Opus is an internet audio codec suitable for music and voice, real time and distribution. RFC 6716. https://opus-codec.org/

Jean-Marc Valin ddd0528ad4 version number bump 13 年之前
celt 38fc0d59bc Fix some inconsequential file descriptor leaks in the demo tools to silence cppcheck warnings. 13 年之前
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include c81b510239 Fix a number of multistream decoder bugs; add some very basic multistream decoder tests. 13 年之前
silk a44e95abd0 SILK encoder description 13 年之前
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INSTALL 4468c75adb draft update, instructions 15 年之前
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README a5ff49ecdc Renames test_opus to opus_demo and adds the test_opus_api, test_opus_encode, test_opus_decode test programs. 13 年之前
README.draft a5ff49ecdc Renames test_opus to opus_demo and adds the test_opus_api, test_opus_encode, test_opus_decode test programs. 13 年之前
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opus.pc.in 9f71309abe Add pkg-config support files. 14 年之前
opus.sln c37499090b renames the libcelt/ directory to celt/ 14 年之前
opus_headers.txt 4923f3f80e Moves the main headers from src/ to include/ 13 年之前
opus_sources.mk d4e9340591 Adds support for multi-stream encoding/decoding 14 年之前
silk_headers.txt 1c2f5633d1 Removed all the silk_ prefixes in source file names (not symbols) 14 年之前
silk_sources.mk e1c3eac46b oops, forgot to update the source list 13 年之前

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 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 <2.5|5|10|20|40|60> : 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 16-bit PCM files (machine endian) or opus bitstreams
with simple opus_demo propritary framing.