From cc445067d1f61e12dbea4e6458f2c85ba58f01bf Mon Sep 17 00:00:00 2001
From: houxiao <houxiao@454eff88-639b-444f-9e54-f578c98de674>
Date: 星期五, 30 十二月 2016 14:28:14 +0800
Subject: [PATCH] fix config, fix some log and todo
---
RtspFace/live555/testProgs/testRTSPClient.hpp | 702 +++++++++++++++++++++++++++++++--------------------------
1 files changed, 379 insertions(+), 323 deletions(-)
diff --git a/RtspFace/live555/testProgs/testRTSPClient.hpp b/RtspFace/live555/testProgs/testRTSPClient.hpp
index 8f33f1a..ff4a861 100644
--- a/RtspFace/live555/testProgs/testRTSPClient.hpp
+++ b/RtspFace/live555/testProgs/testRTSPClient.hpp
@@ -33,10 +33,11 @@
// 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:
-#define DUMMY_SINK_RECEIVE_BUFFER_SIZE 1920*1080*3
+#define DUMMY_SINK_RECEIVE_BUFFER_SIZE 1920*1080*3//#todo
// If you don't want to see debugging output for each received frame, then comment out the following line:
-#define DEBUG_PRINT_EACH_RECEIVED_FRAME 1
+//#define DEBUG_PRINT_EACH_RECEIVED_FRAME 1
+//#define DEBUG_PRINT_NPT 1
// Forward function definitions:
@@ -49,10 +50,10 @@
void subsessionAfterPlaying(void* clientData); // called when a stream's subsession (e.g., audio or video substream) ends
void subsessionByeHandler(void* clientData); // called when a RTCP "BYE" is received for a subsession
void streamTimerHandler(void* clientData);
- // called at the end of a stream's expected duration (if the stream has not already signaled its end using a RTCP "BYE")
+// called at the end of a stream's expected duration (if the stream has not already signaled its end using a RTCP "BYE")
// The main streaming routine (for each "rtsp://" URL):
-void openURL(UsageEnvironment& env, const RTSPConfig& _rtspConfig);
+void openURL(UsageEnvironment& env, const PL_RTSPClient_Config& _rtspConfig);
// Used to iterate through each stream's 'subsessions', setting up each one:
void setupNextSubsession(RTSPClient* rtspClient);
@@ -61,75 +62,82 @@
void shutdownStream(RTSPClient* rtspClient, int exitCode = 1);
// A function that outputs a string that identifies each stream (for debugging output). Modify this if you wish:
-UsageEnvironment& operator<<(UsageEnvironment& env, const RTSPClient& rtspClient) {
- return env << "[URL:\"" << rtspClient.url() << "\"]: ";
+log4cpp::CategoryStream& operator<<(log4cpp::CategoryStream& logRoot, const RTSPClient& rtspClient)
+{
+ return logRoot << "[URL:\"" << rtspClient.url() << "\"]: ";
}
// A function that outputs a string that identifies each subsession (for debugging output). Modify this if you wish:
-UsageEnvironment& operator<<(UsageEnvironment& env, const MediaSubsession& subsession) {
- return env << subsession.mediumName() << "/" << subsession.codecName();
+log4cpp::CategoryStream& operator<<(log4cpp::CategoryStream& logRoot, const MediaSubsession& subsession)
+{
+ return logRoot << subsession.mediumName() << "/" << subsession.codecName();
}
-void usage(UsageEnvironment& env, char const* progName) {
- env << "Usage: " << progName << " <rtsp-url-1> ... <rtsp-url-N>\n";
- env << "\t(where each <rtsp-url-i> is a \"rtsp://\" URL)\n";
+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)";
}
char eventLoopWatchVariable = 0;
-int test_main(int argc, char** argv) {
- // Begin by setting up our usage environment:
- TaskScheduler* scheduler = BasicTaskScheduler::createNew();
- UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
+int test_main(int argc, char** argv)
+{
+ // Begin by setting up our usage environment:
+ TaskScheduler* scheduler = BasicTaskScheduler::createNew();
+ UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
- // We need at least one "rtsp://" URL argument:
- if (argc < 2) {
- usage(*env, argv[0]);
- return 1;
- }
+ // We need at least one "rtsp://" URL argument:
+ if (argc < 2)
+ {
+ usage(*env, argv[0]);
+ return 1;
+ }
- RTSPConfig rtspConfig;
+ PL_RTSPClient_Config rtspConfig;
rtspConfig.progName = argv[0];
rtspConfig.rtspURL = "";
rtspConfig.aux = false;
rtspConfig.verbosityLevel = RTSP_CLIENT_VERBOSITY_LEVEL;
rtspConfig.tunnelOverHTTPPortNum = 0;
rtspConfig.args = nullptr;
-
- // There are argc-1 URLs: argv[1] through argv[argc-1]. Open and start streaming each one:
- for (int i = 1; i <= argc-1; ++i) {
- rtspConfig.rtspURL = argv[i];
- openURL(*env, rtspConfig);
- }
- // All subsequent activity takes place within the event loop:
- env->taskScheduler().doEventLoop(&eventLoopWatchVariable);
- // This function call does not return, unless, at some point in time, "eventLoopWatchVariable" gets set to something non-zero.
+ // There are argc-1 URLs: argv[1] through argv[argc-1]. Open and start streaming each one:
+ for (int i = 1; i <= argc-1; ++i)
+ {
+ rtspConfig.rtspURL = argv[i];
+ openURL(*env, rtspConfig);
+ }
- return 0;
+ // All subsequent activity takes place within the event loop:
+ env->taskScheduler().doEventLoop(&eventLoopWatchVariable);
+ // This function call does not return, unless, at some point in time, "eventLoopWatchVariable" gets set to something non-zero.
- // If you choose to continue the application past this point (i.e., if you comment out the "return 0;" statement above),
- // and if you don't intend to do anything more with the "TaskScheduler" and "UsageEnvironment" objects,
- // then you can also reclaim the (small) memory used by these objects by uncommenting the following code:
- /*
- env->reclaim(); env = NULL;
- delete scheduler; scheduler = NULL;
- */
+ return 0;
+
+ // If you choose to continue the application past this point (i.e., if you comment out the "return 0;" statement above),
+ // and if you don't intend to do anything more with the "TaskScheduler" and "UsageEnvironment" objects,
+ // then you can also reclaim the (small) memory used by these objects by uncommenting the following code:
+ /*
+ env->reclaim(); env = NULL;
+ delete scheduler; scheduler = NULL;
+ */
}
// Define a class to hold per-stream state that we maintain throughout each stream's lifetime:
-class StreamClientState {
+class StreamClientState
+{
public:
- StreamClientState();
- virtual ~StreamClientState();
+ StreamClientState();
+ virtual ~StreamClientState();
public:
- MediaSubsessionIterator* iter;
- MediaSession* session;
- MediaSubsession* subsession;
- TaskToken streamTimerTask;
- double duration;
+ MediaSubsessionIterator* iter;
+ MediaSession* session;
+ MediaSubsession* subsession;
+ TaskToken streamTimerTask;
+ double duration;
};
// If you're streaming just a single stream (i.e., just from a single URL, once), then you can define and use just a single
@@ -137,18 +145,19 @@
// showing how to play multiple streams, concurrently, we can't do that. Instead, we have to have a separate "StreamClientState"
// structure for each "RTSPClient". To do this, we subclass "RTSPClient", and add a "StreamClientState" field to the subclass:
-class ourRTSPClient: public RTSPClient {
+class ourRTSPClient: public RTSPClient
+{
public:
- static ourRTSPClient* createNew(UsageEnvironment& env, const RTSPConfig& _rtspConfig);
+ static ourRTSPClient* createNew(UsageEnvironment& env, const PL_RTSPClient_Config& _rtspConfig);
protected:
- ourRTSPClient(UsageEnvironment& env, const RTSPConfig& _rtspConfig);
- // called only by createNew();
- virtual ~ourRTSPClient();
+ ourRTSPClient(UsageEnvironment& env, const PL_RTSPClient_Config& _rtspConfig);
+ // called only by createNew();
+ virtual ~ourRTSPClient();
public:
- StreamClientState scs;
- const RTSPConfig& rtspConfig;
+ StreamClientState scs;
+ const PL_RTSPClient_Config& rtspConfig;
};
// Define a data sink (a subclass of "MediaSink") to receive the data for each subsession (i.e., each audio or video 'substream').
@@ -159,25 +168,25 @@
class DummySink: public MediaSink
{
public:
- static DummySink* createNew(UsageEnvironment& env,
- const RTSPConfig& _rtspConfig,
- MediaSubsession& subsession, // identifies the kind of data that's being received
- char const* streamId = NULL); // identifies the stream itself (optional)
+ static DummySink* createNew(UsageEnvironment& env,
+ const PL_RTSPClient_Config& _rtspConfig,
+ MediaSubsession& subsession, // identifies the kind of data that's being received
+ char const* streamId = NULL); // identifies the stream itself (optional)
private:
- DummySink(UsageEnvironment& env, const RTSPConfig& _rtspConfig, MediaSubsession& subsession, char const* streamId);
+ DummySink(UsageEnvironment& env, const PL_RTSPClient_Config& _rtspConfig, MediaSubsession& subsession, char const* streamId);
// called only by "createNew()"
virtual ~DummySink();
static void afterGettingFrame(void* clientData, unsigned frameSize,
- unsigned numTruncatedBytes,
- struct timeval presentationTime,
- unsigned durationInMicroseconds);
+ unsigned numTruncatedBytes,
+ struct timeval presentationTime,
+ unsigned durationInMicroseconds);
void afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes,
- struct timeval presentationTime, unsigned durationInMicroseconds);
+ struct timeval presentationTime, unsigned durationInMicroseconds);
public:
- const RTSPConfig& rtspConfig;
+ const PL_RTSPClient_Config& rtspConfig;
private:
// redefined virtual functions:
@@ -191,23 +200,23 @@
static unsigned rtspClientCount = 0; // Counts how many streams (i.e., "RTSPClient"s) are currently in use.
-void openURL(UsageEnvironment& env, const RTSPConfig& _rtspConfig)
+void openURL(UsageEnvironment& env, const PL_RTSPClient_Config& _rtspConfig)
{
// 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)
- {
- env << "Failed to create a RTSP client for URL \"" << _rtspConfig.rtspURL.c_str() << "\": " << env.getResultMsg() << "\n";
- return;
- }
+ {
+ LOG_ERROR << "Failed to create a RTSP client for URL \"" << _rtspConfig.rtspURL.c_str() << "\": " << env.getResultMsg();
+ return;
+ }
++rtspClientCount;
// Next, send a RTSP "DESCRIBE" command, to get a SDP description for the stream.
// Note that this command - like all RTSP commands - is sent asynchronously; we do not block, waiting for a response.
// Instead, the following function call returns immediately, and we handle the RTSP response later, from within the event loop:
- rtspClient->sendDescribeCommand(continueAfterDESCRIBE);
+ rtspClient->sendDescribeCommand(continueAfterDESCRIBE);
}
@@ -216,41 +225,42 @@
void continueAfterDESCRIBE(RTSPClient* rtspClient, int resultCode, char* resultString)
{
do
- {
- UsageEnvironment& env = rtspClient->envir(); // alias
- StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
-
- if (resultCode != 0)
{
- env << *rtspClient << "Failed to get a SDP description: " << resultString << "\n";
- delete[] resultString;
- break;
- }
+ UsageEnvironment& env = rtspClient->envir(); // alias
+ StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
- char* const sdpDescription = resultString;
- env << *rtspClient << "Got a SDP description:\n" << sdpDescription << "\n";
+ if (resultCode != 0)
+ {
+ LOG_WARN << *rtspClient << "Failed to get a SDP description: " << resultString;
+ delete[] resultString;
+ break;
+ }
- // 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)
- {
- env << *rtspClient << "Failed to create a MediaSession object from the SDP description: " << env.getResultMsg() << "\n";
- break;
- }
- else if (!scs.session->hasSubsessions())
- {
- env << *rtspClient << "This session has no media subsessions (i.e., no \"m=\" lines)\n";
- break;
- }
+ char* const sdpDescription = resultString;
+ LOG_INFO << *rtspClient << "Got a SDP description:\n" << sdpDescription;
- // Then, create and set up our data source objects for the session. We do this by iterating over the session's 'subsessions',
- // calling "MediaSubsession::initiate()", and then sending a RTSP "SETUP" command, on each one.
- // (Each 'subsession' will have its own data source.)
- scs.iter = new MediaSubsessionIterator(*scs.session);
- setupNextSubsession(rtspClient);
- return;
- } while (0);
+ // 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();
+ break;
+ }
+ else if (!scs.session->hasSubsessions())
+ {
+ LOG_WARN << *rtspClient << "This session has no media subsessions (i.e., no \"m=\" lines)";
+ break;
+ }
+
+ // Then, create and set up our data source objects for the session. We do this by iterating over the session's 'subsessions',
+ // calling "MediaSubsession::initiate()", and then sending a RTSP "SETUP" command, on each one.
+ // (Each 'subsession' will have its own data source.)
+ scs.iter = new MediaSubsessionIterator(*scs.session);
+ setupNextSubsession(rtspClient);
+ return;
+ }
+ while (0);
// An unrecoverable error occurred with this stream.
shutdownStream(rtspClient);
@@ -258,319 +268,365 @@
void setupNextSubsession(RTSPClient* rtspClient)
{
- UsageEnvironment& env = rtspClient->envir(); // alias
- StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
-
- scs.subsession = scs.iter->next();
- if (scs.subsession != NULL) {
- if (!scs.subsession->initiate()) {
- env << *rtspClient << "Failed to initiate the \"" << *scs.subsession << "\" subsession: " << env.getResultMsg() << "\n";
- setupNextSubsession(rtspClient); // give up on this subsession; go to the next one
- } else {
- env << *rtspClient << "Initiated the \"" << *scs.subsession << "\" subsession (";
- if (scs.subsession->rtcpIsMuxed()) {
- env << "client port " << scs.subsession->clientPortNum();
- } else {
- env << "client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum()+1;
- }
- env << ")\n";
+ UsageEnvironment& env = rtspClient->envir(); // alias
+ StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
- // Continue setting up this subsession, by sending a RTSP "SETUP" command:
- rtspClient->sendSetupCommand(*scs.subsession, continueAfterSETUP, False, REQUEST_STREAMING_OVER_TCP);
- }
- return;
- }
+ scs.subsession = scs.iter->next();
+ if (scs.subsession != NULL)
+ {
+ if (!scs.subsession->initiate())
+ {
+ LOG_ERROR << *rtspClient << "Failed to initiate the \"" << *scs.subsession << "\" subsession: " << env.getResultMsg();
+ setupNextSubsession(rtspClient); // give up on this subsession; go to the next one
+ }
+ else
+ {
+ LOG_INFO << *rtspClient << "Initiated the \"" << *scs.subsession << "\" subsession (";
+ if (scs.subsession->rtcpIsMuxed())
+ LOG_INFO << "client port " << scs.subsession->clientPortNum();
+ else
+ LOG_INFO << "client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum()+1;
+ LOG_INFO << ")";
- // We've finished setting up all of the subsessions. Now, send a RTSP "PLAY" command to start the streaming:
- if (scs.session->absStartTime() != NULL) {
- // Special case: The stream is indexed by 'absolute' time, so send an appropriate "PLAY" command:
- rtspClient->sendPlayCommand(*scs.session, continueAfterPLAY, scs.session->absStartTime(), scs.session->absEndTime());
- } else {
- scs.duration = scs.session->playEndTime() - scs.session->playStartTime();
- rtspClient->sendPlayCommand(*scs.session, continueAfterPLAY);
- }
+ // Continue setting up this subsession, by sending a RTSP "SETUP" command:
+ rtspClient->sendSetupCommand(*scs.subsession, continueAfterSETUP, False, REQUEST_STREAMING_OVER_TCP);
+ }
+ return;
+ }
+
+ // We've finished setting up all of the subsessions. Now, send a RTSP "PLAY" command to start the streaming:
+ if (scs.session->absStartTime() != NULL)
+ {
+ // Special case: The stream is indexed by 'absolute' time, so send an appropriate "PLAY" command:
+ rtspClient->sendPlayCommand(*scs.session, continueAfterPLAY, scs.session->absStartTime(), scs.session->absEndTime());
+ }
+ else
+ {
+ scs.duration = scs.session->playEndTime() - scs.session->playStartTime();
+ rtspClient->sendPlayCommand(*scs.session, continueAfterPLAY);
+ }
}
-void continueAfterSETUP(RTSPClient* rtspClient, int resultCode, char* resultString) {
- do {
- UsageEnvironment& env = rtspClient->envir(); // alias
- StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
+void continueAfterSETUP(RTSPClient* rtspClient, int resultCode, char* resultString)
+{
+ do
+ {
+ UsageEnvironment& env = rtspClient->envir(); // alias
+ StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
- if (resultCode != 0) {
- env << *rtspClient << "Failed to set up the \"" << *scs.subsession << "\" subsession: " << resultString << "\n";
- break;
- }
+ if (resultCode != 0)
+ {
+ LOG_ERROR << *rtspClient << "Failed to set up the \"" << *scs.subsession << "\" subsession: " << resultString;
+ break;
+ }
- env << *rtspClient << "Set up the \"" << *scs.subsession << "\" subsession (";
- if (scs.subsession->rtcpIsMuxed()) {
- env << "client port " << scs.subsession->clientPortNum();
- } else {
- env << "client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum()+1;
- }
- env << ")\n";
+ LOG_INFO << *rtspClient << "Set up the \"" << *scs.subsession << "\" subsession (";
+ if (scs.subsession->rtcpIsMuxed())
+ {
+ LOG_INFO << "client port " << scs.subsession->clientPortNum();
+ }
+ else
+ {
+ LOG_INFO << "client ports " << scs.subsession->clientPortNum() << "-" << scs.subsession->clientPortNum()+1;
+ }
+ LOG_INFO << ")";
- // 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.)
+ // 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());
- // perhaps use your own custom "MediaSink" subclass instead
- if (scs.subsession->sink == NULL) {
- env << *rtspClient << "Failed to create a data sink for the \"" << *scs.subsession
- << "\" subsession: " << env.getResultMsg() << "\n";
- break;
- }
+ 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();
+ break;
+ }
- env << *rtspClient << "Created a data sink for the \"" << *scs.subsession << "\" subsession\n";
- 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);
- // 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);
- }
- } while (0);
- delete[] resultString;
+ LOG_INFO << *rtspClient << "Created a data sink for the \"" << *scs.subsession << "\" subsession";
+ 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);
+ // 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);
+ }
+ }
+ while (0);
+ delete[] resultString;
- // Set up the next subsession, if any:
- setupNextSubsession(rtspClient);
+ // Set up the next subsession, if any:
+ setupNextSubsession(rtspClient);
}
-void continueAfterPLAY(RTSPClient* rtspClient, int resultCode, char* resultString) {
- Boolean success = False;
+void continueAfterPLAY(RTSPClient* rtspClient, int resultCode, char* resultString)
+{
+ Boolean success = False;
- do {
- UsageEnvironment& env = rtspClient->envir(); // alias
- StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
+ do
+ {
+ UsageEnvironment& env = rtspClient->envir(); // alias
+ StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
- if (resultCode != 0) {
- env << *rtspClient << "Failed to start playing session: " << resultString << "\n";
- break;
- }
+ if (resultCode != 0)
+ {
+ LOG_ERROR << *rtspClient << "Failed to start playing session: " << resultString;
+ break;
+ }
- // Set a timer to be handled at the end of the stream's expected duration (if the stream does not already signal its end
- // using a RTCP "BYE"). This is optional. If, instead, you want to keep the stream active - e.g., so you can later
- // 'seek' back within it and do another RTSP "PLAY" - then you can omit this code.
- // (Alternatively, if you don't want to receive the entire stream, you could set this timer for some shorter value.)
- if (scs.duration > 0) {
- unsigned const delaySlop = 2; // number of seconds extra to delay, after the stream's expected duration. (This is optional.)
- scs.duration += delaySlop;
- unsigned uSecsToDelay = (unsigned)(scs.duration*1000000);
- scs.streamTimerTask = env.taskScheduler().scheduleDelayedTask(uSecsToDelay, (TaskFunc*)streamTimerHandler, rtspClient);
- }
+ // Set a timer to be handled at the end of the stream's expected duration (if the stream does not already signal its end
+ // using a RTCP "BYE"). This is optional. If, instead, you want to keep the stream active - e.g., so you can later
+ // 'seek' back within it and do another RTSP "PLAY" - then you can omit this code.
+ // (Alternatively, if you don't want to receive the entire stream, you could set this timer for some shorter value.)
+ if (scs.duration > 0)
+ {
+ unsigned const delaySlop = 2; // number of seconds extra to delay, after the stream's expected duration. (This is optional.)
+ scs.duration += delaySlop;
+ unsigned uSecsToDelay = (unsigned)(scs.duration*1000000);
+ scs.streamTimerTask = env.taskScheduler().scheduleDelayedTask(uSecsToDelay, (TaskFunc*)streamTimerHandler, rtspClient);
+ }
- env << *rtspClient << "Started playing session";
- if (scs.duration > 0) {
- env << " (for up to " << scs.duration << " seconds)";
- }
- env << "...\n";
+ LOG_INFO << *rtspClient << "Started playing session";
+ if (scs.duration > 0)
+ {
+ LOG_INFO << " (for up to " << scs.duration << " seconds)";
+ }
+ LOG_INFO << "...";
- success = True;
- } while (0);
- delete[] resultString;
+ success = True;
+ }
+ while (0);
+ delete[] resultString;
- if (!success) {
- // An unrecoverable error occurred with this stream.
- shutdownStream(rtspClient);
- }
+ if (!success)
+ {
+ // An unrecoverable error occurred with this stream.
+ shutdownStream(rtspClient);
+ }
}
// Implementation of the other event handlers:
-void subsessionAfterPlaying(void* clientData) {
- MediaSubsession* subsession = (MediaSubsession*)clientData;
- RTSPClient* rtspClient = (RTSPClient*)(subsession->miscPtr);
+void subsessionAfterPlaying(void* clientData)
+{
+ MediaSubsession* subsession = (MediaSubsession*)clientData;
+ RTSPClient* rtspClient = (RTSPClient*)(subsession->miscPtr);
- // Begin by closing this subsession's stream:
- Medium::close(subsession->sink);
- subsession->sink = NULL;
-
- // Next, check whether *all* subsessions' streams have now been closed:
- MediaSession& session = subsession->parentSession();
- MediaSubsessionIterator iter(session);
- while ((subsession = iter.next()) != NULL) {
- if (subsession->sink != NULL) return; // this subsession is still active
- }
-
- // All subsessions' streams have now been closed, so shutdown the client:
- shutdownStream(rtspClient);
-}
-
-void subsessionByeHandler(void* clientData) {
- MediaSubsession* subsession = (MediaSubsession*)clientData;
- RTSPClient* rtspClient = (RTSPClient*)subsession->miscPtr;
- UsageEnvironment& env = rtspClient->envir(); // alias
-
- env << *rtspClient << "Received RTCP \"BYE\" on \"" << *subsession << "\" subsession\n";
-
- // Now act as if the subsession had closed:
- subsessionAfterPlaying(subsession);
-}
-
-void streamTimerHandler(void* clientData) {
- ourRTSPClient* rtspClient = (ourRTSPClient*)clientData;
- StreamClientState& scs = rtspClient->scs; // alias
-
- scs.streamTimerTask = NULL;
-
- // Shut down the stream:
- shutdownStream(rtspClient);
-}
-
-void shutdownStream(RTSPClient* rtspClient, int exitCode) {
- UsageEnvironment& env = rtspClient->envir(); // alias
- StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
-
- // First, check whether any subsessions have still to be closed:
- if (scs.session != NULL) {
- Boolean someSubsessionsWereActive = False;
- MediaSubsessionIterator iter(*scs.session);
- MediaSubsession* subsession;
-
- while ((subsession = iter.next()) != NULL) {
- if (subsession->sink != NULL) {
+ // Begin by closing this subsession's stream:
Medium::close(subsession->sink);
subsession->sink = NULL;
- if (subsession->rtcpInstance() != NULL) {
- subsession->rtcpInstance()->setByeHandler(NULL, NULL); // in case the server sends a RTCP "BYE" while handling "TEARDOWN"
- }
+ // Next, check whether *all* subsessions' streams have now been closed:
+ MediaSession& session = subsession->parentSession();
+ MediaSubsessionIterator iter(session);
+ while ((subsession = iter.next()) != NULL)
+ {
+ if (subsession->sink != NULL) return; // this subsession is still active
+ }
- someSubsessionsWereActive = True;
- }
- }
+ // All subsessions' streams have now been closed, so shutdown the client:
+ shutdownStream(rtspClient);
+}
- if (someSubsessionsWereActive) {
- // Send a RTSP "TEARDOWN" command, to tell the server to shutdown the stream.
- // Don't bother handling the response to the "TEARDOWN".
- rtspClient->sendTeardownCommand(*scs.session, NULL);
- }
- }
+void subsessionByeHandler(void* clientData)
+{
+ MediaSubsession* subsession = (MediaSubsession*)clientData;
+ RTSPClient* rtspClient = (RTSPClient*)subsession->miscPtr;
+ UsageEnvironment& env = rtspClient->envir(); // alias
- env << *rtspClient << "Closing the stream.\n";
- Medium::close(rtspClient);
- // Note that this will also cause this stream's "StreamClientState" structure to get reclaimed.
+ LOG_INFO << *rtspClient << "Received RTCP \"BYE\" on \"" << *subsession << "\" subsession";
- if (--rtspClientCount == 0) {
- // 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);
- }
+ // Now act as if the subsession had closed:
+ subsessionAfterPlaying(subsession);
+}
+
+void streamTimerHandler(void* clientData)
+{
+ ourRTSPClient* rtspClient = (ourRTSPClient*)clientData;
+ StreamClientState& scs = rtspClient->scs; // alias
+
+ scs.streamTimerTask = NULL;
+
+ // Shut down the stream:
+ shutdownStream(rtspClient);
+}
+
+void shutdownStream(RTSPClient* rtspClient, int exitCode)
+{
+ UsageEnvironment& env = rtspClient->envir(); // alias
+ StreamClientState& scs = ((ourRTSPClient*)rtspClient)->scs; // alias
+
+ // First, check whether any subsessions have still to be closed:
+ if (scs.session != NULL)
+ {
+ Boolean someSubsessionsWereActive = False;
+ MediaSubsessionIterator iter(*scs.session);
+ MediaSubsession* subsession;
+
+ while ((subsession = iter.next()) != NULL)
+ {
+ if (subsession->sink != NULL)
+ {
+ Medium::close(subsession->sink);
+ subsession->sink = NULL;
+
+ if (subsession->rtcpInstance() != NULL)
+ {
+ subsession->rtcpInstance()->setByeHandler(NULL, NULL); // in case the server sends a RTCP "BYE" while handling "TEARDOWN"
+ }
+
+ someSubsessionsWereActive = True;
+ }
+ }
+
+ if (someSubsessionsWereActive)
+ {
+ // Send a RTSP "TEARDOWN" command, to tell the server to shutdown the stream.
+ // Don't bother handling the response to the "TEARDOWN".
+ rtspClient->sendTeardownCommand(*scs.session, NULL);
+ }
+ }
+
+ LOG_NOTICE << *rtspClient << "Closing the stream.";
+ Medium::close(rtspClient);
+ // Note that this will also cause this stream's "StreamClientState" structure to get reclaimed.
+
+ if (--rtspClientCount == 0)
+ {
+ // 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);
+ }
}
// Implementation of "ourRTSPClient":
-ourRTSPClient* ourRTSPClient::createNew(UsageEnvironment& env, const RTSPConfig& _rtspConfig)
+ourRTSPClient* ourRTSPClient::createNew(UsageEnvironment& env, const PL_RTSPClient_Config& _rtspConfig)
{
- return new ourRTSPClient(env, _rtspConfig);
+ return new ourRTSPClient(env, _rtspConfig);
}
-ourRTSPClient::ourRTSPClient(UsageEnvironment& env, const RTSPConfig& _rtspConfig)
- : RTSPClient(env, _rtspConfig.rtspURL.c_str(), _rtspConfig.verbosityLevel, _rtspConfig.progName.c_str(),
- _rtspConfig.tunnelOverHTTPPortNum, -1), rtspConfig(_rtspConfig)
+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)
{
}
-ourRTSPClient::~ourRTSPClient() {
+ourRTSPClient::~ourRTSPClient()
+{
}
// Implementation of "StreamClientState":
StreamClientState::StreamClientState()
- : iter(NULL), session(NULL), subsession(NULL), streamTimerTask(NULL), duration(0.0) {
+ : iter(NULL), session(NULL), subsession(NULL), streamTimerTask(NULL), duration(0.0)
+{
}
-StreamClientState::~StreamClientState() {
- delete iter;
- if (session != NULL) {
- // We also need to delete "session", and unschedule "streamTimerTask" (if set)
- UsageEnvironment& env = session->envir(); // alias
+StreamClientState::~StreamClientState()
+{
+ delete iter;
+ if (session != NULL)
+ {
+ // We also need to delete "session", and unschedule "streamTimerTask" (if set)
+ UsageEnvironment& env = session->envir(); // alias
- env.taskScheduler().unscheduleDelayedTask(streamTimerTask);
- Medium::close(session);
- }
+ env.taskScheduler().unscheduleDelayedTask(streamTimerTask);
+ Medium::close(session);
+ }
}
// Implementation of "DummySink":
-DummySink* DummySink::createNew(UsageEnvironment& env, const RTSPConfig& _rtspConfig, MediaSubsession& subsession, char const* streamId)
+DummySink* DummySink::createNew(UsageEnvironment& env, const PL_RTSPClient_Config& _rtspConfig, MediaSubsession& subsession, char const* streamId)
{
- return new DummySink(env, _rtspConfig, subsession, streamId);
+ return new DummySink(env, _rtspConfig, subsession, streamId);
}
-DummySink::DummySink(UsageEnvironment& env, const RTSPConfig& _rtspConfig, MediaSubsession& subsession, char const* streamId)
- : MediaSink(env), rtspConfig(_rtspConfig), fSubsession(subsession)
+DummySink::DummySink(UsageEnvironment& env, const PL_RTSPClient_Config& _rtspConfig, MediaSubsession& subsession, char const* streamId)
+ : MediaSink(env), rtspConfig(_rtspConfig), fSubsession(subsession)
{
fStreamId = strDup(streamId);
fReceiveBuffer = new u_int8_t[DUMMY_SINK_RECEIVE_BUFFER_SIZE];
// 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;
+ }
//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;
}
-DummySink::~DummySink() {
- delete[] fReceiveBuffer;
- delete[] fStreamId;
+DummySink::~DummySink()
+{
+ delete[] fReceiveBuffer;
+ delete[] fStreamId;
}
void DummySink::afterGettingFrame(void* clientData, unsigned frameSize, unsigned numTruncatedBytes,
- struct timeval presentationTime, unsigned durationInMicroseconds)
+ struct timeval presentationTime, unsigned durationInMicroseconds)
{
DummySink* sink = (DummySink*)clientData;
if (frameSize > 0)
- {
- unsigned s = frameSize;
- if (sink->rtspConfig.aux)
- s += 4;
- rtsp_client_frame_callback(sink->rtspConfig.args, sink->fReceiveBuffer, s, presentationTime);
- }
-
+ {
+ unsigned s = frameSize;
+ if (sink->rtspConfig.aux)
+ s += 4;
+ rtsp_client_frame_callback(sink->rtspConfig.args, sink->fReceiveBuffer, s, presentationTime);
+ }
+
sink->afterGettingFrame(frameSize, numTruncatedBytes, presentationTime, durationInMicroseconds);
}
void DummySink::afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes,
- struct timeval presentationTime, unsigned /*durationInMicroseconds*/) {
- // We've just received a frame of data. (Optionally) print out information about it:
+ struct timeval presentationTime, unsigned /*durationInMicroseconds*/)
+{
+ // We've just received a frame of data. (Optionally) print out information about it:
#ifdef DEBUG_PRINT_EACH_RECEIVED_FRAME
- if (fStreamId != NULL) envir() << "Stream \"" << fStreamId << "\"; ";
- envir() << fSubsession.mediumName() << "/" << fSubsession.codecName() << ":\tReceived " << frameSize << " bytes";
- if (numTruncatedBytes > 0) envir() << " (with " << numTruncatedBytes << " bytes truncated)";
- char uSecsStr[6+1]; // used to output the 'microseconds' part of the presentation time
- sprintf(uSecsStr, "%06u", (unsigned)presentationTime.tv_usec);
- envir() << ".\tPresentation time: " << (int)presentationTime.tv_sec << "." << uSecsStr;
- if (fSubsession.rtpSource() != NULL && !fSubsession.rtpSource()->hasBeenSynchronizedUsingRTCP()) {
- envir() << "!"; // mark the debugging output to indicate that this presentation time is not RTCP-synchronized
- }
+ if (fStreamId != NULL)
+ LOG_DEBUG << "Stream \"" << fStreamId << "\"; ";
+ LOG_DEBUG << "\t" << fSubsession.mediumName() << "/" << fSubsession.codecName() << ":\tReceived " << frameSize << " bytes";
+ if (numTruncatedBytes > 0)
+ LOG_DEBUG << " (with " << numTruncatedBytes << " bytes truncated)";
+
+ 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;
+ 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
+ }
#ifdef DEBUG_PRINT_NPT
- envir() << "\tNPT: " << fSubsession.getNormalPlayTime(presentationTime);
+ LOG_DEBUG << "\tNPT: " << fSubsession.getNormalPlayTime(presentationTime);
#endif
- envir() << "\n";
#endif
-
- // Then continue, to request the next frame of data:
- continuePlaying();
+
+ // Then continue, to request the next frame of data:
+ continuePlaying();
}
Boolean DummySink::continuePlaying()
{
- if (fSource == NULL) return False; // sanity check (should not happen)
+ if (fSource == NULL)
+ return False; // sanity check (should not happen)
rtsp_client_continue_callback(rtspConfig.args);
@@ -580,8 +636,8 @@
// Request the next frame of data from our input source. "afterGettingFrame()" will get called later, when it arrives:
fSource->getNextFrame(b, DUMMY_SINK_RECEIVE_BUFFER_SIZE,
- afterGettingFrame, this,
- onSourceClosure, this);
+ afterGettingFrame, this,
+ onSourceClosure, this);
return True;
}
--
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