5

There are two issues here. Both claim to be 44.1k FLAC files, yet the first graph clearly goes over 22KHz... which is, if not impossible, rather unlikely. Technically, it could all smooth out before it reaches the limit at 22.05, or it could all brick-wall, but we can't see that from the chart. You cannot judge anything except very basic frequency content ...


3

You can use Coagula. I believe it is the same software Aphex Twin used in Windowlicker.


2

Total phase cancellation [which is what you appear to be looking for] only works in mono. Each time one channel has a positive voltage, the other has an identical negative voltage. When summed to mono, this equals a constant 0 volts, i.e. silence. In stereo you can still hear both signals, because in open air there is a time-difference between the channels ...


2

Do you know what mp3 codec is being used? Frauhenhoffer/lame etc all use different techniques to compress the audio. For example, the Lame VBR, even at its highest setting V0 will cut off anything above 20KHz and use the bandwidth saved to keep more of the sounds you can actually hear. (Remember most human hearing only goes up to 16KHz) The non VBR Lame ...


2

The upper one is definitey the better one. Spectrograms (like the ones you produced with Spek) are used to display and inspect tendencies over time. For example if some noisy signal is present all the time, it will show up as a straight horizontal line somewhere: You can determine some general properties, like that 20 Khz cut, but an average frequency ...


1

There's also this project with some examples http://arss.sourceforge.net/examples.shtml


1

I've been looking at this for a couple of days and it just came to me. You could subtract the two waveforms and get the spectrogram of the difference. In theory it should be the same as what you asked for. To do this you need to invert one of the tracks and mix(add) them in equal volume. If you did that with a single waveform and its inverse you should get ...


1

lol, those "strange lines" (horizontal) are harmonics; and because this spectrogram is scaled linearly, they're evenly spaced; if it was scaled logarithmically, it would closer represent what we hear. ATM, all of the detail is crunched down at the bottom, down past 6 kHz, which we actually hear as quite a high frequency. I don't know about the crackling ...


1

A few guesses: MetaSynth Virtual ANS HighC


1

MP3 encoding employs psychoacoustic techniques like masking (in both frequency and time domain) to reduce the complexity of encoded audio whilst not perceptually sacrificing quality. What you're seeing is the visual representation of quantised high frequency content not being encoded to save on arguably unneeded bits. The encoder model decides this, and it'...


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