Move talk/app/webrtc to webrtc/api

The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.

License headers will be updated in a follow-up CL.

Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
  except for these files:
  talk/app/webrtc/peerconnectionendtoend_unittest.cc
  talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
  talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
  webrtc/media/devices/win32devicemanager.cc

The HAVE_SCTP define was added for the peerconnection_unittests target
in api_tests.gyp.

I also checked that none of
SRTP_RELATIVE_PATH
HAVE_SRTP
HAVE_WEBRTC_VIDEO
HAVE_WEBRTC_VOICE
were used by the talk/app/webrtc code.

For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle:
https://codereview.chromium.org/1615433002

BUG=webrtc:5418
NOPRESUBMIT=True
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1610243002 .

Cr-Commit-Position: refs/heads/master@{#11545}
diff --git a/webrtc/api/peerconnectionproxy.h b/webrtc/api/peerconnectionproxy.h
new file mode 100644
index 0000000..9faf014
--- /dev/null
+++ b/webrtc/api/peerconnectionproxy.h
@@ -0,0 +1,88 @@
+/*
+ * 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,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef WEBRTC_API_PEERCONNECTIONPROXY_H_
+#define WEBRTC_API_PEERCONNECTIONPROXY_H_
+
+#include "webrtc/api/peerconnectioninterface.h"
+#include "webrtc/api/proxy.h"
+
+namespace webrtc {
+
+// Define proxy for PeerConnectionInterface.
+BEGIN_PROXY_MAP(PeerConnection)
+  PROXY_METHOD0(rtc::scoped_refptr<StreamCollectionInterface>,
+                local_streams)
+  PROXY_METHOD0(rtc::scoped_refptr<StreamCollectionInterface>,
+                remote_streams)
+  PROXY_METHOD1(bool, AddStream, MediaStreamInterface*)
+  PROXY_METHOD1(void, RemoveStream, MediaStreamInterface*)
+  PROXY_METHOD2(rtc::scoped_refptr<RtpSenderInterface>,
+                AddTrack,
+                MediaStreamTrackInterface*,
+                std::vector<MediaStreamInterface*>)
+  PROXY_METHOD1(bool, RemoveTrack, RtpSenderInterface*)
+  PROXY_METHOD1(rtc::scoped_refptr<DtmfSenderInterface>,
+                CreateDtmfSender, AudioTrackInterface*)
+  PROXY_METHOD2(rtc::scoped_refptr<RtpSenderInterface>,
+                CreateSender,
+                const std::string&,
+                const std::string&)
+  PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpSenderInterface>>,
+                     GetSenders)
+  PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpReceiverInterface>>,
+                     GetReceivers)
+  PROXY_METHOD3(bool, GetStats, StatsObserver*,
+                MediaStreamTrackInterface*,
+                StatsOutputLevel)
+  PROXY_METHOD2(rtc::scoped_refptr<DataChannelInterface>,
+                CreateDataChannel, const std::string&, const DataChannelInit*)
+  PROXY_CONSTMETHOD0(const SessionDescriptionInterface*, local_description)
+  PROXY_CONSTMETHOD0(const SessionDescriptionInterface*, remote_description)
+  PROXY_METHOD2(void, CreateOffer, CreateSessionDescriptionObserver*,
+                const MediaConstraintsInterface*)
+  PROXY_METHOD2(void, CreateAnswer, CreateSessionDescriptionObserver*,
+                const MediaConstraintsInterface*)
+  PROXY_METHOD2(void, SetLocalDescription, SetSessionDescriptionObserver*,
+                SessionDescriptionInterface*)
+  PROXY_METHOD2(void, SetRemoteDescription, SetSessionDescriptionObserver*,
+                SessionDescriptionInterface*)
+  PROXY_METHOD1(bool,
+                SetConfiguration,
+                const PeerConnectionInterface::RTCConfiguration&);
+  PROXY_METHOD1(bool, AddIceCandidate, const IceCandidateInterface*)
+  PROXY_METHOD1(void, RegisterUMAObserver, UMAObserver*)
+  PROXY_METHOD0(SignalingState, signaling_state)
+  PROXY_METHOD0(IceState, ice_state)
+  PROXY_METHOD0(IceConnectionState, ice_connection_state)
+  PROXY_METHOD0(IceGatheringState, ice_gathering_state)
+  PROXY_METHOD0(void, Close)
+END_PROXY()
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_API_PEERCONNECTIONPROXY_H_