Y axis to display in dB rather than 1 to -1 floating point Fixes the 100% that's really a 'nearly 100%' display Regards, -Paul
  2. Thanks, Dennis. I got your multi-generation 32-bit floating point files and they load fine in DW. I also see that 24-bit and 32-bit integer sample files do not, and I know why. It'll be fixed ASAP. If you do have a file or two that are 32-bit floating point and fail to load, I'd like to see those.
  3. I'll check again on 32-bit floating, but that's a format I've been using extensively during testing. It's not the same as 32-bit signed PCM -- that's an integer format, not floating point.
  4. Thanks, guys. As I said, 32-bit floating point format has been working, it's 32-bit/24-bit integer formats. I've tracked down a problem and will have a fix ASAP. Meanwhile, I'm following up on the tests that Frank reported to try to understand what could've caused them. The likely contender for the lower than expected null for 16-bit files appears to be the use of overlap-add and the choice of the FFT window. This is similar to what I had fixed before in other parts of the software, but appears to behave slightly differently, so I'll need more time to investigate.
  5. OK, Frank, so how did you do this? Here's what I get in Audacity when I try to repeat your steps. Please also share your Audacity Quality settings during processing, as that's the only difference I can think of between our two processes: 1. Load original Bob Marley B track (16/44) 2. Amplify by -0.38dB 3. Export as 16 bit PCM (16/44) 4. Load both the original and the exported files from disk 5. Amplify the modified file by 0.38dB, invert it 6. Mix the inverted file with the original. Here's what I get: Note the RMS value of the null track: -79.3dBFS. When I run the same two files through DeltaWave, the result is as follows: Gain= -0.3843dB (0.9567x) Phase offset=0ms (0 samples) Difference (rms) = -89.52dB [-96.63dBA] Correlated Null Depth=87.98dB [95.99dBA] So, how did you get -95.9dB in Audacity?
  6. Never mind. It was a 32-bit signed PCM rather than floating point that caused this. Try saving as 32-bit floating point, instead, as that works. I'll investigate why this audio library can't handle 32 bit integer samples, but I have a suspicion...
  7. 770dB is a very loud track, don’t try to play it!
  8. Can you please upload a snippet of this file? Somehow I can't reproduce it when I save to 32-bit floating point in Audacity.
  9. It's not bit perfect, but 99.99999% of the samples match. Round it to a whole number and you get 100%. Don't have the room to display all the decimal places on the status bar, but maybe I'll stop rounding and truncate instead, then it'll show up as 99%, which is less accurate but maybe a bit less misleading?
  10. No, 32 bit float and it's been working for me from the beginning That's the result I posted earlier. In Audacity it looks like this when exporting: If you have a short snipped of a file that gives such a large dB peak, can you please share it? Somehow these are either not recognized or processed properly by the audio library I'm using if it's giving 700dB peak
  11. Frank, you keep running into Audacity issues. To avoid any kind of truncation or dithering, please use the following settings when processing and exporting from Audacity: Then, when exporting, use WAV 32-bit floating PCM format. Here's what I get when I do that with the -0.38dB processed file (note the nulls). A little better? Note that it shows NOT BIT PERFECT but 100% of the samples matched. That's because some small fraction of one percent of samples didn't match due to the level change:
  12. Frank, I like that you are testing these different scenarios, but I'm not clear what it is you are trying to achieve. Can you describe what it is you are trying to get to? What nulls are you getting, and with which files? One thing to remember: when you upscale you are not adding any additional information to the waveform. It's a form of averaging and interpolation that, in effect, produces some smoothing of the data by inserting artificially computed values (new samples). The null value is computed as an average (RMS for difference, mean for correlated null). What that means is if you add more values that are 'smoothed' or averaged in between the real ones, you are increasing the number of matching samples, but they are not real. So it's not very surprising that null results improve with upscaled data -- the smoothed, interpolated data is skewing them. By the way, that's why downsampling is the default in DeltaWave for matching sampling rates -- upsampling has an impact on the resulting match.
  13. Gents, I like the idea of doing your own testing and not relying on someone else to tell you that it sounds better. That's one of the reasons I developed DeltaWave. The comparator in DW is one tool where I intend to keep adding functionality and new ideas to improve on audio evaluation techniques. The tool supports the two-channel simultaneous comparison that Jud is proposing here, as well as standard ABX test, and a subjective preference test. What's more the tool allows any of the tests to be run in 'learning' mode where you can see and tell what track is playing, and to then repeat the test in blind mode, where the tracks are randomized. DW also creates a signed report of your blind test, including hashes of the files and all processing DW itself did to match them, so these can be validated by others. You can do a sequenced A/B test, A/B/X test, a simultaneous Stereo X-Y test, or a sequential preference test and get the resulting statistics that demonstrate that your result was not produced by guessing or random choice. DeltaWave is still under development and probably will be for a long time (I like to tinker) but basic functions are there for anyone to try. Regards, -Paul
  14. If you play it from the Play menu, then yes. If you use the Comparator Stereo XY mode, you can switch left and right through the menu Swap Reference and Comparison. The results may be different when swapping Reference and Comparison files on the main screen and doing another match. That's because the algorithm will adjust everything, including level to the other file. It's possible that the result will not sound the same, even if because of a slightly different level. They will be similar, but not exactly the same. Best to use the comparator to swap.
  15. I haven't been following MQA discussions closely, so can't say. It'll be an interesting experiment. I assume you don't want to resample or otherwise modify MQA track, as any bit changes will likely result in it not being 'authenticated', right?
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