Revise README, update AUTHORS emails.

AUTHORS

-Jean-Marc Valin (jean-marc.valin@octasic.com)
+Jean-Marc Valin (jmvalin@jmvalin.ca)
 Koen Vos (koen.vos@skype.net)
 Timothy Terriberry (tterribe@xiph.org)
 Karsten Vandborg Sorensen (karsten.vandborg.sorensen@skype.net)
 Soren Skak Jensen (ssjensen@gn.com)
-Gregory Maxwell (gmaxwell@juniper.net)
+Gregory Maxwell (greg@xiph.org)

README

+== Opus audio codec ==
+
 Opus is a codec for interactive speech and audio transmission over the Internet.
 
   Opus can handle a wide range of interactive audio applications, including
@@ -5,11 +7,16 @@ Voice over IP, videoconferencing, in-game  chat, and even remote live music
 performances. It can scale from low bit-rate narrowband speech to very high
 quality stereo music.
 
-The IETF draft covering Opus can be found at:
- http://tools.ietf.org/id/draft-ietf-codec-opus
+  Opus, when coupled with an appropriate container format, is also suitable
+for non-realtime  stored-file applications such as music distribution, game
+soundtracks, portable music players, jukeboxes, and other applications that
+have historically used high latency formats such as MP3, AAC, or Vorbis.
+
+                    Opus is specified by IETF RFC 6716:
+                    http://tools.ietf.org/html/rfc6716
 
-Opus is subject to the royalty-free patent and copyright licenses specified
-in the file COPYING.
+  The Opus format and this implementation of it are subject to the royalty-
+free patent and copyright licenses specified in the file COPYING.
 
 This package implements a shared library for encoding and decoding raw Opus
 bitstreams. Raw Opus bitstreams should be used over RTP according to
@@ -26,10 +33,14 @@ described at:
 
 An opus-tools package is available which provides encoding and decoding of
 Ogg encapsulated Opus files and includes a number of useful features.
+
 Opus-tools can be found at:
- http://git.xiph.org/?p=users/greg/opus-tools.git
+ https://git.xiph.org/?p=opus-tools.git
+or on the main Opus website:
+ http://opus-codec.org/
 
 == Compiling libopus ==
+
 To build from a distribution tarball, you only need to do the following:
 
 % ./configure
@@ -42,14 +53,14 @@ To build from the git repository, the following steps are necessary:
 % git clone git://git.opus-codec.org/opus.git
 % cd opus
 
-1) Compiling
+2) Compiling the source
 
 % ./autogen.sh
 % ./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 opus_demo executable
+in the top directory.
 
 Usage: opus_demo [-e] <application> <sampling rate (Hz)> <channels (1/2)>
          <bits per second> [options] <input> <output>
@@ -58,27 +69,32 @@ Usage: opus_demo [-e] <application> <sampling rate (Hz)> <channels (1/2)>
 
 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.
+  -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.
 
 == Testing ==
 
 This package includes a collection of automated unit and system tests
-which should be run after compiling the package especially the first
+which SHOULD be run after compiling the package especially the first
 time it is run on a new platform.
 
 To run the integrated tests:
@@ -89,6 +105,8 @@ included in this package for size reasons but can be obtained from:
 http://opus-codec.org/testvectors/opus_testvectors-draft11.tar.gz
 
 To run compare the code to these test vectors:
+
+% curl -O http://opus-codec.org/testvectors/opus_testvectors-draft11.tar.gz
 % tar -zxf opus_testvectors-draft11.tar.gz
 % ./tests/run_vectors.sh ./ opus_testvectors 48000