我的应用。正在计算输入声音的噪音水平和频率峰值。 我用FFT来获取short []缓冲区数组,这就是代码: bufferSize = 1024,sampleRate = 44100
int bufferSize = AudioRecord.getMinBufferSize(sapleRate,
channelConfiguration, audioEncoding);
AudioRecord audioRecord = new AudioRecord(
MediaRecorder.AudioSource.DEFAULT, sapleRate,
channelConfiguration, audioEncoding, bufferSize);
这是转换代码:
short[] buffer = new short[blockSize];
try {
audioRecord.startRecording();
} catch (IllegalStateException e) {
Log.e("Recording failed", e.toString());
}
while (started) {
int bufferReadResult = audioRecord.read(buffer, 0, blockSize);
/*
* Noise level meter begins here
*/
// Compute the RMS value. (Note that this does not remove DC).
double rms = 0;
for (int i = 0; i < buffer.length; i++) {
rms += buffer[i] * buffer[i];
}
rms = Math.sqrt(rms / buffer.length);
mAlpha = 0.9; mGain = 0.0044;
/*Compute a smoothed version for less flickering of the
// display.*/
mRmsSmoothed = mRmsSmoothed * mAlpha + (1 - mAlpha) * rms;
double rmsdB = 20.0 * Math.log10(mGain * mRmsSmoothed);
现在我想知道这个算法是否正常工作还是我错过了什么? 我想知道它是否正确,我在手机上显示的声音是dB,如何测试? 我需要任何帮助,提前致谢:)
答案 0 :(得分:1)
代码看起来正确但你应该处理缓冲区最初包含零的情况,这可能导致Math.log10
失败,例如改变:
double rmsdB = 20.0 * Math.log10(mGain * mRmsSmoothed);
为:
double rmsdB = mGain * mRmsSmoothed >.0 0 ?
20.0 * Math.log10(mGain * mRmsSmoothed) :
-999.99; // choose some appropriate large negative value here for case where you have no input signal