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fraz
Hi all,

Well it's been a long night and now I feel like a bit of a schmuck, but here goes.

I've recently switched over to the Mac platform and as it turns out I was in the middle of a large vinyl ripping project when the Mac arrived. I had previously been recording from a decent Pro-ject turntable to my PC (24/96 using a Terratec soundcard), then to soundforge and one of the GUI Win frontends for LAME. The ultimate goal for now is the family mp3 collection, though I have also archived the original WAV files. I have always used --preset standard .

OK, so I installed an M-Audio 2496 Audiophile soundcard on the G5, same recording settings (the AIF files sound great). I'm using the Demo of Peak 4 LE and it can't directly convert to mp3.

And thus begins the adventure. Most of the (few) Mac AIF/WAV -> Mp3 frontends did not give me enough options for LAME settings. Interestingly, two of them just gave me hideous hiss. So I fiddled, pulled out my hair and finally compiled and installed lame 3.93 via fink. Amazingly that all worked .

My first conversion (straight --preset standard) also gave this pure hiss blink.gif , I began to woder if the fact that the raw AIFF files recorded from Peak were the problem (24/96). So I saved them again, but this time at 16/44. Of course, now things worked fine in my command-line setup.

Am I missing something fundamental? Does LAME really require the AIFF/WAV input to be CD-spec or am I screwing up?

Oh well, at least I have a working CLI LAME setup on OS X. Speaking of which, does anyone have some "new" LAME settings for my music? I've been using "standard" for years and I've been wondering if there are any inputs that would retain more of the warmer sound of the vinyl.

Oh and **any** pointers to a way to do batch conversions from the command line would be much appreciated.

Thanks in advance,

JT
Tec9SD
I believe this might help.

QUOTE
-x    swapbytes
Swap bytes in the input file or ouptut file when using --decode.
For sorting out little endian/big endian type problems. If your encodings sounds like static, try this first


(from the LAME "Full command line switch reference" document in LAME 3.96\ HTML\switchs.html)

See how this does for you. I knew reading that would be useful one day! biggrin.gif

See ya, tec

Edit:
Just to make it clear, you need to specify this after [Alt-]Preset Standard.
e.g. --Alt-preset Standard -x sample.wav sample.mp3

Also, my personal suggestion would be, add any "effects" afterward.
(Preferably only during playback.)

I can't help you with a Mac front-end but no doubt there are many.
I bet you, you can find some nice starts by searching the forum plus you don't have to wait for replies.

There is a LAME 3.96.1 version at RareWares for OS X if you wish to try something new.

P.S. For my 150th post, I'm glad I could be useful. laugh.gif
fraz
QUOTE
-x    swapbytes
Swap bytes in the input file or ouptut file when using --decode.
For sorting out little endian/big endian type problems. If your encodings sounds like static, try this first


Hi Tec!

Hmmm, no dice with the -x flag, but..... your comment made me look a little closer at the comments when lame started encoding an it was clear that it was not recognizing the format of the AIF file properly, something like "Assuming PCM" , not encouraging. wink.gif So, I went back in and saved the file as a WAV and that did the trick. Very odd indeed.

Since I record at 96khz/24-bit it sort of bugs me to save as 44/16 but I think that may just be the way to go.

Now that I've figured out that the problems I was having with the front ends were due to the input file I may try the Mac front end programs again.... or just write a shell scrit (if I can remember how....)

Thanks!

JT
pacohaas
you can compile ssrc for mac, just uncomment this line:
#CFLAGS += -DBIGENDIAN
in the makefile and type make in the ssrc directory and it should make ssrc and ssrc_hq

from there you can pipe the output of ssrc directly to lame, which should speed things up a bit and save you some space on the intermediate files.

something like:

ssrc --rate 44100 --bits 16 <inputfile> - | lame --preset standard - <outputfile>

though this might require a wave input file, but i'm not entirely sure. I wouldn't worry much about having to go to 44.1/16bit, I'm not 100% certain, but i think LAME was doing this internally with your conversions on the PC anyways. You might want to consider just capturing at this samplerate as i don't think you will gain anything coming from a record player to such a high sampling.

also, LAME is extremely easy to compile on the mac, you can follow the instructions on the cvs page to get the latest source, then use the standard ./configure; make; sudo make install
then you're good to go.
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