Hi everyone,
I'm trying to convert this APE 24bit/96KHz to FLAC 16/44.1
I did the following and generated the wav file
ffmpeg -i a.ape -ar 44100 -sample_fmt s16 a.wav
But when I try to split the wav with ACDIR or CUETools they say the audio file is invalid. But the wav file is ok 16/44.1
Audio
Format : Monkey's Audio
Bit rate : 2 430 Kbps
Channel(s) : 2 channels
Sampling rate : 96.0 KHz
Bit depth : 24 bits
Compression mode : Lossless
Compression ratio : 1.897
Encoding settings : Normal
Anyone?
Try :
ffmpeg -i a.ape -ar 44100 -acodec pcm_s16le -f wav a.wav
Thanks dutch!!! worked!
hmm them bits!
https://en.wikipedia.org/wiki/Endianness (https://en.wikipedia.org/wiki/Endianness)
Why not go immediately for FLAC 16/44.1, if that's what you want?
ffmpeg -i a.ape -ar 44100 -sample_fmt s16 -acodec flac a.flac
Going from 24/96 I would use Sox. It's resampler is up there with amongst the best. It's also very good at reducing 24 bit to 16 bits (using dither).
Something like this will serve you well:
sox [infile] -b 16 [outfile.wav] rate -v -s 44100 dither -s
Going from 24/96 I would use Sox. It's resampler is up there with amongst the best. It's also very good at reducing 24 bit to 16 bits (using dither).
Something like this will serve you well:
sox [infile] -b 16 [outfile.wav] rate -v -s 44100 dither -s
Newer versions of ffmpeg actually include the libsoxr resampler, although its not the default.