Convert WAV to OPUS

WAV is one of the most common audio formats and stands for Waveform Audio File Format. It is intended for operation with a digitalized audio stream. However, it has also been used for non-audio data due to its sampling rate (1 Hz to 4.3 GHz). It contains musical compositions, voice recordings, and various audio effects. Developed by programmers from Microsoft and IBM, it is a crucial audio format for placing uncompressed sound files on Windows PCs. The wav wave files can be played back in QuickTime, Windows Media Player, Wav Player, and some other programs. In addition to that, they can be processed in audio editing apps.

When a file is compressed into WAV, the data are not supposed to be lost, and the quality is excellent. However, the format did not have a considerable market share due to its larger size than MP3. It must have enough time and disc space to upload and send such files via the Internet. One of WAV's major advantages is linked to Linear Pulse Code Modulation (LPCM) to store an audio stream. Consequently, a copy is just as good as an original, highly appraised by experts in music and professional users.

Sound files with this extension are recorded into 8 or 16 bit per sample. A standard option for CD Audio is an audio stream of 16 bit per sample and a sampling frequency of 44.1 KHz. One second of sound corresponds to 88 Kb of internal memory. WAV files can store metadata in the INFO chunk, and they also include integrated IFF lists.

In some cases, the standard format may be used for broadcasting. For instance, BBC stereo data of 44 100 Hz and 16 bit are generally accepted within the VCS system.

The IETF community has created the format for online broadcasting and is being used for lossy compression of audio data. It provides excellent quality of sound and a perfect compression level due to dynamically adjusted streaming technology. Being an essential audio codec of Skype, OPUS is based on the Skype message-transfer algorithms. As a container, they use .ogg extension.

The format uses CELT and SILK codecs, which allow easy change of bitrate within the range of 6-510 Kbit/s. That is why it has gained prevalence in video chats and video conferences, as well as IP translations.