我的应用程序记录来自手机麦克风的声音。我正在使用android标准类（android.media.AudioRecord）来做到这一点。当我按开始按钮应用程序开始记录并按停止时，应用程序有2个按钮“开始”和“停止” strong>应用程序停止录制并给我回缓冲区，语音数据为.wav格式。一切正常。
我想以这种方式更改我的应用程序，当应用程序开始工作时它开始分析来自麦克风的声音，如果用户保持静音应用程序继续分析来自麦克风的收入声音，如果用户开始说话应用程序开始录制声音来自麦克风，然后用户完成通话应用程序停止录制，并返回相同的缓冲区，语音数据为.wav格式。
答案 0 :(得分：26)
只需将此代码添加到您的应用程序中，您就会检测到用户何时开始说话以及何时停止。
public void onCreate(Bundle savedInstanceState)
{
super.onCreate(savedInstanceState);
setContentView(R.layout.main);
// Get the minimum buffer size required for the successful creation of an AudioRecord object.
int bufferSizeInBytes = AudioRecord.getMinBufferSize( RECORDER_SAMPLERATE,
RECORDER_CHANNELS,
RECORDER_AUDIO_ENCODING
);
// Initialize Audio Recorder.
AudioRecord audioRecorder = new AudioRecord( MediaRecorder.AudioSource.MIC,
RECORDER_SAMPLERATE,
RECORDER_CHANNELS,
RECORDER_AUDIO_ENCODING,
bufferSizeInBytes
);
// Start Recording.
audioRecorder.startRecording();
int numberOfReadBytes = 0;
byte audioBuffer[] = new byte[bufferSizeInBytes];
boolean recording = false;
float tempFloatBuffer[] = new float[3];
int tempIndex = 0;
int totalReadBytes = 0;
byte totalByteBuffer[] = new byte[60 * 44100 * 2];
// While data come from microphone.
while( true )
{
float totalAbsValue = 0.0f;
short sample = 0;
numberOfReadBytes = audioRecorder.read( audioBuffer, 0, bufferSizeInBytes );
// Analyze Sound.
for( int i=0; i<bufferSizeInBytes; i+=2 )
{
sample = (short)( (audioBuffer[i]) | audioBuffer[i + 1] << 8 );
totalAbsValue += Math.abs( sample ) / (numberOfReadBytes/2);
}
// Analyze temp buffer.
tempFloatBuffer[tempIndex%3] = totalAbsValue;
float temp = 0.0f;
for( int i=0; i<3; ++i )
temp += tempFloatBuffer[i];
if( (temp >=0 && temp <= 350) && recording == false )
{
Log.i("TAG", "1");
tempIndex++;
continue;
}
if( temp > 350 && recording == false )
{
Log.i("TAG", "2");
recording = true;
}
if( (temp >= 0 && temp <= 350) && recording == true )
{
Log.i("TAG", "Save audio to file.");
// Save audio to file.
String filepath = Environment.getExternalStorageDirectory().getPath();
File file = new File(filepath,"AudioRecorder");
if( !file.exists() )
file.mkdirs();
String fn = file.getAbsolutePath() + "/" + System.currentTimeMillis() + ".wav";
long totalAudioLen = 0;
long totalDataLen = totalAudioLen + 36;
long longSampleRate = RECORDER_SAMPLERATE;
int channels = 1;
long byteRate = RECORDER_BPP * RECORDER_SAMPLERATE * channels/8;
totalAudioLen = totalReadBytes;
totalDataLen = totalAudioLen + 36;
byte finalBuffer[] = new byte[totalReadBytes + 44];
finalBuffer[0] = 'R'; // RIFF/WAVE header
finalBuffer[1] = 'I';
finalBuffer[2] = 'F';
finalBuffer[3] = 'F';
finalBuffer[4] = (byte) (totalDataLen & 0xff);
finalBuffer[5] = (byte) ((totalDataLen >> 8) & 0xff);
finalBuffer[6] = (byte) ((totalDataLen >> 16) & 0xff);
finalBuffer[7] = (byte) ((totalDataLen >> 24) & 0xff);
finalBuffer[8] = 'W';
finalBuffer[9] = 'A';
finalBuffer[10] = 'V';
finalBuffer[11] = 'E';
finalBuffer[12] = 'f'; // 'fmt ' chunk
finalBuffer[13] = 'm';
finalBuffer[14] = 't';
finalBuffer[15] = ' ';
finalBuffer[16] = 16; // 4 bytes: size of 'fmt ' chunk
finalBuffer[17] = 0;
finalBuffer[18] = 0;
finalBuffer[19] = 0;
finalBuffer[20] = 1; // format = 1
finalBuffer[21] = 0;
finalBuffer[22] = (byte) channels;
finalBuffer[23] = 0;
finalBuffer[24] = (byte) (longSampleRate & 0xff);
finalBuffer[25] = (byte) ((longSampleRate >> 8) & 0xff);
finalBuffer[26] = (byte) ((longSampleRate >> 16) & 0xff);
finalBuffer[27] = (byte) ((longSampleRate >> 24) & 0xff);
finalBuffer[28] = (byte) (byteRate & 0xff);
finalBuffer[29] = (byte) ((byteRate >> 8) & 0xff);
finalBuffer[30] = (byte) ((byteRate >> 16) & 0xff);
finalBuffer[31] = (byte) ((byteRate >> 24) & 0xff);
finalBuffer[32] = (byte) (2 * 16 / 8); // block align
finalBuffer[33] = 0;
finalBuffer[34] = RECORDER_BPP; // bits per sample
finalBuffer[35] = 0;
finalBuffer[36] = 'd';
finalBuffer[37] = 'a';
finalBuffer[38] = 't';
finalBuffer[39] = 'a';
finalBuffer[40] = (byte) (totalAudioLen & 0xff);
finalBuffer[41] = (byte) ((totalAudioLen >> 8) & 0xff);
finalBuffer[42] = (byte) ((totalAudioLen >> 16) & 0xff);
finalBuffer[43] = (byte) ((totalAudioLen >> 24) & 0xff);
for( int i=0; i<totalReadBytes; ++i )
finalBuffer[44+i] = totalByteBuffer[i];
FileOutputStream out;
try {
out = new FileOutputStream(fn);
try {
out.write(finalBuffer);
out.close();
} catch (IOException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
} catch (FileNotFoundException e1) {
// TODO Auto-generated catch block
e1.printStackTrace();
}
//*/
tempIndex++;
break;
}
// -> Recording sound here.
Log.i( "TAG", "Recording Sound." );
for( int i=0; i<numberOfReadBytes; i++ )
totalByteBuffer[totalReadBytes + i] = audioBuffer[i];
totalReadBytes += numberOfReadBytes;
//*/
tempIndex++;
}
}
选中link。
答案 1 :(得分：7)
之前我回答了类似的问题，一种方法是在录制时使用线程，对录制的字节使用语音功率分析过程，有一个示例代码供您参考：http://musicg.googlecode.com/files/musicg_android_demo.zip
答案 2 :(得分：2)
最好使用private static final int RECORDER_SAMPLERAT = 8000 ; 它对我有用。我认为它对你有帮助
答案 3 :(得分：1)
当我替换totalAbsValue += Math.abs( sample ) / (numberOfReadBytes/2) by totalAbsValue += (float)Math.abs( sample ) / ((float)numberOfReadBytes/(float)2)时，它工作正常。