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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
/** * 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 { @Override public int getPadding() { return 2; } @Override 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; } }