I recorded and mixed down a CD worth of homemade music with Ardour and stupidly exported all the songs in 48 kHz and 24 bit. Now I need it in 44,1 and 16 bit in order to have CDBaby take it and give it over to iTunes, Spotify and whatnot. I expected they'd want mp3 for that but no. I guess I'll be doing something like:

ffmpeg -i song.wav 

and set the new sampling rate with:

-ar 44100 

But how do I get the bitrate down to 16? I haven't been able to find any hints ... most of ffmpeg questions are about video and I get lost following first one possible thread and then another one.

3 Answers

Use the default

Default for WAV output is a 16-bit encoder (pcm_s16le), so all you need to do is:

ffmpeg -i input.wav -ar 44100 output.wav 

Or manually declare a 16-bit encoder

ffmpeg -i input.wav -c:a pcm_s16le -ar 44100 output.wav 

• See a list of encoders with ffmpeg -encoders
• See what audio sample formats (bit depth) an encoder supports with ffmpeg -h encoder=pcm_s16le

Or manually set the audio sample format

With the -sample_fmt option.

ffmpeg -i input.wav -sample_fmt s16 -ar 44100 output.wav 

• See a list of audio sample formats (bit depth) with ffmpeg -sample_fmts

Or use the aformat filter

ffmpeg -i input.wav -af "aformat=sample_fmts=s16:sample_rates=44100" output.wav 

Can this be related to Sample formats?

To see options: ffmpeg -sample_fmts

For you it will be something like,

ffmpeg -i input -sample_fmt s16 -ar 44000 output

Ref:

I suspect that SoX might be a better tool for this job. I created a sample file with the sampling rate of 48.0 kHz and Bit depth of 24 bits, I have arrowed in the relevant sections:

andrew@ilium~/tmp$ mediainfo luckynight_48_24.wav General Complete name : luckynight_48_24.wav Format : Wave File size : 16.6 MiB Duration : 1 min 0 s Overall bit rate mode : Constant Overall bit rate : 2 304 kb/s Audio Format : PCM Format settings : Little / Signed Codec ID : 00000001-0000-0010-8000-00AA00389B71 Duration : 1 min 0 s Bit rate mode : Constant Bit rate : 2 304 kb/s Channel(s) : 2 channels Channel layout : L R Sampling rate : 48.0 kHz <----- Bit depth : 24 bits <----- Stream size : 16.6 MiB (100%) 

Now I am no SoX master but the following command certainly converted the above file to a sampling rate of 44.1 kHz and Bit depth of 16 bits (as you have requested):

sox luckynight_48_24.wav -r 44100 -b 16 luckynight_44_16.wav 

This accomplished the following, and again I have arrowed in the relevant sections:

andrew@ilium~/tmp$ mediainfo luckynight_44_16.wav General Complete name : luckynight_44_16.wav Format : Wave File size : 10.2 MiB Duration : 1 min 0 s Overall bit rate mode : Constant Overall bit rate : 1 411 kb/s Audio Format : PCM Format settings : Little / Signed Codec ID : 1 Duration : 1 min 0 s Bit rate mode : Constant Bit rate : 1 411.2 kb/s Channel(s) : 2 channels Sampling rate : 44.1 kHz <----- Bit depth : 16 bits <----- Stream size : 10.2 MiB (100%) 

And this is exactly what you are after :)