MediaCodec编译pcm为aac

先说说写这份博客的原因吧：

1.自己的第一篇博客，一味的拿来主义者太过于不厚道

2.记录下来也为自己以后使用，利人利己

3.代码都是亲自测试过

另外说一下需要改动的地方，

1，package是肯定需要改动的，你不可能跟我定义的一样吧

2，mediaFormat.setInteger(MediaFormat.KEY_MAX_INPUT_SIZE, 8192);这个参数是pcm buffer大小，可以根据自己的作适当修改,

3，如果还有不明白的地方可以参考以下链接

http://m.blog.csdn.NET/blog/cc_xueqin/41862131

里面讲解十分清楚，很好的文章

废话少说，代码献上

package com.sabine.codec;

import android.media.MediaCodec;
import android.media.MediaCodecInfo;
import android.media.MediaFormat;
import android.os.Environment;
import android.util.Log;


import java.io.BufferedOutputStream;
import java.io.File;
import java.io.FileOutputStream;
import java.io.IOException;
import java.nio.ByteBuffer;


public class AudioEncoder {


private MediaCodec mediaCodec;
private BufferedOutputStream outputStream;
private String mediaType = "OMX.google.aac.encoder";


ByteBuffer[] inputBuffers = null;
ByteBuffer[] outputBuffers = null;


// "OMX.qcom.audio.decoder.aac";
// "audio/mp4a-latm";


public AudioEncoder() {
File f = new File(Environment.getExternalStorageDirectory(),
"sabine/audio_encoded.aac");
touch(f);
try {
outputStream = new BufferedOutputStream(new FileOutputStream(f));
Log.e("AudioEncoder", "outputStream initialized");
} catch (Exception e) {
e.printStackTrace();
}


// mediaCodec = MediaCodec.createEncoderByType("audio/mp4a-latm");
mediaCodec = MediaCodec.createByCodecName(mediaType);
final int kSampleRates[] = { 8000, 11025, 22050, 44100, 48000 };
final int kBitRates[] = { 64000,96000,128000 };
MediaFormat mediaFormat = MediaFormat.createAudioFormat(
"audio/mp4a-latm", kSampleRates[3], 2);
mediaFormat.setInteger(MediaFormat.KEY_AAC_PROFILE,
MediaCodecInfo.CodecProfileLevel.AACObjectLC);
mediaFormat.setInteger(MediaFormat.KEY_BIT_RATE, kBitRates[1]);
mediaFormat.setInteger(MediaFormat.KEY_MAX_INPUT_SIZE, 8192);// It will
// increase
// capacity
// of
// inputBuffers
mediaCodec.configure(mediaFormat, null, null,
MediaCodec.CONFIGURE_FLAG_ENCODE);
mediaCodec.start();


inputBuffers = mediaCodec.getInputBuffers();
outputBuffers = mediaCodec.getOutputBuffers();
}


public void close() {
try {
mediaCodec.stop();
mediaCodec.release();
outputStream.flush();
outputStream.close();
} catch (Exception e) {
e.printStackTrace();
}
}


// called AudioRecord's read
public synchronized void offerEncoder(byte[] input) {
Log.e("AudioEncoder", input.length + " is coming");


int inputBufferIndex = mediaCodec.dequeueInputBuffer(-1);
if (inputBufferIndex >= 0) {
ByteBuffer inputBuffer = inputBuffers[inputBufferIndex];
inputBuffer.clear();


inputBuffer.put(input);


mediaCodec
.queueInputBuffer(inputBufferIndex, 0, input.length, 0, 0);
}


MediaCodec.BufferInfo bufferInfo = new MediaCodec.BufferInfo();
int outputBufferIndex = mediaCodec.dequeueOutputBuffer(bufferInfo, 0);


// //trying to add a ADTS
while (outputBufferIndex >= 0) {
int outBitsSize = bufferInfo.size;
int outPacketSize = outBitsSize + 7; // 7 is ADTS size
ByteBuffer outputBuffer = outputBuffers[outputBufferIndex];


outputBuffer.position(bufferInfo.offset);
outputBuffer.limit(bufferInfo.offset + outBitsSize);


byte[] outData = new byte[outPacketSize];
addADTStoPacket(outData, outPacketSize);


outputBuffer.get(outData, 7, outBitsSize);
outputBuffer.position(bufferInfo.offset);


// byte[] outData = new byte[bufferInfo.size];
try {
outputStream.write(outData, 0, outData.length);
} catch (IOException e) {
// TODO Auto-generated catch block
e.printStackTrace();
}
Log.e("AudioEncoder", outData.length + " bytes written");


mediaCodec.releaseOutputBuffer(outputBufferIndex, false);
outputBufferIndex = mediaCodec.dequeueOutputBuffer(bufferInfo, 0);


}


// Without ADTS header
/*
* while (outputBufferIndex >= 0) { ByteBuffer outputBuffer =
* outputBuffers[outputBufferIndex]; byte[] outData = new
* byte[bufferInfo.size];
* 
* outputBuffer.get(outData); try { outputStream.write(outData, 0,
* outData.length); } catch (IOException e) { // TODO Auto-generated
* catch block e.printStackTrace(); } Log.e("AudioEncoder",
* outData.length + " bytes written");
* 
* mediaCodec.releaseOutputBuffer(outputBufferIndex, false);
* outputBufferIndex = mediaCodec.dequeueOutputBuffer(bufferInfo, 0);
* 
* }
*/
}


/**
* Add ADTS header at the beginning of each and every AAC packet. This is
* needed as MediaCodec encoder generates a packet of raw AAC data.
* 
* Note the packetLen must count in the ADTS header itself.
**/
public void addADTStoPacket(byte[] packet, int packetLen) {
int profile = 2; // AAC LC
// 39=MediaCodecInfo.CodecProfileLevel.AACObjectELD;
int freqIdx = 4; // 44.1KHz
int chanCfg = 2; // CPE


// fill in ADTS data
packet[0] = (byte) 0xFF;
packet[1] = (byte) 0xF9;
packet[2] = (byte) (((profile - 1) << 6) + (freqIdx << 2) + (chanCfg >> 2));
packet[3] = (byte) (((chanCfg & 3) << 6) + (packetLen >> 11));
packet[4] = (byte) ((packetLen & 0x7FF) >> 3);
packet[5] = (byte) (((packetLen & 7) << 5) + 0x1F);
packet[6] = (byte) 0xFC;
}


public void touch(File f) {
try {
if (!f.exists())
f.createNewFile();
} catch (IOException e) {
e.printStackTrace();
}
}

}