Better training

[opus.git] / README.draft

diff --git a/README.draft b/README.draft
index 9e1b53e..061af23 100644 (file)
--- a/README.draft
+++ b/README.draft
@@ -2,42 +2,53 @@ To build this source code, simply type:
 
 % make
 
-If this does not work, or to change the default configuration (e.g. compile for
-a fixed-point architecture), simply edit the options in the Makefile
+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.
 
-To build from the git repository instead of using this draft, the following 
-steps are necessary
+An up-to-date implementation conforming to this standard is available in a
+Git repository at git://git.xiph.org/opus.git or on a website at:
+http://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:
+1) Clone the repository (latest implementation of this standard at the time 
+of publication)
 
-% git clone git://git.xiph.org/users/jm/ietfcodec.git
-% cd ietfcodec
+% git clone git://git.opus-codec.org/opus.git
+% cd opus
 
-2) Get the celt and silk submodules:
-
-% git submodule update --init
-
-3) Compile
+2) Compile
 
 % ./autogen.sh
-% ./configure --disable-shared
+% ./configure
 % make
 
+Once you have compiled the codec, there will be a opus_demo executable in
+the top directory.
 
-Once you have compiled the codec, there will be a test_opus executable in
-the src/ directory. This can be in the following way:
-
-% ./test_opus <mode (0/1/2)> <sampling rate (Hz)> <channels> <bits per second>  [options] <input> <output>
+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: 0 for audo, 1 for voice, 2 for audio:
+mode: voip | audio | restricted-lowdelay
 options:
--cbr                 : enable constant bitrate; default: VBR
--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 
+-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 16-bit PCM files (machine endian)
+input and output are little endian signed 16-bit PCM files or opus bitstreams
+with simple opus_demo propritary framing.