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/peerconnectionfactory.h b/webrtc/api/peerconnectionfactory.h
new file mode 100644
index 0000000..7011736
--- /dev/null
+++ b/webrtc/api/peerconnectionfactory.h
@@ -0,0 +1,132 @@
+/*
+ * libjingle
+ * Copyright 2011 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_PEERCONNECTIONFACTORY_H_
+#define WEBRTC_API_PEERCONNECTIONFACTORY_H_
+
+#include <string>
+
+#include "talk/session/media/channelmanager.h"
+#include "webrtc/api/dtlsidentitystore.h"
+#include "webrtc/api/mediacontroller.h"
+#include "webrtc/api/mediastreaminterface.h"
+#include "webrtc/api/peerconnectioninterface.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/thread.h"
+
+namespace rtc {
+class BasicNetworkManager;
+class BasicPacketSocketFactory;
+}
+
+namespace webrtc {
+
+typedef rtc::RefCountedObject<DtlsIdentityStoreImpl>
+    RefCountedDtlsIdentityStore;
+
+class PeerConnectionFactory : public PeerConnectionFactoryInterface {
+ public:
+  virtual void SetOptions(const Options& options) {
+    options_ = options;
+  }
+
+  rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
+      const PeerConnectionInterface::RTCConfiguration& configuration,
+      const MediaConstraintsInterface* constraints,
+      rtc::scoped_ptr<cricket::PortAllocator> allocator,
+      rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+      PeerConnectionObserver* observer) override;
+
+  bool Initialize();
+
+  rtc::scoped_refptr<MediaStreamInterface>
+      CreateLocalMediaStream(const std::string& label) override;
+
+  rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
+      const MediaConstraintsInterface* constraints) override;
+
+  rtc::scoped_refptr<VideoSourceInterface> CreateVideoSource(
+      cricket::VideoCapturer* capturer,
+      const MediaConstraintsInterface* constraints) override;
+
+  rtc::scoped_refptr<VideoTrackInterface>
+      CreateVideoTrack(const std::string& id,
+                       VideoSourceInterface* video_source) override;
+
+  rtc::scoped_refptr<AudioTrackInterface>
+      CreateAudioTrack(const std::string& id,
+                       AudioSourceInterface* audio_source) override;
+
+  bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
+  void StopAecDump() override;
+  bool StartRtcEventLog(rtc::PlatformFile file) override;
+  void StopRtcEventLog() override;
+
+  virtual webrtc::MediaControllerInterface* CreateMediaController() const;
+  virtual rtc::Thread* signaling_thread();
+  virtual rtc::Thread* worker_thread();
+  const Options& options() const { return options_; }
+
+ protected:
+  PeerConnectionFactory();
+  PeerConnectionFactory(
+      rtc::Thread* worker_thread,
+      rtc::Thread* signaling_thread,
+      AudioDeviceModule* default_adm,
+      cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
+      cricket::WebRtcVideoDecoderFactory* video_decoder_factory);
+  virtual ~PeerConnectionFactory();
+
+ private:
+  cricket::MediaEngineInterface* CreateMediaEngine_w();
+
+  bool owns_ptrs_;
+  bool wraps_current_thread_;
+  rtc::Thread* signaling_thread_;
+  rtc::Thread* worker_thread_;
+  Options options_;
+  // External Audio device used for audio playback.
+  rtc::scoped_refptr<AudioDeviceModule> default_adm_;
+  rtc::scoped_ptr<cricket::ChannelManager> channel_manager_;
+  // External Video encoder factory. This can be NULL if the client has not
+  // injected any. In that case, video engine will use the internal SW encoder.
+  rtc::scoped_ptr<cricket::WebRtcVideoEncoderFactory>
+      video_encoder_factory_;
+  // External Video decoder factory. This can be NULL if the client has not
+  // injected any. In that case, video engine will use the internal SW decoder.
+  rtc::scoped_ptr<cricket::WebRtcVideoDecoderFactory>
+      video_decoder_factory_;
+  rtc::scoped_ptr<rtc::BasicNetworkManager> default_network_manager_;
+  rtc::scoped_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_;
+
+  rtc::scoped_refptr<RefCountedDtlsIdentityStore> dtls_identity_store_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_API_PEERCONNECTIONFACTORY_H_