On date Thursday 2007-08-23 02:47:52 -0500, Michael Poythress encoded:

Before I get flamed for asking an obvious question, please understand that I am not asking for step-by-step help in resolving anything. I am *not* running the absolute latest SVN of ffmpeg, because this is for a production environment where we cannot take the risk of breaking something each day.

I would just like a quick answer, if anyone is familiar with this issue. When running the command to convert audio formats...

ffmpeg -i "AudioInput.aif" -ab 160 "AudioOutput.m4a" ^^^^^^^

At first if this works at all then you're using an old FF-mpeg version, which *isn't supported here* (whit recent versions you should use rater -ab 160k or -ab 160000).

...the maximum bitrate I can get is 152 kbps. No matter whether I change the -ab value to "256", "160k", or "160000" (that last one doesn't work at all; incorrect parameters error). With any bitrate value less than 152, ffmpeg nails it.

Is this some sort of limitation within ffmpeg? Libfaac? Something else?

I think this is the way libfaac deals with the required bitrate value, in this case ffmpeg simply passes the input bitrate to the library. The library doesn't complain about the (eventually invalid) bitrate, but uses some default value for it or use the best valid approximation of the input value. For example I could get a *lower* bitrate (setting for example -ab 32k and getting a corresponding effective bitrate of 61 kbps) but I couldn't get a bitrate higher than 152 kbps. So it seems you have to ask to them...

I have spent a great deal of time searching around for the answer to this before posting, but came up empty handed. So your consideration would be greatly appreciated, lest I unnecessarily begin taking this system apart piece by piece to locate the problem....

Thanks!

HTH, cheers. -- ffmpeg-user random tip #15 Please submit bugreports to: http(s)://roundup.mplayerhq.hu/roundup/ffmpeg/ instead of this mailinglist.