An example of Unity native plugin of webrtc for Windows OS

Unity native plugin has to use Pinvoke technology in its APIs This plugin dll can also be used by Windows C# applications other than Unity. BUG=webrtc:7389 Review-Url: https://codereview.webrtc.org/2823783002 Cr-Commit-Position: refs/heads/master@{#18108}
2017-05-11 16:10:03 -07:00
parent 38605965bd
commit ad7cad8aba
7 changed files with 1146 additions and 0 deletions
--- a/webrtc/examples/unityplugin/simple_peer_connection.h
+++ b/webrtc/examples/unityplugin/simple_peer_connection.h
@ -0,0 +1,125 @@
+/*
+ *  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"
+
+class SimplePeerConnection : public webrtc::PeerConnectionObserver,
+                             public webrtc::CreateSessionDescriptionObserver,
+                             public webrtc::DataChannelObserver,
+                             public webrtc::AudioTrackSinkInterface {
+ public:
+  SimplePeerConnection() {}
+  ~SimplePeerConnection() {}
+
+  bool InitializePeerConnection(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 RegisterOnVideoFramReady(VIDEOFRAMEREADY_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 ReceivedSdp(const char* sdp);
+  bool ReceivedIceCandidate(const char* ice_candidate);
+
+  bool SetHeadPosition(float x, float y, float z);
+  bool SetHeadRotation(float rx, float ry, float rz, float rw);
+  bool SetRemoteAudioPosition(float x, float y, float z);
+  bool SetRemoteAudioRotation(float rx, float ry, float rz, float rw);
+
+ protected:
+  bool CreatePeerConnection(bool 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_;
+
+  webrtc::MediaStreamInterface* remote_stream_ = nullptr;
+
+  VIDEOFRAMEREADY_CALLBACK OnVideoFrameReady = nullptr;
+  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_