AudioRecord錄音幫助類

Java · 發表 2018-10-30 11:40:20
摘要：使用方法： 1、初始化：AudioRecordManager.getInstance().init(); 2、開始錄音：AudioRecordManager.getInstance().startRecord(); 3、停止錄音：AudioReco...
使用方法：
1、初始化：AudioRecordManager.getInstance().init();
2、開始錄音：AudioRecordManager.getInstance().startRecord();
3、停止錄音：AudioRecordManager.getInstance().stopRecord();
4、錄音時會儲存.pcm和.wav兩種格式在手機的根目錄下。
public class AudioRecordManager {

private AudioRecordManager(){
}

private static AudioRecordManager INSTANCE;

public static AudioRecordManager getInstance(){
if(INSTANCE==null){
synchronized (AudioRecordManager.class){
if(INSTANCE==null){
INSTANCE = new AudioRecordManager();
}
}
}
return INSTANCE;
}

// 音訊獲取源
private int audioSource = MediaRecorder.AudioSource.MIC;
// 設定音訊取樣率，44100是目前的標準，但是某些裝置仍然支援22050，16000，11025
private static int sampleRateInHz = 16000;
// 設定音訊的錄製的聲道CHANNEL_IN_STEREO為雙聲道，CHANNEL_CONFIGURATION_MONO為單聲道
private static int channelConfig = AudioFormat.CHANNEL_IN_MONO;
// 音訊資料格式:PCM 16位每個樣本。保證裝置支援。PCM 8位每個樣本。不一定能得到裝置支援。
private static int audioFormat = AudioFormat.ENCODING_PCM_16BIT;
// 緩衝區位元組大小
private int bufferSizeInBytes = 0;
private AudioRecord audioRecord;
private boolean isRecord = false;// 設定正在錄製的狀態
//AudioName裸音訊資料檔案
private static String AudioName = Environment.getExternalStorageDirectory().getPath()+"/0.raw";
//NewAudioName可播放的音訊檔案
private static String NewAudioName =Environment.getExternalStorageDirectory().getPath()+"/1.wav";

public void init() {
// 獲得緩衝區位元組大小
bufferSizeInBytes = AudioRecord.getMinBufferSize(sampleRateInHz,
channelConfig, audioFormat);
// 建立AudioRecord物件
audioRecord = new AudioRecord(audioSource, sampleRateInHz,
channelConfig, audioFormat, bufferSizeInBytes);
}

public void stopRecord() {
close();
}

private void close() {
if (audioRecord != null) {
System.out.println("stopRecord");
isRecord = false;//停止檔案寫入
audioRecord.stop();
audioRecord.release();//釋放資源
audioRecord = null;
}
}

public void startRecord(String pathName){
NewAudioName =Environment.getExternalStorageDirectory().getPath()+ "/"+pathName+".wav";
AudioName =Environment.getExternalStorageDirectory().getPath()+ "/"+pathName+".pcm";
startRecord();
}

public void startRecord() {
if(audioRecord == null){
init();
}
audioRecord.startRecording();
// 讓錄製狀態為true
isRecord = true;
// 開啟音訊檔案寫入執行緒
new Thread(new AudioRecordThread()).start();
}

class AudioRecordThread implements Runnable {
@Override
public void run() {
writeDateTOFile();//往檔案中寫入裸資料
copyWaveFile(AudioName, NewAudioName);//給裸資料加上標頭檔案
}
}

/**
* 這裡將資料寫入檔案，但是並不能播放，因為AudioRecord獲得的音訊是原始的裸音訊，
* 如果需要播放就必須加入一些格式或者編碼的頭資訊。但是這樣的好處就是你可以對音訊的 裸資料進行處理，比如你要做一個愛說話的TOM
* 貓在這裡就進行音訊的處理，然後重新封裝 所以說這樣得到的音訊比較容易做一些音訊的處理。
*/
public void writeDateTOFile() {
// new一個byte陣列用來存一些位元組資料，大小為緩衝區大小
byte[] audiodata = new byte[bufferSizeInBytes];
FileOutputStream fos = null;
int readsize = 0;
try {
File file = new File(AudioName);
if (file.exists()) {
file.delete();
}
fos = new FileOutputStream(file);// 建立一個可存取位元組的檔案
} catch (Exception e) {
e.printStackTrace();
}
while (isRecord == true) {
readsize = audioRecord.read(audiodata, 0, bufferSizeInBytes);
if (AudioRecord.ERROR_INVALID_OPERATION != readsize) {
try {
fos.write(audiodata);
} catch (IOException e) {
e.printStackTrace();
}
}
}
try {
fos.close();// 關閉寫入流
} catch (IOException e) {
e.printStackTrace();
}
}

// 這裡得到可播放的音訊檔案
public void copyWaveFile(String inFilename, String outFilename) {
FileInputStream in = null;
FileOutputStream out = null;
long totalAudioLen = 0;
long totalDataLen = totalAudioLen + 36;
long longSampleRate = sampleRateInHz;
int channels = 1;
long byteRate = 16 * sampleRateInHz * channels / 8;
byte[] data = new byte[bufferSizeInBytes];
try {
in = new FileInputStream(inFilename);
out = new FileOutputStream(outFilename);
totalAudioLen = in.getChannel().size();
totalDataLen = totalAudioLen + 36;
WriteWaveFileHeader(out, totalAudioLen, totalDataLen,
longSampleRate, channels, byteRate);
while (in.read(data) != -1) {
out.write(data);
}
in.close();
out.close();
} catch (FileNotFoundException e) {
e.printStackTrace();
} catch (IOException e) {
e.printStackTrace();
}
}

/**
* 這裡提供一個頭資訊。插入這些資訊就可以得到可以播放的檔案。
* 為我為啥插入這44個位元組，這個還真沒深入研究，不過你隨便開啟一個wav
* 音訊的檔案，可以發現前面的標頭檔案可以說基本一樣哦。每種格式的檔案都有
* 自己特有的標頭檔案。
*/
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] = 16; // 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);
}

}
AudioRecord錄音幫助類

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