Android live555组播拉流客户端

背景：

live555作为知名的流媒体开源框架，在实际项目中，经常使用到。在Android播放器中，可以使用其作为流媒体部分的拉流端，特别是对于RTSP及组播播放，live555相对还是很稳定的。
这次将其移植到Android SDK上，并完成RTSP及组播拉流小程序，权当玩乐及熟悉live555之用。
RTSP拉流小程序基本就是原来live555测试代码testRTSPClient.cpp，仅对其做了点小修改，让其能完成对电视节目RTSP流的获取，所以后面有机会再讲live555 RTSP内部实现流程吧。
这次就讲Android上移植live555及实现组播简单拉流代码。

Android移植live555

live555 Android平台移植方法CSDN上很多可以参考，并不难，我简单说下我的移植方法

首先官网下载live555源码 http://www.live555.com/liveMedia/public/
对于Android，可以删除不必要的目录，我仅保留如下目录：

编写Android.mk文件，如下，
将liveMedia目录中的CPP和C代码全部包含进来，另外其他几个目录需要用到的文件也添加进来。
指定LOCAL_CPPFLAGS 和LOCAL_LDFLAGS ，主要是对STL的支持。
这样编译即可，会在live555源码out目录下生成liblive555.so库。
编译过程中可能有些小错误，不同live555的错误可能略有不同，主要是对NDK的兼容问题，网上查查修改即可，比较简单。

LOCAL_PATH := $(call my-dir)include $(CLEAR_VARS)LOCAL_MODULE := liblive555
LOCAL_MODULE_PATH := $(LOCAL_PATH)/outLOCAL_C_INCLUDES := \$(LOCAL_PATH) \$(LOCAL_PATH)/BasicUsageEnvironment/include \$(LOCAL_PATH)/BasicUsageEnvironment \$(LOCAL_PATH)/UsageEnvironment/include \$(LOCAL_PATH)/UsageEnvironment \$(LOCAL_PATH)/groupsock/include \$(LOCAL_PATH)/groupsock \$(LOCAL_PATH)/liveMedia/include \$(LOCAL_PATH)/liveMedia \LOCAL_MODULE_TAGS := optionalprebuilt_stdcxx_PATH := prebuilts/ndk/current/sources/cxx-stl/gnu-libstdc++
SRC_LIST := $(wildcard $(LOCAL_PATH)/liveMedia/*.cpp)
SRC_LIST += $(wildcard $(LOCAL_PATH)/liveMedia/*.c)LOCAL_SRC_FILES := $(SRC_LIST:$(LOCAL_PATH)/%=%)LOCAL_SRC_FILES += \groupsock/GroupsockHelper.cpp \groupsock/GroupEId.cpp \groupsock/inet.c \groupsock/Groupsock.cpp \groupsock/NetInterface.cpp \groupsock/NetAddress.cpp \groupsock/IOHandlers.cpp \UsageEnvironment/UsageEnvironment.cpp \UsageEnvironment/HashTable.cpp \UsageEnvironment/strDup.cpp \BasicUsageEnvironment/BasicUsageEnvironment0.cpp \BasicUsageEnvironment/BasicUsageEnvironment.cpp \BasicUsageEnvironment/BasicTaskScheduler0.cpp \BasicUsageEnvironment/BasicTaskScheduler.cpp \BasicUsageEnvironment/DelayQueue.cpp \BasicUsageEnvironment/BasicHashTable.cpp \LOCAL_LDLIBS :=  -lm -llogLOCAL_CPPFLAGS := -fexceptions -DXLOCALE_NOT_USED=1 -DNULL=0 -DNO_SSTREAM=1 -UIP_ADD_SOURCE_MEMBERSHIP
LOCAL_LDFLAGS := -L$(prebuilt_stdcxx_PATH)/libs/$(TARGET_CPU_ABI) -lgnustl_static -lsupc++include $(BUILD_SHARED_LIBRARY)

组播拉流程序

代码参考测试代码testMPEG2TransportReceiver.cpp实现。
1、
Source 和 Sink 在live555中是两个非常重要的概念。
Source 发送端，流的起点，可直观理解为生产者，负责读取文件或网络流的信息。
Sink 接收端，流的终点，可理解为是消费者。
可以又多级source，对上级source在进行处理，也可以成为filter（实际上也是source），数据流向简单来说如下：

2、
Source 对于RTP封装的组播和UDP组播裸流传递的有所不同。
有RTP封装的情况下，创建SimpleRTPSource；
UDP裸流传递的情况下，则是创建BasicUDPSource；
同时创建MPEG2TransportStreamFramer这个filter来处理从source接受到TS流。当然也可以不用这个filter，testMPEG2TransportReceiver.cpp中就没有使用，经过MPEG2TransportStreamFramer可以对TS进行一些解析和处理等。
代码如下：

int main(int argc, char** argv) {if (argc < 2) {return 1;}char ipStr[64] = "";unsigned short portNum = 0;Boolean isRTP = (strstr(argv[1],"rtp://") != 0)? True:False;if(isRTP){sscanf(argv[1],"rtp://%[^:]:%hu",ipStr,&portNum);}else{sscanf(argv[1],"udp://%[^:]:%hu",ipStr,&portNum);}if(portNum == 0 || ipStr[0] == 0){return 1;   }TaskScheduler* scheduler = BasicTaskScheduler::createNew();env = BasicUsageEnvironment::createNew(*scheduler);struct in_addr sessionAddress;sessionAddress.s_addr = our_inet_addr(ipStr);const unsigned char ttl = 1; // low, in case routers don't admin scope//RTP & UDP Socketconst Port port(portNum);Groupsock inputsock(*env, sessionAddress, port, ttl);//RTCP Socketconst Port rtcpPort(portNum+1);Groupsock rtcpGroupsock(*env, sessionAddress, rtcpPort, ttl);#if 1
if(isRTP){//RTP// Create the data source: a "MPEG-2 TransportStream RTP source" (which uses a 'simple' RTP payload format):sessionState.udpSource = NULL;sessionState.rtpSource = SimpleRTPSource::createNew(*env, &inputsock, 33, 90000, "video/MP2T", 0, False /*no 'M' bit*/);sessionState.readSource= MPEG2TransportStreamFramer::createNew(*env, sessionState.rtpSource);if(sessionState.readSource == NULL){*env << "create source error...\n";return 1;}// Create (and start) a 'RTCP instance' for the RTP source:const unsigned estimatedSessionBandwidth = 5000; // in kbps; for RTCP b/w shareconst unsigned maxCNAMElen = 100;unsigned char CNAME[maxCNAMElen+1];gethostname((char*)CNAME, maxCNAMElen);CNAME[maxCNAMElen] = '\0'; // just in casesessionState.rtcpInstance= RTCPInstance::createNew(*env, &rtcpGroupsock,estimatedSessionBandwidth, CNAME,NULL /* we're a client */, sessionState.rtpSource);// Note: This starts RTCP running automatically}else{//UDP // Create the data source: a "MPEG-2 TransportStream udp source"sessionState.rtpSource = NULL;sessionState.udpSource = BasicUDPSource::createNew(*env, &inputsock);sessionState.readSource = MPEG2TransportStreamFramer::createNew(*env, sessionState.udpSource);if(sessionState.readSource == NULL){*env << "create source error...\n";return 1;}}
......省略
}

3、
Sink 实现如下：
igmpSink继承MediaSink ，每次收取一帧数据，会调用到afterGettingFrame，通过continuePlaying又会处理获取下一帧数据，从而成为一个循环。所以在afterGettingFrame将流数据dump到文件mDumpFile之中。

// igmpSink //
class igmpSink: public MediaSink {
public:// "bufferSize" should be at least as large as the largest expected input frame.static igmpSink* createNew(UsageEnvironment& env, char const* fileName,unsigned bufferSize = 20000);protected:igmpSink(UsageEnvironment& env, FILE* fid, unsigned bufferSize);virtual ~igmpSink();protected: // redefined virtual functions:virtual Boolean continuePlaying();protected:static void afterGettingFrame(void* clientData, unsigned frameSize,unsigned numTruncatedBytes,struct timeval presentationTime,unsigned durationInMicroseconds);virtual void afterGettingFrame(unsigned frameSize,unsigned numTruncatedBytes,struct timeval presentationTime);FILE* mDumpFile;unsigned char* fBuffer;unsigned fBufferSize;
};// igmpSink //
igmpSink::igmpSink(UsageEnvironment& env, FILE* fid, unsigned bufferSize): MediaSink(env), mDumpFile(fid), fBufferSize(bufferSize){fBuffer = new unsigned char[bufferSize];
}igmpSink::~igmpSink() {delete[] fBuffer;if (mDumpFile != NULL) fclose(mDumpFile);
}igmpSink* igmpSink::createNew(UsageEnvironment& env, char const* fileName, unsigned bufferSize) {FILE* fid;fid = fopen(fileName, "wb");if (fid == NULL) return NULL;return new igmpSink(env, fid, bufferSize);
}Boolean igmpSink::continuePlaying() {if (fSource == NULL) return False;fSource->getNextFrame(fBuffer, fBufferSize,afterGettingFrame, this,onSourceClosure, this);return True;
}void igmpSink::afterGettingFrame(void* clientData, unsigned frameSize,unsigned numTruncatedBytes,struct timeval presentationTime,unsigned /*durationInMicroseconds*/) {igmpSink* sink = (igmpSink*)clientData;sink->afterGettingFrame(frameSize, numTruncatedBytes, presentationTime);
}void igmpSink::afterGettingFrame(unsigned frameSize,unsigned numTruncatedBytes,struct timeval presentationTime) {if (numTruncatedBytes > 0) {envir() << "igmpSink::afterGettingFrame(): The input frame data was too large for our buffer size ("<< fBufferSize << ").  "<< numTruncatedBytes << " bytes of trailing data was dropped!  Correct this by increasing the \"bufferSize\" parameter in the \"createNew()\" call to at least "<< fBufferSize + numTruncatedBytes << "\n";}//envir() << "afterGettingFrame, Write to  file\n";if (mDumpFile != NULL && fBuffer != NULL) {fwrite(fBuffer, 1, frameSize, mDumpFile);}if (mDumpFile == NULL || fflush(mDumpFile) == EOF) {// The output file has closed.  Handle this the same way as if the input source had closed:if (fSource != NULL) fSource->stopGettingFrames();onSourceClosure();return;}// Then try getting the next frame:continuePlaying();
}

4、
最后只要将source和sink绑定，启动即可。
数据流为：udpSource/ rtpSource–>readSource（MPEG2TransportStreamFramer）–>igmpSink。
env->taskScheduler().doEventLoop(); 在live555内部会通过其SingleStep一直循环。（以后分析RTSP流程时，再分析live555内部流程。）组播流会被dump到指定文件。

  sessionState.sink = igmpSink::createNew(*env, "/storage/external_storage/sda1/testIGMP.ts", 5120);// Finally, start receiving the multicast stream:*env << "Beginning receiving multicast stream...\n";sessionState.sink->startPlaying(*sessionState.readSource, afterPlaying, NULL);env->taskScheduler().doEventLoop(); // does not return

5、
测试时可以使用VLC推流，RTP和UDP均可，可以看到流会被dump到我们指定的目录，拿出来播放，看是否一样即可。