mirror of
https://github.com/musix-org/musix-oss
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143 lines
5.2 KiB
Plaintext
143 lines
5.2 KiB
Plaintext
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== Opus audio codec ==
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Opus is a codec for interactive speech and audio transmission over the Internet.
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Opus can handle a wide range of interactive audio applications, including
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Voice over IP, videoconferencing, in-game chat, and even remote live music
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performances. It can scale from low bit-rate narrowband speech to very high
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quality stereo music.
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Opus, when coupled with an appropriate container format, is also suitable
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for non-realtime stored-file applications such as music distribution, game
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soundtracks, portable music players, jukeboxes, and other applications that
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have historically used high latency formats such as MP3, AAC, or Vorbis.
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Opus is specified by IETF RFC 6716:
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https://tools.ietf.org/html/rfc6716
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The Opus format and this implementation of it are subject to the royalty-
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free patent and copyright licenses specified in the file COPYING.
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This package implements a shared library for encoding and decoding raw Opus
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bitstreams. Raw Opus bitstreams should be used over RTP according to
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https://tools.ietf.org/html/rfc7587
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The package also includes a number of test tools used for testing the
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correct operation of the library. The bitstreams read/written by these
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tools should not be used for Opus file distribution: They include
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additional debugging data and cannot support seeking.
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Opus stored in files should use the Ogg encapsulation for Opus which is
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described at:
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https://wiki.xiph.org/OggOpus
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An opus-tools package is available which provides encoding and decoding of
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Ogg encapsulated Opus files and includes a number of useful features.
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Opus-tools can be found at:
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https://git.xiph.org/?p=opus-tools.git
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or on the main Opus website:
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https://opus-codec.org/
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== Compiling libopus ==
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To build from a distribution tarball, you only need to do the following:
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% ./configure
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% make
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To build from the git repository, the following steps are necessary:
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1) Clone the repository:
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% git clone https://git.xiph.org/opus.git
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% cd opus
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2) Compiling the source
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% ./autogen.sh
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% ./configure
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% make
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3) Install the codec libraries (optional)
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% sudo make install
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Once you have compiled the codec, there will be a opus_demo executable
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in the top directory.
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Usage: opus_demo [-e] <application> <sampling rate (Hz)> <channels (1/2)>
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<bits per second> [options] <input> <output>
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opus_demo -d <sampling rate (Hz)> <channels (1/2)> [options]
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<input> <output>
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mode: voip | audio | restricted-lowdelay
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options:
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-e : only runs the encoder (output the bit-stream)
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-d : only runs the decoder (reads the bit-stream as input)
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-cbr : enable constant bitrate; default: variable bitrate
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-cvbr : enable constrained variable bitrate; default:
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unconstrained
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-bandwidth <NB|MB|WB|SWB|FB>
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: audio bandwidth (from narrowband to fullband);
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default: sampling rate
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-framesize <2.5|5|10|20|40|60>
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: frame size in ms; default: 20
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-max_payload <bytes>
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: maximum payload size in bytes, default: 1024
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-complexity <comp>
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: complexity, 0 (lowest) ... 10 (highest); default: 10
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-inbandfec : enable SILK inband FEC
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-forcemono : force mono encoding, even for stereo input
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-dtx : enable SILK DTX
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-loss <perc> : simulate packet loss, in percent (0-100); default: 0
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input and output are little-endian signed 16-bit PCM files or opus
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bitstreams with simple opus_demo proprietary framing.
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== Testing ==
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This package includes a collection of automated unit and system tests
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which SHOULD be run after compiling the package especially the first
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time it is run on a new platform.
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To run the integrated tests:
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% make check
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There is also collection of standard test vectors which are not
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included in this package for size reasons but can be obtained from:
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https://opus-codec.org/testvectors/opus_testvectors.tar.gz
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To run compare the code to these test vectors:
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% curl -O https://opus-codec.org/testvectors/opus_testvectors.tar.gz
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% tar -zxf opus_testvectors.tar.gz
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% ./tests/run_vectors.sh ./ opus_testvectors 48000
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== Portability notes ==
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This implementation uses floating-point by default but can be compiled to
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use only fixed-point arithmetic by setting --enable-fixed-point (if using
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autoconf) or by defining the FIXED_POINT macro (if building manually).
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The fixed point implementation has somewhat lower audio quality and is
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slower on platforms with fast FPUs, it is normally only used in embedded
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environments.
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The implementation can be compiled with either a C89 or a C99 compiler.
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While it does not rely on any _undefined behavior_ as defined by C89 or
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C99, it relies on common _implementation-defined behavior_ for two's
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complement architectures:
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o Right shifts of negative values are consistent with two's
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complement arithmetic, so that a>>b is equivalent to
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floor(a/(2^b)),
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o For conversion to a signed integer of N bits, the value is reduced
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modulo 2^N to be within range of the type,
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o The result of integer division of a negative value is truncated
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towards zero, and
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o The compiler provides a 64-bit integer type (a C99 requirement
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which is supported by most C89 compilers).
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