armv7上的Live555（RTP / RTSP）

Question

可以在armv7s中使用live555 lib吗？我试着用config

编译它

# Change the following version number, if necessary, before running "genMakefiles iphoneos"
IOS_VERSION =       7.0

DEVELOPER_PATH =    /Applications/Xcode.app/Contents/Developer/Platforms/iPhoneOS.platform/Developer
TOOL_PATH =     $(DEVELOPER_PATH)/usr/bin
SDK_PATH =      $(DEVELOPER_PATH)/SDKs
SDK =           $(SDK_PATH)/iPhoneOS$(IOS_VERSION).sdk
COMPILE_OPTS =          $(INCLUDES) -I. $(EXTRA_LDFLAGS) -DBSD=1 -O2 -DSOCKLEN_T=socklen_t -DHAVE_SOCKADDR_LEN=1 -D_LARGEFILE_SOURCE=1 -D_FILE_OFFSET_BITS=64 -fPIC -arch armv7s --sysroot=$(SDK)
C =                     c
C_COMPILER =            $(TOOL_PATH)/gcc
C_FLAGS =               $(COMPILE_OPTS)
CPP =                   cpp
CPLUSPLUS_COMPILER =    $(TOOL_PATH)/g++
CPLUSPLUS_FLAGS =       $(COMPILE_OPTS) -Wall
OBJ =                   o
LINK =                  $(TOOL_PATH)/g++ -o 
LINK_OPTS =             -L. -arch armv7s --sysroot=$(SDK) -L$(SDK)/usr/lib/system
CONSOLE_LINK_OPTS =     $(LINK_OPTS)
LIBRARY_LINK =          libtool -s -o 
LIBRARY_LINK_OPTS =
LIB_SUFFIX =            a
LIBS_FOR_CONSOLE_APPLICATION =
LIBS_FOR_GUI_APPLICATION =
EXE =

然后我尝试将程序 testMP3Reciver 复制并粘贴到我的iOS项目中（是的，我使用.mm作为后缀而不是.m，并且需要包含每个标题），但仍然有14个错误{ {1}}

我的代码：

Undefinded symbols for architecture armv7s

所以我再问一遍，甚至可以在armv7s上使用live555？也许我应该使用另一个lib？

---

解决方案/更新

我必须抛弃armv7s，而不是使用this项目来为iOS创建胖lib。

分步解决方案：

1. 克隆this项目#import "TViewController.h" #include "liveMedia.hh" #include "GroupsockHelper.hh" #include "BasicUsageEnvironment.hh" @interface TViewController () @end @implementation TViewController UsageEnvironment* env; - (void)viewDidLoad { [super viewDidLoad]; // Do any additional setup after loading the view, typically from a nib. } - (void)didReceiveMemoryWarning { [super didReceiveMemoryWarning]; // Dispose of any resources that can be recreated. } struct sessionState_t { FramedSource* source; FileSink* sink; RTCPInstance* rtcpInstance; } sessionState; - (IBAction)start:(id)sender { if (!wasClicked) { //start [self startButton]; wasClicked = true; } else { //stop [self stopButton]; wasClicked = false; } } -(void)startButton{ // Begin by setting up our usage environment: TaskScheduler* scheduler = BasicTaskScheduler::createNew(); env = BasicUsageEnvironment::createNew(*scheduler); // Create the data sink for 'stdout': sessionState.sink = FileSink::createNew(*env, "stdout"); // Note: The string "stdout" is handled as a special case. // A real file name could have been used instead. // Create 'groupsocks' for RTP and RTCP: char const* sessionAddressStr #ifdef USE_SSM = "232.255.42.42"; #else = "239.255.42.42"; // Note: If the session is unicast rather than multicast, // then replace this string with "0.0.0.0" #endif const unsigned short rtpPortNum = 6666; const unsigned short rtcpPortNum = rtpPortNum+1; #ifndef USE_SSM const unsigned char ttl = 1; // low, in case routers don't admin scope #endif struct in_addr sessionAddress; sessionAddress.s_addr = our_inet_addr(sessionAddressStr); const Port rtpPort(rtpPortNum); const Port rtcpPort(rtcpPortNum); #ifdef USE_SSM char* sourceAddressStr = "aaa.bbb.ccc.ddd"; // replace this with the real source address struct in_addr sourceFilterAddress; sourceFilterAddress.s_addr = our_inet_addr(sourceAddressStr); Groupsock rtpGroupsock(*env, sessionAddress, sourceFilterAddress, rtpPort); Groupsock rtcpGroupsock(*env, sessionAddress, sourceFilterAddress, rtcpPort); rtcpGroupsock.changeDestinationParameters(sourceFilterAddress,0,~0); // our RTCP "RR"s are sent back using unicast #else Groupsock rtpGroupsock(*env, sessionAddress, rtpPort, ttl); Groupsock rtcpGroupsock(*env, sessionAddress, rtcpPort, ttl); #endif RTPSource* rtpSource; #ifndef STREAM_USING_ADUS // Create the data source: a "MPEG Audio RTP source" rtpSource = MPEG1or2AudioRTPSource::createNew(*env, &rtpGroupsock); #else // Create the data source: a "MP3 *ADU* RTP source" unsigned char rtpPayloadFormat = 96; // a dynamic payload type rtpSource = MP3ADURTPSource::createNew(*env, &rtpGroupsock, rtpPayloadFormat); #endif // Create (and start) a 'RTCP instance' for the RTP source: const unsigned estimatedSessionBandwidth = 160; // 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 */, rtpSource); // Note: This starts RTCP running automatically sessionState.source = rtpSource; #ifdef STREAM_USING_ADUS // Add a filter that deinterleaves the ADUs after depacketizing them: sessionState.source = MP3ADUdeinterleaver::createNew(*env, sessionState.source); if (sessionState.source == NULL) { *env << "Unable to create an ADU deinterleaving filter for the source\n"; exit(1); } // Add another filter that converts these ADUs to MP3s: sessionState.source = MP3FromADUSource::createNew(*env, sessionState.source); if (sessionState.source == NULL) { *env << "Unable to create an ADU->MP3 filter for the source\n"; exit(1); } #endif // Finally, start receiving the multicast stream: *env << "Beginning receiving multicast stream...\n"; sessionState.sink->startPlaying(*sessionState.source, afterPlaying, NULL); env->taskScheduler().doEventLoop(); // does not return } void afterPlaying(void* /*clientData*/) { *env << "...done receiving\n"; // End by closing the media: Medium::close(sessionState.rtcpInstance); // Note: Sends a RTCP BYE Medium::close(sessionState.sink); Medium::close(sessionState.source); } -(void)stopButton{ afterPlaying; } @end

2. 构建lib，并复制到您的项目。一开始我遇到了找到编译的lib的问题。所以对于那些不知道如何找到它的人。在Xcode文件树中打开lib后打开文件夹Products并右键单击libLive555.a，然后选择“在Finder中显示”。

3. 每隔.hh添加到您的项目中。这些文件位于 Live555 文件夹的每个子文件夹中 include 文件夹中。现在一切都应该没问题。）