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/*
* 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 EXAMPLES_UNITYPLUGIN_SIMPLE_PEER_CONNECTION_H_
#define EXAMPLES_UNITYPLUGIN_SIMPLE_PEER_CONNECTION_H_
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "api/data_channel_interface.h"
#include "api/media_stream_interface.h"
#include "api/peer_connection_interface.h"
#include "examples/unityplugin/unity_plugin_apis.h"
#include "examples/unityplugin/video_observer.h"
class SimplePeerConnection : public webrtc::PeerConnectionObserver,
public webrtc::CreateSessionDescriptionObserver,
public webrtc::DataChannelObserver,
public webrtc::AudioTrackSinkInterface {
public:
SimplePeerConnection() {}
~SimplePeerConnection() {}
bool InitializePeerConnection(const char** turn_urls,
int no_of_urls,
const char* username,
const char* credential,
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 RegisterOnLocalI420FrameReady(I420FRAMEREADY_CALLBACK callback);
void RegisterOnRemoteI420FrameReady(I420FRAMEREADY_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 RegisterOnIceCandidateReadytoSend(
ICECANDIDATEREADYTOSEND_CALLBACK callback);
bool SetRemoteDescription(const char* type, const char* sdp);
bool AddIceCandidate(const char* sdp,
int sdp_mlineindex,
const char* sdp_mid);
protected:
// create a peerconneciton and add the turn servers info to the configuration.
bool CreatePeerConnection(const char** turn_urls,
int no_of_urls,
const char* username,
const char* credential);
void CloseDataChannel();
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(webrtc::RTCError 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_;
std::unique_ptr<VideoObserver> local_video_observer_;
std::unique_ptr<VideoObserver> remote_video_observer_;
rtc::scoped_refptr<webrtc::MediaStreamInterface> remote_stream_ = nullptr;
webrtc::PeerConnectionInterface::RTCConfiguration config_;
LOCALDATACHANNELREADY_CALLBACK OnLocalDataChannelReady = nullptr;
DATAFROMEDATECHANNELREADY_CALLBACK OnDataFromDataChannelReady = nullptr;
FAILURE_CALLBACK OnFailureMessage = nullptr;
AUDIOBUSREADY_CALLBACK OnAudioReady = nullptr;
LOCALSDPREADYTOSEND_CALLBACK OnLocalSdpReady = nullptr;
ICECANDIDATEREADYTOSEND_CALLBACK OnIceCandidateReady = nullptr;
bool is_mute_audio_ = false;
bool is_record_audio_ = false;
bool mandatory_receive_ = false;
// disallow copy-and-assign
SimplePeerConnection(const SimplePeerConnection&) = delete;
SimplePeerConnection& operator=(const SimplePeerConnection&) = delete;
};
#endif // EXAMPLES_UNITYPLUGIN_SIMPLE_PEER_CONNECTION_H_
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