Chromium Code Reviews| OLD | NEW |
|---|---|
| (Empty) | |
| 1 /* | |
| 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. | |
| 3 * | |
| 4 * Use of this source code is governed by a BSD-style license | |
| 5 * that can be found in the LICENSE file in the root of the source | |
| 6 * tree. An additional intellectual property rights grant can be found | |
| 7 * in the file PATENTS. All contributing project authors may | |
| 8 * be found in the AUTHORS file in the root of the source tree. | |
| 9 */ | |
| 10 | |
| 11 #ifndef WEBRTC_EXAMPLES_UNITYPLUGIN_SIMPLE_PEER_CONNECTION_H_ | |
| 12 #define WEBRTC_EXAMPLES_UNITYPLUGIN_SIMPLE_PEER_CONNECTION_H_ | |
| 13 | |
| 14 #include <map> | |
| 15 #include <memory> | |
| 16 #include <string> | |
| 17 #include <vector> | |
| 18 | |
| 19 #include "webrtc/api/datachannelinterface.h" | |
| 20 #include "webrtc/api/mediastreaminterface.h" | |
| 21 #include "webrtc/api/peerconnectioninterface.h" | |
| 22 #include "webrtc/examples/unityplugin/unity_plugin_apis.h" | |
| 23 | |
| 24 class Conductor : public webrtc::PeerConnectionObserver, | |
| 25 public webrtc::CreateSessionDescriptionObserver, | |
| 26 public webrtc::DataChannelObserver, | |
| 27 public webrtc::AudioTrackSinkInterface { | |
| 28 public: | |
| 29 Conductor() {} | |
| 30 ~Conductor() {} | |
| 31 | |
| 32 bool InitializePeerConnection(bool is_receiver); | |
| 33 void DeletePeerConnection(); | |
| 34 void AddStreams(bool audio_only); | |
| 35 bool CreateDataChannel(); | |
| 36 bool CreateOffer(); | |
| 37 bool CreateAnswer(); | |
| 38 bool SendDataViaDataChannel(const std::string& data); | |
| 39 void SetAudioControl(bool is_mute, bool is_record); | |
| 40 | |
| 41 // Register callback functions. | |
| 42 void RegisterOnVideoFramReady(VIDEOFRAMEREADY_CALLBACK callback); | |
| 43 void RegisterOnLocalDataChannelReady(LOCALDATACHANNELREADY_CALLBACK callback); | |
| 44 void RegisterOnDataFromDataChannelReady( | |
| 45 DATAFROMEDATECHANNELREADY_CALLBACK callback); | |
| 46 void RegisterOnFailure(FAILURE_CALLBACK callback); | |
| 47 void RegisterOnAudioBusReady(AUDIOBUSREADY_CALLBACK callback); | |
| 48 void RegisterOnLocalSdpReadytoSend(LOCALSDPREADYTOSEND_CALLBACK callback); | |
| 49 void RegisterOnIceCandiateReadytoSend( | |
| 50 ICECANDIDATEREADYTOSEND_CALLBACK callback); | |
| 51 bool ReceivedSdp(const char* sdp); | |
| 52 bool ReceivedIceCandidate(const char* ice_candidate); | |
| 53 | |
| 54 bool SetHeadPosition(float x, float y, float z); | |
| 55 bool SetHeadRotation(float rx, float ry, float rz, float rw); | |
| 56 bool SetRemoteAudioPosition(float x, float y, float z); | |
| 57 bool SetRemoteAudioRotation(float rx, float ry, float rz, float rw); | |
| 58 | |
| 59 protected: | |
| 60 bool CreatePeerConnection(bool receiver); | |
| 61 void CloseDataChannel(); | |
| 62 std::unique_ptr<cricket::VideoCapturer> OpenVideoCaptureDevice(); | |
| 63 void SetAudioControl(); | |
| 64 | |
| 65 // PeerConnectionObserver implementation. | |
| 66 void OnSignalingChange( | |
| 67 webrtc::PeerConnectionInterface::SignalingState new_state) override{}; | |
| 68 void OnAddStream( | |
| 69 rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override; | |
| 70 void OnRemoveStream( | |
| 71 rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override{}; | |
| 72 void OnDataChannel( | |
| 73 rtc::scoped_refptr<webrtc::DataChannelInterface> channel) override; | |
| 74 void OnRenegotiationNeeded() override {} | |
| 75 void OnIceConnectionChange( | |
| 76 webrtc::PeerConnectionInterface::IceConnectionState new_state) override{}; | |
| 77 void OnIceGatheringChange( | |
| 78 webrtc::PeerConnectionInterface::IceGatheringState new_state) override{}; | |
| 79 void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override; | |
| 80 void OnIceConnectionReceivingChange(bool receiving) override{}; | |
| 81 | |
| 82 // CreateSessionDescriptionObserver implementation. | |
| 83 void OnSuccess(webrtc::SessionDescriptionInterface* desc) override; | |
| 84 void OnFailure(const std::string& error) override; | |
| 85 | |
| 86 // DataChannelObserver implementation. | |
| 87 void OnStateChange() override; | |
| 88 void OnMessage(const webrtc::DataBuffer& buffer) override; | |
| 89 | |
| 90 // AudioTrackSinkInterface implementation. | |
| 91 void OnData(const void* audio_data, | |
| 92 int bits_per_sample, | |
| 93 int sample_rate, | |
| 94 size_t number_of_channels, | |
| 95 size_t number_of_frames) override; | |
| 96 | |
| 97 // Get remote audio tracks ssrcs. | |
| 98 std::vector<uint32_t> GetRemoteAudioTrackSsrcs(); | |
| 99 | |
| 100 private: | |
| 101 static int peer_count_; | |
|
tommi
2017/04/21 09:47:51
should these static vars perhaps be in an anonymou
GeorgeZ
2017/04/21 22:00:04
I moved those static variables to the cc file.
Cl
| |
| 102 static std::unique_ptr<rtc::Thread> worker_thread_; | |
| 103 static std::unique_ptr<rtc::Thread> signaling_thread_; | |
| 104 static rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> | |
| 105 peer_connection_factory_; | |
| 106 | |
| 107 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_ = | |
|
tommi
2017/04/21 09:47:51
initializing scoped_refptr like this, isn't necess
GeorgeZ
2017/04/21 22:00:04
Done.
| |
| 108 nullptr; | |
| 109 rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel_ = nullptr; | |
| 110 std::map<std::string, rtc::scoped_refptr<webrtc::MediaStreamInterface> > | |
| 111 active_streams_; | |
| 112 | |
| 113 webrtc::MediaStreamInterface* remote_stream_ = nullptr; | |
| 114 | |
| 115 VIDEOFRAMEREADY_CALLBACK OnVideoFrameReady = nullptr; | |
| 116 LOCALDATACHANNELREADY_CALLBACK OnLocalDataChannelReady = nullptr; | |
| 117 DATAFROMEDATECHANNELREADY_CALLBACK OnDataFromDataChannelReady = nullptr; | |
| 118 FAILURE_CALLBACK OnFailureMessage = nullptr; | |
| 119 AUDIOBUSREADY_CALLBACK OnAudioReady = nullptr; | |
| 120 | |
| 121 LOCALSDPREADYTOSEND_CALLBACK OnLocalSdpReady = nullptr; | |
| 122 ICECANDIDATEREADYTOSEND_CALLBACK OnIceCandiateReady = nullptr; | |
| 123 | |
| 124 bool is_mute_audio_ = false; | |
| 125 bool is_record_audio_ = false; | |
| 126 | |
| 127 // disallow copy-and-assign | |
| 128 Conductor(const Conductor&) = delete; | |
| 129 Conductor& operator=(const Conductor&) = delete; | |
| 130 }; | |
| 131 | |
| 132 #endif // WEBRTC_EXAMPLES_UNITYPLUGIN_SIMPLE_PEER_CONNECTION_H_ | |
| OLD | NEW |