android音频编辑（裁剪，合成）（2）

.. .-..—…-. -.—–..- (“I love you ”莫斯电码)，这是逛知乎的时候看到程序员的表白情书，感觉我们码农情商好高啊！哈哈，好了开始搬砖。

上一篇介绍了音频的采集过程。之后产品经理找我谈了话，表示功能跟界面都凑合！但是（听到“但是”表示头皮发麻），需要再加一个功能，就是音频在录制的过程中，可以暂停，并且可以删除到上次暂停的地方（此刻内心亿万头草泥马飞奔而过，官大一级压死人啊！）。下面先对上次的音频采集过程中出现的bug进行一下简单的修改。

一、音频的标记操作
上次的操作标记的移动速度是固定的，也就是每次surfaceView进行绘制（每隔20ms）时，标记点位向左平移3个像素，这就导致标记点位的移动速度与波形图不能形成相同速度的平移，由于刷新的频率较高，所以标记点在以肉眼的可见的偏移量偏离标记的位置。

这就很尴尬了。做出的修改就是在，计算录音采集的字节数跟总的画布长度的时候，计算每次移除list集合的字节数，再进行标记点位移。

下面是源码截图，只需要加一个参数即可：

二、音频采集的回删操作，下面是定下来的界面：

主要的改变也只是在音频采集的下面加了一个跑动的条形，每次暂停的时候，会在在这个条上画一个分隔线，删除的时候，条向右移动，

录制的时候，条形是向左平移的，它的平移速度跟上面的时间刻度条是一致的。

回删的操作，其实是在录制完成后，剪辑形成合并形成的，所以请看下面。

三、进入正题，音频的编辑。
先看界面如下：

操作者可以在底部那个左右滑动，控制切割点的位置，时间轴的生成方式与编辑的时间轴是不一样的，这个时间轴是动态的，是用

linerLayout动态添加子View生成的，很简单，每个刻度我这里的是60dp,你自己可以根据需要更改，ll_wave_content是包裹

timeLine的父控件。代码如下：

/**     * 音频的时间刻度     */    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); //字体加粗设置        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);        }

相对其他格式的音频文件，wav格式的相对比较简单，只是在pcm之上添加了头部，wav的头部格式如下：

好，看的不明白的同学可自行百度活谷歌，有很多文章介绍；

既然pcm格式加上wav的头部就可，那剪辑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;        }

剪辑的类，注意的是，需要先将你操作的wav文件塞进去，进行头文件的格式解析，之后就可算出你需要删除的帧区间，之后就是相关的逻辑运算了，这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，最近一段时间确实太忙了，其他项目的维护升级什么的，搞的头皮发麻。有什么问题可以留言交流。

声明：音频的裁剪这个类的原作者的一个开源小项目叫音乐快剪，我只是在其基础上进行了修改！其他格式的音频MP3的话还好，但是ACC或者M4a格式的裁剪就比较麻烦，需要进行重新编码，建议使用FFMPEG进行格式重新编码裁剪，至于FFMPEG的android平台移植，GITHUB上有很多，很多人的博客也有介绍，个人建议不要自己编译（您时间富裕除外），很多已经编译好了，直接使用即可。

Github地址（大家下载的时候顺便给个star也是对作者劳动成果的肯定，谢谢）：
https://github.com/T-chuangxin/VideoMergeDemo