首先利用AudioRecord进行麦克风采集:
int bufferSize = AudioRecord.getMinBufferSize(frequence, channelConfig, audioEncoding);
//实例化AudioRecord
final AudioRecord record = new AudioRecord(MediaRecorder.AudioSource.MIC, frequence, channelConfig, audioEncoding, bufferSize);
//开始录制
record.startRecording();
AacEncode aacMediaEncode = new AacEncode();
//定义缓冲
byte[] buffer = new byte[bufferSize];
//定义循环,根据isRecording的值来判断是否继续录制
while (isRecording) {
//从bufferSize中读取字节。
int bufferReadResult = record.read(buffer, 0, bufferSize);
//获取字节流
if (AudioRecord.ERROR_INVALID_OPERATION != bufferReadResult) {
//转成AAC编码
final byte[] ret = aacMediaEncode.offerEncoder(buffer);
/* if (ret.length > 0) {
byte[] out = aacDecode.offerDecoder(ret);
//发送数据到VLC,这个方法在视频发送那篇文章有,这里就不重复了。需要的可以去看看
netSendTask.pushBuf(ret, ret.length);
}*/
}
}
这里只是贴出一部分代码 buffer 就是读出原始PCM码流
接下来就是将PCM码流转化成为wav
private static void writeWaveFileHeader(FileOutputStream out, long totalAudioLen,
long totalDataLen, int sampleRate, int channels, long byteRate) throws IOException {
byte[] header = new byte[44];
header[0] = 'R'; // RIFF
header[1] = 'I';
header[2] = 'F';
header[3] = 'F';
header[4] = (byte) (totalDataLen & 0xff);//数据大小
header[5] = (byte) ((totalDataLen >> 8) & 0xff);
header[6] = (byte) ((totalDataLen >> 16) & 0xff);
header[7] = (byte) ((totalDataLen >> 24) & 0xff);
header[8] = 'W';//WAVE
header[9] = 'A';
header[10] = 'V';
header[11] = 'E';
//FMT Chunk
header[12] = 'f'; // 'fmt '
header[13] = 'm';
header[14] = 't';
header[15] = ' ';//过渡字节
//数据大小
header[16] = 16; // 4 bytes: size of 'fmt ' chunk
header[17] = 0;
header[18] = 0;
header[19] = 0;
//编码方式 10H为PCM编码格式
header[20] = 1; // format = 1
header[21] = 0;
//通道数
header[22] = (byte) channels;
header[23] = 0;
//采样率,每个通道的播放速度
header[24] = (byte) (sampleRate & 0xff);
header[25] = (byte) ((sampleRate >> 8) & 0xff);
header[26] = (byte) ((sampleRate >> 16) & 0xff);
header[27] = (byte) ((sampleRate >> 24) & 0xff);
//音频数据传送速率,采样率*通道数*采样深度/8
header[28] = (byte) (byteRate & 0xff);
header[29] = (byte) ((byteRate >> 8) & 0xff);
header[30] = (byte) ((byteRate >> 16) & 0xff);
header[31] = (byte) ((byteRate >> 24) & 0xff);
// 确定系统一次要处理多少个这样字节的数据,确定缓冲区,通道数*采样位数
header[32] = (byte) (channels * 16 / 8);
header[33] = 0;
//每个样本的数据位数
header[34] = 16;
header[35] = 0;
//Data chunk
header[36] = 'd';//data
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);
}
上面操作其实也很简单,只要你知道了WAV文件头信息的格式,将采样率,声道数,采样位数,PCM音频数据大小等信息填充进去,然后将这个44个字节数据拼接到PCM文件的开头,就得到了一个可播放的WAV文件了。