From b022b91c0c6fa807424b6c12cc92ac5946838083 Mon Sep 17 00:00:00 2001
From: houxiao <houxiao@454eff88-639b-444f-9e54-f578c98de674>
Date: 星期四, 13 七月 2017 16:34:39 +0800
Subject: [PATCH] update pipeline
---
RtspFace/live555/testProgs/testRTSPClient.hpp | 177 +++++++++++++++++++++++++++++++++++++----------------------
1 files changed, 111 insertions(+), 66 deletions(-)
diff --git a/RtspFace/live555/testProgs/testRTSPClient.hpp b/RtspFace/live555/testProgs/testRTSPClient.hpp
index 75cc022..c1c4765 100644
--- a/RtspFace/live555/testProgs/testRTSPClient.hpp
+++ b/RtspFace/live555/testProgs/testRTSPClient.hpp
@@ -20,8 +20,9 @@
// client application. For a full-featured RTSP client application - with much more functionality, and many options - see
// "openRTSP": http://www.live555.com/openRTSP/
-#include <liveMedia.hh>
-#include <BasicUsageEnvironment.hh>
+#include <liveMedia/liveMedia.hh>
+#include <BasicUsageEnvironment/BasicUsageEnvironment.hh>
+#include <groupsock/GroupsockHelper.hh>
#include <iostream>
@@ -29,7 +30,7 @@
// By default, we request that the server stream its data using RTP/UDP.
// If, instead, you want to request that the server stream via RTP-over-TCP, change the following to True:
-#define REQUEST_STREAMING_OVER_TCP True
+//#define REQUEST_STREAMING_OVER_TCP True
// Even though we're not going to be doing anything with the incoming data, we still need to receive it.
// Define the size of the buffer that we'll use:
@@ -75,8 +76,8 @@
void usage(UsageEnvironment& env, char const* progName)
{
- 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;
+ LOG_DEBUG << "Usage: " << progName << " <rtsp-url-1> ... <rtsp-url-N>" << LOG_ENDL;
+ LOG_DEBUG << "\t(where each <rtsp-url-i> is a \"rtsp://\" URL)" << LOG_ENDL;
}
char eventLoopWatchVariable = 0;
@@ -158,6 +159,7 @@
public:
StreamClientState scs;
const PL_RTSPClient_Config& rtspConfig;
+ int desiredPortNum;
};
// Define a data sink (a subclass of "MediaSink") to receive the data for each subsession (i.e., each audio or video 'substream').
@@ -202,12 +204,38 @@
void openURL(UsageEnvironment& env, const PL_RTSPClient_Config& _rtspConfig)
{
+ if (!_rtspConfig.receivingInterfaceAddr.empty())
+ {
+ NetAddressList addresses(_rtspConfig.receivingInterfaceAddr.c_str());
+ if (addresses.numAddresses() == 0)
+ {
+ LOG_ERROR << "Failed to find network address for " << _rtspConfig.receivingInterfaceAddr << LOG_ENDL;
+ return;
+ }
+ else
+ {
+ ReceivingInterfaceAddr = *(unsigned*)(addresses.firstAddress()->data()); // declared in live555
+ LOG_INFO << "Use receiving interface addr " << _rtspConfig.receivingInterfaceAddr << LOG_ENDL;
+ }
+ }
+
+ if (_rtspConfig.desiredPortNum != 0)
+ {
+ if (_rtspConfig.desiredPortNum <= 0 || _rtspConfig.desiredPortNum >= 65536 || _rtspConfig.desiredPortNum&1)
+ {
+ LOG_ERROR << "bad port number: " << _rtspConfig.desiredPortNum << " (must be even, and in the range (0,65536))" << LOG_ENDL;
+ return;
+ }
+ else
+ LOG_INFO << "Use desired port num " << _rtspConfig.desiredPortNum << LOG_ENDL;
+ }
+
// Begin by creating a "RTSPClient" object. Note that there is a separate "RTSPClient" object for each stream that we wish
// to receive (even if more than stream uses the same "rtsp://" URL).
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() << std::endl;
+ LOG_ERROR << "Failed to create a RTSP client for URL \"" << _rtspConfig.rtspURL.c_str() << "\": " << env.getResultMsg() << LOG_ENDL;
return;
}
@@ -231,25 +259,25 @@
if (resultCode != 0)
{
- LOG_WARN << *rtspClient << "Failed to get a SDP description: " << resultString << std::endl;
+ LOG_WARN << *rtspClient << "Failed to get a SDP description: " << resultString << LOG_ENDL;
delete[] resultString;
break;
}
char* const sdpDescription = resultString;
- LOG_INFO << *rtspClient << "Got a SDP description:\n" << sdpDescription << std::endl;
+ LOG_INFO << *rtspClient << "Got a SDP description:\n" << sdpDescription << LOG_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() << std::endl;
+ LOG_ERROR << *rtspClient << "Failed to create a MediaSession object from the SDP description: " << env.getResultMsg() << LOG_ENDL;
break;
}
else if (!scs.session->hasSubsessions())
{
- LOG_WARN << *rtspClient << "This session has no media subsessions (i.e., no \"m=\" lines)" << std::endl;
+ LOG_WARN << *rtspClient << "This session has no media subsessions (i.e., no \"m=\" lines)" << LOG_ENDL;
break;
}
@@ -269,27 +297,34 @@
void setupNextSubsession(RTSPClient* rtspClient)
{
UsageEnvironment& env = rtspClient->envir(); // alias
- StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
+ ourRTSPClient* _ourRTSPClient = (ourRTSPClient*)rtspClient;
+ StreamClientState& scs = _ourRTSPClient->scs; // alias
scs.subsession = scs.iter->next();
if (scs.subsession != NULL)
{
+ if (_ourRTSPClient->desiredPortNum != 0)
+ {
+ scs.subsession->setClientPortNum(_ourRTSPClient->desiredPortNum);
+ _ourRTSPClient->desiredPortNum += 2;
+ }
+
if (!scs.subsession->initiate())
{
- LOG_ERROR << *rtspClient << "Failed to initiate the \"" << *scs.subsession << "\" subsession: " << env.getResultMsg() << std::endl;
+ LOG_ERROR << *rtspClient << "Failed to initiate the \"" << *scs.subsession << "\" subsession: " << env.getResultMsg() << LOG_ENDL;
setupNextSubsession(rtspClient); // give up on this subsession; go to the next one
}
else
{
- LOG_INFO << *rtspClient << "Initiated the \"" << *scs.subsession << "\" subsession (" << std::endl;
+ LOG_INFO << *rtspClient << "Initiated the \"" << *scs.subsession << "\" subsession (" << LOG_ENDL;
if (scs.subsession->rtcpIsMuxed())
- LOG_INFO << "client port " << scs.subsession->clientPortNum() << std::endl;
+ LOG_INFO << "client port " << scs.subsession->clientPortNum() << LOG_ENDL;
else
- LOG_INFO << "client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum()+1 << std::endl;
- LOG_INFO << ")" << std::endl;
+ LOG_INFO << "client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum()+1 << LOG_ENDL;
+ LOG_INFO << ")" << LOG_ENDL;
// Continue setting up this subsession, by sending a RTSP "SETUP" command:
- rtspClient->sendSetupCommand(*scs.subsession, continueAfterSETUP, False, REQUEST_STREAMING_OVER_TCP);
+ rtspClient->sendSetupCommand(*scs.subsession, continueAfterSETUP, False, _ourRTSPClient->rtspConfig.requestStreamingOverTcp);
}
return;
}
@@ -316,44 +351,48 @@
if (resultCode != 0)
{
- LOG_ERROR << *rtspClient << "Failed to set up the \"" << *scs.subsession << "\" subsession: " << resultString << std::endl;
+ LOG_ERROR << *rtspClient << "Failed to set up the \"" << *scs.subsession << "\" subsession: " << resultString << LOG_ENDL;
break;
}
- LOG_INFO << *rtspClient << "Set up the \"" << *scs.subsession << "\" subsession (" << std::endl;
+ //#todo temp usage
+ std::string sess_mime(scs.subsession->mediumName());
+ if (sess_mime != "video")
+ break;
+
+ LOG_INFO << *rtspClient << "Set up the \"" << *scs.subsession << "\" subsession (" << LOG_ENDL;
if (scs.subsession->rtcpIsMuxed())
- {
- LOG_INFO << "client port " << scs.subsession->clientPortNum() << std::endl;
- }
+ {
+ LOG_INFO << "client port " << scs.subsession->clientPortNum() << LOG_ENDL;
+ }
else
- {
- LOG_INFO << "client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum()+1 << std::endl;
- }
- LOG_INFO << ")" << std::endl;
+ {
+ LOG_INFO << "client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum()+1 << LOG_ENDL;
+ }
+ LOG_INFO << ")" << LOG_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,
// after we've sent a RTSP "PLAY" command.)
- scs.subsession->sink = DummySink::createNew(env, ((ourRTSPClient*)rtspClient)->rtspConfig,
- *scs.subsession, rtspClient->url());
+ scs.subsession->sink = DummySink::createNew(env, ((ourRTSPClient*)rtspClient)->rtspConfig, *scs.subsession, rtspClient->url());
// perhaps use your own custom "MediaSink" subclass instead
if (scs.subsession->sink == NULL)
- {
- LOG_ERROR << *rtspClient << "Failed to create a data sink for the \"" << *scs.subsession
- << "\" subsession: " << env.getResultMsg() << std::endl;
- break;
- }
+ {
+ LOG_ERROR << *rtspClient << "Failed to create a data sink for the \"" << *scs.subsession
+ << "\" subsession: " << env.getResultMsg() << LOG_ENDL;
+ break;
+ }
- LOG_INFO << *rtspClient << "Created a data sink for the \"" << *scs.subsession << "\" subsession" << std::endl;
+ LOG_INFO << *rtspClient << "Created a data sink for the \"" << *scs.subsession << "\" subsession" << LOG_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);
+ Boolean startPlayingRet = scs.subsession->sink->startPlaying(*(scs.subsession->readSource()), subsessionAfterPlaying, scs.subsession);
+ LOG_INFO << "startPlayingRet=" << (bool)startPlayingRet << LOG_ENDL;
// Also set a handler to be called if a RTCP "BYE" arrives for this subsession:
if (scs.subsession->rtcpInstance() != NULL)
- {
- scs.subsession->rtcpInstance()->setByeHandler(subsessionByeHandler, scs.subsession);
- }
+ {
+ scs.subsession->rtcpInstance()->setByeHandler(subsessionByeHandler, scs.subsession);
+ }
}
while (0);
delete[] resultString;
@@ -373,7 +412,7 @@
if (resultCode != 0)
{
- LOG_ERROR << *rtspClient << "Failed to start playing session: " << resultString << std::endl;
+ LOG_ERROR << *rtspClient << "Failed to start playing session: " << resultString << LOG_ENDL;
break;
}
@@ -389,12 +428,12 @@
scs.streamTimerTask = env.taskScheduler().scheduleDelayedTask(uSecsToDelay, (TaskFunc*)streamTimerHandler, rtspClient);
}
- LOG_INFO << *rtspClient << "Started playing session" << std::endl;
+ LOG_INFO << *rtspClient << "Started playing session" << LOG_ENDL;
if (scs.duration > 0)
{
- LOG_INFO << " (for up to " << scs.duration << " seconds)" << std::endl;
+ LOG_INFO << " (for up to " << scs.duration << " seconds)" << LOG_ENDL;
}
- LOG_INFO << "..." << std::endl;
+ LOG_INFO << "..." << LOG_ENDL;
success = True;
}
@@ -438,7 +477,7 @@
RTSPClient* rtspClient = (RTSPClient*)subsession->miscPtr;
UsageEnvironment& env = rtspClient->envir(); // alias
- LOG_INFO << *rtspClient << "Received RTCP \"BYE\" on \"" << *subsession << "\" subsession" << std::endl;
+ LOG_INFO << *rtspClient << "Received RTCP \"BYE\" on \"" << *subsession << "\" subsession" << LOG_ENDL;
// Now act as if the subsession had closed:
subsessionAfterPlaying(subsession);
@@ -491,7 +530,7 @@
}
}
- LOG_NOTICE << *rtspClient << "Closing the stream." << std::endl;
+ LOG_NOTICE << *rtspClient << "Closing the stream." << LOG_ENDL;
Medium::close(rtspClient);
// Note that this will also cause this stream's "StreamClientState" structure to get reclaimed.
@@ -500,7 +539,8 @@
// The final stream has ended, so exit the application now.
// (Of course, if you're embedding this code into your own application, you might want to comment this out,
// and replace it with "eventLoopWatchVariable = 1;", so that we leave the LIVE555 event loop, and continue running "main()".)
- exit(exitCode);
+ //exit(exitCode);
+ eventLoopWatchVariable = 1;
}
}
@@ -514,7 +554,7 @@
ourRTSPClient::ourRTSPClient(UsageEnvironment& env, const PL_RTSPClient_Config& _rtspConfig)
: RTSPClient(env, _rtspConfig.rtspURL.c_str(), _rtspConfig.verbosityLevel, _rtspConfig.progName.c_str(),
- _rtspConfig.tunnelOverHTTPPortNum, -1), rtspConfig(_rtspConfig)
+ _rtspConfig.tunnelOverHTTPPortNum, -1), scs(), rtspConfig(_rtspConfig), desiredPortNum(_rtspConfig.desiredPortNum)
{
}
@@ -526,7 +566,7 @@
// Implementation of "StreamClientState":
StreamClientState::StreamClientState()
- : iter(NULL), session(NULL), subsession(NULL), streamTimerTask(NULL), duration(0.0)
+ : iter(NULL), session(NULL), subsession(NULL), streamTimerTask(), duration(0.0)
{
}
@@ -558,20 +598,25 @@
// ffmpeg need AUX header
if (rtspConfig.aux)
- {
- fReceiveBuffer[0]=0x00;
- fReceiveBuffer[1]=0x00;
- fReceiveBuffer[2]=0x00;
- fReceiveBuffer[3]=0x01;
- }
+ {
+ fReceiveBuffer[0]=0x00;
+ fReceiveBuffer[1]=0x00;
+ fReceiveBuffer[2]=0x00;
+ fReceiveBuffer[3]=0x01;
+ }
+ RtspClientParam param;
+
//parse sdp
- const char* strSDP = fSubsession.savedSDPLines();
- rtsp_client_sdp_callback(rtspConfig.args, strSDP);
-
- const char* strFmtp = fSubsession.fmtp_spropparametersets();
- rtsp_client_fmtp_callback(rtspConfig.args, strFmtp);
- //std::cout << strFmtp << std::endl;
+ param.sdp = fSubsession.savedSDPLines();
+ param.fmtp = fSubsession.fmtp_spropparametersets();
+ param.width = fSubsession.videoWidth();
+ param.height = fSubsession.videoHeight();
+ param.fps = fSubsession.videoFPS();
+ param.codecName = fSubsession.codecName();
+ param.bandwidth = fSubsession.bandwidth();
+
+ rtsp_client_set_param_callback(rtspConfig.args, param);
}
DummySink::~DummySink()
@@ -602,20 +647,20 @@
// 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 << "\"; " << std::endl;
- LOG_DEBUG << "\t" << fSubsession.mediumName() << "/" << fSubsession.codecName() << ":\tReceived " << frameSize << " bytes" << std::endl;
+ LOG_DEBUG << "Stream \"" << fStreamId << "\"; " << LOG_ENDL;
+ LOG_DEBUG << "\t" << fSubsession.mediumName() << "/" << fSubsession.codecName() << ":\tReceived " << frameSize << " bytes" << LOG_ENDL;
if (numTruncatedBytes > 0)
- LOG_DEBUG << " (with " << numTruncatedBytes << " bytes truncated)" << std::endl;
+ LOG_DEBUG << " (with " << numTruncatedBytes << " bytes truncated)" << LOG_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 << std::endl;
+ LOG_DEBUG << "\tPresentation time: " << (int)presentationTime.tv_sec << "." << uSecsStr << LOG_ENDL;
if (fSubsession.rtpSource() != NULL && !fSubsession.rtpSource()->hasBeenSynchronizedUsingRTCP())
{
- LOG_DEBUG << "\tPTS not RTCP-synchronized" << std::endl; // mark the debugging output to indicate that this presentation time is not RTCP-synchronized
+ LOG_DEBUG << "\tPTS not RTCP-synchronized" << LOG_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) << std::endl;
+ LOG_DEBUG << "\tNPT: " << fSubsession.getNormalPlayTime(presentationTime) << LOG_ENDL;
#endif
#endif
--
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