I was trying to analyse the lossy compression of MP3 audio files through their Spectrograms and the results were something like this (where the upper is 80Kbps and lower is 128Kbps).
It is obvious that the compression works by deleting higher frequencies from the audio. This is visible from the Spectrogram itself so I wanted to know
- How exactly does the compression take place behind the simple ffmpeg commands?
- Are spectrograms the best way to analyse the change in bitrates of an audio file?
- The STFT used, gives us the phase and the amplitude from which we take the magnitude and plot the spectrogram. So how different can I expect the phases of the higher and lower bitrates to be?
- Is it possible to somehow recover the lost frequencies because ffmpeg based bitrate upscaling doesn't produce accurate results on the Spectrogram
I am from a CSE background so I will be grateful if someone can answer these questions or atleast refer me to some article or paper regarding the same.