@gjsify/webrtc 0.4.0 → 0.4.4

package/src/rtc-rtp-receiver.ts

@@ -1,122 +0,0 @@
-// W3C RTCRtpReceiver for GJS.
-//
-// Wraps GstWebRTC.WebRTCRTPReceiver. Phase 2.5: incoming media pipeline
-// is managed by ReceiverBridge (Vala) which handles decodebin's
-// streaming-thread signals natively and emits media-flowing on the
-// main thread when decoded media replaces the muted source.
-//
-// Reference: refs/node-gst-webrtc/src/webrtc/RTCRtpReceiver.ts (ISC)
-// Reference: W3C WebRTC spec § 5.3
-
-import type GstWebRTC from 'gi://GstWebRTC?version=1.0';
-import type Gst from 'gi://Gst?version=1.0';
-
-import {
-    ReceiverBridge,
-    type ReceiverBridge as ReceiverBridgeType,
-} from '@gjsify/webrtc-native';
-
-import { MediaStreamTrack } from './media-stream-track.js';
-import { getRtpCapabilities } from './rtp-capabilities.js';
-import type { RTCStatsReport } from './rtc-stats-report.js';
-import type { RTCDtlsTransport } from './rtc-dtls-transport.js';
-import type { RTCRtpCapabilities, RTCRtpCodecParameters, RTCRtpHeaderExtensionParameters, RTCRtcpParameters } from './rtc-rtp-sender.js';
-
-export interface RTCRtpReceiveParameters {
-    codecs: RTCRtpCodecParameters[];
-    headerExtensions: RTCRtpHeaderExtensionParameters[];
-    rtcp: RTCRtcpParameters;
-}
-
-export interface RTCRtpContributingSource {
-    timestamp: number;
-    source: number;
-    audioLevel?: number;
-    rtpTimestamp: number;
-}
-
-export interface RTCRtpSynchronizationSource extends RTCRtpContributingSource {
-    voiceActivityFlag?: boolean;
-}
-
-const MAX_JITTER_BUFFER_TARGET = 4000;
-
-export class RTCRtpReceiver {
-    private _gstReceiver: GstWebRTC.WebRTCRTPReceiver | null;
-    private _track: MediaStreamTrack;
-    private _jitterBufferTarget: number | null = null;
-    private _pipeline: any = null;
-    private _receiverBridge: ReceiverBridgeType | null = null;
-    /** @internal — stats callback set by RTCPeerConnection */
-    _getStatsForTrack: ((track: MediaStreamTrack) => Promise<RTCStatsReport>) | null = null;
-
-    constructor(kind: 'audio' | 'video', gstReceiver: GstWebRTC.WebRTCRTPReceiver | null, pipeline?: any) {
-        this._gstReceiver = gstReceiver;
-        this._pipeline = pipeline ?? null;
-        this._track = new MediaStreamTrack({ kind, muted: true });
-    }
-
-    /** @internal — called from RTCPeerConnection._handlePadAdded */
-    _connectToPad(pad: Gst.Pad): void {
-        if (!this._pipeline || this._receiverBridge) return;
-        this._receiverBridge = new (ReceiverBridge as any)({
-            pipeline: this._pipeline,
-            kind: this._track.kind,
-        });
-        this._receiverBridge!.connect_to_pad(pad);
-        this._receiverBridge!.connect('media-flowing', () => {
-            this._track._setMuted(false);
-        });
-    }
-
-    /** @internal — called from RTCPeerConnection.close() */
-    _dispose(): void {
-        try { this._receiverBridge?.dispose_bridge(); } catch { /* ignore */ }
-        this._receiverBridge = null;
-    }
-
-    /** @internal — set by RTCPeerConnection */
-    _transport: RTCDtlsTransport | null = null;
-
-    get track(): MediaStreamTrack { return this._track; }
-    get transport(): RTCDtlsTransport | null { return this._transport; }
-
-    get jitterBufferTarget(): number | null { return this._jitterBufferTarget; }
-    set jitterBufferTarget(v: number | null) {
-        if (v === null) {
-            this._jitterBufferTarget = null;
-            return;
-        }
-        const n = Number(v);
-        if (!Number.isFinite(n) || n < 0) {
-            throw new RangeError(`Failed to set jitterBufferTarget: ${v} is negative or not finite`);
-        }
-        if (n > MAX_JITTER_BUFFER_TARGET) {
-            throw new RangeError(`Failed to set jitterBufferTarget: ${v} exceeds maximum of ${MAX_JITTER_BUFFER_TARGET}`);
-        }
-        this._jitterBufferTarget = n;
-    }
-
-    getParameters(): RTCRtpReceiveParameters {
-        return {
-            codecs: [],
-            headerExtensions: [],
-            rtcp: {},
-        };
-    }
-
-    getContributingSources(): RTCRtpContributingSource[] { return []; }
-    getSynchronizationSources(): RTCRtpSynchronizationSource[] { return []; }
-
-    async getStats(): Promise<RTCStatsReport> {
-        if (this._getStatsForTrack && this._track) {
-            return this._getStatsForTrack(this._track);
-        }
-        const { RTCStatsReport: Report } = await import('./rtc-stats-report.js');
-        return new Report();
-    }
-
-    static getCapabilities(kind: string): RTCRtpCapabilities | null {
-        return getRtpCapabilities(kind);
-    }
-}

package/src/rtc-rtp-sender.ts

@@ -1,471 +0,0 @@
-// W3C RTCRtpSender for GJS.
-//
-// Phase 2: API surface wrapping GstWebRTC.WebRTCRTPSender.
-// Phase 3: outgoing media pipeline — builds explicit encoder chains
-// (source → valve → convert → encode → payloader → webrtcbin sink pad)
-// entirely on the main thread. No Vala bridge needed.
-//
-// Reference: refs/node-gst-webrtc/src/webrtc/RTCRtpSender.ts (ISC)
-// Reference: W3C WebRTC spec § 5.2
-
-import type GstNs from 'gi://Gst?version=1.0';
-import type GstWebRTC from 'gi://GstWebRTC?version=1.0';
-
-import { Gst } from './gst-init.js';
-import { getRtpCapabilities } from './rtp-capabilities.js';
-import { RTCDTMFSender } from './rtc-dtmf-sender.js';
-import { TeeMultiplexer } from './tee-multiplexer.js';
-import {
-    asValveElement,
-    asRtpPayloaderElement,
-    asCapsFilterElement,
-    asVp8EncElement,
-    type ValveElement,
-} from './internal/gst-types.js';
-import type { RTCStatsReport } from './rtc-stats-report.js';
-import type { RTCDtlsTransport } from './rtc-dtls-transport.js';
-import type { MediaStreamTrack } from './media-stream-track.js';
-import type { MediaStream } from './media-stream.js';
-
-// Standard RTP payload types used in WebRTC SDP
-const OPUS_PAYLOAD_TYPE = 111;
-const VP8_PAYLOAD_TYPE = 96;
-
-export type RTCRtpTransceiverDirection = 'sendrecv' | 'sendonly' | 'recvonly' | 'inactive' | 'stopped';
-
-export interface RTCRtpCodecCapability {
-    mimeType: string;
-    clockRate: number;
-    channels?: number;
-    sdpFmtpLine?: string;
-}
-
-export interface RTCRtpHeaderExtensionCapability {
-    uri: string;
-}
-
-export interface RTCRtpCapabilities {
-    codecs: RTCRtpCodecCapability[];
-    headerExtensions: RTCRtpHeaderExtensionCapability[];
-}
-
-export interface RTCRtpEncodingParameters {
-    rid?: string;
-    active?: boolean;
-    maxBitrate?: number;
-    maxFramerate?: number;
-    scaleResolutionDownBy?: number;
-}
-
-export interface RTCRtpCodecParameters {
-    payloadType: number;
-    mimeType: string;
-    clockRate: number;
-    channels?: number;
-    sdpFmtpLine?: string;
-}
-
-export interface RTCRtpHeaderExtensionParameters {
-    uri: string;
-    id: number;
-    encrypted?: boolean;
-}
-
-export interface RTCRtcpParameters {
-    cname?: string;
-    reducedSize?: boolean;
-}
-
-export interface RTCRtpSendParameters {
-    transactionId: string;
-    encodings: RTCRtpEncodingParameters[];
-    codecs: RTCRtpCodecParameters[];
-    headerExtensions: RTCRtpHeaderExtensionParameters[];
-    rtcp: RTCRtcpParameters;
-}
-
-let _txCounter = 0;
-
-export class RTCRtpSender {
-    private _gstSender: GstWebRTC.WebRTCRTPSender | null;
-    private _track: MediaStreamTrack | null = null;
-    private _lastParams: RTCRtpSendParameters | null = null;
-
-    /** @internal GStreamer pipeline references (set by RTCPeerConnection) */
-    private _pipeline: GstNs.Pipeline | null = null;
-    private _webrtcbin: GstNs.Element | null = null;
-    private _mlineIndex: number = -1;
-    private _elements: GstNs.Element[] = [];
-    private _valve: ValveElement | null = null;
-    _linked = false;
-    /** @internal — tee src pad if this sender uses a shared source */
-    private _teeSrcPad: GstNs.Pad | null = null;
-    /** @internal — stats callback set by RTCPeerConnection */
-    _getStatsForTrack: ((track: MediaStreamTrack) => Promise<RTCStatsReport>) | null = null;
-    /** @internal — set by RTCPeerConnection */
-    _transport: RTCDtlsTransport | null = null;
-    /** @internal — DTMF sender, created lazily for audio senders */
-    private _dtmf: RTCDTMFSender | null = null;
-    /** @internal — the kind of media this sender handles */
-    _kind: 'audio' | 'video' | null = null;
-    /** @internal — back-reference for DTMF stopped/direction checks */
-    _transceiver: { stopped: boolean; currentDirection: string | null } | null = null;
-    /** @internal — callback to notify RTCPeerConnection when pipeline changes (cross-pipeline fix) */
-    _onPipelineChanged: ((newPipeline: GstNs.Pipeline) => void) | null = null;
-
-    constructor(
-        gstSender: GstWebRTC.WebRTCRTPSender | null,
-        pipeline?: GstNs.Pipeline,
-        webrtcbin?: GstNs.Element,
-    ) {
-        this._gstSender = gstSender;
-        this._pipeline = pipeline ?? null;
-        this._webrtcbin = webrtcbin ?? null;
-    }
-
-    get track(): MediaStreamTrack | null { return this._track; }
-
-    /** Returns the DTMF sender for audio senders, null for video. */
-    get dtmf(): RTCDTMFSender | null {
-        // Determine kind from track or from what was set by PC
-        const kind = this._track?.kind ?? this._kind;
-        if (kind !== 'audio') return null;
-        if (!this._dtmf) {
-            const dtmf = new RTCDTMFSender();
-            // Wire transceiver state checks for insertDTMF validation
-            dtmf._isStopped = () => this._transceiver?.stopped ?? false;
-            dtmf._getCurrentDirection = () => this._transceiver?.currentDirection ?? null;
-            this._dtmf = dtmf;
-        }
-        return this._dtmf;
-    }
-    get transport(): RTCDtlsTransport | null { return this._transport; }
-
-    /** @internal */
-    _setTrack(track: MediaStreamTrack | null): void {
-        if (track === null && this._linked) {
-            this._teardownPipeline();
-        }
-        this._track = track;
-    }
-
-    /** @internal — called by RTCPeerConnection._createTransceiverWrapper */
-    _setMlineIndex(index: number): void { this._mlineIndex = index; }
-
-    /** @internal — build the outgoing encoder chain and link to webrtcbin */
-    _wirePipeline(track: MediaStreamTrack): void {
-        if (this._linked || !this._pipeline || !this._webrtcbin) return;
-        const source = track._gstSource as GstNs.Element | null;
-        if (!source) return; // No GStreamer backing — nothing to wire
-
-        // MediaStreamTrack's GStreamer-backing fields are intentionally typed
-        // `any` on the track itself (they are runtime-attached by the
-        // get-user-media / VideoBridge code paths). We narrow them locally
-        // here to keep the rest of this method strongly typed.
-        const trackGst = track as MediaStreamTrack & {
-            _gstSource: GstNs.Element | null;
-            _gstPipeline: GstNs.Pipeline | null;
-            _gstTee: GstNs.Element | null;
-            _teeMultiplexer: TeeMultiplexer | null;
-        };
-        let sourceForChain: GstNs.Element | null; // source directly or null when tee branch
-
-        if (trackGst._gstTee && trackGst._gstPipeline && trackGst._gstPipeline !== this._pipeline) {
-            // Source has a tee from VideoBridge (preview) in a different pipeline.
-            // Instead of moving the source, we add our encoder chain elements to
-            // the SOURCE's pipeline and request a new branch from its tee.
-            // The webrtcbin must also move to the source pipeline.
-            const sourcePipeline = trackGst._gstPipeline;
-            const tee = trackGst._gstTee;
-
-            // Move webrtcbin to the source pipeline so everything is in one pipeline
-            if (this._webrtcbin.get_parent() === this._pipeline) {
-                this._pipeline.set_state(Gst.State.NULL);
-                this._pipeline.remove(this._webrtcbin);
-            }
-            sourcePipeline.add(this._webrtcbin);
-            this._webrtcbin.sync_state_with_parent();
-
-            // Switch our pipeline reference to the source's pipeline
-            this._pipeline = sourcePipeline;
-            // Notify the owning RTCPeerConnection to update its pipeline reference
-            this._onPipelineChanged?.(sourcePipeline);
-
-            // Request a new branch from the existing tee
-            const teeSrcPad = tee.request_pad_simple
-                ? tee.request_pad_simple('src_%u')
-                : tee.get_request_pad('src_%u');
-            this._teeSrcPad = teeSrcPad;
-            sourceForChain = null; // We'll link via pad below
-        } else if (trackGst._teeMultiplexer) {
-            // Track already has a TeeMultiplexer (shared with another PC) — request a new branch
-            const tee = trackGst._teeMultiplexer as TeeMultiplexer;
-            const teeSrcPad = tee.requestSrcPad();
-            this._teeSrcPad = teeSrcPad;
-            sourceForChain = null; // We'll link via pad below
-        } else if (trackGst._gstPipeline && trackGst._gstPipeline !== this._pipeline) {
-            // Source is in another pipeline (no tee from VideoBridge) — this is the
-            // second PC using this track. Insert a tee between source and the
-            // existing consumer. Move source to this pipeline and create a tee.
-            const oldPipeline = trackGst._gstPipeline;
-
-            // First, unlink source from its current peer (the first sender's valve)
-            const sourceSrcPad = source.get_static_pad('src');
-            const oldPeer = sourceSrcPad?.get_peer?.();
-            if (oldPeer) sourceSrcPad.unlink(oldPeer);
-
-            source.set_state(Gst.State.NULL);
-            oldPipeline.remove(source);
-            this._pipeline.add(source);
-            trackGst._gstPipeline = this._pipeline;
-
-            // Create tee in this pipeline
-            const tee = new TeeMultiplexer(this._pipeline, source);
-            trackGst._teeMultiplexer = tee;
-
-            // Reconnect the old consumer (first sender) via a tee branch
-            if (oldPeer) {
-                const firstBranch = tee.requestSrcPad();
-                if (firstBranch) firstBranch.link(oldPeer);
-            }
-
-            // Request a branch for this sender
-            const teeSrcPad = tee.requestSrcPad();
-            this._teeSrcPad = teeSrcPad;
-            sourceForChain = null;
-        } else {
-            // First PC to use this track — move source directly
-            const oldPipeline = trackGst._gstPipeline;
-            if (oldPipeline && oldPipeline !== this._pipeline) {
-                source.set_state(Gst.State.NULL);
-                oldPipeline.remove(source);
-                trackGst._gstPipeline = this._pipeline;
-            }
-            if (source.get_parent() !== this._pipeline) {
-                this._pipeline.add(source);
-            }
-            sourceForChain = source;
-        }
-
-        // Valve element for Track.enabled control
-        const valve = asValveElement(Gst.ElementFactory.make('valve', null)!);
-        valve.drop = !track.enabled;
-        this._valve = valve;
-        this._pipeline.add(valve);
-
-        const elements: GstNs.Element[] = [valve];
-        let lastElement: GstNs.Element;
-
-        if (track.kind === 'audio') {
-            const convert = Gst.ElementFactory.make('audioconvert', null)!;
-            const resample = Gst.ElementFactory.make('audioresample', null)!;
-            const encoder = Gst.ElementFactory.make('opusenc', null)!;
-            const payloader = asRtpPayloaderElement(Gst.ElementFactory.make('rtpopuspay', null)!);
-            payloader.pt = OPUS_PAYLOAD_TYPE;
-
-            // capsfilter tells webrtcbin the RTP caps immediately so createOffer
-            // can generate the m=audio line without waiting for data to flow.
-            const capsfilter = asCapsFilterElement(Gst.ElementFactory.make('capsfilter', null)!);
-            capsfilter.caps = Gst.Caps.from_string(
-                `application/x-rtp,media=audio,encoding-name=OPUS,clock-rate=48000,payload=${OPUS_PAYLOAD_TYPE}`,
-            );
-
-            elements.push(convert, resample, encoder, payloader, capsfilter);
-            for (const el of elements) this._pipeline.add(el);
-
-            // Link source/tee → valve
-            if (this._teeSrcPad) {
-                const valveSinkPad = valve.get_static_pad('sink');
-                this._teeSrcPad.link(valveSinkPad);
-            } else if (sourceForChain) {
-                sourceForChain.link(valve);
-            }
-            valve.link(convert);
-            convert.link(resample);
-            resample.link(encoder);
-            encoder.link(payloader);
-            payloader.link(capsfilter);
-            lastElement = capsfilter;
-        } else {
-            // Video
-            const convert = Gst.ElementFactory.make('videoconvert', null)!;
-            const scale = Gst.ElementFactory.make('videoscale', null)!;
-            const encoder = asVp8EncElement(Gst.ElementFactory.make('vp8enc', null)!);
-            encoder.deadline = 1; // Realtime encoding
-            encoder.keyframe_max_dist = 60;
-            const payloader = asRtpPayloaderElement(Gst.ElementFactory.make('rtpvp8pay', null)!);
-            payloader.pt = VP8_PAYLOAD_TYPE;
-
-            const capsfilter = asCapsFilterElement(Gst.ElementFactory.make('capsfilter', null)!);
-            capsfilter.caps = Gst.Caps.from_string(
-                `application/x-rtp,media=video,encoding-name=VP8,clock-rate=90000,payload=${VP8_PAYLOAD_TYPE}`,
-            );
-
-            elements.push(convert, scale, encoder, payloader, capsfilter);
-            for (const el of elements) this._pipeline.add(el);
-
-            // Link source/tee → valve
-            if (this._teeSrcPad) {
-                const valveSinkPad = valve.get_static_pad('sink');
-                this._teeSrcPad.link(valveSinkPad);
-            } else if (sourceForChain) {
-                sourceForChain.link(valve);
-            }
-            valve.link(convert);
-            convert.link(scale);
-            scale.link(encoder);
-            encoder.link(payloader);
-            payloader.link(capsfilter);
-            lastElement = capsfilter;
-        }
-
-        // Link payloader output to webrtcbin sink pad.
-        // Use the mline index if assigned (>= 0), otherwise request next available.
-        const padName = this._mlineIndex >= 0 ? `sink_${this._mlineIndex}` : 'sink_%u';
-        const sinkPad = this._webrtcbin.request_pad_simple
-            ? this._webrtcbin.request_pad_simple(padName)
-            : this._webrtcbin.get_request_pad(padName);
-        if (sinkPad) {
-            const srcPad = lastElement.get_static_pad('src');
-            srcPad.link(sinkPad);
-        }
-
-        // Sync states — elements added to a PLAYING pipeline
-        // If using a tee branch, don't include the shared source in our element list
-        const ownedElements = this._teeSrcPad ? elements : [source, ...elements];
-        for (const el of ownedElements) {
-            el.sync_state_with_parent();
-        }
-
-        this._elements = ownedElements;
-        this._linked = true;
-
-        // Wire Track.enabled → valve.drop
-        track._setEnableCallback((enabled: boolean) => {
-            if (this._valve) this._valve.drop = !enabled;
-        });
-    }
-
-    /** @internal — tear down the encoder chain on close/removeTrack */
-    _teardownPipeline(): void {
-        if (!this._linked) return;
-        // Disconnect enable callback from the track
-        if (this._track) {
-            this._track._setEnableCallback(null);
-        }
-        // Release tee branch if using shared source
-        if (this._teeSrcPad && this._track?._teeMultiplexer) {
-            try {
-                (this._track._teeMultiplexer as TeeMultiplexer).releaseSrcPad(this._teeSrcPad);
-            } catch { /* ignore */ }
-            this._teeSrcPad = null;
-        }
-        for (const el of [...this._elements].reverse()) {
-            try {
-                el.set_state(Gst.State.NULL);
-                this._pipeline?.remove(el);
-            } catch { /* ignore cleanup errors */ }
-        }
-        this._elements = [];
-        this._valve = null;
-        this._linked = false;
-    }
-
-    getParameters(): RTCRtpSendParameters {
-        if (!this._lastParams) {
-            this._lastParams = {
-                transactionId: String(++_txCounter),
-                encodings: [],
-                codecs: [],
-                headerExtensions: [],
-                rtcp: {},
-            };
-        }
-        return { ...this._lastParams, encodings: [...this._lastParams.encodings] };
-    }
-
-    async setParameters(params: RTCRtpSendParameters): Promise<void> {
-        if (!this._lastParams) {
-            throw new DOMException(
-                'getParameters must be called before setParameters',
-                'InvalidStateError',
-            );
-        }
-        if (params.transactionId !== this._lastParams.transactionId) {
-            throw new DOMException(
-                'transactionId mismatch',
-                'InvalidModificationError',
-            );
-        }
-        this._lastParams = null;
-    }
-
-    async replaceTrack(track: MediaStreamTrack | null): Promise<void> {
-        if (track === null) {
-            this._teardownPipeline();
-            this._track = null;
-            return;
-        }
-        if (this._track !== null && track.kind !== this._track.kind) {
-            throw new TypeError('Cannot replace track with different kind');
-        }
-        // Same narrowing as `_wirePipeline` — the GStreamer-backing fields on
-        // `MediaStreamTrack` are runtime-attached and typed `any` on the class.
-        const trackGst = track as MediaStreamTrack & {
-            _gstSource: GstNs.Element | null;
-            _gstPipeline: GstNs.Pipeline | null;
-        };
-        if (this._linked && trackGst._gstSource) {
-            // Atomic source swap: old source → new source, keep rest of chain
-            const oldSource = this._elements[0];
-            const newSource = trackGst._gstSource;
-
-            // Move new source from its pipeline
-            const oldPipeline = trackGst._gstPipeline;
-            if (oldPipeline && oldPipeline !== this._pipeline) {
-                newSource.set_state(Gst.State.NULL);
-                oldPipeline.remove(newSource);
-                trackGst._gstPipeline = this._pipeline;
-            }
-
-            // Swap: unlink old, link new
-            oldSource.set_state(Gst.State.NULL);
-            if (this._valve) oldSource.unlink(this._valve);
-            this._pipeline?.remove(oldSource);
-
-            this._pipeline?.add(newSource);
-            if (this._valve) newSource.link(this._valve);
-            newSource.sync_state_with_parent();
-
-            this._elements[0] = newSource;
-        } else if (trackGst._gstSource) {
-            this._wirePipeline(track);
-        }
-
-        // Disconnect old track's enable callback, wire new one
-        if (this._track) this._track._setEnableCallback(null);
-        this._track = track;
-        if (this._linked) {
-            track._setEnableCallback((enabled: boolean) => {
-                if (this._valve) this._valve.drop = !enabled;
-            });
-        }
-    }
-
-    async getStats(): Promise<RTCStatsReport> {
-        if (this._getStatsForTrack && this._track) {
-            return this._getStatsForTrack(this._track);
-        }
-        // Fallback: return empty report if no PC or no track
-        const { RTCStatsReport: Report } = await import('./rtc-stats-report.js');
-        return new Report();
-    }
-
-    setStreams(..._streams: MediaStream[]): void {
-        // Phase 3: no-op — webrtcbin manages msid in SDP automatically
-    }
-
-    static getCapabilities(kind: string): RTCRtpCapabilities | null {
-        return getRtpCapabilities(kind);
-    }
-}