Unlike MP3, Wav is a lossless format, so sound quality should not have been lost in decoding MP3 to Wav. (Not sure why you'd want that anyway in many cases, as that will greatly increase the file size and not increase the quality.) On the other hand information will be lost if you encode it back to MP3 from Wav, but you can of course try it Set the input file.format with -i followed by the output file.format: ffmpeg -i input.wav output.ogg output.mp3 output.flac Batch conversion: As a simple one liner with putting each format in a separate folder: You wanna demux the video.. This seperates the audio, subtitles and video from eachtoher.. With ffmpeg the command would be : ffmpeg -i file.avi -acodec copy -vn file.ac3-> Assuming AC3 is the audio codec ofc course!~ 2. If you have an MP3 input file: ffmpeg -i input.mp3 -c:a g723_1 -ar 8000 -ac 1 -b:a 6.3K output.wav. You need to specify the sample rate of 8 kHz, mono audio, and a bitrate of 6.3 kB/s. In the other direction, from any audio file to AAC audio – which is the most common for MP4 containers – it's a little easier: sudo apt-get install ffmpeg lame flac vorbis-tools On Mac OS X: brew install ffmpeg lame Examples. Example 1 - converting from MP3 to OGG: ftransc -f ogg filename.mp3 The output file name for the above example will be 'filename.ogg' Example 2 - converting from MP3 to AAC, removing original file on success, using high quality preset:
Code Revisions 1 Stars 39 Forks 1. Embed. Download ZIP. Convert mp4 to WAV with ffmpeg. Raw. mp4-to-wav. ffmpeg -i -ac 2 -f wav .
Another option is the WAV format, embedding a PCM stream. A pcm stream just consists in the uncompressed audio samples. When creating it you have to specify the sample format (16 or 24 bits integer, or 32 bits float). In ffmpeg for instance this is: -c:a pcm_s24le (signed 24 bits integer, little endian). FLAC can not handle 32 bits floats (but
1 Answer. First you should use Libswresample to resample audio data. Then you can save audio raw data with wav format. Thank you for your answer, but I want to do it by ffmpeg library. @MohammadrezaKashi Libswresample is a part of ffmpeg library.

On my server therer is ffmpeg 4.2.4. When I run apt install ffmpeg it says ffmpeg is already the newest version (7:4.2.4-1ubuntu0.1).. How to upgrade to version 4.4 on the command line? – The ffmpeg download page states "7:4.4.1-2ubuntu1" as the latest. –

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    • convert mp3 to wav ffmpeg