import android.media.MediaCodec;
import android.media.MediaCodecInfo;
import android.media.MediaFormat;
import android.util.Log;
import java.io.ByteArrayOutputStream;
import java.io.File;
import java.io.FileOutputStream;
import java.io.IOException;
import java.nio.ByteBuffer;
/**
* pcm转AAC编码
* Created by gj on 2017/4/7.
*/
public class AacEncode {
private MediaCodec mediaCodec;
private String mediaType = "OMX.google.aac.encoder";
//解码后保存文件
private File file;
ByteBuffer[] inputBuffers = null;
ByteBuffer[] outputBuffers = null;
MediaCodec.BufferInfo bufferInfo;
FileOutputStream encFi = null;
//pts时间基数
long presentationTimeUs = 0;
//创建一个输出流用来输出转换的数据
ByteArrayOutputStream outputStream = new ByteArrayOutputStream();
public AacEncode() {
}
//设置输出路径
public void setOutputPath(String outPath){
File file = new File(outPath);
try {
encFi = new FileOutputStream(file);
}catch (IOException e){
e.printStackTrace();
}
}
public void start(){
try {
mediaCodec = MediaCodec.createEncoderByType(MediaFormat.MIMETYPE_AUDIO_AAC);
//mediaCodec = MediaCodec.createByCodecName(mediaType);
} catch (IOException e) {
e.printStackTrace();
}
// 设置音频采样率，44100是目前的标准，但是某些设备仍然支持22050，16000，11025
final int kSampleRates[] = {8000, 11025, 22050, 44100, 48000};
//比特率 声音中的比特率是指将模拟声音信号转换成数字声音信号后，单位时间内的二进制数据量，是间接衡量音频质量的一个指标
final int kBitRates[] = {64000, 96000, 128000};
//初始化 此格式使用的音频编码技术、音频采样率、使用此格式的音频信道数（单声道为 1，立体声为 2）
MediaFormat mediaFormat = MediaFormat.createAudioFormat(
MediaFormat.MIMETYPE_AUDIO_AAC, kSampleRates[3], 2);
mediaFormat.setString(MediaFormat.KEY_MIME, MediaFormat.MIMETYPE_AUDIO_AAC);
mediaFormat.setInteger(MediaFormat.KEY_AAC_PROFILE,MediaCodecInfo.CodecProfileLevel.AACObjectLC);
//比特率 声音中的比特率是指将模拟声音信号转换成数字声音信号后，单位时间内的二进制数据量，是间接衡量音频质量的一个指标
mediaFormat.setInteger(MediaFormat.KEY_BIT_RATE, kBitRates[1]);
//传入的数据大小
mediaFormat.setInteger(MediaFormat.KEY_MAX_INPUT_SIZE, 4096);// It will
//设置相关参数
mediaCodec.configure(mediaFormat, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
//开始
mediaCodec.start();
inputBuffers = mediaCodec.getInputBuffers();
outputBuffers = mediaCodec.getOutputBuffers();
bufferInfo = new MediaCodec.BufferInfo();
}
/**
* 关闭释放资源
*
* @author：gj
* @date: 2017/4/25
* @time: 16:19
**/
public void close() {
try {
mediaCodec.stop();
mediaCodec.release();
outputStream.flush();
outputStream.close();
encFi.close();
} catch (Exception e) {
e.printStackTrace();
}
}
/**
* 开始编码
* offset 偏移量
* size 数据大小
**/
public void encode(byte[] input, int offset, int size) throws Exception {
int inputBufferIndex = mediaCodec.dequeueInputBuffer(-1);//其中需要注意的有dequeueInputBuffer（-1），参数表示需要得到的毫秒数，-1表示一直等，0表示不需要等，传0的话程序不会等待，但是有可能会丢帧。
if (inputBufferIndex >= 0) {
ByteBuffer inputBuffer = inputBuffers[inputBufferIndex];
inputBuffer.clear();
inputBuffer.put(input);
inputBuffer.limit(input.length);
//计算pts
long pts = computePresentationTime(presentationTimeUs);
mediaCodec.queueInputBuffer(inputBufferIndex, offset, size, pts, 0);
presentationTimeUs += 1;
}
int outputBufferIndex = mediaCodec.dequeueOutputBuffer(bufferInfo, 0);
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);
//添加ADTS头
byte[] outData = new byte[outPacketSize];
addADTStoPacket(outData, outPacketSize);
outputBuffer.get(outData, 7, outBitsSize);
outputBuffer.position(bufferInfo.offset);
//写到输出流里
outputStream.write(outData);
// Log.e("AudioEncoder", outData.length + " bytes written");
mediaCodec.releaseOutputBuffer(outputBufferIndex, false);
outputBufferIndex = mediaCodec.dequeueOutputBuffer(bufferInfo, 0);
}
//输出流的数据转成byte[]
byte[] out = outputStream.toByteArray();
encFi.write(out);
//写完以后重置输出流，否则数据会重复
outputStream.flush();
outputStream.reset();
}
/**
* 给编码出的aac裸流添加adts头字段
*
* @param packet 要空出前7个字节，否则会搞乱数据
* @param packetLen
*/
private void addADTStoPacket(byte[] packet, int packetLen) {
int profile = 2; //AAC LC
int freqIdx = 4; //44.1KHz
int chanCfg = 2; //CPE
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;
}
//计算PTS，实际上这个pts对应音频来说作用并不大，设置成0也是没有问题的
private long computePresentationTime(long frameIndex) {
return frameIndex * 90000 * 1024 / 44100;
}
}
//调用
import android.app.Activity;
import android.os.Bundle;
import android.util.Log;
import java.io.File;
import java.io.FileInputStream;
import java.io.FileOutputStream;
import java.io.IOException;
public class PcmAac extends Activity {
@Override
protected void onCreate(Bundle savedInstanceState) {
super.onCreate(savedInstanceState);
setContentView(R.layout.activity_pcm_aac);
AacEncode aacEncode = new AacEncode();
aacEncode.start();
aacEncode.setOutputPath("/data/local/tmp/yi.aac");
int size = 4096;
byte[] inBuffer = new byte[size];
File inFile = new File("/data/local/tmp/yi.pcm");
FileInputStream in = null;
int len = 0;
try {
in = new FileInputStream(inFile);
while ((len = in.read(inBuffer, 0, 2048)) > 0){
aacEncode.encode(inBuffer, 1024, len);
}
aacEncode.close();
}catch (IOException e){
e.printStackTrace();
}catch (java.lang.Exception e){
e.printStackTrace();
}
}
}
