![]() ![]() VBR example: ffmpeg -i input.wma -c:a libfdk_aac -vbr 5 output.m4a However, it is considered incompatible with the GPL, so you'll have to compile it (although you may be able to find a build ). This is the best AAC encoder supported by ffmpeg. Then experiment to see what is acceptable for you. ![]() Use a good encoder implementation with sane encoding options. Windows Media Audio V8 - 128 kbps, 44.1 kHz, 2 channels, 16 bits ![]() The wma files have the following properties: I am neutral between mp3 or m4A (or aac, but I understood this is just a different file-extension for the same codec?), so would it make a difference to convert the wma to either of them? (I am aware of a specific mp3 vs m4a question here, so my question is really related to having wma as a sourcefile. 128 kbps? Quid sampling rate, also 44.1 or higher?Ĭonvert through an intermediate format (wav or flac), so in a first step all possible audio info is extracted, and then in a second step encoded again. Straight conversion using ffmpeg (but what settings to use? would it help to use 192 kbps or higher, or is this pointless, and it would be better to actually use exactly the same bitrate as the original file, i.e. And yes, I know that lossless to lossless is generally not a good idea, and that I probably will have some deterioration, but I try to find the best method. So I know that ffmpeg, realplayer, mediamonkey, itunes all can convert wma to other formats, but my question is (i) what would be best "workflow" to minimize quality loss and (ii) what target format/codec to use. I have a few hundreds wma files that I would like to convert to a different format, mainly because my entire library consists of mp3, aac, or m4a, so all compatible with Apple portable devices, and I use iTunes as interface with these devices. ![]()
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