Transcoding from AAC to mp3 uses low-quality bitrate

[dinkster]

When downloading audio from YouTube, the audio stream initially downloaded is an .m4a file encoded at 128kbps. Opening the file in VLC reveals it's a 128kbps AAC stream, which is higher quality than 128kbps mp3.

However, when converting to mp3, the software always converts the file down to 128kbps mp3, even if something like 320kbps is specified with a custom conversion preset. I believe what's happening is the bitrate is being copied from the source to reduce filesize, but this results in a pretty noticeable downgrade in audio quality since 128kbps is far below the level of transparency with MP3.

Is there a way to force it to convert to 320kbps mp3 regardless of the bitrate of the input file? I'm pretty sure this is what https://cobalt.tools/ is doing by default for YouTube audio streams, comparing the AAC downloaded by this and the mp3 downloaded by that