Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

examples/unityplugin/simple_peer_connection.h

@@ -0,0 +1,136 @@
+/*
+ *  Copyright (c) 2017 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.
+ */
+
+#ifndef WEBRTC_EXAMPLES_UNITYPLUGIN_SIMPLE_PEER_CONNECTION_H_
+#define WEBRTC_EXAMPLES_UNITYPLUGIN_SIMPLE_PEER_CONNECTION_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "webrtc/api/datachannelinterface.h"
+#include "webrtc/api/mediastreaminterface.h"
+#include "webrtc/api/peerconnectioninterface.h"
+#include "webrtc/examples/unityplugin/unity_plugin_apis.h"
+#include "webrtc/examples/unityplugin/video_observer.h"
+
+class SimplePeerConnection : public webrtc::PeerConnectionObserver,
+                             public webrtc::CreateSessionDescriptionObserver,
+                             public webrtc::DataChannelObserver,
+                             public webrtc::AudioTrackSinkInterface {
+ public:
+  SimplePeerConnection() {}
+  ~SimplePeerConnection() {}
+
+  bool InitializePeerConnection(const char** turn_urls,
+                                const int no_of_urls,
+                                const char* username,
+                                const char* credential,
+                                bool is_receiver);
+  void DeletePeerConnection();
+  void AddStreams(bool audio_only);
+  bool CreateDataChannel();
+  bool CreateOffer();
+  bool CreateAnswer();
+  bool SendDataViaDataChannel(const std::string& data);
+  void SetAudioControl(bool is_mute, bool is_record);
+
+  // Register callback functions.
+  void RegisterOnLocalI420FrameReady(I420FRAMEREADY_CALLBACK callback);
+  void RegisterOnRemoteI420FrameReady(I420FRAMEREADY_CALLBACK callback);
+  void RegisterOnLocalDataChannelReady(LOCALDATACHANNELREADY_CALLBACK callback);
+  void RegisterOnDataFromDataChannelReady(
+      DATAFROMEDATECHANNELREADY_CALLBACK callback);
+  void RegisterOnFailure(FAILURE_CALLBACK callback);
+  void RegisterOnAudioBusReady(AUDIOBUSREADY_CALLBACK callback);
+  void RegisterOnLocalSdpReadytoSend(LOCALSDPREADYTOSEND_CALLBACK callback);
+  void RegisterOnIceCandiateReadytoSend(
+      ICECANDIDATEREADYTOSEND_CALLBACK callback);
+  bool SetRemoteDescription(const char* type, const char* sdp);
+  bool AddIceCandidate(const char* sdp,
+                       const int sdp_mlineindex,
+                       const char* sdp_mid);
+
+ protected:
+  // create a peerconneciton and add the turn servers info to the configuration.
+  bool CreatePeerConnection(const char** turn_urls,
+                            const int no_of_urls,
+                            const char* username,
+                            const char* credential,
+                            bool is_receiver);
+  void CloseDataChannel();
+  std::unique_ptr<cricket::VideoCapturer> OpenVideoCaptureDevice();
+  void SetAudioControl();
+
+  // PeerConnectionObserver implementation.
+  void OnSignalingChange(
+      webrtc::PeerConnectionInterface::SignalingState new_state) override {}
+  void OnAddStream(
+      rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override;
+  void OnRemoveStream(
+      rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override {}
+  void OnDataChannel(
+      rtc::scoped_refptr<webrtc::DataChannelInterface> channel) override;
+  void OnRenegotiationNeeded() override {}
+  void OnIceConnectionChange(
+      webrtc::PeerConnectionInterface::IceConnectionState new_state) override {}
+  void OnIceGatheringChange(
+      webrtc::PeerConnectionInterface::IceGatheringState new_state) override {}
+  void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
+  void OnIceConnectionReceivingChange(bool receiving) override {}
+
+  // CreateSessionDescriptionObserver implementation.
+  void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
+  void OnFailure(const std::string& error) override;
+
+  // DataChannelObserver implementation.
+  void OnStateChange() override;
+  void OnMessage(const webrtc::DataBuffer& buffer) override;
+
+  // AudioTrackSinkInterface implementation.
+  void OnData(const void* audio_data,
+              int bits_per_sample,
+              int sample_rate,
+              size_t number_of_channels,
+              size_t number_of_frames) override;
+
+  // Get remote audio tracks ssrcs.
+  std::vector<uint32_t> GetRemoteAudioTrackSsrcs();
+
+ private:
+  rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
+  rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel_;
+  std::map<std::string, rtc::scoped_refptr<webrtc::MediaStreamInterface> >
+      active_streams_;
+
+  std::unique_ptr<VideoObserver> local_video_observer_;
+  std::unique_ptr<VideoObserver> remote_video_observer_;
+
+  webrtc::MediaStreamInterface* remote_stream_ = nullptr;
+  webrtc::PeerConnectionInterface::RTCConfiguration config_;
+
+  LOCALDATACHANNELREADY_CALLBACK OnLocalDataChannelReady = nullptr;
+  DATAFROMEDATECHANNELREADY_CALLBACK OnDataFromDataChannelReady = nullptr;
+  FAILURE_CALLBACK OnFailureMessage = nullptr;
+  AUDIOBUSREADY_CALLBACK OnAudioReady = nullptr;
+
+  LOCALSDPREADYTOSEND_CALLBACK OnLocalSdpReady = nullptr;
+  ICECANDIDATEREADYTOSEND_CALLBACK OnIceCandiateReady = nullptr;
+
+  bool is_mute_audio_ = false;
+  bool is_record_audio_ = false;
+
+  // disallow copy-and-assign
+  SimplePeerConnection(const SimplePeerConnection&) = delete;
+  SimplePeerConnection& operator=(const SimplePeerConnection&) = delete;
+};
+
+#endif  // WEBRTC_EXAMPLES_UNITYPLUGIN_SIMPLE_PEER_CONNECTION_H_