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 top directory.

Usage: test_opus [-e] <application> <sampling rate (Hz)> <channels (1/2)>
         <bits per second> [options] <input> <output>
       test_opus -d <sampling rate (Hz)> <channels (1/2)> [options]
         <input> <output>

mode: voip | audio | restricted-lowdelay
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) or opus bitstreams
with simple test_opus propritary framing.