@camstack/addon-pipeline 1.2.51 → 1.2.52

package/dist/stream-broker/index.mjs

@@ -1,5 +1,5 @@
 import { r as __toESM, t as __commonJSMin } from "../chunk-DnnnRqeS.mjs";
-import { $ as webrtcSessionCapability, At as union, B as maskUrlCredentials, C as addonWidgetsSourceCapability, Ct as discriminatedUnion, D as createHwAccelCache, Dt as object, E as cameraStreamsCapability, Et as number, F as egressTransportFromRequest, J as resolveEgressDecodeHwAccel, Ot as record, P as egressTranscodeSharingKey, St as boolean, Tt as literal, Z as streamBrokerCapability, _t as parseProfileBrokerId, at as isSoftwareDecode, bt as _enum, ct as DeviceFeature, et as errMsg, f as EncodeProfileSchema, ft as isEvent, ht as nodePin, it as invocationFromEncodeProfile, jt as EventCategory, kt as string, lt as DeviceType, mt as makeSourceBrokerId, nt as Fmp4BoxSplitter, ot as BaseAddon, pt as makeProfileBrokerId, rt as buildFfmpegArgs$1, st as CAM_PROFILE_ORDER, tt as AUDIO_PRESETS, ut as createEvent, v as RATE_CONTROL_RELAXED, x as RingBuffer, xt as array, y as RATE_CONTROL_TIGHT } from "../dist-Dkvk5ykl.mjs";
+import { $ as webrtcSessionCapability, At as union, B as maskUrlCredentials, C as addonWidgetsSourceCapability, Ct as discriminatedUnion, D as createHwAccelCache, Dt as object, E as cameraStreamsCapability, Et as number, F as egressTransportFromRequest, J as resolveEgressDecodeHwAccel, Ot as record, P as egressTranscodeSharingKey, St as boolean, Tt as literal, Z as streamBrokerCapability, _t as parseProfileBrokerId, at as isSoftwareDecode, bt as _enum, ct as DeviceFeature, et as errMsg, f as EncodeProfileSchema, ft as isEvent, ht as nodePin, it as invocationFromEncodeProfile, jt as EventCategory, kt as string, lt as DeviceType, mt as makeSourceBrokerId, nt as Fmp4BoxSplitter, ot as BaseAddon, pt as makeProfileBrokerId, rt as buildFfmpegArgs$1, st as CAM_PROFILE_ORDER, tt as AUDIO_PRESETS, ut as createEvent, v as RATE_CONTROL_RELAXED, x as RingBuffer, xt as array, y as RATE_CONTROL_TIGHT } from "../dist-gB3o0cyM.mjs";
 import { n as profileForStreamId } from "../remote-restream-BeHi78PZ.mjs";
 import { n as createFileDataPlaneHandler } from "../addon-utils-QuxtOdK1.mjs";
 import { createRequire } from "node:module";
@@ -1040,6 +1040,275 @@ function buildDerivedTranscodeArgs(profile, loopbackUrl, decodeHwAccel = null) {
 	}));
 }
 //#endregion
+//#region src/stream-broker/paced-rtp-replay.ts
+var PACED_REPLAY_DEFAULTS = {
+	singleShotMax: 64,
+	tickMs: 5,
+	targetDurationMs: 300,
+	minChunkSize: 32,
+	maxChunkSize: 128
+};
+var defaultSchedule = (fn, ms) => {
+	const t = setTimeout(fn, ms);
+	return () => clearTimeout(t);
+};
+/**
+* Compute the per-tick chunk size for a burst of `total` packets: enough to
+* drain within `targetDurationMs` at one chunk per `tickMs`, clamped to
+* [minChunkSize, maxChunkSize]. Bursts ≤ `singleShotMax` are one chunk.
+*/
+function computeReplayChunkSize(total, tuning) {
+	if (total <= tuning.singleShotMax) return Math.max(total, 1);
+	const maxTicks = Math.max(1, Math.floor(tuning.targetDurationMs / tuning.tickMs));
+	const ideal = Math.ceil(total / maxTicks);
+	return Math.min(tuning.maxChunkSize, Math.max(tuning.minChunkSize, ideal));
+}
+/**
+* One in-flight paced replay. Create per burst; not reusable after
+* completion or abort.
+*/
+var PacedRtpReplay = class {
+	send;
+	onDone;
+	tuning;
+	schedule;
+	now;
+	queue = [];
+	cancelTimer = null;
+	started = false;
+	finished = false;
+	startedAt = 0;
+	chunkSize = 0;
+	initialPackets = 0;
+	enqueuedLive = 0;
+	sentCount = 0;
+	chunkCount = 0;
+	constructor(send, onDone, options) {
+		this.send = send;
+		this.onDone = onDone;
+		this.tuning = {
+			...PACED_REPLAY_DEFAULTS,
+			...options?.tuning
+		};
+		this.schedule = options?.schedule ?? defaultSchedule;
+		this.now = options?.now ?? Date.now;
+	}
+	/** True while packets remain to be drained (live packets must be enqueued,
+	*  not sent directly, to preserve order). */
+	get active() {
+		return this.started && !this.finished;
+	}
+	/**
+	* Begin the replay: sends the first chunk synchronously, then paces the
+	* rest. Calling `start` more than once is a no-op.
+	*/
+	start(initial) {
+		if (this.started) return;
+		this.started = true;
+		this.startedAt = this.now();
+		this.initialPackets = initial.length;
+		this.queue = [...initial];
+		this.chunkSize = computeReplayChunkSize(initial.length, this.tuning);
+		this.drainChunk();
+	}
+	/**
+	* Append a live packet behind the still-queued replay tail. No-op (returns
+	* false) when the replay is not active — the caller must then send the
+	* packet directly.
+	*/
+	enqueue(item) {
+		if (!this.active) return false;
+		this.queue.push(item);
+		this.enqueuedLive++;
+		return true;
+	}
+	/** Cancel the replay: pending packets are dropped, `onDone` fires with
+	*  `aborted: true`. Safe to call multiple times / before `start`. */
+	abort() {
+		if (this.finished) return;
+		if (!this.started) {
+			this.started = true;
+			this.startedAt = this.now();
+		}
+		this.finish(true);
+	}
+	drainChunk() {
+		if (this.finished) return;
+		this.chunkCount++;
+		const n = Math.min(this.chunkSize, this.queue.length);
+		for (let i = 0; i < n; i++) {
+			if (this.finished) return;
+			if (this.send(this.queue[i])) this.sentCount++;
+		}
+		this.queue = this.queue.slice(n);
+		if (this.queue.length === 0) {
+			this.finish(false);
+			return;
+		}
+		this.cancelTimer = this.schedule(() => {
+			this.cancelTimer = null;
+			this.drainChunk();
+		}, this.tuning.tickMs);
+	}
+	finish(aborted) {
+		if (this.finished) return;
+		this.finished = true;
+		if (this.cancelTimer) {
+			this.cancelTimer();
+			this.cancelTimer = null;
+		}
+		this.queue = [];
+		this.onDone({
+			packets: this.initialPackets,
+			enqueuedLive: this.enqueuedLive,
+			sent: this.sentCount,
+			chunks: this.chunkCount,
+			chunkSize: this.chunkSize,
+			durationMs: this.now() - this.startedAt,
+			aborted
+		});
+	}
+};
+//#endregion
+//#region src/stream-broker/rtsp/annexb-deframer.ts
+/**
+* Annex-B NAL unit deframer.
+*
+* ffmpeg outputs Annex-B via stdout in arbitrary chunk sizes — a single
+* `data` event may contain partial NALs, multiple NALs, or NALs split
+* across events. This class accumulates bytes and emits complete NAL units.
+*
+* Usage:
+*   const deframer = new AnnexBDeframer((nal, isKeyframe) => { ... })
+*   proc.stdout.on('data', chunk => deframer.push(chunk))
+*/
+var AnnexBDeframer = class {
+	onNal;
+	buffer = Buffer.alloc(0);
+	codec;
+	constructor(codec, onNal) {
+		this.onNal = onNal;
+		this.codec = codec;
+	}
+	/** Push raw Annex-B bytes. Complete NAL units are emitted via onNal callback. */
+	push(chunk) {
+		this.buffer = this.buffer.length > 0 ? Buffer.concat([this.buffer, chunk]) : chunk;
+		this.drain();
+	}
+	/** Flush remaining buffer (call on stream end). */
+	flush() {
+		if (this.buffer.length > 0) {
+			const nal = this.stripLeadingStartCode(this.buffer);
+			if (nal.length > 0) this.onNal(nal, this.isKeyframeNal(nal));
+			this.buffer = Buffer.alloc(0);
+		}
+	}
+	/** Extract complete NAL units from the buffer, leaving partial data for next push. */
+	drain() {
+		while (true) {
+			const firstSc = this.findStartCode(0);
+			if (firstSc < 0) return;
+			const nalStart = firstSc + (this.isStartCode4(firstSc) ? 4 : 3);
+			const nextSc = this.findStartCode(nalStart);
+			if (nextSc < 0) {
+				if (firstSc > 0) this.buffer = Buffer.from(this.buffer.subarray(firstSc));
+				return;
+			}
+			const nal = this.buffer.subarray(nalStart, nextSc);
+			if (nal.length > 0) this.onNal(nal, this.isKeyframeNal(nal));
+			this.buffer = Buffer.from(this.buffer.subarray(nextSc));
+		}
+	}
+	/** Find the byte offset of the next start code (00 00 01 or 00 00 00 01) at or after `from`. */
+	findStartCode(from) {
+		const buf = this.buffer;
+		const len = buf.length - 2;
+		for (let i = from; i < len; i++) {
+			if (buf[i + 2] > 1) {
+				i += 2;
+				continue;
+			}
+			if (buf[i] === 0 && buf[i + 1] === 0) {
+				if (buf[i + 2] === 1) return i;
+				if (buf[i + 2] === 0 && i + 3 < buf.length && buf[i + 3] === 1) return i;
+			}
+		}
+		return -1;
+	}
+	/** Check if start code at `pos` is 4-byte (00 00 00 01) vs 3-byte (00 00 01). */
+	isStartCode4(pos) {
+		return pos + 3 < this.buffer.length && this.buffer[pos] === 0 && this.buffer[pos + 1] === 0 && this.buffer[pos + 2] === 0 && this.buffer[pos + 3] === 1;
+	}
+	/** Strip leading start code from a buffer (if present). */
+	stripLeadingStartCode(buf) {
+		if (buf.length >= 4 && buf[0] === 0 && buf[1] === 0 && buf[2] === 0 && buf[3] === 1) return buf.subarray(4);
+		if (buf.length >= 3 && buf[0] === 0 && buf[1] === 0 && buf[2] === 1) return buf.subarray(3);
+		return buf;
+	}
+	/** Check if a NAL unit (without start code) is a keyframe. */
+	isKeyframeNal(nal) {
+		if (nal.length === 0) return false;
+		if (this.codec === "h264") return (nal[0] & 31) === 5;
+		const type = nal[0] >> 1 & 63;
+		return type >= 16 && type <= 21;
+	}
+};
+//#endregion
+//#region src/stream-broker/rtsp/broker-device-id.ts
+/**
+* Parse the numeric deviceId out of a canonical brokerId
+* (`${deviceId}/${camStreamId}` or `${deviceId}/${profile}`). Returns
+* `undefined` for a malformed id — so every device-scoped log line on the RTSP
+* plane can carry `tags.deviceId` without risking a bogus tag value.
+*
+* Every log line about a device carries `tags: { deviceId }`, always the same
+* key and always the numeric id: a miss rate, a media gap or a routing fault is
+* always asked per-camera ("why is 617 worse than 615?"), and a line that omits
+* the tag cannot answer it.
+*/
+function deviceIdFromBrokerId$1(brokerId) {
+	const parsed = Number(brokerId.split("/")[0]);
+	return Number.isFinite(parsed) ? parsed : void 0;
+}
+/**
+* Extract the `m=audio` media block from an ffmpeg-generated SDP, normalised
+* for grafting (port 0, non-colliding payload type, own `a=control`, explicit
+* connection line). Returns `null` when the SDP has no audio media section.
+*/
+function extractAudioMediaBlock(ffmpegSdp, options = {}) {
+	const trackId = options.trackId ?? 1;
+	const lines = ffmpegSdp.split(/\r?\n/);
+	const startIdx = lines.findIndex((l) => l.startsWith("m=audio"));
+	if (startIdx < 0) return null;
+	const block = [];
+	for (let i = startIdx; i < lines.length; i++) {
+		const line = lines[i];
+		if (i > startIdx && line.startsWith("m=")) break;
+		if (line.trim().length === 0) continue;
+		if (line.startsWith("a=control:")) continue;
+		block.push(line);
+	}
+	const mParts = block[0].split(/\s+/);
+	const origPt = Number.parseInt(mParts[3] ?? "", 10);
+	if (!Number.isInteger(origPt)) return null;
+	const newPt = origPt >= 96 ? 97 : origPt;
+	const remapped = block.map((line) => {
+		if (line.startsWith("m=audio")) return `m=audio 0 ${mParts.slice(2).join(" ").replace(String(origPt), String(newPt))}`;
+		if (origPt !== newPt && (line.startsWith("a=rtpmap:") || line.startsWith("a=fmtp:"))) return line.replace(`:${origPt} `, `:${newPt} `);
+		return line;
+	});
+	if (!remapped.some((l) => l.startsWith("c="))) remapped.splice(1, 0, "c=IN IP4 0.0.0.0");
+	remapped.push(`a=control:trackID=${trackId}`);
+	return remapped.join("\r\n");
+}
+/**
+* Append a grafted audio media block to a video-only SDP, preserving CRLF
+* framing and the trailing blank line.
+*/
+function appendAudioMediaBlock(videoSdp, audioBlock) {
+	return `${videoSdp.replace(/[\r\n]+$/, "")}\r\n${audioBlock}\r\n`;
+}
+//#endregion
 //#region src/stream-broker/rtsp/rtsp-types.ts
 /** RTSP method constants. */
 var RTSP_METHODS = [
@@ -1127,129 +1396,45 @@ var RtpPacketizer = class {
 			packets.push(this.buildRtpPacket(fragment, timestamp, isLast && marker));
 			offset = end;
 		}
-		return packets;
-	}
-	/** H.265 FU fragmentation (RFC 7798 Section 4.4.3). */
-	fragmentH265FU(nal, timestamp, marker) {
-		const nalType = nal[0] >> 1 & 63;
-		const tidLayerId = nal[1];
-		const payload = nal.subarray(2);
-		const maxFragment = RTP_MAX_PAYLOAD - 3;
-		const packets = [];
-		let offset = 0;
-		while (offset < payload.length) {
-			const end = Math.min(offset + maxFragment, payload.length);
-			const isFirst = offset === 0;
-			const isLast = end === payload.length;
-			const payloadHdr0 = 98 | nal[0] & 129;
-			const payloadHdr1 = tidLayerId;
-			const fuHeader = (isFirst ? 128 : 0) | (isLast ? 64 : 0) | nalType;
-			const fragment = Buffer.allocUnsafe(3 + (end - offset));
-			fragment[0] = payloadHdr0;
-			fragment[1] = payloadHdr1;
-			fragment[2] = fuHeader;
-			payload.copy(fragment, 3, offset, end);
-			packets.push(this.buildRtpPacket(fragment, timestamp, isLast && marker));
-			offset = end;
-		}
-		return packets;
-	}
-	buildRtpPacket(payload, timestamp, marker) {
-		const header = Buffer.allocUnsafe(12);
-		header[0] = 128;
-		header[1] = (marker ? 128 : 0) | this.payloadType & 127;
-		header.writeUInt16BE(this.sequenceNumber & 65535, 2);
-		this.sequenceNumber = this.sequenceNumber + 1 & 65535;
-		header.writeUInt32BE(timestamp >>> 0, 4);
-		header.writeUInt32BE(this.ssrc >>> 0, 8);
-		return {
-			data: Buffer.concat([header, payload]),
-			marker
-		};
-	}
-};
-//#endregion
-//#region src/stream-broker/rtsp/annexb-deframer.ts
-/**
-* Annex-B NAL unit deframer.
-*
-* ffmpeg outputs Annex-B via stdout in arbitrary chunk sizes — a single
-* `data` event may contain partial NALs, multiple NALs, or NALs split
-* across events. This class accumulates bytes and emits complete NAL units.
-*
-* Usage:
-*   const deframer = new AnnexBDeframer((nal, isKeyframe) => { ... })
-*   proc.stdout.on('data', chunk => deframer.push(chunk))
-*/
-var AnnexBDeframer = class {
-	onNal;
-	buffer = Buffer.alloc(0);
-	codec;
-	constructor(codec, onNal) {
-		this.onNal = onNal;
-		this.codec = codec;
-	}
-	/** Push raw Annex-B bytes. Complete NAL units are emitted via onNal callback. */
-	push(chunk) {
-		this.buffer = this.buffer.length > 0 ? Buffer.concat([this.buffer, chunk]) : chunk;
-		this.drain();
-	}
-	/** Flush remaining buffer (call on stream end). */
-	flush() {
-		if (this.buffer.length > 0) {
-			const nal = this.stripLeadingStartCode(this.buffer);
-			if (nal.length > 0) this.onNal(nal, this.isKeyframeNal(nal));
-			this.buffer = Buffer.alloc(0);
-		}
-	}
-	/** Extract complete NAL units from the buffer, leaving partial data for next push. */
-	drain() {
-		while (true) {
-			const firstSc = this.findStartCode(0);
-			if (firstSc < 0) return;
-			const nalStart = firstSc + (this.isStartCode4(firstSc) ? 4 : 3);
-			const nextSc = this.findStartCode(nalStart);
-			if (nextSc < 0) {
-				if (firstSc > 0) this.buffer = Buffer.from(this.buffer.subarray(firstSc));
-				return;
-			}
-			const nal = this.buffer.subarray(nalStart, nextSc);
-			if (nal.length > 0) this.onNal(nal, this.isKeyframeNal(nal));
-			this.buffer = Buffer.from(this.buffer.subarray(nextSc));
-		}
+		return packets;
 	}
-	/** Find the byte offset of the next start code (00 00 01 or 00 00 00 01) at or after `from`. */
-	findStartCode(from) {
-		const buf = this.buffer;
-		const len = buf.length - 2;
-		for (let i = from; i < len; i++) {
-			if (buf[i + 2] > 1) {
-				i += 2;
-				continue;
-			}
-			if (buf[i] === 0 && buf[i + 1] === 0) {
-				if (buf[i + 2] === 1) return i;
-				if (buf[i + 2] === 0 && i + 3 < buf.length && buf[i + 3] === 1) return i;
-			}
+	/** H.265 FU fragmentation (RFC 7798 Section 4.4.3). */
+	fragmentH265FU(nal, timestamp, marker) {
+		const nalType = nal[0] >> 1 & 63;
+		const tidLayerId = nal[1];
+		const payload = nal.subarray(2);
+		const maxFragment = RTP_MAX_PAYLOAD - 3;
+		const packets = [];
+		let offset = 0;
+		while (offset < payload.length) {
+			const end = Math.min(offset + maxFragment, payload.length);
+			const isFirst = offset === 0;
+			const isLast = end === payload.length;
+			const payloadHdr0 = 98 | nal[0] & 129;
+			const payloadHdr1 = tidLayerId;
+			const fuHeader = (isFirst ? 128 : 0) | (isLast ? 64 : 0) | nalType;
+			const fragment = Buffer.allocUnsafe(3 + (end - offset));
+			fragment[0] = payloadHdr0;
+			fragment[1] = payloadHdr1;
+			fragment[2] = fuHeader;
+			payload.copy(fragment, 3, offset, end);
+			packets.push(this.buildRtpPacket(fragment, timestamp, isLast && marker));
+			offset = end;
 		}
-		return -1;
-	}
-	/** Check if start code at `pos` is 4-byte (00 00 00 01) vs 3-byte (00 00 01). */
-	isStartCode4(pos) {
-		return pos + 3 < this.buffer.length && this.buffer[pos] === 0 && this.buffer[pos + 1] === 0 && this.buffer[pos + 2] === 0 && this.buffer[pos + 3] === 1;
-	}
-	/** Strip leading start code from a buffer (if present). */
-	stripLeadingStartCode(buf) {
-		if (buf.length >= 4 && buf[0] === 0 && buf[1] === 0 && buf[2] === 0 && buf[3] === 1) return buf.subarray(4);
-		if (buf.length >= 3 && buf[0] === 0 && buf[1] === 0 && buf[2] === 1) return buf.subarray(3);
-		return buf;
+		return packets;
 	}
-	/** Check if a NAL unit (without start code) is a keyframe. */
-	isKeyframeNal(nal) {
-		if (nal.length === 0) return false;
-		if (this.codec === "h264") return (nal[0] & 31) === 5;
-		const type = nal[0] >> 1 & 63;
-		return type >= 16 && type <= 21;
+	buildRtpPacket(payload, timestamp, marker) {
+		const header = Buffer.allocUnsafe(12);
+		header[0] = 128;
+		header[1] = (marker ? 128 : 0) | this.payloadType & 127;
+		header.writeUInt16BE(this.sequenceNumber & 65535, 2);
+		this.sequenceNumber = this.sequenceNumber + 1 & 65535;
+		header.writeUInt32BE(timestamp >>> 0, 4);
+		header.writeUInt32BE(this.ssrc >>> 0, 8);
+		return {
+			data: Buffer.concat([header, payload]),
+			marker
+		};
 	}
 };
 //#endregion
@@ -1287,191 +1472,6 @@ function buildSdp(options) {
 	lines.push("");
 	return lines.join("\r\n");
 }
-/**
-* Extract the `m=audio` media block from an ffmpeg-generated SDP, normalised
-* for grafting (port 0, non-colliding payload type, own `a=control`, explicit
-* connection line). Returns `null` when the SDP has no audio media section.
-*/
-function extractAudioMediaBlock(ffmpegSdp, options = {}) {
-	const trackId = options.trackId ?? 1;
-	const lines = ffmpegSdp.split(/\r?\n/);
-	const startIdx = lines.findIndex((l) => l.startsWith("m=audio"));
-	if (startIdx < 0) return null;
-	const block = [];
-	for (let i = startIdx; i < lines.length; i++) {
-		const line = lines[i];
-		if (i > startIdx && line.startsWith("m=")) break;
-		if (line.trim().length === 0) continue;
-		if (line.startsWith("a=control:")) continue;
-		block.push(line);
-	}
-	const mParts = block[0].split(/\s+/);
-	const origPt = Number.parseInt(mParts[3] ?? "", 10);
-	if (!Number.isInteger(origPt)) return null;
-	const newPt = origPt >= 96 ? 97 : origPt;
-	const remapped = block.map((line) => {
-		if (line.startsWith("m=audio")) return `m=audio 0 ${mParts.slice(2).join(" ").replace(String(origPt), String(newPt))}`;
-		if (origPt !== newPt && (line.startsWith("a=rtpmap:") || line.startsWith("a=fmtp:"))) return line.replace(`:${origPt} `, `:${newPt} `);
-		return line;
-	});
-	if (!remapped.some((l) => l.startsWith("c="))) remapped.splice(1, 0, "c=IN IP4 0.0.0.0");
-	remapped.push(`a=control:trackID=${trackId}`);
-	return remapped.join("\r\n");
-}
-/**
-* Append a grafted audio media block to a video-only SDP, preserving CRLF
-* framing and the trailing blank line.
-*/
-function appendAudioMediaBlock(videoSdp, audioBlock) {
-	return `${videoSdp.replace(/[\r\n]+$/, "")}\r\n${audioBlock}\r\n`;
-}
-//#endregion
-//#region src/stream-broker/rtsp/broker-device-id.ts
-/**
-* Parse the numeric deviceId out of a canonical brokerId
-* (`${deviceId}/${camStreamId}` or `${deviceId}/${profile}`). Returns
-* `undefined` for a malformed id — so every device-scoped log line on the RTSP
-* plane can carry `tags.deviceId` without risking a bogus tag value.
-*
-* Every log line about a device carries `tags: { deviceId }`, always the same
-* key and always the numeric id: a miss rate, a media gap or a routing fault is
-* always asked per-camera ("why is 617 worse than 615?"), and a line that omits
-* the tag cannot answer it.
-*/
-function deviceIdFromBrokerId$1(brokerId) {
-	const parsed = Number(brokerId.split("/")[0]);
-	return Number.isFinite(parsed) ? parsed : void 0;
-}
-//#endregion
-//#region src/stream-broker/paced-rtp-replay.ts
-var PACED_REPLAY_DEFAULTS = {
-	singleShotMax: 64,
-	tickMs: 5,
-	targetDurationMs: 300,
-	minChunkSize: 32,
-	maxChunkSize: 128
-};
-var defaultSchedule = (fn, ms) => {
-	const t = setTimeout(fn, ms);
-	return () => clearTimeout(t);
-};
-/**
-* Compute the per-tick chunk size for a burst of `total` packets: enough to
-* drain within `targetDurationMs` at one chunk per `tickMs`, clamped to
-* [minChunkSize, maxChunkSize]. Bursts ≤ `singleShotMax` are one chunk.
-*/
-function computeReplayChunkSize(total, tuning) {
-	if (total <= tuning.singleShotMax) return Math.max(total, 1);
-	const maxTicks = Math.max(1, Math.floor(tuning.targetDurationMs / tuning.tickMs));
-	const ideal = Math.ceil(total / maxTicks);
-	return Math.min(tuning.maxChunkSize, Math.max(tuning.minChunkSize, ideal));
-}
-/**
-* One in-flight paced replay. Create per burst; not reusable after
-* completion or abort.
-*/
-var PacedRtpReplay = class {
-	send;
-	onDone;
-	tuning;
-	schedule;
-	now;
-	queue = [];
-	cancelTimer = null;
-	started = false;
-	finished = false;
-	startedAt = 0;
-	chunkSize = 0;
-	initialPackets = 0;
-	enqueuedLive = 0;
-	sentCount = 0;
-	chunkCount = 0;
-	constructor(send, onDone, options) {
-		this.send = send;
-		this.onDone = onDone;
-		this.tuning = {
-			...PACED_REPLAY_DEFAULTS,
-			...options?.tuning
-		};
-		this.schedule = options?.schedule ?? defaultSchedule;
-		this.now = options?.now ?? Date.now;
-	}
-	/** True while packets remain to be drained (live packets must be enqueued,
-	*  not sent directly, to preserve order). */
-	get active() {
-		return this.started && !this.finished;
-	}
-	/**
-	* Begin the replay: sends the first chunk synchronously, then paces the
-	* rest. Calling `start` more than once is a no-op.
-	*/
-	start(initial) {
-		if (this.started) return;
-		this.started = true;
-		this.startedAt = this.now();
-		this.initialPackets = initial.length;
-		this.queue = [...initial];
-		this.chunkSize = computeReplayChunkSize(initial.length, this.tuning);
-		this.drainChunk();
-	}
-	/**
-	* Append a live packet behind the still-queued replay tail. No-op (returns
-	* false) when the replay is not active — the caller must then send the
-	* packet directly.
-	*/
-	enqueue(item) {
-		if (!this.active) return false;
-		this.queue.push(item);
-		this.enqueuedLive++;
-		return true;
-	}
-	/** Cancel the replay: pending packets are dropped, `onDone` fires with
-	*  `aborted: true`. Safe to call multiple times / before `start`. */
-	abort() {
-		if (this.finished) return;
-		if (!this.started) {
-			this.started = true;
-			this.startedAt = this.now();
-		}
-		this.finish(true);
-	}
-	drainChunk() {
-		if (this.finished) return;
-		this.chunkCount++;
-		const n = Math.min(this.chunkSize, this.queue.length);
-		for (let i = 0; i < n; i++) {
-			if (this.finished) return;
-			if (this.send(this.queue[i])) this.sentCount++;
-		}
-		this.queue = this.queue.slice(n);
-		if (this.queue.length === 0) {
-			this.finish(false);
-			return;
-		}
-		this.cancelTimer = this.schedule(() => {
-			this.cancelTimer = null;
-			this.drainChunk();
-		}, this.tuning.tickMs);
-	}
-	finish(aborted) {
-		if (this.finished) return;
-		this.finished = true;
-		if (this.cancelTimer) {
-			this.cancelTimer();
-			this.cancelTimer = null;
-		}
-		this.queue = [];
-		this.onDone({
-			packets: this.initialPackets,
-			enqueuedLive: this.enqueuedLive,
-			sent: this.sentCount,
-			chunks: this.chunkCount,
-			chunkSize: this.chunkSize,
-			durationMs: this.now() - this.startedAt,
-			aborted
-		});
-	}
-};
 //#endregion
 //#region src/stream-broker/rtsp/rtsp-restreamer.ts
 /**
@@ -1637,6 +1637,20 @@ var RtspRestreamer = class {
 	/** Throttled drop/withhold accounting — see {@link reportDrops}. */
 	dropsWithheldSpan = 0;
 	dropsWithheldPackets = 0;
+	/**
+	* Muted (local-decoder loopback) joins refused the ring burst because it was
+	* not near-live — see {@link LIVE_EDGE_MAX_SPAN_MS}. Its own counter: this is
+	* a different rule from the span/packet bound, it fires on a DIFFERENT
+	* population (only decode sessions), and conflating it with
+	* {@link dropsWithheldSpan} would hide that a camera's decoder starts blind
+	* on nearly every dial while its viewers are served normally.
+	*
+	* Counts withhold DECISIONS, not joins — `servePendingSessionsRtp` runs per
+	* packet, so one session withheld for a whole GOP contributes many. Same
+	* semantics as {@link dropsWithheldSpan}; the signal to read is "nonzero on
+	* THIS camera while its viewers are being served".
+	*/
+	dropsWithheldMutedLiveEdge = 0;
 	dropsPrimeBacklog = 0;
 	dropsLiveLatched = 0;
 	dropsLiveClosed = 0;
@@ -1905,7 +1919,11 @@ var RtspRestreamer = class {
 			const session = this.sessions.get(sessionId);
 			if (!session) continue;
 			if (session.isMuted()) {
-				if (!fromRing || spanMs > LIVE_EDGE_MAX_SPAN_MS) continue;
+				if (!fromRing || spanMs > LIVE_EDGE_MAX_SPAN_MS) {
+					this.dropsWithheldMutedLiveEdge++;
+					withheld++;
+					continue;
+				}
 			} else if (overSpan || overPackets) {
 				if (overSpan) this.dropsWithheldSpan++;
 				else this.dropsWithheldPackets++;
@@ -2028,6 +2046,8 @@ var RtspRestreamer = class {
 				boundPackets: bound.maxPackets,
 				primeWithheldSpanMs: this.dropsWithheldSpan,
 				primeWithheldPackets: this.dropsWithheldPackets,
+				primeWithheldMutedLiveEdge: this.dropsWithheldMutedLiveEdge,
+				liveEdgeMaxSpanMs: LIVE_EDGE_MAX_SPAN_MS,
 				primeAbortedOnBacklog: this.dropsPrimeBacklog,
 				liveDroppedUntilKeyframe: this.dropsLiveLatched,
 				clientsClosedOnBacklog: this.dropsLiveClosed,
@@ -2036,6 +2056,7 @@ var RtspRestreamer = class {
 		});
 		this.dropsWithheldSpan = 0;
 		this.dropsWithheldPackets = 0;
+		this.dropsWithheldMutedLiveEdge = 0;
 		this.dropsPrimeBacklog = 0;
 		this.dropsLiveLatched = 0;
 		this.dropsLiveClosed = 0;
@@ -13060,7 +13081,7 @@ var TranscodePipelineManager = class {
 				}
 			});
 			const passVideoCodec = sourceCodec ?? "H264";
-			const url = this.resolveSourceUrl(input.deviceId, sourceStream.camStreamId);
+			const url = this.resolveSourceUrl(input.deviceId, sourceStream.camStreamId, wantedAudio === "none");
 			const key = pipelineKeyFor(input.deviceId, sourceStream.camStreamId, "copy", wantedAudio, outResolution, outputArgs, false);
 			return this.shareOrCreatePassthrough(key, () => ({
 				url,
@@ -13216,10 +13237,11 @@ var TranscodePipelineManager = class {
 	* source dial. Throws when no restream entry exists — a targeted source
 	* must be dialed before it can be consumed.
 	*/
-	resolveSourceUrl(deviceId, camStreamId) {
+	resolveSourceUrl(deviceId, camStreamId, videoOnly = false) {
 		const brokerId = makeSourceBrokerId(deviceId, camStreamId);
 		const entry = this.rtspEntryLookup(brokerId);
-		if (entry?.url) return entry.url;
+		const url = videoOnly ? entry?.mutedUrl ?? entry?.url : entry?.url;
+		if (url) return url;
 		throw new Error(`getStreamWithCodec: no broker restream for ${brokerId} (source not dialed)`);
 	}
 	/**
@@ -13836,7 +13858,10 @@ var StreamBrokerManager = class StreamBrokerManager {
 			cameraStreamLookup: (deviceId) => this.getCameraStreamsForDevice(deviceId),
 			rtspEntryLookup: (brokerId) => {
 				const entry = this.rtspProvider.getEntry(brokerId);
-				return entry ? { url: entry.url } : null;
+				return entry ? {
+					url: entry.url,
+					mutedUrl: entry.mutedUrl
+				} : null;
 			},
 			resolveProfileSource: (deviceId, profile) => this.assignments.get(deviceId)?.map[profile] ?? null,
 			ffmpegConfig: () => this.ffmpegConfig,