Remove ScopedVector from content/renderer/.

content/renderer/media/rtc_peer_connection_handler.cc

 // Copyright (c) 2012 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/rtc_peer_connection_handler.h"
 
 #include <string.h>
 
 #include <string>
 #include <utility>
@@ skipping 1499 matching lines @@
   }
 
   return webkit_request.requestSucceeded();
 }
 
 bool RTCPeerConnectionHandler::addStream(
     const blink::WebMediaStream& stream,
     const blink::WebMediaConstraints& options) {
   DCHECK(thread_checker_.CalledOnValidThread());
   TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::addStream");
-  for (ScopedVector<WebRtcMediaStreamAdapter>::iterator adapter_it =
-      local_streams_.begin(); adapter_it != local_streams_.end();
-      ++adapter_it) {
-    if ((*adapter_it)->IsEqual(stream)) {
+  for (const auto& adapter : local_streams_) {
+    if (adapter->IsEqual(stream)) {
       DVLOG(1) << "RTCPeerConnectionHandler::addStream called with the same "
                << "stream twice. id=" << stream.id().utf8();
       return false;
     }
   }
 
   if (peer_connection_tracker_) {
     peer_connection_tracker_->TrackAddStream(
         this, stream, PeerConnectionTracker::SOURCE_LOCAL);
   }
 
   PerSessionWebRTCAPIMetrics::GetInstance()->IncrementStreamCounter();
 
-  WebRtcMediaStreamAdapter* adapter =
-      new WebRtcMediaStreamAdapter(stream, dependency_factory_);
-  local_streams_.push_back(adapter);
+  local_streams_.push_back(
+      base::MakeUnique<WebRtcMediaStreamAdapter>(stream, dependency_factory_));
 
-  webrtc::MediaStreamInterface* webrtc_stream = adapter->webrtc_media_stream();
+  webrtc::MediaStreamInterface* webrtc_stream =
+      local_streams_.back()->webrtc_media_stream();
   track_metrics_.AddStream(MediaStreamTrackMetrics::SENT_STREAM,
                            webrtc_stream);
 
   if (!options.isEmpty()) {
     // TODO(perkj): |mediaConstraints| is the name of the optional constraints
     // argument in RTCPeerConnection.idl. It has been removed from the spec and
     // should be removed from blink as well.
     LOG(WARNING)
         << "mediaConstraints is not a supported argument to addStream.";
     LOG(WARNING) << "mediaConstraints was " << options.toString().utf8();
   }
 
   return native_peer_connection_->AddStream(webrtc_stream);
 }
 
 void RTCPeerConnectionHandler::removeStream(
     const blink::WebMediaStream& stream) {
   DCHECK(thread_checker_.CalledOnValidThread());
   TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::removeStream");
   // Find the webrtc stream.
   scoped_refptr<webrtc::MediaStreamInterface> webrtc_stream;
-  for (ScopedVector<WebRtcMediaStreamAdapter>::iterator adapter_it =
-           local_streams_.begin(); adapter_it != local_streams_.end();
-       ++adapter_it) {
+  for (auto adapter_it = local_streams_.begin();
+       adapter_it != local_streams_.end(); ++adapter_it) {
     if ((*adapter_it)->IsEqual(stream)) {
       webrtc_stream = (*adapter_it)->webrtc_media_stream();
       local_streams_.erase(adapter_it);
       break;
     }
   }
   DCHECK(webrtc_stream.get());
   // TODO(tommi): Make this async (PostTaskAndReply).
   native_peer_connection_->RemoveStream(webrtc_stream.get());
 
@@ skipping 114 matching lines @@
 
 blink::WebRTCDTMFSenderHandler* RTCPeerConnectionHandler::createDTMFSender(
     const blink::WebMediaStreamTrack& track) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(!track.isNull());
   TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createDTMFSender");
   DVLOG(1) << "createDTMFSender.";
 
   // Find the WebRtc track referenced by the blink track's ID.
   webrtc::AudioTrackInterface* webrtc_track = nullptr;
-  for (const WebRtcMediaStreamAdapter* s : local_streams_) {
-    webrtc_track = s->webrtc_media_stream()->FindAudioTrack(track.id().utf8());
+  for (const auto& adapter : local_streams_) {
+    webrtc_track =
+        adapter->webrtc_media_stream()->FindAudioTrack(track.id().utf8());
     if (webrtc_track)
       break;
   }
   if (!webrtc_track) {
     DLOG(ERROR) << "Audio track with ID '" << track.id().utf8()
                 << "' has no known WebRtc sink.";
     return nullptr;
   }
 
   rtc::scoped_refptr<webrtc::DtmfSenderInterface> sender(
@@ skipping 293 matching lines @@
 }
 
 void RTCPeerConnectionHandler::ResetUMAStats() {
   DCHECK(thread_checker_.CalledOnValidThread());
   num_local_candidates_ipv6_ = 0;
   num_local_candidates_ipv4_ = 0;
   ice_connection_checking_start_ = base::TimeTicks();
   memset(ice_state_seen_, 0, sizeof(ice_state_seen_));
 }
 }  // namespace content