Added DtlsCertificate, a ref counted object owning an SSLIdentity

talk/app/webrtc/webrtcsession.cc

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 499 matching lines @@
     SignalVoiceChannelDestroyed();
     channel_manager_->DestroyVoiceChannel(voice_channel_.release(), nullptr);
   }
   if (data_channel_) {
     SignalDataChannelDestroyed();
     channel_manager_->DestroyDataChannel(data_channel_.release());
   }
   for (size_t i = 0; i < saved_candidates_.size(); ++i) {
     delete saved_candidates_[i];
   }
-  delete identity();
 }
 
 bool WebRtcSession::Initialize(
     const PeerConnectionFactoryInterface::Options& options,
-    const MediaConstraintsInterface*  constraints,
+    const MediaConstraintsInterface* constraints,
     DTLSIdentityServiceInterface* dtls_identity_service,
     const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
+  // Default |dtls_enabled_| value, may get overwritten in InitializeCommon.
+  dtls_enabled_ = (dtls_identity_service != nullptr);
+
+  if (!InitializeCommon(options, constraints, rtc_configuration)) {
+    if (dtls_identity_service) {
+      // Since we have the ownership of |dtls_identity_service| and will not
+      // pass it to WebRtcSessionDescriptionFactory it is our job to delete it.
+      delete dtls_identity_service;
+    }
+    return false;
+  }
+
+  if (dtls_enabled_) {
+    // Note: the factory takes ownership of |dtls_identity_service|.
+    webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
+          signaling_thread(),
+          worker_thread(),
+          channel_manager_,
+          mediastream_signaling_,
+          dtls_identity_service,
+          this,
+          id(),
+          data_channel_type_));
+  } else {
+    if (dtls_identity_service) {
+      // Since we have the ownership of |dtls_identity_service| and will not
+      // pass it to WebRtcSessionDescriptionFactory it is our job to delete it.
+      delete dtls_identity_service;
+    }
+
+    webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
+          signaling_thread(),
+          worker_thread(),
+          channel_manager_,
+          mediastream_signaling_,
+          this,
+          id(),
+          data_channel_type_));
+  }
+  InitializeFactoryAfterConstruction(options);
+  return true;
+}
+
+bool WebRtcSession::Initialize(
+    const PeerConnectionFactoryInterface::Options& options,
+    const MediaConstraintsInterface* constraints,
+    rtc::scoped_refptr<webrtc::DtlsCertificate> certificate,
+    const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
+  DCHECK(certificate.get());
+
+  // Default |dtls_enabled_| value, may get overwritten in InitializeCommon.
+  dtls_enabled_ = true;
+
+  if (!InitializeCommon(options, constraints, rtc_configuration)) {
+    return false;
+  }
+
+  if (dtls_enabled_) {
+    webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
+          signaling_thread(),
+          worker_thread(),
+          channel_manager_,
+          mediastream_signaling_,
+          certificate,
+          this,
+          id(),
+          data_channel_type_));
+  } else {
+    webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
+          signaling_thread(),
+          worker_thread(),
+          channel_manager_,
+          mediastream_signaling_,
+          this,
+          id(),
+          data_channel_type_));
+  }
+  InitializeFactoryAfterConstruction(options);
+  return true;
+}
+
+void WebRtcSession::InitializeFactoryAfterConstruction(
+    const PeerConnectionFactoryInterface::Options& options) {
+  webrtc_session_desc_factory_->SignalCertificateReady.connect(
+      this, &WebRtcSession::OnCertificateReady);
+  if (options.disable_encryption) {
+    webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
+  }
+}
+
+bool WebRtcSession::InitializeCommon(
+    const PeerConnectionFactoryInterface::Options& options,
+    const MediaConstraintsInterface* constraints,
+    const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
   bundle_policy_ = rtc_configuration.bundle_policy;
   rtcp_mux_policy_ = rtc_configuration.rtcp_mux_policy;
   SetSslMaxProtocolVersion(options.ssl_max_version);
 
   // TODO(perkj): Take |constraints| into consideration. Return false if not all
   // mandatory constraints can be fulfilled. Note that |constraints|
   // can be null.
   bool value;
 
   if (options.disable_encryption) {
     dtls_enabled_ = false;
   } else {
-    // Enable DTLS by default if |dtls_identity_service| is valid.
-    dtls_enabled_ = (dtls_identity_service != NULL);
     // |constraints| can override the default |dtls_enabled_| value.
     if (FindConstraint(
           constraints,
           MediaConstraintsInterface::kEnableDtlsSrtp,
-          &value, NULL)) {
+          &value, nullptr)) {
       dtls_enabled_ = value;
     }
   }
 
   // Enable creation of RTP data channels if the kEnableRtpDataChannels is set.
   // It takes precendence over the disable_sctp_data_channels
   // PeerConnectionFactoryInterface::Options.
   if (FindConstraint(
       constraints, MediaConstraintsInterface::kEnableRtpDataChannels,
       &value, NULL) && value) {
@@ skipping 96 matching lines @@
   const cricket::VideoCodec default_codec(
       JsepSessionDescription::kDefaultVideoCodecId,
       JsepSessionDescription::kDefaultVideoCodecName,
       JsepSessionDescription::kMaxVideoCodecWidth,
       JsepSessionDescription::kMaxVideoCodecHeight,
       JsepSessionDescription::kDefaultVideoCodecFramerate,
       JsepSessionDescription::kDefaultVideoCodecPreference);
   channel_manager_->SetDefaultVideoEncoderConfig(
       cricket::VideoEncoderConfig(default_codec));
 
-  webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
-      signaling_thread(),
-      channel_manager_,
-      mediastream_signaling_,
-      dtls_identity_service,
-      this,
-      id(),
-      data_channel_type_,
-      dtls_enabled_));
-
-  webrtc_session_desc_factory_->SignalIdentityReady.connect(
-      this, &WebRtcSession::OnIdentityReady);
-
-  if (options.disable_encryption) {
-    webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
-  }
   port_allocator()->set_candidate_filter(
       ConvertIceTransportTypeToCandidateFilter(rtc_configuration.type));
   return true;
 }
 
 void WebRtcSession::Terminate() {
   SetState(STATE_RECEIVEDTERMINATE);
   RemoveUnusedChannelsAndTransports(NULL);
   ASSERT(!voice_channel_);
   ASSERT(!video_channel_);
@@ skipping 608 matching lines @@
 }
 
 bool WebRtcSession::IceRestartPending() const {
   return ice_restart_latch_->Get();
 }
 
 void WebRtcSession::ResetIceRestartLatch() {
   ice_restart_latch_->Reset();
 }
 
-void WebRtcSession::OnIdentityReady(rtc::SSLIdentity* identity) {
-  SetIdentity(identity);
+void WebRtcSession::OnCertificateReady(
+    rtc::scoped_refptr<DtlsCertificate> certificate) {
+  certificate_ = certificate;
+  SetCertificate(certificate_);
 }
 
-bool WebRtcSession::waiting_for_identity() const {
-  return webrtc_session_desc_factory_->waiting_for_identity();
+bool WebRtcSession::waiting_for_certificate() const {
+  return webrtc_session_desc_factory_->waiting_for_certificate();
+}
+
+rtc::scoped_refptr<DtlsCertificate> WebRtcSession::get_certificate() const {
+  return certificate_;
 }
 
 void WebRtcSession::SetIceConnectionState(
       PeerConnectionInterface::IceConnectionState state) {
   if (ice_connection_state_ == state) {
     return;
   }
 
   // ASSERT that the requested transition is allowed.  Note that
   // WebRtcSession does not implement "kIceConnectionClosed" (that is handled
@@ skipping 642 matching lines @@
 
   if (!srtp_cipher.empty()) {
     metrics_observer_->AddHistogramSample(srtp_name, srtp_cipher);
   }
   if (!ssl_cipher.empty()) {
     metrics_observer_->AddHistogramSample(ssl_name, ssl_cipher);
   }
 }
 
 }  // namespace webrtc