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Topic: Is it possible to listen to lower bits in an audio file? (Read 3786 times) previous topic - next topic
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Is it possible to listen to lower bits in an audio file?

Hello. I was wondering if there's a way (preferably free ::) ) to isolate lower bits, say -60 dB and down in an audio file and "lowpass" the loud part to be able to listen to what there is in the quiet part? ::)
Or is it not possible at all?
Would come in handy in evaluating the hires audio and to hear what's beneath.

Re: Is it possible to listen to lower bits in an audio file?

Reply #1
Yes, via simple subtraction.
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Re: Is it possible to listen to lower bits in an audio file?

Reply #3
No, this info is reserved only for audio elite.

Just subtract from hi-depth audio the 16bit audio that you created from hi-depth audio.
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Re: Is it possible to listen to lower bits in an audio file?

Reply #4
Just subtract from hi-depth audio the 16bit audio that you created from hi-depth audio.
That's not quite what I was asking about. I meant can you cut out the signal above certain db level and leave the rest. Not just substract a 16 bit audio difference from a 24 bit.


Re: Is it possible to listen to lower bits in an audio file?

Reply #6
So you want to isolate the fine ripples that sit on top of the large waves and a filter that intelligently decides what should be filtered out, regardless of the frequency.
That sounds like an impossible task to me. Maybe you could train an AI to try. But the result would certainly contain many harmonics of the many fundamental frequencies present in the signal.

Re: Is it possible to listen to lower bits in an audio file?

Reply #7
That's not quite what I was asking about.
Yes, this is what you were asked about.
Well, I apologize if I wasn't clear before. What I want to know is if there's a possibility "to cut out the signal above certain db level and leave the rest". By "certain db level" I mean just that, user defined db level eg -70dB, -82dB etc.

Re: Is it possible to listen to lower bits in an audio file?

Reply #8
Yes, but your ears would certainly be hurt listening to such signal. Or you mean to use something like broadband noise gate ?
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Re: Is it possible to listen to lower bits in an audio file?

Reply #9
So you want to isolate the fine ripples that sit on top of the large waves and a filter that intelligently decides what should be filtered out, regardless of the frequency.
That sounds like an impossible task to me. Maybe you could train an AI to try. But the result would certainly contain many harmonics of the many fundamental frequencies present in the signal.
I get it, thank you. "The tool doesn't exist" would suffice though.

Re: Is it possible to listen to lower bits in an audio file?

Reply #10
@mycroft  ok, back to yourpoint with substraction. Is there a tool that lets you save a 24 bit audio file into 24 bit allowing you to adjust how much to truncate from the bottom? Say I input 24 bit file and save it into a 24 bit container that only holds 21.5 bits. Substracting it from the original would give me what I want, right? But I wouldn't be restricted to only 24, 16, 8 bit steps.

Re: Is it possible to listen to lower bits in an audio file?

Reply #11
You can manipulate bits in fixed point audio whatever you like.

Just set bits you "hate" always to 0 and you are basically done.

if you set lower bits to 0 you will get lower quality, to like 8bit audio, the one without any quality dither, if you change higher bits, the most significant bits (sign bits and others) you will get unpleasant distortions.

With ffmpeg for example:

ffmpeg -i input.audio -af "aeval=round(val(ch)*256)/256:c=same" -c:a pcm_s24le output.wav

This will basically quantize to 8bit audio from whetever it previously was.

Now if you  want to hear what was removed from X-bit to give you 8-bit just do this instead:

ffmpeg -i input.audio -af "aeval=val(ch)-round(val(ch)*256)/256:c=same" -c:a pcm_s24le output.wav

I even developed filter that is able to make low-bit audio listenable again via adequant filter in librempeg.

There is no way to trim fractional bit in audio AFAIU.
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Re: Is it possible to listen to lower bits in an audio file?

Reply #12
Maybe try and describe with an example what you want to achieve without assuming a concrete technical solution.
The way you describe it doesn't make any sense.

Re: Is it possible to listen to lower bits in an audio file?

Reply #13
Thanks @mycroft, much appreciated. I'll try that, I guess.
Is there by any chance a gui for that? That command line you wrote doesn't look like something I could decypher and tweak to my liking. :-[

Re: Is it possible to listen to lower bits in an audio file?

Reply #14
@darkbyte wow, I thought it was pretty easy, I guess not)))
I want to be able to listen to the (user defined in db) lower amplitude signal in a file muting the higher amplitude one.
As I've already found out the simplest way to do it is to define and strip the specified amount of lower bits (apparently only whole bits) and then substract the difference signal from the original.

Re: Is it possible to listen to lower bits in an audio file?

Reply #15
Thanks @mycroft, much appreciated. I'll try that, I guess.
Is there by any chance a gui for that? That command line you wrote doesn't look like something I could decypher and tweak to my liking. :-[

Yes, you can use this GUI for libavfilter: https://github.com/richardpl/lavfi-preview

Probably someone made or not like VST plugin with such functionality (but have no exact name of it) thus you could use any DAW software or Audacity clones/forks.
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Re: Is it possible to listen to lower bits in an audio file?

Reply #16
@darkbyte wow, I thought it was pretty easy, I guess not)))
I want to be able to listen to the (user defined in db) lower amplitude signal in a file muting the higher amplitude one.

At each instant in time a signal has only a single amplitude, which is simply the value of each sample.  There are infinitely many ways you can take one signal and split it into two, but none are obviously going to result in "the" higher/lower amplitude signals because there is no one specific way to do that. 

Re: Is it possible to listen to lower bits in an audio file?

Reply #17
It's not going to work because every wave crosses-through zero.   Big-loud waves have small samples and and little-quiet waves have small samples.

You can "smooth" the waveform but that gives you a weird kind of constantly-changing low-pass filter.   (I don't know of a plug-in that does that.)

MP3 sort-of does the opposite...  It throws-away quiet parts that are masked (drown-out) by louder parts.     But there are phase-time shifts and some other things going-on so subtracting the MP3 from the original is only an approximation of the "quiet parts" that were thrown-away.

Re: Is it possible to listen to lower bits in an audio file?

Reply #18
Thanks @mycroft, much appreciated. I'll try that, I guess.
Is there by any chance a gui for that? That command line you wrote doesn't look like something I could decypher and tweak to my liking. :-[

Yes, you can use this GUI for libavfilter: https://github.com/richardpl/lavfi-preview

Probably someone made or not like VST plugin with such functionality (but have no exact name of it) thus you could use any DAW software or Audacity clones/forks.
Thanks so much, man. I'll try that.

Re: Is it possible to listen to lower bits in an audio file?

Reply #19
@saratoga, @DVDdoug thanks a lot, guys. I appreciate your input.

Re: Is it possible to listen to lower bits in an audio file?

Reply #20
What I want to know is if there's a possibility "to cut out the signal above certain db level and leave the rest". By "certain db level" I mean just that, user defined db level eg -70dB, -82dB etc.
Yes, this is called ducking (it's the opposite of gating). You can do it in Reaper with the ReaGate plugin, for example.

Is there a tool that lets you save a 24 bit audio file into 24 bit allowing you to adjust how much to truncate from the bottom? Say I input 24 bit file and save it into a 24 bit container that only holds 21.5 bits.
In Reaper, when you export to 24bit FLAC or WavPack, it gives you the option to choose an effective bit depth between 17 and 24 bits (whole bits only, however).

Re: Is it possible to listen to lower bits in an audio file?

Reply #21
Quote from: Brand
In Reaper, when you export to 24bit FLAC or WavPack, it gives you the option to choose an effective bit depth between 17 and 24 bits (whole bits only, however).
Yes, thank you. This is exactly what I've been looking for. ;)
PS As it turned out, the way I worded the question was probably lame and confused a lot of people. Sorry for that ::) . And again, many thanks to everyone.

Re: Is it possible to listen to lower bits in an audio file?

Reply #22
With ffmpeg for example:

ffmpeg -i input.audio -af "aeval=round(val(ch)*256)/256:c=same" -c:a pcm_s24le output.wav

This will basically quantize to 8bit audio from whetever it previously was.

Now if you  want to hear what was removed from X-bit to give you 8-bit just do this instead:

ffmpeg -i input.audio -af "aeval=val(ch)-round(val(ch)*256)/256:c=same" -c:a pcm_s24le output.wav
Don't you have the * and the / the wrong way around, ie ...round(val(ch)/256)*256... ?
It's your privilege to disagree, but that doesn't make you right and me wrong.

Re: Is it possible to listen to lower bits in an audio file?

Reply #23
Quote from: Brand
In Reaper, when you export to 24bit FLAC or WavPack, it gives you the option to choose an effective bit depth between 17 and 24 bits (whole bits only, however).
Yes, thank you. This is exactly what I've been looking for. ;)
PS As it turned out, the way I worded the question was probably lame and confused a lot of people. Sorry for that ::) . And again, many thanks to everyone.
What you'll get from that is an audio file with the least significant bits removed which you can then ABX compare with the intact file so you can test whether you can hear the difference – it won't make the LSBs audible in isolation.
It's your privilege to disagree, but that doesn't make you right and me wrong.

Re: Is it possible to listen to lower bits in an audio file?

Reply #24
Don't you have the * and the / the wrong way around, ie ...round(val(ch)/256)*256... ?
I suspect "val(ch)" returns fp number in [-1, 1] range, in which case your version would always produce 0.