Convert MP3s to WAV first before resizing?

Brackish

New member
I've got some MP3s I'm working with that
need to be either shortened or stretched a
bit in order to sync exactly with the
master guide track.
Would I be better off converting the
MP3s to WAV prior to shortening
their length? Or doesn't it really
help any? I'd rather just leave them
as MP3s if there is nothing to
be gained by adding the extra
step.
 
Brackish said:
I've got some MP3s I'm working with that
need to be either shortened or stretched a
bit in order to sync exactly with the
master guide track.
Would I be better off converting the
MP3s to WAV prior to shortening
their length? Or doesn't it really
help any? I'd rather just leave them
as MP3s if there is nothing to
be gained by adding the extra
step.

I would stick to working in MP3 format.

If you save as a WAV, then re-render to MP3 again, you're losing quality.
 
Massive Master said:
If I'm not mistaken, doing pretty much *anything* to a MP3 file is going to mean a reconversion.

Yea that's probably true.
 
More than likely your software will have an easier time dealing with it if it's in .wav format. I say convert!
 
danny.guitar said:
If you save as a WAV, then re-render to MP3 again, you're losing quality.


Oh, no, it's not going back to MP3. The plan
was to just convert the whole thing
to WAV as the last step.
It would be easier to just work with
the tracks as MP3s and then just do
the one conversion at the end to
WAV rather than converting all the
MP3s to WAV first.

I'm just wondering if there would be
any difference in the end result sound
if the tracks were worked
with (strecth/EQ) as WAVs
instead of MP3s -- or if it would
really make no difference at all.
 
Even those editors that "edit MP3s directly" need to convert the lossy data to a bitmap proxy before the editing can be done. One may not be converting/reconverting the entire file, but where the editing takes place, spot reconversions are indeed happening.

But if you are keeping the edits as WAVs, you're OK, go ahead and edit them a WAVs. The WAV proxy will be as close to an exact bitmap copy of the MP3 as one would ever need. It's only when having to reconvert back to MP3 that the extra information loss is an issue.

I guess my question is why not edit the source WAV file instead of it's MP3? All MP3s have to start as bitmaps somewhere. Unless you're talking about a commercial MP3, in which case you're on your own.

G.
 
SouthSIDE Glen said:
Even those editors that "edit MP3s directly" need to convert the lossy data to a bitmap proxy before the editing can be done.
G.

G,

Do you have any further info on this? I've heard of bitmap proxies in regard to image data, but not audio.
 
MadAudio said:
More than likely your software will have an easier time dealing with it if it's in .wav format.


Thanks, MadAudio. I hadn't thought about this
(but now that you mention it) the editing software
does seem to bog down a bit sometimes on
playback of mp3, which I haven't noticed much
with wav playback.
 
SouthSIDE Glen said:
The WAV proxy will be as close to an exact bitmap copy of the MP3 as one would ever need.

I guess my question is why not edit the source WAV file instead of it's MP3? All MP3s have to start as bitmaps somewhere.

G.

Thanks, Glen. I guess I'll just convert all the individual tracks
to wav before working with them.

As to your question "why not edit the source WAV
file" ... the MP3 IS the source file. The
material was originally recorded live to MP3.
 
masteringhouse said:
G,

Do you have any further info on this? I've heard of bitmap proxies in regard to image data, but not audio.
Common examples of proxies in audio are when you import an MP3 into a standard WAV editor. For example, open an MP3 in Sound Forge. It'll pop up a dialog box stating it's creating a proxy for the MP3 before the final waveform display appears for editing. On faster CPUs this popup may be only momentary and perhaps even so fast as to be un-readable. (And in fact I'm referring to version 6, which is what I still use; it's possible they may have even removed this popup in later versions, I don't know. But I'm sure the underlying process is basically the same.)

The central issue is that MP3 is a format based upon a lossy codec. Codec meaning encoder/decoder. MP3 is to WAV audio what JPG is to BMP or TIFF graphics in this regard (It's also like what MPEG2 is to AVI in video; in fact MP3 is actually shorthand slang for "MPEG1, Layer 3"). An integral part of the MP3 encoding is the shortening of the file by, when possible, replacing long strings of bits with shorthand (and sometimes only approximate) mathematical descriptions of those bits. It's kind of like replacing the string "0000000000000000" with the string "0x16". Instead of taking 16 characters to describe the string of 16 zeros, the length of that particular string has been shortened by a factor of four to four characters by "encoding" that information using an agreed upon code.

A more lossy form might be replacing all the bits it would take to describe the upslope of the initial attack of a wave form 10 samples long - which uncompressed would require a full 16-bit description of every sample in the slope - with a string that gave the initial amplitude, an algebraic description of the curve of the slope between the two (e.g. "y=2x+3") and the length of the slope in number of samples (in this case, the number 10). In such a case the bitmap description of the slope that took 160 characters (16 bits times 10 samples), has been reduced to somewhere around 30 characters (when you include stop bits and such)

These are a gross oversimplifications of what the MP3 codec is actually doing, of course, and that is not technically exactly how it works (we're ignoring bits vs. bytes, for example), but it's a fair demonstration of the general concept of how mathematical shorthand and approximation are used to compress bitmapped information (audio, video, or graphic) into smaller sizes via lossy compression. It's not totally unlike a stenographer using shorthand notation to record actual uncompressed conversation.

But what happens when we want to edit the encoded information? What happens if "0000000000000000" needs to be changed to "0000000000001000". It's not a simple matter of just going in and changing the "spelling" of the phrase "0x16". That phrase is now completly inadequate for describing the uncoded information, because there is no longer a string of sixteen identical characters. It needs to, at some level within the software, decoded so that the new "1" can be added, and then re-encoded to something like "0x12[stop bit]1[stop bit]0x3".

To the best of my understanding, even those packages that tout themselves as "MP3 editors" have to decode (or translate, if you will) at least part of the file back to individual bitmap so that editing can actually be done at the bit level. Sound Forge and "standard" editors that let you drag MP3s in for edting create bitmap proxies on which to perform the actual editing. "MP3 editors" have to do the sam to at least part of the file, if not all of it.

It's like when you go into Photoshop and edit a compressed JPG file. The picture you see on your monitor is actually a pixel-for-pixel representation of how the JPG decodes. It's a bitmap proxy of the JPG information. It has to be for you to be able to do pixel-level editing of the image. Then when you save your edit, the visible proxy is freshly encoded as a new version of the JPG file.

G.
 
Except the compression you're describing "000000000000000" = "0x16" is commonly referred to as RLE-type compression, which is essentially lossless because it can be decompressed to its original form. MP3 is not a (completely) lossless codec, like FLAC for example.

MP3 does include lossless compression as well, but it also removes frequencies that are not in the "audible range".

Us musicians who record and use MP3s on a regular basis can hear the difference easily at around 128KBPS-192KBPS, but most other people can't.

Just thought I'd add my 2¢ which you probably knew already :D
 
Grilled_Cheese said:
Except the compression you're describing "000000000000000" = "0x16" is commonly referred to as RLE-type compression, which is essentially lossless because it can be decompressed to its original form. MP3 is not a (completely) lossless codec, like FLAC for example.

MP3 does include lossless compression as well, but it also removes frequencies that are not in the "audible range".
You are correct on all counts.

The lossy compression come in with the mathematical approximations like in the second example of "slope" that I gave. It's in these approximations where at least some, if not much, of the lost HF content lost in MP3s is indeed lost. The more detail one wants to save, the more exact a mathematical approximation must be. The more exact the description, the more bits it takes to make that description, and the less the amount of bandwidth savings. This exactly describes the difference between the different bitrates of MP3 encoding. The higher the bitrate, the more accurate the description, and the larger the file.

In digital audio, HF requires detail. A clean, 20Hz sine wave is easy to mathematically approximate with a good degree of accuracy. Modulate that sine wave with low-volume, high frequency noise, and you can still approxinate the signal with the same sine wave function, but the HF content will be lost. The higher the frequency you want to keep, the higher the resolution of the approximation must be, and the less the amount of data compression you can apply.

But even in simple run-length encoding (RLE), as well as in approximation, it's not possible to modify/edit individual bits of digital information unless you first convert the compressed information back into a bit-resolution representation (a bitmap) of the signal that information represents.

Again, think of editing a JPG file in Photoshop. The image you are editing on screen is a decoded, single-pixel-resolution version of the encoded JPG file. you don't have 16-pixel blocks of red, you have 16 individual pixels that can be edited individually. It's the same with audio proxying. When you drag an MP3 into Sound Forge, you do not have an entire, indivisible, solid line representing an attack slope of 2x+3 that's 10 samples long, you have a decoded (proxy) version of that that breaks that slope back into 10 individual, editable samples. Then when you're done editing, if you save it as an MP3 again, the MP3 encoder has to re-encode the bitmap proxy in memory back into a newly compressed file.

G.
 
SouthSIDE Glen said:
To the best of my understanding, even those packages that tout themselves as "MP3 editors" have to decode (or translate, if you will) at least part of the file back to individual bitmap so that editing can actually be done at the bit level. Sound Forge and "standard" editors that let you drag MP3s in for edting create bitmap proxies on which to perform the actual editing. "MP3 editors" have to do the sam to at least part of the file, if not all of it.
G, thanks for the info. I suppose that it depends on the type of editing that you are going to perform. From the little doc that's available for mp3trim it mentions that it works with the MPEG audio frames directly unlike Sound Forge etc. If you are just trimming off tops and tails, removing silence, and performing volume changes it seems like this software can do this directly from the MPEG audio frames without conversion (other than listening back to your edits). This is not a visual editor like Sound Forge, so there is no need for the software to show a display of the wave data.

If we're talking about time compression and expansion it's a different animal. Final Cut Pro's native editing for example allows one to edit compressed MPEG-2 video without conversion, at least as it is described. I would imagine that a similar type of editng could be done with mp3 audio, but with so many wav editors it wouldn't be much of marketable product except for needs where the audio has to remain entirely in the mp3 domain.
 
masteringhouse said:
G, thanks for the info. I suppose that it depends on the type of editing that you are going to perform. From the little doc that's available for mp3trim it mentions that it works with the MPEG audio frames directly unlike Sound Forge etc. If you are just trimming off tops and tails, removing silence, and performing volume changes it seems like this software can do this directly from the MPEG audio frames without conversion (other than listening back to your edits). This is not a visual editor like Sound Forge, so there is no need for the software to show a display of the wave data.
Ah, OK. Yeah, one can indeed edit at the frame level, however - I *think*, though I am admittedly not 100% sure of this - the individual frames would still need to be proxied for the very reasons I gave in the previous posts. This proxying may not be visible to the end user, it may just be a processing step internal to the software.

G.
 
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Lots of folk can hear the loss generated by MP3 compression IF they've experienced sound/audio/music etc pre MP3. There's a generation out there that hear MP3 as right just the way I hear vinyl & tape as right (except I like to think of my self as righ right right but not right wing).
Why work in mP3 unless you set out to deliberately not achieve the best quality you can or hear MP3 as right?
 
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