Removes useless test in test_opus

[opus.git] / README

diff --git a/README b/README
index ac83683..b271f16 100644 (file)
--- a/README
+++ b/README
@@ -2,35 +2,44 @@ 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 --disable-shared
+% ./configure
 % make
 
-To build from the git repository, the following steps are necessary
+To build from the git repository, the following steps are necessary:
 
-0) Clone the repository:
+1) Clone the repository:
 
-% git clone git://git.xiph.org/users/jm/ietfcodec.git
-% cd ietfcodec
+% git clone git://git.opus-codec.org/opus.git
+% cd opus
 
-1) Get the celt and silk submodules:
-
-% git submodule update --init
-
-2) Compiling
+1) Compiling
 
 % ./autogen.sh
-% ./configure --disable-shared
+% ./configure
 % make
 
 
-Once you have compiled the codec, there will be a test_hybrid executable in
-the src/ directory. This can be in the following way:
-
-% ./test_hybrid 48000 1 960 80 input.sw output.sw
-
-The arguments are:
-1) The sampling rate (only 48000 supported for now)
-2) The number of channels (only mono supported for now)
-3) The frame size in samples (480 and 960 supported)
-4) Input PCM file (16-bit, machine endian)
-5) Output PCM file after encoding and decoding (16-bit machine endian)
+Once you have compiled the codec, there will be a test_opus executable in
+the src/ directory.
+
+Usage: ./test_opus [-e | -d] <application (0/1)> <sampling rate (Hz)> <channels 
+(1/2)> <bits per second>  [options] <input> <output>
+
+mode: 0 for VoIP, 1 for audio:
+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 16-bit PCM files (machine endian)