content::RTCCertificateGenerator and WebRTC-EnableWebRtcEcdsa update.

content/renderer/media/webrtc/peer_connection_dependency_factory.cc

 // Copyright 2014 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
 
 #include <stddef.h>
 
 #include <utility>
 #include <vector>
@@ skipping 19 matching lines @@
 #include "content/public/common/webrtc_ip_handling_policy.h"
 #include "content/public/renderer/content_renderer_client.h"
 #include "content/renderer/media/media_stream.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
 #include "content/renderer/media/media_stream_audio_processor_options.h"
 #include "content/renderer/media/media_stream_audio_source.h"
 #include "content/renderer/media/media_stream_constraints_util.h"
 #include "content/renderer/media/media_stream_video_source.h"
 #include "content/renderer/media/media_stream_video_track.h"
 #include "content/renderer/media/peer_connection_identity_store.h"
+#include "content/renderer/media/rtc_certificate_generator.h"
 #include "content/renderer/media/rtc_peer_connection_handler.h"
 #include "content/renderer/media/rtc_video_decoder_factory.h"
 #include "content/renderer/media/rtc_video_encoder_factory.h"
 #include "content/renderer/media/webaudio_capturer_source.h"
 #include "content/renderer/media/webrtc/media_stream_remote_audio_track.h"
 #include "content/renderer/media/webrtc/stun_field_trial.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
@@ skipping 446 matching lines @@
   std::unique_ptr<P2PPortAllocator> port_allocator(new P2PPortAllocator(
       p2p_socket_dispatcher_, std::move(network_manager), socket_factory_.get(),
       port_config, requesting_origin, chrome_worker_thread_.task_runner()));
 
   return GetPcFactory()
       ->CreatePeerConnection(config, std::move(port_allocator),
                              std::move(identity_store), observer)
       .get();
 }
 
-// static
-rtc::scoped_refptr<rtc::RTCCertificate>
-PeerConnectionDependencyFactory::GenerateDefaultCertificate() {
-  std::unique_ptr<rtc::SSLIdentity> identity(rtc::SSLIdentity::Generate(
-      webrtc::kIdentityName, rtc::KeyParams::ECDSA(rtc::EC_NIST_P256)));
-  return rtc::RTCCertificate::Create(std::move(identity));
-}
-
 scoped_refptr<webrtc::MediaStreamInterface>
 PeerConnectionDependencyFactory::CreateLocalMediaStream(
     const std::string& label) {
   return GetPcFactory()->CreateLocalMediaStream(label).get();
 }
 
 scoped_refptr<webrtc::AudioSourceInterface>
 PeerConnectionDependencyFactory::CreateLocalAudioSource(
     const cricket::AudioOptions& options) {
   scoped_refptr<webrtc::AudioSourceInterface> source =
@@ skipping 255 matching lines @@
 }
 
 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
   if (audio_device_.get())
     return;
 
   audio_device_ = new WebRtcAudioDeviceImpl();
 }
 
 }  // namespace content