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- This is a prototype codec and for now it has limited functionality.
- To build from a distribution tarball, you only need to do the following:
- % ./configure
- % make
- To build from the git repository, the following steps are necessary:
- 1) Clone the repository:
- % git clone git://git.opus-codec.org/opus.git
- % cd opus
- 1) Compiling
- % ./autogen.sh
- % ./configure
- % make
- Once you have compiled the codec, there will be a test_opus executable in
- the src/ directory.
- Usage: ./test_opus [-e | -d] <application (0/1)> <sampling rate (Hz)> <channels
- (1/2)> <bits per second> [options] <input> <output>
- mode: 0 for VoIP, 1 for audio:
- options:
- -e : only runs the encoder (output the bit-stream)
- -d : only runs the decoder (reads the bit-stream as input)
- -cbr : enable constant bitrate; default: variable bitrate
- -cvbr : enable constrained variable bitrate;
- default: unconstrained
- -bandwidth <NB|MB|WB|SWB|FB> : audio bandwidth (from narrowband to fullband);
- default: sampling rate
- -framesize <2.5|5|10|20|40|60> : frame size in ms; default: 20
- -max_payload <bytes> : maximum payload size in bytes, default: 1024
- -complexity <comp> : complexity, 0 (lowest) ... 10 (highest); default: 10
- -inbandfec : enable SILK inband FEC
- -forcemono : force mono encoding, even for stereo input
- -dtx : enable SILK DTX
- -loss <perc> : simulate packet loss, in percent (0-100); default: 0
- input and output are 16-bit PCM files (machine endian)
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