.. .-..---...-. -.-----..- (“I love you ”莫斯電碼)，這是逛知乎的時候看到程序員的表白情書，感覺我們碼農情商好高啊！哈哈，好了開始搬磚。
上一篇介紹了音頻的采集過程。之后產品經理找我談了話，表示功能跟界面都湊合！但是（聽到“但是”表示頭皮發麻），需要再加一個功能，就是音頻在錄制的過程中，可以暫停，并且可以刪除到上次暫停的地方（此刻內心億萬頭草泥馬飛奔而過，官大一級壓死人啊！）。下面先對上次的音頻采集過程中出現的bug進行一下簡單的修改。
一、音頻的標記操作
上次的操作標記的移動速度是固定的，也就是每次surfaceView進行繪制（每隔20ms）時，標記點位向左平移3個像素，這就導致標記點位的移動速度與波形圖不能形成相同速度的平移，由于刷新的頻率較高，所以標記點在以肉眼的可見的偏移量偏離標記的位置。
這就很尷尬了。做出的修改就是在，計算錄音采集的字節數跟總的畫布長度的時候，計算每次移除list集合的字節數，再進行標記點位移。
下面是源碼截圖，只需要加一個參數即可：
這里寫圖片描述
這里寫圖片描述
二、音頻采集的回刪操作，下面是定下來的界面：
主要的改變也只是在音頻采集的下面加了一個跑動的條形，每次暫停的時候，會在在這個條上畫一個分隔線，刪除的時候，條向右移動，
錄制的時候，條形是向左平移的，它的平移速度跟上面的時間刻度條是一致的。
回刪的操作，其實是在錄制完成后，剪輯形成合并形成的，所以請看下面。
這里寫圖片描述
三、進入正題，音頻的編輯。
先看界面如下：
這里寫圖片描述
操作者可以在底部那個左右滑動，控制切割點的位置，時間軸的生成方式與編輯的時間軸是不一樣的，這個時間軸是動態的，是用
linerLayout動態添加子View生成的，很簡單，每個刻度我這里的是60dp,你自己可以根據需要更改，ll_wave_content是包裹
timeLine的父控件。代碼如下：
/**
* 音頻的時間刻度
*/
private void timeSize() {
timeLine = (LinearLayout)this.findViewById(R.id.ll_time_counter);
tv_totalTime.setText(formatTime(totalTime)+"");
timeLine.removeAllViews();
totleLength = totalTime*DensityUtil.dip2px(60);
// timeLine1.removeAllViews();
ll_wave_content1.setLayoutParams(new FrameLayout.LayoutParams(totalTime*DensityUtil.dip2px
(60),LayoutParams.MATCH_PARENT));
ll_wave_content.setLayoutParams(new FrameLayout.LayoutParams(totalTime*DensityUtil.dip2px
(60),LayoutParams.MATCH_PARENT));
timeLine1.setLayoutParams(new RelativeLayout.LayoutParams(totalTime*DensityUtil.dip2px
(60),LayoutParams.MATCH_PARENT));
for(int i=0;i<totalTime;i++){
LinearLayout line1=new LinearLayout(this);
line1.setOrientation(LinearLayout.HORIZONTAL);
line1.setLayoutParams(new LayoutParams(DensityUtil.dip2px
(60),LinearLayout.LayoutParams.WRAP_CONTENT));
line1.setGravity(Gravity.CENTER);
TextView timeText=new TextView(this);
timeText.setText(formatTime(i));
timeText.setWidth(DensityUtil.dip2px(60)-2);
timeText.setGravity(Gravity.CENTER_HORIZONTAL);
TextPaint paint = timeText.getPaint();
paint.setFakeBoldText(true); //字體加粗設置
timeText.setTextColor(Color.rgb(204, 204, 204));
View line2=new View(this);
line2.setBackgroundColor(Color.rgb(204, 204, 204));
line2.setPadding(0, 10, 0, 0);
line1.addView(timeText);
line1.addView(line2);
timeLine.addView(line1);
}
相對其他格式的音頻文件，wav格式的相對比較簡單，只是在pcm之上添加了頭部，wav的頭部格式如下：
這里寫圖片描述
好，看的不明白的同學可自行百度活谷歌，有很多文章介紹；
既然pcm格式加上wav的頭部就可，那剪輯或者合成就很方便了，合成的方法奉上：
/**
* merge *.wav files
* @param target output file
* @param paths the files that need to merge
* @return whether merge files success
*/
public static boolean mergeAudioFiles(String target,List<String> paths) {
try {
FileOutputStream fos = new FileOutputStream(target);
int size=0;
byte[] buf = new byte[1024 * 1000];
int PCMSize = 0;
for(int i=0;i<paths.size();i++){
FileInputStream fis = new FileInputStream(paths.get(i));
size = fis.read(buf);
while (size != -1){
PCMSize += size;
size = fis.read(buf);
}
fis.close();
}
PCMSize=PCMSize-paths.size()*44;
WaveHeader header = new WaveHeader();
header.fileLength = PCMSize + (44 - 8);
header.FmtHdrLeth = 16;
header.BitsPerSample = 16;
header.Channels = 1;
header.FormatTag = 0x0001;
header.SamplesPerSec = 16000;
header.BlockAlign = (short) (header.Channels * header.BitsPerSample / 8);
header.AvgBytesPerSec = header.BlockAlign * header.SamplesPerSec;
header.DataHdrLeth = PCMSize;
byte[] h = header.getHeader();
assert h.length == 44;
fos.write(h, 0, h.length);
for(int j=0;j<paths.size();j++){
FileInputStream fis = new FileInputStream(paths.get(j));
size = fis.read(buf);
boolean isFirst=true;
while (size != -1){
if(isFirst){
fos.write(buf, 44, size-44);
size = fis.read(buf);
isFirst=false;
}else{
fos.write(buf, 0, size);
size = fis.read(buf);
}
}
fis.close();
}
fos.close();
} catch (Exception e) {
e.printStackTrace();
return false;
}
return true;
}
剪輯的類，注意的是，需要先將你操作的wav文件塞進去，進行頭文件的格式解析，之后就可算出你需要刪除的幀區間，之后就是相關的邏輯運算了，這里我就不一一啰嗦了：
public class CheapWAV extends CheapSoundFile {
public static Factory getFactory() {
return new Factory() {
public CheapSoundFile create() {
return new CheapWAV();
}
public String[] getSupportedExtensions() {
return new String[] { "wav" };
}
};
}
// Member variables containing frame info
private int mNumFrames;
private int[] mFrameOffsets;
private int[] mFrameLens;
private int[] mFrameGains;
private int mFrameBytes;
private int mFileSize;
private int mSampleRate;
private int mChannels;
// Member variables used during initialization
private int mOffset;
public CheapWAV() {
}
public int getNumFrames() {
return mNumFrames;
}
public int getSamplesPerFrame() {
return mSampleRate / 50;
}
public int[] getFrameOffsets() {
return mFrameOffsets;
}
public int[] getFrameLens() {
return mFrameLens;
}
public int[] getFrameGains() {
return mFrameGains;
}
public int getFileSizeBytes() {
return mFileSize;
}
public int getAvgBitrateKbps() {
return mSampleRate * mChannels * 2 / 1024;
}
public int getSampleRate() {
return mSampleRate;
}
public int getChannels() {
return mChannels;
}
public String getFiletype() {
return "WAV";
}
// public int secondsToFrames(double seconds) {
// return (int)(1.0 * seconds * mSampleRate / mSamplesPerFrame + 0.5);
// }
public void ReadFile(File inputFile)
throws java.io.FileNotFoundException,
java.io.IOException {
super.ReadFile(inputFile);
mFileSize = (int)mInputFile.length();
if (mFileSize < 128) {
throw new java.io.IOException("File too small to parse");
}
FileInputStream stream = new FileInputStream(mInputFile);
byte[] header = new byte[12];
stream.read(header, 0, 12);
mOffset += 12;
if (header[0] != 'R' ||
header[1] != 'I' ||
header[2] != 'F' ||
header[3] != 'F' ||
header[8] != 'W' ||
header[9] != 'A' ||
header[10] != 'V' ||
header[11] != 'E') {
throw new java.io.IOException("Not a WAV file");
}
mChannels = 0;
mSampleRate = 0;
while (mOffset + 8 <= mFileSize) {
byte[] chunkHeader = new byte[8];
stream.read(chunkHeader, 0, 8);
mOffset += 8;
int chunkLen =
((0xff & chunkHeader[7]) << 24) |
((0xff & chunkHeader[6]) << 16) |
((0xff & chunkHeader[5]) << 8) |
((0xff & chunkHeader[4]));
if (chunkHeader[0] == 'f' &&
chunkHeader[1] == 'm' &&
chunkHeader[2] == 't' &&
chunkHeader[3] == ' ') {
if (chunkLen < 16 || chunkLen > 1024) {
throw new java.io.IOException(
"WAV file has bad fmt chunk");
}
byte[] fmt = new byte[chunkLen];
stream.read(fmt, 0, chunkLen);
mOffset += chunkLen;
int format =
((0xff & fmt[1]) << 8) |
((0xff & fmt[0]));
mChannels =
((0xff & fmt[3]) << 8) |
((0xff & fmt[2]));
mSampleRate =
((0xff & fmt[7]) << 24) |
((0xff & fmt[6]) << 16) |
((0xff & fmt[5]) << 8) |
((0xff & fmt[4]));
if (format != 1) {
throw new java.io.IOException(
"Unsupported WAV file encoding");
}
} else if (chunkHeader[0] == 'd' &&
chunkHeader[1] == 'a' &&
chunkHeader[2] == 't' &&
chunkHeader[3] == 'a') {
if (mChannels == 0 || mSampleRate == 0) {
throw new java.io.IOException(
"Bad WAV file: data chunk before fmt chunk");
}
int frameSamples = (mSampleRate * mChannels) / 50;
mFrameBytes = frameSamples * 2;
mNumFrames = (chunkLen + (mFrameBytes - 1)) / mFrameBytes;
mFrameOffsets = new int[mNumFrames];
mFrameLens = new int[mNumFrames];
mFrameGains = new int[mNumFrames];
byte[] oneFrame = new byte[mFrameBytes];
int i = 0;
int frameIndex = 0;
while (i < chunkLen) {
int oneFrameBytes = mFrameBytes;
if (i + oneFrameBytes > chunkLen) {
i = chunkLen - oneFrameBytes;
}
stream.read(oneFrame, 0, oneFrameBytes);
int maxGain = 0;
for (int j = 1; j < oneFrameBytes; j += 4 * mChannels) {
int val = java.lang.Math.abs(oneFrame[j]);
if (val > maxGain) {
maxGain = val;
}
}
mFrameOffsets[frameIndex] = mOffset;
mFrameLens[frameIndex] = oneFrameBytes;
mFrameGains[frameIndex] = maxGain;
frameIndex++;
mOffset += oneFrameBytes;
i += oneFrameBytes;
if (mProgressListener != null) {
boolean keepGoing = mProgressListener.reportProgress(
i * 1.0 / chunkLen);
if (!keepGoing) {
break;
}
}
}
} else {
stream.skip(chunkLen);
mOffset += chunkLen;
}
}
}
public void WriteFile(File outputFile, int startFrame, int numFrames)
throws java.io.IOException {
outputFile.createNewFile();
FileInputStream in = new FileInputStream(mInputFile);
FileOutputStream out = new FileOutputStream(outputFile);
long totalAudioLen = 0;
for (int i = 0; i < numFrames; i++) {
totalAudioLen += mFrameLens[startFrame + i];
}
long totalDataLen = totalAudioLen + 36;
long longSampleRate = mSampleRate;
long byteRate = mSampleRate * 2 * mChannels;
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) mChannels;
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 * mChannels); // 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);
byte[] buffer = new byte[mFrameBytes];
int pos = 0;
for (int i = 0; i < numFrames; i++) {
int skip = mFrameOffsets[startFrame + i] - pos;
int len = mFrameLens[startFrame + i];
if (skip < 0) {
continue;
}
if (skip > 0) {
in.skip(skip);
pos += skip;
}
in.read(buffer, 0, len);
out.write(buffer, 0, len);
pos += len;
}
in.close();
out.close();
}
};
好了，最近一段時間確實太忙了，其他項目的維護升級什么的，搞的頭皮發麻。有什么問題可以留言交流。
聲明：音頻的裁剪這個類的原作者的一個開源小項目叫音樂快剪，我只是在其基礎上進行了修改！其他格式的音頻MP3的話還好，但是ACC或者M4a格式的裁剪就比較麻煩，需要進行重新編碼，建議使用FFMPEG進行格式重新編碼裁剪，至于FFMPEG的android平臺移植，GITHUB上有很多，很多人的博客也有介紹，個人建議不要自己編譯（您時間富裕除外），很多已經編譯好了，直接使用即可。
Github地址（大家下載的時候順便給個star也是對作者勞動成果的肯定，謝謝）：
https://github.com/T-chuangxin/VideoMergeDemo
