关于android 使用audiorecord 录制pcm文件 音频速度变快的问题求教

录制音频文件

我使用aduiorecord录制pcm文件 录制代码如下

创建实例

mAudioRecorder = new AudioRecord(
						mRecorderBuilder.mAudioSource,
						sample_rate,
						channel_config, 
						format,
						mRecorderBuilder.bufferSize);
						

其中参数含义

private int mAudioSource = MediaRecorder.AudioSource.VOICE_COMMUNICATION;//自动降噪,也尝试过 MIC
		private int mSampleRate = SAMPLE_RATE_44K_HZ;//44100
		private int mChannelConfig = AudioFormat.CHANNEL_IN_MONO;//单声道
		private int mAudioFormat = AudioFormat.ENCODING_PCM_16BIT;//每次采样位数
		mRecorderBuilder.bufferSize//缓冲区大小,其值大于 用AudioRecord.getMinBufferSize(sample_rate, channel_config, format) 获取的最小值

录音

//mPcmBuffer.size就是上面的mRecorderBuilder.bufferSize 大小
//private short[] mPcmBuffer
	mAudioRecorder.read(mPcmBuffer, 0, mPcmBuffer.length);

存储

我是通过随机存储文件对象,进行数据的存储.
由于需要存储为wav格式的文件,因此需要写头文件.
头文件中需要计算 当前录音的格式 数据
第一次写,网上找的格式写的¹

RandomAccessFile rand = new RandomAccessFile(saveFile, "rw");

RandomAccessFile rand = randomAccessFile;


		long totalDataLen = totalAudioLen + 36;
		rand.seek(0);

		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) (nChannels & 0xff);
		header[23] = (byte) ((nChannels >> 8) & 0xff);

		header[24] = (byte) (sampleRate & 0xff);//采样率
		header[25] = (byte) ((sampleRate >> 8) & 0xff);
		header[26] = (byte) ((sampleRate >> 16) & 0xff);
		header[27] = (byte) ((sampleRate >> 24) & 0xff);

		header[28] = (byte) (byteRate & 0xff);//取八位
		header[29] = (byte) ((byteRate >> 8) & 0xff);
		header[30] = (byte) ((byteRate >> 16) & 0xff);
		header[31] = (byte) ((byteRate >> 24) & 0xff);

		int b = weikuan * nChannels / 8;//每次采样的大小
		header[32] = (byte) (b & 0xff); // block align
		header[33] = (byte) ((b >> 8) & 0xff);
		header[34] = (byte) (weikuan & 0xff);
		header[35] = (byte) ((weikuan >> 8) & 0xff);
		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);

		rand.write(header, 0, 44);

写完之后将写入pcm数据流 我这里分装了一个对象,单独开了一个写文件线程.
通过handler 开启工作线程(之前的录音也是一个单独的线程具体看demo)

saveFileThread = new SaveFileThread();
		saveFileThread.start();
		saveHandler = new SaveFileThread.SaveHandler(saveFileThread);

				if (msg.what == SAVEING) {
					saving = true;
					while (saveFileThread.doWork() > 0) ;
//					Log.e(SaveFileThreadTAG, "收到 msg");

					saving = false;
				} else if (msg.what == STOP) {
					saveFileThread.finish();
					removeCallbacksAndMessages(null);
					getLooper().quit();

				}

			SaveTask remove = saveDatas.remove(0);
			int mSize = remove.mSize;
			saveFile.saveByet(remove.saveData, 0, mSize);

至此wav文件录制完毕

遇见的问题

录出来的wav文件,播放时 语速非常快, 感觉是快进了两倍.
找了很多原因都找不到,网上搜了几天,无果.只有一篇帖子很像²

但是该帖子并没有解决我的问题

特此发帖,一方面是记录一下自己的demo
二来,希望哪位大神可以帮我看一下,我的demo到底出了说明问题

ps :demo中还实现了 lame库 将 pcm格式转换为mp3 格式的文件.使用 pcm格式文件转换可以正常使用

我的demo地址 https://github.com/MartinLi89/myrecorder.git

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1. wav头文件说明(https://blog.csdn.net/hsy12342611/article/details/80075836) ↩︎

2. 帖子https://blog.csdn.net/marller/article/details/52882387 ↩︎