Removes useless test in test_opus

[opus.git] / README

diff --git a/README b/README
index b957da8..b271f16 100644 (file)
--- a/README
+++ b/README
@@ -1,20 +1,45 @@
-CELT is a very low delay audio codec designed for high-quality communications.
-Its potential uses include video-conferencing and network music performance.
-The code is still in early stage, so it may be broken from time to time and
-the bit-stream is not frozen yet, so it is different from one version to 
-another. Oh, and don't complain if it sets your house on fire.
+This is a prototype codec and for now it has limited functionality. 
 
-To compile:
+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
 
-To test the encoder:
-% testcelt <rate> <channels> <frame size> <bytes per packet> input.sw output.sw
 
-where input.sw is a 16-bit (machine endian) audio file sampled at 
-44.1 kHz or 48 kHz. The output file is already decompressed. 
-the <rate> parameter is the number of bytes per packet to use.
+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
 
-Alternatively, there are now Ogg-based tools in tools/. Use 
-the --help option for details. These are quite similar to the
-speexenc/speexdec tools.
+input and output are 16-bit PCM files (machine endian)