Opus Codec

The Opus codec is designed for interactive speech and audio transmission over the Internet. More...

Modules
	Error codes
	Generic CTLs
	These macros are used with the celt_opus_decoder_ctl and opus_encoder_ctl calls to generate a particular request.
	Opus Decoder
	This page describes the process and functions used to decode Opus.
	Opus Encoder
	This page describes the process and functions used to encode Opus.
	Opus library information functions
	Pre-defined values for CTL interface
	Repacketizer
	The repacketizer can be used to merge multiple Opus packets into a single packet or alternatively to split Opus packets that have previously been merged.

Detailed Description

The Opus codec is designed for interactive speech and audio transmission over the Internet.

It is designed by the IETF Codec Working Group and incorporates technology from Skype's SILK codec and Xiph.Org's CELT codec.

The Opus codec is designed to handle a wide range of interactive audio applications, including Voice over IP, videoconferencing, in-game chat, and even remote live music performances. It can scale from low bit-rate narrowband speech to very high quality stereo music. Its main features are:

• Sampling rates from 8 to 48 kHz
• Bit-rates from 6 kb/s to 510 kb/s
• Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
• Audio bandwidth from narrowband to full-band
• Support for speech and music
• Support for mono and stereo
• Support for multichannel (up to 255 channels)
• Frame sizes from 2.5 ms to 60 ms
• Good loss robustness and packet loss concealment (PLC)
• Floating point and fixed-point implementation

Documentation sections:

• Opus Encoder
• Opus Decoder
• Repacketizer
• opus_multistream
• Opus library information functions
• opus_custom