我已阅读this link的手册，但仍无法解决您的问题。
使用代码混合两个wav文件后，我得到一个wav文件（sdcard上的music.wav），但将其播放为 noise 。我真的不知道如何从music.wav文件中删除它。这对我来说是个难题。
这是我的源代码：
public class MainActivity extends Activity {
public static final int FREQUENCY = 44100;
private InputStream in1, in2;
private byte[] output;
@Override
protected void onCreate(Bundle savedInstanceState) {
super.onCreate(savedInstanceState);
setContentView(R.layout.activity_main);
try {
mixSound();
} catch (IOException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
}
private void mixSound() throws IOException {
// AudioTrack audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, 44100, AudioFormat.CHANNEL_OUT_STEREO, AudioFormat.ENCODING_PCM_16BIT, 44100, AudioTrack.MODE_STREAM);
in1 = getResources().openRawResource(R.raw.media_b);
in2 = getResources().openRawResource(R.raw.media_c);
byte[] arrayMusic1 = null;
arrayMusic1 = new byte[in1.available()];
arrayMusic1 = createMusicArray(in1);
in1.close();
byte[] arrayMusic2 = null;
arrayMusic2 = new byte[in2.available()];
arrayMusic2 = createMusicArray(in2);
in2.close();
output = new byte[arrayMusic1.length];
/** old
audioTrack.play();
*/
for (int i = 0; i < output.length; i++) {
float samplef1 = arrayMusic1[i] / 128.0f;
float samplef2 = arrayMusic2[i] / 128.0f;
float mixed = samplef1 + samplef2;
// reduce the volume a bit:
mixed *= 0.8;
// hard clipping
if (mixed > 1.0f) mixed = 1.0f;
if (mixed < -1.0f) mixed = -1.0f;
byte outputSample = (byte) (mixed * 128.0f);
output[i] = outputSample;
}
/** old
audioTrack.write(output, 0, output.length);
convertByteToFile(output);
save();
*/
saveToFile();
}
public static byte[] createMusicArray(InputStream is) throws IOException {
ByteArrayOutputStream baos = new ByteArrayOutputStream();
//10*1024 = 10KB
byte[] byteBuffer = new byte[10*1024];
int i = Integer.MAX_VALUE;
while ((i = is.read(byteBuffer, 0, byteBuffer.length)) > 0) {
baos.write(byteBuffer, 0, i);
}
return baos.toByteArray();
}
/**
* .WAV
* more info --> https://ccrma.stanford.edu/courses/422/projects/WaveFormat/
The canonical WAVE format starts with the RIFF header:
| FILE OFFSET | FIELD SIZE | FIELD NAME
0 4 ChunkID Contains the letters "RIFF" in ASCII form
(0x52494646 big-endian form).
4 4 ChunkSize 36 + SubChunk2Size, or more precisely:
4 + (8 + SubChunk1Size) + (8 + SubChunk2Size)
This is the size of the rest of the chunk
following this number. This is the size of the
entire file in bytes minus 8 bytes for the
two fields not included in this count:
ChunkID and ChunkSize.
8 4 Format Contains the letters "WAVE"
(0x57415645 big-endian form).
The "WAVE" format consists of two subchunks: "fmt " and "data":
The "fmt " subchunk describes the sound data's format:
12 4 Subchunk1ID Contains the letters "fmt "
(0x666d7420 big-endian form).
16 4 Subchunk1Size 16 for PCM. This is the size of the
rest of the Subchunk which follows this number.
20 2 AudioFormat PCM = 1 (i.e. Linear quantization)
Values other than 1 indicate some
form of compression.
22 2 NumChannels Mono = 1, Stereo = 2, etc.
24 4 SampleRate 8000, 44100, etc.
28 4 ByteRate == SampleRate * NumChannels * BitsPerSample/8
32 2 BlockAlign == NumChannels * BitsPerSample/8
The number of bytes for one sample including
all channels. I wonder what happens when
this number isn't an integer?
34 2 BitsPerSample 8 bits = 8, 16 bits = 16, etc.
2 ExtraParamSize if PCM, then doesn't exist
X ExtraParams space for extra parameters
The "data" subchunk contains the size of the data and the actual sound:
36 4 Subchunk2ID Contains the letters "data"
(0x64617461 big-endian form).
40 4 Subchunk2Size == NumSamples * NumChannels * BitsPerSample/8
This is the number of bytes in the data.
You can also think of this as the size
of the read of the subchunk following this
number.
44 * Data The actual sound data.
*/
public void saveToFile() {
try {
long mySubChunk1Size = 16;
int myBitsPerSample = 16;
int myFormat = 1;
long myChannels = 2;
long mySampleRate = 44100;
long myByteRate = mySampleRate * myChannels * (myBitsPerSample/8);
int myBlockAlign = (int) (myChannels * myBitsPerSample/8);
long myDataSize = output.length;
long myChunk2Size = myDataSize * myChannels * myBitsPerSample/8;
long myChunkSize = 36 + myChunk2Size;
OutputStream os = new FileOutputStream(new File(Environment.getExternalStorageDirectory().getPath()+"/videokit/mixed.wav"));
BufferedOutputStream bos = new BufferedOutputStream(os);
DataOutputStream outFile = new DataOutputStream(bos);
outFile.writeBytes("RIFF"); // 00 - RIFF
outFile.write(intToByteArray((int)myChunkSize), 0, 4); // 04 - how big is the rest of this file?
outFile.writeBytes("WAVE"); // 08 - WAVE
outFile.writeBytes("fmt "); // 12 - fmt
outFile.write(intToByteArray((int)mySubChunk1Size), 0, 4); // 16 - size of this chunk
outFile.write(shortToByteArray((short)myFormat), 0, 2); // 20 - what is the audio format? 1 for PCM = Pulse Code Modulation
outFile.write(shortToByteArray((short)myChannels), 0, 2); // 22 - mono or stereo? 1 or 2? (or 5 or ???)
outFile.write(intToByteArray((int)mySampleRate), 0, 4); // 24 - samples per second (numbers per second)
outFile.write(intToByteArray((int)myByteRate), 0, 4); // 28 - bytes per second
outFile.write(shortToByteArray((short)myBlockAlign), 0, 2); // 32 - # of bytes in one sample, for all channels
outFile.write(shortToByteArray((short)myBitsPerSample), 0, 2); // 34 - how many bits in a sample(number)? usually 16 or 24
outFile.writeBytes("data"); // 36 - data
outFile.write(intToByteArray((int)myDataSize), 0, 4); // 40 - how big is this data chunk
outFile.write(output); // 44 - the actual data itself - just a long string of numbers
outFile.flush();
outFile.close();
} catch (IOException e) {
e.printStackTrace();
}
}
public static byte[] getBytesFromFile(File file) throws IOException {
InputStream is = new FileInputStream(file);
long length = file.length();
if (length > Integer.MAX_VALUE) {
}
byte[] bytes = new byte[(int)length];
int offset = 0;
int numRead = 0;
while (offset < bytes.length && (numRead = is.read(bytes, offset, Math.min(bytes.length - offset, 512*1024))) >= 0) {
offset += numRead;
}
if (offset < bytes.length) {
Log.i("xxx", "Could not completely read file");
}
is.close();
return bytes;
}
private static byte[] intToByteArray(int i) {
byte[] b = new byte[4];
b[0] = (byte) (i & 0x00FF);
b[1] = (byte) ((i >> 8) & 0x000000FF);
b[2] = (byte) ((i >> 16) & 0x000000FF);
b[3] = (byte) ((i >> 24) & 0x000000FF);
return b;
}
public static byte[] shortToByteArray(short data) {
return new byte[]{(byte)(data & 0xff),(byte)((data >>> 8) & 0xff)};
}
public static long byteArrayToLong(byte[] b) {
int start = 0;
int i = 0;
int len = 4;
int cnt = 0;
byte[] tmp = new byte[len];
for (i = start; i < (start + len); i++) {
tmp[cnt] = b[i];
cnt++;
}
long accum = 0;
i = 0;
for (int shiftBy = 0; shiftBy < 32; shiftBy += 8) {
accum |= ((long) (tmp[i] & 0xff)) << shiftBy;
i++;
}
return accum;
}
}
文件mixed.wav，噪声范围视图的光谱分析 enter link description here
如果您遇到此问题，请给我一个解决方案。我该如何处理我的代码。你能帮我吗。我期待着您的建议。 非常感谢你。
答案 0 :(得分：0)
您没有提供足够的代码以确定，但至少有一个问题：
在saveToFile函数中，您将样本解释为16位（myBitsPerSample），在混合代码中，您将样本解释为字节（可能是8位）。
其中一个是不正确的（我怀疑它是你的混合代码，但如果它们真的是8位文件，请记住wav文件不支持带符号的8位音频，所以你需要改变你的做法你的转换。）
