答案 0 :(得分:15)
在您的应用程序中试用此代码......
private int RECORDER_CHANNELS = AudioFormat.CHANNEL_CONFIGURATION_MONO;
private int RECORDER_AUDIO_ENCODING = AudioFormat.ENCODING_PCM_16BIT;
private int RECORDER_SAMPLERATE = 44100;
private byte RECORDER_BPP = (byte) 16;
private AudioRecord audioRecorder;
public void arm() {
// 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.
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++;
}
}
有关详细信息,您还可以查看此演示...
答案 1 :(得分:3)
请访问Android开发者页面,了解Audio Capture。
它有一个示例代码,演示如何捕获和播放声音。
答案 2 :(得分:1)
我正在尝试类似的东西。从我所看到的你必须开始录制才能读出音频电平。因此,我试图测量振幅,然后增加录制时间,如果MaxAmplitude达到可听见的点,则将文件保存到其他位置。
话虽如此,我对编程非常陌生,并且非常感谢任何额外的提示或技巧。