if ab=64 =>64 k bits per second and samples per second=8 k samples then bits per sample=64/8=8 bits per sample but the problem is that the output file always has samples of 16 bits per sample. I know that sample can contain bits from 8 , 16, 24 to 32 bits per sample. i want to get 8 bits per sample file how can this be done ? If compressing using MP3 or another audio-specific algorithm isn't acceptable, you could reduce the sampling rate or other parameters in your WAV recording. If you are recording at 2-channel (stereo), 44KHz, 16-bit, for instance, the file would be larger than if you recorded at, say, 1-channel (mono), 22KHz, 16-bit. You can also convert other audio files to WAV and vice versa. To create a Wav file, simply open the audio in any program that supports the format and save. Recording tools and apps also render in WAV. Virtually all players that support MP3s also support WAV, so opening a WAV audio file should be effortless. Windows and Mac users can open WAV If your phone requires files in other than WAV or MP3 format the best course after editing the file is to export it as a mono, 16-bit 44,100 Hz PCM WAV file, then convert that WAV to the required format with an appropriate conversion application of your choice. Below are softwares and online converter you can use to convert to 16 bit Mono 8k PCM waf format. Option 1: g711.org. This free online tool will convert just about any DRM-free media file into audio that's compatible with most telephony vendors' Music on Hold and IVR Announcements. 2 Answers. for followers, ffmpeg -i lame1.mp3 -acodec pcm_s16le yo.wav converts it to wav with the WAV headers. For those stuck on Unable to find a suitable output format for 'output.raw', note that the order of arguments is significant for FFmpeg, and hence you must keep the -i argument here as the first argument. When using ffmpeg -i input.wav -ar 48000 -ac 2 -acodec pcm_s32le output.wav the Audio Format PCM = 0xfffe. SDL2 (as seen in the parent question) only allows files to play with Linear PCM Audio Format (1), and I am unsure how using sox or ffmpeg how to convert my 24bit WAV files upwards to 32bit (as SDL2 only plays 32bit and 16bit). How to convert any mp3 file to .wav 16khz mono 16bit. Please, help to choose solution for converting any mp3 file to special .wav - I'm a newbie with Linux command line tools, so It's hard for me right now. I need to get wav with 16khz mono 16bit sound properties from any mp3 file. I was trying. Bounce and dither your source from 24-bit to 16-bit to start, if it's not already there. I'd keep it WAV format still. Then, it depends on your DAW. I can bounce out from Logic to a 64 stereo/32k mono and if I set the Stereo out to Mono I get that 32kbps file. Audacity can *export* to fixed rate mono that low, and it's free. dNUwDmj.

convert mp3 to wav mono 16 bit