mediasoup-demo-v3中添加flexfec需要RtpStreamSend.hpp如何修改

要在mediasoup-demo-v3中添加FlexFEC，需要修改RtpStreamSend.hpp文件中的代码。具体的步骤如下：

1. 打开mediasoup-demo-v3源代码目录中的`worker/src/RtpStreamSend.hpp`文件。

2. 在该文件中找到如下代码：

class RtpStreamSend :
	public RTC::VideoSinkInterface<webrtc::VideoFrame>,
	public RTC::AudioSinkInterface<webrtc::AudioFrame>,
	public webrtc::Transport, public RtcpProcessor::Listener
{
public:
	RtpStreamSend(
		const RTC::scoped_refptr<webrtc::PeerConnectionFactoryInterface>& peerConnectionFactory,
		const RTC::scoped_refptr<webrtc::AudioTrackInterface>& audioTrack,
		const RTC::scoped_refptr<webrtc::VideoTrackInterface>& videoTrack,
		const RTC::scoped_refptr<webrtc::AudioEncoderFactory>& audioEncoderFactory,
		const RTC::scoped_refptr<webrtc::VideoEncoderFactory>& videoEncoderFactory,
		const RTC::scoped_refptr<webrtc::AudioDecoderFactory>& audioDecoderFactory,
		const RTC::scoped_refptr<webrtc::VideoDecoderFactory>& videoDecoderFactory,
		const RTC::scoped_refptr<webrtc::AudioProcessing>& audioProcessing,
		const RTC::scoped_refptr<webrtc::TaskQueueFactory>& taskQueueFactory,
		const RTC::scoped_refptr<webrtc::Call>& call,
		std::function<void(const uint8_t*, size_t)> sendVideoCallback,
		std::function<void(const uint8_t*, size_t)> sendAudioCallback,
		std::function<void(const uint8_t*, size_t)> sendRtpCallback,
		std::function<void(const uint8_t*, size_t)> sendRtcpCallback,
		std::function<void(const uint8_t*, size_t)> sendSrtpCallback,
		std::function<void(const uint8_t*, size_t)> sendSrtcpCallback,
		const Json::Value& peerCapabilities,
		const Json::Value& appData);

	~RtpStreamSend();

	// Implementations of webrtc::Transport.
	bool SendRtp(const uint8_t* data, size_t len, const webrtc::PacketOptions& options) override;
	bool SendRtcp(const uint8_t* data, size_t len) override;

private:
	// ...
};

3. 在该代码中添加FlexFEC相关的成员变量和方法，如下所示：

class RtpStreamSend :
	public RTC::VideoSinkInterface<webrtc::VideoFrame>,
	public RTC::AudioSinkInterface<webrtc::AudioFrame>,
	public webrtc::Transport, public RtcpProcessor::Listener
{
public:
	RtpStreamSend(
		// ...
	);

	~RtpStreamSend();

	// Implementations of webrtc::Transport.
	bool SendRtp(const uint8_t* data, size_t len, const webrtc::PacketOptions& options) override;
	bool SendRtcp(const uint8_t* data, size_t len) override;

	// FlexFEC related members and methods
private:
	std::unique_ptr<webrtc::FlexfecSender> _flexfecSender;
	std::map<uint32_t, std::unique_ptr<webrtc::RtpPacketToSend>> _flexfecPackets;

	void handleFlexfecPackets(std::map<uint32_t, std::unique_ptr<webrtc::RtpPacketToSend>>& packets);
	void sendFlexfecPacket(std::unique_ptr<webrtc::RtpPacketToSend> packet);
};

4. 在该代码中实现FlexFEC相关的方法，如下所示：

void RtpStreamSend::handleFlexfecPackets(std::map<uint32_t, std::unique_ptr<webrtc::RtpPacketToSend>>& packets)
{
    if (_flexfecSender)
    {
        std::vector<webrtc::RtpSequenceNumberMap::SequenceNumberWithTimestamp> fec_packets;

        for (auto& kv : packets)
        {
            std::unique_ptr<webrtc::RtpPacketToSend>& packet = kv.second;
            const webrtc::RtpPacket& rtp_packet = packet->packet;

            // Add packet to FlexFEC
            if (_flexfecSender->AddRtpPacket(rtp_packet) == webrtc::FlexfecSender::kSuccess)
            {
                fec_packets.push_back(
                    webrtc::RtpSequenceNumberMap::SequenceNumberWithTimestamp{
                        rtp_packet.SequenceNumber(), rtp_packet.Timestamp()});
            }
        }

        if (!fec_packets.empty())
        {
            // Generate FEC packet
            webrtc::RtpPacketToSend fec_packet(_flexfecSender->BuildFlexfecPacket(fec_packets));
            if (fec_packet.size() > 0)
            {
                sendFlexfecPacket(std::make_unique<webrtc::RtpPacketToSend>(std::move(fec_packet)));
            }
        }
    }
}

void RtpStreamSend::sendFlexfecPacket(std::unique_ptr<webrtc::RtpPacketToSend> packet)
{
    const uint8_t* data = packet->data();
    size_t length = packet->size();
    
    // Send FlexFEC packet via transport
    if (_sendRtpCallback)
    {
        _sendRtpCallback(data, length);
    }
}

bool RtpStreamSend::SendRtp(const uint8_t* data, size_t len, const webrtc::PacketOptions& options)
{
    // Create RTP packet
    webrtc::RtpPacketToSend packet(&_rtpConfig);
    if (!packet.Parse(data, len))
    {
        RTC_LOG(LS_ERROR) << "Failed to parse RTP packet.";
        return false;
    }

    // Check if packet is FEC
    if (packet.PayloadType() == _flexfecPayloadType)
    {
        // Add packet to FlexFEC buffer
        uint32_t ssrc = packet.Ssrc();
        _flexfecPackets[ssrc] = std::make_unique<webrtc::RtpPacketToSend>(packet);
        handleFlexfecPackets(_flexfecPackets);
        _flexfecPackets.erase(ssrc);
    }
    else
    {
        // Send RTP packet via transport
        if (_sendRtpCallback)
        {
            _sendRtpCallback(data, len);
        }
    }

    return true;
}

bool RtpStreamSend::SendRtcp(const uint8_t* data, size_t len)
{
    // ...
}

5. 保存文件并重新编译mediasoup-demo-v3。

注意：上述代码只是添加了FlexFEC的基本实现，具体的实现方式可能根据实际情况有所不同。此外，代码中的`_flexfecPayloadType`需要根据实际情况设置为正确的FlexFEC负载类型。