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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}
125 lines
4.9 KiB
C++
125 lines
4.9 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_EXAMPLES_UNITYPLUGIN_SIMPLE_PEER_CONNECTION_H_
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#define WEBRTC_EXAMPLES_UNITYPLUGIN_SIMPLE_PEER_CONNECTION_H_
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#include <map>
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#include <memory>
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#include <string>
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#include <vector>
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#include "webrtc/api/datachannelinterface.h"
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#include "webrtc/api/mediastreaminterface.h"
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#include "webrtc/api/peerconnectioninterface.h"
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#include "webrtc/examples/unityplugin/unity_plugin_apis.h"
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class SimplePeerConnection : public webrtc::PeerConnectionObserver,
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public webrtc::CreateSessionDescriptionObserver,
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public webrtc::DataChannelObserver,
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public webrtc::AudioTrackSinkInterface {
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public:
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SimplePeerConnection() {}
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~SimplePeerConnection() {}
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bool InitializePeerConnection(bool is_receiver);
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void DeletePeerConnection();
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void AddStreams(bool audio_only);
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bool CreateDataChannel();
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bool CreateOffer();
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bool CreateAnswer();
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bool SendDataViaDataChannel(const std::string& data);
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void SetAudioControl(bool is_mute, bool is_record);
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// Register callback functions.
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void RegisterOnVideoFramReady(VIDEOFRAMEREADY_CALLBACK callback);
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void RegisterOnLocalDataChannelReady(LOCALDATACHANNELREADY_CALLBACK callback);
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void RegisterOnDataFromDataChannelReady(
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DATAFROMEDATECHANNELREADY_CALLBACK callback);
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void RegisterOnFailure(FAILURE_CALLBACK callback);
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void RegisterOnAudioBusReady(AUDIOBUSREADY_CALLBACK callback);
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void RegisterOnLocalSdpReadytoSend(LOCALSDPREADYTOSEND_CALLBACK callback);
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void RegisterOnIceCandiateReadytoSend(
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ICECANDIDATEREADYTOSEND_CALLBACK callback);
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bool ReceivedSdp(const char* sdp);
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bool ReceivedIceCandidate(const char* ice_candidate);
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bool SetHeadPosition(float x, float y, float z);
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bool SetHeadRotation(float rx, float ry, float rz, float rw);
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bool SetRemoteAudioPosition(float x, float y, float z);
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bool SetRemoteAudioRotation(float rx, float ry, float rz, float rw);
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protected:
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bool CreatePeerConnection(bool receiver);
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void CloseDataChannel();
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std::unique_ptr<cricket::VideoCapturer> OpenVideoCaptureDevice();
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void SetAudioControl();
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// PeerConnectionObserver implementation.
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void OnSignalingChange(
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webrtc::PeerConnectionInterface::SignalingState new_state) override {}
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void OnAddStream(
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rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override;
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void OnRemoveStream(
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rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override {}
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void OnDataChannel(
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rtc::scoped_refptr<webrtc::DataChannelInterface> channel) override;
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void OnRenegotiationNeeded() override {}
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void OnIceConnectionChange(
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webrtc::PeerConnectionInterface::IceConnectionState new_state) override {}
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void OnIceGatheringChange(
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webrtc::PeerConnectionInterface::IceGatheringState new_state) override {}
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void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
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void OnIceConnectionReceivingChange(bool receiving) override {}
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// CreateSessionDescriptionObserver implementation.
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void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
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void OnFailure(const std::string& error) override;
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// DataChannelObserver implementation.
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void OnStateChange() override;
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void OnMessage(const webrtc::DataBuffer& buffer) override;
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// AudioTrackSinkInterface implementation.
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void OnData(const void* audio_data,
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int bits_per_sample,
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int sample_rate,
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size_t number_of_channels,
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size_t number_of_frames) override;
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// Get remote audio tracks ssrcs.
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std::vector<uint32_t> GetRemoteAudioTrackSsrcs();
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private:
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rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
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rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel_;
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std::map<std::string, rtc::scoped_refptr<webrtc::MediaStreamInterface> >
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active_streams_;
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webrtc::MediaStreamInterface* remote_stream_ = nullptr;
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VIDEOFRAMEREADY_CALLBACK OnVideoFrameReady = nullptr;
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LOCALDATACHANNELREADY_CALLBACK OnLocalDataChannelReady = nullptr;
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DATAFROMEDATECHANNELREADY_CALLBACK OnDataFromDataChannelReady = nullptr;
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FAILURE_CALLBACK OnFailureMessage = nullptr;
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AUDIOBUSREADY_CALLBACK OnAudioReady = nullptr;
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LOCALSDPREADYTOSEND_CALLBACK OnLocalSdpReady = nullptr;
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ICECANDIDATEREADYTOSEND_CALLBACK OnIceCandiateReady = nullptr;
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bool is_mute_audio_ = false;
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bool is_record_audio_ = false;
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// disallow copy-and-assign
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SimplePeerConnection(const SimplePeerConnection&) = delete;
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SimplePeerConnection& operator=(const SimplePeerConnection&) = delete;
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};
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#endif // WEBRTC_EXAMPLES_UNITYPLUGIN_SIMPLE_PEER_CONNECTION_H_
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