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做ppt的模板的网站,做好的网站怎么发布,网页设计与制作的岗位职责,如何建设好一个网站导航app引导中经常遇到破音，这里也将之前经历过的方案收集以下，方便以后选择： 1 对于开始和结尾破音： 可以用升降音来处理两种方式一种是直接对开始和结束的时间段进行音量直接渐进改变。这里配的是200ms的渐变。 VolumeSha…

导航app引导中经常遇到破音，这里也将之前经历过的方案收集以下，方便以后选择：

1 对于开始和结尾破音：可以用升降音来处理

两种方式

一种是直接对开始和结束的时间段进行音量直接渐进改变。这里配的是200ms的渐变。
VolumeShaper.Configuration cfg_out= null;
if (android.os.Build.VERSION.SDK_INT >= android.os.Build.VERSION_CODES.O) {
cfg_out = new VolumeShaper.Configuration.Builder()
.setCurve(new float[]{0f,1f},new float[]{1f,0f})
.setInterpolatorType(VolumeShaper.Configuration.INTERPOLATOR_TYPE_LINEAR)
.setDuration(200)
.build();
VolumeShaper vShaper = mAudioTrack.createVolumeShaper(cfg_out);
vShaper.apply(VolumeShaper.Operation.PLAY);
}

一种是开始的那帧数据进行音量从零渐进增加到当前音量，结束的那几帧数据进行音量从当前音量降到零
/**
* 对音频数据做 fade out
* @param byteBuffer byteBuffer
* @param channelCount channelCount
*/
private ByteBuffer shortFadeOut(ByteBuffer byteBuffer, int channelCount) {
int shortCount = byteBuffer.limit() / 2;
if(1 == channelCount) {
for(int i = 0; i < shortCount; i++) {
short data = (short) (byteBuffer.getShort(i * 2) * 1.0f * (shortCount - i) / (2*shortCount));
byteBuffer.putShort(i * 2, data);
}
} else {
for(int i = 0; i < shortCount; i += 2) {
short data = (short) (byteBuffer.getShort(i * 2) * 1.0f * (shortCount - i) / (2*shortCount));
byteBuffer.putShort(i * 2, data);
data = (short)(byteBuffer.getShort((i + 1) * 2) * 1.0f * (shortCount - i) / (2*shortCount));
byteBuffer.putShort((i + 1) * 2, data);
}
}
byteBuffer.rewind();
return byteBuffer;
}

2 适用于自己的tts引擎

tts放入app进程会受当前app的业务影响，导致tts 不稳定，尤其是导航app，大量的cpu，内存占用是常有的事，可单独放到一个独立进程里，并且启动个前台服务提高优先级。
怎么两个进程沟通呢，由于是低频的沟通，直接广播即可。

3 不固定位置的破音：直接控制tts解析出来的数据块

原理：破音由于系统处理的数据不足，或数据塞入间隔时间过长过短，我们这里直接控制每次写入的数据大小及间隔数据：
详细看下代码（系统不同，代码效果也不一样，要和系统tts端配合，而且要能拿到tts解析数据，我们是自己的tts引擎）：

public class AudioTrackManager {
public static final String TAG = "AudioTrackManager";
private AudioTrack audioTrack;
private static AudioTrackManager mInstance;
private int bufferSize;
private byte[] simpleBytes = null;
private int writeRate = 180;
private int pushRate = 90;
//系统一次处理的数据块的最小值，小于的话，就会破音
private static int RateSize = 1900;

private SyncStack syncStack = new SyncStack();
private long oldTime = 0;
private ExecutorService pool = Executors.newSingleThreadExecutor();

//类似生产者，消费者的一个读写类（每写一次，都给一次取的机会，目的是不耽误取出播报的节奏）
class SyncStack {

LinkedBlockingQueue<byte[]> datas = new LinkedBlockingQueue<byte[]>();
long oldTime = 0;

public void clearData(){
datas.clear();
}

public synchronized void push(byte[] data) {
try {
datas.put(data);
long time = System.currentTimeMillis()-oldTime;
//空出机会给写入线程机会
if (time > pushRate) {
time = 5;
} else {
time = pushRate - time;
}

if(time>0) {
wait(time);
}
oldTime = System.currentTimeMillis();
} catch (InterruptedException e) {
e.printStackTrace();
}
// this.notify();
}

public synchronized byte[] pop() throws InterruptedException {
if (datas == null || datas.size() == 0) {
//50ms后不再等待数据，自动结束流程
if (datas == null || datas.size() == 0) {
wait(50);
}
if(datas==null||datas.size()==0) {
return null;
}
}
return datas.take();
}
}

public AudioTrackManager() {
bufferSize = AudioTrack.getMinBufferSize(8000, AudioFormat.CHANNEL_IN_STEREO, AudioFormat.ENCODING_PCM_16BIT);
audioTrack = new AudioTrack(AudioPolicyManager.STREAM_NAVI, 8000, AudioFormat.CHANNEL_IN_STEREO, AudioFormat.ENCODING_PCM_16BIT, bufferSize, AudioTrack.MODE_STREAM);
}

private void initTrack() {
if (audioTrack == null) {
bufferSize = AudioTrack.getMinBufferSize(8000, AudioFormat.CHANNEL_IN_STEREO, AudioFormat.ENCODING_PCM_16BIT);
audioTrack = new AudioTrack(AudioPolicyManager.STREAM_NAVI, 8000, AudioFormat.CHANNEL_IN_STEREO, AudioFormat.ENCODING_PCM_16BIT, bufferSize, AudioTrack.MODE_STREAM);
}
}

public static AudioTrackManager getInstance() {
if (mInstance == null) {
synchronized (AudioTrackManager.class) {
if (mInstance == null) {
mInstance = new AudioTrackManager();
}
}
}
return mInstance;
}

public void startReady() {
initTrack();
if(syncStack!=null) {
syncStack.clearData();
}else{
syncStack = new SyncStack();
}
}

//System.arraycopy()方法
public static byte[] byteMerger(byte[] bt1, byte[] bt2) {
byte[] bt3 = new byte[bt1.length + bt2.length];
System.arraycopy(bt1, 0, bt3, 0, bt1.length);
System.arraycopy(bt2, 0, bt3, bt1.length, bt2.length);
return bt3;
}
/**
* 停止播放
*/
public void stopPlay() {
try {
//destroyThread();
Log.v(TAG, "yangtest--stopTTS");
if(syncStack!=null){
syncStack.clearData();
}
if (audioTrack != null) {
if (audioTrack.getState() == AudioRecord.STATE_INITIALIZED) {
audioTrack.stop();
}
if (audioTrack != null) {
audioTrack.release();
}
audioTrack = null;
}
} catch (Exception e) {
e.printStackTrace();
}
}

//tts 服务会不停的传过来解析出来的据
public void startPush(byte[] data) {
syncStack.push(data);
}
//启动播报线程
public void startPop() {
Log.e("yangtest","startpop-bufferSize-"+bufferSize);
pool.execute(
new Runnable(){

public void run() {

android.os.Process.setThreadPriority(Process.THREAD_PRIORITY_AUDIO);
try {
//等待先写入数据一定的数据，防止进来就破音
Thread.sleep(getStartTime());
} catch (InterruptedException e) {
e.printStackTrace();
}

audioTrack.play();
try {
while ((simpleBytes = syncStack.pop()) != null) {

while (simpleBytes.length < RateSize) {
try {
//一次取的不够，先等待最小间隔时间再操作
Thread.sleep(writeRate);
} catch (InterruptedException e) {
e.printStackTrace();
}
byte[] temp = syncStack.pop();
if (temp != null) {
simpleBytes = byteMerger(simpleBytes, temp);
} else {
Log.e("yangtest", "no-data");
break;
}
}
startWrite();
}
} catch (InterruptedException e) {
e.printStackTrace();
}

if (endPlay != null) {
endPlay.onEnd();
}
}

});
}
/**
* 启动播放线程
*/
private void startWrite() {
//需先等待最小的间隔时间，保持播报节奏
long timelen = System.currentTimeMillis() - oldTime;
if (timelen < writeRate) {
try {
Thread.sleep(writeRate - timelen);
} catch (InterruptedException e) {
e.printStackTrace();
}
}
oldTime = System.currentTimeMillis();
audioTrack.write(simpleBytes, 0, simpleBytes.length);
simpleBytes = null;
}