Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface.

webrtc/api/peerconnectioninterface.h

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 50 matching lines @@
 #include "webrtc/api/dtmfsenderinterface.h"
 #include "webrtc/api/jsep.h"
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/rtpreceiverinterface.h"
 #include "webrtc/api/rtpsenderinterface.h"
 #include "webrtc/api/statstypes.h"
 #include "webrtc/api/umametrics.h"
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/network.h"
 #include "webrtc/base/rtccertificate.h"
+#include "webrtc/base/rtccertificategenerator.h"
 #include "webrtc/base/socketaddress.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/p2p/base/portallocator.h"
 
 namespace rtc {
 class SSLIdentity;
 class Thread;
 }
 
@@ skipping 492 matching lines @@
     // party supports DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used.
     rtc::SSLProtocolVersion ssl_max_version;
   };
 
   virtual void SetOptions(const Options& options) = 0;
 
   virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
       const PeerConnectionInterface::RTCConfiguration& configuration,
       const MediaConstraintsInterface* constraints,
       std::unique_ptr<cricket::PortAllocator> allocator,
+      std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
+      PeerConnectionObserver* observer) = 0;
+  // TODO(hbos): To be removed in favor of the |cert_generator| version as soon
+  // as Chromium stops using this version. See bugs.webrtc.org/5707,
+  // bugs.webrtc.org/5708.
+  rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
+      const PeerConnectionInterface::RTCConfiguration& configuration,
+      const MediaConstraintsInterface* constraints,
+      std::unique_ptr<cricket::PortAllocator> allocator,
       std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
-      PeerConnectionObserver* observer) = 0;
+      PeerConnectionObserver* observer) {
+    return CreatePeerConnection(
+        configuration,
+        constraints,
+        std::move(allocator),
+        std::unique_ptr<rtc::RTCCertificateGeneratorInterface>(
+            dtls_identity_store ? new RTCCertificateGeneratorStoreWrapper(
+                std::move(dtls_identity_store)) : nullptr),
+        observer);
+  }
 
   virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
       const PeerConnectionInterface::RTCConfiguration& configuration,
       std::unique_ptr<cricket::PortAllocator> allocator,
+      std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
+      PeerConnectionObserver* observer) = 0;
+  // TODO(hbos): To be removed in favor of the |cert_generator| version as soon
+  // as Chromium stops using this version. See bugs.webrtc.org/5707,
+  // bugs.webrtc.org/5708.
+  rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
+      const PeerConnectionInterface::RTCConfiguration& configuration,
+      std::unique_ptr<cricket::PortAllocator> allocator,
       std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
-      PeerConnectionObserver* observer) = 0;
+      PeerConnectionObserver* observer) {
+    return CreatePeerConnection(
+        configuration,
+        std::move(allocator),
+        std::unique_ptr<rtc::RTCCertificateGeneratorInterface>(
+            dtls_identity_store ? new RTCCertificateGeneratorStoreWrapper(
+                std::move(dtls_identity_store)) : nullptr),
+        observer);
+  }
 
   virtual rtc::scoped_refptr<MediaStreamInterface>
       CreateLocalMediaStream(const std::string& label) = 0;
 
   // Creates a AudioSourceInterface.
   // |constraints| decides audio processing settings but can be NULL.
   virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
       const cricket::AudioOptions& options) = 0;
   // Deprecated - use version above.
   virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
@@ skipping 100 matching lines @@
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     cricket::WebRtcVideoDecoderFactory* decoder_factory) {
   return CreatePeerConnectionFactory(
       worker_and_network_thread, worker_and_network_thread, signaling_thread,
       default_adm, encoder_factory, decoder_factory);
 }
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_PEERCONNECTIONINTERFACE_H_