package com.honghe.record; import java.io.File; import java.io.FileInputStream; import java.io.FileNotFoundException; import java.io.FileOutputStream; import java.io.IOException; import android.media.AudioFormat; import android.media.AudioRecord; public class AudioRecordFunc { // 缓冲区字节大小 private int bufferSizeInBytes = 0; //AudioName裸音频数据文件 ,麦克风 private String AudioName = ""; //NewAudioName可播放的音频文件 private String NewAudioName = ""; private AudioRecord audioRecord; private boolean isRecord = false;// 设置正在录制的状态 private static AudioRecordFunc mInstance; private AudioRecordFunc() { } public synchronized static AudioRecordFunc getInstance() { if (mInstance == null) mInstance = new AudioRecordFunc(); return mInstance; } public int startRecordAndFile() { //判断是否有外部存储设备sdcard if (AudioFileFunc.isSdcardExit()) { if (isRecord) { return ErrorCode.E_STATE_RECODING; } else { if (audioRecord == null) creatAudioRecord(); audioRecord.startRecording(); // 让录制状态为true isRecord = true; // 开启音频文件写入线程 new Thread(new AudioRecordThread()).start(); return ErrorCode.SUCCESS; } } else { return ErrorCode.E_NOSDCARD; } } public void stopRecordAndFile() { close(); } public long getRecordFileSize() { return AudioFileFunc.getFileSize(NewAudioName); } private void close() { if (audioRecord != null) { System.out.println("stopRecord"); isRecord = false;//停止文件写入 audioRecord.stop(); audioRecord.release();//释放资源 audioRecord = null; } } private void creatAudioRecord() { // 获取音频文件路径 AudioName = AudioFileFunc.getRawFilePath(); NewAudioName = AudioFileFunc.getWavFilePath(); // 获得缓冲区字节大小 bufferSizeInBytes = AudioRecord.getMinBufferSize(AudioFileFunc.AUDIO_SAMPLE_RATE, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT); // 创建AudioRecord对象 audioRecord = new AudioRecord(AudioFileFunc.AUDIO_INPUT, AudioFileFunc.AUDIO_SAMPLE_RATE, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT, bufferSizeInBytes); } class AudioRecordThread implements Runnable { @Override public void run() { writeDateTOFile();//往文件中写入裸数据 copyWaveFile(AudioName, NewAudioName);//给裸数据加上头文件 } } /** * 这里将数据写入文件,但是并不能播放,因为AudioRecord获得的音频是原始的裸音频, * 如果需要播放就必须加入一些格式或者编码的头信息。但是这样的好处就是你可以对音频的 裸数据进行处理,比如你要做一个爱说话的TOM * 猫在这里就进行音频的处理,然后重新封装 所以说这样得到的音频比较容易做一些音频的处理。 */ private void writeDateTOFile() { // new一个byte数组用来存一些字节数据,大小为缓冲区大小 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);// 建立一个可存取字节的文件 } catch (Exception e) { e.printStackTrace(); } while (isRecord == true) { readsize = audioRecord.read(audiodata, 0, bufferSizeInBytes); if (AudioRecord.ERROR_INVALID_OPERATION != readsize && fos != null) { try { fos.write(audiodata); } catch (IOException e) { e.printStackTrace(); } } } try { if (fos != null) fos.close();// 关闭写入流 } catch (IOException e) { e.printStackTrace(); } } // 这里得到可播放的音频文件 private void copyWaveFile(String inFilename, String outFilename) { FileInputStream in = null; FileOutputStream out = null; long totalAudioLen = 0; long totalDataLen = totalAudioLen + 36; long longSampleRate = AudioFileFunc.AUDIO_SAMPLE_RATE; int channels = 1; long byteRate = 16 * AudioFileFunc.AUDIO_SAMPLE_RATE * 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(); } } /** * 这里提供一个头信息。插入这些信息就可以得到可以播放的文件。 * 为我为啥插入这44个字节,这个还真没深入研究,不过你随便打开一个wav * 音频的文件,可以发现前面的头文件可以说基本一样哦。每种格式的文件都有 * 自己特有的头文件。 */ 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); } }