An M4A file is the go-to audio format for compressed audio, due to the creation and popularity of iTunes. It was developed by Apple, largely as a result of the popularity of the MP3 file. This is the most basic FFmpeg command to convert input to AAC audio using the highest quality AAC FFmpeg has. Stream #0:1: Audio: none, 0 channels, 128 kb/sĮncoder (codec none) not found for output stream #0:0Īlso while using Faac independently I get this weird error for any file. An M4A file format is also called an MPEG-4 audio file, and is an audio file saved in the MPEG-4 standard. Estimating duration from bitrate, this may be inaccurateĭuration: 00:05:50.60, start: 0.000000, bitrate: 191 kb/s The other to the M4As you wish to convert. max_analyze_duration 5000000 reached at 5015510 The first navigated to the directory where ffmpeg and the batch scripts are stored. ~]$ ffmpeg -i Kalimba.mp3 -c:a libfaac Kalimba.m4aįfmpeg version 0.11.1 Copyright (c) 2000-2012 the FFmpeg developersīuilt on 09:33:27 with gcc 4.4.6 20120305 (Red Hat 4.4.6-4)Ĭonfiguration: -enable-libfaac -enable-nonfree -disable-yasm Please provide a resolution to this problem. I have compiled ffmpeg with faac enabled. I am fully aware the legal constraints in using libfaac but this is just for my testing purpose. This is a joined command that will (you can split it up at the '&'s and run separately if you want): (1) combine the mp3s into a single mp3 called 'output.
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