1

I'm on Windows x64 converting MP3 file to AAC using FFmpeg.

I tried these commands:

ffmpeg -i file.mp3 -acodec aac -strict -2 file.aac

ffmpeg -i file.mp3 -c:a libvo_aacenc file.aac

Both commands result in an increased file size. The output file is is about 50% bigger than the file source file.mp3; the second command increases the size more than the first.

Is this as good as it gets or are there any options to convert to smaller size without losing quality?

Improve this question

1 Answer 1

Reset to default
4

file.acc is 50% bigger than source file.mp3. Is it ok?

Default option settings will be used if you do not choose a quality or bitrate. For the aac and libvo_aacenc encoders this will result in ffmpeg automatically using -b:a 128k. Since these encoders only accept a bitrate instead of a quality level then you must adjust the bitrate value higher or lower to get your desired output quality or file size. Alternatively, depending on your ffmpeg build, you can use another AAC encoder, such as libfdk_aac or libfaac, that does allow you to target a specific quality with the -q:a option.

Or are there any options to convert to smaller size without loosing quality?

Technically, no. This is because you are going from a lossy format to another lossy format. General recommendation is to avoid re-encoding. Depending on what you want, and if you must re-encode, then the goal is usually to use a good enough quality that you hopefully do not notice a change. If file size is important then you must find a good balance between quality and file size. Quality is subjective so I can not tell you exactly what will work for you, but you can find some useful information and examples for each AAC encoder supported by ffmpeg at the FFmpeg and AAC Encoding Guide.

Improve this answer
9
  • Thak you for you response. If my mp3 say has 81k bitrate and I do conversion to aac with the same bitrate of 81, i'm I supposed to fully maintain the quality?
    – WHITECOLOR
    Feb 12, 2013 at 7:01
  • @WHITECOLOR No. In addition that AAC is a lossy format, not every encoder is equal (same goes for formats) in terms of bitrate requirements to achieve a certain quality level. Why do you want AAC anyway?
    – llogan
    Feb 12, 2013 at 7:05
  • Well, encoded files are going to be distributed (via internet) for commercial project mp3 in not for free for that, aac licencing is less strict. If ACC is lossy why it is "(AAC) is the successor format to MP3"?
    – WHITECOLOR
    Feb 12, 2013 at 7:23
  • 2
    @WHITECOLOR AAC achieves better subjective quality at the same bitrates when compared to MP3, however that will only be noticeable when you a) convert from the same uncompressed source and b) choose a low enough bitrate. If you're doing MP3 → AAC conversion and stay with the same bitrate, it might just work out for you, but you will have to listen to the files. There's no rule of thumb here as to how big the degradation will be.
    – slhck
    Feb 12, 2013 at 7:28
  • 1
    @WHITECOLOR Furthermore, libvo_aacenc is considered a pretty bad quality AAC encoder. If you can, choose libfdk_aac or libfaac, which unfortunately (due to licensing reasons) don't come with prebuilt FFmpeg binaries, so you'd have to compile FFmpeg yourself. If you're up for Japanese: d.hatena.ne.jp/kamedo2/20120729/1343545890
    – slhck
    Feb 12, 2013 at 7:29

You must log in to answer this question.

Not the answer you're looking for? Browse other questions tagged .