plectrum

Plectrum: instrument tuner for Android
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AudioEvent.java (6135B)


      1 /*
      2 *      _______                       _____   _____ _____  
      3 *     |__   __|                     |  __ \ / ____|  __ \ 
      4 *        | | __ _ _ __ ___  ___  ___| |  | | (___ | |__) |
      5 *        | |/ _` | '__/ __|/ _ \/ __| |  | |\___ \|  ___/ 
      6 *        | | (_| | |  \__ \ (_) \__ \ |__| |____) | |     
      7 *        |_|\__,_|_|  |___/\___/|___/_____/|_____/|_|     
      8 *                                                         
      9 * -------------------------------------------------------------
     10 *
     11 * TarsosDSP is developed by Joren Six at IPEM, University Ghent
     12 *  
     13 * -------------------------------------------------------------
     14 *
     15 *  Info: http://0110.be/tag/TarsosDSP
     16 *  Github: https://github.com/JorenSix/TarsosDSP
     17 *  Releases: http://0110.be/releases/TarsosDSP/
     18 *  
     19 *  TarsosDSP includes modified source code by various authors,
     20 *  for credits and info, see README.
     21 * 
     22 */
     23 
     24 
     25 package be.tarsos.dsp;
     26 
     27 import java.util.Arrays;
     28 
     29 import be.tarsos.dsp.io.TarsosDSPAudioFloatConverter;
     30 import be.tarsos.dsp.io.TarsosDSPAudioFormat;
     31 
     32 /**
     33  * An audio event flows through the processing pipeline. The object is reused for performance reasons.
     34  * The arrays with audio information are also reused, so watch out when using the buffer getter and setters. 
     35  * 
     36  * @author Joren Six
     37  */
     38 public class AudioEvent {
     39 	/**
     40 	 * The format specifies a particular arrangement of data in a sound stream. 
     41 	 */
     42 	private final TarsosDSPAudioFormat format;
     43 	
     44 	private final TarsosDSPAudioFloatConverter converter;
     45 	
     46 	/**
     47 	 * The audio data encoded in floats from -1.0 to 1.0.
     48 	 */
     49 	private float[] floatBuffer;
     50 	
     51 	/**
     52 	 * The audio data encoded in bytes according to format.
     53 	 */
     54 	private byte[] byteBuffer;
     55 	
     56 	/**
     57 	 * The overlap in samples. 
     58 	 */
     59 	private int overlap;
     60 	
     61 	/**
     62 	 * The length of the stream, expressed in sample frames rather than bytes
     63 	 */
     64 	private long frameLength;
     65 	
     66 	/**
     67 	 * The number of bytes processed before this event. It can be used to calculate the time stamp for when this event started.
     68 	 */
     69 	private long bytesProcessed;
     70 
     71 	private int bytesProcessing;
     72 	
     73 	
     74 	public AudioEvent(TarsosDSPAudioFormat format){
     75 		this.format = format;
     76 		this.converter = TarsosDSPAudioFloatConverter.getConverter(format);
     77 		this.overlap = 0;
     78 	}
     79 	
     80 	public float getSampleRate(){
     81 		return format.getSampleRate();
     82 	}
     83 	
     84 	public int getBufferSize(){
     85 		return getFloatBuffer().length;
     86 	}
     87 	
     88 	/**
     89 	 * @return  The length of the stream, expressed in sample frames rather than bytes
     90 	 */
     91 	public long getFrameLength(){
     92 		return frameLength;
     93 	}
     94 	
     95 	public int getOverlap(){
     96 		return overlap;
     97 	}
     98 	
     99 	public void setOverlap(int newOverlap){
    100 		overlap = newOverlap;
    101 	}
    102 	
    103 	public void setBytesProcessed(long bytesProcessed){
    104 		this.bytesProcessed = bytesProcessed;		
    105 	}
    106 	
    107 	/**
    108 	 * Calculates and returns the time stamp at the beginning of this audio event.
    109 	 * @return The time stamp at the beginning of the event in seconds.
    110 	 */
    111 	public double getTimeStamp(){
    112 		return bytesProcessed / format.getFrameSize() / format.getSampleRate();
    113 	}
    114 	
    115 	public double getEndTimeStamp(){
    116 		return(bytesProcessed + bytesProcessing) / format.getFrameSize() / format.getSampleRate();
    117 	}
    118 	
    119 	public long getSamplesProcessed(){
    120 		return bytesProcessed / format.getFrameSize();
    121 	}
    122 
    123 	/**
    124 	 * Calculate the progress in percentage of the total number of frames.
    125 	 * 
    126 	 * @return a percentage of processed frames or a negative number if the
    127 	 *         number of frames is not known beforehand.
    128 	 */
    129 	public double getProgress(){
    130 		return bytesProcessed / format.getFrameSize() / (double) frameLength;
    131 	}
    132 	
    133 	/**
    134 	 * Return a byte array with the audio data in bytes.
    135 	 *  A conversion is done from float, cache accordingly on the other side...
    136 	 * 
    137 	 * @return a byte array with the audio data in bytes.
    138 	 */
    139 	public byte[] getByteBuffer(){
    140 		int length = getFloatBuffer().length * format.getFrameSize();
    141 		if(byteBuffer == null || byteBuffer.length != length){
    142 			byteBuffer = new byte[length];
    143 		}
    144 		converter.toByteArray(getFloatBuffer(), byteBuffer);
    145 		return byteBuffer;
    146 	}
    147 	
    148 	public void setFloatBuffer(float[] floatBuffer) {
    149 		this.floatBuffer = floatBuffer;
    150 	}
    151 	
    152 	public float[] getFloatBuffer(){
    153 		return floatBuffer;
    154 	}
    155 	
    156 	/**
    157 	 * Calculates and returns the root mean square of the signal. Please
    158 	 * cache the result since it is calculated every time.
    159 	 * @return The <a
    160 	 *         href="http://en.wikipedia.org/wiki/Root_mean_square">RMS</a> of
    161 	 *         the signal present in the current buffer.
    162 	 */
    163 	public double getRMS() {
    164 		return calculateRMS(floatBuffer);
    165 	}
    166 	
    167 	
    168 	/**
    169 	 * Returns the dBSPL for a buffer.
    170 	 * 
    171 	 * @return The dBSPL level for the buffer.
    172 	 */
    173 	public double getdBSPL() {
    174 		return soundPressureLevel(floatBuffer);
    175 	}
    176 	
    177 	/**
    178 	 * Calculates and returns the root mean square of the signal. Please
    179 	 * cache the result since it is calculated every time.
    180 	 * @param floatBuffer The audio buffer to calculate the RMS for.
    181 	 * @return The <a
    182 	 *         href="http://en.wikipedia.org/wiki/Root_mean_square">RMS</a> of
    183 	 *         the signal present in the current buffer.
    184 	 */
    185 	public static double calculateRMS(float[] floatBuffer){
    186 		double rms = 0.0;
    187 		for (int i = 0; i < floatBuffer.length; i++) {
    188 			rms += floatBuffer[i] * floatBuffer[i];
    189 		}
    190 		rms = rms / Double.valueOf(floatBuffer.length);
    191 		rms = Math.sqrt(rms);
    192 		return rms;
    193 	}
    194 
    195 	public void clearFloatBuffer() {
    196 		Arrays.fill(floatBuffer, 0);
    197 	}
    198 
    199 		/**
    200 	 * Returns the dBSPL for a buffer.
    201 	 * 
    202 	 * @param buffer
    203 	 *            The buffer with audio information.
    204 	 * @return The dBSPL level for the buffer.
    205 	 */
    206 	private static double soundPressureLevel(final float[] buffer) {
    207 		double rms = calculateRMS(buffer);
    208 		return linearToDecibel(rms);
    209 	}
    210 	
    211 	/**
    212 	 * Converts a linear to a dB value.
    213 	 * 
    214 	 * @param value
    215 	 *            The value to convert.
    216 	 * @return The converted value.
    217 	 */
    218 	private static double linearToDecibel(final double value) {
    219 		return 20.0 * Math.log10(value);
    220 	}
    221 
    222 	public boolean isSilence(double silenceThreshold) {
    223 		return soundPressureLevel(floatBuffer) < silenceThreshold;
    224 	}
    225 
    226 	public void setBytesProcessing(int bytesProcessing) {
    227 		this.bytesProcessing = bytesProcessing;
    228 		
    229 	}
    230 	
    231 }