Opus is an internet audio codec suitable for music and voice, real time and distribution. RFC 6716. https://opus-codec.org/
Jean-Marc Valin
68e7ec9420
Adds padding code to fill up SILK frames to CBR
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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 test_opus executable in
the top directory.
Usage: test_opus [-e]
[options]
test_opus -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 test_opus propritary framing.
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]
[options]
test_opus -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 test_opus propritary framing.