I'm unable to rip some flac files to mp3 because of this problem. Any suggestions how I might go about it?

Foobar2000 can be your friend with that.

Use the search function of this forum. "resample" is the keyword you'd want to use.


Cheers.


Sergio

BTW, you don't "rip" from FLAC to mp3. You "rip" the content from some physical format (CD, DVD, vinyl) to a digital representation of it on your PC. You CONVERT from a format to another, and when you convert changing the sampling rate you also RESAMPLE. Also, when you convert from a lossy format to another lossy format you are TRANSCODING, while when you convert from a lossless (or original) format to a lossy format you are ENCODING. At least this how I see it...

I knew someone would get me with that. I only realised after I posted that I used the wrong word. Just too early in the morning I suppose.

Okay, so I've installed foobar2000 Special, and I have this SSRC zip file thing. But now I have no idea what to do. Explanation please? I'm no music software guru.

The SSRC zip? You don't need that if you have installed foobar2000 "special". Otherwise I think you have to unzip it to the foobar2000's "components" folder.

Then, IIRC:

in fb2k preferences:

Diskwriter
- Select and correctly setup an "Output preset" (mine is for LAME 3.97 and its parameters string is "--preset fast standard --noreplaygain - %d")
- Under "Processing" check "Use DSP"

Resampler (just "Resampler", not "Resampler (SRC)" or "Resampler (PPHS)")
- Set "Target sample rate" to 44100Hz

Playback, DSP Manager
- Add "Resampler (SSRC)" to the list of active DSPs (ONLY this must be active)

now...

drag the files you want to convert/resample to foobar2000, select them all and then right-click, "Convert", "Run conversion"


Remember to remove the "Use DSP" setting from diskwriter and the Resampler from active DSP's when you've finished converting.

BTW, you don't need to resample, MP3 supports a sampling rate of 48kHz just fine.

Well, can anyone show me how to use LAME 3.97 to encode these wavs? I keep getting an error message in EAC.

Just like you would encode 44.1kHz .wav files, there's no particular switch to use. If you get an error then either the error lies somewhere else, or your .wav files don't have the specs you think.

i wonder what the error message is

i thought you were trying to go from flac to mp3? now its wav to mp3?... anyways

lame --resample 44 input.wav output.mp3
i dunno if that will help you or not

Why resample?

The error:

The error message you are getting is not from foobar2000, but EAC.
Furthermore, you mentioned earlier that you're trying to convert FLAC to mp3, and on the screenshot I can see that EAC is telling you about .WAV files.

Are you certain those files are FLAC? If they are, the first thing you should do is rename them to .FLAC.

After that, load those files in foobar2000's playlist. Go to Preferences -> Diskwriter and tick the "Use DSP" box. Just next to the box you have "Go to DSP settings" button, and be sure to enable the Resampler (keep away from other DSPs, except maybe from gap killer and similar, but you needn't use those). Set Resampler to 44100 Hz.
After that, return to the diskwriter and choose "Commandline encoder - LAME... something". Click "edit", and in Lame's preferences locate the lame.exe on your hard disk (if you don't do that, you won't be able to convert).

Now select the files in your playlist, right click, convert and select your configured commandline encoder - lame.

One more thing. Since you have foobar2000 now, you actually don't need to resample your source files.

I'm sorry, I'm not being very clear here - that's my fault. I decoded the flac files to wav, and I just wanted to encode them to mp3. The way I always have.

Forget about resampling, 48 khz works fine if you go with --preset extreme. The 48Khz wav/flac/pcm which I converted to 48Khz mp3 sounds great, and I believe they sound better because of the oversampling. Nice article here: http://www.earlevel.com/Digital%20Audio/Oversampling.html

There seems to be some debate about resampling. Resample to 44.1 for lower bitrates. 48Khz @ >256Kbs would not seem to make sense.

To keep things simple, I use a separate program to convert files (really). The software I use to encode mp3's is called All2Lame, and can be found here: http://members.home.nl/w.speek/all2lame.htm

Note:
Either put the lame.exe files (the lame encoder itself) into the ALL2Lame folder, OR map its location via the locations button.
Type "--preset extreme" in the box labeled "Switches" for excellent mp3.