node-av 3.1.3 → 5.0.0

package/dist/api/media-output.js.map

@@ -1 +0,0 @@
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package/dist/api/webrtc.d.ts

@@ -1,664 +0,0 @@
-import { RTCRtpCodecParameters, RtpPacket } from 'werift';
-import { MediaInput } from './media-input.js';
-import type { AVCodecID, AVHWDeviceType } from '../constants/constants.js';
-/**
- * Codec information for WebRTC streaming.
- *
- * Contains RTP codec parameters and FFmpeg codec IDs for video and audio streams.
- * Used for codec negotiation in WebRTC peer connections.
- */
-export interface WebRTCCodecInfo {
-    /**
-     * Video codec configuration.
-     * Combines RTP parameters (mimeType, clockRate, etc.) with FFmpeg codec ID.
-     */
-    video: Partial<RTCRtpCodecParameters> & {
-        codecId: AVCodecID;
-    };
-    /**
-     * Optional audio codec configuration.
-     * Combines RTP parameters (mimeType, clockRate, channels) with FFmpeg codec ID.
-     */
-    audio?: Partial<RTCRtpCodecParameters> & {
-        codecId: AVCodecID;
-    };
-}
-/**
- * Options for configuring WebRTC streaming.
- */
-export interface WebRTCStreamOptions {
-    /**
-     * Callback invoked for each video RTP packet.
-     * Use this to send packets to your WebRTC implementation.
-     *
-     * @param packet - Serialized RTP packet ready for transmission
-     */
-    onVideoPacket?: (packet: RtpPacket) => void;
-    /**
-     * Callback invoked for each audio RTP packet.
-     * Use this to send packets to your WebRTC implementation.
-     *
-     * @param packet - Serialized RTP packet ready for transmission
-     */
-    onAudioPacket?: (packet: RtpPacket) => void;
-    /**
-     * Maximum transmission unit (MTU) size in bytes.
-     * RTP packets will be fragmented to fit within this size.
-     *
-     * @default 1200
-     */
-    mtu?: number;
-    /**
-     * Hardware acceleration configuration.
-     *
-     * - `'auto'` - Automatically detect and use available hardware acceleration
-     * - Object with deviceType - Manually specify hardware acceleration type
-     *
-     * @default { deviceType: AV_HWDEVICE_TYPE_NONE }
-     */
-    hardware?: 'auto' | {
-        deviceType: AVHWDeviceType;
-        device?: string;
-        options?: Record<string, string>;
-    };
-    /**
-     * FFmpeg input options passed directly to the input.
-     *
-     * @default { flags: 'low_delay' }
-     */
-    inputOptions?: Record<string, string | number | boolean | null | undefined>;
-}
-/**
- * High-level WebRTC streaming with automatic codec detection and transcoding.
- *
- * Provides library-agnostic RTP streaming for WebRTC applications.
- * Automatically detects input codecs and transcodes non-WebRTC-compatible formats.
- * Handles video (H.264, H.265, VP8, VP9) and audio (Opus, PCMA, PCMU) codecs.
- * Supports hardware acceleration for video transcoding.
- * Essential component for building WebRTC streaming servers without direct WebRTC library coupling.
- *
- * @example
- * ```typescript
- * import { WebRTCStream } from 'node-av/api';
- *
- * // Create stream with RTP packet callbacks
- * const stream = await WebRTCStream.create('rtsp://camera.local/stream', {
- *   mtu: 1200,
- *   hardware: 'auto',
- *   onVideoPacket: (rtp) => {
- *     // Send RTP packet to WebRTC peer connection
- *     videoTrack.writeRtp(rtp);
- *   },
- *   onAudioPacket: (rtp) => {
- *     audioTrack.writeRtp(rtp);
- *   }
- * });
- *
- * // Get detected codecs for SDP negotiation
- * const codecs = stream.getCodecs();
- * console.log('Video:', codecs.video.mimeType);
- * console.log('Audio:', codecs.audio?.mimeType);
- *
- * // Start streaming
- * await stream.start();
- * ```
- *
- * @example
- * ```typescript
- * // Stream with hardware acceleration
- * import { AV_HWDEVICE_TYPE_CUDA } from 'node-av/constants';
- *
- * const stream = await WebRTCStream.create('video.mp4', {
- *   hardware: {
- *     deviceType: AV_HWDEVICE_TYPE_CUDA,
- *     device: '/dev/nvidia0'
- *   },
- *   onVideoPacket: (rtp) => sendToWebRTC(rtp)
- * });
- *
- * await stream.start();
- * stream.stop();
- * stream.dispose();
- * ```
- *
- * @see {@link WebRTCSession} For complete WebRTC session management with werift
- * @see {@link MediaInput} For input media handling
- * @see {@link HardwareContext} For GPU acceleration
- */
-export declare class WebRTCStream implements Disposable {
-    readonly input: MediaInput;
-    private codecInfo;
-    private options;
-    private videoOutput;
-    private audioOutput;
-    private hardwareContext;
-    private videoDecoder;
-    private videoEncoder;
-    private audioDecoder;
-    private audioFilter;
-    private audioEncoder;
-    private streamActive;
-    private disposed;
-    /**
-     * @param input - Media input source
-     *
-     * @param options - Stream configuration options
-     *
-     * Use {@link create} factory method
-     *
-     * @internal
-     */
-    private constructor();
-    /**
-     * Create a WebRTC stream from a media source.
-     *
-     * Opens the input media, detects video and audio codecs, and prepares
-     * transcoding pipelines for non-WebRTC-compatible formats.
-     * Automatically configures H.264 encoding for unsupported video codecs
-     * and Opus encoding for unsupported audio codecs.
-     *
-     * @param inputUrl - Media source URL (RTSP, file path, HTTP, etc.)
-     *
-     * @param options - Stream configuration options
-     *
-     * @returns Configured WebRTC stream instance
-     *
-     * @throws {Error} If no video stream found in input
-     *
-     * @throws {FFmpegError} If input cannot be opened
-     *
-     * @example
-     * ```typescript
-     * // Stream from RTSP camera
-     * const stream = await WebRTCStream.create('rtsp://camera.local/stream', {
-     *   mtu: 1200,
-     *   onVideoPacket: (rtp) => videoTrack.writeRtp(rtp),
-     *   onAudioPacket: (rtp) => audioTrack.writeRtp(rtp)
-     * });
-     * ```
-     *
-     * @example
-     * ```typescript
-     * // Stream file with auto hardware acceleration
-     * const stream = await WebRTCStream.create('video.mp4', {
-     *   hardware: 'auto'
-     * });
-     * ```
-     */
-    static create(inputUrl: string, options?: WebRTCStreamOptions): Promise<WebRTCStream>;
-    /**
-     * Check if the stream is active.
-     *
-     * @returns True if the stream is active, false otherwise
-     */
-    get isStreamActive(): boolean;
-    /**
-     * Get detected codec information for SDP negotiation.
-     *
-     * Returns RTP codec parameters and FFmpeg codec IDs for video and audio.
-     * Use this information to configure WebRTC peer connections with matching codecs.
-     *
-     * @returns Codec configuration for video and audio streams
-     *
-     * @example
-     * ```typescript
-     * const stream = await WebRTCStream.create('input.mp4');
-     * const codecs = stream.getCodecs();
-     *
-     * console.log('Video codec:', codecs.video.mimeType);
-     * console.log('Audio codec:', codecs.audio?.mimeType);
-     * ```
-     */
-    getCodecs(): WebRTCCodecInfo;
-    /**
-     * Start streaming media to RTP packets.
-     *
-     * Begins the media processing pipeline, reading packets from input,
-     * transcoding if necessary, and invoking RTP packet callbacks.
-     * Automatically handles video and audio streams in parallel.
-     * Flushes all buffers at the end of stream.
-     * This method blocks until streaming completes or {@link stop} is called.
-     *
-     * @returns Promise that resolves when streaming completes
-     *
-     * @throws {FFmpegError} If transcoding or muxing fails
-     *
-     * @example
-     * ```typescript
-     * const stream = await WebRTCStream.create('rtsp://camera.local/stream', {
-     *   onVideoPacket: (rtp) => sendRtp(rtp)
-     * });
-     *
-     * // Start streaming (blocks until complete or stopped)
-     * await stream.start();
-     * ```
-     *
-     * @example
-     * ```typescript
-     * // Non-blocking start with background promise
-     * const stream = await WebRTCStream.create('input.mp4');
-     * const streamPromise = stream.start();
-     *
-     * // Later: stop streaming
-     * stream.stop();
-     * await streamPromise;
-     * ```
-     */
-    start(): Promise<void>;
-    /**
-     * Stop streaming gracefully.
-     *
-     * Signals the streaming loop to exit after the current packet is processed.
-     * Does not immediately close resources - use {@link dispose} for cleanup.
-     * Safe to call multiple times.
-     *
-     * @example
-     * ```typescript
-     * const stream = await WebRTCStream.create('input.mp4');
-     * const streamPromise = stream.start();
-     *
-     * // Stop after 10 seconds
-     * setTimeout(() => stream.stop(), 10000);
-     *
-     * await streamPromise; // Resolves when stopped
-     * stream.dispose();
-     * ```
-     */
-    stop(): void;
-    /**
-     * Clean up all resources and close the stream.
-     *
-     * Stops streaming if active and releases all FFmpeg resources including
-     * decoders, encoders, filters, outputs, and input. Should be called when
-     * done with the stream to prevent memory leaks.
-     * Safe to call multiple times.
-     *
-     * @example
-     * ```typescript
-     * const stream = await WebRTCStream.create('input.mp4');
-     * await stream.start();
-     * stream.dispose();
-     * ```
-     *
-     * @example
-     * ```typescript
-     * // Using automatic cleanup
-     * {
-     *   await using stream = await WebRTCStream.create('input.mp4');
-     *   await stream.start();
-     * } // Automatically disposed
-     * ```
-     */
-    dispose(): void;
-    /**
-     * Check if the given audio codec is compatible with WebRTC.
-     *
-     * @param codecId - The AVCodecID to check
-     *
-     * @returns True if the codec is WebRTC compatible, false otherwise
-     *
-     * @internal
-     */
-    private isAudioCodecSupported;
-    /**
-     * Check if the given video codec is compatible with WebRTC.
-     *
-     * @param codecId - The AVCodecID to check
-     *
-     * @returns True if the codec is WebRTC compatible, false otherwise
-     *
-     * @internal
-     */
-    private isVideoCodecSupported;
-    /**
-     * Get the audio codec configuration for WebRTC.
-     *
-     * @param codecId - The AVCodecID of the audio codec
-     *
-     * @returns An object containing MIME type, clock rate, and channels, or null if unsupported
-     *
-     * @internal
-     */
-    private getAudioCodecConfig;
-    /**
-     * Get the video codec configuration for WebRTC.
-     *
-     * @param codecId - The AVCodecID of the video codec
-     *
-     * @returns An object containing MIME type and clock rate, or null if unsupported
-     *
-     * @internal
-     */
-    private getVideoCodecConfig;
-    /**
-     * Flush video encoder pipeline.
-     *
-     * @param videoStreamIndex - Output video stream index
-     *
-     * @internal
-     */
-    private flushVideo;
-    /**
-     * Flush audio encoder pipeline.
-     *
-     * @param audioStreamIndex - Output audio stream index
-     *
-     * @param hasAudio - Whether audio stream exists
-     *
-     * @internal
-     */
-    private flushAudio;
-    /**
-     * Symbol.dispose implementation for automatic cleanup.
-     *
-     * @internal
-     */
-    [Symbol.dispose](): void;
-}
-/**
- * Options for configuring WebRTC session with werift integration.
- *
- * Extends WebRTCStreamOptions but excludes RTP packet callbacks since
- * they are automatically handled by the session's media tracks.
- */
-export interface WebRTCSessionOptions extends Omit<WebRTCStreamOptions, 'onVideoPacket' | 'onAudioPacket'> {
-    /**
-     * ICE servers for NAT traversal and STUN/TURN configuration.
-     *
-     * @default []
-     *
-     * @example
-     * ```typescript
-     * const session = await WebRTCSession.create('input.mp4', {
-     *   iceServers: [
-     *     { urls: 'stun:stun.l.google.com:19302' },
-     *     { urls: 'turn:turn.example.com:3478' }
-     *   ]
-     * });
-     * ```
-     */
-    iceServers?: {
-        urls: string;
-    }[];
-}
-/**
- * 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 WebRTCStream} but handles all WebRTC protocol details.
- * Integrates with werift library for RTCPeerConnection and media track handling.
- * Ideal for building complete WebRTC streaming applications with minimal code.
- *
- * @example
- * ```typescript
- * import { WebRTCSession } from 'node-av/api';
- *
- * // Create session from media source
- * const session = await WebRTCSession.create('rtsp://camera.local/stream', {
- *   mtu: 1200,
- *   hardware: 'auto',
- *   iceServers: [{ urls: 'stun:stun.l.google.com:19302' }]
- * });
- *
- * // Setup ICE candidate handler
- * session.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 WebRTCSession.create(msg.url, {
- *       hardware: 'auto'
- *     });
- *
- *     session.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 WebRTCStream} For library-agnostic RTP streaming
- * @see {@link MediaInput} For input media handling
- * @see {@link HardwareContext} For GPU acceleration
- */
-export declare class WebRTCSession implements Disposable {
-    private stream;
-    private pc;
-    private videoTrack;
-    private audioTrack;
-    private options;
-    /**
-     * Callback invoked when a new ICE candidate is discovered.
-     * Send this candidate to the remote peer via signaling channel.
-     *
-     * @param candidate - ICE candidate string to send to remote peer
-     *
-     * @example
-     * ```typescript
-     * session.onIceCandidate = (candidate) => {
-     *   ws.send(JSON.stringify({ type: 'candidate', value: candidate }));
-     * };
-     * ```
-     */
-    onIceCandidate: ((candidate: string) => void) | null;
-    /**
-     * @param options - Session configuration options
-     *
-     * Use {@link create} factory method
-     *
-     * @internal
-     */
-    private constructor();
-    /**
-     * 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 inputUrl - Media source URL (RTSP, file path, HTTP, etc.)
-     *
-     * @param options - Session configuration options
-     *
-     * @returns Configured WebRTC session instance
-     *
-     * @throws {Error} If no video stream found in input
-     *
-     * @throws {FFmpegError} If input cannot be opened
-     *
-     * @example
-     * ```typescript
-     * const session = await WebRTCSession.create('rtsp://camera.local/stream', {
-     *   mtu: 1200,
-     *   hardware: 'auto',
-     *   iceServers: [{ urls: 'stun:stun.l.google.com:19302' }]
-     * });
-     * ```
-     *
-     * @example
-     * ```typescript
-     * // Session from file with hardware acceleration
-     * const session = await WebRTCSession.create('video.mp4', {
-     *   hardware: {
-     *     deviceType: AV_HWDEVICE_TYPE_CUDA
-     *   }
-     * });
-     * ```
-     */
-    static create(inputUrl: string, options?: WebRTCSessionOptions): Promise<WebRTCSession>;
-    /**
-     * 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.
-     *
-     * @returns Codec configuration for video and audio streams
-     *
-     * @example
-     * ```typescript
-     * const session = await WebRTCSession.create('input.mp4');
-     * const codecs = session.getCodecs();
-     *
-     * console.log('Video:', codecs.video.mimeType);
-     * console.log('Audio:', codecs.audio?.mimeType);
-     * ```
-     */
-    getCodecs(): WebRTCCodecInfo;
-    /**
-     * 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 {@link onIceCandidate} callback.
-     * Must be called before {@link start}.
-     *
-     * @param offerSdp - SDP offer string from remote WebRTC peer
-     *
-     * @returns SDP answer string to send back to remote peer
-     *
-     * @example
-     * ```typescript
-     * const session = await WebRTCSession.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 });
-     * ```
-     */
-    setOffer(offerSdp: string): Promise<string>;
-    /**
-     * 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.
-     *
-     * @param candidate - ICE candidate string from remote peer
-     *
-     * @throws {Error} If peer connection not initialized (call {@link setOffer} first)
-     *
-     * @example
-     * ```typescript
-     * // In signaling message handler
-     * if (msg.type === 'candidate') {
-     *   session.addIceCandidate(msg.value);
-     * }
-     * ```
-     */
-    addIceCandidate(candidate: string): void;
-    /**
-     * Start streaming media to WebRTC peer connection.
-     *
-     * Begins the media processing pipeline, reading packets from input,
-     * transcoding if necessary, and sending RTP packets to media tracks.
-     * Must call {@link setOffer} before starting.
-     * This method blocks until streaming completes or {@link stop} is called.
-     *
-     * @returns Promise that resolves when streaming completes
-     *
-     * @throws {FFmpegError} If transcoding or muxing fails
-     *
-     * @example
-     * ```typescript
-     * const session = await WebRTCSession.create('input.mp4');
-     * session.onIceCandidate = (c) => sendToRemote(c);
-     *
-     * const answer = await session.setOffer(remoteOffer);
-     * sendToRemote(answer);
-     *
-     * // Start streaming (blocks until complete)
-     * await session.start();
-     * ```
-     *
-     * @example
-     * ```typescript
-     * // Non-blocking start
-     * const session = await WebRTCSession.create('input.mp4');
-     * const streamPromise = session.start();
-     *
-     * // Later: stop streaming
-     * session.stop();
-     * await streamPromise;
-     * ```
-     */
-    start(): Promise<void>;
-    /**
-     * Stop streaming gracefully.
-     *
-     * Signals the streaming loop to exit after the current packet is processed.
-     * Does not immediately close resources - use {@link dispose} for cleanup.
-     * Safe to call multiple times.
-     *
-     * @example
-     * ```typescript
-     * const session = await WebRTCSession.create('input.mp4');
-     * const streamPromise = session.start();
-     *
-     * // Stop after 10 seconds
-     * setTimeout(() => session.stop(), 10000);
-     *
-     * await streamPromise;
-     * session.dispose();
-     * ```
-     */
-    stop(): void;
-    /**
-     * Clean up all resources and close the session.
-     *
-     * Stops streaming if active, releases all FFmpeg resources, closes peer connection,
-     * and cleans up media tracks. Should be called when done with the session to prevent
-     * memory leaks. Safe to call multiple times.
-     *
-     * @example
-     * ```typescript
-     * const session = await WebRTCSession.create('input.mp4');
-     * await session.start();
-     * session.dispose();
-     * ```
-     *
-     * @example
-     * ```typescript
-     * // Using automatic cleanup
-     * {
-     *   await using session = await WebRTCSession.create('input.mp4');
-     *   await session.start();
-     * } // Automatically disposed
-     * ```
-     */
-    dispose(): void;
-    /**
-     * Symbol.dispose implementation for automatic cleanup.
-     *
-     * @internal
-     */
-    [Symbol.dispose](): void;
-}