I downloaded a small wav (It's the THX "Deep Note"). The specs are:

bitrate = 352
samplerate = 11025
channels = 2
codec = PCM
bitspersample = 16
----------
147547 samples @ 11025Hz

I tried to compress this file with several lossy codecs and the results were all very different.

The first one I tried was OggEncGT3b1.exe with a setting of -q 5. I didn't do any blind testing or anything (please don't shoot), but the results were quite close to the original (meaning it SOUNDED pretty much close to what it should (using Bose Tri-port headphones)). The file decoded to the exact same size and sounded resonably correct. I tried to compare the wavs with EAC, but it would not open the original. "EAC processes only uncompressed 44.1 khz stereo WAVs" was the message.

I then tried PsyTEL® MPEG-4 AAC Encoder V2.15 (build Mar 2 2002) with a setting of --normal. The file was encoded, but the volume was VERY low. I did it several times to make sure I hadn't done something out of the ordinary. The decoded wav had the same low volume and the file size was 4 kb larger than the original.

Next up, LAME version 3.90.3 MMX -aps. The file was encoded, but was very garbled on playback. (foobar 2000 v0.7.1 was used for all of the playback). The funny thing is (funny hmmmm, not funny ha-ha) that the decoded wav (again) SOUNDED resonably close to the original.

I tried MPC Encoder 1.14 -Beta-, but it said "ERROR: Sampling frequency of 11.025 kHz is not supported!"

Having said all of this I forgot what the question was, but I would enjoy any feedback that might ensue.

None of those encoders is optimized for non-red-book audio (44.1 kHz, 16-bit, stereo)

Interesting results, though.

My question about this is one to the LAME gurus. I have read elsewhere, and ataque just said that most encoders are optimised for 44100Hz sampled audio data. If I want to encode, say, a 22050Hz sampled file, which of these is the best option:

1) Just use the --alt-preset and no other flags
2) Use LAME's --resample flag (in addition to --alt-preset whatever) and tell it to resample to 44.1kHz
3) Resample the audio first with something like sox then feed it to lame with just the --alt-preset flag

What about 48000Hz sampled audio?

Of course the other problem with say 22050 sampling is on playback. Some of the players out there will then play it 2x as fast ...

I would go this way.

I think that a decent audio editor would do a better job resampling this than LAME.
And then use -aps