From 0688756b71b40e0ac60c68af2fa1fe4aaeb1718d Mon Sep 17 00:00:00 2001
From: houxiao <houxiao@454eff88-639b-444f-9e54-f578c98de674>
Date: 星期一, 13 二月 2017 16:27:41 +0800
Subject: [PATCH] replace log to support android
---
RtspFace/live555/testProgs/testRTSPClient.hpp | 62 +++++++++++++++---------------
1 files changed, 31 insertions(+), 31 deletions(-)
diff --git a/RtspFace/live555/testProgs/testRTSPClient.hpp b/RtspFace/live555/testProgs/testRTSPClient.hpp
index ff4a861..f346165 100644
--- a/RtspFace/live555/testProgs/testRTSPClient.hpp
+++ b/RtspFace/live555/testProgs/testRTSPClient.hpp
@@ -75,8 +75,8 @@
void usage(UsageEnvironment& env, char const* progName)
{
- LOG_DEBUG << "Usage: " << progName << " <rtsp-url-1> ... <rtsp-url-N>";
- LOG_DEBUG << "\t(where each <rtsp-url-i> is a \"rtsp://\" URL)";
+ LOG_DEBUG << "Usage: " << progName << " <rtsp-url-1> ... <rtsp-url-N>" << std::endl;
+ LOG_DEBUG << "\t(where each <rtsp-url-i> is a \"rtsp://\" URL)" << std::endl;
}
char eventLoopWatchVariable = 0;
@@ -207,7 +207,7 @@
RTSPClient* rtspClient = ourRTSPClient::createNew(env, _rtspConfig);
if (rtspClient == NULL)
{
- LOG_ERROR << "Failed to create a RTSP client for URL \"" << _rtspConfig.rtspURL.c_str() << "\": " << env.getResultMsg();
+ LOG_ERROR << "Failed to create a RTSP client for URL \"" << _rtspConfig.rtspURL.c_str() << "\": " << env.getResultMsg() << std::endl;
return;
}
@@ -231,25 +231,25 @@
if (resultCode != 0)
{
- LOG_WARN << *rtspClient << "Failed to get a SDP description: " << resultString;
+ LOG_WARN << *rtspClient << "Failed to get a SDP description: " << resultString << std::endl;
delete[] resultString;
break;
}
char* const sdpDescription = resultString;
- LOG_INFO << *rtspClient << "Got a SDP description:\n" << sdpDescription;
+ LOG_INFO << *rtspClient << "Got a SDP description:\n" << sdpDescription << std::endl;
// Create a media session object from this SDP description:
scs.session = MediaSession::createNew(env, sdpDescription);
delete[] sdpDescription; // because we don't need it anymore
if (scs.session == NULL)
{
- LOG_ERROR << *rtspClient << "Failed to create a MediaSession object from the SDP description: " << env.getResultMsg();
+ LOG_ERROR << *rtspClient << "Failed to create a MediaSession object from the SDP description: " << env.getResultMsg() << std::endl;
break;
}
else if (!scs.session->hasSubsessions())
{
- LOG_WARN << *rtspClient << "This session has no media subsessions (i.e., no \"m=\" lines)";
+ LOG_WARN << *rtspClient << "This session has no media subsessions (i.e., no \"m=\" lines)" << std::endl;
break;
}
@@ -276,17 +276,17 @@
{
if (!scs.subsession->initiate())
{
- LOG_ERROR << *rtspClient << "Failed to initiate the \"" << *scs.subsession << "\" subsession: " << env.getResultMsg();
+ LOG_ERROR << *rtspClient << "Failed to initiate the \"" << *scs.subsession << "\" subsession: " << env.getResultMsg() << std::endl;
setupNextSubsession(rtspClient); // give up on this subsession; go to the next one
}
else
{
- LOG_INFO << *rtspClient << "Initiated the \"" << *scs.subsession << "\" subsession (";
+ LOG_INFO << *rtspClient << "Initiated the \"" << *scs.subsession << "\" subsession (" << std::endl;
if (scs.subsession->rtcpIsMuxed())
- LOG_INFO << "client port " << scs.subsession->clientPortNum();
+ LOG_INFO << "client port " << scs.subsession->clientPortNum() << std::endl;
else
- LOG_INFO << "client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum()+1;
- LOG_INFO << ")";
+ LOG_INFO << "client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum()+1 << std::endl;
+ LOG_INFO << ")" << std::endl;
// Continue setting up this subsession, by sending a RTSP "SETUP" command:
rtspClient->sendSetupCommand(*scs.subsession, continueAfterSETUP, False, REQUEST_STREAMING_OVER_TCP);
@@ -316,20 +316,20 @@
if (resultCode != 0)
{
- LOG_ERROR << *rtspClient << "Failed to set up the \"" << *scs.subsession << "\" subsession: " << resultString;
+ LOG_ERROR << *rtspClient << "Failed to set up the \"" << *scs.subsession << "\" subsession: " << resultString << std::endl;
break;
}
- LOG_INFO << *rtspClient << "Set up the \"" << *scs.subsession << "\" subsession (";
+ LOG_INFO << *rtspClient << "Set up the \"" << *scs.subsession << "\" subsession (" << std::endl;
if (scs.subsession->rtcpIsMuxed())
{
- LOG_INFO << "client port " << scs.subsession->clientPortNum();
+ LOG_INFO << "client port " << scs.subsession->clientPortNum() << std::endl;
}
else
{
- LOG_INFO << "client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum()+1;
+ LOG_INFO << "client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum()+1 << std::endl;
}
- LOG_INFO << ")";
+ LOG_INFO << ")" << std::endl;
// Having successfully setup the subsession, create a data sink for it, and call "startPlaying()" on it.
// (This will prepare the data sink to receive data; the actual flow of data from the client won't start happening until later,
@@ -341,11 +341,11 @@
if (scs.subsession->sink == NULL)
{
LOG_ERROR << *rtspClient << "Failed to create a data sink for the \"" << *scs.subsession
- << "\" subsession: " << env.getResultMsg();
+ << "\" subsession: " << env.getResultMsg() << std::endl;
break;
}
- LOG_INFO << *rtspClient << "Created a data sink for the \"" << *scs.subsession << "\" subsession";
+ LOG_INFO << *rtspClient << "Created a data sink for the \"" << *scs.subsession << "\" subsession" << std::endl;
scs.subsession->miscPtr = rtspClient; // a hack to let subsession handler functions get the "RTSPClient" from the subsession
scs.subsession->sink->startPlaying(*(scs.subsession->readSource()),
subsessionAfterPlaying, scs.subsession);
@@ -373,7 +373,7 @@
if (resultCode != 0)
{
- LOG_ERROR << *rtspClient << "Failed to start playing session: " << resultString;
+ LOG_ERROR << *rtspClient << "Failed to start playing session: " << resultString << std::endl;
break;
}
@@ -389,12 +389,12 @@
scs.streamTimerTask = env.taskScheduler().scheduleDelayedTask(uSecsToDelay, (TaskFunc*)streamTimerHandler, rtspClient);
}
- LOG_INFO << *rtspClient << "Started playing session";
+ LOG_INFO << *rtspClient << "Started playing session" << std::endl;
if (scs.duration > 0)
{
- LOG_INFO << " (for up to " << scs.duration << " seconds)";
+ LOG_INFO << " (for up to " << scs.duration << " seconds)" << std::endl;
}
- LOG_INFO << "...";
+ LOG_INFO << "..." << std::endl;
success = True;
}
@@ -438,7 +438,7 @@
RTSPClient* rtspClient = (RTSPClient*)subsession->miscPtr;
UsageEnvironment& env = rtspClient->envir(); // alias
- LOG_INFO << *rtspClient << "Received RTCP \"BYE\" on \"" << *subsession << "\" subsession";
+ LOG_INFO << *rtspClient << "Received RTCP \"BYE\" on \"" << *subsession << "\" subsession" << std::endl;
// Now act as if the subsession had closed:
subsessionAfterPlaying(subsession);
@@ -491,7 +491,7 @@
}
}
- LOG_NOTICE << *rtspClient << "Closing the stream.";
+ LOG_NOTICE << *rtspClient << "Closing the stream." << std::endl;
Medium::close(rtspClient);
// Note that this will also cause this stream's "StreamClientState" structure to get reclaimed.
@@ -602,20 +602,20 @@
// We've just received a frame of data. (Optionally) print out information about it:
#ifdef DEBUG_PRINT_EACH_RECEIVED_FRAME
if (fStreamId != NULL)
- LOG_DEBUG << "Stream \"" << fStreamId << "\"; ";
- LOG_DEBUG << "\t" << fSubsession.mediumName() << "/" << fSubsession.codecName() << ":\tReceived " << frameSize << " bytes";
+ LOG_DEBUG << "Stream \"" << fStreamId << "\"; " << std::endl;
+ LOG_DEBUG << "\t" << fSubsession.mediumName() << "/" << fSubsession.codecName() << ":\tReceived " << frameSize << " bytes" << std::endl;
if (numTruncatedBytes > 0)
- LOG_DEBUG << " (with " << numTruncatedBytes << " bytes truncated)";
+ LOG_DEBUG << " (with " << numTruncatedBytes << " bytes truncated)" << std::endl;
char uSecsStr[6+1]; // used to output the 'microseconds' part of the presentation time
sprintf(uSecsStr, "%06u", (unsigned)presentationTime.tv_usec);
- LOG_DEBUG << "\tPresentation time: " << (int)presentationTime.tv_sec << "." << uSecsStr;
+ LOG_DEBUG << "\tPresentation time: " << (int)presentationTime.tv_sec << "." << uSecsStr << std::endl;
if (fSubsession.rtpSource() != NULL && !fSubsession.rtpSource()->hasBeenSynchronizedUsingRTCP())
{
- LOG_DEBUG << "\tPTS not RTCP-synchronized"; // mark the debugging output to indicate that this presentation time is not RTCP-synchronized
+ LOG_DEBUG << "\tPTS not RTCP-synchronized" << std::endl; // mark the debugging output to indicate that this presentation time is not RTCP-synchronized
}
#ifdef DEBUG_PRINT_NPT
- LOG_DEBUG << "\tNPT: " << fSubsession.getNormalPlayTime(presentationTime);
+ LOG_DEBUG << "\tNPT: " << fSubsession.getNormalPlayTime(presentationTime) << std::endl;
#endif
#endif
--
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