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/*
*
* Copyright (c) 2025 Project CHIP Authors
* All rights reserved.
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#include "webrtc-abstract.h"
#include <lib/support/logging/CHIPLogging.h>
#include <rtc/rtc.hpp>
#include <webrtc-transport.h>
SDPType RtcTypeToSDPType(rtc::Description::Type type)
{
switch (type)
{
case rtc::Description::Type::Offer:
return SDPType::Offer;
case rtc::Description::Type::Answer:
return SDPType::Answer;
case rtc::Description::Type::Pranswer:
return SDPType::Pranswer;
case rtc::Description::Type::Rollback:
return SDPType::Rollback;
default:
return SDPType::Offer;
}
}
WebrtcTransport::WebrtcTransport()
{
ChipLogProgress(Camera, "WebrtcTransport created");
mRequestArgs = { 0, 0, 0, 0, 0, 0 }; // Initialize request arguments to zero
}
WebrtcTransport::~WebrtcTransport()
{
ClosePeerConnection();
ChipLogProgress(Camera, "WebrtcTransport destroyed for sessionID: [%u]", mRequestArgs.sessionId);
}
void WebrtcTransport::SetCallbacks(OnTransportLocalDescriptionCallback onLocalDescription,
OnTransportConnectionStateCallback onConnectionState)
{
mOnLocalDescription = onLocalDescription;
mOnConnectionState = onConnectionState;
}
void WebrtcTransport::SetRequestArgs(const RequestArgs & args)
{
mRequestArgs = args;
}
WebrtcTransport::RequestArgs & WebrtcTransport::GetRequestArgs()
{
return mRequestArgs;
}
void WebrtcTransport::SendVideo(const chip::ByteSpan & data, int64_t timestamp, uint16_t videoStreamID)
{
if (mLocalVideoTrack)
{
mLocalVideoTrack->SendFrame(data, timestamp);
}
}
// Implementation of SendAudio method
void WebrtcTransport::SendAudio(const chip::ByteSpan & data, int64_t timestamp, uint16_t audioStreamID)
{
if (mLocalAudioTrack)
{
mLocalAudioTrack->SendFrame(data, timestamp);
}
}
// Implementation of SendAudioVideo method
void WebrtcTransport::SendAudioVideo(const chip::ByteSpan & data, uint16_t videoStreamID, uint16_t audioStreamID)
{
// Placeholder for actual WebRTC implementation to send synchronized audio/video data
}
// Implementation of CanSendVideo method
bool WebrtcTransport::CanSendVideo()
{
return mLocalVideoTrack != nullptr;
}
// Implementation of CanSendAudio method
bool WebrtcTransport::CanSendAudio()
{
return mLocalAudioTrack != nullptr;
}
const char * WebrtcTransport::GetStateStr() const
{
switch (mState)
{
case State::Idle:
return "Idle";
case State::SendingOffer:
return "SendingOffer";
case State::SendingAnswer:
return "SendingAnswer";
case State::SendingICECandidates:
return "SendingICECandidates";
case State::SendingEnd:
return "SendingEnd";
}
return "N/A";
}
void WebrtcTransport::MoveToState(const State targetState)
{
mState = targetState;
ChipLogProgress(Camera, "WebrtcTransport moving to [ %s ]", GetStateStr());
}
void WebrtcTransport::SetCommandType(const CommandType commandtype)
{
mCommandType = commandtype;
}
void WebrtcTransport::Start()
{
if (mPeerConnection.get())
{
ChipLogProgress(Camera, "Start, mPeerConnection is already created");
return;
}
mPeerConnection = CreateWebRTCPeerConnection();
mPeerConnection->SetCallbacks([this](const std::string & sdp, SDPType type) { this->OnLocalDescription(sdp, type); },
[this](const std::string & candidate) { this->OnICECandidate(candidate); },
[this](bool connected) { this->OnConnectionStateChanged(connected); },
[this](std::shared_ptr<WebRTCTrack> track) { this->OnTrack(track); });
}
void WebrtcTransport::Stop()
{
if (mPeerConnection != nullptr)
{
mPeerConnection->Close();
}
mLocalVideoTrack = nullptr;
mLocalAudioTrack = nullptr;
}
void WebrtcTransport::AddVideoTrack(const std::string & videoMid, int payloadType)
{
if (mPeerConnection != nullptr)
{
// Adding local tracks to send video data to remote peer
mLocalVideoTrack = mPeerConnection->AddTrack(MediaType::Video, videoMid, payloadType);
}
}
void WebrtcTransport::AddAudioTrack(const std::string & audioMid, int payloadType)
{
if (mPeerConnection != nullptr)
{
// Adding local tracks to send audio data to remote peer
mLocalAudioTrack = mPeerConnection->AddTrack(MediaType::Audio, audioMid, payloadType);
}
}
void WebrtcTransport::AddRemoteCandidate(const std::string & candidate, const std::string & mid)
{
ChipLogProgress(Camera, "Adding remote candidate for sessionID: %u", mRequestArgs.sessionId);
mPeerConnection->AddRemoteCandidate(candidate, mid);
}
// WebRTC Callbacks
void WebrtcTransport::OnLocalDescription(const std::string & sdp, SDPType type)
{
ChipLogProgress(Camera, "Local description received for sessionID: %u", mRequestArgs.sessionId);
mLocalSdp = sdp;
mLocalSdpType = type;
if (mOnLocalDescription)
mOnLocalDescription(sdp, type, mRequestArgs.sessionId);
}
bool WebrtcTransport::ClosePeerConnection()
{
if (mPeerConnection == nullptr)
{
return false;
}
mPeerConnection->Close();
mPeerConnection.reset();
return true;
}
void WebrtcTransport::OnICECandidate(const std::string & candidate)
{
ChipLogProgress(Camera, "ICE Candidate received for sessionID: %u", mRequestArgs.sessionId);
mLocalCandidates.push_back(candidate);
ChipLogProgress(Camera, "Local Candidate:");
ChipLogProgress(Camera, "%s", candidate.c_str());
}
void WebrtcTransport::OnConnectionStateChanged(bool connected)
{
ChipLogProgress(Camera, "Connection state changed for sessionID: %u", mRequestArgs.sessionId);
if (mOnConnectionState)
mOnConnectionState(connected, mRequestArgs.sessionId);
}
void WebrtcTransport::OnTrack(std::shared_ptr<WebRTCTrack> track)
{
// Only logging the track addition here as it's not used in the current implementation. In future, we can add functionality to
// handle
ChipLogProgress(Camera, "Remote track added for the sessionID: %u, type: %s", mRequestArgs.sessionId, track->GetType().c_str());
}