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You might get a warning about "-acodec vorbis" being experimental. You can use "-acodec libvorbis" instead.
Be careful with the "-aq" option, the value 100 (as suggested by a previous post in this thread) is definitely too high. The supported values are listed in the "Technical details" section of the Vorbis wikipedia page.
I used "-qa 4" (roughly 128 kbit/s):
for wma in *.wma; do ffmpeg -i "$wma" -acodec libvorbis -aq 4 "$wma"_aq4.ogg; done
Wma is lossy, ogg vorbis is lossy, lossy-> lossy transcodes will make the files sound worse.
Its better to rerip as ogg vorbis, and if thats not an option, then maybe its possible to find the same files in MP3, ogg vorbis or a lossless codec.
Originally Posted by 1kyle
Why not convert to FLAC (free lossless codec). You can convert WMA to FLAC directly --even on Windows and if you are in Windows Winamp (free) can play FLAC files directly.
Wma is lossy (and I dont trust wma 'lossless' to be lossless). Transcoding a lossy-> lossless file is just a good way to use space, its not going to sound any better.
For sane transcodes (lossless->lossy, lossless->lossless) you can normally use soundconvter (GTK) or soundkonverter (Qt). Some of the linux media players will transcode as well. No need to go poking around the command line to transcode, unless you want to.
Personally I dont need thousands of $$$ worth of stereo to be able to tell the difference between low bitrate lossy files and lossless. I cant deal with any lossy file at lower than 160-192K, to me they sound flat, lifeless and bland.
Most wma files I've ever seen have been 192k/sec, vorbis -q 4 is 128k/sec. AFAIK the ffmpeg 'aq' is a 0-100 quality level, not the q-2 (32k/sec) to q10 (500k/sec) of standard vorbis. It thats right, I'd love to know what bitrate those vorbis files encoded at, it could be a tiny bitrate.
Vorbis is a variable bit rate (VBR) codec, which means that you tell it to create audio of a given quality and it uses however many bytes of data it takes to achieve that goal. The (undocumented in the man page!) "-aq" option of ffmpeg is how you tell it what audio quality to ask of a VBR codec, and the numerical value you give is interpreted differently by different codecs. In the case of the Vorbis codec, "0" is the poorest quality available, getting better with larger values. The value of "60" used here results in audio at roughly 160 kbit/s, which is very good.
Even if wma and ogg vorbis used the same method for discarding information from the audio track (they dont) the ogg vorbis file should sound worse than the wma.
If you cant pick the difference, thats fine for your use. But I wouldnt assume that the limit is your ears, laptops tend to have pretty lame sound chips, and the 'bookshelf'/compact speakers wont be helping.