* #18668: audiortp: remove duplicated code

daemon/src/audio/audiortp/audio_rtp_session.cpp

@@ -51,6 +51,7 @@ AudioRtpSession::AudioRtpSession(SIPCall &call, ost::RTPDataQueue &queue) :
     , remote_ip_()
     , remote_port_(0)
     , timestampCount_(0)
+    , rtpSendThread_(*this)
 {
     queue_.setTypeOfService(ost::RTPDataQueue::tosEnhanced);
 }
@@ -268,4 +269,52 @@ void AudioRtpSession::startRtpThreads(const std::vector<AudioCodec*> &audioCodec
     startSendThread();
 }
 
+AudioRtpSession::AudioRtpSendThread::AudioRtpSendThread(AudioRtpSession &session) :
+    running_(false), rtpSession_(session), thread_(0), timer_()
+{}
+
+AudioRtpSession::AudioRtpSendThread::~AudioRtpSendThread()
+{
+    running_ = false;
+    if (thread_)
+        pthread_join(thread_, NULL);
+}
+
+void AudioRtpSession::AudioRtpSendThread::start()
+{
+    running_ = true;
+    pthread_create(&thread_, NULL, &runCallback, this);
+}
+
+void *
+AudioRtpSession::AudioRtpSendThread::runCallback(void *data)
+{
+    AudioRtpSession::AudioRtpSendThread *context = static_cast<AudioRtpSession::AudioRtpSendThread*>(data);
+    context->run();
+    return NULL;
+}
+
+void AudioRtpSession::AudioRtpSendThread::run()
+{
+    timer_.setTimer(rtpSession_.transportRate_);
+    const int MS_TO_USEC = 1000;
+
+    while (running_) {
+        // Send session
+        if (rtpSession_.hasDTMFPending())
+            rtpSession_.sendDtmfEvent();
+        else
+            rtpSession_.sendMicData();
+
+        usleep(timer_.getTimer() * MS_TO_USEC);
+
+        timer_.incTimer(rtpSession_.transportRate_);
+    }
+}
+
+
+void AudioRtpSession::startSendThread()
+{
+    rtpSendThread_.start();
+}
 }

daemon/src/audio/audiortp/audio_rtp_session.h

@@ -86,19 +86,6 @@ class AudioRtpSession : public AudioRtpRecordHandler {
 
         bool onRTPPacketRecv(ost::IncomingRTPPkt&);
 
-        /**
-         * Send DTMF over RTP (RFC2833). The timestamp and sequence number must be
-         * incremented as if it was microphone audio. This function change the payload type of the rtp session,
-         * send the appropriate DTMF digit using this payload, discard coresponding data from mainbuffer and get
-         * back the codec payload for further audio processing.
-         */
-        void sendDtmfEvent();
-
-        /**
-         * Send encoded data to peer
-         */
-        virtual void sendMicData();
-
         SIPCall &call_;
 
         /**
@@ -121,7 +108,37 @@ class AudioRtpSession : public AudioRtpRecordHandler {
     private:
         NON_COPYABLE(AudioRtpSession);
         virtual void startReceiveThread() = 0;
-        virtual void startSendThread() = 0;
+        void startSendThread();
+
+        /**
+         * Send DTMF over RTP (RFC2833). The timestamp and sequence number must be
+         * incremented as if it was microphone audio. This function change the payload type of the rtp session,
+         * send the appropriate DTMF digit using this payload, discard coresponding data from mainbuffer and get
+         * back the codec payload for further audio processing.
+         */
+        void sendDtmfEvent();
+
+        /**
+         * Send encoded data to peer
+         */
+        virtual void sendMicData();
+
+
+        class AudioRtpSendThread {
+            public:
+                AudioRtpSendThread(AudioRtpSession &session);
+                ~AudioRtpSendThread();
+                void start();
+                bool running_;
+
+            private:
+                static void *runCallback(void *data);
+                void run();
+                NON_COPYABLE(AudioRtpSendThread);
+                AudioRtpSession &rtpSession_;
+                pthread_t thread_;
+                ost::TimerPort timer_;
+        };
 
         /**
          * Set RTP Sockets send/receive timeouts
@@ -152,6 +169,8 @@ class AudioRtpSession : public AudioRtpRecordHandler {
          * Timestamp reset frequency specified in number of packet sent
          */
         short timestampCount_;
+
+        AudioRtpSendThread rtpSendThread_;
 };
 
 }

daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp

@@ -41,63 +41,14 @@ namespace sfl {
 AudioSymmetricRtpSession::AudioSymmetricRtpSession(SIPCall &call) :
     ost::SymmetricRTPSession(ost::InetHostAddress(call.getLocalIp().c_str()), call.getLocalAudioPort())
     , AudioRtpSession(call, *this)
-    , rtpSendThread_(*this)
 {
     DEBUG("Setting new RTP session with destination %s:%d",
             call_.getLocalIp().c_str(), call_.getLocalAudioPort());
     audioRtpRecord_.callId_ = call_.getCallId();
 }
 
-AudioSymmetricRtpSession::AudioRtpSendThread::AudioRtpSendThread(AudioSymmetricRtpSession &session) :
-    running_(false), rtpSession_(session), thread_(0)
-{}
-
-AudioSymmetricRtpSession::AudioRtpSendThread::~AudioRtpSendThread()
-{
-    running_ = false;
-    if (thread_)
-        pthread_join(thread_, NULL);
-}
-
-void AudioSymmetricRtpSession::AudioRtpSendThread::start()
-{
-    running_ = true;
-    pthread_create(&thread_, NULL, &runCallback, this);
-}
-
-void *
-AudioSymmetricRtpSession::AudioRtpSendThread::runCallback(void *data)
-{
-    AudioSymmetricRtpSession::AudioRtpSendThread *context = static_cast<AudioSymmetricRtpSession::AudioRtpSendThread*>(data);
-    context->run();
-    return NULL;
-}
-
-void AudioSymmetricRtpSession::AudioRtpSendThread::run()
-{
-    ost::TimerPort::setTimer(rtpSession_.transportRate_);
-    const int MS_TO_USEC = 1000;
-
-    while (running_) {
-        // Send session
-        if (rtpSession_.hasDTMFPending())
-            rtpSession_.sendDtmfEvent();
-        else
-            rtpSession_.sendMicData();
-
-        usleep(ost::TimerPort::getTimer() * MS_TO_USEC);
-
-        ost::TimerPort::incTimer(rtpSession_.transportRate_);
-    }
-}
-
 void AudioSymmetricRtpSession::startReceiveThread()
 {
     ost::SymmetricRTPSession::start();
 }
-
-void AudioSymmetricRtpSession::startSendThread()
-{
-    rtpSendThread_.start();
-}
 }

daemon/src/audio/audiortp/audio_symmetric_rtp_session.h

@@ -78,25 +78,7 @@ class AudioSymmetricRtpSession : public ost::TimerPort, public ost::SymmetricRTP
     private:
         NON_COPYABLE(AudioSymmetricRtpSession);
 
-        class AudioRtpSendThread : public ost::TimerPort {
-            public:
-                AudioRtpSendThread(AudioSymmetricRtpSession &session);
-                ~AudioRtpSendThread();
-                void start();
-                bool running_;
-
-            private:
-                NON_COPYABLE(AudioRtpSendThread);
-                static void *runCallback(void *data);
-                void run();
-                pthread_t thread_;
-                AudioSymmetricRtpSession &rtpSession_;
-        };
-
         void startReceiveThread();
-        void startSendThread();
-
-        AudioRtpSendThread rtpSendThread_ ;
 };
 
 }

daemon/src/audio/audiortp/audio_zrtp_session.cpp

@@ -50,7 +50,6 @@ AudioZrtpSession::AudioZrtpSession(SIPCall &call, const std::string &zidFilename
     ost::SymmetricZRTPSession(ost::InetHostAddress(call.getLocalIp().c_str()), call.getLocalAudioPort())
     , AudioRtpSession(call, *this)
     , zidFilename_(zidFilename)
-    , rtpSendThread_(*this)
 {
     initializeZid();
     DEBUG("Setting new RTP session with destination %s:%d",
@@ -112,52 +111,6 @@ void AudioZrtpSession::sendMicData()
     queue_.sendImmediate(timestamp_, getMicDataEncoded(), compSize);
 }
 
-AudioZrtpSession::AudioZrtpSendThread::AudioZrtpSendThread(AudioZrtpSession &session) :
-    running_(true), zrtpSession_(session), thread_(0)
-{}
-
-AudioZrtpSession::AudioZrtpSendThread::~AudioZrtpSendThread()
-{
-    running_ = false;
-    if (thread_)
-        pthread_join(thread_, NULL);
-}
-
-void
-AudioZrtpSession::AudioZrtpSendThread::start()
-{
-    pthread_create(&thread_, NULL, &runCallback, this);
-}
-
-void *
-AudioZrtpSession::AudioZrtpSendThread::runCallback(void *data)
-{
-    AudioZrtpSession::AudioZrtpSendThread *context = static_cast<AudioZrtpSession::AudioZrtpSendThread*>(data);
-    context->run();
-    return NULL;
-}
-
-void AudioZrtpSession::AudioZrtpSendThread::run()
-{
-    DEBUG("Entering Audio zrtp thread main loop %s", running_ ? "running" : "not running");
-
-    TimerPort::setTimer(zrtpSession_.transportRate_);
-
-    while (running_) {
-        // Send session
-        if (zrtpSession_.hasDTMFPending())
-            zrtpSession_.sendDtmfEvent();
-        else
-            zrtpSession_.sendMicData();
-
-        Thread::sleep(TimerPort::getTimer());
-
-        TimerPort::incTimer(zrtpSession_.transportRate_);
-    }
-
-    DEBUG("Leaving audio rtp thread loop");
-}
-
 int AudioZrtpSession::getIncrementForDTMF() const
 {
     return 160;
@@ -168,9 +121,4 @@ void AudioZrtpSession::startReceiveThread()
     ost::SymmetricZRTPSession::start();
 }
 
-void AudioZrtpSession::startSendThread()
-{
-    rtpSendThread_.start();
-}
-
 }

daemon/src/audio/audiortp/audio_zrtp_session.h

@@ -42,7 +42,6 @@ using std::ptrdiff_t;
 
 #include "global.h"
 #include "audio_rtp_session.h"
-// #include <commoncpp/numbers.h> // OST::Time
 
 class SIPCall;
 class AudioCodec;
@@ -76,28 +75,11 @@ class AudioZrtpSession :
     private:
         NON_COPYABLE(AudioZrtpSession);
 
-        class AudioZrtpSendThread : public ost::TimerPort {
-            public:
-                AudioZrtpSendThread(AudioZrtpSession &session);
-                ~AudioZrtpSendThread();
-                void start();
-                bool running_;
-
-            private:
-                static void *runCallback(void *data);
-                void run();
-                NON_COPYABLE(AudioZrtpSendThread);
-                AudioZrtpSession &zrtpSession_;
-                pthread_t thread_;
-        };
         void sendMicData();
         void initializeZid();
         std::string zidFilename_;
         void startReceiveThread();
-        void startSendThread();
         virtual int getIncrementForDTMF() const;
-
-        AudioZrtpSendThread rtpSendThread_;
 };
 
 }