1、通過MediaRecord和AudioRecord同時分別錄製出無聲MP4視頻和raw音頻。
2、將raw音量增益並封裝成wav
3、將wav轉碼成AAC，並與MP4視頻合成爲目標錄像。(此處參考http://blog.csdn.net/smile3670/article/details/41279749)
錄音增益封裝成WAV音頻部分代碼如下
public class AudioMessageRecord {
private static final String TAG = "MessageRecord";
// 音頻獲取源
private int audioSource = MediaRecorder.AudioSource.MIC;
// 設置音頻採樣率，44100是目前的標準，但是某些設備仍然支持22050，16000，11025
private static int sampleRateInHz = 8000;
// 設置音頻的錄製的聲道CHANNEL_IN_STEREO爲雙聲道，CHANNEL_CONFIGURATION_MONO爲單聲道(CHANNEL_IN_MONO)
private static int channelConfig = AudioFormat.CHANNEL_IN_STEREO;
// 音頻數據格式:PCM 16位每個樣本。保證設備支持。PCM 8位每個樣本。不一定能得到設備支持。
private static int audioFormat = AudioFormat.ENCODING_PCM_16BIT;
// 緩衝區字節大小
private int bufferSizeInBytes = 0;
private AudioRecord audioRecord;
private boolean isRecord = false;// 設置正在錄製的狀態
// AudioName裸音頻數據文件
private static final String AudioName = "/sdcard/audio.raw";
// NewAudioName可播放的音頻文件
private String outFileName = "/sdcard/audio.wav";
public AudioMessageRecord(){
creatAudioRecord();
}
private void creatAudioRecord() {
// 獲得緩衝區字節大小
bufferSizeInBytes = AudioRecord.getMinBufferSize(sampleRateInHz,
channelConfig, audioFormat);
Log.e(TAG, "bufferSizeInBytes = " + bufferSizeInBytes);
// 創建AudioRecord對象
audioRecord = new AudioRecord(audioSource, sampleRateInHz,
channelConfig, audioFormat, bufferSizeInBytes);
}
public void startRecord(String fileName) {
outFileName = fileName;
audioRecord.startRecording();
// 讓錄製狀態爲true
isRecord = true;
// 開啓音頻文件寫入線程
new Thread(new AudioRecordThread()).start();
}
public void stopRecord() {
close();
}
private void close() {
if (audioRecord != null) {
System.out.println("stopRecord");
isRecord = false;// 停止文件寫入
audioRecord.stop();
audioRecord.release();// 釋放資源
audioRecord = null;
}
}
class AudioRecordThread implements Runnable {
@Override
public void run() {
writeDateTOFile();// 往文件中寫入裸數據
copyWaveFile(AudioName, outFileName);// 給裸數據加上頭文件\WAV格式
if (mAudioMessageRecordCallback != null) {
mAudioMessageRecordCallback.audioRecordEnd();
}
}
}
/**
* 這裏將數據寫入文件，但是並不能播放，因爲AudioRecord獲得的音頻是原始的裸音頻，
* 如果需要播放就必須加入一些格式或者編碼的頭信息。但是這樣的好處就是你可以對音頻的 裸數據進行處理，比如你要做一個愛說話的TOM
* 貓在這裏就進行音頻的處理，然後重新封裝 所以說這樣得到的音頻比較容易做一些音頻的處理。
*/
private void writeDateTOFile() {
// new一個byte數組用來存一些字節數據，大小爲緩衝區大小
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);// 建立一個可存取字節的文件
} catch (Exception e) {
e.printStackTrace();
}
while (isRecord == true) {
readsize = audioRecord.read(audiodata, 0, bufferSizeInBytes);
/*for(int i=0;i<audiodata.length;i++)
{
//音量大小,此種方法放大聲音會有底噪聲
audiodata[i]= (byte) (audiodata[i] * 5);//數字決定大小
}*/
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();
}
}
// 這裏得到可播放的音頻文件
private void copyWaveFile(String inFilename, String outFilename) {
FileInputStream in = null;
FileOutputStream out = null;
long totalAudioLen = 0;
long totalDataLen = totalAudioLen + 36;
long longSampleRate = sampleRateInHz;
int channels = 2;
long byteRate = 16 * sampleRateInHz * channels / 8;
//JNI增益處理必須是320
byte[] data = new byte[320];
try {
in = new FileInputStream(inFilename);
out = new FileOutputStream(outFilename);
totalAudioLen = in.getChannel().size();
totalDataLen = totalAudioLen + 36;
WriteWaveFileHeader(out, totalAudioLen, totalDataLen,
longSampleRate, channels, byteRate);
//音頻會比視頻晚一秒結束，經過試驗，減去最後一秒的數據
Log.i("count", "in.available() = " + in.available());
int count = in.available() / 320;
Log.i("count", "all count = " + count);
while (in.read(data) != -1) {
if (count --> 90) {
//通過JNI接口增益
SettingsJni.retrieveSettings().kotiNsAndAgc(data);
out.write(data);
}
}
in.close();
out.close();
} catch (FileNotFoundException e) {
e.printStackTrace();
} catch (IOException e) {
e.printStackTrace();
}
}
/**
* 這裏提供一個頭信息。插入這些信息就可以得到可以播放的文件。 爲我爲啥插入這44個字節，這個還真沒深入研究，不過你隨便打開一個wav
* 音頻的文件，可以發現前面的頭文件可以說基本一樣哦。每種格式的文件都有 自己特有的頭文件。
*/
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);
}
AudioMessageRecordCallback mAudioMessageRecordCallback;
public void setAudioCallback(AudioMessageRecordCallback audioMessageRecordCallback){
mAudioMessageRecordCallback = audioMessageRecordCallback;
}
public interface AudioMessageRecordCallback{
/**
* 錄音結束
*/
void audioRecordEnd();
}
}