Hi every one! Registered here as need some specialist help and don't have any one to get help from. My life and reletionship is in doubt because of one recording. There is some noise involved and can't work out if my partner cheated on me or not. Weird I know but never the less want to get to bottom of it.If any of you could help me am happy to send that recording or put it online for someone whos willing to help. Please msg me or answer to my post. Any help (but not life advise) welcome.
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If it is not too embarrassing for you, you're welcome to post it to the forum using the upload files button below
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Dialog Isolate in iZotope RX 6 would probably help but I don't have it myself. I've got RX 3, perhaps something there will work, I'll take a look.
A quick listen through shows a small passage of conversation at the beginning and again at the end. Now where did I put my headphones ... -
I´m by myself quite new at audio so I can only give some general recommendations. Most programs offer to use the full-functionality of it´s program for some weeks for free to try it. So you might go this way. Often noise reduction functions are called de-noise, de-hiss, ... this might help to make an own research in google, ... .
Also words as "restoration" will help.
There are a lot of programs like:
- Wave Arts Restoration (trial)
- Acon Restoration suite
Especially Acon is a not so complex program and there is a de-noise tool.
Good luck! -
Sounds pretty obvious to me even not being native English... sorry
It can be processed further but source is mp3 compressed and it will be difficult to improve this recording due related compression artefacts. -
Source is 128kbps after processing with bellow script it is 52kbps in average (encoded with q 2).
Code:@set aproc="pan=1c|c0 < 1*c0+1*c1,firequalizer=gain='if(gte(f,200),0,-INF)+if(lte(f,5000),0,-INF)',dynaudnorm=p=1/sqrt(2):m=100:s=9,firequalizer=gain='if(gte(f,200),0,-INF)+if(lte(f,5000),0,-INF)',aresample=resampler=soxr:osr=22050:cutoff=0.990:dither_method=0,asplit=2[wav][mp3]" @ffmpeg -y -i %filename% -vn -sn -dn -filter_complex %aproc% -map [wav] -c:a pcm_f32le -f wav "%~n1_dyn.wav" -map [mp3] -c:a libmp3lame -q:a 2 "%~n1_dyn.mp3"
Signal itself is passed trough bandpass filter (quite sharp 200Hz - 5kHz, under 200Hz are only distortions and human voice can be easily identified within this limited spectrum).
Source is decoded to wav and mp3 (mp3 is smaller and more versatile nowadays then ancient PCM audio, wav can be used to be processed further in some sound editor but as i said - due source being mp3 compressed it will be extremely difficult to gain significant quality improvement). -
This should be put on a Voyager III probe... let THEM figure it out ! :^p
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