I have a library which
- Loads audio files
- Does filtering on the PCM data
- Saves the new PCM back to files
it does that using:
- Using its own implementation of load/save (raw files and uncompressed WAVE files)
- Using the LAME library (MP3)
- Using the ffmpeg library (all other formats)
Now I want to test that the ffmpeg implementation works as expected. I am thinking of:
Creating some audio samples in memory
Save them to an uncompressed WAVE file
Convert them to another format that the ffmpeg library supports using a command line such as:
ffmpeg -i my-file.wav my-file.ogg
Load the
my-file.ogg
with my libraryCompare to the audio samples I created in memory (first step above)
I'm not testing whether the ffmpeg library and tools work. I'm testing whether my software properly uses the ffmpeg library. So I would imagine that trying to load that my-file.ogg
even though it was created by the ffmpeg
command line tool, is a valid test as far as my software is concerned.
Am I correct or is that process flawed?