homebridge-nest-accfactory 0.4.3 → 0.4.4-beta.2

package/dist/HomeKitDevice.js

@@ -112,7 +112,7 @@ export default class HomeKitDevice extends EventEmitter {
   static EVEHOME = undefined; // HomeKitHistory object
   static LOGGER = undefined; // Logging object
   static TYPE = 'base'; // String naming type of device
-  static VERSION = '2026.05.26'; // Code version
+  static VERSION = '2026.08.18'; // Code version
 
   // Backend types
   static HOMEBRIDGE = 'homebridge';
@@ -313,7 +313,7 @@ export default class HomeKitDevice extends EventEmitter {
 
     // If using HAP-NodeJS library, publish accessory on local network
     if (this.accessory !== undefined && this.backend === HomeKitDevice.HAP_NODEJS) {
-      this.accessory.publish({
+      await this.accessory.publish({
         username: this.accessory.username,
         pincode: this.accessory.pincode,
         category: this.accessory.category,
@@ -623,7 +623,15 @@ export default class HomeKitDevice extends EventEmitter {
           }
 
           if (this.accessory !== undefined && this.#platform === undefined) {
-            this.accessory.unpublish();
+            try {
+              await this.accessory.unpublish();
+            } catch (error) {
+              this?.log?.warn?.(
+                'Failed to unpublish standalone accessory "%s": %s',
+                this.deviceData.description,
+                String(error?.stack || error),
+              );
+            }
           }
 
           this.deviceData = {};
@@ -727,18 +735,20 @@ export default class HomeKitDevice extends EventEmitter {
           }
 
           if (skipHistory === false) {
-            this.historyService.addHistory(
+            let historyResult = await this.historyService.addHistory(
               target,
               entry,
               Number.isFinite(Number(options?.timegap)) === true ? Number(options.timegap) : undefined,
             );
+
+            if (historyResult !== false) {
+              // Notify hooks only after the entry was accepted by the history service.
+              await callLifecycleHook('onHistory', target, entry, options);
+              await callLifecycleHook(['handler for HISTORY', handler], target, entry, options);
+            }
           }
         }
 
-        // Call the onHistory method and after any static handler registered via HomeKitDevice.message(uuid, type, handler)
-        await callLifecycleHook('onHistory', target, entry, options);
-        await callLifecycleHook(['handler for HISTORY', handler], target, entry, options);
-
         handled = true;
       }
 
@@ -893,8 +903,14 @@ export default class HomeKitDevice extends EventEmitter {
 
     // delay + interval => fire once after delay, then repeat
     entry.timeout = setTimeout(() => {
-      fire();
       entry.timeout = undefined;
+      fire();
+
+      // A synchronous first callback may remove or replace this timer. Do not
+      // create an interval that is no longer owned by the timer registry.
+      if (entry.cancelled === true || this.#timers.get(timerHandle) !== entry) {
+        return;
+      }
 
       entry.intervalHandle = setInterval(() => {
         fire();
@@ -1132,10 +1148,9 @@ export default class HomeKitDevice extends EventEmitter {
 
     return typeof name === 'string'
       ? name
-          .replace(/[^\p{L}\p{N}\p{Zs}\u2019'&!._:;()\/,-]/gu, '')
-          .replace(/^[^\p{L}\p{N}]+/gu, '')
-          .replace(/[^\p{L}\p{N}]+$/gu, '')
-          .trim()
+        .replace(/[^\p{L}\p{N}\p{Zs}\u2019'&!._:;()\/,-]/gu, '')
+        .replace(/^[^\p{L}\p{N}]+|[^\p{L}\p{N}]+$/gu, '')
+        .trim() || 'Unknown Device'
       : name;
   }
 

package/dist/ffmpeg.js

@@ -29,7 +29,7 @@
 // - Logging and debugging are handled by the calling module
 // - Binary validation and capability checks are performed during initialisation
 //
-// Code version 2026.05.06
+// Code version 2026.09.05
 // Mark Hulskamp
 'use strict';
 
@@ -298,6 +298,30 @@ export default class FFmpeg {
 
       // Linux: prioritise nvenc > qsv > vaapi > v4l2m2m, only if required devices exist
       else if (platform === 'linux') {
+        // Pi codec nodes are dynamically numbered; identify the encoder rather than a camera or decoder.
+        try {
+          if (fs.readFileSync('/sys/firmware/devicetree/base/model', 'utf8').startsWith('Raspberry Pi') === true) {
+            hasVideo = false;
+            hasVideo = fs.readdirSync('/sys/class/video4linux').some((device) => {
+              if (/^video\d+$/.test(device) === false) {
+                return false;
+              }
+              try {
+                if (fs.readFileSync('/sys/class/video4linux/' + device + '/name', 'utf8').trim() !== 'bcm2835-codec-encode') {
+                  return false;
+                }
+                fs.accessSync('/dev/' + device, fs.constants.R_OK | fs.constants.W_OK);
+                return true;
+              } catch {
+                // A missing or inaccessible node is not available to this process.
+                return false;
+              }
+            });
+          }
+        } catch {
+          // Missing host metadata keeps the generic check; missing Pi codec metadata leaves acceleration disabled.
+        }
+
         let linuxEncoders = [
           { key: 'h264_nvenc', device: hasDri },
           { key: 'h264_qsv', device: hasIntelQSV },

package/dist/streamer.js

@@ -67,7 +67,7 @@
 // - MediaTimeline owns retained buffer/index stats
 // - Streamer owns per-output write/drop/playout stats
 //
-// Code version 2026.05.18
+// Code version 2026.08.20
 // Mark Hulskamp
 'use strict';
 
@@ -119,6 +119,9 @@ const OUTPUT_PLAYOUT_POLICY = {
     requireKeyFrameStart: true,
     allowAudioBeforeKeyFrame: false,
     playoutDelayMs: 200,
+    minPlayoutDelayMs: 200,
+    maxPlayoutDelayMs: 200,
+    playoutAdjustStepMs: 20,
     maxLagBehindLiveMs: 1200,
     dueTolerance: 12,
     dueSlack: 20,
@@ -893,30 +896,33 @@ export default class Streamer {
       // - preserves continuity while draining retained media
       policy: { ...(OUTPUT_PLAYOUT_POLICY[type] ?? OUTPUT_PLAYOUT_POLICY.live) },
 
-      // Debug / instrumentation stats (used for tuning pacing behaviour).
-      stats: {
-        startedAt: Date.now(),
-        firstWriteAt: undefined,
-        firstVideoWriteAt: undefined,
-        firstAudioWriteAt: undefined,
-        writes: { total: 0, video: 0, audio: 0 },
-        drops: { videoBeforeKeyframe: 0, audioBeforeKeyframe: 0, bufferTrimmed: 0 },
-        diagnostics: {
-          lastVideoWriteAt: undefined,
-          lastAudioWriteAt: undefined,
-          maxVideoWriteGapMs: 0,
-          maxAudioWriteGapMs: 0,
-          audioWriteGapsOver100Ms: 0,
-          audioWriteGapsOver250Ms: 0,
-          audioWriteGapsOver500Ms: 0,
-          audioBlockedBehindVideo: 0,
-          videoBackpressureStops: 0,
-          maxAudioQueuedBytes: 0,
-          maxBlockedAudioLagMs: 0,
-          lastAudioDiagnosticWriteAt: undefined,
-          lastAudioDiagnosticLogTime: undefined,
-        },
-      },
+      // Debug instrumentation is not allocated for normal streaming.
+      stats:
+        this.supportDump === true
+          ? {
+              startedAt: Date.now(),
+              firstWriteAt: undefined,
+              firstVideoWriteAt: undefined,
+              firstAudioWriteAt: undefined,
+              writes: { total: 0, video: 0, audio: 0 },
+              drops: { videoBeforeKeyframe: 0, audioBeforeKeyframe: 0, bufferTrimmed: 0 },
+              diagnostics: {
+                lastVideoWriteAt: undefined,
+                lastAudioWriteAt: undefined,
+                maxVideoWriteGapMs: 0,
+                maxAudioWriteGapMs: 0,
+                audioWriteGapsOver100Ms: 0,
+                audioWriteGapsOver250Ms: 0,
+                audioWriteGapsOver500Ms: 0,
+                audioBlockedBehindVideo: 0,
+                videoBackpressureStops: 0,
+                maxAudioQueuedBytes: 0,
+                maxBlockedAudioLagMs: 0,
+                lastAudioDiagnosticWriteAt: undefined,
+                lastAudioDiagnosticLogTime: undefined,
+              },
+            }
+          : undefined,
     };
 
     // Attach talkback handling for live streams.
@@ -1210,6 +1216,10 @@ export default class Streamer {
   }
 
   #statsWrite(output, type, dateNow) {
+    if (this.supportDump !== true) {
+      return;
+    }
+
     let outputStats = output?.stats;
     let outputWrites = outputStats?.writes;
     let diagnostics = outputStats?.diagnostics;
@@ -1221,7 +1231,6 @@ export default class Streamer {
         gapMs = dateNow - diagnostics.lastAudioDiagnosticWriteAt;
 
         if (
-          this.supportDump === true &&
           gapMs >= STREAMER_AUDIO_GAP_LOG_MS &&
           (typeof diagnostics.lastAudioDiagnosticLogTime !== 'number' ||
             dateNow - diagnostics.lastAudioDiagnosticLogTime >= STREAMER_AUDIO_GAP_LOG_INTERVAL_MS)
@@ -1247,10 +1256,6 @@ export default class Streamer {
       diagnostics.lastAudioDiagnosticWriteAt = dateNow;
     }
 
-    if (this.supportDump !== true) {
-      return;
-    }
-
     if (typeof outputStats !== 'object' || outputStats === null) {
       return;
     }
@@ -1410,7 +1415,6 @@ export default class Streamer {
     let includeAudio = false;
     let isH264Output = false;
     let isLiveOutput = false;
-    let isRecordOutput = false;
 
     let lastSPS = undefined;
     let lastPPS = undefined;
@@ -1477,16 +1481,9 @@ export default class Streamer {
     includeAudio = output.includeAudio === true && outputAudio !== null;
     isH264Output = this.codecs?.video === StreamTransport.CODEC_TYPE.H264;
     isLiveOutput = output.type === Streamer.STREAM_TYPE.LIVE;
-    isRecordOutput = output.type === Streamer.STREAM_TYPE.RECORD;
-
-    // Resolve output policy.
-    // Allows per-output tuning (live vs record) while falling back to sensible defaults.
-    policy =
-      typeof output?.policy === 'object' && output.policy !== null
-        ? output.policy
-        : (OUTPUT_PLAYOUT_POLICY[output?.type] ?? OUTPUT_PLAYOUT_POLICY.live);
 
-    // Extract policy values with safety checks.
+    // Output policy is normalised once when the output is created.
+    policy = output.policy;
     //
     // Even when live policy does not normally require keyframe startup, force
     // decoder-safe startup whenever this output has not yet seen a real keyframe.
@@ -1497,50 +1494,18 @@ export default class Streamer {
     allowAudioBeforeKeyFrame = policy.allowAudioBeforeKeyFrame === true;
 
     playoutDelayMs = Number.isFinite(output.playoutDelayMs) === true ? output.playoutDelayMs : policy.playoutDelayMs;
-    playoutDelayMs = Number.isFinite(playoutDelayMs) === true && playoutDelayMs >= 0 ? playoutDelayMs : 120;
-    minPlayoutDelayMs =
-      Number.isFinite(policy.minPlayoutDelayMs) === true && policy.minPlayoutDelayMs >= 0 ? policy.minPlayoutDelayMs : playoutDelayMs;
-    maxPlayoutDelayMs =
-      Number.isFinite(policy.maxPlayoutDelayMs) === true && policy.maxPlayoutDelayMs >= minPlayoutDelayMs
-        ? policy.maxPlayoutDelayMs
-        : Math.max(minPlayoutDelayMs, playoutDelayMs);
-    playoutAdjustStepMs =
-      Number.isFinite(policy.playoutAdjustStepMs) === true && policy.playoutAdjustStepMs > 0 ? policy.playoutAdjustStepMs : 20;
+    minPlayoutDelayMs = policy.minPlayoutDelayMs;
+    maxPlayoutDelayMs = policy.maxPlayoutDelayMs;
+    playoutAdjustStepMs = policy.playoutAdjustStepMs;
     playoutDelayMs = Math.min(maxPlayoutDelayMs, Math.max(minPlayoutDelayMs, playoutDelayMs));
-    maxLagBehindLiveMs =
-      Number.isFinite(policy.maxLagBehindLiveMs) === true && policy.maxLagBehindLiveMs > 0 ? policy.maxLagBehindLiveMs : 750;
-    dueTolerance = Number.isFinite(policy.dueTolerance) === true && policy.dueTolerance >= 0 ? policy.dueTolerance : 10;
-    dueSlack = Number.isFinite(policy.dueSlack) === true && policy.dueSlack >= 0 ? policy.dueSlack : 10;
-    catchupExitThresholdMs =
-      Number.isFinite(policy.catchupExitThresholdMs) === true && policy.catchupExitThresholdMs >= 0 ? policy.catchupExitThresholdMs : 250;
-
-    catchupAudioBurstLimit =
-      Number.isFinite(policy.catchupAudioBurstLimit) === true && policy.catchupAudioBurstLimit > 0
-        ? policy.catchupAudioBurstLimit
-        : isRecordOutput === true
-          ? 4
-          : 2;
-
-    catchupVideoBurstLimit =
-      Number.isFinite(policy.catchupVideoBurstLimit) === true && policy.catchupVideoBurstLimit > 0
-        ? policy.catchupVideoBurstLimit
-        : isRecordOutput === true
-          ? 8
-          : 4;
-
-    normalAudioBurstLimit =
-      Number.isFinite(policy.normalAudioBurstLimit) === true && policy.normalAudioBurstLimit > 0
-        ? policy.normalAudioBurstLimit
-        : isRecordOutput === true
-          ? 4
-          : 2;
-
-    normalVideoBurstLimit =
-      Number.isFinite(policy.normalVideoBurstLimit) === true && policy.normalVideoBurstLimit > 0
-        ? policy.normalVideoBurstLimit
-        : isRecordOutput === true
-          ? 3
-          : 2;
+    maxLagBehindLiveMs = policy.maxLagBehindLiveMs;
+    dueTolerance = policy.dueTolerance;
+    dueSlack = policy.dueSlack;
+    catchupExitThresholdMs = policy.catchupExitThresholdMs;
+    catchupAudioBurstLimit = policy.catchupAudioBurstLimit;
+    catchupVideoBurstLimit = policy.catchupVideoBurstLimit;
+    normalAudioBurstLimit = policy.normalAudioBurstLimit;
+    normalVideoBurstLimit = policy.normalVideoBurstLimit;
 
     // Cached clean SPS/PPS for H264 keyframe bootstrap.
     // Transport media itself remains Annex-B and is written directly.
@@ -1635,6 +1600,11 @@ export default class Streamer {
       }
     }
 
+    // Keep the unconsumed media candidate between loop iterations. Only the
+    // selected media cursor advances, so resolving both again wastes a search.
+    videoItem = state.videoCursor < timelineEnd ? timeline.nextVideoFrom(state.videoCursor) : undefined;
+    audioItem = includeAudio === true && state.audioCursor < timelineEnd ? timeline.nextAudioFrom(state.audioCursor) : undefined;
+
     // Main processing loop.
     // MediaTimeline gives us indexed video/audio lookup so we no longer walk
     // the mixed shared buffer looking for the next item of each type.
@@ -1648,9 +1618,6 @@ export default class Streamer {
         break;
       }
 
-      videoItem = state.videoCursor < timelineEnd ? timeline.nextVideoFrom(state.videoCursor) : undefined;
-      audioItem = includeAudio === true && state.audioCursor < timelineEnd ? timeline.nextAudioFrom(state.audioCursor) : undefined;
-
       dueVideo = false;
       dueAudio = false;
       capVideo = false;
@@ -1809,6 +1776,7 @@ export default class Streamer {
             this.#statsDrop(output, Streamer.MEDIA_TYPE.VIDEO);
             state.videoCursor = nextCursor;
             processed++;
+            videoItem = state.videoCursor < timelineEnd ? timeline.nextVideoFrom(state.videoCursor) : undefined;
             continue;
           }
 
@@ -1876,6 +1844,7 @@ export default class Streamer {
           break;
         }
 
+        videoItem = state.videoCursor < timelineEnd ? timeline.nextVideoFrom(state.videoCursor) : undefined;
         continue;
       }
 
@@ -1884,6 +1853,7 @@ export default class Streamer {
         if (includeAudio !== true || outputAudio === null) {
           state.audioCursor = nextCursor;
           processed++;
+          audioItem = state.audioCursor < timelineEnd ? timeline.nextAudioFrom(state.audioCursor) : undefined;
           continue;
         }
 
@@ -1893,14 +1863,14 @@ export default class Streamer {
           this.#statsDrop(output, Streamer.MEDIA_TYPE.AUDIO);
           state.audioCursor = nextCursor;
           processed++;
+          audioItem = state.audioCursor < timelineEnd ? timeline.nextAudioFrom(state.audioCursor) : undefined;
           continue;
         }
 
-        queuedBytes =
-          (Number.isFinite(outputAudio.writableLength) === true ? outputAudio.writableLength : 0) +
-          (Number.isFinite(outputAudio.readableLength) === true ? outputAudio.readableLength : 0);
-
         if (this.supportDump === true && typeof output?.stats?.diagnostics === 'object' && output.stats.diagnostics !== null) {
+          queuedBytes =
+            (Number.isFinite(outputAudio.writableLength) === true ? outputAudio.writableLength : 0) +
+            (Number.isFinite(outputAudio.readableLength) === true ? outputAudio.readableLength : 0);
           output.stats.diagnostics.maxAudioQueuedBytes = Math.max(output.stats.diagnostics.maxAudioQueuedBytes ?? 0, queuedBytes);
         }
 
@@ -1916,14 +1886,15 @@ export default class Streamer {
         state.audioCursor = nextCursor;
         processed++;
 
-        queuedBytes =
-          (Number.isFinite(outputAudio.writableLength) === true ? outputAudio.writableLength : 0) +
-          (Number.isFinite(outputAudio.readableLength) === true ? outputAudio.readableLength : 0);
-
         if (this.supportDump === true && typeof output?.stats?.diagnostics === 'object' && output.stats.diagnostics !== null) {
+          queuedBytes =
+            (Number.isFinite(outputAudio.writableLength) === true ? outputAudio.writableLength : 0) +
+            (Number.isFinite(outputAudio.readableLength) === true ? outputAudio.readableLength : 0);
           output.stats.diagnostics.maxAudioQueuedBytes = Math.max(output.stats.diagnostics.maxAudioQueuedBytes ?? 0, queuedBytes);
         }
 
+        audioItem = state.audioCursor < timelineEnd ? timeline.nextAudioFrom(state.audioCursor) : undefined;
+
         continue;
       }
     }
@@ -1959,11 +1930,9 @@ export default class Streamer {
 
   #processOutput(dateNow, budgetMs) {
     let timeline = this.#timeline;
-    let itemsLength = 0;
     let hasOutputs = false;
     let cutoffTime = 0;
     let latestItemTime = undefined;
-    let trimCount = 0;
     let oldestProtectedCursor = this.#itemIndex;
     let fallbackFrame = undefined;
     let output = undefined;
@@ -1972,11 +1941,10 @@ export default class Streamer {
       return;
     }
 
-    itemsLength = typeof timeline.size === 'number' ? timeline.size : 0;
     hasOutputs = this.#outputs.size !== 0;
 
     // Determine if timeline contains items older than retention window.
-    if (itemsLength !== 0) {
+    if (timeline.size !== 0) {
       latestItemTime = timeline.latestTime();
       latestItemTime = Number.isFinite(latestItemTime) === true ? latestItemTime : dateNow;
       cutoffTime = latestItemTime - this.#bufferDuration;
@@ -1986,11 +1954,7 @@ export default class Streamer {
       oldestProtectedCursor = timeline.protectedStart(this.#outputs);
 
       // Trim expired media while respecting the oldest protected output cursor.
-      trimCount = timeline.trim(cutoffTime, oldestProtectedCursor);
-
-      if (trimCount !== 0) {
-        itemsLength -= trimCount;
-      }
+      timeline.trim(cutoffTime, oldestProtectedCursor);
     }
 
     // No outputs attached means we only need retention maintenance this tick.

package/dist/streamtransport.js

@@ -66,7 +66,7 @@
 // - CLOSING
 // - CLOSED
 //
-// Code version 2026.05.18
+// Code version 2026.08.20
 // Mark Hulskamp
 'use strict';
 
@@ -497,12 +497,12 @@ export default class StreamTransport {
     for (let packet of group.packets) {
       if (typeof expectedSequence === 'number' && packet.sequenceNumber !== expectedSequence) {
         group.hasSequenceGap = true;
-        break;
       }
 
       expectedSequence = (packet.sequenceNumber + 1) & jitter.sequenceMask;
     }
 
+    group.lastSequence = group.packets.at(-1)?.sequenceNumber;
     group.packetsSorted = true;
 
     return group.hasSequenceGap === true;
@@ -1340,9 +1340,10 @@ export default class StreamTransport {
         rtpTimestamp: rtpTimestamp,
         firstReceivedAt: receivedAt,
         lastReceivedAt: receivedAt,
+        lastSequence: undefined,
         markerSeen: false,
         hasSequenceGap: false,
-        packetsSorted: false,
+        packetsSorted: true,
         sequences: new Set(),
         packets: [],
       };
@@ -1362,7 +1363,17 @@ export default class StreamTransport {
     group.firstReceivedAt = Math.min(group.firstReceivedAt, receivedAt);
     group.lastReceivedAt = Math.max(group.lastReceivedAt, receivedAt);
     group.markerSeen = group.markerSeen === true || packetInfo.marker === true;
-    group.packetsSorted = false;
+
+    // Preserve the sorted fast path for the overwhelmingly common case where
+    // RTP packets arrive in sequence. Re-sort only after actual reordering.
+    if (
+      group.packets.length !== 0 &&
+      (group.packetsSorted !== true ||
+        typeof group.lastSequence !== 'number' ||
+        sequenceNumber !== ((group.lastSequence + 1) & jitter.sequenceMask))
+    ) {
+      group.packetsSorted = false;
+    }
 
     // Protocol-specific grouped metadata can be supplied in packetInfo.group.
     // Boolean flags accumulate across packets so later fragments cannot clear
@@ -1380,6 +1391,9 @@ export default class StreamTransport {
       receivedAt: receivedAt,
     });
     group.sequences.add(sequenceNumber);
+    if (group.packetsSorted === true) {
+      group.lastSequence = sequenceNumber;
+    }
     jitter.packetCount++;
 
     while (jitter.packetCount > jitter.maxPackets && jitter.queue.size > 0) {

package/dist/webrtc.js

@@ -50,7 +50,7 @@
 // - Output playout timing, catch-up, and live latency policy are owned by Streamer
 // - Incomplete keyframes and pathological access units are dropped/recovered locally rather than blocking the plugin process
 //
-// Code version 2026.05.20
+// Code version 2026.08.20
 // Mark Hulskamp
 'use strict';
 
@@ -102,8 +102,9 @@ const VIDEO_RTP_REORDER_DELAY_MS = 250; // Hold video RTP briefly so reordered f
 const VIDEO_RTP_REORDER_MAX_PACKETS = 4096; // Bound video RTP queue while allowing a fragmented 4MiB keyframe plus packet overhead
 const VIDEO_RTP_DRAIN_MAX_GROUPS = 8; // Bound synchronous video jitter release work per callback
 const VIDEO_RTP_DRAIN_MAX_PACKETS = 128; // Bound synchronous H264 assembly work per callback
-const STARTUP_KEYFRAME_PLI_INTERVAL_MS = 1500; // Retry startup keyframe requests while waiting for first decodable IDR
-const STARTUP_KEYFRAME_PLI_MAX_MS = 12000; // Bound startup PLI retries so slow sources do not spam RTCP forever
+const STARTUP_KEYFRAME_REQUEST_INTERVAL_MS = 1500; // Retry startup keyframe requests while waiting for first decodable IDR
+const STARTUP_KEYFRAME_REQUEST_MAX_MS = 12000; // Bound startup retries so slow sources do not spam RTCP forever
+const VIDEO_REMB_BITRATE = 4_000_000; // Raise the sender's conservative startup estimate while keyframes are requested
 
 // WebRTC object
 export default class WebRTC extends StreamTransport {
@@ -201,7 +202,9 @@ export default class WebRTC extends StreamTransport {
       return;
     }
 
-    let homeFoyerResponse = await this.#grpcTransport.command(GOOGLE_HOME_FOYER_PREFIX, 'CameraService', 'SendCameraViewIntent', {
+    // Wake the camera while local certificate, SDP, and ICE candidate work runs.
+    // These operations are independent until JoinStream sends the completed offer.
+    let viewIntentPromise = this.#grpcTransport.command(GOOGLE_HOME_FOYER_PREFIX, 'CameraService', 'SendCameraViewIntent', {
       request: {
         googleDeviceId: {
           value: this.#googleHomeDeviceUUID,
@@ -210,16 +213,6 @@ export default class WebRTC extends StreamTransport {
       },
     });
 
-    if (this.#closeInProgress === true || this.#peerConnection !== undefined) {
-      return;
-    }
-
-    if (homeFoyerResponse?.status !== 0) {
-      this?.log?.debug?.('Request to start camera viewing was not accepted for uuid "%s"', this.uuid);
-      this.setState(StreamTransport.STATE.CLOSED, { reason: 'view-intent-failed' });
-      return;
-    }
-
     // Create our local WebRTC peer connection and advertise the codecs we support.
     // We receive H264 video and Opus audio from the camera, then convert that into
     // Streamer media items for live view and recording.
@@ -275,6 +268,29 @@ export default class WebRTC extends StreamTransport {
     // If accepted, we will get an SDP answer back plus a streamId for later extend/end/talkback calls.
     let webRTCOffer = await peerConnection.createOffer();
     await peerConnection.setLocalDescription(webRTCOffer);
+    let localOfferSdp = peerConnection.localDescription?.sdp ?? webRTCOffer.sdp;
+
+    // Both overlapped operations must complete successfully before joining.
+    let homeFoyerResponse = await viewIntentPromise;
+
+    if (this.#peerConnection !== peerConnection || this.#closeInProgress === true) {
+      try {
+        await peerConnection?.close?.();
+      } catch {
+        // Empty
+      }
+      return;
+    }
+
+    if (homeFoyerResponse?.status !== 0) {
+      await peerConnection?.close?.();
+      this.#peerConnection = undefined;
+      this.#videoTransceiver = undefined;
+      this.#audioTransceiver = undefined;
+      this?.log?.debug?.('Request to start camera viewing was not accepted for uuid "%s"', this.uuid);
+      this.setState(StreamTransport.STATE.CLOSED, { reason: 'view-intent-failed' });
+      return;
+    }
 
     homeFoyerResponse = await this.#grpcTransport.command(GOOGLE_HOME_FOYER_PREFIX, 'CameraService', 'JoinStream', {
       command: 'offer',
@@ -283,7 +299,9 @@ export default class WebRTC extends StreamTransport {
       streamContext: 'STREAM_CONTEXT_DEFAULT',
       // Request highest possible resolution; actual delivered resolution may be lower.
       requestedVideoResolution: 'VIDEO_RESOLUTION_FULL_HIGH',
-      sdp: webRTCOffer.sdp,
+      // setLocalDescription() gathers ICE candidates and updates the peer's SDP.
+      // The earlier createOffer() value does not contain those candidates.
+      sdp: localOfferSdp,
     });
 
     if (this.#peerConnection !== peerConnection) {
@@ -830,7 +848,7 @@ export default class WebRTC extends StreamTransport {
       // Start bounded startup keyframe requests. The first call may happen
       // before the RTP SSRC is known, so retries continue until the first IDR.
       this.#startStartupKeyframeTimer();
-      this.#sendVideoPLI();
+      this.#requestVideoKeyframe();
 
       this.#refreshStallTimer();
       return;
@@ -851,32 +869,19 @@ export default class WebRTC extends StreamTransport {
   }
 
   #ensurePlaybackVideoTrack() {
-    // Build or repair the per-session video state once, keeping packet hot paths
-    // focused on RTP/H264 work instead of repeated object-shape checks.
-    if (typeof this.#tracks !== 'object' || this.#tracks === null) {
-      this.#tracks = {};
-    }
-
-    if (typeof this.#tracks.video !== 'object' || this.#tracks.video === null) {
-      this.#tracks.video = {};
-    }
-
+    // onTrack calls this before subscribing RTP callbacks, so packet handlers
+    // can use stable session state without repairing object shapes per packet.
     let video = this.#tracks.video;
 
-    video.id = typeof video.id === 'number' ? video.id : RTP_H264_VIDEO_PAYLOAD_TYPE;
-    video.rtxId = typeof video.rtxId === 'number' ? video.rtxId : RTP_H264_VIDEO_RTX_PAYLOAD_TYPE;
-    video.rtxSsrc = typeof video.rtxSsrc === 'number' ? video.rtxSsrc : undefined;
-    video.codec = typeof video.codec === 'string' ? video.codec : StreamTransport.CODEC_TYPE.H264;
-    video.sampleRate = typeof video.sampleRate === 'number' ? video.sampleRate : 90000;
-    video.lastPLITime = typeof video.lastPLITime === 'number' ? video.lastPLITime : undefined;
-    video.keyframeRequestInFlight = typeof video.keyframeRequestInFlight === 'boolean' ? video.keyframeRequestInFlight : false;
-    video.lastKeyframeEventTime = typeof video.lastKeyframeEventTime === 'number' ? video.lastKeyframeEventTime : undefined;
-
-    video.rtp = typeof video.rtp === 'object' && video.rtp !== null ? video.rtp : { lastSequence: undefined, lastTimestamp: undefined };
-    video.output = typeof video.output === 'object' && video.output !== null ? video.output : { lastTimestamp: undefined };
-    video.health = typeof video.health === 'object' && video.health !== null ? video.health : this.getMediaState('video');
-    video.deltaAudit =
-      typeof video.deltaAudit === 'object' && video.deltaAudit !== null ? video.deltaAudit : { hasAcceptedKeyframe: false };
+    video.id ??= RTP_H264_VIDEO_PAYLOAD_TYPE;
+    video.rtxId ??= RTP_H264_VIDEO_RTX_PAYLOAD_TYPE;
+    video.codec ??= StreamTransport.CODEC_TYPE.H264;
+    video.sampleRate ??= 90000;
+    video.keyframeRequestInFlight ??= false;
+    video.rtp ??= { lastSequence: undefined, lastTimestamp: undefined };
+    video.output ??= { lastTimestamp: undefined };
+    video.health ??= this.getMediaState('video');
+    video.deltaAudit ??= { hasAcceptedKeyframe: false };
 
     if (typeof video.jitter !== 'object' || video.jitter === null || video.jitter.groupByTimestamp !== true) {
       video.jitter = this.createJitterBuffer({
@@ -894,27 +899,16 @@ export default class WebRTC extends StreamTransport {
   }
 
   #ensurePlaybackAudioTrack() {
-    // Build or repair the per-session audio state once so packet handling only
-    // deals with RTP timing, Opus decode, and decoded-frame emission.
-    if (typeof this.#tracks !== 'object' || this.#tracks === null) {
-      this.#tracks = {};
-    }
-
-    if (typeof this.#tracks.audio !== 'object' || this.#tracks.audio === null) {
-      this.#tracks.audio = {};
-    }
-
+    // Audio follows the same onTrack-before-packets invariant as video.
     let audio = this.#tracks.audio;
 
-    audio.id = typeof audio.id === 'number' ? audio.id : RTP_OPUS_AUDIO_PAYLOAD_TYPE;
-    audio.codec = typeof audio.codec === 'string' ? audio.codec : StreamTransport.CODEC_TYPE.OPUS;
-    audio.sampleRate = typeof audio.sampleRate === 'number' ? audio.sampleRate : 48000;
-    audio.channels = typeof audio.channels === 'number' ? audio.channels : 2;
-    audio.packetTime = typeof audio.packetTime === 'number' ? audio.packetTime : 20;
-    audio.rtp = typeof audio.rtp === 'object' && audio.rtp !== null ? audio.rtp : { lastSequence: undefined, lastTimestamp: undefined };
-    audio.output = typeof audio.output === 'object' && audio.output !== null ? audio.output : { lastTimestamp: undefined };
-    audio.lastDecodeFallbackLogTime = typeof audio.lastDecodeFallbackLogTime === 'number' ? audio.lastDecodeFallbackLogTime : undefined;
-    audio.lastDecodeErrorLogTime = typeof audio.lastDecodeErrorLogTime === 'number' ? audio.lastDecodeErrorLogTime : undefined;
+    audio.id ??= RTP_OPUS_AUDIO_PAYLOAD_TYPE;
+    audio.codec ??= StreamTransport.CODEC_TYPE.OPUS;
+    audio.sampleRate ??= 48000;
+    audio.channels ??= 2;
+    audio.packetTime ??= 20;
+    audio.rtp ??= { lastSequence: undefined, lastTimestamp: undefined };
+    audio.output ??= { lastTimestamp: undefined };
 
     if (typeof audio.jitter !== 'object' || audio.jitter === null || audio.jitter.groupByTimestamp === true) {
       audio.jitter = this.createJitterBuffer({
@@ -1038,12 +1032,14 @@ export default class WebRTC extends StreamTransport {
             );
           }
 
-          this.#sendVideoPLI();
-          this.markMediaIssue('video', 'jitter-keyframe-incomplete');
-          this.clearJitterBuffer(jitter);
+          this.#requestVideoRecovery('jitter-keyframe-incomplete');
           this.#resetFragmentedVideoFrame();
           this.#resetPendingVideoFrame();
-          break;
+
+          // releaseJitterGroups already removed only the failed timestamp.
+          // Keep later groups: one may contain the complete recovery keyframe,
+          // whereas clearing them here can turn one damaged IDR into a long gap.
+          continue;
         }
 
         continue;
@@ -1131,6 +1127,7 @@ export default class WebRTC extends StreamTransport {
     let payloadInfo = undefined;
     let queueSequenceNumber = 0;
     let packetReceivedAt = 0;
+    let learnedPrimarySsrc = false;
 
     // Ensure we have a valid RTP packet with a payload before processing video data
     if (
@@ -1154,7 +1151,7 @@ export default class WebRTC extends StreamTransport {
     let ssrc = Number.isInteger(header.ssrc) === true ? header.ssrc >>> 0 : undefined;
     packetReceivedAt = Number.isFinite(rtpPacket.receivedAt) === true ? rtpPacket.receivedAt : Date.now();
 
-    let video = this.#ensurePlaybackVideoTrack();
+    let video = this.#tracks.video;
     let h264 = video.h264;
     let videoRtp = video.rtp;
     let deltaAudit = video.deltaAudit;
@@ -1192,6 +1189,7 @@ export default class WebRTC extends StreamTransport {
       // request a fresh IDR even while the first timestamp group is still queued.
       if (typeof video.ssrc !== 'number' && typeof ssrc === 'number') {
         video.ssrc = ssrc;
+        learnedPrimarySsrc = true;
       }
 
       if (typeof video.ssrc === 'number' && typeof ssrc === 'number' && ssrc !== video.ssrc) {
@@ -1200,6 +1198,10 @@ export default class WebRTC extends StreamTransport {
 
       payloadInfo = RtpH264.getPayloadInfo(payload);
 
+      if (learnedPrimarySsrc === true && this.ready !== true && payloadInfo.hasKeyFrame !== true) {
+        this.#requestVideoKeyframe();
+      }
+
       if (
         this.pushJitterPacket(jitter, {
           rtpTimestamp: rtpTimestamp,
@@ -1253,7 +1255,7 @@ export default class WebRTC extends StreamTransport {
     }
 
     if (this.ready !== true && h264?.hasIDR !== true && typeof video.ssrc === 'number') {
-      this.#sendVideoPLI();
+      this.#requestVideoKeyframe();
     }
 
     // Drop duplicate or clearly late/out-of-order packets before they touch assembly state.
@@ -1315,8 +1317,7 @@ export default class WebRTC extends StreamTransport {
         );
 
         if (h264.pendingKeyFrame === true || (Number.isFinite(pendingAgeMs) === true && pendingAgeMs >= 300)) {
-          this.#sendVideoPLI();
-          this.markMediaIssue('video', 'pending-incomplete');
+          this.#requestVideoRecovery('pending-incomplete');
         }
 
         this.#resetPendingVideoFrame();
@@ -1353,8 +1354,7 @@ export default class WebRTC extends StreamTransport {
         );
 
         if (h264.fuNalType === H264.NALUS.TYPES.IDR || (Number.isFinite(fuAgeMs) === true && fuAgeMs >= 600)) {
-          this.#sendVideoPLI();
-          this.markMediaIssue('video', 'fu-incomplete');
+          this.#requestVideoRecovery('fu-incomplete');
         }
       }
 
@@ -1422,44 +1422,30 @@ export default class WebRTC extends StreamTransport {
       });
 
       if (fuResult?.ok !== true) {
-        if (fuResult?.reason === 'orphan') {
-          if (fuResult?.nalType === H264.NALUS.TYPES.IDR) {
-            this.recordVideoDrop('fu-keyframe-incomplete');
-
-            this?.log?.debug?.(
-              'Dropping orphaned WebRTC FU-A keyframe fragment for uuid "%s": seq="%s" ts="%s"',
-              this.uuid,
-              sequenceNumber,
-              rtpTimestamp,
-            );
-
-            this.#sendVideoPLI();
-            this.markMediaIssue('video', 'fu-incomplete');
-          }
-
-          this.#resetFragmentedVideoFrame();
-          if (h264.pendingRtpTimestamp === rtpTimestamp && h264.pendingHasVcl !== true) {
-            this.#resetPendingVideoFrame();
+        // Orphans and sequence gaps both invalidate the retained FU-A fragments.
+        // Handle them together while preserving the stronger keyframe recovery:
+        // gaps count delta losses, while IDR loss also requests a replacement.
+        if (fuResult?.reason === 'orphan' || fuResult?.reason === 'gap') {
+          let keyframeFragment = fuResult?.nalType === H264.NALUS.TYPES.IDR;
+
+          if (fuResult.reason === 'gap' || keyframeFragment === true) {
+            this.recordVideoDrop(keyframeFragment === true ? 'fu-keyframe-incomplete' : 'fu-delta-incomplete');
           }
-          return;
-        }
-
-        if (fuResult?.reason === 'gap') {
-          this.recordVideoDrop(fuResult?.nalType === H264.NALUS.TYPES.IDR ? 'fu-keyframe-incomplete' : 'fu-delta-incomplete');
 
-          if (fuResult?.nalType === H264.NALUS.TYPES.IDR) {
-            h264.pendingCorrupt = true;
+          if (keyframeFragment === true) {
+            if (fuResult.reason === 'gap') {
+              h264.pendingCorrupt = true;
+            }
 
             this?.log?.debug?.(
-              'Dropping gapped WebRTC FU-A keyframe for uuid "%s": expectedSeq="%s" seq="%s" ts="%s"',
+              'Dropping %s WebRTC FU-A keyframe fragment for uuid "%s": expectedSeq="%s" seq="%s" ts="%s"',
+              fuResult.reason,
               this.uuid,
               fuResult.expectedSequenceNumber,
               sequenceNumber,
               rtpTimestamp,
             );
-
-            this.#sendVideoPLI();
-            this.markMediaIssue('video', 'fu-incomplete');
+            this.#requestVideoRecovery('fu-incomplete');
           }
 
           this.#resetFragmentedVideoFrame();
@@ -1488,8 +1474,7 @@ export default class WebRTC extends StreamTransport {
         );
 
         if (fuResult.previousNalType === H264.NALUS.TYPES.IDR) {
-          this.#sendVideoPLI();
-          this.markMediaIssue('video', 'fu-incomplete');
+          this.#requestVideoRecovery('fu-incomplete');
         }
       }
 
@@ -1656,8 +1641,7 @@ export default class WebRTC extends StreamTransport {
         keyframeByteLimit,
       );
 
-      this.#sendVideoPLI();
-      this.markMediaIssue('video', 'oversized-keyframe');
+      this.#requestVideoRecovery('oversized-keyframe');
       this.#resetPendingVideoFrame();
       return;
     }
@@ -1689,8 +1673,7 @@ export default class WebRTC extends StreamTransport {
       );
 
       if (pendingKeyFrame === true) {
-        this.#sendVideoPLI();
-        this.markMediaIssue('video', 'corrupt-keyframe');
+        this.#requestVideoRecovery('corrupt-keyframe');
       }
 
       this.#resetPendingVideoFrame();
@@ -1757,8 +1740,7 @@ export default class WebRTC extends StreamTransport {
           keyframeAssemblyLimitMs,
         );
 
-        this.#sendVideoPLI();
-        this.markMediaIssue('video', 'shock-keyframe');
+        this.#requestVideoRecovery('shock-keyframe');
         this.#resetPendingVideoFrame();
         return;
       }
@@ -1913,18 +1895,18 @@ export default class WebRTC extends StreamTransport {
         return;
       }
 
-      if (now - this.#startupKeyframeStartedAt > STARTUP_KEYFRAME_PLI_MAX_MS) {
+      if (now - this.#startupKeyframeStartedAt > STARTUP_KEYFRAME_REQUEST_MAX_MS) {
         this.#stopStartupKeyframeTimer();
         this?.log?.debug?.(
           'Stopped WebRTC startup keyframe requests for uuid "%s" after %sms without accepted keyframe',
           this.uuid,
-          STARTUP_KEYFRAME_PLI_MAX_MS,
+          STARTUP_KEYFRAME_REQUEST_MAX_MS,
         );
         return;
       }
 
-      this.#sendVideoPLI();
-    }, STARTUP_KEYFRAME_PLI_INTERVAL_MS);
+      this.#requestVideoKeyframe();
+    }, STARTUP_KEYFRAME_REQUEST_INTERVAL_MS);
   }
 
   #stopStartupKeyframeTimer() {
@@ -1995,7 +1977,7 @@ export default class WebRTC extends StreamTransport {
     let ssrc = Number.isInteger(header.ssrc) === true ? header.ssrc >>> 0 : undefined;
     now = fromJitterBuffer === true && Number.isFinite(rtpPacket.receivedAt) === true ? rtpPacket.receivedAt : now;
 
-    let audio = this.#ensurePlaybackAudioTrack();
+    let audio = this.#tracks.audio;
     let audioRtp = audio.rtp;
     let audioOutput = audio.output;
 
@@ -2122,12 +2104,20 @@ export default class WebRTC extends StreamTransport {
     });
   }
 
-  #sendVideoPLI() {
+  #requestVideoRecovery(reason) {
+    // A damaged keyframe needs both an upstream replacement request and a
+    // health signal so StreamTransport can apply its existing recovery policy.
+    this.#requestVideoKeyframe();
+    this.markMediaIssue('video', reason);
+  }
+
+  #requestVideoKeyframe() {
     let video = this.#tracks?.video;
     let now = Date.now();
-    let sendResult = undefined;
+    let pliSendResult = undefined;
+    let receiver = this.#videoTransceiver?.receiver;
 
-    if (this.#videoTransceiver === undefined || video === undefined || typeof video?.ssrc !== 'number') {
+    if (receiver === undefined || video === undefined || typeof video?.ssrc !== 'number') {
       return;
     }
 
@@ -2135,19 +2125,19 @@ export default class WebRTC extends StreamTransport {
       return;
     }
 
-    if (typeof video.lastPLITime === 'number' && now - video.lastPLITime < 1500) {
+    if (typeof video.lastKeyframeRequestTime === 'number' && now - video.lastKeyframeRequestTime < STARTUP_KEYFRAME_REQUEST_INTERVAL_MS) {
       return;
     }
 
-    video.lastPLITime = now;
+    video.lastKeyframeRequestTime = now;
     video.keyframeRequestInFlight = true;
     this.recordKeyframeRequest();
 
     try {
-      sendResult = this.#videoTransceiver?.receiver?.sendRtcpPLI?.(video.ssrc);
+      pliSendResult = receiver.sendRtcpPLI?.(video.ssrc);
 
-      if (sendResult instanceof Promise) {
-        sendResult
+      if (pliSendResult instanceof Promise) {
+        pliSendResult
           .catch((error) => {
             this?.log?.debug?.('Error sending WebRTC PLI for uuid "%s": %s', this.uuid, String(error));
           })
@@ -2156,13 +2146,60 @@ export default class WebRTC extends StreamTransport {
               video.keyframeRequestInFlight = false;
             }
           });
-        return;
       }
     } catch (error) {
       this?.log?.debug?.('Error sending WebRTC PLI for uuid "%s": %s', this.uuid, String(error));
     }
 
-    video.keyframeRequestInFlight = false;
+    if (pliSendResult instanceof Promise === false) {
+      video.keyframeRequestInFlight = false;
+    }
+
+    if (typeof receiver?.rtcpSsrc !== 'number' || typeof receiver?.dtlsTransport?.sendRtcp !== 'function') {
+      return;
+    }
+
+    try {
+      let firBuffer = Buffer.alloc(16);
+      let rembExponent = 0;
+      let rembMantissa = VIDEO_REMB_BITRATE;
+
+      firBuffer.writeUInt32BE(receiver.rtcpSsrc >>> 0, 0);
+      firBuffer.writeUInt32BE(video.ssrc >>> 0, 4);
+      firBuffer.writeUInt32BE(video.ssrc >>> 0, 8);
+      firBuffer[12] = this.stats.media.keyframeRequests & 0xff;
+
+      while (rembMantissa > 0x3ffff) {
+        rembMantissa = Math.floor(rembMantissa / 2);
+        rembExponent++;
+      }
+
+      receiver.dtlsTransport
+        .sendRtcp([
+          new werift.RtcpPayloadSpecificFeedback({
+            feedback: {
+              count: 4,
+              length: firBuffer.length / 4 - 1,
+              serialize: () => firBuffer,
+            },
+          }),
+          new werift.RtcpPayloadSpecificFeedback({
+            feedback: new werift.ReceiverEstimatedMaxBitrate({
+              senderSsrc: receiver.rtcpSsrc,
+              mediaSsrc: 0,
+              ssrcNum: 1,
+              brExp: rembExponent,
+              brMantissa: rembMantissa,
+              ssrcFeedbacks: [video.ssrc],
+            }),
+          }),
+        ])
+        .catch((error) => {
+          this?.log?.debug?.('Error sending WebRTC FIR/REMB for uuid "%s": %s', this.uuid, String(error));
+        });
+    } catch (error) {
+      this?.log?.debug?.('Error building WebRTC FIR/REMB for uuid "%s": %s', this.uuid, String(error));
+    }
   }
 
   #requestReconnect(reason) {

package/package.json

@@ -2,7 +2,7 @@
     "name": "homebridge-nest-accfactory",
     "displayName": "Nest Accfactory",
     "type": "module",
-    "version": "0.4.3",
+    "version": "0.4.4-beta.2",
     "description": "Homebridge support for Nest/Google devices including HomeKit Secure Video (HKSV) support for doorbells and cameras",
     "author": "n0rt0nthec4t",
     "license": "Apache-2.0",
@@ -36,8 +36,8 @@
     ],
     "main": "dist/index.js",
     "engines": {
-        "node": "^22 || ^24",
-        "homebridge": "^1.11.4 || ^2.3.1"
+        "node": "^22 || ^24 || ^26",
+        "homebridge": "^1.11.4 || ^2.4.0"
     },
     "files": [
         "LICENSE",
@@ -48,27 +48,30 @@
     ],
     "scripts": {
         "clean": "rimraf dist",
-        "format": "prettier --write \"src/**/*.{js,mjs}\" && eslint --fix \"src/**/*.{js,mjs}\"",
-        "lint": "eslint \"src/**/*.{js,mjs}\" --max-warnings=20",
+        "format": "prettier --write \"{src,test}/**/*.{js,mjs}\" && eslint --fix \"src/**/*.{js,mjs}\" \"test/**/*.js\"",
+        "lint": "eslint \"src/**/*.{js,mjs}\" \"test/**/*.js\" --max-warnings=20",
+        "test": "node --test",
+        "test:coverage": "node --test --experimental-test-coverage",
+        "check": "npm run lint && npm test && npm run build",
         "build": "npm run clean && copyfiles -u 1 src/*.js dist && copyfiles -u 2 src/HomeKitDevice/*.js dist && copyfiles -u 2 src/HomeKitHistory/*.js dist && copyfiles -u 1 src/res/*.h264 dist && copyfiles -u 1 src/res/*.jpg dist && copyfiles -u 1 'src/protobuf/**/*.proto' dist && copyfiles -u 1 src/plugins/*.js dist && copyfiles -u 1 src/plugins/*.mjs dist",
-        "prepublishOnly": "npm run format && npm run lint && npm run build"
+        "prepublishOnly": "npm run format && npm run lint && npm test && npm run build"
     },
     "devDependencies": {
         "@eslint/js": "^10.0.1",
         "@stylistic/eslint-plugin": "^5.10.0",
-        "@types/node": "^26.2.0",
-        "@typescript-eslint/parser": "^8.66.0",
+        "@types/node": "^26.4.1",
+        "@typescript-eslint/parser": "^8.69.0",
         "copyfiles": "^2.4.1",
-        "eslint": "^10.8.1",
-        "homebridge": "^1.11.4 || ^2.3.1",
+        "eslint": "^10.10.0",
+        "homebridge": "^1.11.4 || ^2.4.0",
         "prettier": "^3.9.6",
-        "prettier-eslint": "^17.1.1",
+        "prettier-eslint": "^17.1.2",
         "rimraf": "^6.1.3"
     },
     "dependencies": {
         "@evan/opus": "^1.0.3",
         "chalk": "^6.0.0",
-        "protobufjs": "^8.7.2",
+        "protobufjs": "^8.8.0",
         "werift": "^0.24.4"
     }
 }