* #6124 : add VideoRtpFactory class

Currently a dummy class.

sflphone-common/libs/video/Makefile.am

@@ -5,7 +5,8 @@ noinst_LTLIBRARIES = libvideo_endpoint.la
 
 libvideo_endpoint_la_SOURCES = video_endpoint.cpp video_endpoint.h \
 							   libav_utils.cpp libav_utils.h \
-							   video_rtp_session.cpp video_rtp_session.h
+							   video_rtp_session.cpp video_rtp_session.h \
+							   video_rtp_factory.cpp video_rtp_factory.h
 libvideo_endpoint_la_LIBADD = @LIBAVFORMAT_LIBS@ 
 
 AM_CXXFLAGS=@LIBAVCODEC_CFLAGS@ @LIBAVFORMAT_CFLAGS@ @LIBAVDEVICE_CFLAGS@ @LIBSWSCALE_CFLAGS@

sflphone-common/libs/video/video_rtp_factory.cpp

@@ -0,0 +1,48 @@
+/*
+ *  Copyright (C) 2004, 2005, 2006, 2009, 2008, 2009, 2010, 2011 Savoir-Faire Linux Inc.
+ *  Author: Tristan Matthews <tristan.matthews@savoirfairelinux.com>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; either version 3 of the License, or
+ *  (at your option) any later version.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ *  Additional permission under GNU GPL version 3 section 7:
+ *
+ *  If you modify this program, or any covered work, by linking or
+ *  combining it with the OpenSSL project's OpenSSL library (or a
+ *  modified version of that library), containing parts covered by the
+ *  terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
+ *  grants you additional permission to convey the resulting work.
+ *  Corresponding Source for a non-source form of such a combination
+ *  shall include the source code for the parts of OpenSSL used as well
+ *  as that of the covered work.
+ */
+
+#include "video_rtp_factory.h"
+
+namespace sfl_video 
+{
+VideoRtpFactory::VideoRtpFactory(SIPCall * ca)
+{
+}
+
+void VideoRtpFactory::start()
+{
+    // starting
+}
+
+void VideoRtpFactory::stop()
+{
+    // stop
+}
+} // end namespace sfl

sflphone-common/libs/video/video_rtp_factory.h

@@ -0,0 +1,56 @@
+/*
+ *  Copyright (C) 2004, 2005, 2006, 2009, 2008, 2009, 2010, 2011 Savoir-Faire Linux Inc.
+ *  Author: Tristan Matthews <tristan.matthews@savoirfairelinux.com>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; either version 3 of the License, or
+ *  (at your option) any later version.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ *  Additional permission under GNU GPL version 3 section 7:
+ *
+ *  If you modify this program, or any covered work, by linking or
+ *  combining it with the OpenSSL project's OpenSSL library (or a
+ *  modified version of that library), containing parts covered by the
+ *  terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
+ *  grants you additional permission to convey the resulting work.
+ *  Corresponding Source for a non-source form of such a combination
+ *  shall include the source code for the parts of OpenSSL used as well
+ *  as that of the covered work.
+ */
+
+#ifndef __VIDEO_RTP_SESSION_H__
+#define __VIDEO_RTP_SESSION_H__
+
+class SIPCall;
+
+namespace sfl_video {
+class VideoRtpFactory {
+    public:
+        VideoRtpFactory(SIPCall * ca);
+
+        /**
+         * Start the video RTP thread of the type specified in the configuration
+         * file.
+         * @param None
+         */
+        void start();
+        /**
+         * Stop the audio rtp thread of the type specified in the configuration
+         * file. initAudioRtpSession must have been called prior to that.
+         * @param None
+         */
+        void stop();
+};
+}
+
+#endif // __VIDEO_RTP_SESSION_H__

sflphone-common/src/audio/audiortp/AudioRtpSession.cpp

@@ -69,7 +69,6 @@ AudioRtpSession::~AudioRtpSession()
 
 void AudioRtpSession::final()
 {
-
     delete _rtpThread;
 
     delete static_cast<AudioRtpSession *> (this);

sflphone-common/src/sip/sipcall.h

@@ -45,6 +45,11 @@ namespace sfl
 class AudioRtpFactory;
 }
 
+namespace sfl_video
+{
+class VideoRtpFactory;
+}
+
 /**
  * @file sipcall.h
  * @brief SIPCall are SIP implementation of a normal Call
@@ -161,6 +166,13 @@ class SIPCall : public Call
             return _audiortp;
         }
 
+        /**
+         * Returns a pointer to the VideoRtp object
+         */
+        sfl_video::VideoRtpFactory * getVideoRtp (void) {
+            return videortp_;
+        }
+
         /**
          * Return the local memory pool for this call
          */
@@ -206,6 +218,11 @@ class SIPCall : public Call
          */
         sfl::AudioRtpFactory * _audiortp;
 
+        /**
+         * Video Rtp Session factory
+         */
+        sfl_video::VideoRtpFactory * videortp_;
+
         /**
          * Event subscription structure
          */

sflphone-common/src/sip/sipvoiplink.cpp

@@ -51,6 +51,8 @@
 #include "audio/audiolayer.h"
 #include "audio/audiortp/AudioRtpFactory.h"
 
+#include "video/video_rtp_factory.h"
+
 #include "pjsip/sip_endpoint.h"
 #include "pjsip/sip_transport_tls.h"
 #include "pjsip/sip_transport_tls.h"
@@ -717,6 +719,8 @@ Call *SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl)
 		call->getAudioRtp()->initLocalCryptoInfo (call);
 		_info ("UserAgent: Start audio rtp session");
 		call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec));
+		_info ("UserAgent: Start video rtp session");
+		call->getVideoRtp()->start();
 	} catch (...) {
 		throw VoipLinkException ("Could not start rtp session for early media");
 	}
@@ -855,6 +859,7 @@ SIPVoIPLink::hangup (const CallID& id) throw (VoipLinkException)
     try {
         if (Manager::instance().isCurrentCall (id)) {
             call->getAudioRtp()->stop();
+            call->getVideoRtp()->stop();
         }
     }
     catch(...) {
@@ -901,6 +906,7 @@ SIPVoIPLink::peerHungup (const CallID& id) throw (VoipLinkException)
         if (Manager::instance().isCurrentCall (id)) {
             _debug ("UserAgent: Stopping AudioRTP for hangup");
             call->getAudioRtp()->stop();
+            call->getVideoRtp()->stop();
         }
     }
     catch(...) {
@@ -942,11 +948,11 @@ SIPVoIPLink::onhold (const CallID& id) throw (VoipLinkException)
     }
 
     // Stop sound
-    call->setAudioStart (false);
     call->setState (Call::Hold);
 
     try {
         call->getAudioRtp()->stop();
+        call->getVideoRtp()->stop();
     }
     catch (...) {
     	throw VoipLinkException("Could not stop audio rtp session");
@@ -1020,6 +1026,7 @@ SIPVoIPLink::offhold (const CallID& id) throw (VoipLinkException)
         call->getAudioRtp()->initAudioRtpConfig (call);
         call->getAudioRtp()->initAudioRtpSession (call);
         call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec));
+        call->getVideoRtp()->start();
 
     }
     catch (SdpException &e) {
@@ -1314,6 +1321,7 @@ SIPVoIPLink::refuse (const CallID& id)
 
     // Stop Audio RTP session
     call->getAudioRtp()->stop();
+    call->getVideoRtp()->stop();
 
     // User refuse current call. Notify peer
     status = pjsip_inv_end_session (call->getInvSession(), PJSIP_SC_DECLINE, NULL, &tdata);   //603
@@ -1652,6 +1660,10 @@ SIPVoIPLink::SIPCallServerFailure (SIPCall *call)
         if (call->getAudioRtp ()) {
             call->getAudioRtp()->stop();
         }
+
+        if (call->getVideoRtp ()) {
+            call->getVideoRtp()->stop();
+        }
     }
 }
 
@@ -1668,9 +1680,9 @@ SIPVoIPLink::SIPCallClosed (SIPCall *call)
     CallID id = call->getCallId();
 
     if (Manager::instance().isCurrentCall (id)) {
-        call->setAudioStart (false);
         _debug ("UserAgent: Stopping AudioRTP when closing");
         call->getAudioRtp()->stop();
+        call->getVideoRtp()->stop();
     }
 
     Manager::instance().peerHungupCall (id);
@@ -1795,6 +1807,7 @@ bool SIPVoIPLink::SIPNewIpToIpCall (const CallID& id, const std::string& to)
             call->getAudioRtp()->initAudioRtpSession (call);
             call->getAudioRtp()->initLocalCryptoInfo (call);
             call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec));
+            call->getVideoRtp()->start ();
         } catch (...) {
             _debug ("UserAgent: Unable to create RTP Session in new IP2IP call (%s:%d)", __FILE__, __LINE__);
         }
@@ -3518,8 +3531,6 @@ void sdp_media_update_cb (pjsip_inv_session *inv, pj_status_t status)
     AudioCodecType pl = (AudioCodecType) sessionMedia->getPayloadType();
 
     try {
-        call->setAudioStart (true);
-
         Manager::instance().audioLayerMutexLock();
         Manager::instance().getAudioDriver()->startStream();
         Manager::instance().audioLayerMutexUnlock();