Create local copy of Opus v1.1.2

webrtc/modules/audio_coding/codecs/opus/opus/src/README.draft

Index: webrtc/modules/audio_coding/codecs/opus/opus/src/README.draft
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+To build this source code, simply type:
+
+% make
+
+If this does not work, or if you want to change the default configuration
+(e.g., to compile for a fixed-point architecture), simply edit the options
+in the Makefile.
+
+An up-to-date implementation conforming to this standard is available in a
+Git repository at https://git.xiph.org/opus.git or on a website at:
+https://opus-codec.org/
+However, although that implementation is expected to remain conformant
+with the standard, it is the code in this RFC that shall remain normative.
+To build from the git repository instead of using this RFC, follow these
+steps:
+
+1) Clone the repository (latest implementation of this standard at the time
+of publication)
+
+% git clone https://git.xiph.org/opus.git
+% cd opus
+
+2) Compile
+
+% ./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] <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.