Put All Audio Questions Here

Use a non-broken recording software that supports user-specified pre-gap length.

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Depends, m4a indicates only an MPEG4-style container. Could contains audio encoded in AAC (lossly) or MPEG-4 ALS (audio lossless coding) material. You should check this with foobar2k, usually it displays the compression type in the preferences. Or try VLC with verbose messages activated, should also give you a clue what type the audio is.

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Originally Posted by Audiophile

iTunes tracks are all lossy. Unless you just want to remove the DRM, and have a useable m4a file, the only way not to lose quality is to convert to lossless. Probably the easiest (and only) way I know of is to burn it to a CD, and then rip it to FLAC (which is better than WAV; same quality, less size). I think there used to be a program to strip off the DRM, but it's pretty old, and I don't think it's been updated for the new scheme. Hope this helps!

Wrong, there are a multitude of projects that can strip the DRM encapsulation of the audio file if you provide a key to do the decryption.
One example is this project:
hymn -- decrypt iTunes and iPod music / unprotect AAC files (m4p --> m4a)

If you know the key and algorithm to the problem then the only problem left is obfuscation.

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Originally Posted by Audiophile

Whoops! I knew about hymn, I just didn't know that it was working with the newest form of the protection on iTunes. The last time I had checked, it hadn't been updated for a while.

It was only an example. Like the cracking/reverse engineering community is always lagging behind the copy-protection industry, it's the same here. You just have to wait, or start hacking yourself.

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No, it won't affect audio quality. A larger buffer does not modify the digital data that resides inside. Latency is only interesting when dealing with both recording and playback where it matters if you hear the incoming audio data some hundreds of ms later.

I was speaking idiomatically.

Try using playgsf-0.7.1 with WAVE output and then feed the result into LAME.

What kind of toxic man-thing is happening now?

You may want to read this:
USF Central

Extracting audio from N64 catridges is even harder than extraction from Playstation ISOs.

FELIPE NO

You repeat yourself...

The Gameshark does nothing more than to modify memory location, either manipulating game code or game data (or both, as we know that a lot of N64 catridges do run-time code transformation).

If you have read the link I gave you, you now know that the N64 has no standard way of playing back SFX and music data, that's the main reason why it's so hard to rip music.

Should be clear by now why no universal memory hack exists for disable SFX playback. You would have to figure out (by disassembling and tracing) for each game which part of the gamecode generates SFX and passes it to the DSP of the N64. Then you patch that part of the code and make a diff, resulting in your gameshark code.
That's not easier than creating a USF.

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Or enable C2 with secure mode (usually most guides advice you to disable it), if the drive logic correctly implements C2 error reporting and there are any (which can't be detected through C1), then EAC should provide a more precise report of possible error positions.

@sup!: I would advice against cutting out frames, which results in audible dropouts. Filtering the frames plus surrounding frames should be better.

Jam it back in, in the dark.

Never transcode lossy -> lossy

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Well, the problem with your question is that you can't really answer it.

Plus: the question is incomplete. "ideal bitrate" <- ideal for WHAT? To retain perceived audio quality? To retain previous filesize?

So what you should ask yourself is: why do I even need to reencode? And if it's absolutely necessary, why isn't there no (uncompressed) source material available?

Additionaly the answer to the question also depends on more than just the source bitrate of the file. What encoder was used? Which version, what options, and so on.

And keep in mind that Ogg Vorbis is inherently VBR. Some GUIs displays something like a target bitrate, but that's purely based on some test encodes.

If you just need to transcode to give someone a demo of Vorbis I'd say: use quality level 5.0 and hope that the source material was properly encoded.

If you're up to transcode some music collection -> JUST DON'T DO IT

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

Originally Posted by Dark Nation

I have some sound effects files and also music files. I'm using them for a project of mine, but for reasons I'd rather not get into, I cannot use .mp3 right now.

I suspect patent issues

Originally Posted by Dark Nation

I have the option of MIDI, Ogg Vorbis, WAV.... or .wma (Which I do not deal with at all just out of principle).

I wouldn't be surprised it you'd also end up with patent issues when using WMA.

Originally Posted by Dark Nation

While I would go with WAV, the file-size will likely be prohibitively large for the music files at least.

Yeah, usually not a good idea to include purely uncompressed music. Have you though about using some lossless codec, like FLAC? You already mention it below, but not on your list with codecs available.

Originally Posted by Dark Nation

Plus, .ogg files in my experience tend to sound well even at lower bitrates, but we come back to my original question: What is the average bitrate that would be most appropriate? As you just said above, the reason is perceived audio quality.

I would aim for quality setting 5.0, but probably you should approach the problem from another side. You mentioned that this is some sort of (software?) project. So you possibly have a target filesize for the final (setup) package. Maybe you should figure out the maximum size of the audio you can include and then tweak the Vorbis quality setting so you end up with something around that filesize.

Originally Posted by Dark Nation

The files were orignally .mp3 and I don't think I can find an uncompressed source.

Well. Like I already said: It's hard to answer that kind of target bitrate questions when transcoding is done. You probably want to check some quality settings and then decide what your minimum quality target should be.

Originally Posted by Dark Nation

However a second sub-question arises: Should I be able to find lossless versions of said files, what is the best file-type to, again, convert to .wav or .ogg? FLAC? APE? WAV (CD Audio)?

Sorry, I'm not sure I understand.

So you assume you have source material in lossless form. What exactly is your question? What the target format should be, or is it about the source format?

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Like AVI and WAV also OGG is only a container format, so it can contain various formats. Most of the time it contains Vorbis data (and Vorbis is NOT lossless), but can also contain FLAC (which IS lossless).

I was speaking idiomatically.