node-av-lgpl 0.0.0 → 7.1.0-beta.2

package/dist/api/webrtc-stream.js

@@ -0,0 +1,546 @@
+import { MediaStreamTrack, RTCIceCandidate, RTCPeerConnection, RTCRtpCodecParameters, RTCSessionDescription } from 'werift';
+import { AV_CODEC_ID_AV1, AV_CODEC_ID_H264, AV_CODEC_ID_HEVC, AV_CODEC_ID_OPUS, AV_CODEC_ID_PCM_ALAW, AV_CODEC_ID_PCM_MULAW, AV_CODEC_ID_VP8, AV_CODEC_ID_VP9, } from '../constants/constants.js';
+import { isMediaFrameSource, RTPStream } from './rtp-stream.js';
+/**
+ * Complete WebRTC session management with werift integration.
+ *
+ * Provides end-to-end WebRTC streaming with automatic SDP negotiation,
+ * ICE candidate handling, and peer connection management.
+ * Built on top of {@link RTPStream} for generic RTP streaming with WebRTC-specific
+ * protocol details handled automatically.
+ * Integrates with werift library for RTCPeerConnection and media track handling.
+ * Ideal for building complete WebRTC streaming applications with minimal code.
+ *
+ * @example
+ * ```typescript
+ * import { WebRTCStream } from 'node-av/api';
+ *
+ * // Create session from media source
+ * const session = await WebRTCStream.create('rtsp://camera.local/stream', {
+ *   mtu: 1200,
+ *   hardware: 'auto',
+ *   iceServers: [{ urls: 'stun:stun.l.google.com:19302' }],
+ *   onIceCandidate: (candidate) => {
+ *     sendToClient({ type: 'candidate', value: candidate });
+ *   }
+ * });
+ *
+ * // Process SDP offer from client
+ * const answer = await session.setOffer(clientOffer);
+ * sendToClient({ type: 'answer', value: answer });
+ *
+ * // Start streaming
+ * await session.start();
+ * ```
+ *
+ * @example
+ * ```typescript
+ * // Complete WebSocket signaling server
+ * import { WebSocket } from 'ws';
+ *
+ * ws.on('message', async (data) => {
+ *   const msg = JSON.parse(data);
+ *
+ *   if (msg.type === 'offer') {
+ *     const session = await WebRTCStream.create(msg.url, {
+ *       hardware: 'auto',
+ *       onIceCandidate: (candidate) => {
+ *         ws.send(JSON.stringify({ type: 'candidate', value: candidate }));
+ *       }
+ *     });
+ *
+ *     const answer = await session.setOffer(msg.value);
+ *     ws.send(JSON.stringify({ type: 'answer', value: answer }));
+ *
+ *     await session.start();
+ *   } else if (msg.type === 'candidate') {
+ *     session.addIceCandidate(msg.value);
+ *   }
+ * });
+ * ```
+ *
+ * @see {@link RTPStream} For library-agnostic RTP streaming
+ * @see {@link Demuxer} For input media handling
+ */
+export class WebRTCStream {
+    stream;
+    pc = null;
+    videoTrack = null;
+    audioTrack = null;
+    options;
+    pendingIceCandidates = [];
+    frameSource;
+    stopping = false;
+    /**
+     * @param options - Session configuration options
+     *
+     * Use {@link create} factory method
+     *
+     * @internal
+     */
+    constructor(options) {
+        this.options = {
+            onIceCandidate: options.onIceCandidate ?? (() => { }),
+            onClose: options.onClose ?? (() => { }),
+            iceServers: options.iceServers ?? [],
+            ...options,
+        };
+    }
+    /**
+     * Create a WebRTC session from a media source.
+     *
+     * Opens the input media, creates internal streaming components, and prepares
+     * for WebRTC peer connection negotiation. Does not start streaming yet.
+     * Call {@link setOffer} to negotiate SDP and {@link start} to begin streaming.
+     *
+     * @param input - Media source URL (RTSP, file path, HTTP, etc.) or a pre-opened {@link Demuxer}
+     *
+     * @param options - Session configuration options
+     *
+     * @returns Configured WebRTC session instance
+     *
+     * @example
+     * ```typescript
+     * const session = WebRTCStream.create('rtsp://camera.local/stream', {
+     *   mtu: 1200,
+     *   hardware: 'auto',
+     *   iceServers: [{ urls: 'stun:stun.l.google.com:19302' }]
+     * });
+     * ```
+     *
+     * @example
+     * ```typescript
+     * // Session from a pre-opened Demuxer (e.g. screen capture)
+     * const demuxer = await Device.openScreen({ frameRate: 30 });
+     * const session = WebRTCStream.create(demuxer, {
+     *   hardware: 'auto'
+     * });
+     * ```
+     */
+    static create(input, options = {}) {
+        const session = new WebRTCStream(options);
+        // Remember a frame source so negotiation can advertise the encoder's target
+        // codecs (there's no input stream to read a codec from).
+        if (isMediaFrameSource(input)) {
+            session.frameSource = input;
+        }
+        // Create stream with WebRTC-specific codec support
+        session.stream = RTPStream.create(input, {
+            video: options.video,
+            audio: options.audio,
+            hardware: options.hardware,
+            inputOptions: options.inputOptions,
+            supportedVideoCodecs: [AV_CODEC_ID_H264, AV_CODEC_ID_HEVC, AV_CODEC_ID_VP8, AV_CODEC_ID_VP9, AV_CODEC_ID_AV1],
+            supportedAudioCodecs: [AV_CODEC_ID_OPUS, AV_CODEC_ID_PCM_ALAW, AV_CODEC_ID_PCM_MULAW],
+            onVideoPacket: (rtp) => {
+                session.videoTrack?.writeRtp(rtp);
+            },
+            onAudioPacket: (rtp) => {
+                session.audioTrack?.writeRtp(rtp);
+            },
+            onClose: (error) => {
+                session.options.onClose?.(error);
+            },
+        });
+        return session;
+    }
+    /**
+     * Start streaming media to the WebRTC peer connection.
+     *
+     * Begins the media processing pipeline, reading packets from input,
+     * transcoding if necessary, and sending RTP packets to the media tracks.
+     * {@link setOffer} calls this automatically after negotiating, so calling it
+     * explicitly is optional. It is idempotent - a useful pattern for demuxer
+     * inputs is calling `start()` *before* `setOffer()` so the input is open and
+     * its actual codecs are advertised in the answer.
+     *
+     * @returns Promise that resolves once streaming has started (runs in the background)
+     *
+     * @throws {FFmpegError} If transcoding or muxing fails
+     *
+     * @example
+     * ```typescript
+     * const session = WebRTCStream.create('rtsp://camera.local/stream');
+     *
+     * await session.start(); // open input so getCodecs() sees the real codecs
+     * const answer = await session.setOffer(remoteOffer);
+     * sendToRemote(answer);
+     * ```
+     *
+     * @see {@link setOffer} Which starts the stream automatically after negotiation
+     */
+    async start() {
+        await this.stream.start();
+    }
+    /**
+     * Stop streaming gracefully and clean up all resources.
+     *
+     * Stops the stream, closes peer connection, and releases all resources.
+     * Safe to call multiple times. After stopping, you can call start() again
+     * to restart the session.
+     *
+     * @example
+     * ```typescript
+     * const session = await WebRTCStream.create('input.mp4');
+     * await session.start();
+     *
+     * // Stop after 10 seconds
+     * setTimeout(async () => await session.stop(), 10000);
+     * ```
+     */
+    async stop() {
+        if (this.stopping) {
+            return;
+        }
+        this.stopping = true;
+        try {
+            await this.stream.stop();
+            try {
+                await this.pc?.close();
+            }
+            catch {
+                // Peer connection may already be closed.
+            }
+            this.videoTrack = null;
+            this.audioTrack = null;
+            this.pc = null;
+            this.pendingIceCandidates = [];
+        }
+        finally {
+            this.stopping = false;
+        }
+    }
+    /**
+     * Get detected codec information.
+     *
+     * Returns RTP codec parameters and FFmpeg codec IDs for video and audio.
+     * Useful for inspecting what codecs will be used in the WebRTC session.
+     * The input is automatically opened during {@link create}, so codecs can be
+     * detected immediately after session creation.
+     *
+     * @returns Codec configuration for video and audio streams
+     *
+     * @example
+     * ```typescript
+     * const session = await WebRTCStream.create('input.mp4');
+     * const codecs = session.getCodecs();
+     *
+     * console.log('Video:', codecs.video.mimeType);
+     * console.log('Audio:', codecs.audio?.mimeType);
+     * ```
+     */
+    getCodecs() {
+        // Frame source: no input stream - advertise the encoder's target codecs
+        // (RTPStream's frame path encodes video to H264 and audio to OPUS), gated on
+        // which frame streams are present.
+        if (this.frameSource) {
+            return {
+                video: this.frameSource.video ? this.getVideoCodecConfig(AV_CODEC_ID_H264) : undefined,
+                audio: this.frameSource.audio ? this.getAudioCodecConfig(AV_CODEC_ID_OPUS) : undefined,
+            };
+        }
+        const input = this.stream.getInput();
+        if (!input) {
+            return {
+                video: undefined,
+                audio: undefined,
+            };
+        }
+        const videoStream = input.video();
+        const audioStream = input.audio();
+        // Try the input codec first; if unsupported (e.g. rawvideo from device capture),
+        // fall back to the first supported video codec (the transcode target)
+        let videoConfig = this.getVideoCodecConfig(videoStream?.codecpar.codecId);
+        if (!videoConfig && videoStream) {
+            const supportedVideoCodecs = [AV_CODEC_ID_H264, AV_CODEC_ID_HEVC, AV_CODEC_ID_VP8, AV_CODEC_ID_VP9, AV_CODEC_ID_AV1];
+            for (const codecId of supportedVideoCodecs) {
+                videoConfig = this.getVideoCodecConfig(codecId);
+                if (videoConfig)
+                    break;
+            }
+        }
+        return {
+            video: videoConfig,
+            audio: this.getAudioCodecConfig(audioStream?.codecpar.codecId),
+        };
+    }
+    /**
+     * Process SDP offer from remote peer and generate SDP answer.
+     *
+     * Creates RTCPeerConnection with detected codecs, sets up media tracks,
+     * processes the remote SDP offer, and generates a local SDP answer.
+     * Also configures ICE candidate handling via `onIceCandidate` callback in options.
+     * Must be called before {@link start}.
+     * Calling it again renegotiates: the previous peer connection is closed and
+     * replaced by a new one built from the new offer, while streaming continues.
+     *
+     * @param offerSdp - SDP offer string from remote WebRTC peer
+     *
+     * @returns SDP answer string to send back to remote peer
+     *
+     * @example
+     * ```typescript
+     * const session = await WebRTCStream.create('input.mp4');
+     *
+     * // Setup ICE candidate handler first
+     * session.onIceCandidate = (candidate) => {
+     *   sendToRemote({ type: 'candidate', value: candidate });
+     * };
+     *
+     * // Process offer and send answer
+     * const answer = await session.setOffer(remoteOffer);
+     * sendToRemote({ type: 'answer', value: answer });
+     *
+     * // Start streaming
+     * await session.start();
+     * ```
+     */
+    async setOffer(offerSdp) {
+        // Renegotiation: dispose any previous peer connection first. Left alive, its
+        // connectionstatechange handler would later report 'failed'/'closed' and tear
+        // down the new session via stop(). Buffered candidates belong to the old ICE
+        // session, so drop them too.
+        if (this.pc) {
+            const oldPc = this.pc;
+            this.pc = null;
+            this.videoTrack = null;
+            this.audioTrack = null;
+            this.pendingIceCandidates = [];
+            try {
+                await oldPc.close();
+            }
+            catch {
+                // Old connection may already be gone - proceed with the new one.
+            }
+        }
+        const codecs = this.getCodecs();
+        const videoConfig = codecs.video ?? {
+            mimeType: 'video/H264',
+            clockRate: 90000,
+            payloadType: 96,
+        };
+        delete videoConfig.codecId;
+        const audioConfig = codecs.audio ?? {
+            mimeType: 'audio/opus',
+            clockRate: 48000,
+            channels: 2,
+            payloadType: 111,
+        };
+        delete audioConfig.codecId;
+        // Create PeerConnection with detected codecs
+        const codecParams = {
+            video: [
+                new RTCRtpCodecParameters({
+                    ...videoConfig,
+                }),
+            ],
+            audio: [
+                new RTCRtpCodecParameters({
+                    ...audioConfig,
+                }),
+            ],
+        };
+        const pc = new RTCPeerConnection({
+            codecs: codecParams,
+            iceServers: this.options.iceServers,
+        });
+        this.pc = pc;
+        // Tear down when the peer goes away. werift verifies ICE consent (RFC 7675)
+        // and reports 'failed' once the browser disconnects (tab closed/reloaded) -
+        // without this, the pipeline would keep encoding into the void forever.
+        pc.connectionStateChange.subscribe((state) => {
+            // Ignore stale events from a connection that was replaced by a newer
+            // setOffer() - they must not tear down the current session.
+            if (this.pc !== pc) {
+                return;
+            }
+            if ((state === 'failed' || state === 'closed') && !this.stopping) {
+                this.stop().catch(() => {
+                    // Best-effort teardown - errors already surface via onClose(error).
+                });
+            }
+        });
+        // Setup ICE candidate handling
+        pc.onIceCandidate.subscribe((candidate) => {
+            if (candidate?.candidate && this.options.onIceCandidate) {
+                this.options.onIceCandidate(candidate.candidate);
+            }
+        });
+        // Setup tracks
+        pc.onRemoteTransceiverAdded.subscribe(async (transceiver) => {
+            if (transceiver.kind === 'video') {
+                this.videoTrack = new MediaStreamTrack({ kind: 'video' });
+                transceiver.sender.replaceTrack(this.videoTrack);
+                transceiver.setDirection('sendonly');
+                // Answer PLI feedback with a fresh keyframe - a receiver that connected
+                // after the stream started (or lost packets) can't decode until the next
+                // IDR, and it asks for one via RTCP PLI.
+                transceiver.sender.onPictureLossIndication.subscribe(() => {
+                    this.stream.requestVideoKeyframe();
+                });
+            }
+            else if (transceiver.kind === 'audio' && this.audioTrack === null) {
+                this.audioTrack = new MediaStreamTrack({ kind: 'audio' });
+                transceiver.sender.replaceTrack(this.audioTrack);
+                transceiver.setDirection('sendonly');
+            }
+            else if (transceiver.kind === 'audio') {
+                // Backchannel
+                const [track] = await transceiver.onTrack.asPromise();
+                const input = this.stream.getInput();
+                const ctx = input?.getFormatContext();
+                const streams = ctx?.getRTSPStreamInfo();
+                const backchannel = streams?.find((s) => s.direction === 'sendonly' && s.mediaType === 'audio');
+                track.onReceiveRtp.subscribe(async (rtp) => {
+                    if (backchannel && this.stream.isStreamActive) {
+                        try {
+                            await ctx?.sendRTSPPacket(backchannel.streamIndex, rtp.serialize());
+                        }
+                        catch {
+                            // Ignore send errors
+                        }
+                    }
+                });
+            }
+        });
+        // Set remote description and create answer
+        await pc.setRemoteDescription(new RTCSessionDescription(offerSdp, 'offer'));
+        // The sender transmits with the FIRST negotiated codec, and browsers offer
+        // several H264 variants (packetization-mode=1 and =0). FFmpeg's RTP payloader
+        // fragments large NALs (FU-A), which is invalid under packetization-mode=0 -
+        // if that entry happened to be first, the receiver would silently discard
+        // every fragmented (key)frame. Prefer a packetization-mode=1 H264 entry.
+        for (const transceiver of pc.getTransceivers()) {
+            if (transceiver.kind !== 'video') {
+                continue;
+            }
+            transceiver.codecs = [...transceiver.codecs].sort((a, b) => {
+                const score = (codec) => codec.mimeType.toLowerCase() === 'video/h264' && !(codec.parameters ?? '').includes('packetization-mode=1') ? 1 : 0;
+                return score(a) - score(b);
+            });
+        }
+        const answer = await pc.createAnswer();
+        // Must be awaited: localDescription is only complete afterwards, and an
+        // un-awaited rejection would otherwise be unhandled.
+        await pc.setLocalDescription(answer);
+        // Apply any buffered ICE candidates now that remote description is set
+        if (this.pendingIceCandidates.length > 0) {
+            for (const candidate of this.pendingIceCandidates) {
+                pc.addIceCandidate(new RTCIceCandidate({ candidate }));
+            }
+            this.pendingIceCandidates = [];
+        }
+        // Start streaming - negotiation is done and the tracks exist, so the caller
+        // doesn't have to sequence start() themselves. Idempotent: calling start()
+        // before setOffer() (e.g. to have the input open for codec detection) is fine.
+        await this.start();
+        return pc.localDescription?.sdp ?? '';
+    }
+    /**
+     * Add ICE candidate from remote peer.
+     *
+     * Processes ICE candidates received from the remote peer via signaling channel.
+     * Should be called whenever a new candidate message arrives from remote peer.
+     * Can be called multiple times as candidates are discovered.
+     *
+     * Supports ICE trickling: If called before {@link setOffer}, candidates are buffered
+     * and applied automatically once the remote description is set.
+     *
+     * @param candidate - ICE candidate string from remote peer
+     *
+     * @example
+     * ```typescript
+     * // In signaling message handler
+     * if (msg.type === 'candidate') {
+     *   session.addIceCandidate(msg.value);
+     * }
+     * ```
+     */
+    addIceCandidate(candidate) {
+        // Buffer candidates if peer connection not ready yet (ICE trickling)
+        if (!this.pc?.remoteDescription) {
+            this.pendingIceCandidates.push(candidate);
+            return;
+        }
+        // Apply candidate immediately if peer connection is ready
+        this.pc.addIceCandidate(new RTCIceCandidate({ candidate }));
+    }
+    /**
+     * Get video codec configuration from input stream.
+     *
+     * @param codecId - FFmpeg codec ID
+     *
+     * @returns RTP codec parameters with codec ID or undefined if unsupported
+     *
+     * @internal
+     */
+    getVideoCodecConfig(codecId) {
+        let mimeType;
+        const clockRate = 90000;
+        let payloadType;
+        switch (codecId) {
+            case AV_CODEC_ID_H264:
+                mimeType = 'video/H264';
+                payloadType = 96;
+                break;
+            case AV_CODEC_ID_HEVC:
+                mimeType = 'video/H265';
+                payloadType = 96;
+                break;
+            case AV_CODEC_ID_VP8:
+                mimeType = 'video/VP8';
+                payloadType = 96;
+                break;
+            case AV_CODEC_ID_VP9:
+                mimeType = 'video/VP9';
+                payloadType = 98;
+                break;
+            case AV_CODEC_ID_AV1:
+                mimeType = 'video/AV1';
+                payloadType = 98;
+                break;
+            default:
+                return undefined;
+        }
+        return { mimeType, clockRate, payloadType, codecId };
+    }
+    /**
+     * Get audio codec configuration from input stream.
+     *
+     * @param codecId - FFmpeg codec ID
+     *
+     * @returns RTP codec parameters with codec ID or undefined if unsupported
+     *
+     * @internal
+     */
+    getAudioCodecConfig(codecId) {
+        let mimeType;
+        let clockRate;
+        let channels;
+        let payloadType;
+        switch (codecId) {
+            case AV_CODEC_ID_OPUS:
+                mimeType = 'audio/opus';
+                clockRate = 48000;
+                channels = 2;
+                payloadType = 111;
+                break;
+            case AV_CODEC_ID_PCM_ALAW:
+                mimeType = 'audio/PCMA';
+                clockRate = 8000;
+                channels = 1;
+                payloadType = 8;
+                break;
+            case AV_CODEC_ID_PCM_MULAW:
+                mimeType = 'audio/PCMU';
+                clockRate = 8000;
+                channels = 1;
+                payloadType = 0;
+                break;
+            default:
+                return undefined;
+        }
+        return { mimeType, clockRate, channels, payloadType, codecId };
+    }
+}
+//# sourceMappingURL=webrtc-stream.js.map

package/dist/api/webrtc-stream.js.map

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