speechflow 1.7.1 → 2.0.1

package/speechflow-cli/src/speechflow-node-xio-webrtc.ts

@@ -0,0 +1,535 @@
+/*
+**  SpeechFlow - Speech Processing Flow Graph
+**  Copyright (c) 2024-2025 Dr. Ralf S. Engelschall <rse@engelschall.com>
+**  Licensed under GPL 3.0 <https://spdx.org/licenses/GPL-3.0-only>
+*/
+
+/*  standard dependencies  */
+import Stream from "node:stream"
+import http   from "node:http"
+import crypto from "node:crypto"
+
+/*  external dependencies  */
+import { DateTime }    from "luxon"
+import * as arktype    from "arktype"
+import { OpusEncoder } from "@discordjs/opus"
+import {
+    RTCPeerConnection, MediaStreamTrack,
+    RtpPacket, RtpHeader,
+    useAbsSendTime, useSdesMid
+} from "werift"
+
+/*  internal dependencies  */
+import SpeechFlowNode, { SpeechFlowChunk } from "./speechflow-node"
+import * as util                           from "./speechflow-util"
+
+/*  WebRTC peer connection state  */
+interface WebRTCConnection {
+    pc:           RTCPeerConnection
+    track:        MediaStreamTrack | null
+    resourceId:   string
+    subscription: { unSubscribe: () => void } | null
+}
+
+/*  SpeechFlow node for WebRTC networking (WHIP/WHEP)  */
+export default class SpeechFlowNodeXIOWebRTC extends SpeechFlowNode {
+    /*  declare official node name  */
+    public static name = "xio-webrtc"
+
+    /*  internal state  */
+    private peerConnections                                       = new Map<string, WebRTCConnection>()
+    private httpServer:  http.Server                       | null = null
+    private chunkQueue:  util.SingleQueue<SpeechFlowChunk> | null = null
+    private opusEncoder: OpusEncoder                       | null = null
+    private opusDecoder: OpusEncoder                       | null = null
+    private pcmBuffer                                             = Buffer.alloc(0)
+    private rtpSequence                                           = 0
+    private rtpTimestamp                                          = 0
+    private rtpSSRC                                               = 0
+    private maxConnections                                        = 10
+
+    /*  Opus codec configuration: 48kHz, mono, 16-bit  */
+    private readonly OPUS_SAMPLE_RATE = 48000
+    private readonly OPUS_CHANNELS    = 1
+    private readonly OPUS_BIT_DEPTH   = 16
+    private readonly OPUS_FRAME_SIZE  = 960              /*  20ms at 48kHz = 960 samples  */
+    private readonly OPUS_FRAME_BYTES = 960 * 2          /*  16-bit = 2 bytes per sample  */
+
+    /*  construct node  */
+    constructor (id: string, cfg: { [ id: string ]: any }, opts: { [ id: string ]: any }, args: any[]) {
+        super(id, cfg, opts, args)
+
+        /*  declare node configuration parameters  */
+        this.configure({
+            listen:     { type: "string", pos: 0, val: "8085",    match: /^(?:\d+|.+?:\d+)$/ },
+            path:       { type: "string", pos: 1, val: "/webrtc", match: /^\/.+$/ },
+            mode:       { type: "string", pos: 2, val: "r",       match: /^(?:r|w)$/ },
+            iceServers: { type: "string", pos: 3, val: "",        match: /^.*$/ }
+        })
+
+        /*  declare node input/output format  */
+        if (this.params.mode === "r") {
+            this.input  = "none"
+            this.output = "audio"
+        }
+        else if (this.params.mode === "w") {
+            this.input  = "audio"
+            this.output = "none"
+        }
+    }
+
+    /*  parse address:port string  */
+    private parseAddress (addr: string, defaultPort: number): { host: string, port: number } {
+        if (addr.match(/^\d+$/))
+            return { host: "0.0.0.0", port: Number.parseInt(addr, 10) }
+        const m = addr.match(/^(.+?):(\d+)$/)
+        if (m === null)
+            return { host: addr, port: defaultPort }
+        return { host: m[1], port: Number.parseInt(m[2], 10) }
+    }
+
+    /*  read HTTP request body  */
+    private readRequestBody (req: http.IncomingMessage): Promise<string> {
+        return new Promise((resolve, reject) => {
+            const chunks: Buffer[] = []
+            const maxBodySize = 1024 * 1024  /*  1 MB limit for SDP  */
+            let totalSize = 0
+            const onData = (chunk: Buffer) => {
+                totalSize += chunk.length
+                if (totalSize > maxBodySize) {
+                    req.removeListener("data",  onData)
+                    req.removeListener("end",   onEnd)
+                    req.removeListener("error", onError)
+                    req.destroy()
+                    reject(new Error("request body too large"))
+                    return
+                }
+                chunks.push(chunk)
+            }
+            const onEnd = () =>
+                resolve(Buffer.concat(chunks).toString("utf8"))
+            const onError = (err: Error) =>
+                reject(err)
+            req.on("data", onData)
+            req.on("end", onEnd)
+            req.on("error", onError)
+        })
+    }
+
+    /*  decode Opus packet to PCM and enqueue as SpeechFlowChunk  */
+    private decodeOpusToChunk (opusPacket: Buffer) {
+        if (this.opusDecoder === null || this.chunkQueue === null)
+            return
+        if (this.params.mode === "w")
+            return
+        try {
+            /*  decode Opus to PCM (16-bit signed, little-endian, 48kHz)  */
+            const pcmBuffer = this.opusDecoder.decode(opusPacket)
+
+            /*  create chunk with timing information (use Opus codec rates, not config)  */
+            const now = DateTime.now()
+            const start = now.diff(this.timeZero)
+            const duration = util.audioBufferDuration(pcmBuffer,
+                this.OPUS_SAMPLE_RATE, this.OPUS_BIT_DEPTH, this.OPUS_CHANNELS)
+            const end = start.plus(duration * 1000)
+            const chunk = new SpeechFlowChunk(start, end, "final", "audio", pcmBuffer)
+            this.chunkQueue.write(chunk)
+        }
+        catch (err: unknown) {
+            this.log("warning", `Opus decode error: ${util.ensureError(err).message}`)
+        }
+    }
+
+    /*  buffer PCM and encode to Opus frames, send to all viewers  */
+    private bufferAndEncode (chunk: SpeechFlowChunk) {
+        if (this.opusEncoder === null)
+            return
+        const pcm = chunk.payload as Buffer
+        this.pcmBuffer = Buffer.concat([ this.pcmBuffer, pcm ])
+
+        /*  prevent unbounded buffer growth  */
+        const maxBufferSize = this.OPUS_FRAME_BYTES * 10
+        if (this.pcmBuffer.length > maxBufferSize) {
+            this.log("warning", `PCM buffer overflow (${this.pcmBuffer.length} bytes), discarding excess`)
+            this.pcmBuffer = this.pcmBuffer.subarray(this.pcmBuffer.length - maxBufferSize)
+        }
+
+        /*  process full Opus frames from buffer  */
+        while (this.pcmBuffer.length >= this.OPUS_FRAME_BYTES) {
+            const frame = this.pcmBuffer.subarray(0, this.OPUS_FRAME_BYTES)
+            this.pcmBuffer = this.pcmBuffer.subarray(this.OPUS_FRAME_BYTES)
+            try {
+                /*  encode PCM to Opus  */
+                const opusPacket = this.opusEncoder.encode(frame)
+                this.sendOpusToAllViewers(opusPacket)
+            }
+            catch (err: unknown) {
+                this.log("warning", `Opus encode error: ${util.ensureError(err).message}`)
+            }
+        }
+    }
+
+    /*  send Opus packet to all connected WHEP viewers  */
+    private sendOpusToAllViewers (opusPacket: Buffer) {
+        /*  build RTP header  */
+        const rtpHeader = new RtpHeader({
+            version:        2,
+            padding:        false,
+            paddingSize:    0,
+            extension:      false,
+            marker:         true,
+            payloadType:    111, /*  Opus payload type  */
+            sequenceNumber: this.rtpSequence++ & 0xFFFF,
+            timestamp:      this.rtpTimestamp,
+            ssrc:           this.rtpSSRC,
+            csrc:           [],
+            extensions:     []
+        })
+
+        /*  build RTP packet  */
+        const rtpPacket = new RtpPacket(rtpHeader, opusPacket)
+
+        /*  advance timestamp by frame duration  */
+        this.rtpTimestamp = (this.rtpTimestamp + this.OPUS_FRAME_SIZE) >>> 0
+
+        /*  send to all connected viewers (snapshot to avoid concurrent modification)  */
+        const connections = Array.from(this.peerConnections.values())
+        for (const conn of connections) {
+            if (conn.track !== null) {
+                try {
+                    conn.track.writeRtp(rtpPacket)
+                }
+                catch (err: unknown) {
+                    this.log("warning", `failed to send RTP to WebRTC peer: ${util.ensureError(err).message}`)
+                }
+            }
+        }
+    }
+
+    /*  parse ICE servers configuration  */
+    private parseIceServers (): { urls: string }[] {
+        if (this.params.iceServers === "")
+            return []
+        let servers: { urls: string }[] = []
+        try {
+            servers = util.importObject("WebRTC ICE servers",
+                this.params.iceServers,
+                arktype.type({ urls: "string" }).array())
+        }
+        catch (err: unknown) {
+            this.log("warning", `invalid iceServers JSON: ${util.ensureError(err).message}`)
+            servers = []
+        }
+        return servers
+    }
+
+    /*  create a new RTCPeerConnection with standard configuration  */
+    private createPeerConnection (resourceId: string): { pc: RTCPeerConnection, subscription: { unSubscribe: () => void } } {
+        const pc = new RTCPeerConnection({
+            iceServers: this.parseIceServers(),
+            headerExtensions: {
+                audio: [ useSdesMid(), useAbsSendTime() ]
+            }
+        })
+        const subscription = pc.connectionStateChange.subscribe((state: string) => {
+            this.log("info", `WebRTC connection ${resourceId}: ${state}`)
+            if (state === "failed" || state === "closed" || state === "disconnected")
+                setImmediate(() => {
+                    if (this.peerConnections.has(resourceId))
+                        this.cleanupConnection(resourceId)
+                })
+        })
+        return { pc, subscription }
+    }
+
+    /*  safely close a peer connection  */
+    private closePeerConnection (pc: RTCPeerConnection) {
+        util.shield(() => { pc.close() })
+    }
+
+    /*  perform SDP negotiation and establish connection  */
+    private async performSDPNegotiation (
+        res:      http.ServerResponse,
+        offer:    string,
+        protocol: "WHIP" | "WHEP",
+        setupFn:  (pc: RTCPeerConnection, resourceId: string) => MediaStreamTrack | null
+    ) {
+        /*  enforce connection limit  */
+        if (this.peerConnections.size >= this.maxConnections) {
+            res.writeHead(503, { "Content-Type": "text/plain" })
+            res.end("Service Unavailable: Maximum connections reached")
+            return
+        }
+
+        /*  create peer connection  */
+        const resourceId = crypto.randomUUID()
+        const { pc, subscription } = this.createPeerConnection(resourceId)
+
+        /*  protocol-specific setup  */
+        const track = setupFn(pc, resourceId)
+
+        /*  complete SDP offer/answer exchange and establish connection  */
+        try {
+            /*  set remote description (offer from client)  */
+            await pc.setRemoteDescription({ type: "offer", sdp: offer })
+
+            /*  create and set local description (answer)  */
+            const answer = await pc.createAnswer()
+            await pc.setLocalDescription(answer)
+
+            /*  store connection  */
+            this.peerConnections.set(resourceId, { pc, track, resourceId, subscription })
+
+            /*  return SDP answer  */
+            if (pc.localDescription === null || pc.localDescription === undefined)
+                throw new Error("local description is missing")
+            res.writeHead(201, {
+                "Content-Type": "application/sdp",
+                "Location":     `${this.params.path}/${resourceId}`
+            })
+            res.end(pc.localDescription.sdp)
+            this.log("info", `${protocol} connection established: ${resourceId}`)
+        }
+        catch (err: unknown) {
+            util.shield(() => { subscription.unSubscribe() })
+            this.closePeerConnection(pc)
+            this.log("error", `${protocol} negotiation failed: ${util.ensureError(err).message}`)
+            res.writeHead(500, { "Content-Type": "text/plain" })
+            res.end("Internal Server Error")
+        }
+    }
+
+    /*  handle WHIP POST (receiving audio from publisher)  */
+    private async handleWHIP (res: http.ServerResponse, offer: string) {
+        await this.performSDPNegotiation(res, offer, "WHIP", (pc, _resourceId) => {
+            /*  handle incoming audio track  */
+            pc.ontrack = (event: { track: MediaStreamTrack }) => {
+                const track = event.track
+                if (track.kind === "audio") {
+                    this.log("info", `WebRTC audio track received from publisher`)
+
+                    /*  subscribe to incoming RTP packets  */
+                    track.onReceiveRtp.subscribe((rtpPacket: RtpPacket) => {
+                        this.decodeOpusToChunk(rtpPacket.payload)
+                    })
+                }
+            }
+            return null
+        })
+    }
+
+    /*  handle WHEP POST (sending audio to viewer)  */
+    private async handleWHEP (res: http.ServerResponse, offer: string) {
+        await this.performSDPNegotiation(res, offer, "WHEP", (pc, _resourceId) => {
+            /*  create outbound audio track  */
+            const outboundTrack = new MediaStreamTrack({ kind: "audio" })
+            pc.addTrack(outboundTrack)
+            return outboundTrack
+        })
+    }
+
+    /*  handle DELETE (connection teardown)  */
+    private handleDELETE (res: http.ServerResponse, resourceId: string) {
+        if (this.peerConnections.has(resourceId)) {
+            this.cleanupConnection(resourceId)
+            res.writeHead(200)
+            res.end()
+            this.log("info", `WebRTC connection terminated: ${resourceId}`)
+        }
+        else {
+            res.writeHead(404, { "Content-Type": "text/plain" })
+            res.end("Not Found")
+        }
+    }
+
+    /*  cleanup a peer connection  */
+    private cleanupConnection (resourceId: string) {
+        const conn = this.peerConnections.get(resourceId)
+        if (conn === undefined)
+            return
+        this.peerConnections.delete(resourceId)
+        if (conn.subscription !== null)
+            util.shield(() => { conn.subscription?.unSubscribe() })
+        if (conn.track !== null)
+            util.shield(() => { conn.track?.stop() })
+        this.closePeerConnection(conn.pc)
+    }
+
+    /*  open node  */
+    async open () {
+        /*  setup Opus codec  */
+        this.opusEncoder = new OpusEncoder(this.OPUS_SAMPLE_RATE, this.OPUS_CHANNELS)
+        this.opusDecoder = new OpusEncoder(this.OPUS_SAMPLE_RATE, this.OPUS_CHANNELS)
+
+        /*  initialize RTP state  */
+        this.rtpSequence  = Math.floor(Math.random() * 0x10000)
+        this.rtpTimestamp = Math.floor(Math.random() * 0x100000000) >>> 0
+        this.rtpSSRC      = Math.floor(Math.random() * 0x100000000) >>> 0
+
+        /*  setup chunk queue for incoming audio  */
+        this.chunkQueue = new util.SingleQueue<SpeechFlowChunk>()
+
+        /*  parse listen address  */
+        const listen = this.parseAddress(this.params.listen, 8085)
+
+        /*  setup HTTP server for WHIP/WHEP signaling  */
+        const self = this
+        this.httpServer = http.createServer(async (req, res) => {
+            /*  determine URL  */
+            if (req.url === undefined) {
+                res.writeHead(400, { "Content-Type": "text/plain" })
+                res.end("Bad Request")
+                return
+            }
+            const host = req.headers.host?.replace(/[^a-zA-Z0-9:.\-_]/g, "") ?? "localhost"
+            const url = new URL(req.url, `http://${host}`)
+            const pathMatch = url.pathname === self.params.path
+            const resourceMatch = url.pathname.startsWith(self.params.path + "/")
+
+            /*  CORS headers for browser clients  */
+            res.setHeader("Access-Control-Allow-Origin",   "*")
+            res.setHeader("Access-Control-Allow-Methods",  "POST, DELETE, OPTIONS")
+            res.setHeader("Access-Control-Allow-Headers",  "Content-Type")
+            res.setHeader("Access-Control-Expose-Headers", "Location")
+
+            /*  handle CORS preflight  */
+            if (req.method === "OPTIONS") {
+                res.writeHead(204)
+                res.end()
+                return
+            }
+
+            /*  handle requests...  */
+            if (req.method === "POST" && pathMatch) {
+                /*  handle WHIP/WHEP POST  */
+                const body = await self.readRequestBody(req)
+
+                /*  sanity check content type  */
+                const contentType = req.headers["content-type"]
+                if (contentType !== "application/sdp") {
+                    res.writeHead(415, { "Content-Type": "text/plain" })
+                    res.end("Unsupported Media Type")
+                    return
+                }
+
+                /*  determine if WHIP (receiving) or WHEP (sending) based on SDP content  */
+                const hasSendonly  = /\ba=sendonly\b/m.test(body)
+                const hasSendrecv  = /\ba=sendrecv\b/m.test(body)
+                const hasRecvonly  = /\ba=recvonly\b/m.test(body)
+                const isPublisher  = hasSendonly || hasSendrecv
+                const isViewer     = hasRecvonly
+
+                /*  handle protocol based on mode  */
+                if (self.params.mode === "r" && isPublisher)
+                    /*  in read mode, accept WHIP publishers  */
+                    await self.handleWHIP(res, body)
+                else if (self.params.mode === "w" && isViewer)
+                    /*  in write mode, accept WHEP viewers  */
+                    await self.handleWHEP(res, body)
+                else {
+                    res.writeHead(403, { "Content-Type": "text/plain" })
+                    res.end("Forbidden")
+                }
+            }
+            else if (req.method === "DELETE" && resourceMatch) {
+                /*  handle DELETE for connection teardown  */
+                const resourceId = url.pathname.substring(self.params.path.length + 1)
+                self.handleDELETE(res, resourceId)
+            }
+            else {
+                /*  handle unknown requests  */
+                res.writeHead(404, { "Content-Type": "text/plain" })
+                res.end("Not Found")
+            }
+        })
+
+        /*  start HTTP server  */
+        await new Promise<void>((resolve) => {
+            this.httpServer!.listen(listen.port, listen.host, () => {
+                const mode = this.params.mode === "r" ? "WHIP" : "WHEP"
+                this.log("info", `WebRTC ${mode} server listening on http://${listen.host}:${listen.port}${this.params.path}`)
+                resolve()
+            })
+        })
+
+        /*  create duplex stream  */
+        const reads = new util.PromiseSet<void>()
+        this.stream = new Stream.Duplex({
+            writableObjectMode: true,
+            readableObjectMode: true,
+            decodeStrings:      false,
+            highWaterMark:      1,
+            write (chunk: SpeechFlowChunk, encoding, callback) {
+                if (self.params.mode === "r") {
+                    callback(new Error("write operation on read mode node"))
+                    return
+                }
+                if (chunk.type !== "audio") {
+                    callback(new Error("WebRTC node only supports audio type"))
+                    return
+                }
+                if (self.peerConnections.size === 0) {
+                    /*  silently drop if no viewers connected  */
+                    callback()
+                    return
+                }
+                self.bufferAndEncode(chunk)
+                callback()
+            },
+            async final (callback) {
+                await reads.awaitAll()
+                callback()
+            },
+            read (size: number) {
+                if (self.params.mode === "w") {
+                    self.log("error", "read operation on write mode node")
+                    this.push(null)
+                    return
+                }
+                reads.add(self.chunkQueue!.read().then((chunk) => {
+                    this.push(chunk, "binary")
+                }).catch((err: Error) => {
+                    self.log("warning", `read on chunk queue operation failed: ${err}`)
+                    this.push(null)
+                }))
+            }
+        })
+    }
+
+    /*  close node  */
+    async close () {
+        /*  close all peer connections  */
+        for (const resourceId of Array.from(this.peerConnections.keys()))
+            this.cleanupConnection(resourceId)
+
+        /*  close HTTP server  */
+        if (this.httpServer !== null) {
+            await new Promise<void>((resolve, reject) => {
+                this.httpServer!.close((err) => {
+                    if (err) reject(err)
+                    else     resolve()
+                })
+            }).catch((err: Error) => {
+                this.log("warning", `failed to close HTTP server: ${err.message}`)
+            })
+            this.httpServer = null
+        }
+
+        /*  drain and clear chunk queue  */
+        if (this.chunkQueue !== null) {
+            this.chunkQueue.drain()
+            this.chunkQueue = null
+        }
+
+        /*  cleanup codec instances  */
+        this.opusEncoder = null
+        this.opusDecoder = null
+        this.pcmBuffer   = Buffer.alloc(0)
+
+        /*  shutdown stream  */
+        if (this.stream !== null) {
+            await util.destroyStream(this.stream)
+            this.stream = null
+        }
+    }
+}

package/speechflow-cli/src/speechflow-node-xio-websocket.ts

@@ -15,7 +15,7 @@ import ReconnWebSocket, { ErrorEvent } from "@opensumi/reconnecting-websocket"
 import SpeechFlowNode, { SpeechFlowChunk } from "./speechflow-node"
 import * as util                           from "./speechflow-util"
 
-/*  SpeechFlow node for Websocket networking  */
+/*  SpeechFlow node for WebSocket networking  */
 export default class SpeechFlowNodeXIOWebSocket extends SpeechFlowNode {
     /*  declare official node name  */
     public static name = "xio-websocket"
@@ -38,9 +38,9 @@ export default class SpeechFlowNodeXIOWebSocket extends SpeechFlowNode {
 
         /*  sanity check parameters  */
         if (this.params.listen !== "" && this.params.connect !== "")
-            throw new Error("Websocket node cannot listen and connect at the same time")
+            throw new Error("WebSocket node cannot listen and connect at the same time")
         else if (this.params.listen === "" && this.params.connect === "")
-            throw new Error("Websocket node requires either listen or connect mode")
+            throw new Error("WebSocket node requires either listen or connect mode")
 
         /*  declare node input/output format  */
         if (this.params.mode === "rw") {
@@ -121,7 +121,7 @@ export default class SpeechFlowNodeXIOWebSocket extends SpeechFlowNode {
                     else if (chunk.type !== self.params.type)
                         callback(new Error(`written chunk is not of ${self.params.type} type`))
                     else if (websockets.size === 0)
-                        callback(new Error("still no Websocket connections available"))
+                        callback(new Error("still no WebSocket connections available"))
                     else {
                         const data = util.streamChunkEncode(chunk)
                         const results: Promise<void>[] = []
@@ -168,10 +168,10 @@ export default class SpeechFlowNodeXIOWebSocket extends SpeechFlowNode {
                 connectionTimeout:           4000,
                 minUptime:                   5000
             })
-            this.client.addEventListener("open", (ev) => {
+            this.client.addEventListener("open", (_ev) => {
                 this.log("info", `connection opened to URL ${this.params.connect}`)
             })
-            this.client.addEventListener("close", (ev) => {
+            this.client.addEventListener("close", (_ev) => {
                 this.log("info", `connection closed to URL ${this.params.connect}`)
             })
             this.client.addEventListener("error", (ev: ErrorEvent) => {
@@ -208,7 +208,7 @@ export default class SpeechFlowNodeXIOWebSocket extends SpeechFlowNode {
                     else if (chunk.type !== self.params.type)
                         callback(new Error(`written chunk is not of ${self.params.type} type`))
                     else if (!self.client!.OPEN)
-                        callback(new Error("still no Websocket connection available"))
+                        callback(new Error("still no WebSocket connection available"))
                     else {
                         const data = util.streamChunkEncode(chunk)
                         self.client!.send(data)
@@ -234,7 +234,7 @@ export default class SpeechFlowNodeXIOWebSocket extends SpeechFlowNode {
 
     /*  close node  */
     async close () {
-        /*  close Websocket server  */
+        /*  close WebSocket server  */
         if (this.server !== null) {
             await new Promise<void>((resolve, reject) => {
                 this.server!.close((error) => {
@@ -245,7 +245,7 @@ export default class SpeechFlowNodeXIOWebSocket extends SpeechFlowNode {
             this.server = null
         }
 
-        /*  close Websocket client  */
+        /*  close WebSocket client  */
         if (this.client !== null) {
             this.client.close()
             this.client = null

package/speechflow-cli/src/speechflow-util-audio.ts

@@ -10,6 +10,9 @@ import path from "node:path"
 /*  external dependencies  */
 import { AudioContext, AudioWorkletNode } from "node-web-audio-api"
 
+/*  internal dependencies  */
+import { shield } from "./speechflow-util-error"
+
 /*  calculate duration of an audio buffer  */
 export function audioBufferDuration (
     buffer: Buffer,
@@ -159,6 +162,9 @@ export function updateEnvelopeForChannel (
         else
             currentEnv = alphaR * currentEnv + (1 - alphaR) * det
     }
+
+    /*  store updated envelope value back  */
+    env[chan] = currentEnv
     return Math.sqrt(Math.max(currentEnv, 1e-12))
 }
 
@@ -170,6 +176,7 @@ export function dB2lin (db: number): number {
     return Math.pow(10, db / 20)
 }
 
+/*  Web Audio API wrapper class  */
 export class WebAudio {
     /*  internal state  */
     public audioContext: AudioContext
@@ -278,18 +285,16 @@ export class WebAudio {
         })
     }
 
+    /*  destroy object  */
     public async destroy (): Promise<void> {
         /*  reject all pending promises  */
-        try {
+        shield(() => {
             this.pendingPromises.forEach(({ reject, timeout }) => {
                 clearTimeout(timeout)
                 reject(new Error("WebAudio destroyed"))
             })
             this.pendingPromises.clear()
-        }
-        catch (_err) {
-            /*  ignored -- cleanup during shutdown  */
-        }
+        })
 
         /*  disconnect nodes  */
         if (this.sourceNode !== null) {

package/speechflow-cli/src/speechflow-util-error.ts

@@ -188,3 +188,12 @@ export function runner<T> (
             return run(() => action(...args), oncatch, onfinally)
     }
 }
+
+/*  shield cleanup operation, ignoring errors  */
+export function shield<T extends (void | Promise<void>)> (op: () => T) {
+    return run(
+        "shielded operation",
+        () => { op() },
+        (_err) => { /* ignore error */ }
+    )
+}