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| author | Jean-Marc Valin <jean-marc.valin@octasic.com> | 2011-03-09 21:24:24 +0300 |
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| committer | Jean-Marc Valin <jean-marc.valin@octasic.com> | 2011-03-09 21:24:24 +0300 |
| commit | 080620047cb68a2adebf5b68b74ecb8ec2bfa3a7 (patch) | |
| tree | 27eaadd231652f7f8236f350964e6952dcb67982 /README | |
| parent | ebf9c03229d6160528415e2e0f872582883917ae (diff) | |
Minor draft distribution update
Diffstat (limited to 'README')
| -rw-r--r-- | README | 24 |
1 files changed, 14 insertions, 10 deletions
@@ -26,13 +26,17 @@ 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: -% ./test_opus 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) Number of compressed bytes per frame (80 here means 32 kb/s) -5) Input PCM file (16-bit, machine endian) -6) Output PCM file after encoding and decoding (16-bit machine endian) - +% ./test_opus <mode (0/1/2)> <sampling rate (Hz)> <channels> <bits per second> [options] <input> <output> + +mode: 0 for audo, 1 for voice, 2 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 +-max_payload <bytes> : maximum payload size in bytes, default: 1024 +-complexity <comp> : complexity, 0 (lowest) ... 10 (highest); default: 10 +-inbandfec : enable SILK inband FEC +-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) |