[FFmpeg-user] transcoding mp3 to wav

Dave Rice dave at dericed.com
Fri Jan 23 15:54:04 CET 2015

> On Jan 22, 2015, at 12:52 PM, Moritz Barsnick <barsnick at gmx.net> wrote:
> On Thu, Jan 22, 2015 at 10:44:01 -0700, jd1008 wrote:
>> I tried with the params:  -ac 2  -ar 44.1k -ab 1600k   but to no avail.
> [...]
>>     Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
> I think you don't understand: A 16bit stereo WAV file with 44.1 kHz
> sampling rate inheritently has a bitrate of 1411.2kbits/s. There is
> nothing you can change about that without e.g. increasing the sample
> rate. But what for? What are you trying to achieve? And why?

Just to be picky, the WAV container can contain compressed audio data so if pcm is not needed, one could control a lossy bitrate if using a lossy codec that WAV supports. Not sure if ffmpeg supports any lossy codec in WAV but the specification does allow it.
Dave Rice

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