Android/Java 实现PCM与G.711编码互转
16lz
2021-01-26
背景
最近做一个UDP局域网通讯的项目，是 Android 手机和一个室内硬件外设通讯，因为这个外设的音频传输采用的是 G.711 a率 编码，Android 的 AudioRecord 又不能直接采集 G.711 a率 编码的音频，所以就想采集之后转成 G.711 来传输。
但是网络上搜了好几圈，都是使用的 Android jni 去调用 C++ 代码实现，我去尝试调用了几次都没成功，所以自己仿照着写了一个，经测试可以使用，特此分享。
格式参数
PCM:采样率：8000采样大小：16 BIT 声道：双声道 G.711采样率：8000 采样大小：16 BIT 声道：单声道 转换工具类： G711Code
/** * 核心转换 * Created by onlygx */ public class G711Code { private final static int SIGN_BIT = 0x80; private final static int QUANT_MASK = 0xf; private final static int SEG_SHIFT = 4; private final static int SEG_MASK = 0x70; static short[] seg_end = {0xFF, 0x1FF, 0x3FF, 0x7FF,0xFFF, 0x1FFF, 0x3FFF, 0x7FFF}; static short search(short val,short[] table,short size){ for (short i = 0 ; i < size; i++) { if(val <= table[i]){ return i; } } return size; } static byte linear2alaw(short pcm_val){ short mask; short seg; char aval; if(pcm_val >= 0){ mask = 0xD5; }else{ mask = 0x55; pcm_val = (short) (-pcm_val - 1); if(pcm_val < 0){ pcm_val = 32767; } } /* Convert the scaled magnitude to segment number. */ seg = search(pcm_val, seg_end, (short) 8); /* Combine the sign, segment, and quantization bits. */ if (seg >= 8) /* out of range, return maximum value. */ return (byte) (0x7F ^ mask); else { aval = (char) (seg << SEG_SHIFT); if (seg < 2) aval |= (pcm_val >> 4) & QUANT_MASK; else aval |= (pcm_val >> (seg + 3)) & QUANT_MASK; return (byte) (aval ^ mask); } } static short alaw2linear(byte a_val){ short t; short seg; a_val ^= 0x55; t = (short) ((a_val & QUANT_MASK) << 4); seg = (short) ((a_val & SEG_MASK) >> SEG_SHIFT); switch (seg) { case 0: t += 8; break; case 1: t += 0x108; break; default: t += 0x108; t <<= seg - 1; } return (a_val & SIGN_BIT) != 0 ? t : (short) -t; } /** * pcm 转 G711 a率 * @param pcm * @param code * @param size */ public static void G711aEncoder(short[] pcm,byte[] code,int size){ for(int i=0;i/** * G.711 转 PCM * @param pcm * @param code * @param size */ public static void G711aDecoder(short[] pcm,byte[] code,int size) { for(int i=0;i 附录1 音频采集工具类
工具类不是我原创的，但是百度太多已经忘记是在哪里看到的，在此感谢那位作者。
（貌似有好几个都是这么写的，不知道谁是原作者。）
简单给小白说一下，这个工具类需要先初始化，然后设置一个回调来处理采集到的数据。
实现 OnAudioFrameCapturedListener 接口，重写 onAudioFrameCaptured 方法，这是个很重要的回调。
如果不用了，要停止采集。
import android.media.AudioFormat; import android.media.AudioRecord; import android.media.MediaRecorder; import android.os.SystemClock; import android.util.Log; public class AudioCapturer { private static final String TAG = "AudioCapturer"; private static final int DEFAULT_SOURCE = MediaRecorder.AudioSource.MIC; private static final int DEFAULT_SAMPLE_RATE = 8000; private static final int DEFAULT_CHANNEL_CONFIG = AudioFormat.CHANNEL_IN_MONO; private static final int DEFAULT_AUDIO_FORMAT = AudioFormat.ENCODING_PCM_16BIT; private AudioRecord mAudioRecord; private int mMinBufferSize = 0; private Thread mCaptureThread; private boolean mIsCaptureStarted = false; private volatile boolean mIsLoopExit = false; private OnAudioFrameCapturedListener mAudioFrameCapturedListener; public interface OnAudioFrameCapturedListener { public void onAudioFrameCaptured(short[] audioData); } public boolean isCaptureStarted() { return mIsCaptureStarted; } public void setOnAudioFrameCapturedListener(OnAudioFrameCapturedListener listener) { mAudioFrameCapturedListener = listener; } public boolean startCapture() { return startCapture(DEFAULT_SOURCE, DEFAULT_SAMPLE_RATE, DEFAULT_CHANNEL_CONFIG, DEFAULT_AUDIO_FORMAT); } public boolean startCapture(int audioSource, int sampleRateInHz, int channelConfig, int audioFormat) { if (mIsCaptureStarted) { Log.e(TAG, "Capture already started !"); return false; } mMinBufferSize = AudioRecord.getMinBufferSize(sampleRateInHz,channelConfig,audioFormat); if (mMinBufferSize == AudioRecord.ERROR_BAD_VALUE) { Log.e(TAG, "Invalid parameter !"); return false; } Log.d(TAG , "getMinBufferSize = "+mMinBufferSize+" bytes !"); mAudioRecord = new AudioRecord(audioSource,sampleRateInHz ,channelConfig,audioFormat,mMinBufferSize); if (mAudioRecord.getState() == AudioRecord.STATE_UNINITIALIZED) { Log.e(TAG, "AudioRecord initialize fail !"); return false; } mAudioRecord.startRecording(); mIsLoopExit = false; mCaptureThread = new Thread(new AudioCaptureRunnable()); mCaptureThread.start(); mIsCaptureStarted = true; Log.d(TAG, "Start audio capture success !"); return true; } public void stopCapture() { if (!mIsCaptureStarted) { return; } mIsLoopExit = true; try { mCaptureThread.interrupt(); mCaptureThread.join(1000); } catch (InterruptedException e) { e.printStackTrace(); } if (mAudioRecord.getRecordingState() == AudioRecord.RECORDSTATE_RECORDING) { mAudioRecord.stop(); } mAudioRecord.release(); mIsCaptureStarted = false; mAudioFrameCapturedListener = null; Log.d(TAG, "Stop audio capture success !"); } private class AudioCaptureRunnable implements Runnable { @Override public void run() { while (!mIsLoopExit) { short[] buffer = new short[320]; int ret = mAudioRecord.read(buffer, 0, 320); if (ret == AudioRecord.ERROR_INVALID_OPERATION) { Log.e(TAG , "Error ERROR_INVALID_OPERATION"); } else if (ret == AudioRecord.ERROR_BAD_VALUE) { Log.e(TAG , "Error ERROR_BAD_VALUE"); } else { if (mAudioFrameCapturedListener != null) { mAudioFrameCapturedListener.onAudioFrameCaptured(buffer); } Log.d(TAG , "OK, Captured "+ret+" bytes !"); } SystemClock.sleep(10); } } } }附录2 音频播放工具类
import android.media.AudioFormat; import android.media.AudioManager; import android.media.AudioTrack; import android.util.Log; public class AudioReader { private int mFrequency;// 采样率 private int mChannel;// 声道 private int mSampBit; // 采样精度 private AudioTrack mAudioTrack; public AudioReader(){ mFrequency = 8000; mChannel = AudioFormat.CHANNEL_OUT_MONO; mSampBit = AudioFormat.ENCODING_PCM_16BIT; } public void init(){ if (mAudioTrack != null){ release(); } // 获得构建对象的最小缓冲区大小 int minBufSize = AudioTrack.getMinBufferSize(mFrequency,mChannel, mSampBit); //STREAM_ALARM：警告声 //STREAM_MUSCI：音乐声，例如music等 //STREAM_RING：铃声 //STREAM_SYSTEM：系统声音 //STREAM_VOCIE_CALL：电话声音 mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, mFrequency,mChannel,mSampBit,minBufSize, AudioTrack.MODE_STREAM); //AudioTrack中有MODE_STATIC和MODE_STREAM两种分类。 //STREAM的意思是由用户在应用程序通过write方式把数据一次一次得写到audiotrack中。 //这个和我们在socket中发送数据一样，应用层从某个地方获取数据， //例如通过编解码得到PCM数据，然后write到audiotrack。 //这种方式的坏处就是总是在JAVA层和Native层交互，效率损失较大。 //而STATIC的意思是一开始创建的时候，就把音频数据放到一个固定的buffer，然后直接传给audiotrack， //后续就不用一次次得write了。AudioTrack会自己播放这个buffer中的数据。 //这种方法对于铃声等内存占用较小，延时要求较高的声音来说很适用。 mAudioTrack.play(); } public void release(){ if (mAudioTrack != null){ mAudioTrack.stop(); mAudioTrack.release(); } } public void playAudioTrack(byte []data, int offset, int length){ if (data == null || data.length == 0){return ;} try { mAudioTrack.write(data, offset, length); } catch (Exception e) { Log.i("MyAudioTrack", "catch exception..."); } } public void playAudioTrack(short []data, int offset, int length){ if (data == null || data.length == 0){return ;} try { mAudioTrack.write(data, offset, length); } catch (Exception e) { Log.i("MyAudioTrack", "catch exception..."); } } public int getPrimePlaySize(){ int minBufSize = AudioTrack.getMinBufferSize(mFrequency, mChannel, mSampBit); return minBufSize * 2; } }更多相关文章
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