可以将 live555 lib 与 armv7s 一起使用吗？因为我尝试用 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，并包含需要的每个标题），但仍然出现 14 个错误Undefinded symbols for architecture armv7s
我的代码：
#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
所以我再问一次，甚至可以在armv7s上使用live555吗？也许我应该使用另一个库？
解决方案/更新
我必须扔掉 armv7s，而不是使用这个项目来为 iOS 创建胖库。
分步解决方案：
- 克隆这个项目
git clone git@github.com:weevilgenius/live555-ios.git 构建 lib，并复制到您的项目中。一开始我在查找已编译的库时遇到问题。所以对于那些不知道如何找到它的人。在 Xcode 文件树中构建 lib 后，打开文件夹 Products 并右键单击 libLive555.a 然后选择“在 Finder 中显示”。
每个.hh 添加到您的项目中。这些文件位于文件夹中，包括在文件夹Live555的每个子文件夹中。现在一切都应该很好:)