483 lines
11 KiB
Go
483 lines
11 KiB
Go
/**
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* package mediaserver
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*
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* WebRTC ↔ RTMP 媒体桥接
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* 功能：
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* 1. WebRTC 流转 RTMP（H5用户 -> 小程序用户）
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* 2. RTMP 流转 WebRTC（小程序用户 -> H5用户）
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* 3. 音频转码：Opus ↔ AAC
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* 4. 视频透传：H.264
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*/
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package mediaserver
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import (
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"fmt"
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"log"
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"sync"
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"time"
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"github.com/pion/rtp"
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"github.com/pion/webrtc/v3"
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)
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// Bridge WebRTC-RTMP 媒体桥
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type Bridge struct {
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roomID string
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mu sync.RWMutex
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running bool
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stopChan chan struct{}
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// WebRTC -> RTMP 转换器
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webrtcToRTMP map[string]*WebRTCToRTMPConverter
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// RTMP -> WebRTC 转换器
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rtmpToWebRTC map[string]*RTMPToWebRTCConverter
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}
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// WebRTCToRTMPConverter WebRTC 转 RTMP 转换器
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type WebRTCToRTMPConverter struct {
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UserID string
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StreamID string
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RoomID string
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Running bool
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stopChan chan struct{}
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// 输入: WebRTC Track
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VideoTrack *webrtc.TrackRemote
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AudioTrack *webrtc.TrackRemote
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// 输出: RTMP 流信息
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OutputURL string // RTMP 推流地址
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// 编解码器
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videoCodec string // h264
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audioCodec string // opus -> aac
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// 缓冲区
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videoBuffer chan *rtp.Packet
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audioBuffer chan *rtp.Packet
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}
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// RTMPToWebRTCConverter RTMP 转 WebRTC 转换器
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type RTMPToWebRTCConverter struct {
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UserID string
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StreamID string
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RoomID string
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Running bool
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stopChan chan struct{}
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// 输入: RTMP 拉流地址
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InputURL string
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// 输出: WebRTC Track
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VideoTrack *webrtc.TrackLocalStaticRTP
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AudioTrack *webrtc.TrackLocalStaticRTP
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// 编解码器
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videoCodec string // h264
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audioCodec string // aac -> opus
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}
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// BridgeManager 桥接管理器
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type BridgeManager struct {
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bridges map[string]*Bridge // roomID -> bridge
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mu sync.RWMutex
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}
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// 全局桥接管理器
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var bridgeManager *BridgeManager
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var bridgeOnce sync.Once
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// GetBridgeManager 获取桥接管理器单例
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func GetBridgeManager() *BridgeManager {
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bridgeOnce.Do(func() {
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bridgeManager = &BridgeManager{
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bridges: make(map[string]*Bridge),
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}
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})
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return bridgeManager
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}
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// GetOrCreateBridge 获取或创建房间的桥接
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func (bm *BridgeManager) GetOrCreateBridge(roomID string) *Bridge {
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bm.mu.Lock()
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defer bm.mu.Unlock()
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if bridge, exists := bm.bridges[roomID]; exists {
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return bridge
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}
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bridge := NewBridge(roomID)
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bm.bridges[roomID] = bridge
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return bridge
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}
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// RemoveBridge 移除房间的桥接
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func (bm *BridgeManager) RemoveBridge(roomID string) {
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bm.mu.Lock()
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defer bm.mu.Unlock()
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if bridge, exists := bm.bridges[roomID]; exists {
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bridge.Stop()
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delete(bm.bridges, roomID)
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}
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}
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// GetBridge 获取桥接
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func (bm *BridgeManager) GetBridge(roomID string) *Bridge {
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bm.mu.RLock()
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defer bm.mu.RUnlock()
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return bm.bridges[roomID]
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}
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// NewBridge 创建新桥接
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func NewBridge(roomID string) *Bridge {
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return &Bridge{
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roomID: roomID,
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stopChan: make(chan struct{}),
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webrtcToRTMP: make(map[string]*WebRTCToRTMPConverter),
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rtmpToWebRTC: make(map[string]*RTMPToWebRTCConverter),
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}
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}
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// Start 启动桥接
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func (b *Bridge) Start() {
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b.mu.Lock()
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defer b.mu.Unlock()
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if b.running {
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return
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}
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b.running = true
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log.Printf("🌉 [Bridge] 启动房间桥接 | Room:%s", b.roomID)
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}
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// Stop 停止桥接
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func (b *Bridge) Stop() {
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b.mu.Lock()
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defer b.mu.Unlock()
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if !b.running {
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return
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}
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// 关闭停止信号
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close(b.stopChan)
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// 停止所有 WebRTC -> RTMP 转换器
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for _, converter := range b.webrtcToRTMP {
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converter.Stop()
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}
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b.webrtcToRTMP = make(map[string]*WebRTCToRTMPConverter)
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// 停止所有 RTMP -> WebRTC 转换器
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for _, converter := range b.rtmpToWebRTC {
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converter.Stop()
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}
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b.rtmpToWebRTC = make(map[string]*RTMPToWebRTCConverter)
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b.running = false
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log.Printf("🌉 [Bridge] 停止房间桥接 | Room:%s", b.roomID)
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}
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// AddWebRTCToRTMP 添加 WebRTC 到 RTMP 的转换
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func (b *Bridge) AddWebRTCToRTMP(userID string, videoTrack, audioTrack *webrtc.TrackRemote, outputURL string) (*WebRTCToRTMPConverter, error) {
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b.mu.Lock()
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defer b.mu.Unlock()
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if !b.running {
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b.running = true
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}
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streamID := fmt.Sprintf("w2r_%s_%s_%d", b.roomID, userID, time.Now().UnixNano())
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converter := &WebRTCToRTMPConverter{
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UserID: userID,
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StreamID: streamID,
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RoomID: b.roomID,
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Running: true,
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stopChan: make(chan struct{}),
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VideoTrack: videoTrack,
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AudioTrack: audioTrack,
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OutputURL: outputURL,
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videoCodec: "h264",
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audioCodec: "opus",
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videoBuffer: make(chan *rtp.Packet, 100),
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audioBuffer: make(chan *rtp.Packet, 100),
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}
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b.webrtcToRTMP[userID] = converter
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// 启动转换协程
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go converter.Run()
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log.Printf("🌉 [Bridge] 添加 WebRTC->RTMP | Room:%s User:%s Stream:%s", b.roomID, userID, streamID)
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return converter, nil
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}
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// AddRTMPToWebRTC 添加 RTMP 到 WebRTC 的转换
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func (b *Bridge) AddRTMPToWebRTC(userID string, inputURL string) (*RTMPToWebRTCConverter, error) {
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b.mu.Lock()
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defer b.mu.Unlock()
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if !b.running {
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b.running = true
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}
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streamID := fmt.Sprintf("r2w_%s_%s_%d", b.roomID, userID, time.Now().UnixNano())
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// 创建本地轨道
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videoTrack, err := webrtc.NewTrackLocalStaticRTP(
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webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeH264},
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fmt.Sprintf("video_%s", streamID),
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fmt.Sprintf("stream_%s", userID),
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)
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if err != nil {
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return nil, fmt.Errorf("create video track failed: %w", err)
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}
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audioTrack, err := webrtc.NewTrackLocalStaticRTP(
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webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeOpus},
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fmt.Sprintf("audio_%s", streamID),
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fmt.Sprintf("stream_%s", userID),
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)
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if err != nil {
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return nil, fmt.Errorf("create audio track failed: %w", err)
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}
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converter := &RTMPToWebRTCConverter{
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UserID: userID,
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StreamID: streamID,
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RoomID: b.roomID,
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Running: true,
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stopChan: make(chan struct{}),
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InputURL: inputURL,
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VideoTrack: videoTrack,
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AudioTrack: audioTrack,
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videoCodec: "h264",
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audioCodec: "aac",
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}
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b.rtmpToWebRTC[userID] = converter
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// 启动转换协程
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go converter.Run()
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log.Printf("🌉 [Bridge] 添加 RTMP->WebRTC | Room:%s User:%s Stream:%s", b.roomID, userID, streamID)
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return converter, nil
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}
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// RemoveWebRTCToRTMP 移除 WebRTC 到 RTMP 的转换
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func (b *Bridge) RemoveWebRTCToRTMP(userID string) {
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b.mu.Lock()
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defer b.mu.Unlock()
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if converter, exists := b.webrtcToRTMP[userID]; exists {
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converter.Stop()
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delete(b.webrtcToRTMP, userID)
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log.Printf("🌉 [Bridge] 移除 WebRTC->RTMP | Room:%s User:%s", b.roomID, userID)
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}
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}
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// RemoveRTMPToWebRTC 移除 RTMP 到 WebRTC 的转换
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func (b *Bridge) RemoveRTMPToWebRTC(userID string) {
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b.mu.Lock()
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defer b.mu.Unlock()
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if converter, exists := b.rtmpToWebRTC[userID]; exists {
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converter.Stop()
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delete(b.rtmpToWebRTC, userID)
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log.Printf("🌉 [Bridge] 移除 RTMP->WebRTC | Room:%s User:%s", b.roomID, userID)
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}
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}
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// GetRTMPToWebRTCTracks 获取 RTMP->WebRTC 转换器的输出轨道
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func (b *Bridge) GetRTMPToWebRTCTracks(userID string) (*webrtc.TrackLocalStaticRTP, *webrtc.TrackLocalStaticRTP) {
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b.mu.RLock()
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defer b.mu.RUnlock()
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if converter, exists := b.rtmpToWebRTC[userID]; exists {
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return converter.VideoTrack, converter.AudioTrack
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}
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return nil, nil
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}
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// --- WebRTCToRTMPConverter 方法 ---
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// Run 执行 WebRTC 到 RTMP 的转换
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func (c *WebRTCToRTMPConverter) Run() {
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log.Printf("🌉 [W2R] 转换开始 | Stream:%s", c.StreamID)
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defer log.Printf("🌉 [W2R] 转换结束 | Stream:%s", c.StreamID)
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// 启动视频读取
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if c.VideoTrack != nil {
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go c.readVideoTrack()
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}
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// 启动音频读取
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if c.AudioTrack != nil {
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go c.readAudioTrack()
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}
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// 启动 RTMP 推流
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go c.pushToRTMP()
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// 等待停止信号
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<-c.stopChan
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}
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// readVideoTrack 读取视频轨道
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func (c *WebRTCToRTMPConverter) readVideoTrack() {
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buf := make([]byte, 1500)
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for c.Running {
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n, _, err := c.VideoTrack.Read(buf)
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if err != nil {
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if c.Running {
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log.Printf("⚠️ [W2R] 读取视频失败: %v", err)
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}
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return
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}
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// 解析 RTP 包
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packet := &rtp.Packet{}
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if err := packet.Unmarshal(buf[:n]); err != nil {
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continue
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}
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select {
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case c.videoBuffer <- packet:
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default:
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// 缓冲区满，丢弃
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}
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}
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}
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// readAudioTrack 读取音频轨道
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func (c *WebRTCToRTMPConverter) readAudioTrack() {
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buf := make([]byte, 1500)
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for c.Running {
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n, _, err := c.AudioTrack.Read(buf)
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if err != nil {
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if c.Running {
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log.Printf("⚠️ [W2R] 读取音频失败: %v", err)
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}
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return
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}
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// 解析 RTP 包
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packet := &rtp.Packet{}
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if err := packet.Unmarshal(buf[:n]); err != nil {
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continue
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}
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select {
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case c.audioBuffer <- packet:
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default:
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// 缓冲区满，丢弃
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}
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}
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}
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// pushToRTMP 推送到 RTMP
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func (c *WebRTCToRTMPConverter) pushToRTMP() {
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// TODO: 实际的 RTMP 推流实现
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// 1. 连接到 RTMP 服务器
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// 2. 从 videoBuffer/audioBuffer 读取 RTP 包
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// 3. 解封装 RTP，获取裸数据
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// 4. 视频: H.264 NAL 单元 -> FLV Tag
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// 5. 音频: Opus -> AAC (需要转码)
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// 6. 发送 FLV Tag 到 RTMP 服务器
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for c.Running {
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select {
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case <-c.stopChan:
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return
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case videoPacket := <-c.videoBuffer:
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// 处理视频包
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_ = videoPacket
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case audioPacket := <-c.audioBuffer:
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// 处理音频包
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_ = audioPacket
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}
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}
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}
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// Stop 停止转换
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func (c *WebRTCToRTMPConverter) Stop() {
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if !c.Running {
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return
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}
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c.Running = false
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close(c.stopChan)
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}
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// --- RTMPToWebRTCConverter 方法 ---
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// Run 执行 RTMP 到 WebRTC 的转换
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func (c *RTMPToWebRTCConverter) Run() {
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log.Printf("🌉 [R2W] 转换开始 | Stream:%s URL:%s", c.StreamID, c.InputURL)
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defer log.Printf("🌉 [R2W] 转换结束 | Stream:%s", c.StreamID)
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// TODO: 实际的 RTMP 拉流实现
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// 1. 连接到 RTMP 服务器
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// 2. 拉取 FLV 流
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// 3. 解析 FLV Tag
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// 4. 视频: H.264 NAL 单元 -> RTP 包
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// 5. 音频: AAC -> Opus (需要转码)
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// 6. 写入 WebRTC Track
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// 等待停止信号
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<-c.stopChan
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}
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// Stop 停止转换
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func (c *RTMPToWebRTCConverter) Stop() {
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if !c.Running {
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return
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}
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c.Running = false
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close(c.stopChan)
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}
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// GetConverterCount 获取转换器数量
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func (b *Bridge) GetConverterCount() (int, int) {
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b.mu.RLock()
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defer b.mu.RUnlock()
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return len(b.webrtcToRTMP), len(b.rtmpToWebRTC)
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}
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// IsRunning 是否运行中
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func (b *Bridge) IsRunning() bool {
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b.mu.RLock()
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defer b.mu.RUnlock()
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return b.running
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}
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// GetAllConverters 获取所有转换器信息
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func (b *Bridge) GetAllConverters() ([]map[string]interface{}, []map[string]interface{}) {
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b.mu.RLock()
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defer b.mu.RUnlock()
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w2r := make([]map[string]interface{}, 0, len(b.webrtcToRTMP))
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for _, c := range b.webrtcToRTMP {
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w2r = append(w2r, map[string]interface{}{
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"user_id": c.UserID,
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"stream_id": c.StreamID,
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"output_url": c.OutputURL,
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"running": c.Running,
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})
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}
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r2w := make([]map[string]interface{}, 0, len(b.rtmpToWebRTC))
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for _, c := range b.rtmpToWebRTC {
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r2w = append(r2w, map[string]interface{}{
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"user_id": c.UserID,
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"stream_id": c.StreamID,
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"input_url": c.InputURL,
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"running": c.Running,
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})
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}
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return w2r, r2w
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}
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