Best Possible Way to Rip MP3s

[Moguta]

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 (http://www.rockbox.org)), several iRiver models, serveral Cowon models, Pocket PCs (with TCPMP (http://www.hpcfactor.com/downloads/tcpmp/)), and many others (http://wiki.xiph.org/VorbisHardware)
PC PLAYERS: WinAmp, Windows Media Player (w/filter (http://tobias.everwicked.com/)), Foobar2000, and many others (http://wiki.xiph.org/index.php/VorbisSoftwarePlayers)


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 (http://www.rockbox.org)), Pocket PCs (with TCPMP (http://www.hpcfactor.com/downloads/tcpmp/)), and others (http://flac.sourceforge.net/links.html#hardware)
PC PLAYERS: WinAmp, Windows Media Player (w/filters (http://www.losslessaudioblog.com/?p=40)), Foobar2000, and others (http://flac.sourceforge.net/links.html#software)


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 (http://www.exactaudiocopy.org)

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 (http://www.rarewares.org/mp3-lame-bundle.php#lame-current)
Ogg Vorbis: Oggenc2.85 using aoTuVb5.7 (http://www.rarewares.org/ogg-oggenc.php#oggenc-aotuv)
FLAC: FLAC for Windows with installer (http://flac.sourceforge.net/download.html) (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 (http://www.foobar2000.org) 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 (http://www.rarewares.org/mp3-lame-bundle.php#lame-current)
Ogg Vorbis: Oggenc2.83 using aoTuVb5.7 (http://www.rarewares.org/ogg-oggenc.php#oggenc-aotuv)
FLAC: FLAC for Windows with installer (http://flac.sourceforge.net/download.html) (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 (http://mp3gain.sourceforge.net/) 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 (http://www.listening-tests.info)
- Roberto's Public Listening Tests (http://www.rjamorim.com/test/index.html)

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? :D
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 (http://wiki.hydrogenaudio.org/index.php?title=LAME#VBR_.28variable_bitrate.29_se ttings) 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 (http://www.hydrogenaudio.org) 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.

[Ian Gillseed]

Okay, I've done all of that. Now...when I want to burn these "exact audio copy" mp3s onto a disc, any recommendations as to the best program(s) for that purpose? Or are VBR mp3s impossible to burn accurately?

[Eriol]

Originally posted by Ian Gillseed
Okay, I've done all of that. Now...when I want to burn these "exact audio copy" mp3s onto a disc, any recommendations as to the best program(s) for that purpose? Or are VBR mp3s impossible to burn accurately?

Now that you have mp3s, you have files, which most CD burner software can handle.

Ahead Nero is among the better burner programs.

[Jodo Kast]

I'm curious - If you have a CDR or the original CD, of what use would mp3's be? (other than posting them online.)

People that rip their cds into the mp3 format and listen to the mp3s is analogous to a hypothetical situation in which people would take their DVDs and reduce them to the quality of VHS. What's the point? - you already have the best. Why break it down?

The mp3 format will be useless ten years from now, in my opinion. Storage capacity and data transfer rates will have reached the point where compression is not necessary.

[Ian Gillseed]

That seems true. In my case, I would be ripping one track from a CD I no longer wish to keep. After I have a CD-sized collection of tracks from various other CDs, I'd burn those tracks into one mix disc. Right now I've got Adaptec's Easy CD Creator, merely because it came installed on my HD. If there are better burning software packages, however, I'll change in a heartbeat.

[Moguta]

Originally posted by Jodo Kast
I'm curious - If you have a CDR or the original CD, of what use would mp3's be? (other than posting them online.)

People that rip their cds into the mp3 format and listen to the mp3s is analogous to a hypothetical situation in which people would take their DVDs and reduce them to the quality of VHS. What's the point? - you already have the best. Why break it down?

The mp3 format will be useless ten years from now, in my opinion. Storage capacity and data transfer rates will have reached the point where compression is not necessary.
Well, that's why some people prefer to use lossless compression, such as Monkey's Audio ( www.monkeysaudio.com ). But as of now, if you have a huge music collection, you really can't even afford to store them all in APE.

I agree, the MP3 format will be useless in ten or so years. But by then I imagine there will be multi-channel recordings, and other enhancements, so that audio files will be even larger. Then, OGG will have a use for its multi-channel capabilites. ^^

One thing to consider, too. Ten years from now, people will still have dialup modems, so don't think everyone would be able to have the bandwidth to download WAVs. =p

[Lentium]

If technology has not evolved enough to give everyone affordable broadband internet access within 10 years, then we're doing something critically wrong here.

[Eriol]

Originally posted by Lentium
If technology has not evolved enough to give everyone affordable broadband internet access within 10 years, then we're doing something critically wrong here.

In 10 years, we may see the installation of Digital Rights Management systems, making everything very inconvenient. In 10 years, more Draconian copyright laws could be in effect.

We might as well be using modems.

[Moguta]

EDIT: There was something else completely different here when I posted it. *is boggled*

[Cain]

Anywho, I just used this method and I must say I'm very pleased. Good FAQ, Mogu ;)

[BumpsTuscadero]

Worked great for me, too. I got a great copy of the Warcraft III soundtrack via that method. :D

And all this nonsense about Palladium and these steps to hinder free file trade, at least in a few senses...it's not only confusing, but worrisome.

[Eriol]

You see, if they make everybody use the hardware and outlaw non-complian hardware, what will we use?

It's kind of hard to crack hardware for anybody other than a determined, and skilled person. If the restrictions are systemwide on a computer, the problem is more difficult.

[SephVBC]

There are new mp3 headers used by LAME to specify which method was used, the quality, and existence of the LAME header extension

LQUA: Lame quality, a number from 0...100
LAM: LAME header existance
MET: LAME VBR encoding method

This is only for people who know how to program and make use of the variables, I'd imagine EncSpot will support it shortly, if it doesn't already in Pro.

** I just tested EncSpot Professional v2.0 and it has support for the header tags, but gives some weird info like vbr-old / vbr-rh. Doesn't matter if it's APS, APE, or r3mix, it gives that. **

*** Also ... I found these on the hydrogen audio boards, as to being best able to determine whether it was encoded in aps or ape or r3mix in EncSpot

lame 3.92: --alt-preset extreme

quality: 78
vbr-old / vbr-rh
lowpass: 19500
nspsytune
safe js: yes
ath: 2
min: unknown (unless -b set)
n.s: 2

lame 3.92: --alt-preset fast extreme

quality: 78
vbr-mtrh
lowpass: 19500
nspsytune
safe js: yes
ath: 2
min: unknown (unless -b set)
n.s: 2

lame 3.92: --alt-preset standard

quality: 78
vbr-old / vbr-rh
lowpass: 19000
nspsytune
safe js: yes
ath: 4
min: unknown (unless -b set)
n.s: 2

lame 3.92: --alt-preset fast standard

quality: 78
vbr-mtrh
lowpass: 19000
nspsytune
safe js: yes
ath: 4
min: unknown (unless -b set)
n.s: 2

lame 3.92: --r3mix

quality: 88
vbr-mtrh
lowpass: 19500
nspsytune
safe js: no
ath: 3
min: unknown (unless -b set)
n.s: 1

This was a reply a few lines down...

"In regards to detecting settings used in LAME via Encspot, this works for everything except for the --alt-presets because the LAME tag (which I've stated before as being poorly designed), does not provide for storing information on any of the internal psymodel tweaks that the --alt-presets use beyond standard command line options.

The best you can get is that it's possible that an mp3 was encoded via --aps if it meets certain criteria, but you cannot be positive because it's quite easy to create an mp3 with all of the normal settings of --alt-preset standard, but without any of the code modifications.

For example, try using --alt-preset standard, and then --alt-preset standard --no-preset-tune to create 2 mp3s of the same clip. The first will use all of the internal tweaks the normal --alt-preset standard vbr modes turn on, but the second will only use the normal LAME settings, disabling the extra stuff.

From encspot, these mp3s should look identical in regards to the LAME tag stuff.

Also, the quality index and color coding are useless as a guide to actual measured quality. For example, I believe --r3mix generates a higher quality index than --alt-preset standard. "

You can safely assume, however, with game music that people aren't changing the settings and they're just reading the FAQs. Unless they really know what they're doing, or they're audiophiles, then they're probably going to default to --alt-preset standard.

[matrixPlus]

wow.. those are some complex directions. I just use cannamp program that I have had for years actually that was freeware and does the job just peachy. I haven;t ever had issues reburning if need to audio cd. I guess if anyone wants the application I could post somewhere. I guess if you would like send me a PM.

[San]

I'm having a few problems. When I push Alt-G to get the tracklist, it asks me if I'm sure I want to delete the data on my audio CD. Also, it doesn't seem to compress very small, at 6 megs for a 4 minute song. Did I do something wrong?

[San]

I ripped my first CD, following your method, and I'm not very happy. When I try to play the MP3s, there is no sound. Why is this? Is there a way to compress them for a lower file size, since I'm not really that big of a quality freak?

[Avaxx]

You can always use the outdated --r3mix format if you -must- have a VBR mp3. If not, you can always just clear the custom command line in EAC and select 192 kBit/s (or 160/128, depending on just how little you care about quality) "Bit rate: [ ___ Pull Down Window ____ ]"

Also, using this format you aren't going to get the same compression you will with a 128cbr mp3 and the reason for that would be that this format is much better than 128cbr (Technically, anyway) and thus the file sizes are bigger.

The file size does vary from file to file as, unlike a CBR MP3, each MP3 has a different bit rate. Usually a 128cbr is a little under a MB per minute of music. With VBR there is no such standard as that.

If you truly can't tell the difference and have a limited harddrive I cannot see any reason why you should feel that you must use high-quality VBR. It's up to you, I s'pose.

[thebeaver]

Originally posted by San
I'm having a few problems. When I push Alt-G to get the tracklist, it asks me if I'm sure I want to delete the data on my audio CD. Also, it doesn't seem to compress very small, at 6 megs for a 4 minute song. Did I do something wrong?

It depends on what kind of compression you selected, 128 kbps usually runs about a 1 per minute while 192 runs about 1.5-1.75 megs per minute if not larger. If you want better sounding files at lower file sizes I recommend using Ogg Vorbis.

[ckain]

Even better. Why don't we use lossless compression (http://flac.sourceforge.net/)? :)

[SephVBC]

A number of reasons.

1.) Compressed, but only to about 67% on the highest and longest compression methods
2.) Takes too long to transfer on current broadband bandwidth limitations. Sure, you might have uncapped download and upload on your cable modem, but for how long?
3.) The audible gain in the differences between lossless and this method of variable bitrate are pretty slim. You need to have a good ear, and a good setup to tell the difference. The average user with a stock stereo for his PC, or even a gaming system speaker set, won't notice the difference.
4.) Lack of available addons and stuff for lossless. APE has tags, but not as extensive as mp3 does (Yeah, picture and lyrics are overkill, but stuff like composer, url, copyright aren't)
5.) Goes hand in hand with #1. Larger files means you need more hd space. Lossless albums vary depending on what you're encoding, but can take up anywhere from 100mb to 500mb per disc. That adds up quick, and unless you have 300$ to be shelling out for a new 120gb hd every couple months, again, not worth it.

You're welcome to encode in lossless, but currently you're going to be hardpressed to find users switching from mainstream mp3 to lossless (or even a lossy codec like ogg or aac) for increases in quality that they probably wouldn't even notice.

[ckain]

Well yep. You're right SephVBC

But what about Ogg Vorbis? Can't people just move on to a better, than just say "MP3 is popular."

[SephVBC]

Well, with ogg, the encoding time is about the same, the sound quality is about the same (very very slight differences), the size is about the same (ogg is smaller, yeah)

But, the support is weak. It's open source, so it'll get bigger, but I can't see anyone switching to ogg either, as it's only very slight differences/improvements.

[Yushiro]

Hmm, awhile back I read ripping the tracks to WAV with EAC, and then encoding them using RazorLame was the best way. What makes using EAC to encode better?

[Eriol]

Originally posted by Yushiro
Hmm, awhile back I read ripping the tracks to WAV with EAC, and then encoding them using RazorLame was the best way. What makes using EAC to encode better?

EAC is simply an audio ripper. You can tell it that you want to use Lame (Razorlame is a GUI for Lame) for mp3 encoding, so after EAC rips the audio track, it can immediately call Lame to encode.
Therefore, you end up with a one-stop shop process: Rip to encode in one pass.

EAC is considered one of the finest rippers, because it takes into account a lot of minute facets of CDR/CDRW/CD-ROM/DVD-ROM drives. It's able to rip the audio in a very precise way with error correction abilities and all the other good stuff to ensure you don't get any pops, clicks, skips, or static in your WAV file. The program is not meant for the "novice" because there are a lot of settings.

[Ninja]

Thank you for remaking this I have been looking for it but could not rember who made it.

[Moguta]

Originally posted by SephVBC
Well, with ogg, the encoding time is about the same, the sound quality is about the same (very very slight differences), the size is about the same (ogg is smaller, yeah)

But, the support is weak. It's open source, so it'll get bigger, but I can't see anyone switching to ogg either, as it's only very slight differences/improvements.
Not quite. OGG is not tuned for high archival-quality, like the --alt-presets are for LAME MP3.

OGG Vorbis, however, does give *much* better quality than respective MP3s at lower bitrates (i.e. less than 128Kbps). And it's not a proprietary format like WMA.

EDIT:
Originally posted by San
I ripped my first CD, following your method, and I'm not very happy. When I try to play the MP3s, there is no sound. Why is this? Is there a way to compress them for a lower file size, since I'm not really that big of a quality freak?
What MP3 player are you using? WinAmp is able to play VBR mp3s, and so should Windows Media Player.

EDIT 2 (after reading backwards yet again!)
Originally posted by San
I'm having a few problems. When I push Alt-G to get the tracklist, it asks me if I'm sure I want to delete the data on my audio CD. Also, it doesn't seem to compress very small, at 6 megs for a 4 minute song. Did I do something wrong?
It is not asking you if you want to delete the data on the CD, that's impossible (besides scratching the CD yourself =p). It's just asking if you want to clear the tracklist that EAC already has showing.

Also, 128Kbps MP3s have about 1MB/min compression. Even 1.5MB/min is great, considering WAVs are about 10MB/min.

[fausty]

edit - never mind, third time was the charm.

[Grass-eatin'me]

I don't understand why everybody should use preset standard & extreme, I think good results come up without using those presets. And ogg is superior to mp3, in quality/bitrate, and size, though it's only 300Kbs difference for 192 KBps (or something like that, cannot remember...) It will begin to serve alot to you with your 200 albums (200albums x 30tracks = 6000, 6000 x 300 = 18.000.000kb) this way you can conserve 18.000.000kb, or 18 Gigabyte. With even higher quality!

Haha, I love this stuff.
(Still use .mp3s cause'of the tags!)

[SephVBC]

Uh, that's a savings of 1.8gb, do your math. 1,800,000kbytes = ~1800mb = ~1.8GB. Besides, not every OGG file is smaller than every mp3 file. You're comparing apples and oranges; it's like saying 300 apples will fit into a box but 300 oranges won't. They might, they might not.

Besides, OGG is not better than mp3 at any bitrates except low quality ones. You cannot accurately compare a 192Kbps OGG file to a 192Kbps mp3 file. They are not the same.

The only thing ogg is good for is the fact that it's free (and maybe freeform tagging, but seriously, who really needs freeform tags? id3v2 has everything you'll ever need), everything else is either too close, or a superficial difference in quality. All of which, might I add, do not warrant a complete change of format. Mp3 is fine, and will not be usurped anytime soon.

[Grass-eatin'me]

Originally posted by SephVBC
Uh, that's a savings of 1.8gb, do your math. 1,800,000kbytes = ~1800mb = ~1.8GB.

Whooops!

Originally posted by SephVBC
Besides, OGG is not better than mp3 at any bitrates except low quality ones. You cannot accurately compare a 192Kbps OGG file to a 192Kbps mp3 file. They are not the same.


Are you talking about the fact that a 128KBps (when encoded) ogg file turns out to be a 111KBps (average) when it's done?

I say that ogg is better, but I do not have a really good reason for why, if not my own listening experience is enough.

[Moguta]

Originally posted by Grass-eatin'me
I don't understand why everybody should use preset standard & extreme, I think good results come up without using those presets. And ogg is superior to mp3, in quality/bitrate, and size, though it's only 300Kbs difference for 192 KBps (or something like that, cannot remember...) It will begin to serve alot to you with your 200 albums (200albums x 30tracks = 6000, 6000 x 300 = 18.000.000kb) this way you can conserve 18.000.000kb, or 18 Gigabyte. With even higher quality!

Haha, I love this stuff.
(Still use .mp3s cause'of the tags!)
Okay, your post doesn't make much sense to me.

1) --alt-preset standard gives the best balance of filesize vs. quality for MP3s. There really is no (tested & confirmed) better setting, although there are settings that will result in smaller files for less quality, yes. And I stand by the advice of the LAME developers that there is barely a difference at all between the sound quality of standard & extreme, so save your HD space with standard.

2) Ogg Vorbis is superior to LAME MP3s only at lower bitrates. Ogg Vorbis has not (yet) been tuned for high quality, and the developers know this.

3) You still use MP3s because of the tags? What's wrong with Vorbis tags? If you want a "composer" or "orginal artist" tag, you can add your own.

And Seph, I seem to remember the LAME developers calling ID3v2 tags "critically flawed", although I'm not sure for what exact reason.

[SephVBC]

It's not critically flawed, the problem is with developers who decide to use it. The difference in the two standards make it hard(er) to write a program to do an id3v2 tag than an id3v1, thus making more errors. We've all seen retarded id3v2 tags, but there's little to no chance of error with id3v1, since programming them is a snap.

[Grass-eatin'me]

Because of batch tagging: there're a lot of batch taggers for mp3s but not that many for oggs, easy as that.

And btw, I just checked (in a way...) the preset data, for extreme and standard, both of them are 32-320KBps, 16bit, vbr quality 2.
So, at the lame screen, (after typing --standard) is it any noticable difference? I'm asking about this because it would be nice to know in case I should use the dll, for any reason :P

[Moguta]

Originally posted by Grass-eatin'me
Because of batch tagging: there're a lot of batch taggers for mp3s but not that many for oggs, easy as that.

And btw, I just checked (in a way...) the preset data, for extreme and standard, both of them are 32-320KBps, 16bit, vbr quality 2.
So, at the lame screen, (after typing --standard) is it any noticable difference? I'm asking about this because it would be nice to know in case I should use the dll, for any reason :P
There *is* a batch tagger for Vorbis (works with about every compression codec), but it's commandline, intended for Linux use. There are two frontends for it, though, listed on the homepage.
Tag (http://www.saunalahti.fi/~cse/html/tag.html)

The second part of your post... I can't figure out what you're asking.

[Grass-eatin'me]

What I mean is this: When you use lame.exe, and sets the preset to standard, the encoding will start, and show what settings that is used for the conversion. In standard, the settings are 32-320KBps, q2. And it is exactly the SAME settings for extreme.
I don't think I can explain this any better :P

I'll be sure to check out the tagger you told me about.. :D

[Moguta]

Originally posted by Grass-eatin'me
What I mean is this: When you use lame.exe, and sets the preset to standard, the encoding will start, and show what settings that is used for the conversion. In standard, the settings are 32-320KBps, q2. And it is exactly the SAME settings for extreme.That's because this short description of the encoding settings can't tell you the all the differences in encoding being used in the --alt-presets. There's more going on behind the scenes in the encoder than those couple of lines tell you. (EDIT: Basically, what I'm trying to say here is that the --alt-presets can't be replicated with any set of commandline options, like the old --r3mix could be.)

Actually, --alt-preset extreme has a higher lowpass filter, if you'll notice. It's listed as the "transition band." I believe the higher lowpass is the only real difference between standard & extreme; most people (read: 99%) won't be able to tell any difference between the two settings.

[TerraEpon]

--alt-preset standard gives the best balance of filesize vs. quality for MP3s. There really is no (tested & confirmed) better setting

Depends on the music, really. For most music that most people care about (ie pop/rock/hiphop/etc), as well as most VGM (ie, chip-synth), it's true.
But get into instrumental music, especially orchestral or other ensembles where the sounds of the varied insrtuments matter, and extreme is the way to go instead of standard.

-Joshua

[Moguta]

Originally posted by TerraEpon
Depends on the music, really. For most music that most people care about (ie pop/rock/hiphop/etc), as well as most VGM (ie, chip-synth), it's true.
But get into instrumental music, especially orchestral or other ensembles where the sounds of the varied insrtuments matter, and extreme is the way to go instead of standard.
Have you confirmed this by doing actual blind listening tests? Or is that your "impression?" Really, orchestral music is typically less demanding in regards to audio compression. Just because it might give you files with significantly lower average bitrates doesn't mean it's any worse.

[Grass-eatin'me]

Thanks for the info, mog!
I ave the impression of that more ambient music (songes with great echo and such) , needs higher bitrates. But I do only think that out of a single cd I encoded in standard, and then listened to in standard... So I really don't know if it is a difference.

[Moguta]

While --alt-preset standard isn't flawless (neither is extreme, for that matter), I doubt it will fail just because the music has a lot of reverb.

If it gives low-bitrate files, then the encoder has decided that the music only needs such low bitrates to sound accurate. Also, just because something *sounds* complex to you does not mean it will be complex to encode & thus need higher bitrates.

And LAME --alt-preset standard has been tested with all kinds of music, including instrumental and symphonic and ambient.

[Grass-eatin'me]

I tried encoding the album I spoke of, in extreme.. and no difference occured (there was a HUGE difference between the regular CDA, and the MP3, believe my words!) So, allright you have convienced me.

[Moguta]

Originally posted by Grass-eatin'me
(there was a HUGE difference between the regular CDA, and the MP3, believe my words!)
A huge difference in sound quality? There really shouldn't be... If you're using WinAmp's equalizer, different internal EQs are used for MP3 & PCM (CDA isn't the CD format, it's how computers see an audio CD's table of contents). Turning the equalizer off should solve that.

Unless you have razor-sharp ears and decent sound equipment, there shouldn't be a HUGE difference... :confused:

[Grass-eatin'me]

Perhaps not a huge difference in quality, but it sounds different. (with that extreme dynamic echo) I have rather uber-sharp ears, and decent head-phones.

[Moguta]

I think you probably hit upon one of the major MP3 artifacts, "pre-echo." ^^

You can always try using Musepack (MPC) instead. ;)

[Grass-eatin'me]

Aren't those MPCs huge in size? Perhaps I should try encoding some...

[Moguta]

At --standard --xlevel, MPC is smaller than --alt-preset standard, encodes faster, and has fewer quality flaws.

The only disadvantage to MPC is it's lack of support. There is a WinAmp plugin though.

MPC encoder/decoder, plugin, recommended settings:
http://www.hydrogenaudio.org/show.php/act/ST/f/11/t/1927

[Grass-eatin'me]

So, now you're telling me to use MPC instead of preset --standard? If --standard --xlevel is smaller, encodes faster and have fewer quality flaws than lame standard, I see no reason to use MP3 anymore :)

[Moguta]

Well, as I said before, it has very little playback support. Portable players certainly aren't going to support MPC anytime soon, compared to the dominance that MP3 has.

If you have no problem with that, then encode away!

[Grass-eatin'me]

I'll still use OGG, as the only times I actually encode something is when I rip PC Games, and those are almost always at 22khz, and thus fitting very well for ogg.

[Flint]

Did somone already mention this (http://www.saunalahti.fi/cse/EAC/index.html) Musepack guide?

[Moguta]

I think the Hydrogen Audio thread on Musepack's recommended settings covers that. (See link a few posts up.)

[Helmholz]

I was ripping a song and it took my computer over an hour to do just one song. It ripped the wav quickly, but compression took forever. I have a 1.3 GHz Celeron, so I don't think it's my processing ability. Any ideas?

[Moguta]

The compression to MP3 took nearly an hour?? That's very odd considering your processor speed...
What method did you use? --alt-preset standard?

[Helmholz]

I quit a few more unnecessary programs and it works like a charm. I was using extreme, and I love the quality. Thanks.

[Mahew]

It seems the Exact Audio Copy (http://www.exactaudiocopy.de/) site has been shut down. Has development been discontinued? What does it say?

[BAMAToNE]

Dear God I hope not. :\

[Moguta]

I was worried too, until I visited Hydrogen Audio & found out the domain was changed.

http://www.exactaudiocopy.org/

I will update my post. (=

[Moguta]

The guide has been given a multiformat overhaul! =D
(see page 1)

[James]

Sorry could i ask a quick question about FLAC. I was wondering how to burn flac files to make a perfect copy of the original audio cd. How do you decode flac files and what program could you use?
Thanks in advance.

[Lowe]

To decode FLAC's you could use the FLAC Frontend (http://members.home.nl/w.speek/flac.htm)
Or to burn FLAC's Download the nero plugin (http://neroplugins.cd-rw.org/files/nxmyfla.zip)

[James]

Thats great thanks, i just cant get the nero plugin to download, is there a problem with the link. Thanks for the help though.

[Lowe]

hmm maybe.

[James]

Thank you so much for the help!! That works perfectly!
Regards

[Moguta]

To make a perfect burn, you'll also want to make sure that Nero is set to not put gaps of silence between each tracks. While on most CDs this setting wouldn't be a big problem, 2-second pauses will quite noticably affect albums that have continuous tracks. ^^

[The Raven]

Thanks for the instructions, they were helpful. But.. Could someone tell me how I can rip a whole album to one big OGG file (with EAC)?

[Moguta]

Why would you want to extract it into one huge Ogg Vorbis file? They will play back gaplessly even if the CD tracks are extracted as individual Ogg Vorbis files.

[The Raven]

They will play back gaplessly even if the CD tracks are extracted as individual Ogg Vorbis files.
Not on my PC, at least not with Winamp. :( And how could they? It's not like the gapping results from the files themselves, right..? But if you know how to play individual music files gaplessly, please tell me. ^^;

[Lowe]

If there all on the play list they should just roll into each other, unless it's a cd-r someone sent you or something and they made 2 second gaps beetween the tracks, then no matter what way you rip it there will always be gaps.

[The Raven]

Ok, I changed the DirectSound output plug-in's "Buffer-ahead on track change" settings in Winamp, and it helped - at least a little. Maybe I should try Foobar 2000 or something..

Thanks for your answers though. :)

[Moguta]

I have WinAmp's "Buffer ahead of track change" set to 666ms, and it has always worked fine like this.

You could also try engaging the silence remover in the same DirectSound options to see if that helps. I'd put the scale somewhere low around -80dB so it doesn't cut out the end of fading songs.

Foobar should definitely play those gapless. Personally, I don't like it. It's a great audio file utitility IMO (ReplayGain, Masstagger, Renamer, Converter), but the player interface needs improvement.

[Kuumies]

Hmm, I have problems with MPC-codec. All my compressed files have "wrong" tags. Like title is %t and artist is %a etc. How can I solve this thing? I use EAC.

Somehow it doesn't understand this additional command line, I believe?
--standard --xlevel --artist "%a" --album "%g" --tag title="%t" --tag year="%y" --tag genre="%m" --tag track="%n" %s %d

Any idea for help?

-Kuumies-

[Moguta]

I haven't updated the guide here for a good while, just brought it up to date now.

Try this slightly different command line.--standard --xlevel --artist "%a" --title "%t" --album "%g" --year "%y" --track "%n" --genre "%m" %s %d

Also, make sure you have "User defined encoder" selected as the parameter passing scheme, not "MPC encoder". If you have all the CD info entered into EAC, then it should substitute the artist name for %a, track titles for %t, and so on.

[Adaman]

Good timing, the first time i needed to use this guide and now it's updated =p

Why is LAME 3.90.3 recommended over the newer versions?

[Kuumies]

Yeah, it works now. Thanks a lot, Moguta!

-Kuumies-

[DML]

Hmm...just browsing over the forums when I noticed Adaman's question went unanswered. I'm guessing you've gotten it answered somewhere by now, but incase you haven't...aside from the fact everyone says the newer versions haven't been tested as thoroughly (which they haven't), the newer versions of LAME encode the files differently so they encode significantly faster (it takes no more than 30 seconds for any given mp3 file to encode with v3.96). The way the algorythm handles the file is different with each version, so 3.90.3 is good for high-bitrate variable mp3s, whereas the newer versions of LAME are good for low-to-mid bitrate mp3s. :)

[Adaman]

I didn't know that...i figured that was the case. Thanks DML (dark monkey lord? =p)

I never got around to ripping a couple of CDs i have in VBR, EAC said they were going to take like 10+ hours to finish...does that seem right? I ripped the Actraiser Symphonic Suite album in under 3 hours i think.

[DML]

Well, depending on the quality of your CD-ROM drive, it can take up to that long especially if it's an older one. I have a 52x LiteOn burner, and it takes up to 2-3 hours for a good length cd. Newer and faster drives should be able to handle it faster. That's the price to pay for when ripping such high quality mp3s. I've never ripped into APE or FLAC or anything of the sort, so I don't know how long that would take.

The newer versions of LAME are by no means bad, but if you're dealing with high-end bitrates and you're nuts about high quality, then the ripping time may be worth the sacrifice. As for 10 hours on your side...good lord, if that's an accurate time, that's nuts. Try running through your settings once again to double check you don't have some funky setting that's throwing off the speed there. Even though my burner takes 2-3 hours to rip a cd, I usually rip my cd's overnight or when I'm out (class, friends, etc.). I can't stand waiting for that thing to finish. :)

[Adaman]

Well, it burns at 0.1x speed or something, i think...i haven't done it in a while. I'm using the suggested settings. I think i have it at 320 VBR instead of 192.

[DML]

0.1x, eh? Hrmm....I think mine goes at like 2x or something, if memory recalls. I don't have any cd's at hand that need ripping (they're in the mail stiill :)), but as soon as I start ripping them, I'll let you know what speed mine goes at. Last cd I ripped was my Lost in Translation OST, which took 2 hours and 4 minutes (I still have the log file, heh). Still, like I said before, I usually rip my stuff overnight, so I don't really pay attention to the numbers. Obviously it's going to vary based on cd length and quality of disc, but I don't think it should take more than 3-4 hours for any given disc that is in good, readable condition.

[Adaman]

I tried on a different computer and now have the opposite problem, it's burning too fast =p, like at 9.7x or something, i'm not sure if that's a problem necessarily, but it seems like it should be burning slower, to be more accurate and to preserve the quality as much as possible. The problem i think is that i tried that using the newer versions of EAC and Lame, which probably have substantial differences from the versions of the programs the guide was written for.

The MP3s sound find as far as i can tell though, maybe they came out how they should of. The peak levels were different for a lot of the tracks, perhaps that's just how it is.

[DML]

Strange, 9.7x, eh? I don't know what to tell you about that - I've never gotten that high of a rip speed. If it's worrying you, you should be able to manually configure EAC to rip at a slower speed, I think. As for the peak level in EAC, its specifically for the audio quality on the cd, not the quality of the rip. The peak level can be anything, but the track quality is what's important when ripping. Mine always rips to 100%, with the occasional 99.9%.

What version are you using of LAME? 3.90.3 is the best for high-variable rips and by far the slowest. If you're looking just to rip something in average-to-good quality, v3.96 flies when ripping. A whole cd should take no more than 5-10 minutes at that setting, but the very high-end suffers slightly (at least as shown by tests).

[Adaman]

Yes i was testing out 3.96.1. I'm going to try and rip with the older version of Lame, i think it will work.

*later that day*

Well i tried 3.90.3 and it still ripped at 8+x speeds, it must be because i have a newer EAC or something...i dunno. Anyways, the track qualities all read 100%, so it can't be too bad.

[hyrule_princess]

I've had EAC for a couple months, didn't really have problems with it until now. I basically left the settings alone and used the default ones for ripping cd's to mp3's. Then yesterday I decided I wanted to rip some tracks into flac. So I did everything exactly like the first post in this thread said, I changed all the settings and checked that it was correct. I then tried to rip a song, and was very fast. So then I played the file in Winamp and the file was actually a wav. So I checked everything again and it looked just like it should have. I tried another track and the same result. I then looked at other websites for an answer but it all said the same thing, and that those parameters should work.

I tried many tracks with no success, so I gave up and put my old settings of lame & mp3 back into EAC. I then tried to rip a track to mp3, and that didn't work either!!! The file would result in a wav, not mp3. I made sure everything was correct, and tried again. Still wav files. So I deleted EAC from my computer completely, even deleted the registry key. I then unzipped EAC again and started from scratch. I installed lame and everything. I then ripped a track to mp3 and it worked! OK so then I get EAC ready for ripping flac files again, try it, and it doesn't work like before, now also ripping to mp3 doesn't work anymore, just like yesterday.

Do you know what causes this? I know when I access flac.exe my computer slows down drastically, like it freezes for a little bit. Do you need a lot of free RAM/memory to rip to flac? Or am I doing something wrong? Please someone help, I'm losing my sanity. Thank you in advance.

(Sorry this post is so long.)

[DML]

Hrmm...very strange. I've personally never used FLAC myself, but it is lossless. Then again, I thought FLAC was .mpc, not .mp3 or .wav, but I could be wrong - not sure on that.

Anyways, try checking out some other websites for information on ripping to FLAC also. You might get more information there on your problem.

[Secret Squirrel]

The only thing I can think, is that you may have clicked on the files and said 'Copy / Uncompressed', which produces .wav files. 'Copy / Compressed' makes the mp3s.

[Grass-eatin'me]

Yes, that is the only thing I can think of as well.

And DML, FLACs extension is .flac, .mpc is Musepack. :p

[DML]

Ahh, gotcha. I don't really use anything other than .mp3 myself or direct cd-r copies, so that's why I didn't know that. :)

[hyrule_princess]

I'm back. Sad to say though, after trying everything again, and doulbe checking myself, things came out the same. I don't know what it is, I guess it's just hard luck. Though at least I can burn mp3's. I'll keep trying though, and thanks everyone for the suggestions and help. :)

[Moguta]

Yikes, I've so forgotten about this thread! I'll respond to all your questions when I have more time on my hands...

[Musouka]

Thanks for the post... I should give the program a try in the near future and see...

I don't see what some got against the MP3 format. The premise of MP3 encoding is that it removes the sounds out of the ear's frequency range of hearing i.e. sounds less than 15-20 hertz and sounds more than 20,000 hertz. It would remove the sounds we cannot hear. So instead of recording every sound there, it tends to be selective.

Personally, I have been using MusicMatch to rip my CDs at 160 kbps (CD quality is 128, 160 comes with over-sampling). The program is rather fast with 12 to 15X. I can rip a CD in 5-10 minutes. I can hear no difference at all in the quality between the original and the rip even if I set the volume to the highest level. Perhaps the only thing MP3 is failing in is 5.1 CDs but even these need special equipments not available to everyone.

The format is widely supported and I can hardly see it dying. It may evolve and thus require less space, but dying; I don't see that coming. The computers' CPUs and Hard Drives keep growing in size and connections keep getting faster. At the same time, companies try to find ways to reduce file size. That is a puzzle in itself.

[Moguta]

I don't see what some got against the MP3 format.
The format is set in stone and aging. Even LAME, the encoder that has been developed the most extensively in recent time, cannot overcome the limitations of the MP3 format specification.

The premise of MP3 encoding is that it removes the sounds out of the ear's frequency range of hearing i.e. sounds less than 15-20 hertz and sounds more than 20,000 hertz. It would remove the sounds we cannot hear. So instead of recording every sound there, it tends to be selective.
Yes, the goal of MP3, and any other lossy encoding format, is to save space by removing all inaudible audio & keeping intact what is audible (this is referred to as being "transparent" to the original audio). However, there is a lot more to it than what you're saying. At 128Kbps, most encoders tend to lowpass around 16KHz rather than 20KHz, which can be distinctly audible as a lack of "sharpness". Also, I don't know of any encoder that highpasses around 15-20Hz, or one that highpasses at all. Most subwoofers won't reproduce frequencies that low, and I don't think much music even contains such low frequency content.

Even with the lowpassing, most of the space savings in MP3s come from other things. One is frequency masking, in which a higher-frequency tone with lower amplitude cannot be heard because of the simultaneous presence of a lower-frequency tone with higher amplitude. Encoders use simplified (for speed) psychoacoustic models to detect what audio content will be masked by other audio content present in the file, and then removes that content it calculates to be inaudible. But since these models aren't perfect (even complex ones are limited by the accuracy of the data collected to make the mathematical model), they can create unintended audible flaws, or "artifacts", in the audio.

Additionally, the audio is not stored as amplitude over time, like in a WAV file. (Otherwise all this audio removing would be for naught.) All the content is converted into the frequency domain... and this is where my understanding gets really foggy. But I believe this process can also lead to audio artifacts based on precise timing (like a "pre-echo")?

MP3 encoding also uses Huffman compression to whittle down the size further, but this part of the process is lossless and will not result in any audio defects.

Personally, I have been using MusicMatch to rip my CDs at 160 kbps (CD quality is 128, 160 comes with over-sampling).
Contrary to popular opinion, 128Kbps is not CD-quality, and 160Kbps certainly does not "over-sample". Every MP3 encoding, even at 320Kbps, removes audio from a CD-quality (44.1KHz 16-bit PCM) source, although 320Kbps MP3s will be audibly transparent for nearly all music & ears.

To demonstrate:
- FLAC and Monkey's Audio are two common audio compression formats that keep all the original audio data 100% intact (known as "lossless"). They typically result in files with 40-70% of the original audio's uncompressed filesize.
- MP3 at a constrant bitrate of 128Kbps results in files only 9% as large as the original. Even 320Kbps MP3s are 23% the size of the original uncompressed audio.

You can see from this dramatic difference alone that the MP3s must be sacrificing data, and retaining less-than-CD quality. And it is true, MP3 is a "lossy" format that closely approximates the original audio in whatever filespace it is given.

The program is rather fast with 12 to 15X. I can rip a CD in 5-10 minutes.
MusicMatch also does not check to ensure that it has not read data incorrectly, something that can easily happen on a CD with scratches, smudges, dirt, dust, lint, or whatever on its surface. (Or if you have a poor CD drive.) Exact Audio Copy results in lower ripping speeds, but it double-checks to make sure there aren't any pops, clicks, or skips in your rip. If it reads a different value when it re-reads the data, it will attempt to correct the error as best as it can.

I can hear no difference at all in the quality between the original and the rip even if I set the volume to the highest level.
Then the weakest link in your audio chain is causing more distortion than the MP3s are, rendering the differences between the MP3 and the original unnoticable. The weakest link could be anything from your sound card to your speakers to your own ears.

Some people care to have higher quality equipment with better clarity, and some are satisfied to be able to hear music period. There are also those who simply can't hear well enough to notice most flaws in MP3 compression. That's simply how it is. This guide is for those who prefer high-fidelity and like to hear the detailed quality of their music. (And for those who would simply like to distribute their music in high quality, even if they personally don't notice the differences.)

The format is widely supported and I can hardly see it dying. It may evolve and thus require less space, but dying; I don't see that coming. The computers' CPUs and Hard Drives keep growing in size and connections keep getting faster. At the same time, companies try to find ways to reduce file size. That is a puzzle in itself.
As I said earlier, any further development of MP3 is restricted by the MP3 specification, which dictates what data MP3s must contain, and how it must be structured, so that an MP3 player can certifiably play back all MP3s. (If this were changed, your current player wouldn't be able to play these "new MP3s".) A change in how the file is structured & what data is stored could help achieve smaller sizes & more transparent compression.

Why reduce sizes while bandwidth increases? How about progress. And efficiency. (And for folks who can't afford the latest and fastest, too??)

If you don't see a need to reduce file sizes, try ripping your audio CDs straight to WAV and see how quickly it can fill your modern hard drive. Or try downloading an album in WAV format over your new superfast connection. Better yet, try serving a WAV album over the Internet, with hundreds of people downloading it over time. (Even with MP3 albums your bandwidth expenditure would add up to some sweet dough.)

Don't worry, there are practical uses for even better filesize reduction. ;)

[Moguta]

Yes i was testing out 3.96.1. I'm going to try and rip with the older version of Lame, i think it will work.

*later that day*

Well i tried 3.90.3 and it still ripped at 8+x speeds, it must be because i have a newer EAC or something...i dunno. Anyways, the track qualities all read 100%, so it can't be too bad.
3.96.1 is supposed to encode much more quickly than 3.90.3. But I'm not sure EAC even factors the encoding time into the speed of the rip. (I could be wrong, though.)

If you think it's going too fast, make certain it is set to rip with Secure Mode, not Burst Mode or anything else... since Secure Mode is the only one that double-checks the data read.

I'm back. Sad to say though, after trying everything again, and doulbe checking myself, things came out the same. I don't know what it is, I guess it's just hard luck. Though at least I can burn mp3's. I'll keep trying though, and thanks everyone for the suggestions and help.
That's a wierd problem you described. I have no idea what's wrong, I can only think that something must be getting entered incorrectly. =/ I really wish EAC would let the DOS-shell window stay open after an error so that we could see exactly what the problem is when this stuff happens.

[Ian Gillseed]

I've been encoding mp3s with dbPowerAmp lately. It has plenty of settings and the results sound great to me. Just out of curiousity, what's the concensus on that program? Does it do a good encoding job for VBR .mp3 files?

[DML]

Wow, hot damn Moguta, I knew you knew your shit, but wow. I hereby certify you as the local SD Audiophile - congratulations!

The only thing I can comment on is the cd quality comment by Musouka. If your ears are good and you think you may be able to hear the difference with high quality gear, pick yourself up a pair of $50+ headphones. Depending on your setup, this will easily make or break the sound quality. Personally, I'm running a 600watt Kenwood home theater surround setup, connected via fiber-optic Monster Cable (toslink). Overkill? Perhaps, but the sound is crystal clear. My old Altec Lansing speakers were good, or so I thought. I actually had to delete some of the 160kbps albums I acquired because the quality was so much lower than I had previously thought. :)

[Moguta]

I've been encoding mp3s with dbPowerAmp lately. It has plenty of settings and the results sound great to me. Just out of curiousity, what's the concensus on that program? Does it do a good encoding job for VBR .mp3 files?
dpPowerAmp is a frontend for encoders rather than an encoder itself. It uses the LAME encoder for MP3 encoding, and you should be able to specify --alt-preset standard somewhere in the options (I think). www.dbpoweamp.com has absolutely no documentation on its functions/settings/etc :( so I can't say for sure.

Wow, hot damn Moguta, I knew you knew your shit, but wow. I hereby certify you as the local SD Audiophile - congratulations!
Haha!
Thanks, I guess. ^_^

[Moguta]

I'm back. Sad to say though, after trying everything again, and doulbe checking myself, things came out the same. I don't know what it is, I guess it's just hard luck. Though at least I can burn mp3's. I'll keep trying though, and thanks everyone for the suggestions and help. :)
EDIT: NEVERMIND! See my next post!

Well, I tried copy/pasting the command I have listed here straight into EAC. And it works fine for me. So I'm not sure what the problem is in your case. =/

Does everything look like this? (except for the *specific* file location of flac.exe)

EDIT: NEVERMIND! See my next post!

[Moguta]

Okay, I discovered the problem with FLAC. Unbeknownst to me, the FLAC binary that I link requires a libmmd.dll due to the compiler used. (This hasn't occured in his previous FLAC compiles.) I've changed the FLAC link to the official Sourceforge package, which is what I've actually been using.

D'oh!

[Musouka]

Yes, I got to admit that the quality is better when using this method. Thanks a lot for that.

I got but one question. The program's track number format is XX/YY. For example, if the CD has 40 tracks, the first track number will be 01/40. How can I set the thing up so it writes 01 for first track, 02 for second and so on?

Thanks!

[Secret Squirrel]

Hm, i think you just have to set the file naming scheme to %N - %T under the naming tab.

[CaptainCommando]

What about 'timing errors' in the log files?

(Example:

Timing problem 0:00:08
Timing problem 0:00:23

Peak level 99.8 %
Copy CRC 1298FE24
Copy finished)

These seem to pull up at least once in a rip, and I'm kind of curious to know if it really matters any (especially if there are several timing errors in there). I'm guessing it may mostly have to do with the CDs being scratched or dirty (started with some of my older discs first). It hasn't pulled up any read errors yet, though. I'm using the standard settings from this help guide.

I have been re-ripping the tracks that have timing errors, but it's a bit more time-consuming and sometimes I have to rip the tracks more than twice. What exactly are the timing errors and do they cause any problems with the tracks?

Nothing seems listed in this post, but I was able to find this bit of info on EAC's FAQ page, though it doesn't seem too informative...:

Q: When using burst mode, EAC also shows up timing problems, are these really errors or what?

A: No, burst mode has no error detection nor error correction. If burst mode brings up a timing problem, the read command needed a lot of time, which could have several reasons, like loosing sync or trying to fix an read error. Of course this is a really poor "error detection" and should not be taken as serious indication.

[Lowe]

What about 'timing errors' in the log files?

(Example:

Timing problem 0:00:08
Timing problem 0:00:23

Peak level 99.8 %
Copy CRC 1298FE24
Copy finished)



Why are you using burst mode? You should always rip in secure mode. This is a great guide too and with pictures http://forums.danomac.org/viewtopic.php?t=27 .

[CaptainCommando]

Yup, that would be it! Thanks! I thought I'd set that, but I guess I hadn't...

Like that forum, too, and the instructions are MUCH more helpful with screenshots (though they could still be cleaned up a bit). It's also nice when you have the only videogame soundtrack on the offset detection list :P (Ultima IX: Ascension, though I've got the promotional demo disc, so I don't know if there's any issues with the public rerelease).

Now even if I have the offset checked, would it still be better to use a drive that has a lower offset number (say +02) rather than a higher number (+705)?

They also have a nice listing of how to detect pirated software, some of which I was only vaguely familiar with, but having actual images of the logos is a big help. I wasn't aware of pirated discs until I actually purchased a couple, but I was frustrated to find out one other disc I had was pirated (I didn't buy it though). Have yet to check the discs I keep in my closet, but it better be the real deal considering A. how much I paid for it and B. that it came from gamemusic.com :P). Is there a listing somewhere with a lot of authentic publishers and their logos? (as it would be nice to have a more complete list in one place). For now, I'll just make extra notations in the ID3 tags if the disc is pirated (or from an actual game disc).

[Moguta]

Now even if I have the offset checked, would it still be better to use a drive that has a lower offset number (say +02) rather than a higher number (+705)?

They also have a nice listing of how to detect pirated software, some of which I was only vaguely familiar with, but having actual images of the logos is a big help. I wasn't aware of pirated discs until I actually purchased a couple, but I was frustrated to find out one other disc I had was pirated (I didn't buy it though). Have yet to check the discs I keep in my closet, but it better be the real deal considering A. how much I paid for it and B. that it came from gamemusic.com :P). Is there a listing somewhere with a lot of authentic publishers and their logos? (as it would be nice to have a more complete list in one place). For now, I'll just make extra notations in the ID3 tags if the disc is pirated (or from an actual game disc).
If you tell EAC the offset it will correct for it, so as long as the drive is ripping CDs fine there's no need to use another.

As far as verifying what soundtracks are authenticly published, check the listed catalog numbers & publishers against the information on sites like Game Music Revolution (www.gmronline.com), Chudah's Corner (http://www.chudahs-corner.com/soundtracks/index.html), or RPGFan (http://www.rpgfan.com/soundtracks1.html). When buying from an online auction and they don't have that info listed, ASK.

SonMay, EverAnime, and Alion are the big 3 VGM bootleggers last I knew.

[]-/|\-oOo-/|\-[]

A bit off topic, but does anyone know how to convert HES files (PCE music) into MP3 or Wav? I'm assuming you would need to do it on Winamp, but I haven't quite figured it out yet.

[dma]

Yeah in WinAmp, set output to "stream-writer" or something, then it'll record what you play in .wav (you maybe even got a choice to record it as mp3 directly).

[Moguta]

-/|\-oOo-/|\-[']A bit off topic, but does anyone know how to convert HES files (PCE music) into MP3 or Wav? I'm assuming you would need to do it on Winamp, but I haven't quite figured it out yet.Hit Ctrl-P in WinAmp.
In the tree view to the left, click on the "Output" suboption under "Plug-ins".
It should now show the output plugin selection screen on the right.
Select "Nullsoft Disk Writer plug-in" and click the "Configure" button at the bottom of the screen.
Click the buttton below "Output directroy" to specify where you want the output files to be placed.
You'll probably want to leave the output format as WAV since the max MP3 bitrate this plugin will automatically convert to is 56Kbps. So click "OK".
Leave "Nullsoft Disk Writer plug-in" selected, click "Close", and then just start to play what you want converted.

Once you are done converting what you want to WAV, go back to the output plugins options, and select "DirectSound output". Your music will now play to your soundcard & speakers as usual.

If you want to convert the WAVs to MP3, you could use Exact Audio Copy if you already have it set up from this topic. Just go to the "Tools" menu and select "Compress WAVs". If you don't have EAC already set up, then I'd download the LAME encoder (http://www.rarewares.org/dancer/dancer.php?f=1) and ALL2LAME frontend (http://members.home.nl/w.speek/all2lame.htm).

[]-/|\-oOo-/|\-[]

Thanks a lot, Moguta!
I will try this as soon as I can get back to my PC.

[Moguta]

LAME 3.97 has become the Hydrogen Audio recommended version after a round of double-blind listening tests. So I updated the guide. Feel free to try out the new version & quality scale if you want. ^_^

[DML]

Ooh, very interesting. Was the LAME algorythm reverted back to the older, slower method? I know that the newer versions could encode an album in --APS very quickly. Hopefully this is the best of both world! Thanks Moguta! :)

[Moguta]

With --vbr-new, I'm getting 16x realtime encoding speed at the moment. (Granted, I'm on a 2400+ Athlon64 with 1GB of PC3200 RAM) --vbr-new is quick. ;)

The non "-new" mode of VBR was used previously in the --alt-presets. --vbr-new is a quicker mode that wasn't recommended in previous LAME versions because there were concerns about its sound quality. It isn't "new" to 3.97 (the --vbr-new command was available in 3.90), it's just *good* in 3.97.

[DML]

Hmm, interesting. Is this the recommended command line to encode at now, Moguta? 16x would be very nice, but first and foremost, quality is the biggest priority. :)

Heh, I have a ton of stuff I need to rip, so if --vbr-new is the preferred method AND it's lightning fast, this is good news!

[dma]

Notice that you can still use "-alt --preset ..." commands, as these are mapped to new recommended presets.
Just beware as the new recommended "-V 2 --vbr-new" preset is equivalent to "-alt --preset fast standard", and not "-alt --preset standard" like what was recommended for LAME3.90.3.

[Datschge]

Such long discusssions about the right settings for an outdated lossy audio format when there is a newer lossy format with better audio quality at less disk use and faster encoding speed? Just had to ask after seeing this thread popping up for the hundredth time. >_>

[dma]

Just a short notice for people using CDEx (if any).

The "--alt-preset xxx" parameters in this CD ripping software are not simple commands send to LAME but a series of hardcoded parameters, so these won't be corresponding to any new "-V xxx" with LAME 3.97b1 DLL.

Therefore, if you want to use CDEx with LAME 3.97b1, take the DLL which uses an .ini file for setting encoding parameters: http://www.rarewares.org/files/mp3/lameDll3.97b1_MOD.zip
The new recommended quality parameter " -V 2 --vbr-new" is number 16 in the ini file.


Yeah i know, i'm using old non updated software with updated but outdated encoder, sorry Datschge.

[DML]

Outdated? Sure, of course mp3 is outdated, but it's the standard. I'm not sure which format you're refering to, specifically, but there's still not much support for OGG. Anything else like FLAC or APE isn't supported at all, for the most part, so having mp3s the highest quality is priority right now. Or are you referring to AAC? I honestly don't know what's the deal with it, personally. I have a #gamemp3s rip of some soundtracks in AAC (albeit, they were ripped a long time ago) and mp3 counterparts. The mp3s sound better; the highs and lows just sound better.

So, yeah, mp3s are going to be around for a little while longer, so might as well get the most out of them. LAME4LIFE, haha. ;)

[Moguta]

Notice that you can still use "-alt --preset ..." commands, as these are mapped to new recommended presets.
Just beware as the new recommended "-V 2 --vbr-new" preset is equivalent to "-alt --preset fast standard", and not "-alt --preset standard" like what was recommended for LAME3.90.3.
This is true, but those are merely to be consistent with the old mappings, since the normal "--alt-preset"s would use the old VBR mode and the "--alt-preset fast" commands always used --vbr-new. It has been suggested that, in version 3.97, --vbr-new is equivalent to or slightly better quality than the old VBR algorithm. Might as well reap the benefit of markedly quicker encoding times!

Such long discusssions about the right settings for an outdated lossy audio format when there is a newer lossy format with better audio quality at less disk use and faster encoding speed? Just had to ask after seeing this thread popping up for the hundredth time. >_>
If you're talking about Ogg Vorbis, MP3 (via LAME) tends to be better at high-quality higher-bitrate encodes. It's much better than MP3 for low-bitrate material, but if you want the best quality period, Vorbis is not the way to go. Not to mention the format's development seems to be moving along at a snail's pace, if it hasn't altogether halted.

MPC or Musepack does provide excellent high-quality sound, however this format has VERY little support (only a few select media players and NO hardware players) and the development also seems to be rather snail-like.

AAC, often hailed as the successor to MP3, is still not nearly as supported as MP3, and from what I hear, the free-to-use encoders don't produce such good quality AAC files.

[DML]

AAC, often hailed as the successor to MP3, is still not nearly as supported as MP3, and from what I hear, the free-to-use encoders don't produce such good quality AAC files.

One of my good friends ripped an album using the AAC encoder included with iTunes 6 and I ripped it using EAC LAME v3.90.3. The LAME encoded mp3 sounded better on my 600watt Kenwood home theater rig than his AAC tracks did. For one, in the specific track we listened to (can't remember what it was off hand, this was a few weeks ago...some electronic album), the low end sounded better on LAME as there was more bass (in a good way). In terms of the high end, the sound from the LAME-encoded mp3 just sounded more rich and vibrant. I played both to him blindly and he felt the LAME track was superior.

Again, this is using the encoder bundled with the new iTunes. I'm sure you can pay for higher-quality encoders, but I'm sure as hell not going to pay for an encoder when I can get something of extremely high quality for free. I've gotta hand it to those developers of LAME. :D

[kingoftheryche]

one more addition: i think you should also add: "-m s" to parametres to have Stereo, not a joint-stereo which is a worse quality. But this is still a doubtful question.

[dma]

kingoftheryche : It seems you're living like 2 or 3 years in the past. ;)

[kingoftheryche]

Oh really? Why so? LOL
How can you prove that j-stereo is better then full stereo?

[dma]

Because the algorythm has been corrected since several years.

[kingoftheryche]

It is a bad proof. LOL "Many years passed and everything improved". j-stereo was used for lower bitrates while stereo for high CBR's.

[POPOBOT5000]

---

[kingoftheryche]

nice page here:

http://harmsy.freeuk.com/mostync/

[kingoftheryche]

and here nice stuff:

http://www.videohelp.com/forum/userguides/130689.php

http://www.hydrogenaudio.org/forums/index.php?showtopic=30637

[POPOBOT5000]

---

[kingoftheryche]

What i know for sure is that j-stereo degrades heavily panoramic sound like dolby surround. While stereo fills each channel equally that can cause garbage in lower bitrates... .

[Moguta]

one more addition: i think you should also add: "-m s" to parametres to have Stereo, not a joint-stereo which is a worse quality. But this is still a doubtful question.
There is often some confusion with joint-stereo mode. It merely decides frame-by-frame whether to use mid-side encoding or left-right stereo depending on which is a more efficient method to represent it (i.e. parts where the left & right channels contain much of the same audio will likely use mid-side, and chunks with high stereo seperation will use independent left-right channels). You 'll see % and * in the bitrate bars when LAME is encoding in VBR with joint-stereo (as well as "LR MS %") which signifies how much of the audio is being encoded as left-right (LR) stereo vs. mid-side (MS). Additionally, whenever this is brought up on the Hydrogen Audio forums, those involved with LAME have always emphasized that forcing truestereo could degrade the audio quality & that joint-stereo is optimal.

The "intensity stereo" method mentioned on some of the sites you linked is not used in the LAME --presets or -V commands, and that *is* indeed to be used in low bitrates only since it results in degredation of the stereo image.

What i know for sure is that j-stereo degrades heavily panoramic sound like dolby surround. While stereo fills each channel equally that can cause garbage in lower bitrates... .
Why would you be using 2-channel encoding for surround sound in the first place? Additionally, as mentioned in a link of yours (http://harmsy.freeuk.com/mostync/), mid-side actually results in the exact same audio upon decode. It's just about whether mid-side or left-right results in a more efficient use of bits.

[CaptainCommando]

I brought this up in another topic, but figured I'd bring it up here too. What's the best way of converting to MP3 WAV files that are encoded at something OTHER than 44 KHz? Should you still aim for something close to LAME VBR (I thought I could convert them with RazorLAME, but I'm hearing LAME doesn't like anything that's not 44k) or use something like CoolEdit to convert them to VBR using the Fraunhaufer encoders? (Gotta love batch file convert :D ) Or should I convert the WAVs to 44K and THEN use LAME to encode them?

Reason I ask is because I've got some wavs that are at 48k, others at 37k, and I might have some more that are at something liek 16k. Somebody had also mentioned 22k files, which apparently LAME DOES like to read.

[Rimo]

Although I'm not the expert about this, I can still answer something to your question. From my experience, LAME can encode wave files that are at 48Khz, 44.1Khz, 32Khz, 22.05KHz, etc. Though I think it only handles the "standard" frequencies (like those I just listed). The XA one, 37.8KHz, isn't standard, so you have to convert it (usually to 44.1Khz, CD-standard). I'd also be interested to know more about this.

[Moguta]

Updated the guide...

- Removed AAC to decrease the clutter, as I have the feeling that absolutely no one was using this format.
- Updated to the newest Musepack encoder 1.16 and added markedly quicker CPU-optimized Vorbis encoders/decoders.
- Expanded the description of MP3Gain's capabilities.
- Added section on encoding from WAV & transcoding.
- Replaced out-of-date links.
- Miscellaneous reorganization.

[Secret Squirrel]

Thanks for cleaning up this thread. It's been a staple here for a long time.

[GoldfishX]

Updated the guide...

- Removed AAC to decrease the clutter, as I have the feeling that absolutely no one was using this format.

Um, I rather love AAC...That's what I rip all of my CD's to through iTunes and/or Nero (I started using iTunes for for the .mp4a, so I could use them in my shuffle). Pretty self-explanatory though.

[Naka]

I notice #gamemp3s are using EAC .95 beta 3 but I cant find this anywhere to download anymore. Is beta 4 okay to use for the most part?

[Moguta]

Um, I rather love AAC...That's what I rip all of my CD's to through iTunes and/or Nero (I started using iTunes for for the .mp4a, so I could use them in my shuffle). Pretty self-explanatory though.(Bold emphasis mine)
Yes, but I was merely referring to those who use this guide. Plus FAAC, the only freely-available command line encoder -- i.e. the only one you can use in conjunction with EAC -- has comparatively poor quality compared to other AAC encoders. Such as the ones in iTunes and Nero. :)

I notice #gamemp3s are using EAC .95 beta 3 but I cant find this anywhere to download anymore. Is beta 4 okay to use for the most part?
I seriously doubt there is any major difference. Honestly, the version of EAC seems to matter much less than the version of the encoder you're using. Exact Audio Copy has always been stable for me with any version I've used, and the ripping method itself has stayed pretty much unchanged.

[CaptainCommando]

Here's a question, what's the alternative to EAC if you're using a Mac? I've found this program, Max, and want to know how it compares with EAC's encoding. They say they use LAME (built-in) for MP3 encoding.

http://sbooth.org/Max/

[CaptainCommando]

I ripped a couple tracks here and compared them with a set of tracks encoded in EAC. The file size was only off by a few dozen k so I think the conversion quality is roughly the same. If you've got a Mac, try this out and see how well it compares with PC rips. For those of us who do not have PC's, this might be an excellent high-quality alternative to EAC.

EDIT:
The MusicBrainz server seems to be really crappy. It has practically zero information on Japanese game CDs. When I load Golden Axe Collection, it can't find any information on tracks and titles. iTunes, on the other hand, will pull this information from the disc itself. I haven't figured out how to get it to take information off the disc in that case, but that's going to be really annoying to have to manually add album information all the time.

Apparently I'm not the only one having this problem and from what I've gathered they haven't fixed it yet: http://forums.sbooth.org/viewtopic.php?f=4&t=2092

Here's what I did to set things up:

Part I: Setting up for encoding:

1. General Tab - Most of this is for checking program updates. You can have Max automatically acquire artist and track names as well as save disc information by checking those boxes. I don't think you'll want to automatically encode all tracks.

2. Formats Tab - Double-click on MP3. Set Encoder quality to Custom. Set Encoder target to Quality. (I think Bitrate will just try to be economical with storage size). Set Encoding engine quality to High. Set Stereo to Default (I think this will do whatever the disc is set at). Set Quality up to 100. Set Variable bitrate mode to Standard. (I think Fast might give quality issues). Set bitrate to 320. Hit OK.

Using these settings, I was able to get a file size that was almost identical to that of a track ripped in EAC, so I think the quality is almost the same. Encoding time is comparable to that on EAC.

Note that you can double-click MP3 from Available output formats again to add another set of MP3 encoding settings.

With FLAC encoding I'm not too sure as I'm not familiar with it. They have compression level from 0 to 8 or you can customize the settings.

EDIT: With a FLAC compression level of 6, resulting files are only about 30k smaller than those ripped using EAC.

3. Output Tab Automatically saves everything to Music, but you can change this. In File Naming, you'll want to delete the {albumArtist} variable - otherwise it will stuff everything into separate folders based on artist and album rather than just album. You can also add a hyphen between {trackNumber} and {trackTitle}. Check "Use two digit track numbers" so everything will be in proper order. I'm not sure what the 'Fallback to album tags if track tags unset' does. I've got 'Save encoder settings in comment'

4. Ripper Tab - I'm not 100% sure what this does here. You can change which ripper to use (Basic, Comparison, and cdparanoia). I checked cdparanoia, though I'm not sure if my G4 PowerBook has error correction (note that file sizes were the same when I tested this on different settings). I turned on the SHA-256 and C2 error correction. In Paranoia Settings, I enabled Error correction and used Full paranoia and Overlap Checking. I checked 'Never allow skipping'.


Part II: Encoding

Now when you actually go to do the encoding, you can try and find CD information from Audio Brain. I'm not sure why it's getting the artist names in Japanese though - if it's getting this off the disc or the database.

You can select all tracks or select them individually. You'll probably have to change the Genre to 'Game'. In Comments, you can add who is doing the encoding and what version of Max.

When you go to encode, just hit 'Encode' and it will run in the ripper window. Oh yeah, it will also encode multiple tracks at the same time. I think this is set by 'Max number of encoders' in the General tab of Preferences.

[CaptainCommando]

Ah. There is an Applescript for getting metadata from iTunes:

http://forums.sbooth.org/viewtopic.php?f=4&t=1930

This will replace the app that was downloaded with 0.8.1. Copy/paste to Apple Script Editor and save it as an application. To make things easier, drag a shortcut to the applications bar.

However, iTunes and Gracenote is pretty worthless for Japanese albums unless you like having half your information in Japanese. (then again, it's not like Golden Axe The Music has been properly translated on VGMDb yet either...). Perhaps somebody could eventually write a script to access Freedb as I think they'll have better information there...

[CaptainCommando]

Well this is interesting as well. Blacktree has an iTunes plugin that will do LAME encoding through here:

http://blacktree.com/?quicksilver

However, the encoder seems a bit slow and does not allow for as many LAME options as are available with EAC, or if it does, it seems to use different code. --alt-preset-extreme seems like the best bet, but I don't think it's as good as EAC considering how the iTunes encoder doesn't give you much control over settings.

[CaptainCommando]

Wow, I seem to have made quite a few posts here in a row. This is a question about kanji display in EAC. I've got a new album that is unlisted on VGMdb and I'm trying to get a program that will properly display the kanji info from the disc so I can get an accurate track list (can't find the artist, but it's Cho Aniki so pretty sure it's Koji Hayama). I don't want to stick the album into my Mac because a) it's a minidisc and b) the slot loader has been smudging discs so I refuse to trust it. I posted a somewhat mirror copy of this on VGMdb:

http://vgmdb.net/forums/showthread.php?p=6514#post6514

[Secret Squirrel]

Where do the tracknames come from? Are they stored on the disc somehow, or is EAC accessing some kind of Freedb?

[CaptainCommando]

The tracknames are stored on the disc somehow as digital tags - it's not in Freedb.

http://vgmdb.net/album/11720

That's the actual Japanese text from the tags, and I just extrapolated the artist as Koji Hayama because it uses his style of recorded clips. I can read it in iTunes, but when it comes to EAC actually being able to read/write the kanji, it can only do it for artist and album name - I can't copy/paste the tag info into the track names themselves. Could just be my software is screwy...

Anyway, if you want a copy for SD, I can send it to you. The album's got some hilariously bad stuff (sometimes so bad it's painful). If you've seen footage of the game in action... Well, let's just say I opted to play Sexy Parodius and Dungeons and Dragons Collection first :P Like I said, scans in the cue...

[Moguta]

If it's pulling names off of the album itself, it would be using the disc's CD-Text (http://en.wikipedia.org/wiki/CD-Text). Also, note that EAC will automatically try to pull an album's information from FreeDB without asking, just in case you thought it was something you have to initiate.

As for the kanji transforming into '?'s, it means EAC needs to be updated for full Unicode (international character) support. Nothing you or I can do until the author writes new code.

[Moguta]

A bit of an update to the guide:
Updated 'Encoding from Files' section to reflect changes in Foobar2000 v0.9.6
Removed all mention of the rare, long-stagnated Musepack codec
Updated LAME to version 3.98.2 and Ogg Vorbis to aoTuV 5.61
Changed recommended Error Recovery read-attempts to Medium