Question

我正在使用android平台，从以下参考问题我知道使用AudioRecord类返回原始数据我可以过滤音频范围取决于我的需要但为此我需要算法，有人可以请帮助我out找到算法来过滤范围b / w 14,400 bph和16,200 bph。

我试过“JTransform”，但我不知道我能用JTransform实现这个吗？目前我正在使用“jfftpack”来显示效果非常好的视觉效果但是我无法使用它实现音频过滤。

帮助表示感谢提前。以下是我上面提到的代码我使用“jfftpack”库来显示你可能会在代码中找到这个库参考请不要混淆

private class RecordAudio extends AsyncTask<Void, double[], Void> {
        @Override
        protected Void doInBackground(Void... params) {
try {
    final AudioRecord audioRecord = findAudioRecord();
                    if(audioRecord == null){
                        return null;
                    }

                    final short[] buffer = new short[blockSize];
                    final double[] toTransform = new double[blockSize];

                    audioRecord.startRecording();


    while (started) {
                        final int bufferReadResult = audioRecord.read(buffer, 0, blockSize);

                        for (int i = 0; i < blockSize && i < bufferReadResult; i++) {
                            toTransform[i] = (double) buffer[i] / 32768.0; // signed 16 bit
                        }

                        transformer.ft(toTransform);
                        publishProgress(toTransform);

                    }
audioRecord.stop();
                audioRecord.release();
} catch (Throwable t) {
                Log.e("AudioRecord", "Recording Failed");
            }
            return null;

/**
         * @param toTransform
         */
        protected void onProgressUpdate(double[]... toTransform) {
            canvas.drawColor(Color.BLACK);
            for (int i = 0; i < toTransform[0].length; i++) {
                int x = i;
                int downy = (int) (100 - (toTransform[0][i] * 10));
                int upy = 100;
                canvas.drawLine(x, downy, x, upy, paint);
            }
            imageView.invalidate();
        }

Answer 1

这个过程中有很多微小的细节可能会让你陷入困境。此代码未经过测试，我不经常进行音频过滤，因此您应该非常怀疑。这是过滤音频所需的基本过程：

获取音频缓冲区
可能的音频缓冲区转换（字节到浮动）
（可选）应用窗口函数，即Hanning
参加FFT
过滤频率
采取逆FFT

我假设你有一些Android和录音的基本知识，所以将在这里介绍步骤4-6。

//it is assumed that a float array audioBuffer exists with even length = to 
//the capture size of your audio buffer

//The size of the FFT will be the size of your audioBuffer / 2
int FFT_SIZE = bufferSize / 2;
FloatFFT_1D mFFT = new FloatFFT_1D(FFT_SIZE); //this is a jTransforms type

//Take the FFT
mFFT.realForward(audioBuffer);

//The first 1/2 of audioBuffer now contains bins that represent the frequency
//of your wave, in a way.  To get the actual frequency from the bin:
//frequency_of_bin = bin_index * sample_rate / FFT_SIZE

//assuming the length of audioBuffer is even, the real and imaginary parts will be
//stored as follows
//audioBuffer[2*k] = Re[k], 0<=k<n/2
//audioBuffer[2*k+1] = Im[k], 0<k<n/2

//Define the frequencies of interest
float freqMin = 14400;
float freqMax = 16200;

//Loop through the fft bins and filter frequencies
for(int fftBin = 0; fftBin < FFT_SIZE; fftBin++){        
    //Calculate the frequency of this bin assuming a sampling rate of 44,100 Hz
    float frequency = (float)fftBin * 44100F / (float)FFT_SIZE;

    //Now filter the audio, I'm assuming you wanted to keep the
    //frequencies of interest rather than discard them.
    if(frequency  < freqMin || frequency > freqMax){
        //Calculate the index where the real and imaginary parts are stored
        int real = 2 * fftBin;
        int imaginary = 2 * fftBin + 1;

        //zero out this frequency
        audioBuffer[real] = 0;
        audioBuffer[imaginary] = 0;
    }
}

//Take the inverse FFT to convert signal from frequency to time domain
mFFT.realInverse(audioBuffer, false);

Android AudioRecord过滤频率范围

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