782 lines
24 KiB
TypeScript
782 lines
24 KiB
TypeScript
/**
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* 录制服务：基于 mediasoup PlainTransport + ffmpeg 子进程
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*
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* 录制管线：
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* ┌─────────────┐ RTP/RTCP ┌──────────────┐ SDP+stdin ┌────────┐ fs ┌─────────┐
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* │ Producer X │ ───────────> │ PlainTransport│ ───────────────> │ ffmpeg │ ──────> │ *.mp4 │
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* │ (audio/video)│ │ + Consumer │ (UDP loopback) │ (子进程)│ └─────────┘
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* └─────────────┘ └──────────────┘ └────────┘
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*
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* 启动顺序（关键）：
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* 1) 为每个 Producer 创建 Consumer 拿到 negotiated rtpParameters
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* （Consumer 立即 pause，避免 RTP 在 ffmpeg 起来前空打）
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* 2) 给每个 Consumer 分配一对 UDP 端口（ffmpeg 监听端）
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* 3) 写 SDP 文件描述所有流
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* 4) spawn ffmpeg 进程，listen 在那些端口
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* 5) 创建对应 PlainTransport（rtcpMux=false, comedia=false）
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* 并 connect 到 ffmpeg 监听端口
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* 6) resume Consumer，RTP 开始流向 ffmpeg
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*
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* 停止顺序：
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* 1) SIGINT ffmpeg（让 ffmpeg 关闭 mp4 moov box；SIGKILL 会导致文件损坏）
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* 2) 等 ffmpeg 进程退出（超时 5s 兜底 SIGKILL）
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* 3) close Consumer / PlainTransport
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* 4) 释放端口
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* 5) stat 输出文件，返回大小 + 时长
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*
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* 关键约束：
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* - PlainTransport 用 rtcpMux=false：避免 RTCP 与 RTP 同端口，ffmpeg 解析更可靠
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* - comedia=false：由 server 主动 connect，IP/Port 是已知的本地回环
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* - 不开启 SRTP（loopback 不需要加密，降低 CPU 与配置复杂度）
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* - Consumer 必须设置 paused=true 创建，否则 RTP 会先于 ffmpeg 启动到达
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*/
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import { spawn, type ChildProcess } from 'node:child_process';
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import { randomUUID } from 'node:crypto';
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import { promises as fs } from 'node:fs';
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import { tmpdir } from 'node:os';
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import { join } from 'node:path';
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import type {
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Consumer,
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PlainTransport,
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RtpCodecParameters,
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RtpParameters,
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} from 'mediasoup/types';
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import { config } from '../config.js';
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import { AppError, notFound } from '../utils/errors.js';
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import { childLogger } from '../utils/logger.js';
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import { allocatePortPair, releasePortPair } from '../utils/port-pool.js';
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import { assertTestOnly } from '../utils/test-guard.js';
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import { getProducer } from './producer.service.js';
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import { getRouter } from './router.service.js';
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const log = childLogger({ module: 'services.recording' });
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// ============ 类型 ============
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interface RecordingTrack {
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/** 来源 producer ID（供日志/排障） */
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producerId: string;
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/** Consumer ID（停止时关闭） */
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consumerId: string;
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/** PlainTransport ID（停止时关闭） */
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transportId: string;
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/** ffmpeg 监听端口（RTP） */
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ffmpegRtpPort: number;
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/** ffmpeg 监听端口（RTCP） */
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ffmpegRtcpPort: number;
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kind: 'audio' | 'video';
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/** payload type，用于 SDP / 跨进程串联 */
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payloadType: number;
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/** Codec mime，例如 audio/opus / video/VP8 */
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codecMime: string;
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/** Codec clock rate */
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clockRate: number;
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/** Codec channels（音频专用） */
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channels?: number;
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/** Codec sdpFmtp 字段（如 minptime=10;useinbandfec=1） */
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sdpFmtp?: string;
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}
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interface RecordingEntry {
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id: string;
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roomCode: string;
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routerId: string;
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startedAt: number;
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/** 各路 producer 对应的 track */
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tracks: RecordingTrack[];
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/** mediasoup Consumer 句柄（停止时 close） */
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consumers: Consumer[];
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/** mediasoup PlainTransport 句柄（停止时 close） */
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transports: PlainTransport[];
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/** ffmpeg 子进程 */
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ffmpeg: ChildProcess;
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/** SDP 临时文件 */
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sdpFilePath: string;
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/** 输出 mp4 文件 */
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outputPath: string;
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/** ffmpeg 退出 promise，stop 时 await */
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ffmpegExited: Promise<void>;
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/** 停止时回填，避免重复 stop */
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stopping: boolean;
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/** 退出码，stop 后填充供 callback 返回 */
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exitCode: number | null;
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/** 失败原因（spawn 阶段或 stop 超时），供 finalize 上报 */
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failureReason: string | null;
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/** ffmpeg 是否在 stop 之前异常退出（watchdog 标记，避免重复触发回调） */
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unexpectedExit: boolean;
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}
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// ==================== Watchdog / 失败回调 ====================
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/**
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* ffmpeg 异常退出回调签名
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*
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* 触发条件：ffmpeg 在 stop 调用之前自行退出（非零退出码或 spawn 失败）
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* 调用方可注册 webhook、上报告警、写入 DB 标记失败等
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*
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* 注意：回调在 ffmpeg 'exit' 事件后异步触发（setImmediate），不会阻塞退出处理；
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* 实现方需保证内部错误不抛出到 Node 主循环，否则会被全局 unhandledRejection 接住。
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*/
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export type RecordingFailureCallback = (info: {
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recordingId: string;
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roomCode: string;
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routerId: string;
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exitCode: number | null;
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failureReason: string;
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}) => void | Promise<void>;
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let failureCallback: RecordingFailureCallback | null = null;
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/**
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* 注册录制失败回调（覆盖式：只保留最近一次注册）
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* 传 null 取消注册
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*/
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export function setRecordingFailureCallback(cb: RecordingFailureCallback | null): void {
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failureCallback = cb;
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}
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/**
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* 默认 webhook 实现：当 process.env.RECORDING_FAILURE_WEBHOOK_URL 设置时，
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* 通过 fetch POST JSON 上报失败。Node 18+ 自带 fetch
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*/
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async function defaultWebhookCallback(info: Parameters<RecordingFailureCallback>[0]): Promise<void> {
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const url = process.env.RECORDING_FAILURE_WEBHOOK_URL;
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if (!url) return;
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try {
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const res = await fetch(url, {
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method: 'POST',
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headers: { 'Content-Type': 'application/json' },
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body: JSON.stringify(info),
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});
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if (!res.ok) {
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log.warn(
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{ recordingId: info.recordingId, status: res.status },
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'recording failure webhook returned non-2xx',
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);
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}
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} catch (err) {
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log.warn(
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{ recordingId: info.recordingId, err: err instanceof Error ? err.message : String(err) },
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'recording failure webhook POST failed',
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);
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}
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}
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/**
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* 内部触发失败回调：异步、错误吞掉
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* 同时尝试用户注册的回调和默认 webhook（两者独立）
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*/
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function fireFailureCallback(entry: RecordingEntry): void {
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if (entry.unexpectedExit) return; // 幂等：每条录制最多触发一次
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entry.unexpectedExit = true;
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const info = {
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recordingId: entry.id,
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roomCode: entry.roomCode,
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routerId: entry.routerId,
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exitCode: entry.exitCode,
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failureReason: entry.failureReason ?? 'ffmpeg exited unexpectedly',
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};
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setImmediate(() => {
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void defaultWebhookCallback(info);
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if (failureCallback) {
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try {
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const r = failureCallback(info);
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if (r && typeof (r as Promise<void>).catch === 'function') {
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(r as Promise<void>).catch((err) => {
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log.warn(
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{ recordingId: info.recordingId, err: err instanceof Error ? err.message : String(err) },
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'recording failure callback rejected',
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);
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});
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}
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} catch (err) {
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log.warn(
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{ recordingId: info.recordingId, err: err instanceof Error ? err.message : String(err) },
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'recording failure callback threw',
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);
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}
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}
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});
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}
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const recordingMap = new Map<string, RecordingEntry>();
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// 默认输出目录；由 RECORDING_OUTPUT_DIR 环境变量覆盖，便于 docker volume 挂载
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const OUTPUT_DIR = process.env.RECORDING_OUTPUT_DIR
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? process.env.RECORDING_OUTPUT_DIR
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: join(tmpdir(), 'echochat-recordings');
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const STOP_GRACE_MS = 5000;
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// ============ 工具函数 ============
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async function ensureOutputDir(): Promise<void> {
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await fs.mkdir(OUTPUT_DIR, { recursive: true });
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}
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function recordingId(): string {
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// crypto.randomUUID 在 node 20+ 可用；移除连字符以让文件名更紧凑、URL-safe
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return randomUUID().replace(/-/g, '');
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}
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/**
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* 从 Consumer 的 rtpParameters 提取录制需要的关键字段
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* - 取第一个 codec：mediasoup consumer 默认每个 Consumer 只协商一个主 codec
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* - sdpFmtp：把 codec.parameters 拼成 SDP a=fmtp 行格式
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*/
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function pickCodec(rtpParameters: RtpParameters): {
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payloadType: number;
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codecMime: string;
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clockRate: number;
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channels?: number;
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sdpFmtp?: string;
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} {
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const codec: RtpCodecParameters | undefined = rtpParameters.codecs[0];
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if (!codec) {
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throw new AppError('MEDIASOUP_ERROR', 'consumer has no codecs in rtpParameters');
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}
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const params = codec.parameters || {};
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const fmtpParts = Object.entries(params)
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.filter(([, v]) => v !== undefined && v !== null && v !== '')
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.map(([k, v]) => `${k}=${typeof v === 'boolean' ? (v ? 1 : 0) : String(v)}`);
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return {
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payloadType: codec.payloadType,
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codecMime: codec.mimeType,
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clockRate: codec.clockRate,
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channels: codec.channels,
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sdpFmtp: fmtpParts.length > 0 ? fmtpParts.join(';') : undefined,
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};
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}
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/**
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* 把 mime（audio/opus / video/VP8）转成 SDP rtpmap 编码名
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* SDP a=rtpmap 不接受 `video/` 前缀，要单独取编码名
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*/
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function mimeToCodecName(mime: string): string {
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const idx = mime.indexOf('/');
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return idx >= 0 ? mime.slice(idx + 1) : mime;
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}
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/**
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* 生成 SDP 内容
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*
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* 注意 SDP 是从 ffmpeg 视角写的：c= 行的地址、m= 行的端口都是 ffmpeg 监听处。
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* mediasoup PlainTransport 之后 connect({ ip, port: ffmpegPort, rtcpPort }) 把流送过来。
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*
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* 一份 SDP 描述了一个会话里 1~N 条流（每条对应一个 producer 的 consumer）
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*/
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function buildSdp(tracks: RecordingTrack[], localIp: string): string {
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const lines: string[] = [
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'v=0',
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`o=- 0 0 IN IP4 ${localIp}`,
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's=echochat-recording',
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`c=IN IP4 ${localIp}`,
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't=0 0',
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];
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for (const t of tracks) {
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const codecName = mimeToCodecName(t.codecMime);
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const rtpmapValue =
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t.kind === 'audio'
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? `${codecName}/${t.clockRate}/${t.channels ?? 2}`
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: `${codecName}/${t.clockRate}`;
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lines.push(`m=${t.kind} ${t.ffmpegRtpPort} RTP/AVP ${t.payloadType}`);
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lines.push(`a=rtcp:${t.ffmpegRtcpPort}`);
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lines.push(`a=rtpmap:${t.payloadType} ${rtpmapValue}`);
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if (t.sdpFmtp) {
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lines.push(`a=fmtp:${t.payloadType} ${t.sdpFmtp}`);
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}
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lines.push('a=recvonly');
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}
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// SDP 必须以 \r\n 分隔，ffmpeg 解析行为更稳定
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return lines.join('\r\n') + '\r\n';
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}
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/**
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* 构造 ffmpeg 命令行参数
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*
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* -protocol_whitelist file,udp,rtp：默认只允许 file，需要显式开 udp/rtp
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* -fflags +genpts：从 RTP 时间戳生成 PTS
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* -use_wallclock_as_timestamps 1：避免 RTP 时间戳跳变导致 mp4 时长异常
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* -c copy：不重编码，直接复用 RTP 载荷
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* - 优点：CPU 占用极低
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* - 缺点：浏览器侧 VP8 进 mp4 容器在某些播放器不支持；后续可改 libx264
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* -y：覆盖已存在文件（防止 recordingId 碰撞）
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*/
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function buildFfmpegArgs(sdpFilePath: string, outputPath: string): string[] {
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return [
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'-loglevel',
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'info',
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'-protocol_whitelist',
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'file,udp,rtp',
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'-fflags',
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'+genpts',
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'-use_wallclock_as_timestamps',
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'1',
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'-i',
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sdpFilePath,
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'-map',
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'0:a?',
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'-map',
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'0:v?',
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'-c',
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'copy',
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'-y',
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outputPath,
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];
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}
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// ============ 对外 API ============
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export interface StartRecordingInput {
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routerId: string;
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roomCode: string;
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/**
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* 要录制的 producer ID 列表。
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* 通常是 host 的 audio + video，或者屏幕共享 video。
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* Phase B1 阶段只录这些指定的 producer，不混合所有成员；后续 Phase B3 再扩展。
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*/
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producerIds: string[];
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}
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export interface StartRecordingOutput {
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recordingId: string;
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outputPath: string;
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tracks: Array<{
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producerId: string;
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kind: 'audio' | 'video';
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}>;
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}
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/**
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* 启动录制
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*
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* 失败处理：
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* - 任一步骤失败都会回滚已创建的资源（端口 / Transport / Consumer / ffmpeg 进程）
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* - 不在 recordingMap 中留下"半态"条目，调用方可直接重试
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*/
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export async function startRecording(input: StartRecordingInput): Promise<StartRecordingOutput> {
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const router = getRouter(input.routerId);
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if (input.producerIds.length === 0) {
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throw new AppError('CONFLICT', 'producerIds is empty');
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}
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await ensureOutputDir();
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const id = recordingId();
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const startedAt = Date.now();
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const localIp = '127.0.0.1';
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const outputPath = join(OUTPUT_DIR, `${id}.mp4`);
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const sdpFilePath = join(OUTPUT_DIR, `${id}.sdp`);
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// 已分配资源，用于失败时回滚
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const allocatedPortPairs: Array<{ rtp: number; rtcp: number }> = [];
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const createdTransports: PlainTransport[] = [];
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const createdConsumers: Consumer[] = [];
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let ffmpegProc: ChildProcess | null = null;
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const rollback = async () => {
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for (const c of createdConsumers) {
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try {
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c.close();
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} catch {
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/* ignored */
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}
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}
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for (const t of createdTransports) {
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try {
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t.close();
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} catch {
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/* ignored */
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}
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}
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for (const p of allocatedPortPairs) {
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releasePortPair(p.rtp, p.rtcp);
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}
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if (ffmpegProc && ffmpegProc.exitCode === null) {
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try {
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ffmpegProc.kill('SIGKILL');
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} catch {
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/* ignored */
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}
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}
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try {
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await fs.unlink(sdpFilePath);
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} catch {
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/* ignored */
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}
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try {
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await fs.unlink(outputPath);
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} catch {
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/* ignored */
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}
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};
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try {
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// ---------- Step 1: 预校验所有 producer 存在且可消费 ----------
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// 不在这里建 Consumer：Consumer 必须依附 PlainTransport，留到 Step 2 一起做
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// 但 canConsume 失败要尽早抛错，避免分配资源后再失败导致回滚开销
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const tracks: RecordingTrack[] = [];
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for (const producerId of input.producerIds) {
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// getProducer 不存在会抛 notFound，等价于校验 producerId 合法
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getProducer(producerId);
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if (!router.canConsume({ producerId, rtpCapabilities: router.rtpCapabilities })) {
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throw new AppError(
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'CONFLICT',
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`router cannot consume producer ${producerId} with its own rtpCapabilities`,
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);
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}
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}
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// ---------- Step 2: 为每个 producer 分配端口对 + 创建 PlainTransport ----------
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for (const producerId of input.producerIds) {
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const producer = getProducer(producerId);
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const portPair = await allocatePortPair();
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allocatedPortPairs.push(portPair);
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const transport = await router.createPlainTransport({
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listenIp: { ip: config.mediasoup.listenIp, announcedIp: undefined },
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rtcpMux: false,
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comedia: false,
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enableSrtp: false,
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appData: {
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purpose: 'recording',
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recordingId: id,
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producerId,
|
||
},
|
||
});
|
||
createdTransports.push(transport);
|
||
|
||
// 创建 consumer：必须传 router 自己的 rtpCapabilities，consumer 会与 producer 协商
|
||
const consumer = await transport.consume({
|
||
producerId,
|
||
rtpCapabilities: router.rtpCapabilities,
|
||
paused: true,
|
||
});
|
||
createdConsumers.push(consumer);
|
||
|
||
const codec = pickCodec(consumer.rtpParameters);
|
||
tracks.push({
|
||
producerId,
|
||
consumerId: consumer.id,
|
||
transportId: transport.id,
|
||
ffmpegRtpPort: portPair.rtp,
|
||
ffmpegRtcpPort: portPair.rtcp,
|
||
kind: producer.kind as 'audio' | 'video',
|
||
...codec,
|
||
});
|
||
}
|
||
|
||
// ---------- Step 3: 写 SDP 文件 ----------
|
||
const sdp = buildSdp(tracks, localIp);
|
||
await fs.writeFile(sdpFilePath, sdp, 'utf-8');
|
||
log.info({ recordingId: id, sdp }, 'SDP file generated');
|
||
|
||
// ---------- Step 4: spawn ffmpeg ----------
|
||
const ffmpegBin = process.env.FFMPEG_BIN || 'ffmpeg';
|
||
const args = buildFfmpegArgs(sdpFilePath, outputPath);
|
||
ffmpegProc = spawn(ffmpegBin, args, { stdio: ['ignore', 'pipe', 'pipe'] });
|
||
|
||
const exited = new Promise<void>((resolve) => {
|
||
ffmpegProc!.once('exit', (code, signal) => {
|
||
const entry = recordingMap.get(id);
|
||
if (entry) {
|
||
entry.exitCode = code;
|
||
if (!entry.stopping && code !== 0) {
|
||
entry.failureReason = `ffmpeg exited unexpectedly code=${code} signal=${signal ?? ''}`;
|
||
log.error(
|
||
{ recordingId: id, code, signal },
|
||
'ffmpeg exited unexpectedly before stop',
|
||
);
|
||
// Phase B watchdog：异步触发失败回调（webhook + 用户回调），不阻塞 exit 处理
|
||
fireFailureCallback(entry);
|
||
}
|
||
}
|
||
resolve();
|
||
});
|
||
});
|
||
|
||
ffmpegProc.stderr?.on('data', (chunk: Buffer) => {
|
||
const text = chunk.toString('utf-8');
|
||
// ffmpeg 默认大部分日志走 stderr；只在 debug 级别落，避免日志爆炸
|
||
log.debug({ recordingId: id }, `[ffmpeg] ${text.trim()}`);
|
||
});
|
||
|
||
ffmpegProc.once('error', (err) => {
|
||
log.error({ recordingId: id, err: err.message }, 'ffmpeg spawn error');
|
||
});
|
||
|
||
// 等 ffmpeg 进入监听就绪：实践经验 200ms 足够 ffmpeg 解析 SDP + 绑定端口
|
||
// 这里加一个 250ms 的小延迟，避免过早 connect 导致前几个 RTP 包丢失
|
||
await new Promise((r) => setTimeout(r, 250));
|
||
|
||
if (ffmpegProc.exitCode !== null) {
|
||
throw new AppError(
|
||
'MEDIASOUP_ERROR',
|
||
`ffmpeg exited prematurely with code ${ffmpegProc.exitCode}`,
|
||
);
|
||
}
|
||
|
||
// ---------- Step 5: PlainTransport.connect 到 ffmpeg ----------
|
||
for (let i = 0; i < input.producerIds.length; i++) {
|
||
const transport = createdTransports[i];
|
||
const t = tracks[i];
|
||
await transport.connect({
|
||
ip: localIp,
|
||
port: t.ffmpegRtpPort,
|
||
rtcpPort: t.ffmpegRtcpPort,
|
||
});
|
||
}
|
||
|
||
// ---------- Step 6: resume consumers，RTP 开始流向 ffmpeg ----------
|
||
for (const c of createdConsumers) {
|
||
await c.resume();
|
||
}
|
||
|
||
const entry: RecordingEntry = {
|
||
id,
|
||
roomCode: input.roomCode,
|
||
routerId: input.routerId,
|
||
startedAt,
|
||
tracks,
|
||
consumers: createdConsumers,
|
||
transports: createdTransports,
|
||
ffmpeg: ffmpegProc,
|
||
sdpFilePath,
|
||
outputPath,
|
||
ffmpegExited: exited,
|
||
stopping: false,
|
||
exitCode: null,
|
||
failureReason: null,
|
||
unexpectedExit: false,
|
||
};
|
||
recordingMap.set(id, entry);
|
||
|
||
log.info(
|
||
{
|
||
recordingId: id,
|
||
roomCode: input.roomCode,
|
||
producerCount: input.producerIds.length,
|
||
outputPath,
|
||
},
|
||
'recording started',
|
||
);
|
||
|
||
return {
|
||
recordingId: id,
|
||
outputPath,
|
||
tracks: tracks.map((t) => ({ producerId: t.producerId, kind: t.kind })),
|
||
};
|
||
} catch (err) {
|
||
log.error(
|
||
{ recordingId: id, err: err instanceof Error ? err.message : String(err) },
|
||
'startRecording failed, rolling back',
|
||
);
|
||
await rollback();
|
||
if (err instanceof AppError) throw err;
|
||
throw new AppError(
|
||
'MEDIASOUP_ERROR',
|
||
`startRecording failed: ${err instanceof Error ? err.message : String(err)}`,
|
||
);
|
||
}
|
||
}
|
||
|
||
export interface StopRecordingOutput {
|
||
recordingId: string;
|
||
outputPath: string;
|
||
sizeBytes: number;
|
||
durationMs: number;
|
||
exitCode: number | null;
|
||
failureReason: string | null;
|
||
}
|
||
|
||
/**
|
||
* 停止录制
|
||
*
|
||
* 幂等：重复调用同一 recordingId 仅第一次执行清理，后续返回 notFound
|
||
*
|
||
* 时间线：
|
||
* - t=0: SIGINT ffmpeg
|
||
* - t≤5s: ffmpeg 优雅退出，写完 moov box
|
||
* - t>5s: 强制 SIGKILL，文件可能损坏（罕见，记录 failureReason）
|
||
* - 之后 close consumers/transports/释放端口
|
||
*/
|
||
export async function stopRecording(id: string): Promise<StopRecordingOutput> {
|
||
const entry = recordingMap.get(id);
|
||
if (!entry) {
|
||
throw notFound('recording', id);
|
||
}
|
||
if (entry.stopping) {
|
||
// 并发重复 stop：返回当前快照，避免重复 SIGINT
|
||
log.warn({ recordingId: id }, 'stopRecording called while already stopping');
|
||
}
|
||
entry.stopping = true;
|
||
|
||
// 1) SIGINT ffmpeg
|
||
let killed = false;
|
||
if (entry.ffmpeg.exitCode === null) {
|
||
try {
|
||
entry.ffmpeg.kill('SIGINT');
|
||
killed = true;
|
||
} catch (err) {
|
||
log.warn(
|
||
{ recordingId: id, err: err instanceof Error ? err.message : String(err) },
|
||
'failed to send SIGINT to ffmpeg',
|
||
);
|
||
}
|
||
}
|
||
|
||
// 2) 等 ffmpeg 退出 + 超时兜底
|
||
if (killed) {
|
||
const timeout = new Promise<void>((resolve) => {
|
||
setTimeout(() => {
|
||
if (entry.ffmpeg.exitCode === null) {
|
||
try {
|
||
entry.ffmpeg.kill('SIGKILL');
|
||
} catch {
|
||
/* ignored */
|
||
}
|
||
entry.failureReason = `ffmpeg did not exit within ${STOP_GRACE_MS}ms, forced SIGKILL`;
|
||
log.error({ recordingId: id }, entry.failureReason);
|
||
}
|
||
resolve();
|
||
}, STOP_GRACE_MS);
|
||
});
|
||
await Promise.race([entry.ffmpegExited, timeout]);
|
||
// race 后再 await exited 拿到最终退出码
|
||
await entry.ffmpegExited;
|
||
}
|
||
|
||
// 3) close consumers + transports
|
||
for (const c of entry.consumers) {
|
||
try {
|
||
c.close();
|
||
} catch {
|
||
/* ignored */
|
||
}
|
||
}
|
||
for (const t of entry.transports) {
|
||
try {
|
||
t.close();
|
||
} catch {
|
||
/* ignored */
|
||
}
|
||
}
|
||
|
||
// 4) 释放端口
|
||
for (const track of entry.tracks) {
|
||
releasePortPair(track.ffmpegRtpPort, track.ffmpegRtcpPort);
|
||
}
|
||
|
||
// 5) 删除 SDP 临时文件（保留 mp4）
|
||
try {
|
||
await fs.unlink(entry.sdpFilePath);
|
||
} catch {
|
||
/* ignored */
|
||
}
|
||
|
||
// 6) stat 输出文件
|
||
let sizeBytes = 0;
|
||
try {
|
||
const st = await fs.stat(entry.outputPath);
|
||
sizeBytes = st.size;
|
||
} catch (err) {
|
||
if (!entry.failureReason) {
|
||
entry.failureReason = `output file missing: ${err instanceof Error ? err.message : String(err)}`;
|
||
}
|
||
}
|
||
|
||
const durationMs = Date.now() - entry.startedAt;
|
||
recordingMap.delete(id);
|
||
|
||
log.info(
|
||
{
|
||
recordingId: id,
|
||
durationMs,
|
||
sizeBytes,
|
||
exitCode: entry.exitCode,
|
||
failureReason: entry.failureReason,
|
||
},
|
||
'recording stopped',
|
||
);
|
||
|
||
return {
|
||
recordingId: id,
|
||
outputPath: entry.outputPath,
|
||
sizeBytes,
|
||
durationMs,
|
||
exitCode: entry.exitCode,
|
||
failureReason: entry.failureReason,
|
||
};
|
||
}
|
||
|
||
export interface RecordingInfo {
|
||
recordingId: string;
|
||
roomCode: string;
|
||
routerId: string;
|
||
startedAt: number;
|
||
durationMs: number;
|
||
outputPath: string;
|
||
trackCount: number;
|
||
ffmpegAlive: boolean;
|
||
}
|
||
|
||
export function getRecording(id: string): RecordingInfo {
|
||
const entry = recordingMap.get(id);
|
||
if (!entry) {
|
||
throw notFound('recording', id);
|
||
}
|
||
return {
|
||
recordingId: entry.id,
|
||
roomCode: entry.roomCode,
|
||
routerId: entry.routerId,
|
||
startedAt: entry.startedAt,
|
||
durationMs: Date.now() - entry.startedAt,
|
||
outputPath: entry.outputPath,
|
||
trackCount: entry.tracks.length,
|
||
ffmpegAlive: entry.ffmpeg.exitCode === null,
|
||
};
|
||
}
|
||
|
||
export function getRecordingStats(): { total: number } {
|
||
return { total: recordingMap.size };
|
||
}
|
||
|
||
/**
|
||
* 列出指定 roomCode 下的所有活跃录制（一般 ≤1）
|
||
* 供 Go 后端 EndRoom / 资源清理时反查
|
||
*/
|
||
export function listRecordingsByRoom(roomCode: string): RecordingInfo[] {
|
||
const out: RecordingInfo[] = [];
|
||
for (const entry of recordingMap.values()) {
|
||
if (entry.roomCode === roomCode) {
|
||
out.push({
|
||
recordingId: entry.id,
|
||
roomCode: entry.roomCode,
|
||
routerId: entry.routerId,
|
||
startedAt: entry.startedAt,
|
||
durationMs: Date.now() - entry.startedAt,
|
||
outputPath: entry.outputPath,
|
||
trackCount: entry.tracks.length,
|
||
ffmpegAlive: entry.ffmpeg.exitCode === null,
|
||
});
|
||
}
|
||
}
|
||
return out;
|
||
}
|
||
|
||
/** 测试专用：复位 recordingMap（停止所有进行中的录制） */
|
||
export async function _clearRecordings(): Promise<void> {
|
||
assertTestOnly('_clearRecordings');
|
||
const ids = [...recordingMap.keys()];
|
||
for (const id of ids) {
|
||
try {
|
||
await stopRecording(id);
|
||
} catch {
|
||
/* ignored */
|
||
}
|
||
}
|
||
recordingMap.clear();
|
||
}
|