Here, we’ll guide you through how to convert WAV to MP3 using FFmpeg. 1. First, locate your target .wav file in File Explorer. Left-click to select the full path of your wav file in the address bar and input CMD to bring up the Command Prompt in the selected directory. 2. Type the following commands in the window: ffmpeg -i input.wav output.mp3. 5. I have WAV data that I'd like to convert to MP3 on the fly with a PHP script. The WAV file originates with the script, so it does not start out as a file. I can run something like this: exec ( "lame --cbr -b 32k in.wav out.mp3" ); But this will require that I first write in.wav to disk, read out.mp3 from disk, and then clean up when I'm .\input\ .\output\ convert.bat .\input\ contains subfolders which in their turn contain .wav and other audio-formats. Im trying to get convert.bat to convert all files in those subfolders to .mp3 in .\output\. Furthermore, I need to name to converted files to the name of the subfolder where it originated from + it's original filename. The way to "mixdown" from stereo to mono in any supported file in ffmpeg is like so: ffmpeg -i file.ext -ac 1 file_mono.ext. The "-ac 1" bit instructs ffmpeg to output just 1 audio channel, i.e. mono. By default, this operation will preserve your file format but will revert your bitrate to the ffmpeg default of 64kbs. Create a txt file and write this command in it. Now, save the file and change extension to .bat. Set the ffmpeg.exe location properly. Make folder named ffmpeg in C drive and put ffmpeg.exe in it. Put the bat file in the folder containing all the mp3 and execute the bat file. It won't keep the folder structure though. Option placement matters with ffmpeg. You're attempting to apply an output option to the input. ffmpeg [input options] input [output options] output Corrected command: fluidsynth -T raw -F - sound_font_file.sf3 input.mid | ffmpeg -y -f s32le -i - -b:a 192k output.mp3 Fore more info about MP3 encoding with ffmpeg see FFmpeg Wiki: MP3. MP3 is a lossy codec. If you want to minimize quality loss, use a high bit rate. For example ffmpeg -i audio.ogg -c:a libmp3lame -q:a 0 newfile.mp3 will use the highest quality VBR (variable bit rate) preset from the Lame MP3 encoder, which will usually give you a file with an average bit rate between 220 and 260 kb/s. If your input file has a The solution written there is to used these arguments when using ffmpeg. > I must use ffmpeg -i -f s16le tmp.dat and then mplayer -demuxer rawaudio -rawaudio rate=44100:channels=2:samplesize=2 -ao pcm tmp.dat to get "normal" wav file, many program can't read ffmpeg's wav file, they read 44 bytes of header, and others data use as sound .

convert wav to mp3 ffmpeg