feat: sync group call livestream support
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A
parent
56d995474c
commit
f1a27996d3
37 changed files with 2219 additions and 83 deletions
192
internal/rpc/phone_group_call_rtmp.go
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192
internal/rpc/phone_group_call_rtmp.go
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package rpc
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import (
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"context"
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"crypto/rand"
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"encoding/binary"
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"encoding/json"
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"github.com/gotd/td/tg"
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"telesrv/internal/domain"
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)
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// RTMP 直播（Live Stream）RPC：createGroupCall(rtmp_stream) 建房后，推流方（OBS）
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// 用 getGroupCallStreamRtmpUrl 拿到的 url/key 推流，观众 join 后经
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// upload.getFile(inputGroupCallStream) 拉 broadcast chunk。
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//
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// 与普通语音聊天（RTC/SFU）关键差异：RTMP join 不建 SFU 连接、不做 ssrc 唯一性
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// 媒体面校验；updateGroupCallConnection.params 返回 {"stream":true,"rtmp":true}
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// 让 TDesktop 切 broadcast 模式（ParseJoinResponse → JoinBroadcastStream）。
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// rtmpConnectionParams 是 RTMP 观众 join 响应的下行 JSON（tgcalls broadcast 分支）。
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// 可管理者附 rtmp_stream_url/key 供直播设置页展示（普通观众不下发，避免泄漏推流凭据）。
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type rtmpConnectionParams struct {
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Stream bool `json:"stream"`
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Rtmp bool `json:"rtmp"`
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RtmpStreamURL string `json:"rtmp_stream_url,omitempty"`
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RtmpStreamKey string `json:"rtmp_stream_key,omitempty"`
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}
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func buildRtmpConnectionParams(url, key string) (string, error) {
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p := rtmpConnectionParams{Stream: true, Rtmp: true, RtmpStreamURL: url, RtmpStreamKey: key}
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out, err := json.Marshal(p)
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if err != nil {
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return "", err
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}
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return string(out), nil
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}
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// joinRtmpGroupCall 处理 RTMP 直播房间的 joinGroupCall。
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func (r *Router) joinRtmpGroupCall(ctx context.Context, scope *groupCallScope, req *tg.PhoneJoinGroupCallRequest) (tg.UpdatesClass, error) {
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// 房间上限：RTMP 观众也计入 participants_count；rejoin（已在会换 ssrc）不受限。
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if max := r.cfg.GroupCallMaxParticipants; max > 0 && scope.call.ParticipantsCount >= max {
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if p, found, _ := r.deps.GroupCalls.Participant(ctx, scope.call.ID, scope.userID); !found || p.Left {
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return nil, groupCallForbiddenErr()
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}
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}
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joinAsChannelID, err := r.groupCallJoinAsChannelID(scope, req.JoinAs)
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if err != nil {
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return nil, err
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}
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now := int(r.clock.Now().Unix())
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// RTMP 观众的 join JSON 仍带 tgcalls ssrc（客户端为拉流也建了本地 controller）；
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// 解析失败/缺失不致命——直播不需要 SFU，用随机 ssrc 记账保证参与者行有效。
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ssrc := int64(0)
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if _, s, err := parseGroupCallJoinPayload(req.Params.Data); err == nil {
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ssrc = s
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}
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if ssrc == 0 {
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ssrc = randomSSRC()
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}
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// RTMP 观众（含创建者，其经 OBS 独立推流，在 group call 里同样是纯观众）一律
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// force-muted：IsAdmin=false 让 store 对 join_muted 房间置 muted+muted_by_admin，
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// self 行输出 muted=true / can_self_unmute=false，TDesktop 才会稳定停在 stream 模式。
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mut, err := r.deps.GroupCalls.Join(ctx, domain.JoinGroupCallRequest{
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CallID: scope.call.ID,
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UserID: scope.userID,
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JoinAsChannelID: joinAsChannelID,
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SSRC: ssrc,
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Muted: true,
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IsAdmin: false,
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Now: now,
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})
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if err != nil {
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return nil, groupCallErr(err)
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}
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// 可管理者拿推流 url/key（直播设置页展示）；普通观众只得 stream:true。
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var url, key string
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if scope.canManage() && r.deps.GroupCalls != nil {
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if k, kerr := r.deps.GroupCalls.RtmpStreamKey(ctx, scope.channel.ID, false, now); kerr == nil {
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key = k
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url = r.rtmpIngestURL()
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}
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}
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params, err := buildRtmpConnectionParams(url, key)
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if err != nil {
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return nil, internalErr()
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}
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channel := r.groupCallMutationFanout(ctx, scope.channel, mut)
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out := r.groupCallUpdateContainer(ctx, scope.userID, channel,
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&tg.UpdateGroupCallParticipants{
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Call: &tg.InputGroupCall{ID: mut.Call.ID, AccessHash: mut.Call.AccessHash},
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Participants: tgGroupCallParticipants([]domain.GroupCallParticipant{mut.Participant}, scope.userID),
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Version: mut.Call.Version,
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}, []int64{scope.userID})
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// updateGroupCall 必须先于 updateGroupCallConnection：TDesktop 按序 applyUpdates，
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// 处理 connection 时若还没从 groupCall 读到 stream_dc_id 会打
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// "Api Error: Empty stream_dc_id" 并 fallback 主 DC。
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callUpdate := &tg.UpdateGroupCall{Call: tgGroupCall(mut.Call, scope.userID, scope.canManage())}
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if channel.ID != 0 {
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callUpdate.SetPeer(&tg.PeerChannel{ChannelID: channel.ID})
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}
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out.Updates = append(out.Updates, callUpdate)
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out.Updates = append(out.Updates, &tg.UpdateGroupCallConnection{Params: tg.DataJSON{Data: params}})
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return out, nil
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}
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// onPhoneGetGroupCallStreamRtmpURL 返回频道的 RTMP 推流 url/key（创建前预览、
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// 直播设置页展示、revoke 轮换）。仅频道管理员可调；revoke=true 生成新 key，旧 key
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// 立即失效并断开正在进行的推流。
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func (r *Router) onPhoneGetGroupCallStreamRtmpURL(ctx context.Context, req *tg.PhoneGetGroupCallStreamRtmpURLRequest) (*tg.PhoneGroupCallStreamRtmpURL, error) {
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if req == nil {
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return nil, inputRequestInvalidErr()
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}
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if r.deps.GroupCalls == nil || r.deps.Channels == nil {
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return nil, notImplementedErr()
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}
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userID, err := r.phoneRequireUser(ctx)
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if err != nil {
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return nil, err
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}
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peer, err := r.checkedDomainPeerFromInputPeer(ctx, userID, req.Peer)
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if err != nil {
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return nil, err
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}
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if peer.Type != domain.PeerTypeChannel || peer.ID == 0 {
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return nil, peerIDInvalidErr()
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}
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view, err := r.deps.Channels.GetChannel(ctx, userID, peer.ID)
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if err != nil {
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return nil, peerIDInvalidErr()
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}
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if view.Self.Status != domain.ChannelMemberActive || !channelMemberIsAdmin(view.Self) {
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return nil, tgerr400("CHAT_ADMIN_REQUIRED")
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}
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now := int(r.clock.Now().Unix())
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key, err := r.deps.GroupCalls.RtmpStreamKey(ctx, peer.ID, req.Revoke, now)
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if err != nil {
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return nil, groupCallErr(err)
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}
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if req.Revoke && r.deps.LiveStreams != nil {
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// 旧 key 失效后仍在推的连接必须断开（否则用旧 key 的推流会继续被接收）。
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r.deps.LiveStreams.DropChannel(peer.ID)
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}
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return &tg.PhoneGroupCallStreamRtmpURL{URL: r.rtmpIngestURL(), Key: key}, nil
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}
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// onPhoneGetGroupCallStreamChannels 返回 RTMP 直播的当前时间轴（unified：channel=1
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// scale=0），供 tgcalls 决定从哪个 time_ms 起拉 chunk。无活跃推流时返回空列表，
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// TDesktop 据此显示"等待推流"占位并循环重试。
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func (r *Router) onPhoneGetGroupCallStreamChannels(ctx context.Context, call tg.InputGroupCallClass) (*tg.PhoneGroupCallStreamChannels, error) {
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scope, err := r.groupCallScopeFrom(ctx, call)
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if err != nil {
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return nil, err
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}
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out := &tg.PhoneGroupCallStreamChannels{Channels: []tg.GroupCallStreamChannel{}}
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if !scope.call.RtmpStream || r.deps.LiveStreams == nil || scope.channel.ID == 0 {
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return out, nil
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}
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for _, ch := range r.deps.LiveStreams.StreamChannels(scope.channel.ID) {
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out.Channels = append(out.Channels, tg.GroupCallStreamChannel{
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Channel: ch.Channel,
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Scale: ch.Scale,
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LastTimestampMs: ch.LastTimestampMs,
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})
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}
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return out, nil
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}
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// rtmpIngestURL 返回展示给推流端的 RTMP 服务器地址。
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func (r *Router) rtmpIngestURL() string {
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if r.cfg.RtmpIngestURL != "" {
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return r.cfg.RtmpIngestURL
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}
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host := r.cfg.IP
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if host == "" {
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host = "127.0.0.1"
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}
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return "rtmp://" + host + ":2400/live"
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}
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func randomSSRC() int64 {
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var buf [4]byte
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if _, err := rand.Read(buf[:]); err != nil {
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return 1
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}
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v := int64(binary.BigEndian.Uint32(buf[:]))
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if v == 0 {
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v = 1
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}
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return v
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}
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