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authorKat Walsh <kat@wikimedia.org>2011-08-16 06:43:51 +0400
committerJean-Marc Valin <jean-marc.valin@usherbrooke.ca>2011-08-16 08:07:44 +0400
commit3269dcec2f1b48046740a150e2797651f847aa54 (patch)
tree42356c3818601cec446a5181527eb7d7ca96d24e
parent66ff26f3565820ce8acea5511a0a6ae7b1246a78 (diff)
Updated README files to reflect current output of tool and tweaked punctuation and wording. Corrected typo in test_opus ("constraint"->"constrained").
-rw-r--r--README21
-rw-r--r--README.draft29
-rw-r--r--src/test_opus.c2
3 files changed, 33 insertions, 19 deletions
diff --git a/README b/README
index a6be1406..b271f163 100644
--- a/README
+++ b/README
@@ -5,7 +5,7 @@ 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
+To build from the git repository, the following steps are necessary:
1) Clone the repository:
@@ -20,18 +20,25 @@ To build from the git repository, the following steps are necessary
Once you have compiled the codec, there will be a test_opus executable in
-the src/ directory. This can be in the following way:
+the src/ directory.
-% ./test_opus <mode (0/1/2)> <sampling rate (Hz)> <channels> <bits per second> [options] <input> <output>
+Usage: ./test_opus [-e | -d] <application (0/1)> <sampling rate (Hz)> <channels
+(1/2)> <bits per second> [options] <input> <output>
-mode: 0 for audo, 1 for voice, 2 for audio:
+mode: 0 for VoIP, 1 for audio:
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
diff --git a/README.draft b/README.draft
index cc8219e0..400180f5 100644
--- a/README.draft
+++ b/README.draft
@@ -2,11 +2,12 @@ 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
+To build from the git repository instead of using this draft, follow these
+steps:
1) Clone the repository:
@@ -19,20 +20,26 @@ steps are necessary
% ./configure
% make
-
Once you have compiled the codec, there will be a test_opus executable in
-the src/ directory. This can be in the following way:
+the src/ directory.
-% ./test_opus <mode (0/1/2)> <sampling rate (Hz)> <channels> <bits per second> [options] <input> <output>
+Usage: ./test_opus [-e | -d] <application (0/1)> <sampling rate (Hz)> <channels
+(1/2)> <bits per second> [options] <input> <output>
-mode: 0 for audo, 1 for voice, 2 for audio:
+mode: 0 for VoIP, 1 for audio:
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
diff --git a/src/test_opus.c b/src/test_opus.c
index d1476d62..dd9fb8c4 100644
--- a/src/test_opus.c
+++ b/src/test_opus.c
@@ -49,7 +49,7 @@ void print_usage( char* argv[] )
fprintf(stderr, "-e : only runs the encoder (output the bit-stream)\n" );
fprintf(stderr, "-d : only runs the decoder (reads the bit-stream as input)\n" );
fprintf(stderr, "-cbr : enable constant bitrate; default: variable bitrate\n" );
- fprintf(stderr, "-cvbr : enable constraint variable bitrate; default: unconstraint\n" );
+ fprintf(stderr, "-cvbr : enable constrained variable bitrate; default: unconstrained\n" );
fprintf(stderr, "-bandwidth <NB|MB|WB|SWB|FB> : audio bandwidth (from narrowband to fullband); default: sampling rate\n" );
fprintf(stderr, "-framesize <2.5|5|10|20|40|60> : frame size in ms; default: 20 \n" );
fprintf(stderr, "-max_payload <bytes> : maximum payload size in bytes, default: 1024\n" );