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/unity_plugin_apis.h

@@ -0,0 +1,105 @@
+/*
+ *  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.
+ */
+
+// This file provides an example of unity native plugin APIs.
+
+#ifndef WEBRTC_EXAMPLES_UNITYPLUGIN_UNITY_PLUGIN_APIS_H_
+#define WEBRTC_EXAMPLES_UNITYPLUGIN_UNITY_PLUGIN_APIS_H_
+
+#include <stdint.h>
+
+// Definitions of callback functions.
+typedef void (*I420FRAMEREADY_CALLBACK)(const uint8_t* data_y,
+                                        const uint8_t* data_u,
+                                        const uint8_t* data_v,
+                                        int stride_y,
+                                        int stride_u,
+                                        int stride_v,
+                                        uint32_t width,
+                                        uint32_t height);
+typedef void (*LOCALDATACHANNELREADY_CALLBACK)();
+typedef void (*DATAFROMEDATECHANNELREADY_CALLBACK)(const char* msg);
+typedef void (*FAILURE_CALLBACK)(const char* msg);
+typedef void (*LOCALSDPREADYTOSEND_CALLBACK)(const char* type, const char* sdp);
+typedef void (*ICECANDIDATEREADYTOSEND_CALLBACK)(const char* candidate,
+                                                 const int sdp_mline_index,
+                                                 const char* sdp_mid);
+typedef void (*AUDIOBUSREADY_CALLBACK)(const void* audio_data,
+                                       int bits_per_sample,
+                                       int sample_rate,
+                                       int number_of_channels,
+                                       int number_of_frames);
+
+#if defined(WEBRTC_WIN)
+#define WEBRTC_PLUGIN_API __declspec(dllexport)
+#elif defined(WEBRTC_ANDROID)
+#define WEBRTC_PLUGIN_API __attribute__((visibility("default")))
+#endif
+extern "C" {
+// Create a peerconnection and return a unique peer connection id.
+WEBRTC_PLUGIN_API int CreatePeerConnection(const char** turn_urls,
+                                           const int no_of_urls,
+                                           const char* username,
+                                           const char* credential);
+// Close a peerconnection.
+WEBRTC_PLUGIN_API bool ClosePeerConnection(int peer_connection_id);
+// Add a audio stream. If audio_only is true, the stream only has an audio
+// track and no video track.
+WEBRTC_PLUGIN_API bool AddStream(int peer_connection_id, bool audio_only);
+// Add a data channel to peer connection.
+WEBRTC_PLUGIN_API bool AddDataChannel(int peer_connection_id);
+// Create a peer connection offer.
+WEBRTC_PLUGIN_API bool CreateOffer(int peer_connection_id);
+// Create a peer connection answer.
+WEBRTC_PLUGIN_API bool CreateAnswer(int peer_connection_id);
+// Send data through data channel.
+WEBRTC_PLUGIN_API bool SendDataViaDataChannel(int peer_connection_id,
+                                              const char* data);
+// Set audio control. If is_mute=true, no audio will playout. If is_record=true,
+// AUDIOBUSREADY_CALLBACK will be called every 10 ms.
+WEBRTC_PLUGIN_API bool SetAudioControl(int peer_connection_id,
+                                       bool is_mute,
+                                       bool is_record);
+// Set remote sdp.
+WEBRTC_PLUGIN_API bool SetRemoteDescription(int peer_connection_id,
+                                            const char* type,
+                                            const char* sdp);
+// Add ice candidate.
+WEBRTC_PLUGIN_API bool AddIceCandidate(const int peer_connection_id,
+                                       const char* candidate,
+                                       const int sdp_mlineindex,
+                                       const char* sdp_mid);
+
+// Register callback functions.
+WEBRTC_PLUGIN_API bool RegisterOnLocalI420FrameReady(
+    int peer_connection_id,
+    I420FRAMEREADY_CALLBACK callback);
+WEBRTC_PLUGIN_API bool RegisterOnRemoteI420FrameReady(
+    int peer_connection_id,
+    I420FRAMEREADY_CALLBACK callback);
+WEBRTC_PLUGIN_API bool RegisterOnLocalDataChannelReady(
+    int peer_connection_id,
+    LOCALDATACHANNELREADY_CALLBACK callback);
+WEBRTC_PLUGIN_API bool RegisterOnDataFromDataChannelReady(
+    int peer_connection_id,
+    DATAFROMEDATECHANNELREADY_CALLBACK callback);
+WEBRTC_PLUGIN_API bool RegisterOnFailure(int peer_connection_id,
+                                         FAILURE_CALLBACK callback);
+WEBRTC_PLUGIN_API bool RegisterOnAudioBusReady(int peer_connection_id,
+                                               AUDIOBUSREADY_CALLBACK callback);
+WEBRTC_PLUGIN_API bool RegisterOnLocalSdpReadytoSend(
+    int peer_connection_id,
+    LOCALSDPREADYTOSEND_CALLBACK callback);
+WEBRTC_PLUGIN_API bool RegisterOnIceCandiateReadytoSend(
+    int peer_connection_id,
+    ICECANDIDATEREADYTOSEND_CALLBACK callback);
+}
+
+#endif  // WEBRTC_EXAMPLES_UNITYPLUGIN_UNITY_PLUGIN_APIS_H_