com.musicg.dsp.Resampler Maven / Gradle / Ivy

Show more of this group Show more artifacts with this name
Show all versions of musicg Show documentation

Musicg is a lightweight audio analysis library, written in Java, with the purpose of extracting both high level and low level audio features.

The newest version!

/*
 * Copyright (C) 2011 Jacquet Wong
 *
 * Licensed under the Apache License, Version 2.0 (the "License");
 * you may not use this file except in compliance with the License.
 * You may obtain a copy of the License at
 *
 * http://www.apache.org/licenses/LICENSE-2.0
 *
 * Unless required by applicable law or agreed to in writing, software
 * distributed under the License is distributed on an "AS IS" BASIS,
 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
 * See the License for the specific language governing permissions and
 * limitations under the License.
 */

package com.musicg.dsp;

/**
 * Resample signal data (base on bytes)
 * 
 * @author jacquet
 *
 */
public class Resampler {

	public Resampler() {
	}

	/**
	 * Do resampling. Currently the amplitude is stored by short such that maximum bitsPerSample is 16 (bytePerSample is 2)
	 * 
	 * @param sourceData	The source data in bytes
	 * @param bitsPerSample	How many bits represents one sample (currently supports max. bitsPerSample=16) 
	 * @param sourceRate	Sample rate of the source data
	 * @param targetRate	Sample rate of the target data
	 * @return re-sampled data
	 */
	public byte[] reSample(byte[] sourceData, int bitsPerSample, int sourceRate, int targetRate) {

		// make the bytes to amplitudes first
		int bytePerSample = bitsPerSample / 8;
		int numSamples = sourceData.length / bytePerSample;
		short[] amplitudes = new short[numSamples];	// 16 bit, use a short to store
		
		int pointer = 0;
		for (int i = 0; i < numSamples; i++) {
			short amplitude = 0;
			for (int byteNumber = 0; byteNumber < bytePerSample; byteNumber++) {
				// little endian
				amplitude |= (short) ((sourceData[pointer++] & 0xFF) << (byteNumber * 8));
			}
			amplitudes[i] = amplitude;
		}
		// end make the amplitudes
		
		// do interpolation
		LinearInterpolation reSample=new LinearInterpolation();
		short[] targetSample = reSample.interpolate(sourceRate, targetRate, amplitudes);
		int targetLength = targetSample.length;
		// end do interpolation
		
		// TODO: Remove the high frequency signals with a digital filter, leaving a signal containing only half-sample-rated frequency information, but still sampled at a rate of target sample rate. Usually FIR is used
		
		// end resample the amplitudes

		// convert the amplitude to bytes
		byte[] bytes;
		if (bytePerSample==1){
			bytes= new byte[targetLength];
			for (int i=0; i> 8) & 0xff);			
			}
		}
		// end convert the amplitude to bytes
		
		return bytes;
	}
}