* from 32, 44.1 or 48 kHz to 8 kHz, if ULAW is defined. Frequencies above 4 kHz\r
* are removed by ignoring higher subbands.\r
*/\r
+@LATTICE("OUT<V,V<SAMPLE,SAMPLE<EQ,EQ<IDX")\r
final class SynthesisFilter {\r
- private int vcount = 0;\r
+\r
+ @LOC("IDX")\r
private int vidx = 1;\r
+ @LOC("V")\r
private float[] v1;\r
+ @LOC("V")\r
private float[] v2;\r
// private float[] actual_v; // v1 or v2\r
+ @LOC("IDX")\r
private int actual_write_pos; // 0-15\r
+ @LOC("SAMPLE")\r
private float[] samples; // 32 new subband samples\r
+ @LOC("V")\r
private int channel;\r
+ @LOC("V")\r
private float scalefactor;\r
+ @LOC("EQ")\r
private float[] eq;\r
\r
/**\r
* Compute PCM Samples.\r
*/\r
\r
- private float[] _tmpOut = new float[32];\r
+ @LOC("OUT") private float[] _tmpOut = new float[32];\r
\r
private void compute_pcm_samples0() {\r
\r
// if (buffer != null) {\r
// buffer.appendSamples(channel, _tmpOut);\r
// }\r
- SampleBufferWrapper.getOutput().appendSamples(channel, _tmpOut);\r
+ SampleBufferWrapper.appendSamples(channel, _tmpOut);\r
\r
/*\r
* // MDM: I was considering putting in quality control for // low-spec\r
* d[] split into subarrays of length 16. This provides for more faster access\r
* by allowing a block of 16 to be addressed with constant offset.\r
**/\r
+ @LOC("V")\r
private static float d16[][] = null;\r
\r
/**\r