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| 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 SimplePeerConnection : public webrtc::PeerConnectionObserver, |
| 25 public webrtc::CreateSessionDescriptionObserver, |
| 26 public webrtc::DataChannelObserver, |
| 27 public webrtc::AudioTrackSinkInterface { |
| 28 public: |
| 29 SimplePeerConnection() {} |
| 30 ~SimplePeerConnection() {} |
| 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 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; |
| 102 rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel_; |
| 103 std::map<std::string, rtc::scoped_refptr<webrtc::MediaStreamInterface> > |
| 104 active_streams_; |
| 105 |
| 106 webrtc::MediaStreamInterface* remote_stream_ = nullptr; |
| 107 |
| 108 VIDEOFRAMEREADY_CALLBACK OnVideoFrameReady = nullptr; |
| 109 LOCALDATACHANNELREADY_CALLBACK OnLocalDataChannelReady = nullptr; |
| 110 DATAFROMEDATECHANNELREADY_CALLBACK OnDataFromDataChannelReady = nullptr; |
| 111 FAILURE_CALLBACK OnFailureMessage = nullptr; |
| 112 AUDIOBUSREADY_CALLBACK OnAudioReady = nullptr; |
| 113 |
| 114 LOCALSDPREADYTOSEND_CALLBACK OnLocalSdpReady = nullptr; |
| 115 ICECANDIDATEREADYTOSEND_CALLBACK OnIceCandiateReady = nullptr; |
| 116 |
| 117 bool is_mute_audio_ = false; |
| 118 bool is_record_audio_ = false; |
| 119 |
| 120 // disallow copy-and-assign |
| 121 SimplePeerConnection(const SimplePeerConnection&) = delete; |
| 122 SimplePeerConnection& operator=(const SimplePeerConnection&) = delete; |
| 123 }; |
| 124 |
| 125 #endif // WEBRTC_EXAMPLES_UNITYPLUGIN_SIMPLE_PEER_CONNECTION_H_ |
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