KLV for edit unit 0 extends into next edit unit - OPAtom misinterpreted as OP1a? File position before avformat_find_stream_info() is 11741 Format mxf probed with size=2048 and score=100 To reproduce this behavior please use pcm_s24le_to_pcm_s16le.mxf sample-file, which was uploaded to ffmpeg public ftp and run ffmpeg with following params:
In ffmpeg 0.7.13 this is working well with the same sample-file. ffmpeg versions used: 1.0 and N-46710-g4facddd (from git). Doing this with FFmpeg can be useful if you are thinking of the automation of some kind of platform that allows users to upload music in WAV format, but instead of serving the RAW WAV files, the transferred file to listen online will be MP3 instead.I am trying to extract audio stream from mxf file and transcode it from pcm_s24le to pcm_s16le audio, but ffmpeg returns broken file instead. In our case, for a WAV file of about 49MB, the output MP3 file has a size of 4.35 MB only. Video:0kB audio:4461kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.016307% Guessed Channel Layout for Input Stream #0.0 : stereoĭuration: 00:03:10.31, bitrate: 2116 kb/s The conversion should start and an output similar to the following one will appear: ffmpeg version 4.3.1- Copyright (c) 2000-2021 the FFmpeg developersīuilt with gcc 10.2.0 (Rev5, Built by MSYS2 project) The command to convert the WAV file given the following explanation would be the following one: ffmpeg -i input-file.wav -vn -ar 44100 -ac 2 -b:a 192k output-file.mp3 So to get the highest quality setting use -b:a 320k. Here you can specify the number of bits per second, for example, -b:a 256k if you want 256 Kbit/s (25.6 KB/s) audio. If you need constant bitrate (CBR) MP3 audio, you need to use the -b:a option instead of -qscale:a. -b:a: Converts the audio bitrate to be exact 192kbit per second.So used here to make sure it is stereo (2 channels) instead of mono (1 channel). For input streams, this option only makes sense for audio grabbing devices and raw demuxers and is mapped to the corresponding demuxer options.
For output streams, it is set by default to the number of input audio channels. -ac: Set the number of audio channels.The most common values for the sampling rate is 8kHz (most common for telephone communications), 44.1kHz (most common for music CDs), and 48kHz (most common for audio tracks in movies). -ar: Set the audio sampling frequency.-vn: Disable any possible video, just to make sure that there won't be any album cover image attached.-i: the input WAV file that will be converted to MP3 using the libmp3lame encoder.The command to convert WAV to MP3 with a good relation between quality and size can be breakdown like this: WAV to MP3 using FFmpegĪs with everything in our blog, you will find the solution right away so you can immediately use it in your own projects.
In this article, I will explain to you how to easily convert a WAV file to MP3 using FFmpeg from the command line. You can easily understand it with the example of a song that in WAV format has a size of about 70MB while in MP3 format, it will have a size of up to 5MB only depending on the bitrate.įFmpeg can be used to convert a huge WAV file into a tiny MP3 file that allows the user to listen to the same song but downloading just a portion of the original size of the WAV file. WAV is used where uncompromised audio quality is required and MP3 where lightweight music files are needed. For people without enough knowledge about these formats will always ask which one is better, WAV or MP3? That's because those are the most common formats of audio nowadays that you can find everywhere.