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[ 原始碼: opus  ]

套件:libopus0(1.2~alpha2-1)

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Opus codec runtime library

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.

It is intended to suit 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. The current features are:

 Bit-rates from 6 kb/s 510 kb/s
 Sampling rates from 8 to 48 kHz
 Frame sizes from 2.5 ms to 60 ms
 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 up to 255 channels (multistream frames)
 Dynamically adjustable bitrate, audio bandwidth, and frame size
 Good loss robustness and packet loss concealment (PLC)
 Floating point and fixed-point implementation

This package provides the Opus runtime library.

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armhf 146。0 kB233。0 kB [檔案列表]