Skip to content
Snippets Groups Projects
Select Git revision
  • test
1 result

README

Blame
  • user avatar
    Felicia Lim authored and Ray Essick committed
    Bug: 63932386
    Test: - verified builds for arm*/x86*
          - checked functionality using an emulator and stagefright
    
    Change-Id: I10c4b267be1c846d8992e3c5f6d2576c2cb258a9
    Signed-off-by: default avatarFelicia Lim <flim@google.com>
    0efcc2be
    History
    README 5.72 KiB
    == 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:
                        https://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
     https://tools.ietf.org/html/rfc7587
    
    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:
     https://tools.ietf.org/html/rfc7845
    
    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:
     https://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:
    
    0) Set up a development environment:
    
    On an Ubuntu or Debian family Linux distribution:
    
        % sudo apt-get install git autoconf automake libtool gcc make
    
    On a Fedora/Redhat based Linux:
    
        % sudo dnf install git autoconf automake libtool gcc make
    
    Or for older Redhat/Centos Linux releases:
    
        % sudo yum install git autoconf automake libtool gcc make
    
    On Apple macOS, install Xcode and brew.sh, then in the Terminal enter:
    
        % brew install autoconf automake libtool
    
    1) Clone the repository:
    
        % git clone https://git.xiph.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] <application> <sampling rate (Hz)> <channels (1/2)>
             <bits per second> [options] <input> <output>
           opus_demo -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 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:
    https://opus-codec.org/docs/opus_testvectors-rfc8251.tar.gz
    
    To run compare the code to these test vectors:
    
        % curl -OL https://opus-codec.org/docs/opus_testvectors-rfc8251.tar.gz
        % tar -zxf opus_testvectors-rfc8251.tar.gz
        % ./tests/run_vectors.sh ./ opus_newvectors 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).