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。
LOCAL_PATH := $(call my-dir)
include $(CLEAR_VARS)
LOCAL_MODULE := liblive555
LOCAL_MODULE_PATH := $(LOCAL_PATH)/out
LOCAL_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 := optional
prebuilt_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 -llog
LOCAL_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)
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：
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 Socket
const Port port(portNum);
Groupsock inputsock(*env, sessionAddress, port, ttl);
//RTCP Socket
const 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 share
const unsigned maxCNAMElen = 100;
unsigned char CNAME[maxCNAMElen+1];
gethostname((char*)CNAME, maxCNAMElen);
CNAME[maxCNAMElen] = '\0'; // just in case
sessionState.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