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Was using a ffmpeg command
ffmpeg -i "$input_file" -ar 16000 -codec:a libmp3lame -qscale:a 9 "$output_file" -y -loglevel quiet
to compress wav files, and it's giving me consistent 7x-9x compression ratio. Was told to modify it so it does it in one second chunks so i wrote this c++ code for it https://pastebin.com/Mesp8Lz0
don't have a lot of experience with this lame thing but the way I see it I've set all parameters pretty much the same in both the scenarios? But the best it gives me is 4x for any file. Is that like a hard ceiling? is there a way to replicate the performance I get from the ffmpeg command more closely? any help appreciated!
(considering they use the same library this seems trivial in my head, but I haven't been able to find anything on it, so yeah)
ffmpeg -i "$input_file" -ar 16000 -codec:a libmp3lame -qscale:a 9 "$output_file" -y -loglevel quiet
to compress wav files, and it's giving me consistent 7x-9x compression ratio. Was told to modify it so it does it in one second chunks so i wrote this c++ code for it https://pastebin.com/Mesp8Lz0
don't have a lot of experience with this lame thing but the way I see it I've set all parameters pretty much the same in both the scenarios? But the best it gives me is 4x for any file. Is that like a hard ceiling? is there a way to replicate the performance I get from the ffmpeg command more closely? any help appreciated!
(considering they use the same library this seems trivial in my head, but I haven't been able to find anything on it, so yeah)
