我的应用程序。是计算噪声水平和输入声音的频率的峰值。 我用FFT得到的短裤[]缓冲区数组,这就是code: BUFFERSIZE = 1024,采样率= 44100
My app. is calculating noise level and peak of frequency of input sound. I used FFT to get array of shorts[] buffer , and this is the code : bufferSize = 1024, sampleRate = 44100
int bufferSize = AudioRecord.getMinBufferSize(sapleRate,
channelConfiguration, audioEncoding);
AudioRecord audioRecord = new AudioRecord(
MediaRecorder.AudioSource.DEFAULT, sapleRate,
channelConfiguration, audioEncoding, bufferSize);
,这是将code:
and this is converting code :
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,如果这个算法正常工作或我失去了一些东西? 我想知道这是否是正确的,我有声音分贝显示在移动,如何测试呢? 我需要什么帮助请在此先感谢:)
Now I want to know if this algorithm works correctly or i'm missing something ? And I want to know if it was correct and i have sound in dB displayed on mobile , how to test it ? I need any help please , Thanks in advance :)
在code看起来是正确的,但你或许应该处理的情况下缓冲区最初包含零,这可能会导致 Math.log10
失败,例如:改变:
The code looks correct but you should probably handle the case where the buffer initially contains zeroes, which could cause Math.log10
to fail, e.g. change:
double rmsdB = 20.0 * Math.log10(mGain * mRmsSmoothed);
到
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
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