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I am wondering if it's possible to rip music from my legally purchased CD's as MP3 and encode them at 320 KBPS quality? I have not done this before on Linux and was wondering if you had any suggestions?
There are many cd ripper gui frontends available that will allow you to do this.Make sure that you have lame installed.I use Asunder for ripping but there are many more.If you prefer command line ripping you could try ripit.
You can also use the OGG Vorbis codec, which is similar, but doesn't have patent issues. It is less compatible with physical players, so MP3 might be more suitable.
You can also use the OGG Vorbis codec, which is similar, but doesn't have patent issues. It is less compatible with physical players, so MP3 might be more suitable.
Yes - sadly I am using an iPod. I can't find any other portable music players as polished as iPods. I don't think they support .ogg sadly. I want to keep my MP3 format but use 320 KBPS encoding sound quality.
There are dozens of CD rippers out there, of varying quality and features. The vast majority of them act as frontends for the command-line programs cdparanoia (for ripping), and lame, oggenc, flacenc, etc (for the encoding).
Simply use whatever package manager you use (synaptic, aptitude, apt-cache search, etc) to search for programs and try a few of them out. Eventually you'll find one you prefer. Some of the more common ones are sound-juicer (gtk/gnome), ripperx (gtk), kaudiocreator (for kde3), and asunder (for kde4). One interesting program I found recently is xfca, which can do rips to multiple formats at once, among other things. And of course there's a variety of cli rippers out there as well.
The k3b disk burning program for kde can also do audio and video ripping. And kde even has a kio-multimedia-plugin that lets you browse virtual folders in konqueror/dolphin that contain mp3, ogg, flac, and wav "files" of the cd tracks, and if you drag&drop them, they will be ripped on the fly to create a new file of that type based on your preset audio settings.
I personally use kaudiocreator and asunder, mostly because they have the best before-the-encoding tag editing features I've found (and only incidentally because I use kde).
As mentioned above, you'll probably have to install some codecs separately, such as mp3 and aac, as they are patent-encumbered and can't be included by default. Assuming your listing of Debian is correct, you'll probably want to add http://www.debian-multimedia.org/ to your repositories in order to get the best multimedia software.
Yes - sadly I am using an iPod. I can't find any other portable music players as polished as iPods. I don't think they support .ogg sadly. I want to keep my MP3 format but use 320 KBPS encoding sound quality.
If your ipod is compatible, you might consider loading it up with the rockbox open-source alternative firmware. It appears to have lots of good features, as well as support for most of the popular formats (but not drm-encumbered stuff, obviously)
According to Wikipedia, iPods support "MP3, AAC/M4A, Protected AAC, AIFF, WAV, Audible audiobook, and Apple Lossless".
Many Linux rippers can be configured to use arbitrary encoding backends, so you could install faac and set it to encode to AAC instead. But AAC on linux doesn't yet have quite the level of support that mp3 does, especially in things like tagging software. Google tells me that there's apple lossless (ALAC) decoding available for linux, but it's another proprietary codec and so there's no encoder available. If you want to go lossless (and not hack your ipod) you'd have to go with uncompressed .wav or .aiff.
IMO though, unless your player+earphones+ear are all audiophile quality, it's a waste of space to go lossless. Your ear simply can't tell the difference (and I think people who say they can are simply psychologically fooling themselves). Even going to 320kbs seems a bit excessive to me most of the time. I usually encode to 192kbs .ogg myself (bumped to 224 for hi-dynamic-range classical stuff) and never notice anything that would diminish my enjoyment of the music. The only reason I would go lossless would be if I wanted to have only one file that I could use on both my portable player and high-fidelity home system.
I usually encode to 192kbs .ogg myself (bumped to 224 for hi-dynamic-range classical stuff) and never notice anything that would diminish my enjoyment of the music. The only reason I would go lossless would be if I wanted to have only one file that I could use on both my portable player and high-fidelity home system.
I agree with you for the loossy quality, 192kps is plenty, but one other reason for lossless (flac) would be for cd backup reason (average 1/3 of the size of a cd) like this you can store a perfect cd quality album for latter burning on cd if needed.
Last edited by Davno; 11-20-2009 at 06:02 AM.
Reason: too many typo
The best tool to extract cdaudio these days seems to be cdda2wav, since cdparanoia seems to be abandon-ware, I can't suggest frontends, I like to keep things simple so I just use the tool directly. I got in touch with the cdda2wav devs some time ago on the Gentoo mailing lists, about this same issue. It was quite illustrative. If you are curious, you can follow the thread at this reader (amongst others probably).
For actual mp3 encoding, there is nothing better than "lame", and there aren't many alternatives AFAIK. Most tools use the lame libs or the lame tool, i.e. the algorithms from the "lame" program. For example, you can encode mp3 files with the "audacity" audio editor using "lame" libs.
For actual mp3 encoding, there is nothing better than "lame", and there aren't many alternatives AFAIK. Most tools use the lame libs or the lame tool, i.e. the algorithms from the "lame" program. For example, you can encode mp3 files with the "audacity" audio editor using "lame" libs.
There's toolame and twolame, which are respectively the fork, and the fork of the fork of lame if I remember right. Surely there's a lot of shared code in the middle, and the basic procedure will be the same, after all the result must be an mp3 file. However there's always some place for creativity and improvement in complex algorithms. You can always make your own tests.
I personally don't compress to mp3 so I don't have any motivation for such tests. I just use vorbis when 1:1 quality is not important, and flac when preserving every single bit is important.
For actual mp3 encoding, there is nothing better than "lame", and there aren't many alternatives AFAIK. Most tools use the lame libs or the lame tool, i.e. the algorithms from the "lame" program. For example, you can encode mp3 files with the "audacity" audio editor using "lame" libs.
GOGO-no-coda (petit) http://homepage1.nifty.com/herumi/gogo_e.html
This is not another front end, its a command line encoder base on Lame 3.9, a lot faster encoding to .mp3 than plain Lame.
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