live555源码分析----关于mp3的处理

    RFC3199定义了MP3的RTP打包规则。首先来看看处理*.mp3的sesseion是如何创建的

static ServerMediaSession* createNewSMS(UsageEnvironment& env,
					char const* fileName, FILE* /*fid*/) {
...
   else if (strcmp(extension, ".mp3") == 0) {
    // Assumed to be a MPEG-1 or 2 Audio file:
    NEW_SMS("MPEG-1 or 2 Audio");


    //去注释STREAM_USING_ADUS宏，传输时使用ADUs，而不是MP3裸帧
    // To stream using 'ADUs' rather than raw MP3 frames, uncomment the following:
//#define STREAM_USING_ADUS 1
    //去注释INTERLEAVE_ADUS宏，在传输之前奖ADUs重排序(交错)
    // To also reorder ADUs before streaming, uncomment the following:
//#define INTERLEAVE_ADUS 1
    // (For more information about ADUs and interleaving,
    //  see <http://www.live555.com/rtp-mp3/>)
    Boolean useADUs = False;
    Interleaving* interleaving = NULL;
#ifdef STREAM_USING_ADUS
    useADUs = True;
#ifdef INTERLEAVE_ADUS
    unsigned char interleaveCycle[] = {0,2,1,3}; // or choose your own...
    unsigned const interleaveCycleSize
      = (sizeof interleaveCycle)/(sizeof (unsigned char));
    interleaving = new Interleaving(interleaveCycleSize, interleaveCycle);  //创建一个用于交错的filter
#endif
#endif
    sms->addSubsession(MP3AudioFileServerMediaSubsession::createNew(env, fileName, reuseSource, useADUs, interleaving)); //注意这里传递的参数


  }
}

    上面的代码打开宏STREAM_USING_ADUS，则会将MP3帧打包成ADU后，再发送。打开宏INTERLEAVE_ADUS，MP3打包成ADU后，将进行交错排列。可以看到默认情况下，这两个选项都是关闭的。在MP3AudioFileServerMediaSubsession::createNewStreamSource函数中会调用一个createNewStreamSourceCommon函数，处理ADU的打包操作。

FramedSource* MP3AudioFileServerMediaSubsession
::createNewStreamSourceCommon(FramedSource* baseMP3Source, unsigned mp3NumBytes, unsigned& estBitrate) {
  FramedSource* streamSource;
  fFileDuration = 0.0;
  do {
    streamSource = baseMP3Source; // by default
    if (streamSource == NULL) break;


    // Use the MP3 file size, plus the duration, to estimate the stream's bitrate:
    if (mp3NumBytes > 0 && fFileDuration > 0.0) {
      estBitrate = (unsigned)(mp3NumBytes/(125*fFileDuration) + 0.5); // kbps, rounded
    } else {
      estBitrate = 128; // kbps, estimate
    }


    if (fGenerateADUs) {  //判断是否打包成ADU后发送
      // Add a filter that converts the source MP3s to ADUs:
      streamSource = ADUFromMP3Source::createNew(envir(), streamSource);
      if (streamSource == NULL) break;


      if (fInterleaving != NULL) {
	// Add another filter that interleaves the ADUs before packetizing:
	streamSource = MP3ADUinterleaver::createNew(envir(), *fInterleaving,
						    streamSource);
	if (streamSource == NULL) break;
      }
    } else if (fFileDuration > 0.0) {   
        //
        //注意了，对于不需要打包成ADU的情况，这里有一个打包再解包的过程，为是的方便定位
        //
      // Because this is a seekable file, insert a pair of filters: one that
      // converts the input MP3 stream to ADUs; another that converts these
      // ADUs back to MP3.  This allows us to seek within the input stream without
      // tripping over the MP3 'bit reservoir':
      streamSource = ADUFromMP3Source::createNew(envir(), streamSource);