Re-encoding - Is there such a thing?
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The MP3 codec is a lossy format. Unfortunately, there is no way back to the original file size without loss. The only real way to fix it would be to encode them at the desired bitrate from uncompressed WAV files.
Jason Hutchinson wrote:
The MP3 codec is a lossy format. Unfortunately, there is no way back to the original file size without loss.
Well, the file started life as 128kbps, then was converted to 320kbps. The question is, how much loss would be incurred when going from 320 back to 128, and comparing that version with the original that already was at 128. Even though, I realize, he no longer has it. But it would be an interesting experiment one way or another - and hang on to all versions each step of the way so they can later be compared.
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Send him here: mp3ornot.com[^] and if he doesn't score (say) 5/5 then he may as well give up, and encode them all to 128. I got the first 2 right, but failed on the third (didn't like the sound anyway) so stopped. I am not an audiophile.
That's an interesting test, although they really need more than just the one sample song. Some recordings can sound fine at 128kbps, and others might sound *terrible* at that resolution. But then I've only tried this with my desktop speakers with the window open, traffic going by and a fan running. Not exactly a great listening setup.
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That's an interesting test, although they really need more than just the one sample song. Some recordings can sound fine at 128kbps, and others might sound *terrible* at that resolution. But then I've only tried this with my desktop speakers with the window open, traffic going by and a fan running. Not exactly a great listening setup.
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I always encode everything to 128kbps. This comes from a time when my HD was only 500 GB (ancient times) and it was full. It's still very relevant for my (old) 160 GB MP3 player today. This may sound like I'm really old skool, or hipster maybe, but my music tastes aren't always on streaming platforms such as Spotify or Bandcamp. I don't really hear the 128 vs. 320 difference though. Maybe with headphones, but not from my laptop speaker or earplugs on my bike. Besides, the brain is great at filling in missing parts. The soundtracks back in the (S)NES days sounded like actual orchestras to me ;) Anyway, "upcoding" isn't possible as far as I know. Tried it sometime, but it gets glitchy. I can't imagine "upcoding" and then decoding back is good for your quality. I'm afraid your friend is out of luck :( But couldn't he just try to decode the 256 ones back to 128 and leave the 320 alone? Or decode the 320 to 256 or 192 to save some space while still having decent quality? He should also check VBR (Variable Bit Rate), which is kind of what you want, but not completely. It will work for the original 320, but probably won't make the "upcoded" stuff better. With VBR you get like 128 (or even less) at the quiet parts and maybe 320 at loud parts with lots of instruments, and everything in between, basically. Your bit rate with VBR can be something like 213kbps because it's an average of the various parts. It's the best of both worlds although, as said, I don't use it myself.
Best, Sander sanderrossel.com Migrating Applications to the Cloud with Azure arrgh.js - Bringing LINQ to JavaScript Object-Oriented Programming in C# Succinctly
Sander Rossel wrote:
I don't really hear the 128 vs. 320 difference though. Maybe with headphones, but not from my laptop speaker or earplugs on my bike.
:-D That much is a given. Laptop speakers are notoriously bad (and I laugh at laptops that have built-in Harman Kardon speakers). Steve Jobs has also done a *fantastic* job at lowering expectations given the hardware he hawks. I'm no audiophile by any stretch, but depending on the material, I'll almost *always* immediately make the distinction between 128 and 320kbps. 256 is where I'll typically start to get it wrong.
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There is more than one sample - by some magic, if one waits, the sample changes on next play - don't ask me how that works! I think you have to do all 3 listens, then select an answer - then the clip changes!
I hit refresh about 20 times, and the same sample keeps coming up. Maybe you do have to listen to them all, rather than immediately jumping to step 2. [Edit] Ok, my mistake was hitting Refresh to try to get a new sample. It seems to restart the whole thing, so refresh is entirely pointless...
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Someone's come to me with what I thought was a good question. He's got some 320kbps MP3s that were upconverted from files that originally were anywhere between 128kbps and 256kbps. X| Don't ask me why this was done. Someone must've thought introducing extra bits would magically improve the lower-res recording. He no longer has the original versions of the files. The question he asked me, and I had no answer for, is this: Is there software that can analyze the audio in a given file, and determine that it's something that does NOT require 320kbps and there would be "no loss" converting it back to 256 or 128kps, or whatever it was originally encoded from? His argument is that his library is now taking roughly 2x+ the amount of disk space it used to, with no benefit to be gained. Of course, he's already got some MP3s that *were* originally ripped at 320kbps, so he doesn't want to bulk-convert everything back to 256kbps or lower - only those that were at the lower resolution to begin with. Obviously this isn't "audio fingerprinting" like [MusicBrainz Picard](https://picard.musicbrainz.org/) can do. And I can't come up with the right keywords for googling.
I think you can tell by looking at the frequency content. It's easy to tell the difference between an "it started as an MP3" file and a "it started as a WAV" in my audio noise clean-up software because the spectrum for the MP3 has a brick-wall low pass filter well below the Nyquist frequency for the file's sample rate. That frequency is probably lower for lower-resolution MP3 files. (But no, I don't think I've ever checked this. And though I'm working from home, I risk far too much distraction from "I'm supposed to be working now" if I turn on the home computer and its associated audio editing tools to run some tests.)
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I think you can tell by looking at the frequency content. It's easy to tell the difference between an "it started as an MP3" file and a "it started as a WAV" in my audio noise clean-up software because the spectrum for the MP3 has a brick-wall low pass filter well below the Nyquist frequency for the file's sample rate. That frequency is probably lower for lower-resolution MP3 files. (But no, I don't think I've ever checked this. And though I'm working from home, I risk far too much distraction from "I'm supposed to be working now" if I turn on the home computer and its associated audio editing tools to run some tests.)
I wondered if the frequency spectrum would be enough to indicate the bit rate, as although I knew that MP3 encoding 'threw away' bits of the sound, I didn't know whether that would show clearly on a spectrum - so I just took the same 5s clip from the sound track to Clockwork Orange (The Thieving Magpie) which I know contains some well defined high-frequency bits (a flute or somesuch - I used this track to compare recordings onto audio cassettes many years ago, to establish which tape quality (cost) level was needed to avoid unacceptable loss). I loaded the orignal WAV file from CD into Audacity, and did a spectrum. Then saved as MP3 at a variety of bitrates. and did the same. Without being able to load pictures here, I can table the maximum frequency seen in the plots for various bit rates, as it is very clear there is a pattern:
bit rate -100dB freq (Hz) Full (WAV) 21,300 320 kbps 20,100 256 19,400 128 16,600I picked the freq at which the signal went below -100dB, as that corresponded roughly with the visual scale on the plot (which only goes down to -90 by default). The cutoff was well-defined, and if there is a place to put them, I can supply screen shots. So one could take a resampled file and, if its frequency cutoff (of maybe a suitable bit) was below, say, 18kHz, one could judge that it came from a 128kbps original. It would be a lot of work to do that for all files, of course. You really need a tool that measures in one go the frequency cutoff. What surprised me was that there was very little difference in the spectrum apart from the upper end - but of course, that is where the sample rate required for faithfull reproduction is greatest, so where the greatest reduction in file size can be obtained. Now off to look at the wavefoms in more detail... I, of course, am retired, so have nothing better to do - and it was fun.
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I wondered if the frequency spectrum would be enough to indicate the bit rate, as although I knew that MP3 encoding 'threw away' bits of the sound, I didn't know whether that would show clearly on a spectrum - so I just took the same 5s clip from the sound track to Clockwork Orange (The Thieving Magpie) which I know contains some well defined high-frequency bits (a flute or somesuch - I used this track to compare recordings onto audio cassettes many years ago, to establish which tape quality (cost) level was needed to avoid unacceptable loss). I loaded the orignal WAV file from CD into Audacity, and did a spectrum. Then saved as MP3 at a variety of bitrates. and did the same. Without being able to load pictures here, I can table the maximum frequency seen in the plots for various bit rates, as it is very clear there is a pattern:
bit rate -100dB freq (Hz) Full (WAV) 21,300 320 kbps 20,100 256 19,400 128 16,600I picked the freq at which the signal went below -100dB, as that corresponded roughly with the visual scale on the plot (which only goes down to -90 by default). The cutoff was well-defined, and if there is a place to put them, I can supply screen shots. So one could take a resampled file and, if its frequency cutoff (of maybe a suitable bit) was below, say, 18kHz, one could judge that it came from a 128kbps original. It would be a lot of work to do that for all files, of course. You really need a tool that measures in one go the frequency cutoff. What surprised me was that there was very little difference in the spectrum apart from the upper end - but of course, that is where the sample rate required for faithfull reproduction is greatest, so where the greatest reduction in file size can be obtained. Now off to look at the wavefoms in more detail... I, of course, am retired, so have nothing better to do - and it was fun.
Thanks for confirming my hunch. WAV file from CD should be sampled at 44.1KHz, which puts Nyquist at 22.05KHz -- pretty close to the 21,300Hz you see. My older ears don't hear the difference between 128kbps and 320kbps unless I'm paying close attention.
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Someone's come to me with what I thought was a good question. He's got some 320kbps MP3s that were upconverted from files that originally were anywhere between 128kbps and 256kbps. X| Don't ask me why this was done. Someone must've thought introducing extra bits would magically improve the lower-res recording. He no longer has the original versions of the files. The question he asked me, and I had no answer for, is this: Is there software that can analyze the audio in a given file, and determine that it's something that does NOT require 320kbps and there would be "no loss" converting it back to 256 or 128kps, or whatever it was originally encoded from? His argument is that his library is now taking roughly 2x+ the amount of disk space it used to, with no benefit to be gained. Of course, he's already got some MP3s that *were* originally ripped at 320kbps, so he doesn't want to bulk-convert everything back to 256kbps or lower - only those that were at the lower resolution to begin with. Obviously this isn't "audio fingerprinting" like [MusicBrainz Picard](https://picard.musicbrainz.org/) can do. And I can't come up with the right keywords for googling.
You can compare decoded raw bitstreams with tools like Audacity but imho it is unnecessary. Re-encoding with VBR (Variable bitrate) encoding would be the way to go. It will use lower bits when there is less sound information, and will use your maximum provided bitrate when it actually needs it. Also, I would strongly advise to migrate from mp3 to a better format unless you're hardware locked. 128 kbps opus is said to be transparent (almost indistinguishable from uncompressed form to 99% ears) while 80-96 kbps opus is somewhat equivalent to 320 kbps mp3. And the format is royalty-free with wide support from various OSes.
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You can compare decoded raw bitstreams with tools like Audacity but imho it is unnecessary. Re-encoding with VBR (Variable bitrate) encoding would be the way to go. It will use lower bits when there is less sound information, and will use your maximum provided bitrate when it actually needs it. Also, I would strongly advise to migrate from mp3 to a better format unless you're hardware locked. 128 kbps opus is said to be transparent (almost indistinguishable from uncompressed form to 99% ears) while 80-96 kbps opus is somewhat equivalent to 320 kbps mp3. And the format is royalty-free with wide support from various OSes.
Never heard of Opus until now. If I can't throw something at a random player and have it "just work", it's not even a contender in my book. I'm reading that "[...] in many respects Ogg Opus is the successor to Ogg Vorbis". That name, I recognized, and I stopped reading there. You can have the most awesome format in the world, if the support isn't there, it's a non-starter. Re-encoding is never a good idea, and going down this path to me sounds like having to re-encode in a different format every couple of years for the sake of using the format-du-jour. No thanks.
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Someone's come to me with what I thought was a good question. He's got some 320kbps MP3s that were upconverted from files that originally were anywhere between 128kbps and 256kbps. X| Don't ask me why this was done. Someone must've thought introducing extra bits would magically improve the lower-res recording. He no longer has the original versions of the files. The question he asked me, and I had no answer for, is this: Is there software that can analyze the audio in a given file, and determine that it's something that does NOT require 320kbps and there would be "no loss" converting it back to 256 or 128kps, or whatever it was originally encoded from? His argument is that his library is now taking roughly 2x+ the amount of disk space it used to, with no benefit to be gained. Of course, he's already got some MP3s that *were* originally ripped at 320kbps, so he doesn't want to bulk-convert everything back to 256kbps or lower - only those that were at the lower resolution to begin with. Obviously this isn't "audio fingerprinting" like [MusicBrainz Picard](https://picard.musicbrainz.org/) can do. And I can't come up with the right keywords for googling.
No, there will not be any such software. In short, your friend cannot return to their original 128k quality files from what they have now. The process you're referring to is called transcoding. Whenever you transcode across bitrates of a lossy codec, you lose information because the quantitization of the samples is different. So, quality was lost in the 128k -> 320k trancoding. More quality will be lost doing another transcoding from 320k -> 128k. Maybe your friend is now at a financial point in their life where they can afford to replace the mangled files?
I live in Oregon, and I'm an engineer.
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No, there will not be any such software. In short, your friend cannot return to their original 128k quality files from what they have now. The process you're referring to is called transcoding. Whenever you transcode across bitrates of a lossy codec, you lose information because the quantitization of the samples is different. So, quality was lost in the 128k -> 320k trancoding. More quality will be lost doing another transcoding from 320k -> 128k. Maybe your friend is now at a financial point in their life where they can afford to replace the mangled files?
I live in Oregon, and I'm an engineer.
patbob wrote:
Maybe your friend is now at a financial point in their life where they can afford to replace the mangled files?
There's "affording" the time, and "affording" the money. Despite the old saying, I hope you're not suggesting both are interchangeable.
