我想通过修改字节来为音频文件（.wav）添加一个过滤器，但我遇到的问题是声音有噪音。
public class MainActivity extends ActionBarActivity {
private int bufferSize = 0;
private static final int RECORDER_BPP = 16;
private static final String AUDIO_RECORDER_FILE_EXT_WAV = ".wav";
private static final String AUDIO_RECORDER_FOLDER = "ks";
private static final int RECORDER_SAMPLERATE = 44100;
private static final int RECORDER_CHANNELS = AudioFormat.CHANNEL_IN_STEREO;
private static final int RECORDER_AUDIO_ENCODING = AudioFormat.ENCODING_PCM_16BIT;
@Override
protected void onCreate(Bundle savedInstanceState) {
super.onCreate(savedInstanceState);
setContentView(R.layout.activity_main);
bufferSize = AudioRecord.getMinBufferSize(RECORDER_SAMPLERATE,
RECORDER_CHANNELS, RECORDER_AUDIO_ENCODING);
String file1 = "/storage/emulated/0/File/rec.wav";
String file2 = "/storage/emulated/0/File/rec2.wav";
copyWaveFile(file1,file2);
}
static final float ALPHA = 0.15f;
private float[] lowPass(float[] input, float[] output) {
if (output == null)
return input;
for (int n = 0; n < input.length; n++) {
output[n] = output[n] + ALPHA * (input[n] - output[n]);
}
return output;
}
private static float[] toFloatArray(byte[] buffer) {
ByteArrayInputStream bas = new ByteArrayInputStream(buffer);
DataInputStream ds = new DataInputStream(bas);
float[] fArr = new float[buffer.length / 4]; // 4 bytes per float
for (int i = 0; i < fArr.length; i++) {
try {
fArr[i] = ds.readFloat();
} catch (IOException e) {
Log.e("E", "eRRO :" + e.getMessage());
e.printStackTrace();
}
}
return fArr;
}
private static byte[] toByteArray(float[] fArr) {
ByteArrayOutputStream bas = new ByteArrayOutputStream();
DataOutputStream ds = new DataOutputStream(bas);
for (float f : fArr)
try {
ds.writeFloat(f);
} catch (IOException e) {
Log.i("", "ERROR:" + e.getMessage());
e.printStackTrace();
}
byte[] bytes = bas.toByteArray();
return bytes;
}
private void copyWaveFile(String file1, String file2) {
FileInputStream in1 = null;
FileOutputStream out = null;
long totalAudioLen = 0;
long totalDataLen = totalAudioLen + 36;
long longSampleRate = RECORDER_SAMPLERATE;
int channels = 2;
long byteRate = RECORDER_BPP * RECORDER_SAMPLERATE * channels / 8;
byte[] data = new byte[bufferSize];
byte[] dataOut = new byte[bufferSize];
float[] floatArray = null;
try {
in1 = new FileInputStream(file1);
out = new FileOutputStream(file2);
totalAudioLen = in1.getChannel().size();
totalDataLen = totalAudioLen + 36;
WriteWaveFileHeader(out, totalAudioLen, totalDataLen,
longSampleRate, channels, byteRate);
while (in1.read(data) != -1) {
floatArray = lowPass(toFloatArray(data), toFloatArray(dataOut));
// out.write(data); original
out.write(toByteArray(floatArray));
}
out.close();
in1.close();
Toast.makeText(this, "Done!!", Toast.LENGTH_LONG).show();
} catch (FileNotFoundException e) {
e.printStackTrace();
} catch (IOException e) {
e.printStackTrace();
}
}
private void WriteWaveFileHeader(FileOutputStream out, long totalAudioLen,
long totalDataLen, long longSampleRate, int channels, long byteRate)
throws IOException {
byte[] header = new byte[44];
header[0] = 'R'; // RIFF/WAVE header
header[1] = 'I';
header[2] = 'F';
header[3] = 'F';
header[4] = (byte) (totalDataLen & 0xff);
header[5] = (byte) ((totalDataLen >> 8) & 0xff);
header[6] = (byte) ((totalDataLen >> 16) & 0xff);
header[7] = (byte) ((totalDataLen >> 24) & 0xff);
header[8] = 'W';
header[9] = 'A';
header[10] = 'V';
header[11] = 'E';
header[12] = 'f'; // 'fmt ' chunk
header[13] = 'm';
header[14] = 't';
header[15] = ' ';
header[16] = 16; // 4 bytes: size of 'fmt ' chunk
header[17] = 0;
header[18] = 0;
header[19] = 0;
header[20] = 1; // format = 1
header[21] = 0;
header[22] = (byte) channels;
header[23] = 0;
header[24] = (byte) (longSampleRate & 0xff);
header[25] = (byte) ((longSampleRate >> 8) & 0xff);
header[26] = (byte) ((longSampleRate >> 16) & 0xff);
header[27] = (byte) ((longSampleRate >> 24) & 0xff);
header[28] = (byte) (byteRate & 0xff);
header[29] = (byte) ((byteRate >> 8) & 0xff);
header[30] = (byte) ((byteRate >> 16) & 0xff);
header[31] = (byte) ((byteRate >> 24) & 0xff);
header[32] = (byte) (2 * 16 / 8); // block align
header[33] = 0;
header[34] = RECORDER_BPP; // bits per sample
header[35] = 0;
header[36] = 'd';
header[37] = 'a';
header[38] = 't';
header[39] = 'a';
header[40] = (byte) (totalAudioLen & 0xff);
header[41] = (byte) ((totalAudioLen >> 8) & 0xff);
header[42] = (byte) ((totalAudioLen >> 16) & 0xff);
header[43] = (byte) ((totalAudioLen >> 24) & 0xff);
out.write(header, 0, 44);
}
}
可以帮助我解决奇怪声音以及如何获得更多音频滤波器算法并实现的人。
答案 0 :(得分：0)
您可以尝试使用Matlab查看波的类型以及噪声所存在的频率类型。然后，您可以将低通滤波器设计为在噪声边界处具有截止频率。
答案 1 :(得分：0)
如果您正在尝试构建无限脉冲响应（IIR）滤波器（我猜您来自代码），滤波器方程将是当前输入样本和先前输出样本的加权和。
由于output[n]尚未在等式的右侧计算，您的噪音很可能会出现。
至于设计过滤器，我希望你的数学很好 - 你需要从第一原理开始研究它，以了解它是如何工作的。许多人使用MatLab在他们决定过滤器类型和顺序后实际计算系数。
关于教程 - 看看免费的大学课件 - 这是电子工程学位的第一和第二年本科材料。
答案 2 :(得分：0)
两件事。
首先，WAV文件不包含浮点数，它包含整数。通过将它们解码为浮点数，你会得到一个完全不合适的信号 - 我很惊讶你可以听到输出中的任何内容。
其次，您的lowPass函数不执行低通滤波器。即使这样，您也需要检查溢出/下溢并将值钳制回有效范围。