Put All Audio Questions Here

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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Originally Posted by LiquidAcid

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.

Oh...I see the connection now. Thanks. Yeah, I did read the link you posted, but I didn't see how that answered my question until you explained it just now. (I repeated myself because I wasn't sure if your post was directed at me to begin with. )

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

Recording net radio: HOW TO?

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

I've begun working on my first .GBS conversion, and I have a few questions.

First off, I saw that this was covered briefly earlier in this thread, but when applying replaygain info to tracks, is there any way to prevent native clipping from a .GBS file? (I guess that would be the best way to describe it.) I did a replaygain scan with Foobar2000 v0.9.4.2 and noticed all tracks are above the ideal 1.00 dB track gain value, not to mention one of the tracks being at 11.15 dB... So, I guess my question is, since I can't apply replaygain to the .GBS file itself (to my knowledge) after converting the tracks to .FLAC (.WAV can't store replaygain info, AFAIK), can I still apply replaygain? Or, would it not make a difference at all?

Second, are there are any recommended freeware audio editing programs you would suggest?

Finally, I've never really done any audio editing before, so any suggested guides/instructions/tips for the audio editing process would be appreciated. Thanks.

Edit: I'm using Audacity to do some experimental editing. I've ripped/encoded a sample track, if anyone wants to critique it and give me any suggestions. Thanks, again.

Kaeru.mp3

I was speaking idiomatically.

What is the best program to edit audio tags?

What kind of toxic man-thing is happening now?

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This track contains artifacts at 1:03, 3:13 and a ripper at 4:35, more at 6:32 and 6:35, as well as skipping at 5:29. But are they rip/handling related or is it a problem with the mastering? The disc's part of a 7CD set that arrived last week, and the others are totally flawless. There's only the barest scuffing on the problem disc.

I've reripped it twice, presuming hardware hiccups, but every time EAC error correction didn't detect anything amiss.

If it's a mastering issue, is there any possiblity for correction? If it's (somehow) physical damage, what else is there to do?

FELIPE NO

Sometimes using burst-mode in EAC helps with badly scratched discs. If there're no scratches then it's probalby a mastering issue.

You could edit the wave file and cut the erroneous frames out.

How ya doing, buddy?

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.

It isn't badly scratched though, but regardless I'll try burst then fiddle with C2 reporting.

How ya doing, buddy?

Well I made a journal entry but since no one responded, I guess I'll ask here:

If I have an mp3 file, what's the ideal bitrate for an .ogg conversion?

96kbps, 128kbps, something else? Thanks if you know.

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

Never transcode lossy -> lossy

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Originally Posted by LiquidAcid

Never transcode lossy -> lossy

While there's truth to this, it doesn't exactly answer his question.

I honestly know nothing about the OGG format, having never worked with it. Talk to Rimo or Killy; they would have the answer for you.

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Originally Posted by LiquidAcid

Never transcode lossy -> lossy

Ok yeah, but that doesn't answer my question. I need to convert some .mp3 files to .ogg and I want to know the appropriate bitrate to do so at.

Man why is it so hard to just get a direct answer to a direct question nowdays?

Ask Rimo or Killy eh? Alright.

What kind of toxic man-thing is happening now?

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

FELIPE NO

Alright, I guess I should have provided more info, but I didn't think it mattered:

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 have the option of MIDI, Ogg Vorbis, WAV.... or .wma (Which I do not deal with at all just out of principle). While I would go with WAV, the file-size will likely be prohibitively large for the music files at least. 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.

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

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)?

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

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?

Jam it back in, in the dark.

As LiquidAcid brought forth, -q 5 should be ideal to get a great sound with a moderate filesize. It is the setting Hydrogenaudio recommends to achieve transparency (no perceivable quality loss), so using a higher setting is somewhat pointless for standard listening purpose.

Since you'll be transcoding from MP3, there will be additional quality loss. If your goal is to get excellent audio quality, you might want to try a higher setting and verify if there would be any difference. But I think going with -q 5 should be good enough. On the other hand, if space limitations would be problematic, you could also use a lower setting. If you're uncertain, do some tests with the different settings and find out which would suit your needs best!

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Alright, that helped me out more, thanks Liquid and Rimo. I'll do a few experiments and see what works best.

As for my second question. The source files were mp3, however, if I were to come across a lossless audio file and I wanted to convert it, which lossless format would be best to convert to .ogg or .wav? If there is no "best" format for lossless, then nevermind

Hope that is more understandable.

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

Lossless is lossless. There is no 'best' format, other than space and compatibility. For instance, using WAV will give maximum compatibility since any PC out there can play it, but it's a huge space hog.

How ya doing, buddy?

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.

I have an audio question. I am having trouble with specific kinds of pops and clicks in the waveforms I edit. Most of them are short, only a few samples long, and usually centered in the 2000 to 8000 and 10000 to 15000hz range. The person whose voice I am recording has a very clicky vocal tract, so there are a ton of these errors. I am editing these files in Adobe Audition 3, and the included pop and click filter is completely useless for removing these errors, unless I do each of them individually. Is there any software out there that has more powerful pop removal that can be applied to an entire waveform, or large parts of it? It is very tedious to remove all of these manually.

How ya doing, buddy?

I have an audio-question about Ipod. I have downloaded some cd's and on my computer they are grouped in nice folders. When i try to import the music folder on my ipod in I-tunes it turns up into a mess. All tracks that start with numer 1 are grouped together. I know that you can manually import the cd-info in I-tunes, but that will be a lot of work. Is there a simple way to import your music folder from the computer to I-tunes, in a way that the cd's stay in the right order? Thanks for the help.

FELIPE NO

The only way to do what you are asking is to install some kind of third party software on the iPod, such as Rockbox. Then you can choose whether to browse by ID3 tag data or using a directory tree.

It will also allow you to simply drag folders directly onto the iPod and browse them normally.

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

Wasn't sure where to post this but I was wondering if anybody can help me find a way to convert XM (extended files I think what they are called) into any other formart (wav, mp3, etc...) Haven't been able to find a good answer that helps me out. Thankz in advance to anybody who helps me out.

Jam it back in, in the dark.

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Alternatively, install http://www.winamp.com/plugins/details/132367 and set Disk Writer as the output on Winamp

There's nowhere I can't reach.