Re: MP3 encoding of wav files from Olympus voice recorder

On Wed, 27 Sep 2006 20:59:40 +0200, Dances With Crows wrote:

On Tue, 26 Sep 2006 19:57:53 +0200, Bernard staggered into the Black Sun
and said:
On Tue, 26 Sep 2006 18:12:55 +0200, Dances With Crows wrote:
Bernard wrote:
My Olympus Voice Recorder VN-960PC generates wav files, and I wonder
why I cannot mp3 encode them using LAME. If I try to do so, I get
clairissa:/usr/share/sounds$ file k3b_error1.wav k3b_error1.wav: RIFF
(little-endian) data, WAVE audio, Microsoft PCM, 8 bit, mono 8000 Hz
file dw_c0106.wav
dw_c0106.wav: RIFF (little-endian) data, WAVE audio, mono 22050 Hz

Should be no problem.

Also give the exact lame command you're using.
below, one typical lame command that I use : lame -r -s 16 --bitwidth
16 -m m -q 0 --noreplaygain dw_c0106.wav trial3.mp3

Let's see now. Your .wav file is *not* raw PCM data, so forget the -r
option. You don't need -s either, since the sampling frequency is
stored in the .wav header, and running 16000 Hz on a 22050 Hz input file
would sound bad. -m m is OK since it's mono. -q 0 should be OK even
though the default is 2 and has probably been more widely tested.
--noreplaygain may or may not have an effect. I'd leave it off since
you reported volume problems on the final mp3s. You should probably use
this instead:

lame -m m file.wav file.mp3

I've just tried this above. I've also tried the same with
--noreplaygain... to no avail. I get mp3 files that are inaudible.
Besides, the playing time is much smaller than that of the original .wav
file (2 sec instead of 22 sec). Only in playing with the -r and -s and
--resample options have I been able to get mp3 audiofiles which playing
time matched that of the wav file... but this was inaudible.

I really don't know where to go from there...


...which allows lame to get the sampling rate directly from the header.
Generally, you don't want to specify the sampling rate unless you know
*exactly* what you're doing.

This one gives something that is maybe less weird than whatever else I
get using other parameters; I can hear that someone is kind of talking,
but much faster than normal.

Yeah, screwing up the sample rate can cause the Chipmunk Effect.

Finally, play one of these .wav files using play or whatever, and make
sure it plays properly.
it does play properly with anything

OK, then the problem is probably in the lame command you're using.
.

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