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About esldude

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  1. It appears the 16/44 has the higher treble. Apodizing filters usually, though not necessarily will result in a subtle treble roll off.
  2. What it looks like to me is noise. Remember how we get these FR charts using music. If the music is very low in level then noise begins to effect the FR charts derived from comparing bins in the FFT. In these 2L tracks being discussed, the levels above 10 khz are way down in level. There isn't much there. So what looks like a FR difference up there is really just more noise in the 16 bit track than the hirez 24 bit track. That was my deleted post last night. I'd looked in Deltawave, and thought it had a mild frequency contour around or over 1 db in the treble. Plenty to be heard as different. Then I realized it was the signal being so close to the noise floor you were seeing mainly a noise difference. So I doubt at those low levels you'll hear the noise alone among music. So I deleted my post about FR.
  3. So do George and I need to have a high resolution "record off" now? Hahahahaha. I haven't claimed to be a great recordist. In fact what surprised me, is once you learn the basics and do get a little experience, just how easy it is to get nice recordings. Key being, decent mikes in the right place, a good room or space and good musicians. You get those and there isn't much to it other than having enough sense to not process it to death. Now when you get to multi-tracking, difficult rooms, bunches of retakes and processing well I'm able to do okay(maybe), but experienced people do much better than me. OTOH, experienced people also turn out work much worse than mine much too often. The how to do it isn't really the main part of what makes commercial recordings sound like they do. There are all these other reasons. And many of the things done means you have no chance of getting the recording to reproduce like the real thing. It isn't on the recording at least 99% of the time.
  4. I've done it at 44 and 48 with 24 bits. And at 88, 96, and 192 rates. Neither I nor the musicians had a preference. The two musicians with the best ears, a young girl and a middle age lady who also has a masters in music when asked to compare listened. Listened some more. And one said, "so what is supposed to be the difference?" I'd asked them to listen to some different versions and tell me which was better. When I told them the sample rates were different, the lady said, "Oh, I thought you changed something else. I've wondered what difference that would make. I've been told it would sound better the more you sample it." Shrug. So I've mostly stuck with 48/24. I'm sure my gear was no good, my mike was wrong, none of our ears were adequate, the moon was out of phase, it was recorded below ground and played back above ground, playing by musicians when facing north and listening was facing southwest, a cold front was coming thru, global warming is causing many issues etc. etc. etc. ................
  5. That must be why you see people say the time and frequency domains are opposite sides of the same coin. Hmmmmm.
  6. https://dspguru.com/dsp/howtos/how-to-interpolate-in-time-domain-by-zero-padding-in-frequency-domain/
  7. Can we get a MQA and/or Tidal Deathwatch thread going. Whats the over/under about 6 months?
  8. Hint: entirely realistic audio is not going to result from going to higher sample rate.
  9. This whatever it is that makes high sample rate better sure is well hidden.
  10. I once had the 3bx version. I wonder how useful it might be today. I also had another similar unit made by Pioneer. The DBX was better, but I really didn't use it much. I'd think software dynamic range expansion would be better. Then again, if you don't know the settings of the compressor in use you have little chance of making it sound really right.
  11. I don't know that I'm any authority to confirm this. But yes if an instrument or other sound caused intermodulation into the audible band, microphones will record it. They have no way to separate intermodulation produced sound from any other sound. If someone has done some mixing and processing that altered the sound and the reason is intermodulation, they couldn't hear that unless it was in the below 20 khz sound. 20 khz will record it. And think about this. Suppose the sound levels of some ultrasonics are so high that the non-linearity of the air causes intermodulation into lower frequencies. And you record it with wideband microphones and reproduce it with wide band speakers capable of playing it back at levels high enough to intermodulate with the air below 20 khz. You've just doubled the audible intermodulation generated tones vs were you there to hear it live. In direct comparisons it may have more air, space, growl or zip than 20 khz reproduction only. But it actually is a distortion to produce those frequencies. You could only hear 20 khz if you were there live. A British company some 15 or so years ago promised to make speakers on the principle. I think it was going to run ultrasonic emitters at 60 khz and vary them so it would intermodulate thru the air and produce sound in the below 20 khz band. It apparently had hurdles that couldn't be fully over come. Dr. Joseph Pompei once of MIT and now of Audio Spotlight has worked on this. The advantage is you could fire two emitters at a distant location and have the beams intermodulate or demodulate into the audible band at a distant location. You could make it sound as if the sound were right beside someone at a distance. For such to work at all sound levels have to be above 100 db SPL for the air to be non-linear enough. https://www.holosonics.com/product-brochure-pdf https://engineering.dartmouth.edu/events/the-audio-spotlight-beams-of-sound-from-ultrasound/
  12. It has been awhile since I looked at those. And depends upon which recordings. I can look perhaps and get back to it later. Okay happened to have this one on file. 2L38. 192/24 minus 44.1/16. The background here is set to go to light gray at -80 dbFS. Light blue is the first color just above that. Very little there. I'm showing the left channel as it had slightly more content left than the right channel. Here is one where I took one channel and implemented a brickwall filter at 22 khz in the upper channel, and left the lower channel alone. Same settings goes to light gray at -80 dbFS. A little more, but not much there above -80 db FS. The RMS value for what is left is - 69 db.
  13. I know this is asking a lot Mani. Could you repeat both captures at least once or twice? Would be good to know how consistent the captures are. And you could listen and see if the same capture twice or thrice all sound the same at least to each other. And offering those re-captures for us to download too.
  14. So peak levels in that are maybe -80 db FS in the 20-25 khz range. So if your 0 dbFS level plays back at 120 db SPL, these ultrasonics would be 40 dbSPL. Considering hearing around 20 khz if you have it has a threshold of about 100 dbSPL, I'm going out on a limb here and saying that isn't going to be heard. This is before we consider masking by louder lower frequencies too.
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