Documentation updates.
diff --git a/README b/README
index b271f16..afcbab3 100644
--- a/README
+++ b/README
@@ -20,18 +20,20 @@
Once you have compiled the codec, there will be a test_opus executable in
-the src/ directory.
+the top directory.
-Usage: ./test_opus [-e | -d] <application (0/1)> <sampling rate (Hz)> <channels
-(1/2)> <bits per second> [options] <input> <output>
+Usage: test_opus [-e] <application> <sampling rate (Hz)> <channels (1/2)>
+ <bits per second> [options] <input> <output>
+ test_opus -d <sampling rate (Hz)> <channels (1/2)> [options]
+ <input> <output>
-mode: 0 for VoIP, 1 for audio:
+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
+-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
@@ -42,4 +44,5 @@
-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 16-bit PCM files (machine endian) or opus bitstreams
+with simple test_opus propritary framing.