Put All Audio Questions Here
 

I decided to make a Q&A thread about any audio problems that you might come up with. I or some other resident audiophile will try to answer your question in a timely manner. Anything from gamerips to LAME development will get answerd in one way or another. This thread's purpose is to help people rip/encode audio with the best advice and help them troubleshoot any problem's they're having.

I'll get the ball rolling here:
Q#1: I got a pair of Koss KSC75 headphones after my Sony MDR-Q55SL phones crapped out. They have good bass, but the treble is muted (meaning it's there but it doesn't sound loud). I can use the equalizer and bass/treble settings on Windows, but with my Sony D-NE710 there is no such setting. Is there any cheap ($15 or less) device that I can get that will boost the treble of my CD/Mp3 player?

Ok the first part of your question confuses me.
Is it your crapped out Sony MDR-Q55SL or the Koss KSC75 earphone that's giving you sound problems?

Another thing, is the earphone doing this with any source its plugged into? Or only on your computer?

If its just the computer try going to: START > Control Panel > Sounds and Audio Devices > under 'volume' tab click 'advance' next to 'speaker settings' > under the 'speaker' tab use the drop-down list and select 'stereo headphones.' and hit ok.

That should fix the problem!

As for boosting treble with a external device, I can't help you there though I'm sure someone else will.

The Sony phones are dead and the Koss one's are currently giving me crap with the treble. Sound on the PC is fine because I can tweak the treble, but my Sony CD/Mp3 player can't.

Hmm well it does appear to have a 60 ohm impedance.. But that's not enough to really make the earphone sound "quiet" in general while unamped.
Personally, it sounds to me like you bought a bad pair. Take them back and get it replaced if you can. I know they're not bad, infact they're very good for their price :).

It's actually part of the phones. If you read reviews by websites, most of them say the high notes are muted.

Ohh, I see. Then its more 'preference of sound' than a actual problem with the earphone. You made the problem sound like you could only hear the bass coming out of the earphone =p.

Well yes, the treble is a little weak. But what can you expect for a 15 dollar earphone? Technically the busted sony earphones might've sounded better to you with its louder sound (that had a 24 ohm impedance btw, which can be enough to affect the response of the sound.)

Anyway, the only thing you can do is turn up the volume (which I wouldn't recommend) or get a better earphone. There is also getting a portable amp but it would be silly to buy a 30 to 100 dollar amp for a 15 dollar earphone.

Hoefully I'll be able to buy a digital music player that can boost treble, afterall a CD/Mp3 player can't hold much of my music anyways.

Huh? An MP3/CD player can hold every piece of music ever created. It just gets tiresome carrying CD's everywhere you go.

Gots a prob you audiophiles might be able to help me with.

I recently raided my old hard drice and found a bunch of old Icewind Dale speechfiles for characters. I figured they might be useful (and some are quite funny). and since they were .wav files, opening them seemed like it would be easy.

No such luck. Winamp, Itunes, Quicktime, and Audion all refuse to play the files and don't recognize them as WAVs. Any ideas? I'll post a sample when I get onto my home machine, since each file is quite small.

It can only hold 700MB of music per CD (12 discs with mp3s @ 128kbps). I have well over 10GB of music, so I have to use CD-RWs to listen to music (I don't want to waste my CD-Rs).

They could be from a wav format your previous machine's Windows installation supported, but the current does not. It's most likely an ADPCM file or something like that. If you give me one of those samples I could analyze the file and see what format it's in.

Here's one of the troublesome WAVs. Any info you can wring out of them will be greatly appriciated.

According to Game Audio Player, your file is not a wav file, but a ACM file. Use Game Audio Player to playback and convert your ACM file into a wav file. When you run the program, click on the playlist icon (to the right of the Play, Stop, and Track/seeking buttons). Then press Alt+F to scan an individual file or Alt+X to scan an entire directory.

An ACM, eh? Figures. I was converting all the Icewind Dale music on my HD from ACM to WAV when I found the files in the first place!

Thanks for the info though. I wonder why it's got the .wav tag (which it had in the game directory) rather than .acm? Oh well.

EDIT: You don't happen to know of an app for OSX that can do the job, do you? It would save me a step and a half or so :D

Unfortunately I don't, but even having a convoluted long process is better than nothing.

MINIGSF/MINIPSF/MINIPSF2/MINIUSF-->MP3
 

I've noticed that some sets of MINIGSF and GSF soundtracks are at a low volume when played. So when I go to use Winamp to encode it to WAV or MP3, the volume is still horribly low.

How can I fix this? All of the Sonic Advance sets have this problem as well as some of the Mega Man Battle Network sets.

EAC related quesiton
 

I read somewhere that using EAC (esp. the Test + copy function) will wear your drives out much quicker than normal cd ripping programs. Someone said they ripped about 100 cds (on a new drive) then the drive crapped out. My question is, what's the life expectancy for cheap to mid-priced drives that use EAC on a daily basis (I rip about 5 cds a day)? Also will I notice a deterioration in the quality of my rips as the drives begin to wear out. Also, I've noticed in the past (When using other Cd extracting programs like EZCDDAX) that if I rip a large amount of cds in a short period of time they have audible flaws (such as pops/skips). Can this happen while using EAC and the CRC check reports no errors after I've ripped a few Cds. One more question I have is: Do you have any idea why I would tend to get a high number of SYNC. errors during cd extraction for the last track of each CD. It never occurs anywhere else, just the last track. When this happens I re rip the last track using Burst mode... no errors occur. Have any idea why I'm experiencing this problem? Thanks in advance.:)

EAC could cause wear by the excessive CD-ROM use that bad CDs can cause. The constant re-reading of the same sector can wear out the motors that look at different track position.
Tell me what CD-ROM drive you use and I could tell if your CD drive lacks important DAE related features. EAc tends to have some read errors near the end of a track, but for me that's only caused by Offset Correction. Your CD-ROM drive's make and model would be a big help in telling me what your drive can and cannot do.

Use foobar2000 and replaygain the files before diskwriting/encoding to WAV/MP3.

I use two crappy Lite-On Drive
s: The DVDRW - SOHW-1693S and
The CDRW - SOHR - 5239V


Does the quality of the cd/DVD drive used for the extraction have any effect on the quality of the rip???

Yes. But as long as the CD was in decent condition and not copy-protected, the differences between a rip from a good drive and from a lousy drive should be inaudible.

Can you suggest some good mid-priced CDR Drives (Internal). I think I'm going to lose the Lite-On... I think it may crap out on me soon, and I can't stand the noise (it's as loud as a vacuum cleaner). Thanks again!

Well, that question would best be answered at the CDFreaks forum.

In my opinion, optical drives are such small investments that you might as well always go with the best available to you.

For instance, check out the Plextor PX-716A (DVD±R/RW/DL) and the Plextor 52/32/52A.

I'd in invesr in two drives, one for reading (DVD/CD-ROM) and one for writing (DVD+-RW or CD-RW). My Sony DDU-1613 has always served me well and has damn near perfect DAE (it's dirt cheap too).

What is replaygain and how do I use it? I haven't heard of the other program either.

Same here, I thought it was just normal that one of the soundtracks I ripped had a low volume... guess not.

I'll have to try out Arrowhead's suggestion sometime and see what comes out of it.

How do I do this?

Double Post:
Nevermind, figured it out.

Also most good plugins have a pre-amp for volume amplification, set that to a desireable volume if you still use Winamp.

So, turning the volume up in Winamp would make it sound louder before I convert it?

Not exactly what I'm talking about. If your plugins don't support a preamp in the plugins, just move all the sliders on your equalizer up evenly until you achive your desired loudness.

Not necessarily a good idea, as it can introduce clipping if you're not very careful about it.

Double Post:
To "replaygain" files in foobar2000, just select them from its window, right-click and go ReplayGain > Scan selection as…

Yeah, but Lady Miyomi's GSF files are only natively playable in Winamp (you can play them in foobar2000 0.8.3, but it requires a Winamp plugin wrapper). You could then edit the spit out wav files and apply Replaygain to them to delete the clipping.

This is exactly what I've been doing and it works perfect! Thank you both!

Nope. Clipping in WAV files is unrepairable.

Double Post:
If only it worked.

It doesn't - as I just posted above, replaygain won't fix clipped WAV files.

I've been using Winamp to convert them to WAV and then some other program to convert them to MP3. Winamp's MP3 plugin freezes after a few conversions and won't work anymore. I use replaygain on the MP3s. It works for me that way. That's what I was talking about.

It's a convoluted path but here goes:
1. Convert your files in Winamp to wav (or some other lossless format)
2. Go into foobar2000 v0.8.3
3. Scan the files for Replaygain (don't worry if they can't have Replaygan applied to them, the database will save the RG values
4. Open up the preferences
5. Go to the Playback options and make sure the following option is checked:
http://img113.imageshack.us/img113/5...83120894ll.png
6. Go to the diskwriter preferences and make sure the following option is checked:
http://img112.imageshack.us/img112/2...83120897mt.png
7. Then covert the file to any format you please, the files will have the clipping cut off and Replaygain applied.

That'll work.

Splitting different file formats...
 

I'm specifically asking about NSF and GBS files. Is there any way to split them into individual NSF or GBS tracks? If so, how do I do it?

I don't know if you'll like this method or if you can even use it, but what I do is use NotSoFatso with Winamp. I convert an NSF to an NSFE through NotSoFatso. Then, I make one copy of the NSFE file for each track I wish to have. I open up each NSFE in NotSoFatso and edit the playlist so it only plays a single track. When you're done, you should have a bunch of NSFE files, each with a different, singular track. You may have to tweak some other playback settings (such as hiding the track selection window, as it will be useless), but I think that's about it.

Not sure what to do with GBS...

I've heard tips that if you open up an NSF or GBS file in Sony Sound Forge, that you can edit them from there. Just highlight a part of the track, use CTRL+X, and then open a new window and paste it there. You should have two files - one being the original NSF without one of the tracks, and the other file being what you cut and pasted.

Of course, you'd have to do this for every track in the NSF or GBS. However, I haven't tried this out myself yet, but I got this information from Kairyu.

You forgot the first part of the instruction; which is to first convert the NSF or GBS file into a wav file. Most of the time I use winamp to accomplish this. After that you should be able to open the file in Sony Sound Forge.

So, does it make one big wav file or does it know to separate the files?

Running it through winamp's diskwriter will make the NSF/GBS file into one big wav file. So I've been spliting the tracks the way Blue_Kirby2 mentioned. I suppose there is a faster way of doing this but my way does work =).

Someone just mentioned to me that maybe I should record it from my soundcard as it's playing. I'm not even sure how to do that, but I'll try your method first to see if I can handle all the steps correctly. Thanks!

From the soundcard, with cables? That can be done, but I guess that will give you an digital->analog->digital recording instead of a pure digital recording. It could sound pretty grisly. Worth a try though especially if you don't plan on sharing the recording.

I have no idea because this person didn't explain the concept clearly. They just told me to look into it. I'll just use Kairyu's method. Thanks.

Does anyone know anything about .srf files? I'm interested in decompressing some of the .srf files from the "You Don't Know Jack" trivia series to get some funny quotes without having to record directly from the game.

I don't really think anyone will know anything--my gut tells me it's a proprietary format--but hope springeth eternal.

EDIT: Included a sample file. I renamed it to .rar, but it's actually a .srf

Does anyone know how to do XA to WAV conversions? I used to remember this about seven years ago, but I forgot. I have an XA file that I want to convert.

Does anyone know the difference between 'joint stereo' and 'stereo' in a mp3 file? I tend to think 'stereo' sounds better than 'joint stereo' but joint stereo is used when I rip using EAC (from that thread by Moguta)

Joint Stereo analyzes the audio for similarities between the left and right channel and will result in better quality for a lower bitrate if the two channels have little or no separation. LAME determines which is best for each frame in an MP3 and so do most other lossy encoders. You get better quality for lower bitrates with joint stereo, so it's best to use joint stereo with mp3s (though non-LAME mp3 encoders make bad choices on what to use joint stereo on).

You think, but can you tell? Have you tried blind testing J-Stereo vs Stereo vs the original?

No I haven't, but these people have. Hydrogen Audio is the site I respect the most for the final word on the best sounding codecs. JStereo will most likely increase quality on a CBR file and lower the bitrate on an ABR/VBR mp3. Any lossy codec that gives you a choice between mid stereo (aka Joint Stereo) and regular stereo (except for non-LAME mp3 encoders) will benefit more from mid stereo.

I know. We all know. But he claims otherwise, so he's got to prove his claim.

I'll give it a try later I guess... but so far all my stereo mp3's seem so much more 'vibrant' than my joint stereo ones (same bitrates)... maybe its my personal preference.

Yup, personal preference.

Once you know how it works in LAME, you'll see that Joint Stereo is superior to "Stereo", period.

Why? Because "Stereo" uses only "Left/Right Stereo" frames, but "Joint Stereo" chooses between "Left/Right Stereo" and "Mid/Side Stereo" on a frame-by frame basis for the best quality.

Joint Stereo: The Myths And The Realities

Anyway, I urge you to do that blind test. Depending on how good your hearing is, you'll find that either a) you can't tell anything apart or b) that Joint Stereo is better.

I've tried finding programs to do this, but XA to WAV program that's on Zophar's Domain site doesn't work on this computer anymore. I've tried other programs, but they don't work either. Some even advertise XA to WAV conversation, but have no such function. I'm having no luck. Anybody have any ideas?

I do belive PsxMC can import XA files, it does work with most popular PS1 formats.

OK, so I know thsi is going to cause people to go OMG TRANSCODING but here goes:

I have some .ogg files that I want to transcode to mp3 for use in an mp3 player.

What settings/quality blah blah blah should I use in LAME to avoid making the files huge/not lose TOO much quality?

Anything at "-V 5" (don't forget to add the vbr new switch, which I conviently forgot the exact command to) or higher should be workable under most circumstances. It'd probably be best to use a similar preset to the bitrate the source Ogg is at, as not to waste space or lose too much quality.

I suppose just saying convert it to 128k would be too much to ask for LAME, wouldn't it.

:p

And I know how to do that, I'd just be using RazorLAME and an older version of LAME.

Yes, but VBR does a better job of determining what's the best quality. CBR (for bitrates below 320kbps) is pretty much useless, it wastes too much potential quality. If one wants to use 128kbps, then all you need to do is supply an input filename, LAME automatically uses 128 CBR as the default. If you want a stable bitrate and the best quality, ABR @ 128kbps is the best way to go.

I think, in the case of just converting it so you can use it in a DAP, quality isn't much of a concern. I mean, I transcode mp3s to 128k for use on my mp3, as a means of compression.

I'm fine with this, because these are just going to be for my mp3 player, and nothing else. And I don't care enough to look for the differences in quality.

He's not saying he's going to keep said transcoded files (unless you have to for whatever reason.)

128k should be fine, unless you're going to be using the files on the mp3 player for a specific purpose that requires them to be in a much higher quality than that.

I think recent versions of LAME use -V2 as the default.

[Edit]Oh, you're right. It does use 128kbps CBR by default.[/edit]

What are the differences between APS and APX VBR? I'm just wondering since I don't know what to encode my future rips in.

Really both will be indistinguishable in almost all cases (there might be a few cases where APX (aka -V 0) sounds better). If you don't keep lossless backups of your CDs, APX might be the way to go. If you don't really obsess about quality to the extent a select few do, then APS will sound transparent in 99.999% of all songs (especially if the words $100 and headphones never enter the same sentence when buying audio hardware).

Well the thing is, I will be backing up my VGM collection eventually and I am a quality-obsessed ripper. I've also seen #gamemp3s ripping most of their music in APS. That, and I kinda got the impression that APX files have a larger filesize than APS files do, though I can't be 100% sure on that.

If others were to burn my projects onto CDs I would go for APS then, eh? And thanks for the help.

Generally APS mp3s average between 190-210kbps and APX goes from 230-280kbps. The quality difference is not that great, especially compared to 128->192kbps. If you have large amounts of space, then go for APX, but if you backup to small mediums (like CD-R), using APS could mean the difference between having 8 full albums as opposed to 6.
Oh and use LAME 3.97b2 with -V 2 for better-than-LAME3.90.3-APS-quality. You'll get faster encodes and better quality in almost all cases. Personally I'd use -V 0 if you're obsessed with quality though, anything higher would be superflous and a waste of space. OR go with the compromise of -V 1 which is around 210kbps in most cases.

We had this extensive and well written hardware guide on the previous incarnation of the GFF - could anyone bring it back?

As soon as the GFF Archive starts working again, I could just hyperlink the guide. But with this crap, someone will have to supply us with a html copy of the thread.

Can somebody please tell me: What is the sampling rate for PSX XA audio files?

44.1kHz, I think. Though I very easily could be wrong.

I'd agree there. Try 44100Hz, if that doesn't work, 48000Hz or 37800Hz should do the trick. PsxMC should be able to correctly detect the sampling rates of said files.

I have a question.

How exactly do I rip music in VBR? I use Goldwave for my ripping needs and it always rips it at 128CBR. Do I need a better ripping program or is there some sort of settings I need to change on Goldwave?

Get a program called Exact Audio Copy. Goldwave is useless unless you like to rip in CBR. Anyway, the tutorial is all right here.

Whoops, I should have reworded that better. I meant, rip/record music from games. Whenever I rip music from games with Goldwave, it comes out in 128. How do I record music in VBR?

Use GoldWave to record WAV's instead of MP3's. Then you can encode the WAV's to MP3's with something like ALL2LAME.

I have no experience in ripping games with GoldWave, but it's one suggestion. I recommend Sony Sound Forge, but there's Audacity or TotalRecorder as well. And yes, ripping to WAVs first and then converting to mp3 works best. (My recommendation for mp3 converters is dBpowerAMP)

I'm wondering how would I go about removing lyrics from a song? A lot of songs I have, I'm only interested in the instrumental.

HELP!
 

How come i cannot access the sheet music posted by other users!!
when i try to click on the links, it redirects me to a login page, but i have already logged in!!!!

You don't. There are lyric removal plugins for WinAMP and probably for some sound editors, but they never do a good job - either not removing the lyrics completely or taking some of the instrumentals along with them, leaving the song distorted in either case.

If you want instrumentals, find an instrumental version of the song.

How do I rip a lossless copy of the audio from a DVD, then split the audio into tracks that I can convert to Mp3? I want to convert my Cyndi Lauper in Paris DVD into a track by track MP3 collection, but I need some sort of editor that I can use to get the stuff chopped into tracks. Recording it song by song is messy, slow, and you miss stuff. Can someone help me?

*edit*

Let me clairify that some. Can someone help me find a good (perferrably free, but not at the cost of quality) program that lets me edit WAV files into chunks that I can individually convert into Mp3?

Sony Sound Forge is one program, but there are others as well, like Audacity. I uploaded SSF version 7 with the password crack:

http://beta.yousendit.com/transfer.p...13DA834CECCA60

It's a 30 megabyte ZIP file. I don't know how to help you out with ripping the audio from the DVD though.

Lucas - I've got just the program you need: Acoustica MP3 Audio Mixer.
It can output to mp3 (You can select Bitrate and Encoder), WAV, WMA, and RealAudio, although I doubt ANYONE would use those last two. Let me know when you downloaded it, so I can give you the serial. The program is like 5 and a half MBs unpacked, so downloading it should be quick.

Thanks for the help guys :)

Okay, I've got another question.

Why do some people rip CDs in 320 CBR? Is it better than Variable Bi-rate or is there some sort of advantage? It also takes up so much space that I can't see why they'd rip it like that. This happened on an Inuyahsa Theme collection album I downloaded. (They ripped it in 320 CBR. It was a Vocal Heavy album).

It might just be personal preference. That, or some people don't know how to rip CDs in VBR. Nonetheless, VBR is still better than CBR because of better quality, doesn't matter what case it is.

I thought so.

The last question on my mind is about Enhanced CDs. While searching through soundtracks. I noticed on Amazon.com that some CDs have [Enhanced] on their description. So, what do they really do differently for enhanced CDs? Do they buff something out or increase pitches or something?

What would be the big differences from Enhanced CDs and non-enhanced?

Not sure, but I think it has something to do with special encoding of the 15th bit of the 16bit CD audio, called a HDCD. It expands the quality to 20bit audio (or 120dB range, the limit of human hearing), making it sound better, but only if you have an audio player that can interpret the data (WMP 9 or 10 is your best bet). More infomation can be found here.
It could also simply mean a Blue-Book Audio CD, meaning a CD with the audio tracks and music videos, flash movies, wallpapers and *ahem* DRM software.

Yes, that's it. They can also be called "CD Extra".

If you do decide to buy one of these CDs, make sure you hold down onthe Shift key when you load it into your PC. That way any copy protection won't attempt to jam itself into your OS, and you'll be free to copy the CD without restrictions.
Also when browsing its contents, be very wary of any non movie/picture files, EXE programs tend have some crappy surprise in store for you.

Okay, I've been having some confusion about VBR.

I've been cruising on the internet for soundtracks and I find that I see some albums that say something like 224kbps VBR. Is that possible? I thought VBR meant variable Kbps. Do people on the internet just make a mistake about VBR or something?

The 224kbps means the average bitrate of the whole song (or the setting used). For example 192kbps VBR could also mean --preset standard or -V 2 in LAME. Even though the bitrate of the mp3s varies from point to point, it's average bitrate is still measured in kbps to help people determine size and quality level of the file.

Okay, to continue my question off :p

So, when you convert an a file format to another. Do you lose quality or anything?

Like Converting a WAV file to an Mp3 file.

You will lose quality if you convert a .wav file into .mp3. This is the whole point with a format such as MP3. Lesser quality, smaller file.

Basicly, unless you convert a file from one lossless format to another you will always loose in quality when converting audio files.

So if a gameripper wishes to keep his own rips as conserved as possible, would he convert .wav to another lossless format? Or would he just keep the original WAVs? Not that I'd start distributing lossless versions of gamerips out to the public.

WAV (PCM anyway) is uncompressed, so think of it like a lossless format. Sure, you can distribute a gamerip in a lossless format.

But let me just say this:
- If your recording setup is mediocre, you might as well use high-quality MP3 instead.
- If you're doing a gamerip from emulation (for example SNES .SPC music files) just share the original files rather than doing a conversion.

Question here for those listening to audio XP machines. What speaker settings should I use if I'm listening through headphones rather than speakers. Been meaning to set it up properly since last night but fell asleep listening to my batch of CDs through my HD595 headphones. :)

Wrong. 320kbps is technically better quality than VBR, for the very simple reason that the bitrate is nailed as high as it can possibly go.

I'm not even sure if that really has any effect on anything. I have it set up for "desktop speakers" and I use the dedicated headphone output on my M-audio Revolution soundcard for my headphones.

Alright, was just really interested on someone else's opinion because I had switched it to headphones and it sounded a bit muffled. I'm trying to find the best settings for my Audigy 2 ZS minus the hub. :doh: Was advertised on newegg for dirt cheap about 2 years ago. Either way thanks for the input.

So then, is there any area of quaility or anything that is a disadvantage of having it in VBR? Is there any reason to have an album in VBR as opposed to 320kbps? (Besides the massive amount of size)

Basically I'm asking, VBR vs. 320kbps?

Double Post:
So far I've heard that 320kbps doesn't have as good 'highs' and 'lows' as VBR does.

No reason really. -V 0 (aka --preset extreme) will go to 320kbps when the audio is suitably complex, not decreasing percieved quality at all. LAME does a very good job with this, which is why rock albums encode at >280kbps and Sega Genesis VGM rips encode <128kbps with -v 0.
If you want anything sounding better than 320kbps LAME mp3s, go with AAC, Musepack, Ogg Vorbis at 256kbps or just go lossless.

Would someone (or some people) be able to review this track for problems? It's a sample rip from Gamecube, and IMO I think the bass sounds a bit scratchy.

I'm not sure whether it's caused from recording too loud, or the music itself.

Just the massive amount of size, yeah.

The way VBR works is by choosing a bitrate between 32kbps and 320kbps on a frame by frame basis. If the signal is simple then it doesn't need a high bitrate, so it gets a lower one. If it's complex, it will need a higher bitrat so it gets a higher bitrate.
CBR 320 just uses 320kbps (the highest possible bitrate, remember) for every frame.

I really doubt that.

So, wouldn't having a 320 CBR for a less complex song hurt it's quality?

No it'd just waste space. Encoding a song with 5 minutes of silence (like bonus tracks on CDs) would be an utter waste, as those silent frames would be perfectly encoded at 32kbps.
It'd be like encoding a 8000Hz 8bit Mono file at 160kbps, which is even above the lossless bitrate, Shure, you'd have the best representation of the original audio, but you could do 99.99% as well with -V 0 or similar preset.

Thanks for all of the help guys. I'm sorry for pestering you all so much will all of my questions but I've had some of these questions for a while. :juggler:

I'm glad you guys were nice enough help me.

I'm using a Batch Processor on Goldwave to take all of my albums and songs in CBR and convert them to VBR. What I'm wondering though is, how is it possible to actually convert a song from CBR to VBR? I though Birate is permanent and that no program can change it when it's ripped in CBR.

Another question is, would it have the same quality as directly ripping a song from the actually CD in VBR?

In any case, it's always best to rip the VBR mp3 from the source material, and not the CBR mp3. The only reason I could see to convert an mp3 from CBR to VBR is to lower the bitrate (like 320kbps to -V 2), which has its purposes, or if you have something like Ogg Vorbis files you can't play on a certain media player.
If you must convert CBR mp3s to VBR, I do believe LAME should be able to accept the mp3 without question. Just use whatever VBR preset you want and LAME will re-encode the file. There's no point in going for a higher bitrate BTW, so if you wanted 128kbps CBR mp3s to sound like 192kbps VBR mp3s, it's not going to happen. Like I've said, re-rip the CD tracks and keep as lossless copy if you can, then re-rip the tracks to mp3. THIS is the reason many people use lossless, so they can re-encode when it proves advantageous without having to do the nasty lossy to lossy process of transcoding.

Long story short... making an MP3 from another MP3 will never improve the sound quality. It will always diminish it. Make the new MP3 from the original source instead.

You might wanna take a look at Mp3packer. It seems convert 320kbps to smaller vbr files without loss. I never tried it myself though.

Unlikely. If you transcode 320kbps files to -V 5 (~130kbps VBR), you won't really hear any differences unless you have extremely good hearing.

Unlikely or not, it's for real. The guys at HydrogenAudio have tested it MP3's shrunken with it give bit-identical input to their original copies.

Mind you, it typically only saves a few kB per minute.

You don't need "extremely good hearing" to tell the difference.

Hi there, i don't know if this is the right place to ask this but here goes, i wanted to know the link to a website that had the directsound resampling plugin for winamp. These forums even had a guide on how to improve quality in winamp and etc. i already grabbed shibatch but i wanted the ssrc for winamp version 2.2.6 if not mistaken. thanks everyone hope someone can help.

I do believe shibatch's directsound plugin inlcludes SSRC, I don't use WinAmp anymore, so I can't confirm this.

Before the GF forums went down there was a pretty good program listed for converting lossless files to .mp3 files. Any recommendations for this, for coverting lossless files to lossy?

I use ALL2LAME or Multi Frontend (by the same author) depending on the situation.

Multi Frontend - Nice for encoding to lossless formats or transcoding lossless to lossy.
ALL2LAME - Can only encode to MP3, but gives you the ability to use LAME's --nogap command which Multi Frontend doesn't.

These programs are just "frontends", meaning that they are workable interfaces for CLI software. You can find just about all the CLI encoders and decoders you'll ever look for at RareWares.

Would anyone know a good program for converting file formats? I have to convert an album in .ogg format to Mp3 format and I'd like to know a good program that would do it well without lossing too much (or any, if possible) quality?

I use dBpowerAMP (a cracked version). However, converting a lossy format to another lossy format will make you lose some quality, but not a lot. Just for the safe side, I'd convert lossy to lossless and then back to lossy again.

Ex: OGG > WAV > MP3

Again, there's no possible method in converting a lossy file to another lossy file without losing quality.

:lolsign:

I'm ashamed to say I actually missed that when skimming through. :(

Oh Well, thanks for the info guys.

Has anyone made a good Spc plugin for foobar2000 yet? The OpenSPC is a real piece of shit, it has so many missing sounds. If I could find something that could do as well as SNESAmp, I could give the Winamp Wrapper plugin the boot.

I use GEP (Game Emu Player). It's very accurate. It will not make your SPC's sound super fake and sparkly like Winamp's SPC plugins tend to.

Is there any way to disable (or override) the other formats GEP supports? I like foo_vgm much more than GEP's VGM.

Go into Preferences, and look at the input plugins' section. There should be a section for Game Emu Player where you can tell it which formats to handle.

Here's an issue that's been driving me bonkers.

I've occasionally seen rips of DVD audio that contain just the music and some SFX from a movie, and none of the dialogue. I assume that this is done by isolating one of the sound channels and recording it with a line-in or some such thing.

My question: how does one do that? My computer's DVD player doesn't seem to have an option for seperating the channels, and I'm not sure where to look for something that does.

Any suggestions?

Maybe using foobar2000 with DSP and its disk writer.

Well, usually the music and sfx come from the left and right speakers, while speech only comes from the center speaker. So you probably are missing the sound that comes center (and surround speakers).
Try setting your soundcard/mediaplayer to stereo. This way the 5.1 channels will be mixed down to stereo and you will hear all sounds.

Thanks, but in this case, I'm not trying to hear all sounds. I want to split the channels so I can get an SFX/music only track, which I know is technically possible.

It's the closest thing to ripping an isolated score...without the isolated score!

You should be able to do that with Besweet. To make your life easier, get the BesweetGUI as well.
I guess you have to play with the output/input at the Azid settings to get the left and right channel only.

http://dspguru.doom9.net/

deleted... sorted it out...
THanks,
Tom

.gsf files are a pain. Is there a certain reason why GBA rips are ripped in that format over mp3? Is it better quality?

Potentially, yes (it would depend on what you're using to play them).

They are "better" quality for the simple reason that they are the actual music data from the game, rather than just a recording of it.

Plus they take up a lot less space. VGM rips are (at their largest) only 4MB, while a 128kbps MP3 of one of the songs could be bigger than that. Also VGM, SPC from what I've noticed are the smallest music files I've seen able to playback 96kHz music.

A small question about NotSoFatso: when I open up an NSFe file it starts the first track at the very beginning. But the next tracks will always play with a small "fade-in" that's very annoying. Anyway to fix that, or should I just stick with Foobar?

hello guys ... i wanna know exactly before i download the programs what types of cables i should buy to convert from a video game cd to mp3 ???

If you can put the game CD into the computer and see files (or better yet, audio CD tracks), then you don't need any special cables.

ok mate but what about these types like nintendo 64 ?

how can i convert their musics ??

and some cd's cotain specefic files (not audio files) ... so, any ideas ?

I don't know, but you can try googling. There ought to be N64 music sets already out there. It would then be just a matter of playing and converting them in a good player like foobar2000.

if there's a folder called sfx, sound, audio, music or anything like that, take a peek and see what extensions the files have. Then google it.

Switched to foobar (092) from Winamp. Says on the homepage that it supports MPC, M4A, AAC, etc.

I put in .m4as, foobar says 'FUCK YOU'. Is it too much to expect a new version of a player will support such formats without having to fag about with plugins?

It should play M4A just fine out of the box. I'm using 0.8.3 and that's how it is for me anyway.

Maybe there's something wrong with your M4A's, or they're not M4A's.

I may have asked this question before, but I was wondering if there is a way to save streaming MP3/OGG/AAC streams directly to the HDD without any transcoding.
I'm getting into streaming radio again and would love to be able to save streams onto disk.

What is the advantage of having a playlist? Does it primarily making playing the mp3's faster? How does it affect mp3 players (like what would be the difference of playing in album without a playlist and one with one).

None really, it's just easier to open a whole album by clicking on one file as opposed to selecting all of them individually (or roping them as is the common method of selecting multiple files with a mouse). It also is very handy for preserving a play order that the filenames wouldn't permit a (such as in Project2612 rips, where the files have no trackname, and thus the m3u playlists preserve the order they should be played back in).
Also (I haven't dabbled in this yet), it's possible to store the metadata of a music file without altering it by storing the information in the playlist. Most newer (meaning post m3u) playlist formats support this natively, while m3u playlists need the player to be able to read or write the metadata to the m3u file.

Kind of a dumb question, but how would I merge two tracks together? I'm working with DSPs and I want to combine the left channel file and the right channel file into one file. Thanks for any help.

First get Audacity 1.2.3 or newer:
Just ignore all sample rate information and Ratchet & Clank references.

Ok, I've got a question.

http://systema.googlepages.com/DelightfulGirlOSTss.gif
I recently downloaded an OST from a k-drama. When looking at the file names in windows, they appear as korean characters (as above), but when I play the files in winamp, they show up random symbols. Do I need some sort of plugin for winamp?

More importantly, when I put the songs in iTunes and transferred them to my iPod, they were again, random symbols. iTunes also messed up the album and made each song into it's own album folder, probably because of the this.

Any help would be appreciated. Thanks.

Support for foreign characters like that in MP3 players is still a horror show.

It'd probably be best to translate the Korean characters to syllables in western lettering. Sorry, that's probably the last thing you wanted to hear.

I don't know anything about MP3 players but I was wondering if any of them support FLAC and if so, any recommendations?

http://en.wikipedia.org/wiki/Flac#Hardware_support

iPod (if you replace the firmware with RockBox firmware), Rio Karma and various iAudio models.

Myself, I'd go with the iPod Video.

Quick question regarding foobar, flac and replaygain:

Would it be wrong to apply RG (albumgain) to a file and then set the preamp to a positive value (over 89dB)?

The reason i am asking is because patchmix (the program monitoring and controlling my E-MU 1212m card) shows a rather low input signal and when i adjust the preamp to a positive value the input signal rises and becomes normal. Spesifically at around +6dB preamp.

But wouldn't that contradict the purpose of RG? I mean lowering the volume.
I don't know. That's why am asking you guys.

Yes, it does sort of contradict the purpose of ReplayGain.

If patchmix is complaining about a weak signal, then either something somewhere is lowering your volume further than it should be lowered, or the monitoring part of patchmix is not very well calibrated.

Can someone suggest me some softwares (preferably with a download link) to convert .minigsf files into .mp3 ones? Thanks!

WinAmp - to convert .minigsf into .wav

dBpowerAMP
Exact Audio Copy - to convert .wav into .mp3


You can download WinAmp and EAC from their respective websites, just do a Google search. As for dBpowerAMP, I got a cracked version via torrent, I can uplaod it for you if you need it.

Are there any websites explaining how to convert the minigsf into mp3s? I can't seem to find any button that does the trick.

In Winamp, type ctrl-p to access the Preferences menu, click on Output (under Plug-ins) and select the Nullsoft Disk Writer plug-in. After configuring it to your preference, put the track(s) you want to convert into your playlist and play them. Instead of playing them, Winamp will begin converting the tracks to wav files. Hopefully you can take it from there.

Yes, they worked perfectly. Thank you guys so much! :) Now I can finally enjoy my music in the mp3 player.

I want to rip music from a PC game, and I know Audacity can do the job. I want to record the music directly, no wires and all those stuff involved. Just play the music of game, hit the record button and wait till the song's finished.

I've worked with Audacity earlier with ripping PC music, and had no problems, but now I have.

I got a new computer and stuff. I needed to download Audacity again, no problem.

Now I want to record music again the same way I did before, but no matter which option I choose to record (Microphone, Line-in) it won't record any music!

I don't know what I'm doing wrong, some help would be nice!

I'm pretty sure you can just hit record without the use of any line-in or microphone options, if you're recording from the PC. I was able to do it yesterday.

That's the problem, I can't.

I know it sounds weird but I can't record any music!

Does it refuse, or is the recording silent? If the latter is the case, make sure you windows volume meters are around 50% for Mic and Line In.

A friend of mine recorded a gig on his mp3 player, the player in question is a Venture. This company seems unheard of, the item cost 160 euro or something but anyway nevermind that.

The tracks are .WAV files but when I attempt to play them they are called unsupported format, the format is invalid on my psp aswell. They play grand on the actual mp3 player but they don't seem to play elsewhere at all. Anyone any thoughts on how this is happening, know of any codec or software which could help? Cheers in advance

They sound like mp3s in a WAV format. Try renaming the wav files to mp3 and see if that works.

My guess would be that the WAV's are in ADPCM format.

foobar2000 ought to be able to play them.

Yeah I thought you had it there actually when I read this today but it turns out doing that just corrupts the file, it seems to be a variation of a .wav file because its requesting a different codec

Cool, gonna download that now, cheers man.

There used to be a great thread here about the best way to listen to your music on winamp and such, and it gave 2 plugins. One was the MAD plugin for winamp, what was the other one?

i think this question would go here. Does anyone know the name of those keyboard guitars everybody used in the 80's?

You don't need special plugins to play MP3 with the highest possible quality on Winamp. The Nullsoft MPEG Decoder doesn't have any problems.

What's the best way to convert FLAC files to MP3?

With something that uses LAME.

My favourite is ALL2LAME

It needs the FLAC and LAME encoders to do its job. You can pick them up here.

I have a digital piano with a 1/8 headphone jack and I'm wondering if i can use an audio cable to connect that with my computer's microphone jack and use recording software to record what I play? Does that work?
Thanks

You should be using the computer's "line in" or "auxiliary" jack if it has one, NOT the microphone jack. If you don't have a line in or auxiliary jack, then you should get a USB audio recording kit.

But yes, a headphone jack can be used as a makeshift line out. If possible, set the volume on the keyboard pretty low. It's better to have a signal that you have to amplify later, than one that is so loud it's washed out.

Thanks, I'll see if I can get a line in recording device. What's the difference between a line in and mike jack? What would happen if I used the mike jack to record...?

Mic in's are usually mono, and not as good quality in's for recording generally. In fact, unless you regularly use a microphone, mute 'microphone' in your windows (or other) mixer.

- Spike

Also, I may be wrong, but I think mic ins also usually expect a lower level signal than line ins, and so it will be tougher to get the recording not to "clip"... might even be more likely to fry the mic in than the line in.

Does anyone know how to extract music from a DVD? I want to record the credits song in a movie.

help please!
 

Few days ago I downloaded from e mule one folder (Beethoven Piano sonatas Vol.1 with Glenn Gould). In that folder I got one big file (235 mgb) which contains music (.ape) and notepad which is called "readme". I opened notepad and it says this :

What is it:

1. APE file - this is original WAV file ripped by EAC and compressed by Monkey's Audio.
Monkey's Audio is lossless(!) compression (much better than RAR -mM), so when you decompress
the file you can burn it back as Audio CD or encode it with your favorite MP3 tool.
Absolutely no quality loss - you have the same (or as close as possible) WAV file as on
original audio CD.

EAC Tutorial : http://www.ping.be/satcp/tutorials.htm
Monkey's Audio : http://www.monkeysaudio.com

2. CUE file - this file can be used to either split WAV file(not APE!) to tracks (e.g. using EAC)
or you can burn it as audio CD.
To burn it as audio CD - decompress APE file, put WAV&CUE in the same dir and use Nero/CDRwin/EAC/etc.

Why APE:
http://www.monkeysaudio.com/comparison_compression.html


So, I'm curious now, how can I decompress the file so that I can burn it back as Audio CD?

Thanks in advance!:)

Very easy, I did this this week- go to the Monkey's Audio website and download their program. Open it, load the APE file, and select it and hit "Decompress".

Walk away from the PC for a sec so it works well of course :)


Yeah- does your DVD have a VOB file?

Get DVD Decrypter and demux it, etc. It's pretty easy to use, there's a tutorial somewhere.

If you still need help after DL'ing it, I'll post how to do it.

- Spike

I use Imtoo's DVD Audio Ripper (Free Download) for this. Pretty straight forward and easy program to use.

Thanks a lot!!!:)

I also have 2 two questions:

1) How can I reduce quality of the song? I decompressed using Monkey's Audio, but this song has 700 mgb, so I want to reduce it..
2) Which program is the best (and easiest:)) for cutting songs and separate it on a few parts?

Thanks!:)

Just use the lossy codec of your choice (such as LAME mp3) and encode the file. If you want to make it easy, use foobar2000 and it'll load the CUE file and make it easy to split it into individual mp3s.

If you're cutting songs, then keep it as a WAV file until you're done :)

I use CoolEdit, but free programs such as Audacity and so forth can be used as well. Of course, I didn't pay for CoolEdit :)


On DVD audio extraction, do you mean from VOB files? I use CoolEdit also for that, just hit 'Extract audio from video", change 'avi' to 'all files', and select and let it go.

- Spike

OK, so here's the deal. I like to record music that I write, but it all sounds extremely soft, although the balance and equalization are fine. I have to turn the volume all the way up on my computer to hear my songs. Would MP3Gain or ReplayGain solve this problem, or will that just give me clipping?

ReplayGain has a built-in mechanism, not only to prevent clipping but to remove clipping already in the waveform. So no matter how much +dB ReplayGain gives you, it shouldn't have the file clip, it should in fact peak perfectly at ~1.000000.

I'm pretty certain I know the answer to this already, but I'd just like to be 100% sure - if I was to decode an APE file to WAV, and then encode to mp3, is there any loss of quality in doing so? Somebody sent me a CD rip last night that was compressed all into a single APE file, and this is the first time I've attempted to decode + encode it to mp3.

If you mean is there any quality loss between APE and WAV, then there should be no loss in quality. Of course mp3, using lossy compression will have some quality loss; but (provided the APE files are not corrupted), CD -> APE -> WAV -> mp3 should be the same as CD -> mp3.

BTW: If there's a single APE file for the whole CD-rip, be aware that you'll probably need to open a CUE sheet to properly find the track positions for the mp3s to split properly. foobar2000 could open the CUE sheet and even do a straight APE -> mp3 conversion without too much hassle.

Yeah, I meant converting APE to WAV for the most part, I'm already aware that mp3 is a lossy format.

I've already done that, considering I searched this forum for APE > WAV tips. Thanks though. :)

Just a quick question. How do you rate these in terms of best quality to least best quality? Uncompressed Wav, 320 CBR MP3, and any of the variety of Lossless File Formats (I guess FLAC is the best). I've been using Moguta's guide to compress my music into VBR MP3s using the setting "-V 0 --vbr-new %s %d" which is like the best 192 VBR setting I guess (I forgot, but the reason I only use 192 VBR is because my portable MP3 player can't play any of those lossless formats, so I don't really bother using them.

I guess my question would be for archival purposes. I just recently bought a CD and make an ISO of it to store on my hard drive. I also ripped it to WAV files and 320 CBR MP3 files, so I'm trying to figure out which one is the best quality and to get rid of the least best quality to free up HD space. I guess if I just want to listen to it on my computer I can try out those lossless compressions like FLAC.

WAV, like any other lossless format is just that - loss-less. For archival purposes though, you'd want whatever is able to compress your .wav files/.iso file the most. It doesn't matter which of the these two you keep as they are of the same quality. The .mp3s though are of lower quality, the MP3 format being a lossy audio format. So try messing around with either FLAC or APE for your .wav files - and for the .iso, just use something like WinRAR.

When you encode using a variable bitrate setting as you're doing (and should be), the CBR value you choose has no importance, as you won't be using a constant bitrate anyway. You can set the drop-down* to whatever [*I'm assuming you're using EAC here].

Is there any way to make the end of a song with a natural fade-out, uh, not fade out? Like some way to make the fade out portion the same volume as the rest of the song? I want to loop a song that fades out, and having it fade then start again is kind of weird.

You can try to edit it, but I don't think it'll give much good. I think that in that case, the volume change must be damaging to the audio quality in the last parts of the fading, and that simply inversing it will render something ugly. You never know before you try though ~

Drakken,

Do you mean, a piece of music is 50 seconds, and instead of the loop ending at 50 seconds it doesn't cut off, but rather, fades the last few seconds at 47-50 or similar?

I think it could be saved. But it'd be tough. Samples please :)

- Spike

This is probably an insanely stupid question, but I haven't been active here in quite some time, and I've forgotten how the HUB thing works... >.<;;;;
If anyone could possibly explain a bit, or direct me to a site that explains....

@_@;; thanks...

How can I get rid of the two-second intertrack buffer Nero always bungs in, without having to burn from cue files (ie. how can I avoid it if I'm inputting separate tracks)? Having a great quartet just stop and then resume shits me up the wall.

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

http://www.raborak.com/galerie/albums/userpics/nero.png

Right click on all tracks, properties, change the Pause value.

That's Nero 6 btw.

Yeah, that's the kind of thing I'm talking about. Here's the song in .wav format:

Soothing Theme

I included some on each end (mainly before the fade) so you could get a good idea of the volume of the rest of the song. Thanks for checking it out.

How about this:

Soothing Defaded

http://img.photobucket.com/albums/v1...ek/Regular.gif

http://img.photobucket.com/albums/v1...amek/Fixed.gif

Been there done that, mate. On burning I just get a prompt telling me Nero (6) can't do the job. Says something about a minimum of two secs.

Also tried alternating between Track- and Disc-at-once. Nuthan.

I'm trying to rip DS music using a rom and VGMTrans, but I can't even open the program. I downloaded and installed the C++ thing and the DirectX deal the site said to, and restarted, but when I double-click the .exe to start the program, I still get this same error. I'm currently using Windows 98se. What should I do to get VGMTrans working?

That's weird, the prompt is usually just about the first track, not the others ...

What version of Nero are you using ? =/

Use Disc at Once, and set the first track to have 2 seconds, the rest, zero. It will work.


The Sendspace link isn't working right now, and I'll wait to see your (Drakken's) response to Piccolo's work. :) Visually it looks promising ;)

- Spike

The sendspace link works for me. I'd really like some feedback on it. Being a professional audio editor, I certainly hope my work is up to par... :)

I'd also like to say that if anyone has any questions or problems concerning waveform audio editing in particular, feel free to ask me, I'm always willing to help. (And always looking for a challenge, too.)

PN: It wasn't actually 'down', it said something like 'this sendspace link is inaccessible for the moment', probably maintenance or something.

And wow, a pro audio editor huh? Makes me jealous :)

Here's a few questions off the top of my head:
1. What plugins do you use for noise reduction (actual reduction based on samples, not just transforms that use defaults and destroy audio)?
2. What are some good methods to enhance the stereo field of tracks (such as making a stereo file where most things are clustered in the centre pan a bit more or at least have a wider field)?
3. Are there any good reverb effects you know of that don't stop the audio sounding crisp (as in, don't saturate the audio)?
4. Is there a real reason to use 'floats' (32-bit) when applying effects to 16-bit audio, or when copying/pasting into a new waveform?

I'm sure there's more, but that'll do for now ;)

- Spike

Heh. "Professional" as in I earn 100% of my income from audio editing. People tell me that my work is also professional quality, but I'm not sure if I think so. Perhaps that's just my insecurity showing.

There's something I should clarify. I very rarely work on music. Most of my work is done with vocal and spoken tracks, and a lot of my expertise centers on precise manipulation of the waveform to correct speech errors and whatnot. For example, I was working on a radio advertisement recently, and the person who was speaking screwed up and said "cylinder block" instead of "cinder block". I seamlessly edited it to say "cinder block". (You can listen to that here). I also fixed the timing of his speech and removed any unnecessary vocal tract noises. My of my work consists of stuff like this. Basically, I am more of a dialogue editor, rather than an effects editor.

Some of these skills can be used for editing music as well (mixing, splicing, correcting errors, etc) but I'm not involved in making music or in the process of creatively applying special effects.

To answer some of your questions, I don't normally perform noise reduction on the tracks that I work with because our equipment has a very low noise floor. When I absolutely need to reduce the noise level, I use Cool Edit's built-in noise reduction, which samples the noise and has many adjustable parameters.

http://img.photobucket.com/albums/v1...ek/Noise-1.gif

In the case of your second question, unless you have access to the original multi-track layout, there is no way to selectively move things within the stereo image, although you can expand the width of the whole stereo image. Cool Edit and similar programs have dedicated tools for manipulating the stereo image of a file, including special effects like making a rotating stereo field or doppler effects.

http://img.photobucket.com/albums/v1...mek/PanExp.gif

As for questions three and four, I really don't know. I'm afraid they are beyond my range of experience.

Hey, that's pretty good. Thanks. :)

Do anyone know to rip 3do music?

Heh, glad I read this thread! I'm currently doing just that.

I've found it impossible to extract/rip the music, but you can get the sound by using an emulator and simply recording with your sound card ('what u hear' or 'stereo mix' or whatever records digitally for your card).

Since the emulator was messing the sound up, I found a workaround- I started recording, then minimised it and it played fine- might not apply to you though.

Which game you doing, out of interest?

- Spike

Just a simple question.

Is m4a lossy or lossless? I got an iTunes exclusive track that I want to convert to mp3, though if m4a is lossy I probably won't do so and I'll just convert it to WAV or FLAC.

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.

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!

Yeah, this track I'm looking at is lossy. Crap. I guess I'll never get an authentic lossless version of the exclusive iTunes track from The Simpsons Movie. :(

I don't exactly know what DRM is, but I usually convert lossy files to WAV with dBpowerAMP, and then recompress to FLAC with EAC.

Thanks, guys~

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.

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.

Also, DRM is an acronym for "Digital Rights Managment". It's basically the copy-protection that's on most online music download services. There are some websites that don't have this, but most do.

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.

I probably already know the answer to this, but does increasing the latency of a sound card have any possible effect on output quality?

I only use my Audiophile 192 to listen to music, so I bumped up the DMA Buffer to 2048 samples to reduce any likelihood of pops and other artifacts. Sometimes, I get the impression that the music is more forward-sounding and cleaner if I reduce the latency, but it's probably my imagination.

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'm using EAC to rip my CDs. I've noticed two of my CDs have the first track title in red, rather than black, indicating a 'track 0'.
(Hybrid - I Choose Noise and Queens of the Stone Age - Songs for the Deaf)
How do I rip these hidden tracks?

EDIT: Problem solved!
Action> Copy Range
Move the sliders to the appropriate time on the cd, then extract!

How can I make a cue file for individual ape files? Never sussed out how to do it in foobar, nor dbPA.

Treat the ape files like wav or mp3. For example, convert the ape files to wav and write the cue file. Then open the cue file and replace .wav to .ape.

I don't know if this is the right place for this question, but here goes:

Are there any Gameshark codes that mute sound effects in N64 games?

Help...
 

Okay. I just can't do it. I would like to convert .minisgf files to .mP3 format. I have used Winamp, gone through numerous extentions and such. It's really starting to eat me up. I was wondering if there was someone who could convert for me.

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

-_-;
 

I got it now. Thanks anyways. I figured out how to do it.

And does anyone yet know if there exist Gameshark codes to mute sound effects in N64 games (for clean music rips)?

You may want to read this:
USF Central

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

Hmm...I'm not sure if that was directed toward my post, but the main thing a friend and I were wondering was whether there existed Gameshark codes to mute N64 sound effects, so that recording music from the games can be cleaner.

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. ;))

Recording net radio: HOW TO?

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

What is the best program to edit audio tags?

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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?

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.

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.

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

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.

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.

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.

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 ;)

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)?

I suspect patent issues ;)

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

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.

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.

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.

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?

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!

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.

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.

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 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.

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.

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.

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.

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

Sorry if this is a dredge but it said to place audio questions here. This is actually for a friend she's trying to rip from Sugoi! Arcana Heart 2 some missing music from the soundtrack and she's totally lost. Here's her question:

First off, I want to rip from a Japanese game. Is there any special plugin required because of the different formatting? Also, what would you recommend to get the job done? I have cube media player v.2 but I also heard of Alternates. I have no experience ripping, so please if you could start from the ground up. I don't know any of the audio terms. Thanks so much!

Any help for us would be appreciated.

I'm interested in Ripping the songs off my Bleach: Blade Battlers 2nd.

I know that MFAudio can play and extract the songs into wav by having the right interleave and offset, but I'm lost into finding the offset of each song.
Right now I can only rip the first song.

Cube Media Player 2 can find all the songs, but plays in heavy static/scratchy songs.
Might I be able to pull the information from here, and make it playable in MFAudio?

Edit: Awesome, I finally got it to work in a different manner.