ref: c681bd0480a8c6a99ff30e240ec52fe129f8eff9
dir: /README/
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)