libopus 1.1

libopus 1.1

TestsTested
LangLanguage CC
License BSD
ReleasedLast Release May 2015

Maintained by Jiayong Ou.


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  • By
  • Xiph.Org, Skype Limited, Octasic, Jean-Marc Valin, Timothy B. Terriberry, CSIRO, Gregory Maxwell, Mark Borgerding and Erik de Castro Lopo

== Opus audio codec ==

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

Opus can 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.

Opus, when coupled with an appropriate container format, is also suitable for non-realtime stored-file applications such as music distribution, game soundtracks, portable music players, jukeboxes, and other applications that have historically used high latency formats such as MP3, AAC, or Vorbis.

                Opus is specified by IETF RFC 6716:
                http://tools.ietf.org/html/rfc6716

The Opus format and this implementation of it are subject to the royalty- free patent and copyright licenses specified in the file COPYING.

This package implements a shared library for encoding and decoding raw Opus bitstreams. Raw Opus bitstreams should be used over RTP according to http://tools.ietf.org/html/draft-spittka-payload-rtp-opus

The package also includes a number of test tools used for testing the correct operation of the library. The bitstreams read/written by these tools should not be used for Opus file distribution: They include additional debugging data and cannot support seeking.

Opus stored in files should use the Ogg encapsulation for Opus which is described at: http://wiki.xiph.org/OggOpus

An opus-tools package is available which provides encoding and decoding of Ogg encapsulated Opus files and includes a number of useful features.

Opus-tools can be found at: https://git.xiph.org/?p=opus-tools.git or on the main Opus website: http://opus-codec.org/

== Compiling libopus ==

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

2) Compiling the source

% ./autogen.sh % ./configure % make

3) Install the codec libraries (optional)

% sudo make install

Once you have compiled the codec, there will be a opus_demo executable in the top directory.

Usage: opus_demo [-e] [options] opus_demo -d [options]

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 : audio bandwidth (from narrowband to fullband); default: sampling rate -framesize : frame size in ms; default: 20 -max_payload : maximum payload size in bytes, default: 1024 -complexity : 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 : simulate packet loss, in percent (0-100); default: 0

input and output are little-endian signed 16-bit PCM files or opus bitstreams with simple opus_demo proprietary framing.

== Testing ==

This package includes a collection of automated unit and system tests which SHOULD be run after compiling the package especially the first time it is run on a new platform.

To run the integrated tests: % make check

There is also collection of standard test vectors which are not included in this package for size reasons but can be obtained from: http://opus-codec.org/testvectors/opus_testvectors.tar.gz

To run compare the code to these test vectors:

% curl -O http://opus-codec.org/testvectors/opus_testvectors.tar.gz % tar -zxf opus_testvectors.tar.gz % ./tests/run_vectors.sh ./ opus_testvectors 48000

== Portability notes ==

This implementation uses floating-point by default but can be compiled to use only fixed-point arithmetic by setting --enable-fixed-point (if using autoconf) or by defining the FIXED_POINT macro (if building manually). The fixed point implementation has somewhat lower audio quality and is slower on platforms with fast FPUs, it is normally only used in embedded environments.

The implementation can be compiled with either a C89 or a C99 compiler. While it does not rely on any undefined behavior as defined by C89 or C99, it relies on common implementation-defined behavior for two's complement architectures:

o Right shifts of negative values are consistent with two's complement arithmetic, so that a>>b is equivalent to floor(a/(2^b)),

o For conversion to a signed integer of N bits, the value is reduced modulo 2^N to be within range of the type,

o The result of integer division of a negative value is truncated towards zero, and

o The compiler provides a 64-bit integer type (a C99 requirement which is supported by most C89 compilers).