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Jean-Marc Valin's avatar
Jean-Marc Valin authored
- Merged the LPC stabilization from NLSF2A_stable.c into NLSF2A.c
- The bandwidth expansion in NLSF2A() now operates on int32 LPC coefficients in
Q17 domain (instead of int16 Q12 coefficients)
- The function bwexpander_32() has a more precise way of updating the chirp
variable (round to nearest, instead of round down)
- Changed a few variables in NLSF_stabilize() from int16 to int32 to avoid signed
wrap-around (no difference in results as the wrap-around would always be reversed
later)
- The LSF codebook for WB speech has a quantization stepsize of 0.15 (was 0.16).
This doesn't break the bitstream, although it slightly limits quality of signals
encoded with the old version and decoded with the new one (I can't really hear it
and PESQ gives high scores as well).  I does improve handling of tonal signals.
- As discussed: the Q-domain of the poly function is now in Q16 (was Q20)
- As discussed: limiting the LSFs in NLSF_decode() to 0...32767
- The silk_NLSF_DELTA_MIN values were lowered to deal with a possible future situation with less or no input HP filtering.
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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
% 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


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 <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)