Ffmpeg to convert from flac to wav

ffmpegflacwav

I need to convert a flac file to a wav file without changing sample rate and bit depth. As far as I know changing these properties may distort the audio, so how do i specify them not to be changed?

Also, is there any way to prevent metadata to be written to the output file?

Edit:
Apparently this is an XY Problem, I'm sorry, I'm new here.
My problem is that I don't want to install flac on my OS X, because I'm trying to sandbox everything I use, so I need a single executable file, such as ffmpeg. I'll try @slhck's suggestion and check whether sample rate and bit depth change.

Edit:
ffmpeg only preserves sample rate. Bit depth needs to be set manually.

Best Answer

ffmpeg will not change sample rate unless you tell it to (or the output codec does not support it, but then it will most probably fail). So this should be enough:

ffmpeg -i input.flac output.wav

ffmepg will however not preserve bit depth and default to 16-bit encoding, so if your input is 24 bit, you have to use:

ffmpeg -i input.flac -c:a pcm_s24le output.wav

As for removing metadata, see Strip metadata from all formats with FFmpeg -- you basically only add the -map_metadata -1 option.

libfdk_aac

libfdk_aac is still the best AAC-LC encoder supported by FFmpeg, and is the only HE-AAC encoder supported by FFmpeg. However, its license is problematic when it comes to distribution, so you'll need to compile ffmpeg to be able to use it.

Native FFmpeg AAC encoder

This encoder has seen many improvements since the answer you refer to. It is no longer considered experimental, but it is not yet as good as libfdk_aac. However, if you give it enough bits it should be acceptable. The advantage of this encoder is that it is always available and comes with FFmpeg.

Pipe

Not always practical, but you can pipe to your favorite, standalone encoder:

ffmpeg -i input -f wav - | fdkaac -I -m 5 - -o output.m4a

Also see

FFmpeg Wiki: AAC

Convert WAV to FLAC in FFmpeg

FFmpeg's FLAC encoder supports sample bit depths of 16 and 24 bits, the latter padded to 32-bit. So for 24-bit, you will have to use a filter in-between.

ffmpeg -i in.wav -af aformat=s32:176000 out.flac

The above encodes to a 176 kHz 24-bit sample, stored as 32-bits. And the command below encodes to 16-bit and 44.1 kHz.

ffmpeg -i in.wav -af aformat=s16:44100 out.flac

Best Answer

Related Solutions

Ffmpeg FLAC to AAC encoding

libfdk_aac

Native FFmpeg AAC encoder

Pipe

Also see

Convert WAV to FLAC in FFmpeg

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