To export to 44100 Hz 16-bit PCM WAV. If required, convert the stereo track to mono. Select menu item File > Export Audio. In the "Export Audio" dialog choose WAV (Microsoft) signed 16-bit PCM in the Format dropdown (default setting). Make sure the Sample Rate is set to 44100 Hz. Click Export. ffmpeg -i video.mkv audio.mp3. For downsampling to 16KHz, converting stereo (2 channels) to mono (1 channel) and converting MP3 to WAV (uncompressed audio samples), one needs to use the -ar (audio In newer Ubuntu releases the Opus codec is included in the libavcodec libraries that will be installed with ffmpeg. Audio encoding is then done with. ffmpeg -i infile.ext outfile.opus. The audio converter shipped with the opus-tools can convert audio in raw, wave or AIFF format. The minimal syntax uses default settings: Mac: From the menu bar at the top of the screen, click on iTunes > Preferences. PC: From the menu bar at the top of the iTunes window, click on Edit > Preferences. In the General Preferences tab, click on Import Settings, located towards the bottom. Click on the menu next to Import Using > WAV Encoder. Then click to change Setting > Custom and Convert to WAV. Using Zamzar it is possible to convert to WAV from a variety of other formats. 264 to wav (H.264 Raw Files) 3g2 to wav (3GPP2 Multimedia File) 3ga to wav (3GA Multimedia File) 3gp to wav (3GPP Multimedia File) 3gpp to wav (3GPP Multimedia File) aac to wav (Advanced Audio Coding File) Convert Audio Formats. 20210607. A simple way to convert between audio formats is to use ffmpeg. Whilst this is a video processing tool it also handles audio, of course, since most video includes audio. The same commands are also useful to extract just the audio track from a video. To convert a mp3 file to wav: ffmpeg -i myaudio.mp3 myaudio.wav. Converting from stereo to mono will mean re-encoding, so keeping the same bit rate would be meaningless. In fact, converting a 128 kbit/s MP3 -> a new 128 kbit/s MP3 will net you godawfully terrible quality, even if the second one is mono (and therefore requires a lower bit rate for the same subjective quality). I am trying to convert .mp3 file to .wav format. when I am using below command in "command prompt" I am able to get desired output. it gives me output file in .wav format sox "C:\Users\Desktop\Audio 8-bit mono 16-bit mono 24-bit mono 32-bit mono (or something completely different). You will need to make your code look at the file's header bytes, in order to determine what kind of format it is. If, for example, the file is a 16-bit stereo, and you find the data in the file, you should be able to read them and directly feed them to the I2S Before we come to the transcription part, we have to first bring our data in the right format. Podcasts or other (long) audio files are usually in mp3 format. However, this is not the format the packages or toolkits can work with. To be here more specific, we need to convert our (mp3) audio in: Wave format (.wav) Mono; 16,000Hz sample rate gh3Hd.