add thread name for debugging

mpromonet · Sep 22, 2018 · d0f47da · d0f47da
1 parent 014c9c8
commit d0f47da
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Showing 6 changed files with 34 additions and 47 deletions.
diff --git a/inc/rtspaudiocapturer.h b/inc/rtspaudiocapturer.h
@@ -19,8 +19,8 @@
 
 class RTSPAudioSource : public webrtc::Notifier<webrtc::AudioSourceInterface>, public rtc::Thread, public RTSPConnection::Callback {
 	public:
-		static rtc::scoped_refptr<RTSPAudioSource> Create(rtc::scoped_refptr<webrtc::AudioDecoderFactory> audioDecoderFactory, const std::string & uri) {
-			rtc::scoped_refptr<RTSPAudioSource> source(new rtc::RefCountedObject<RTSPAudioSource>(audioDecoderFactory, uri));
+		static rtc::scoped_refptr<RTSPAudioSource> Create(rtc::scoped_refptr<webrtc::AudioDecoderFactory> audioDecoderFactory, const std::string & uri, const std::map<std::string,std::string> & opts) {
+			rtc::scoped_refptr<RTSPAudioSource> source(new rtc::RefCountedObject<RTSPAudioSource>(audioDecoderFactory, uri, opts));
 			return source;
 		}
 
@@ -44,11 +44,8 @@ class RTSPAudioSource : public webrtc::Notifier<webrtc::AudioSourceInterface>, p
 		virtual bool onData(const char* id, unsigned char* buffer, ssize_t size, struct timeval presentationTime);
 
 	protected:
-		RTSPAudioSource(rtc::scoped_refptr<webrtc::AudioDecoderFactory> audioDecoderFactory, const std::string & uri) 
-				: m_connection(m_env, this, uri.c_str(), 5, RTSPConnection::RTPUDPUNICAST, rtc::LogMessage::GetLogToDebug()<=3)
-				, m_factory(audioDecoderFactory), m_sink(NULL), m_freq(8000), m_channel(1) { rtc::Thread::Start(); }
-		virtual ~RTSPAudioSource() override { m_env.stop(); rtc::Thread::Stop(); }
-
+		RTSPAudioSource(rtc::scoped_refptr<webrtc::AudioDecoderFactory> audioDecoderFactory, const std::string & uri, const std::map<std::string,std::string> & opts); 
+		virtual ~RTSPAudioSource();
 
 	private:
 		Environment                             m_env;

diff --git a/inc/rtspvideocapturer.h b/inc/rtspvideocapturer.h
@@ -45,15 +45,13 @@ class RTSPVideoCapturer : public cricket::VideoCapturer, public RTSPConnection::
 		// overide RTSPConnection::Callback
 		virtual bool onNewSession(const char* id, const char* media, const char* codec, const char* sdp);
 		virtual bool onData(const char* id, unsigned char* buffer, ssize_t size, struct timeval presentationTime);
-                virtual void    onConnectionTimeout(RTSPConnection& connection) {
-                        connection.start();
-                }
-                virtual void    onDataTimeout(RTSPConnection& connection)       {
-                        connection.start();
-                }
-                virtual void    onError(RTSPConnection& connection,const char* erro)       {
-                        connection.start(1);
-                }		
+		virtual void    onConnectionTimeout(RTSPConnection& connection) {
+				connection.start();
+		}
+		virtual void    onDataTimeout(RTSPConnection& connection) {
+				connection.start();
+		}
+		virtual void    onError(RTSPConnection& connection,const char* erro);
 
 		// overide webrtc::DecodedImageCallback
 		virtual int32_t Decoded(webrtc::VideoFrame& decodedImage);

diff --git a/live555helper b/live555helper
diff --git a/src/PeerConnectionManager.cpp b/src/PeerConnectionManager.cpp
@@ -793,7 +793,7 @@ rtc::scoped_refptr<webrtc::AudioTrackInterface> PeerConnectionManager::CreateAud
 	{
 #ifdef HAVE_LIVE555
 		audioDeviceModule_->Terminate();
-		audioSource = RTSPAudioSource::Create(audioDecoderfactory_, audiourl);
+		audioSource = RTSPAudioSource::Create(audioDecoderfactory_, audiourl, opts);
 #endif
 	}
 	else if (std::regex_match("audiocap://",m_publishFilter)) 

diff --git a/src/rtspaudiocapturer.cpp b/src/rtspaudiocapturer.cpp
@@ -17,6 +17,19 @@
 #include "rtspaudiocapturer.h"
 
 
+RTSPAudioSource::RTSPAudioSource(rtc::scoped_refptr<webrtc::AudioDecoderFactory> audioDecoderFactory, const std::string & uri, const std::map<std::string,std::string> & opts) 
+				: m_connection(m_env, this, uri.c_str(), RTSPConnection::decodeTimeoutOption(opts), RTSPConnection::decodeRTPTransport(opts), rtc::LogMessage::GetLogToDebug()<=3)
+				, m_factory(audioDecoderFactory), m_sink(NULL), m_freq(8000), m_channel(1) {
+	SetName("RTSPAudioSource", NULL);
+	rtc::Thread::Start(); 
+}
+
+RTSPAudioSource::~RTSPAudioSource()  { 
+	m_env.stop(); 
+	rtc::Thread::Stop(); 
+}
+
+
 // overide RTSPConnection::Callback
 bool RTSPAudioSource::onNewSession(const char* id, const char* media, const char* codec, const char* sdp) {
 
@@ -97,7 +110,7 @@ bool RTSPAudioSource::onData(const char* id, unsigned char* buffer, ssize_t size
 			RTC_LOG(LS_ERROR) << "RTSPAudioSource::onData error:No Audio decoder";
 		}
 	} else {
-		RTC_LOG(LS_ERROR) << "RTSPAudioSource::onData error:No Audio Sink";
+		RTC_LOG(LS_VERBOSE) << "RTSPAudioSource::onData error:No Audio Sink";
 	}
 	return success;
 }

diff --git a/src/rtspvideocapturer.cpp b/src/rtspvideocapturer.cpp
@@ -32,35 +32,8 @@
 
 uint8_t marker[] = { 0, 0, 0, 1};
 
-int decodeTimeoutOption(const std::map<std::string,std::string> & opts) {
-	int timeout = 10;
-	if (opts.find("timeout") != opts.end()) 
-	{
-		std::string timeoutString = opts.at("timeout");
-		timeout = std::stoi(timeoutString);
-	}
-	return timeout;
-}
-
-int decodeRTPTransport(const std::map<std::string,std::string> & opts) 
-{
-	int rtptransport = RTSPConnection::RTPUDPUNICAST;
-	if (opts.find("rtptransport") != opts.end()) 
-	{
-		std::string rtpTransportString = opts.at("rtptransport");
-		if (rtpTransportString == "tcp") {
-			rtptransport = RTSPConnection::RTPOVERTCP;
-		} else if (rtpTransportString == "http") {
-			rtptransport = RTSPConnection::RTPOVERHTTP;
-		} else if (rtpTransportString == "multicast") {
-			rtptransport = RTSPConnection::RTPUDPMULTICAST;
-		}
-	}
-	return rtptransport;
-}
-
 RTSPVideoCapturer::RTSPVideoCapturer(const std::string & uri, const std::map<std::string,std::string> & opts) 
-	: m_connection(m_env, this, uri.c_str(), decodeTimeoutOption(opts), decodeRTPTransport(opts), rtc::LogMessage::GetLogToDebug()<=3),
+	: m_connection(m_env, this, uri.c_str(), RTSPConnection::decodeTimeoutOption(opts), RTSPConnection::decodeRTPTransport(opts), rtc::LogMessage::GetLogToDebug()<=3),
 	m_width(0), m_height(0), m_fps(0)
 {
 	RTC_LOG(INFO) << "RTSPVideoCapturer " << uri ;
@@ -230,6 +203,11 @@ bool RTSPVideoCapturer::onData(const char* id, unsigned char* buffer, ssize_t si
 	return (res == 0);
 }
 
+void RTSPVideoCapturer::onError(RTSPConnection& connection, const char* error) {
+	RTC_LOG(LS_ERROR) << "RTSPVideoCapturer:onError url:" << m_connection.getUrl() <<  " error:" << error;
+	connection.start(1);
+}		
+
 void RTSPVideoCapturer::DecoderThread() 
 {
 	while (IsRunning()) {
@@ -313,6 +291,7 @@ cricket::CaptureState RTSPVideoCapturer::Start(const cricket::VideoFormat& forma
 	RTC_LOG(INFO) << "RTSPVideoCapturer::start format" << format.ToString();
 	SetCaptureFormat(&format);
 	SetCaptureState(cricket::CS_RUNNING);
+	SetName("RTSPVideoCapturer", NULL);
 	rtc::Thread::Start();
 	m_decoderthread = std::thread(&RTSPVideoCapturer::DecoderThread, this);
 	return cricket::CS_RUNNING;