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package com.sun.media.sound;
/**
* A resampler that uses first-order (linear) interpolation.
*
* This one doesn't perform float to int casting inside the processing loop.
*
* @author Karl Helgason
*/
public final class SoftLinearResampler2 extends SoftAbstractResampler {
public int getPadding() {
return 2;
}
public void interpolate(float[] in, float[] in_offset, float in_end,
float[] startpitch, float pitchstep, float[] out, int[] out_offset,
int out_end) {
float pitch = startpitch[0];
float ix = in_offset[0];
int ox = out_offset[0];
float ix_end = in_end;
int ox_end = out_end;
// Check if we have do anything
if (!(ix < ix_end && ox < ox_end))
return;
// 15 bit shift was choosed because
// it resulted in no drift between p_ix and ix.
int p_ix = (int) (ix * (1 << 15));
int p_ix_end = (int) (ix_end * (1 << 15));
int p_pitch = (int) (pitch * (1 << 15));
// Pitch needs to recalculated
// to ensure no drift between p_ix and ix.
pitch = p_pitch * (1f / (1 << 15));
if (pitchstep == 0f) {
// To reduce
// while (p_ix < p_ix_end && ox < ox_end)
// into
// while (ox < ox_end)
// We need to calculate new ox_end value.
int p_ix_len = p_ix_end - p_ix;
int p_mod = p_ix_len % p_pitch;
if (p_mod != 0)
p_ix_len += p_pitch - p_mod;
int ox_end2 = ox + p_ix_len / p_pitch;
if (ox_end2 < ox_end)
ox_end = ox_end2;
while (ox < ox_end) {
int iix = p_ix >> 15;
float fix = ix - iix;
float i = in[iix];
out[ox++] = i + (in[iix + 1] - i) * fix;
p_ix += p_pitch;
ix += pitch;
}
} else {
int p_pitchstep = (int) (pitchstep * (1 << 15));
pitchstep = p_pitchstep * (1f / (1 << 15));
while (p_ix < p_ix_end && ox < ox_end) {
int iix = p_ix >> 15;
float fix = ix - iix;
float i = in[iix];
out[ox++] = i + (in[iix + 1] - i) * fix;
ix += pitch;
p_ix += p_pitch;
pitch += pitchstep;
p_pitch += p_pitchstep;
}
}
in_offset[0] = ix;
out_offset[0] = ox;
startpitch[0] = pitch;
}
}