File formats + Details + Wrong encoding
ErfolgreichCharismatisch opened this issue · 1 comments
I have a m4a file that I want to normalize. DAN cannot work with m4a as it doesn't know what it is.
- Which format I can use codec copy with ffmpeg with works with DAN?
- Also, what does a WAV-File have to look like to be an input for DAN - Frequency, endian, resolution...?
- Why are mp3-files huge that DAN outputs and how to minimize them?
DynamicAudioNormalizer CLI supports different input/output modules. These are:
- libsndfile
- libmpg123 – input only
- libopusfile – input only
So, as input, you can open all the various file types supported by libsndfile – which includes Wave Audio files as well as FLAC, Vorbis, AU and AIFF – plus MP3 files (will be decoded via libmpg123) plus Opus files (will be decoded via libopusfile). As far as uncompressed (PCM) Wave Audio files are concerned, pretty much all bit-depths, endianesses and sampling rates should be supported, as libsndfile handles that transparently for us. Wave Audio files containing "a-law", "µ-law" or "ADPCM" should work as well, again via libsndfile.
Supported output formats include Wave Audio, FLAC and Ogg/Vorbis. We do not currently support MP3 output, as we do not currently include any kind of MP3 encoder library (e.g. libmp3lame). So you probably created a PCM/Wave file and gave it a .mp3
extension. This doesn't make it an MP3 file tough 😏
Regards.