我正在使用Android
应用,我需要实时计算音频幅度。截至目前,我正在使用MediaPlayer
来播放曲目。有没有办法在播放时实时计算其幅度?
这是我的代码:
int counterPlayer = 0;
static double[] drawingBufferForPlayer = new double[100];
private byte[] mBytes;
private byte[] mFFTBytes;
private Visualizer mVisualizer;
public void link(final MediaPlayer player)
{
if(player == null)
{
throw new NullPointerException("Cannot link to null MediaPlayer");
}
// Create the Visualizer object and attach it to our media player.
mVisualizer = new Visualizer(player.getAudioSessionId());
mVisualizer.setCaptureSize(Visualizer.getCaptureSizeRange()[1]);
//mVisualizer.setMeasurementMode(Visualizer.MEASUREMENT_MODE_PEAK_RMS);
// Pass through Visualizer data to VisualizerView
Visualizer.OnDataCaptureListener captureListener = new Visualizer.OnDataCaptureListener()
{
@Override
public void onWaveFormDataCapture(Visualizer visualizer, byte[] bytes,
int samplingRate)
{
updateVisualizer(bytes);
}
@Override
public void onFftDataCapture(Visualizer visualizer, byte[] bytes,
int samplingRate)
{
updateVisualizerFFT(bytes);
}
};
mVisualizer.setDataCaptureListener(captureListener,
Visualizer.getMaxCaptureRate() / 2, true, true);
// Enabled Visualizer and disable when we're done with the stream
mVisualizer.setEnabled(true);
player.setOnCompletionListener(new MediaPlayer.OnCompletionListener()
{
@Override
public void onCompletion(MediaPlayer mediaPlayer)
{
mVisualizer.setEnabled(false);
}
});
}
public void updateVisualizer(byte[] bytes) {
int t = calculateRMSLevel(bytes);
Visualizer.MeasurementPeakRms measurementPeakRms = new Visualizer.MeasurementPeakRms();
int x = mVisualizer.getMeasurementPeakRms(measurementPeakRms);
mBytes = bytes;
}
/**
* Pass FFT data to the visualizer. Typically this will be obtained from the
* Android Visualizer.OnDataCaptureListener call back. See
* {@link android.media.audiofx.Visualizer.OnDataCaptureListener#onFftDataCapture }
* @param bytes
*/
public void updateVisualizerFFT(byte[] bytes) {
int t = calculateRMSLevel(bytes);
mFFTBytes = bytes;
}
public int calculateRMSLevel(byte[] audioData) {
//System.out.println("::::: audioData :::::"+audioData);
double amplitude = 0;
for (int i = 0; i < audioData.length; i++) {
amplitude += Math.abs((double) (audioData[i] / 32768.0));
}
amplitude = amplitude / audioData.length;
//Add this data to buffer for display
if (counterPlayer < 100) {
drawingBufferForPlayer[counterPlayer++] = amplitude;
} else {
for (int k = 0; k < 99; k++) {
drawingBufferForPlayer[k] = drawingBufferForPlayer[k + 1];
}
drawingBufferForPlayer[99] = amplitude;
}
updateBufferDataPlayer(drawingBufferForPlayer);
setDataForPlayer(100,100);
return (int)amplitude;
}
答案 0 :(得分:4)
您的问题在于将16位样本的不正确转换为双精度。首先,您需要将两个相邻字节转换为int,然后将转换转换为double。例如
double amplitude = 0;
for (int i = 0; i < audioData.length/2; i++) {
double y = (audioData[i*2] | audioData[i*2+1] << 8) / 32768.0
// depending on your endianness:
// double y = (audioData[i*2]<<8 | audioData[i*2+1]) / 32768.0
amplitude += Math.abs(y);
}
amplitude = amplitude / audioData.length / 2;
请注意,您的代码和我的答案都假设有一个数据通道。如果您有多个通道,则需要小心分离振幅,因为数据将交错L,R,L,R(转换为双精度后)。
答案 1 :(得分:1)
你试过那个solution吗?
// Calc amplitude for this waveform
float accumulator = 0;
for (int i = 0; i < data.bytes.length - 1; i++) {
accumulator += Math.abs(data.bytes[i]);
}
float amp = accumulator/(128 * data.bytes.length);
答案 2 :(得分:0)