Sorted by: 1. First, install the FFmpeg.js library using npm by running the following command in your Angular 6 project directory: npm install @ffmpeg/ffmpeg. Next, import the FFmpeg library and define a function that takes a WebM blob as input and converts it to an MP3 blob using FFmpeg.js: import * as FFmpeg from '@ffmpeg/ffmpeg'; async Recommended settings to convert Opus to AAC. It doesn’t require libfdk_aac; just remove that part of the command and do this: ffmpeg -i input.opus -vbr 5 -cutoff 18000 output.m4a. Or even more simply as this: ffmpeg -i input.opus -vbr 5 output.m4a. What you are asking is just converting from .opus to something else; this is not difficult. Edit. This will convert your WAV files to MP3s. -write_id3v1 1 - This will write metadata to an ID3v1 tag at the head of the file, assuming you’ve embedded metadata into the WAV file. -id3v2_version 3 - This will write metadata to an ID3v2.3 tag at the tail of the file, assuming you’ve embedded metadata into the WAV file. I want to convert the following simple oggenc command to use ffmpeg instead: oggenc -b 8 input.wav -o out.ogg From the ffmpeg libvorbis wrapper doc, it says, The following options are supported by the libvorbis wrapper. The oggenc-equivalent of the options are listed in parentheses. b (-b) I have an audio file with 256 bit rate and sampling rate - 8000Hz. I would just like to reduce the bit rate to 128/64 kbs. I tried converting to mp3 and back to wav, ffmpeg -i input.wav 1.mp3 ffmpeg -i "1.mp3" -acodec pcm_s16le -ar 4000 out.wav but this reduced sampling rate as well. I use the following command to convert m4a format to mp3. ffmpeg -i audio.m4a -acodec libmp3lame audio.mp3 I've 32 x86 cores, however libmp3lame processes in a single thread. I know libmp3lame does not support multithreading, thus I'm open to other alternates that can be executed in ubuntu CLI. This is the most basic FFmpeg command to convert input to AAC audio using the highest quality AAC FFmpeg has. libfdk_aac is the Fraunhofer AAC encoder, and it is available when you compiled FFmpeg with support for it. ffmpeg -i input.mp3 -c:a libfdk_aac output.m4a. To change the quality, you have two options: Without further ado, the script to convert all m4a files to mp3 format. (Because ffmpeg is liberal with its input, this would work for any audio files in the directory, just make sure to use the PEqH.