Cocos2d-x项目移植到WP8系列之五：播放MP3

 原文链接： http://www.cnblogs.com/zouzf/p/3972549.html

 

这一块的细节还是不太了解，只是东凑西拼能跑起来而已

1、网上下载lamb库 生成需要的lib库，详情见 【Advanced Windows Phone Programming】在windows phone 8中录制MP3和AAC及Amr音频  第三点的前半部分，也就是生成lib那一部分，然后在CocosDenshion工程里引用那两个lib。

2、把CocosDenshion工程里的Audio.cpp里的void Audio::PreloadSoundEffect(const char* pszFilePath, bool isMusic) 方法里的m_soundEffects[sound].m_soundID = sound; 这句代码之前的代码改成如下：

    if (m_engineExperiencedCriticalError) {
        return;
    }

    std::string path(pszFilePath);

    int sound = Hash(pszFilePath);

    MediaStreamer mediaStreamer;


    // no MP3 support for CC_PLATFORM_WP8
    std::string::size_type pos = path.find(".mp3");
    if (pos  != path.npos)
    {
        //path.replace(pos, path.length(), ".wav");

        mediaStreamer.Initialize_MP3(CCUtf8ToUnicode(pszFilePath).c_str());
    }
    else
    {
        mediaStreamer.Initialize(CCUtf8ToUnicode(pszFilePath).c_str());
    }

    
//    mediaStreamer.Initialize(CCUtf8ToUnicode(pszFilePath).c_str());
    m_soundEffects[sound].m_soundID = sound;    

3、CocosDenshion工程里的MediaStreamer类加多一个方法：void MediaStreamer::Initialize_MP3(__in const WCHAR* url)，实现如下：

void MediaStreamer::Initialize_MP3(__in const WCHAR* url)
{

    WCHAR filePath[MAX_PATH] = {0};
    if ((wcslen(url) > 1 && url[1] == ':'))
    {
        // path start with "x:", is absolute path
        wcscat_s(filePath, url);
    }
    else if (wcslen(url) > 0 
        && (L'/' == url[0] || L'\\' == url[0]))
    {
        // path start with '/' or '\', is absolute path without driver name
        wcscat_s(filePath, m_locationPath->Data());
        // remove '/' or '\\'
        wcscat_s(filePath, (const WCHAR*)url[1]);
    }else
    {
        wcscat_s(filePath, m_locationPath->Data());
        wcscat_s(filePath, url);
    }


    hip_t hip = hip_decode_init();
    if (!hip)  
    {
        printf("创建mp3解码失败"); 
        return ;
    } 

    mp3data_struct mp3str;//mp3文件编码信息
    std::vector<short*> mp3Buffer;// mp3数据流
    std::vector<int> mp3BufferSize;

    int samples;
    int mp3_bytes;
    int write_bytes = 0;

    const int BUF_SIZE = 512;
    const int INBUF_SIZE = 4096;
    const int MP3BUF_SIZE = (int)(1.25 * BUF_SIZE) + 7200;

    short pcm_l[INBUF_SIZE];
    short pcm_r[INBUF_SIZE];
    unsigned char mp3_buf[MP3BUF_SIZE];

    FILE * MP3File;
    

    std::wstring wstr = std::wstring(filePath);
//    return std::string(wstr.begin(),wstr.end());

    std::string str_filePath = std::string(wstr.begin(),wstr.end());//"C:\\Data\\Programs\\{4AE99D96-62EE-4A1C-A345-F1B0A81D07BC}\\Install\\Assets\\Resources\\islandbg.mp3";
    
    
    auto error = fopen_s(&MP3File, str_filePath.c_str(), "rb");
    mp3data_struct mp3Header;

    while ((mp3_bytes = fread(mp3_buf, 1, 210, MP3File)) > 0)
    {
        samples = hip_decode_headers(hip, mp3_buf, 210, pcm_l, pcm_r, &mp3Header);
        if (samples > 0)
        {
            short *tt = new short[samples*sizeof(short)];
            memcpy((void*)tt, (const void*)pcm_l, samples*sizeof(short));
            mp3Buffer.push_back(tt);
            write_bytes += samples*sizeof(short);
            mp3BufferSize.push_back(samples*sizeof(short));
            //write_bytes = fwrite(pcm_l, sizeof(short), samples, wavfp);
        }
    }


    byte* _mp3Buffer = new byte[write_bytes];
    byte* temp = _mp3Buffer;
    int size = mp3BufferSize.size();
    for(int i=0;i<size;i++)
    {
        memcpy(temp, mp3Buffer[i],mp3BufferSize[i]);
        delete mp3Buffer[i];
        temp += mp3BufferSize[i];
    }
    mp3Buffer.clear();
    hip_decode_exit(hip);
    //m_soundEffects[sound].m_soundEffectBufferData

    m_data.resize(write_bytes);
    for(int i=0;i<write_bytes;i++)
    {
        m_data[i]=_mp3Buffer[i];
    }

    fclose(MP3File);



    m_waveFormat.wFormatTag = WAVE_FORMAT_PCM; //固定
    m_waveFormat.nChannels = 1; //固定
    m_waveFormat.nSamplesPerSec = (DWORD) mp3Header.samplerate;// 固定 

    m_waveFormat.wBitsPerSample = 16;// mp3Header.bitrate * m_waveFormat.nChannels / 8; 
    //m_waveFormat.wBitsPerSample = mp3Header.bitrate * m_waveFormat.nChannels / 8.0; 
    m_waveFormat.nBlockAlign = m_waveFormat.nChannels * m_waveFormat.wBitsPerSample / 8.0; 
    m_waveFormat.nAvgBytesPerSec = m_waveFormat.nSamplesPerSec * m_waveFormat.nBlockAlign; //越大越快
    m_waveFormat.cbSize = 0;

}

完成。

大概流程就是，通过lamb的两个lib库的方法，读取MP3文件的流构建成wav文件格式的流，重点在于通过MP3的文件头信息mp3data_struct 构建出 wav文件的文件头信息WAVEFORMATEX。

另外，因为只读取了一个声道的流，所以m_waveFormat.nChannels = 1;如果对声道有要求的，可以在while循环里把右声道的流也处理一下。

附上一些参考文章： lame mp3解码      Suche MP3 to WAV decoder 

 

原文链接： http://www.cnblogs.com/zouzf/p/3972549.html