/* * _______ _____ _____ _____ * |__ __| | __ \ / ____| __ \ * | | __ _ _ __ ___ ___ ___| | | | (___ | |__) | * | |/ _` | '__/ __|/ _ \/ __| | | |\___ \| ___/ * | | (_| | | \__ \ (_) \__ \ |__| |____) | | * |_|\__,_|_| |___/\___/|___/_____/|_____/|_| * * ------------------------------------------------------------- * * TarsosDSP is developed by Joren Six at IPEM, University Ghent * * ------------------------------------------------------------- * * Info: http://0110.be/tag/TarsosDSP * Github: https://github.com/JorenSix/TarsosDSP * Releases: http://0110.be/releases/TarsosDSP/ * * TarsosDSP includes modified source code by various authors, * for credits and info, see README. * */ package be.tarsos.dsp; public class MultichannelToMono implements AudioProcessor{ private int channels; private boolean mean; public MultichannelToMono(int numberOfChannels,boolean meanOfchannels){ channels = numberOfChannels; mean = meanOfchannels; } @Override public boolean process(AudioEvent audioEvent) { float[] buffer = audioEvent.getFloatBuffer(); float[] newBuffer = new float[buffer.length/channels]; if(mean){ if(channels==2){ for(int i = 0 ; i < buffer.length ; i = i + channels){ newBuffer[i/channels]=(buffer[i]+buffer[i+1])/2.0f; } }else{ for(int i = 0 ; i < buffer.length ; i = i + channels){ double sum = 0; for(int j = 0; j < channels;j++){ sum = sum + buffer[i+j]; } newBuffer[i/channels]=(float) (sum/channels); } } }else{ for(int i = 0 ; i < buffer.length ; i = i + channels){ newBuffer[i/channels]=buffer[i]; } } audioEvent.setFloatBuffer(newBuffer); return true; } @Override public void processingFinished() { } }