Guide to Ripping & Encoding High Quality MP3s

Guide for Encoding Efficient, High Quality Digital Audio

Last updated: November 7th, 2009

- First, decide which audio codec you wish to use!

• MP3 is the most popular format, the most supported, and therefore the first choice for most people. In recent years, the quality of MP3 has been continually pushed to the envelope by the LAME encoder team (yes, it really is called LAME) despite the limits of this aging format. Sometimes, however, the limits cannot be easily overcome. For example, LAME attempts to fix the gap problem, where MP3s typically play an extra bit of silence at the end, something quite noticable when playing tracks that are supposed to flow into the next. Since LAME's fix is not a part of the MP3 standard, however, only a few players will skip that trailing silence.
  PC & HARDWARE PLAYERS: nearly any
  
  
• Ogg Vorbis is an unpatented, open-source, free-as-in-freedom audio codec. It especially excels at low bitrates (less than 128Kbps) compared to the other formats, and it is gapless between tracks. Although official development has crawled along, unofficial Aoyumi's Tuned Vorbis (aoTuV) versions have kept quality marching forward. Also of note, some claim that Vorbis encoding flaws sound less harsh than those of MP3. A fair number of hardware players support Vorbis, but unfortunately not nearly as many as MP3.
  HARDWARE PLAYERS: iPods (with Rockbox), several iRiver models, serveral Cowon models, Pocket PCs (with TCPMP), and many others
  PC PLAYERS: WinAmp, Windows Media Player (w/filter), Foobar2000, and many others
  
  
• FLAC is not a lossy codec like all of the above. Instead, it performs lossless compression, which means FLACs will always output the exact same audio that was put into them. But because FLAC does not selectively discard data like lossy formats, the files are quite larger. However, decoding takes very little CPU power, which makes for fast conversions from FLAC to whatever lossy format your portables may use, or whatever format would be easiest to "share". FLAC is useful for archiving in perfect quality, or for those with huge hard drives.
  HARDWARE PLAYERS: iPods and iRivers (with Rockbox), Pocket PCs (with TCPMP), and others
  PC PLAYERS: WinAmp, Windows Media Player (w/filters), Foobar2000, and others

Encoding from Audio CDs

I. First-time setup:

1) Because reading the CD right matters just as much as how you encode it, download & install the best, Exact Audio Copy, from
Introduction Exact Audio Copy

2) Download the currently recommended encoder for your preferred format, unzip, and place it in a folder you will remember.

MP3: LAME 3.98.2
Ogg Vorbis: Oggenc2.85 using aoTuVb5.7
FLAC: FLAC for Windows with installer (v1.2.1)

3) Run Exact Audio Copy. The Configuration Wizard should pop up (if not, start it from the "EAC" menu).

• When it asks you, select all your CD drives. (Note: CD-RW drives are typically best for ripping purposes.)
• Select "I prefer to have accurate results" for each drive.
• Auto-detect your CD drive features. This requires an audio CD to be in your drive.
• MP3 only: Check "Install & configure the external LAME.EXE compressor." Stop the search, manually find the path where you downloaded LAME, and select either quality option given (this will be changed later).
• Enter the e-mail address.
• Select "I am an expert."

4) Open the "EAC Options" from the "EAC" menu.

• Under the Extraction tab, put "Error recovery quality" at "Medium".
• If you have a constantly-on Internet connection, under the General tab select "On unknown CDs, automatically access online freedb database."
• Under the Filename tab, change the naming scheme to

  %N - %T

  which is basically the format "01 - TrackOneName". You may want to put additional parameters in there too, that's fine, but make sure the track number gets in there somewhere... unless you feel like having the tracks listed in alphabetical order rather than the natural CD order. :worried:
  You can also put your albums in directories, for example

  %A\%C\%N - %T

  which is "\Artist\Album\01 - TrackOneName"

5) Open the "Drive Options" from the "EAC" menu.

• Under the Extraction Method tab, it should already be set to "Secure mode with following drive features." If not, change it.
• Under the Drive tab, hit the "Auto-detect read command now" button.

6) Open the "Compressor Options" from the "EAC" menu.

• Under the External Compression tab, change the "Parameter passing scheme" to "User defined encoder", and enter the appropriate file extension below.
  
  MP3: .mp3
  Ogg Vorbis: .ogg
  FLAC: .flac
  
  (Note: Selecting "User defined encoder" disables the effects of the "high quality/low quality" buttons & the bit rate drop-down menu. So just ignore them.)
  
  
• Browse to the location where you unzipped the encoder. The specific file you're looking for is:
  
  MP3: lame.exe
  Ogg Vorbis: oggenc2.exe
  FLAC: flac.exe (in the "bin" sub-folder)
  
  
• Enter under "Additional command line options":
  (These commands determine what methods will be used to encode the audio)
  
  MP3

  -V 2 --vbr-new %s %d

  Ogg Vorbis

  -q 5.0 -a "%a" -t "%t" -l "%g" -d "%y" -N "%n" -G "%m" %s %d

  FLAC

  -6 -V -T "artist=%a" -T "title=%t" -T "album=%g" -T "date=%y" -T "tracknumber=%n" -T "genre=%m" %s %d

  NOTE: Make sure no extra spaces or discrepancies are included when you enter or copy these commands! This can cause the encoder to fail when it tries to encode the music, and you will just end up with WAV files!
  
• You will probably want to check "Delete WAV after compression". UNcheck "Add ID3 tag" if you are NOT using MP3.
  
• MP3 only: Check "Add ID3 tag." Under the Offset tab, look at "Construction of the ID3 tag comment field", select "Write following text into ID3 tag comment", and then type LAME v3.98 -V 2 --vbr-new or your version of LAME & encoding method used, if different. (This is simply used for identification, so anyone viewing the ID3 comment can tell what quality mode was used to encode the MP3.)
  NOTE: Do NOT also add ID3 tags via the "Additional command line options", or you will end up with possibly-erroneous double-tagged MP3s!

Congrats, you finished the long part! Once everything has been set up, you shouldn't need to go back & mess with any of these settings. =)

II. Ripping each CD:

• Run Exact Audio Copy & put the audio CD in the appropriate drive.
• Either enter the CD/track/artist info yourself, or get it auto-filled from the CDDB (ALT-G) if you're connected to the Internet.
• To rip & encode the entire CD, click the "MP3" button on the left. To get individual tracks, select all the ones you want and press SHIFT-F6.
• If you don't mind taking double the time to be even more confident that your rips are coming out perfectly, then instead of hitting the MP3 button, right-click on the selected tracks and select "Test & Copy Compressed". After all the ripping, if the "CRC" field at the very right says "OK" for each track, all is good! If not, the test & actual copies read differently, which means one or both read something wrong.

That's it. The End! Hopefully you now have a folder full of high quality audio files ready for listening!

Important Lossy Concept

A lossy file (such as MP3, AAC, or Vorbis) can never turn itself back into the original audio it is trying to approximate. Any converting, any burning, and any playing can only use the imperfect audio in that lossy file to do its job, so turning an MP3 to a WAV or burning it to a CD will only result in audio that sounds exactly as imperfect as the MP3. Also, if you were to take that imperfect-sounding WAV and turn it into MP3 again, it will only result in more loss. For this reason, it is inadvisable to convert lossy files to other lossy files. It is always best to use lossy compression on only original full quality audio.

Since lossy compression works by trying to remove the information humans percieve least, such quality degredation may not always be detectable. Indeed, the hope is that the encoded audio will sound exactly the same as the source. But quality reduction does always occur even if it is often inaudible.

Encoding from Files

I. First-time setup:

• Foobar2000 has a reputation as a spartan, utilitarian little audio player. However, it features one of most customizable file converters that I have yet to try. So go ahead and get downloadin' and installin'. Oh, and make sure not to de-select that important little "Converter" component.
  
• Download the currently recommended encoder for your preferred format, unzip, and place it in a folder you will remember.
  
  MP3: LAME 3.98.2
  Ogg Vorbis: Oggenc2.83 using aoTuVb5.7
  FLAC: FLAC for Windows with installer (v1.2.1)
  
  (Wait! Haven't you done this part already? If not, return to the top of the thread, do not pass Go, do not collect $100, and install EAC already!)
• Run Foobar2000 and add to the playlist the file(s) you want to convert. Select them all and right click, choosing "Convert > ..." from the menu.
• Choose your preferred audio format from the drop-down box under "Output format". The default encoding settings are right in line with the suggestions of this guide, so there should be no need to tweak them. If you do wish to tweak them, however, you can hit the "..." button to the right.
• Under the "Output files" section, you can edit the output file names. I would recommend changing the Name Format setting to %filename% so that your converted file will only differ in file extension.
• Under "Output path", tell it where you want to put the files. Then click "OK" at the bottom. Now remember when I told you that you had to memorize where you put the encoder? This is where you have to go find it, Fido.
• After selecting the directory to drop the new file into and watching the progress bars fill to full, you should now be the proud owner of a newly converted file, complete with any tag information that was in the original file.

II. Each subsequent conversion:

• Run Foobar and add the files to the playlist.
• Select, right-click, "Convert > ..."
• Select your desired codec, if it isn't selected already, and hit "OK".
• Wait for moist, delicious cake^Z^Z^Z^Z MP3s.

Foobar2000, especially with all of its components, can convert just about anything you throw at it, to just about any format you want.
Except for emulated game music. For that, just download some a WinAmp plugin and diskwrite it to WAV before throwing it to Foobar.

ReplayGain: Preventing Loudness Jumps & Clipping

MP3Gain

MP3gain is a very useful program that performs volume normalizing, maximizing, and adjustment. Since MP3s are just an approximation of the original file, and since modern CDs are pushed so very close to the maximum volume/amplitude value, at some points the waveform of a decoded/played-back MP3 may calculate as a value above that maximum. This is called clipping, because those higher values must be truncated down to the maximum limit, flattening those segments of the waveform and often introducing annoying pops or static. MP3gain can prevent clipping by reducing the MP3's internal volume level just enough that its peaks will not breach the maximum amplitude.

The program can also "normalize" song loudness, meaning that it will make MP3s sound about equally loud from track to track (in default Track Mode), preventing the constant need to adjust your player's volume when listening to your collection. MP3Gain will even preserve the intended volume differences between songs on the same CD when you use Album Gain mode, instead attempting to equalize the overall loudness of different albums.

Note that for this to work as intended, you must Album Gain or Track Gain all the MP3s in your collection. Also, it will often make your songs quieter. This is because it uses a loudness standard that attempts to minimize the amount of clipping caused by raising dynamic tracks, with high peaks and otherwise low levels, to the same overall volume as more modern tracks, which tend to be so dynamically-compressed that the entire song hugs the maximum amplitude.

NOTE: Winamp has recently gained the ability to read MP3Gain's ReplayGain tags, so the compatability worries previously espoused here are now irrelevant.


Scientific Lossy Audio Codec Comparisons

- Sebastian's Public Listening Tests
- Roberto's Public Listening Tests

Please let me know if there are any other links I should include.

Questions and Explanations

Why is it ripping so slowly? I can get 5-10x faster with MusicMatch/CDex/WinAmp/etc!

• Secure Mode double-reads the CD to make sure there was no read error (i.e. each data sector should be the same when read twice) caused by something on the CD or just a bad CD drive. This is the only mode that can let you know when there is an error, and the only one that will rescan the CD to get a prominent reading when it does find a discrepancy. Speed is what you sacrifice for this quality.
• If you absolutely hate the slower ripping speed, or find that a CD is encountering error after error & taking hours, use Burst Mode with Test & Copy to verify. Occasionally Burst Mode results in a better rip in those cases.

I hate VBR mp3s! They always give me problems, and are certainly not high quality!

• LAME's implementation of VBR (variable bitrate) is markedly improved over any other VBR mp3s you've heard. Audiophile enthusiasts have even done scientific double-blind listening tests to prove that LAME's VBR is audibly "transparent" (sounds no different than the original audio) to most listeners.

What's wrong with 128Kbps? Why use something so LAME? (harharhar) Why not encode with a constant 192Kbps?

• Firstly, if you can't tell the difference between 128Kbps mp3s and the originals, do yourself a long overdue favor: buy yourself a better set of speakers, headphones, or sound card. (Remember that the weakest links in the chain from that audio file to your ears dictate the sound quality.) However, older folks and those with hearing loss may not be able to tell the difference. And honestly, the difference can sometimes be subtle.
• We use the LAME encoder, as it is considered the best encoder out there for bitrates above 112Kbps. LAME produces better quality mp3s than even the official Fraunhoffer encoders. Isn't it great that LAME is free?
• The -V switch invokes a VBR (variable bitrate) encoding mode in LAME. VBR is, in theory, inherently superior to CBR (constant bitrate). CBR mp3s may at times have too few bits available to encode all the audible information in a frame. Or one frame may only need a fractional amount of the available bits to be audibly the same, with the rest of the frame filled with inaudible data to keep the mp3 at a constant bitrate. A variable bitrate cures such problems of lost audio quality & wasted file space by determining how many bits should be used in each different frame to keep the audio audibly unchanged. (Admittedly, the "bit reservoir" in CBR mp3s attempts to resolve this as well, but it is much less effective than pure VBR.) Typically, a VBR file may possess audio quality equivalent to a 10-20% larger CBR file.

What are the typical average bitrates of MP3s using these VBR switches?

• See the LAME VBR bitrate chart on the Hydrogen Audio Wiki.

Why is the encoder set to "User defined encoder" in EAC, when there is already an option for the LAME encoder?

• Choosing "User defined encoder" and entering -V [0/1/2/3/etc] --vbr-new %s %d effectively disables all other mp3 encoding options in the External Compression tab. This is done because certain quality parameters in LAME can interfere with -V's functionality, even some typically associated with producing hi-quality LAME mp3s. We are merely making sure that these specific VBR quality settings, and the source & destination files, are the only parameters passed to LAME.

Why set Error Recovery to Medium? Why use Secure Mode?

• If any kind of error occurs while reading the CD, an Error Recovery of Medium tells EAC to rescan this section of the disc the 2nd-most number of times allowed. EAC will then use the most prominent result found from these reads. ('High' was previously recommended. But if a 'Medium' amount of reads isn't able to give a prominent result, more attempts will very likely be little help & will only slow the ripping process.)
• The Burst & Fast modes may often make copying errors without realizing this or attempting to make any sort of correction. Secure Mode does rip more slowly, but it attempts to make a more perfect copy of the track than either other mode.

Where can I get more information about digital audio?

• The Hydrogen Audio forums are an excellent resource for digital audio discussion, explanation, testing, and development.

Why is this post so long?

• Because it's awesome?? :sweat: :worried:

If you have any questions about all of this, feel absolutely free to reply or contact me.


How ya doing, buddy?

Thanks to Darko for keeping a copy of my thread alive even after the GFF crash! This one has a few changes, such as a totally new section on encoding files already on the hard disk, additional explanation of what MP3Gain does, updated broken links, a short mention of some CPU-optimized Vorbis encoders (much quicker!), and the newest version of the Musepack encoder. Oh yeah, and taking out AAC to reduce clutter, and because I have the feeling absolutely no one who followed my guide used that format.

cubed, I would argue that -V 0 --vbr-new is a bit overdoing it & wasteful of space, since -V 2 --vbr-new doesn't reveal even subtly audible difference in the great majority of cases. But at least you're not using 320Kbps.
*Coughs to clear his throat and dons his mock-superior voice* Yes, cubed, you are a very bad man for using higher quality settings than necessary for your own personal use! Shame, to the highest degree!

There's nowhere I can't reach.

Omigosh it's Drexie! And I get the honor of having my thread stickied by him!

Oh, yes, and hello. ;D

Ah, who cares if it's your first action in four years? Wouldn't want you to break out into a sweat with all that heavy mod work. Nope, nope!
Good to see ya around again.

This thing is sticky, and I don't like it. I don't appreciate it.

Phatcorns, I'm not certain why that is. The option is ghosted on mine as well...

But, if you are using lame.exe in the "External Compression" tab, following the directions in my ripping guide, then it will use those settings when doing the "Compress WAVs" command. And I believe EAC may be able to decode MP3s without LAME.

Additional Post:
And just linking a post I just made in another thread which explains, in a bit of detail, some of the aspects and trivialities of audio encoding. For those who care to know...
http://www.gamingforce.com/forums/be...ost355440.html

I am a dolphin, do you want me on your body?

Sorry, Aevum. Evidently Rarewares.org -- the site I link to for the LAME encoder -- was down for a bit due to a billing dispute. But it's back up again.

I was speaking idiomatically.

Hrm, Cal, that is very odd. EAC should be converting those symbols (like %a) into the actual artist/album/track/etc. strings shown in the main window before it passes the command to the encoder. Perhaps you could try it without the surrounding "quotes"?

EDIT: I just tried to reproduce your problem, but to no avail. Even with the ""s around the symbols, the tags were written fine.

What kind of toxic man-thing is happening now?

I'm not exactly sure what you mean by that, Cal.

EAC and Musepack work together fine for me under these compression settings (and with the rest of the commandline that's off-screen, identical to what I wrote in the guide above).



FELIPE NO

Hrm.... Rocktime, see the screenshot in my last post? Check-mark "Check for external programs return code" at the bottom left. Then, when any error dialog box pops up during extraction, save a screenshot of it. Hopefully we can figure out what the problem is that way.

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That's really weird. Especially so with you saying that it won't work sometimes, and then sometimes those exact same tracks will encode. o.O

I don't understand that at all. Even if EAC was getting different rip results each time... the audio itself shouldn't make LAME error. Only, like, invalid parameters or such... and EAC should be passing LAME the exact same parameters each time.

The only thing I can possibly think of is LAME having trouble with the temporary file names, since that's the only thing that changes each time. Perhaps the ()s in the temp file name that you linked? Although, really, I can't think of why that would be any problem.

Most amazing jew boots

What application are you using to play MP3s? Winamp shouldn't have this problem, nor should Foobar2000. Yet it wouldn't surprise me at all if Windows Media Player messed it up; past versions have handled VBR files rather shoddily.

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Originally Posted by Diversion

It is WMP, actually, I'll try it out in other programs (should of did that in the first place, I assumed it was a bad encoding).

Hmm, does seem to work better now. It sounds exactly the same, don't know why it does that.

Glad to hear you figured out that it's the player (it figures, I've never liked WMP), rather than the MP3. If you're not using the latest version, perhaps updating could help.

Originally Posted by lundeberg

There's nothing about mpeg4 files!! How Do I convert them ? Is there a freeware that does it?

There is no one specific MPEG-4 audio specification, but instead there are a number of versions of AAC. I removed the AAC encoding from my guide because the only encoder that works with Exact Audio Copy produces notably worse quality than the Apple, Quicktime, and Nero AAC encoders.

This thing is sticky, and I don't like it. I don't appreciate it.

Originally Posted by Spikey

Check this out, folks:
http://www.hydrogenaudio.org/forums/...howtopic=54085

Everyone using V2 as a standard yet?

This thread has been recommending -V 2 for quite some time now... in fact, ever since the --preset/--alt-preset modes were abandoned to introduce the -V switch.

I'm not too sure what new you're trying to point out here.

I am a dolphin, do you want me on your body?

Originally Posted by Spikey

Nothing, I just find the sort of relative complexity of MP3 encoding amusing.

Well, it is a highly mathematical process with so many possible implementations.

Originally Posted by Spikey

But I guess what I was hinting at was, is everyone using this standard and does everyone understand it I'm guessing not.

Hehe, no and no. I mean, you just linked that HA thread... and its only one of many more like it. People in that forum get rather tired of all the "newfangled" command line combinations that newbies come up with and try to claim as THE BEST!

I was speaking idiomatically.

Originally Posted by Spikey

But people wouldn't release public beta's and keep updating them if they were worse (hint: 3.98b3 = best MP3 encoder out there).

Then, using your logic, because there's an update out there with a higher version number, by that fact alone it must be better than all versions that came before it? Have you ever heard of "version regression"? For example, LAME 3.90 had been recommended as the preferred version over LAME 3.93 because of a regression in quality. And numerous programs have had flaws introduced, or re-introduced, in successive versions. It's pretty much an inevitable part of developing complex, and even sometimes simple, programs.

And when we're talking about perceptive audio encoders, something so difficult to objectively evaluate, where the "better"-ness of a program is based on how it sounds to the human ear, something not measurable by a computer nor even easily objectified by humans themselves, it needs thorough testing to ensure that the changes introduced have indeed improved the overall quality.

Thank you for the news of the new 3rd beta of 3.98, but as of now, 3.97 stable remains the recommended version of LAME both at HydrogenAudio and in this guide.
Also, thanks for letting us know about the RareWares site redesign. It seems I will need to update the links in my first post.

What kind of toxic man-thing is happening now?

Originally Posted by Dee

I'm getting a weird error from EAC. I recently installed it on my work computer (currently posting from it), and this is what I get when I try to compress it:

http://img413.imageshack.us/img413/9606/untitlednq5.jpg

(I know image quality sucks... don't have photoshop on this computer!). It looks similar to the other person's error on the first page. I still have all the wav files when I ripped the CDs. Any suggestions?

Judging from the command line it's attempting to pass, it appears that you have your Parameter Passing Scheme ("EAC" menu -> Compression Options -> External Compression tab) set to "LAME MP3 Encoder". It should be set to "User Defined Encoder", at the very bottom of the list.

Hope that helps!

FELIPE NO

Originally Posted by Spikey

I take your point man. I'm going to be using b4 because it has advantages over 3.97, and I use high bitrates (apparently there's no issues with b3/4 at high bitrates).

I'm not sure "no issues" is quite accurate. All lossy codecs tend to have audio issues, and I did see some early comparisons in a HA thread where it seemed that 3.98 improved the handling of some problem samples and receded in others. Although, it's probably true that there are no major, glaring issues.

Originally Posted by Spikey

And Moguta- maybe you should consider recommending LAMEDrop, not command line utilities. Then noone is using switches, everyone's using a simple to follow VBR method and it's easy to explain to newbies.

That might be a good idea, at least for those who want to only encode from WAV. I've just never liked the *drop interface (no menu or buttons?), and some may find transcoding useful. But I'll consider.

Originally Posted by Spikey

For example, the recommended Vorbis encoder is an inferior encoder to the best (stable I might add) one out there (it's not even on RareWares!).

o.O I'm not sure what you mean by that. I thought AoTuV beta5 was the latest Vorbis, unless you're referring to all the chipset-optimized compiles.

Originally Posted by Spikey

No problem man, we're ultimately in agreement I think. <offers hand to shake>

I think so, too. No hard feelings here. *Shakels*

What, you don't want my bikini-clad body?

http://www.hydrogenaudio.org/forums/...howtopic=55852
Here's the list of new features, although most people probably don't need to worry about them. The addition of AccurateRip, however, is a quick additional method to ensure the security of your rips.

You'll also notice some bug reports in the topic I just listed, although seemingly only with a couple of the new features. Personally, I'd advise waiting just a little while, until bugs can be found and ironed out.

Jam it back in, in the dark.

Thanks for the compliment, and I'm glad you find this guide useful. And, if you'll notice, the very first thing in the guide is "new news" from 2005... so the similar guide you speak of having used is probably a previous version of this one.

While some people -- the mentioned Windows Media Player users included -- may not care much for the quality of their sound as long as notes and voices can be heard, I'm always eager to help those who wish to preserve and revel in every small aural detail of their music!

There's nowhere I can't reach.

Hydrogen Audio currently recommends --vbr-new because of the noticably quicker encode speed and because in their listening tests, quality generally either slightly improved or was no noticeably different. I'd say go right ahead.

This thing is sticky, and I don't like it. I don't appreciate it.

Sure. Using the latest version of foobar2000 (0.9.4.5), open the Preferences tree under the File menu.
Find and click Converter, directly under Tools.
Check the existing Encoding Presets for the text MP3 (LAME) | 190Kbps | V2, fast.
If it already exists, use that and skip the next step.
If not, click Add New, select the MP3 (LAME) encoder from the drop-down, move the quality slider to ~190Kbps (*) V2, and activate Fast Mode (--vbr-new).

Now, whenever you right-click and Convert your music, select MP3 (LAME), 190Kbps, V2, fast from the drop-down, make sure ReplayGain Processing and DSP Processing are OFF, and let it start chuggin' away!

How ya doing, buddy?

Originally Posted by Lousy

Hello Moguta,

is the LAME V2 better than LAME CBR 192? Many/Most files which i encoded with LAME V2 have an average bitrate of ~160; only few reach an average of 190+. So those 160-VBR's, are they indeed better than a potential CD-rip to 192-CBR?

192-CBR (=average is 192) is higher than ~160 (=average is 160). Isnt higher better? That's why i am asking..

Thanks, best, Lousy

Higher bitrates means that more information is being stored. However, whether a particular higher bitrate is "better" is not an easy question to answer.

The short answer is that the ~32Kbps difference between your -V2 --vbr-new and 192Kbps CBR encodes is very likely a combination of CBR inefficiencies and inaudible information.

The entire point of MP3 is to shrink file sizes. VBR attempts to be as efficient about that goal as possible, by determining what bitrates are necessary for each file to encode them with a certain -V quality. More complex audio requires fewer bits, and less complex audio really doesn't need high bitrates (Note: Though true, that explanation is an extreme simplification). CBR, in contrast, just throws however many bits at a file that you tell it, without regard to how many it actually needs to sound the same as the original. It could be too little, or -- as it seems in your example above -- too much.

By the way, I would stick with -V2. It is a time- and test-proven quality setting. The VBR quality #s above that (-V1 and -V0) will likely begin to store more inaudible, unnecessary information, eroding the VBR's efficiency. Some people do prefer the "overhead", however, feeling safer even if there is no audible difference.

How ya doing, buddy?

Originally Posted by Lousy

Wow, great explanations. I'm enlightened. Thanks for the splendid replies!
Ah, me too i have a foobar related question... Does a plugin (component) exist for converting to Windows Media Audio (*.wma) instead of to LAME V2 (*.mp3)?
Some of my sets and rips are in *.wma, so i would want to convert the inet ape/flac's to wma ... just for the sake of homogenity of file extensions *g*
i googled. and i think that foobar2000 doesnt support the conversion TO wma.

Thanks. I could even go more in-depth, but I figured I'd spare you the long-winded explanation. ;p

And although I would recommend encoding to LAME MP3 rather than WMA, I did find a guide to do exactly what you ask:
How to set up Converter for WMA 9 - Hydrogenaudio Forums

Originally Posted by Rew

I'm not sure if this is the right place for this question, but I've recently ripped some game music into .AUS format. How can I convert .AUS files into WAV or MP3 files? Thanks!

I'm sorry, but I don't know what .AUS files are and couldn't find it in a quick Google.

What kind of toxic man-thing is happening now?

Basil, Audacity is not solely a converter, but is mainly an audio waveform editor. Rew seems to need the ability to fade his recordings as well as encode them.

Rew, perhaps you can try the trial versions of Sound Forge or Cool Edit. Although, I can't remember, one of their trial limitations might be that you can't save your work...

Also, do you notice any difference about the files that Audacity won't open? Is it audio from entire games that won't open, or will only some tracks in the same game not work? Do they have unusual sample rates? And can you play the problematic WAVs fine in your audio player?

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Originally Posted by Lousy

Just to let you know, i did that conversion (lossless ape -> 128kbps WMA) following the options given at that hydrogenaudio page, and i am very pleased with the resulting files. I really have the impression that WMA's at low bitrates are superior in (subjective) sound impression than MP3's (FhG) at the same CBR-bitrate.

LAME V2 is as good of course, since it is VBR, and some files are as small as the low-bitrate-but-great-sounding WMA-CBR's. It all depends on the lossless source audio material --- i guess.

Modern audio encoder implementations do seem pretty competitive around 128Kbps, as demonstrated by the results of this public ~128Kbps double-blind listening test in December 2005:



Although, this test isn't entirely relevant to your statement, since it used WMA Pro in VBR mode rather than CBR WMA. It's just too bad there have been no public double-blind listening tests performed with LAME's -V2 --vbr-new or --alt-preset standard modes. (In the above test, LAME is evaluted by its lower-quality -V5 --vbr-new setting.) I have heard that its simply too fatiguing for most people to try to reliably & repeatably discern between that level of quality and the original.

Additional Post:

Originally Posted by Rew

Thanks, everyone! Actually, I figured out a different sort of trick. The WAV files that would go nowhere in Audacity I opened in iTunes instead and converted them to MP3. Their sound quality remained intact, and this time, as MP3 files, I was able to play them in Audacity and do the five-second fadeouts that I wanted. So all that to say, problem solved!

Moguta: What was weird is that I couldn't find any commonality at all among the WAV files that Audacity wouldn't take. Tracks from literally the same folder of a game would convert splendidly, while a stubborn few just wouldn't at all, and no error messages were given either. Oh well. At least that's over with now. =0)

That is strange, about Audacity not opening files randomly. Perhaps you could report it to the devs and see what they make of it, for the future.

And in case you didn't realize, when you open the MP3 in Audacity and then re-save it after doing the fade-out, you are actually re-encoding the MP3. (WAV -> lossy MP3 -> lossier MP3 w/fade)

To preserve audio quality, you could try:
1. Download Foobar2000 and do a full install... or at least make sure that you install the Converter component.
1b. Download & extract the current FLAC and LAME encoders (links in the 1st post of this thread)
2. Add the desired WAVs to Foobar's playlist, then select them all and choose Convert > Convert to... from the right-click menu.
3. Select FLAC, level 5 from the drop-down box, hit OK, and wait for it to complete. The first time you convert, it will also ask for the location of the FLAC encoder you just downloaded.
4. Import the FLAC files into Audacity, then delete the FLACs once you have encoded to MP3. Alternately, since Audacity only encodes in outdated CBR mode, you could have Audacity export the faded audio to WAV and use Foobar2000 to convert them into efficient, high quality VBR MP3s. Just add the WAVs and proceed like you converted to FLAC, but instead selecting MP3 (LAME), 190 kbps, V2, fast in Foobar's converter.

EDIT: Oooops, I forgot that Foobar's converter doesn't include the encoders themselves! Updated it to work.

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In your first image, everything after --vbr-new and before %s %d is tagging parameters. Note that %a, %t, %g, and such are all EAC-specific parameters that EAC replaces with each track's values before it actually passes that command line to LAME.

To respond to each of your needs:

[1] There's no need to go into the Custom encoder mode in Foobar for this. Just choose MP3 (LAME), 190Kbps, V2, fast when encoding or converting. If it's not already in there, just select MP3 (LAME) for the encoder, drag the quality to the ~190Kbps, V2 tick, and make sure Fast Mode (--vbr-new) is checked. Foobar itself takes care of copying all the necessary tags.

[2] For this, you do need to go to Custom for the encoder. Delete -V2 --vbr-new from the existing LAME command line and replace it with -b 192 instead.

I have to mention, it is strongly recommended not to encode in CBR, unless doing so for an old device that literally does not support VBR. Even using average bitrate mode (ABR) is an improvement over CBR, and gives you the bitrate predictability that quality-oriented VBR lacks. (Encoding at an average bitrate of 190Kbps, for example, would be done --abr 190)

EDIT: Since you were curious, and I didn't know, I looked up the -S parameter in LAME's help. It simply suppresses the text-based encoding progress report (which is what you see when ripping with EAC), because Foobar has its own graphical progress meter. And --noreplaygain simply means it doesn't calculate the track ReplayGain after encoding every file, which is fine. You'd have to use another program to calculate the album ReplayGain anyway.

How ya doing, buddy?