werift 0.24.4 → 0.25.0

package/lib/webrtc/src/media/codec.js.map

@@ -1 +1 @@
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{ channels: codec.channels } : {}),\n    ...(codec.payloadType != undefined\n      ? { payloadType: codec.payloadType }\n      : {}),\n    rtcpFeedback: codec.rtcpFeedback.map((feedback) => ({ ...feedback })),\n    ...(codec.parameters != undefined ? { parameters: codec.parameters } : {}),\n    direction: codec.direction,\n  });\n}\n\nexport const supportedCodecs = [\n  useAV1X(),\n  useVP9(),\n  useVP8(),\n  useH264(),\n  useOPUS(),\n  usePCMU(),\n].map((codec) => codec.mimeType);\n\nexport const supportedVideoCodecs = supportedCodecs.filter((codec) =>\n  codec.toLowerCase().startsWith(\"video/\"),\n);\n\nexport const supportedAudioCodecs = supportedCodecs.filter((codec) =>\n  codec.toLowerCase().startsWith(\"audio/\"),\n);\n"]}

package/lib/webrtc/src/media/codecCompatibility.d.ts

@@ -0,0 +1,23 @@
+import type { Kind } from "../types/domain";
+import type { RTCRtpCodecParameters } from "./parameters";
+export declare function isCodecCompatible(source: RTCRtpCodecParameters, configured: RTCRtpCodecParameters): boolean;
+/**
+ * remote track が実際に配信する codec を返す。
+ * audio の RED は RTCRtpReceiver が primary へ展開してから track へ渡すため、
+ * RED が参照する primary codec (不明なら先頭の primary) を track の codec とする。
+ */
+export declare function deliveredTrackCodec(kind: string, codec: RTCRtpCodecParameters | undefined, negotiated: readonly RTCRtpCodecParameters[]): RTCRtpCodecParameters | undefined;
+/**
+ * RTX / RED を除く primary codec に MIME 一致が 1 つでもあるか。
+ * remote m-line を local capability で扱えるかの判定 (Issue #705) に使う。
+ */
+export declare function hasCommonPrimaryMimeType(local: readonly RTCRtpCodecParameters[], remote: readonly RTCRtpCodecParameters[]): boolean;
+export declare function intersectCodecs(configured: RTCRtpCodecParameters[], source: readonly RTCRtpCodecParameters[]): RTCRtpCodecParameters[];
+export declare function applyCodecPreferences(configured: RTCRtpCodecParameters[], preferences: readonly RTCRtpCodecParameters[] | undefined): RTCRtpCodecParameters[];
+export declare function resolveCodecs(configured: RTCRtpCodecParameters[], source: readonly RTCRtpCodecParameters[] | undefined, preferences: readonly RTCRtpCodecParameters[] | undefined): RTCRtpCodecParameters[];
+export declare function assertCodecsSupported(options: {
+    kind: Kind;
+    configured: RTCRtpCodecParameters[];
+    source: readonly RTCRtpCodecParameters[] | undefined;
+    preferences: readonly RTCRtpCodecParameters[] | undefined;
+}): RTCRtpCodecParameters[];

package/lib/webrtc/src/media/codecCompatibility.js

@@ -0,0 +1,138 @@
+"use strict";
+Object.defineProperty(exports, "__esModule", { value: true });
+exports.isCodecCompatible = isCodecCompatible;
+exports.deliveredTrackCodec = deliveredTrackCodec;
+exports.hasCommonPrimaryMimeType = hasCommonPrimaryMimeType;
+exports.intersectCodecs = intersectCodecs;
+exports.applyCodecPreferences = applyCodecPreferences;
+exports.resolveCodecs = resolveCodecs;
+exports.assertCodecsSupported = assertCodecsSupported;
+const errors_1 = require("../errors");
+const sdp_1 = require("../sdp");
+function codecName(codec) {
+    return codec.name.toLowerCase();
+}
+function optionalEqual(left, right) {
+    return left == undefined || right == undefined || left === right;
+}
+function h264Parameters(codec) {
+    const parameters = (0, sdp_1.codecParametersFromString)(codec.parameters ?? "");
+    return {
+        packetizationMode: parameters["packetization-mode"],
+        profileLevelId: parameters["profile-level-id"]?.toString().toLowerCase(),
+    };
+}
+function isCodecCompatible(source, configured) {
+    if (source.mimeType.toLowerCase() !== configured.mimeType.toLowerCase()) {
+        return false;
+    }
+    if (source.clockRate !== configured.clockRate)
+        return false;
+    if (!optionalEqual(source.channels, configured.channels))
+        return false;
+    if (codecName(source) !== "h264")
+        return true;
+    const sourceParameters = h264Parameters(source);
+    const configuredParameters = h264Parameters(configured);
+    if (!optionalEqual(sourceParameters.packetizationMode, configuredParameters.packetizationMode)) {
+        return false;
+    }
+    return optionalEqual(sourceParameters.profileLevelId, configuredParameters.profileLevelId);
+}
+function isAuxiliary(codec) {
+    return ["rtx", "red"].includes(codecName(codec));
+}
+/**
+ * remote track が実際に配信する codec を返す。
+ * audio の RED は RTCRtpReceiver が primary へ展開してから track へ渡すため、
+ * RED が参照する primary codec (不明なら先頭の primary) を track の codec とする。
+ */
+function deliveredTrackCodec(kind, codec, negotiated) {
+    if (kind !== "audio" || codec == undefined || codecName(codec) !== "red") {
+        return codec;
+    }
+    const [primaryPayloadType] = referencedPayloadTypes(codec);
+    return (negotiated.find((candidate) => !isAuxiliary(candidate) && candidate.payloadType === primaryPayloadType) ??
+        negotiated.find((candidate) => !isAuxiliary(candidate)) ??
+        codec);
+}
+/**
+ * RTX / RED を除く primary codec に MIME 一致が 1 つでもあるか。
+ * remote m-line を local capability で扱えるかの判定 (Issue #705) に使う。
+ */
+function hasCommonPrimaryMimeType(local, remote) {
+    return local.some((localCodec) => !isAuxiliary(localCodec) &&
+        remote.some((remoteCodec) => remoteCodec.mimeType.toLowerCase() ===
+            localCodec.mimeType.toLowerCase()));
+}
+function referencedPayloadTypes(codec) {
+    const parameters = codec.parameters ?? "";
+    if (codecName(codec) === "rtx") {
+        const apt = Number((0, sdp_1.codecParametersFromString)(parameters).apt);
+        return Number.isNaN(apt) ? [] : [apt];
+    }
+    if (codecName(codec) === "red") {
+        return parameters
+            .split("/")
+            .map(Number)
+            .filter((value) => !Number.isNaN(value));
+    }
+    return [];
+}
+function withAuxiliaryCodecs(configured, primary) {
+    const keptPayloadTypes = new Set(primary.map((codec) => codec.payloadType));
+    const auxiliary = configured.filter((codec) => {
+        if (!isAuxiliary(codec))
+            return false;
+        const referenced = referencedPayloadTypes(codec);
+        return (referenced.length > 0 &&
+            referenced.every((payloadType) => keptPayloadTypes.has(payloadType)));
+    });
+    return configured.filter((codec) => primary.includes(codec) || auxiliary.includes(codec));
+}
+function intersectCodecs(configured, source) {
+    const primary = configured.filter((codec) => !isAuxiliary(codec) &&
+        source.some((sourceCodec) => isCodecCompatible(sourceCodec, codec)));
+    return withAuxiliaryCodecs(configured, primary);
+}
+function matchesPreference(candidate, preference) {
+    return (candidate.mimeType.toLowerCase() === preference.mimeType.toLowerCase() &&
+        optionalEqual(candidate.clockRate, preference.clockRate));
+}
+function applyCodecPreferences(configured, preferences) {
+    if (preferences == undefined)
+        return configured;
+    const preferred = preferences.flatMap((preference) => {
+        const match = configured.find((codec) => matchesPreference(codec, preference));
+        return match ? [match] : [];
+    });
+    const uniquePreferred = [...new Set(preferred)];
+    const referencedByPreferredAuxiliary = new Set(uniquePreferred.filter(isAuxiliary).flatMap(referencedPayloadTypes));
+    const primary = configured.filter((codec) => !isAuxiliary(codec) &&
+        (uniquePreferred.includes(codec) ||
+            referencedByPreferredAuxiliary.has(codec.payloadType)));
+    const allowed = withAuxiliaryCodecs(configured, primary);
+    return [
+        ...uniquePreferred.filter((codec) => allowed.includes(codec)),
+        ...allowed.filter((codec) => !uniquePreferred.includes(codec)),
+    ];
+}
+function resolveCodecs(configured, source, preferences) {
+    return applyCodecPreferences(source == undefined ? configured : intersectCodecs(configured, source), preferences);
+}
+function list(codecs) {
+    return codecs?.map((codec) => codec.mimeType).join(", ") || "(none)";
+}
+function assertCodecsSupported(options) {
+    const { kind, configured, source, preferences } = options;
+    const sourceCompatible = source == undefined ? configured : intersectCodecs(configured, source);
+    if (source != undefined && sourceCompatible.length === 0) {
+        throw (0, errors_1.createWebRtcDomException)("NotSupportedError", `Track codec ${list(source)} is not supported by this RTCPeerConnection. Configured ${kind} codecs: ${list(configured)}`);
+    }
+    const effective = applyCodecPreferences(sourceCompatible, preferences);
+    if (effective.length === 0) {
+        throw (0, errors_1.createWebRtcDomException)("NotSupportedError", `No codec remains after applying codec preferences. Source codecs: ${list(source)} Configured codecs: ${list(configured)} Preferred codecs: ${list(preferences)}`);
+    }
+    return effective;
+}
+//# sourceMappingURL=codecCompatibility.js.map

package/lib/webrtc/src/media/codecCompatibility.js.map

@@ -0,0 +1 @@
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{ createWebRtcDomException } from \"../errors\";\nimport { codecParametersFromString } from \"../sdp\";\nimport type { Kind } from \"../types/domain\";\nimport type { RTCRtpCodecParameters } from \"./parameters\";\n\nfunction codecName(codec: RTCRtpCodecParameters) {\n  return codec.name.toLowerCase();\n}\n\nfunction optionalEqual<T>(left: T | undefined, right: T | undefined) {\n  return left == undefined || right == undefined || left === right;\n}\n\nfunction h264Parameters(codec: RTCRtpCodecParameters) {\n  const parameters = codecParametersFromString(codec.parameters ?? \"\");\n  return {\n    packetizationMode: parameters[\"packetization-mode\"],\n    profileLevelId: parameters[\"profile-level-id\"]?.toString().toLowerCase(),\n  };\n}\n\nexport function isCodecCompatible(\n  source: RTCRtpCodecParameters,\n  configured: RTCRtpCodecParameters,\n) {\n  if (source.mimeType.toLowerCase() !== configured.mimeType.toLowerCase()) {\n    return false;\n  }\n  if (source.clockRate !== configured.clockRate) return false;\n  if (!optionalEqual(source.channels, configured.channels)) return false;\n  if (codecName(source) !== \"h264\") return true;\n\n  const sourceParameters = h264Parameters(source);\n  const configuredParameters = h264Parameters(configured);\n  if (\n    !optionalEqual(\n      sourceParameters.packetizationMode,\n      configuredParameters.packetizationMode,\n    )\n  ) {\n    return false;\n  }\n  return optionalEqual(\n    sourceParameters.profileLevelId,\n    configuredParameters.profileLevelId,\n  );\n}\n\nfunction isAuxiliary(codec: RTCRtpCodecParameters) {\n  return [\"rtx\", \"red\"].includes(codecName(codec));\n}\n\n/**\n * remote track が実際に配信する codec を返す。\n * audio の RED は RTCRtpReceiver が primary へ展開してから track へ渡すため、\n * RED が参照する primary codec (不明なら先頭の primary) を track の codec とする。\n */\nexport function deliveredTrackCodec(\n  kind: string,\n  codec: RTCRtpCodecParameters | undefined,\n  negotiated: readonly RTCRtpCodecParameters[],\n) {\n  if (kind !== \"audio\" || codec == undefined || codecName(codec) !== \"red\") {\n    return codec;\n  }\n  const [primaryPayloadType] = referencedPayloadTypes(codec);\n  return (\n    negotiated.find(\n      (candidate) =>\n        !isAuxiliary(candidate) && candidate.payloadType === primaryPayloadType,\n    ) ??\n    negotiated.find((candidate) => !isAuxiliary(candidate)) ??\n    codec\n  );\n}\n\n/**\n * RTX / RED を除く primary codec に MIME 一致が 1 つでもあるか。\n * remote m-line を local capability で扱えるかの判定 (Issue #705) に使う。\n */\nexport function hasCommonPrimaryMimeType(\n  local: readonly RTCRtpCodecParameters[],\n  remote: readonly RTCRtpCodecParameters[],\n) {\n  return local.some(\n    (localCodec) =>\n      !isAuxiliary(localCodec) &&\n      remote.some(\n        (remoteCodec) =>\n          remoteCodec.mimeType.toLowerCase() ===\n          localCodec.mimeType.toLowerCase(),\n      ),\n  );\n}\n\nfunction referencedPayloadTypes(codec: RTCRtpCodecParameters) {\n  const parameters = codec.parameters ?? \"\";\n  if (codecName(codec) === \"rtx\") {\n    const apt = Number(codecParametersFromString(parameters).apt);\n    return Number.isNaN(apt) ? [] : [apt];\n  }\n  if (codecName(codec) === \"red\") {\n    return parameters\n      .split(\"/\")\n      .map(Number)\n      .filter((value) => !Number.isNaN(value));\n  }\n  return [];\n}\n\nfunction withAuxiliaryCodecs(\n  configured: RTCRtpCodecParameters[],\n  primary: RTCRtpCodecParameters[],\n) {\n  const keptPayloadTypes = new Set(primary.map((codec) => codec.payloadType));\n  const auxiliary = configured.filter((codec) => {\n    if (!isAuxiliary(codec)) return false;\n    const referenced = referencedPayloadTypes(codec);\n    return (\n      referenced.length > 0 &&\n      referenced.every((payloadType) => keptPayloadTypes.has(payloadType))\n    );\n  });\n  return configured.filter(\n    (codec) => primary.includes(codec) || auxiliary.includes(codec),\n  );\n}\n\nexport function intersectCodecs(\n  configured: RTCRtpCodecParameters[],\n  source: readonly RTCRtpCodecParameters[],\n) {\n  const primary = configured.filter(\n    (codec) =>\n      !isAuxiliary(codec) &&\n      source.some((sourceCodec) => isCodecCompatible(sourceCodec, codec)),\n  );\n  return withAuxiliaryCodecs(configured, primary);\n}\n\nfunction matchesPreference(\n  candidate: RTCRtpCodecParameters,\n  preference: RTCRtpCodecParameters,\n) {\n  return (\n    candidate.mimeType.toLowerCase() === preference.mimeType.toLowerCase() &&\n    optionalEqual(candidate.clockRate, preference.clockRate)\n  );\n}\n\nexport function applyCodecPreferences(\n  configured: RTCRtpCodecParameters[],\n  preferences: readonly RTCRtpCodecParameters[] | undefined,\n) {\n  if (preferences == undefined) return configured;\n  const preferred = preferences.flatMap((preference) => {\n    const match = configured.find((codec) =>\n      matchesPreference(codec, preference),\n    );\n    return match ? [match] : [];\n  });\n  const uniquePreferred = [...new Set(preferred)];\n  const referencedByPreferredAuxiliary = new Set(\n    uniquePreferred.filter(isAuxiliary).flatMap(referencedPayloadTypes),\n  );\n  const primary = configured.filter(\n    (codec) =>\n      !isAuxiliary(codec) &&\n      (uniquePreferred.includes(codec) ||\n        referencedByPreferredAuxiliary.has(codec.payloadType)),\n  );\n  const allowed = withAuxiliaryCodecs(configured, primary);\n  return [\n    ...uniquePreferred.filter((codec) => allowed.includes(codec)),\n    ...allowed.filter((codec) => !uniquePreferred.includes(codec)),\n  ];\n}\n\nexport function resolveCodecs(\n  configured: RTCRtpCodecParameters[],\n  source: readonly RTCRtpCodecParameters[] | undefined,\n  preferences: readonly RTCRtpCodecParameters[] | undefined,\n) {\n  return applyCodecPreferences(\n    source == undefined ? configured : intersectCodecs(configured, source),\n    preferences,\n  );\n}\n\nfunction list(codecs: readonly RTCRtpCodecParameters[] | undefined) {\n  return codecs?.map((codec) => codec.mimeType).join(\", \") || \"(none)\";\n}\n\nexport function assertCodecsSupported(options: {\n  kind: Kind;\n  configured: RTCRtpCodecParameters[];\n  source: readonly RTCRtpCodecParameters[] | undefined;\n  preferences: readonly RTCRtpCodecParameters[] | undefined;\n}) {\n  const { kind, configured, source, preferences } = options;\n  const sourceCompatible =\n    source == undefined ? configured : intersectCodecs(configured, source);\n  if (source != undefined && sourceCompatible.length === 0) {\n    throw createWebRtcDomException(\n      \"NotSupportedError\",\n      `Track codec ${list(source)} is not supported by this RTCPeerConnection. Configured ${kind} codecs: ${list(configured)}`,\n    );\n  }\n  const effective = applyCodecPreferences(sourceCompatible, preferences);\n  if (effective.length === 0) {\n    throw createWebRtcDomException(\n      \"NotSupportedError\",\n      `No codec remains after applying codec preferences. Source codecs: ${list(source)} Configured codecs: ${list(configured)} Preferred codecs: ${list(preferences)}`,\n    );\n  }\n  return effective;\n}\n"]}

package/lib/webrtc/src/media/index.d.ts

@@ -1,4 +1,5 @@
 export * from "./codec";
+export * from "./codecCompatibility";
 export * from "./extension/rtcpFeedback";
 export * from "./extension/rtpExtension";
 export * from "./parameters";

package/lib/webrtc/src/media/index.js

@@ -15,6 +15,7 @@ var __exportStar = (this && this.__exportStar) || function(m, exports) {
 };
 Object.defineProperty(exports, "__esModule", { value: true });
 __exportStar(require("./codec"), exports);
+__exportStar(require("./codecCompatibility"), exports);
 __exportStar(require("./extension/rtcpFeedback"), exports);
 __exportStar(require("./extension/rtpExtension"), exports);
 __exportStar(require("./parameters"), exports);

package/lib/webrtc/src/media/index.js.map

@@ -1 +1 @@
-{"version":3,"file":"index.js","sourceRoot":"","sources":["../../../../src/media/index.ts"],"names":[],"mappings":";;;;;;;;;;;;;;;;AAAA,0CAAwB;AACxB,2DAAyC;AACzC,2DAAyC;AACzC,+CAA6B;AAC7B,2CAAyB;AACzB,gDAA8B;AAC9B,8CAA4B;AAC5B,mDAAiC;AACjC,0CAAwB;AACxB,wDAAsC;AACtC,0CAAwB","sourcesContent":["export * from \"./codec\";\nexport * from \"./extension/rtcpFeedback\";\nexport * from \"./extension/rtpExtension\";\nexport * from \"./parameters\";\nexport * from \"./router\";\nexport * from \"./rtpReceiver\";\nexport * from \"./rtpSender\";\nexport * from \"./rtpTransceiver\";\nexport * from \"./stats\";\nexport * from \"../transceiverManager\";\nexport * from \"./track\";\n"]}
+{"version":3,"file":"index.js","sourceRoot":"","sources":["../../../../src/media/index.ts"],"names":[],"mappings":";;;;;;;;;;;;;;;;AAAA,0CAAwB;AACxB,uDAAqC;AACrC,2DAAyC;AACzC,2DAAyC;AACzC,+CAA6B;AAC7B,2CAAyB;AACzB,gDAA8B;AAC9B,8CAA4B;AAC5B,mDAAiC;AACjC,0CAAwB;AACxB,wDAAsC;AACtC,0CAAwB","sourcesContent":["export * from \"./codec\";\nexport * from \"./codecCompatibility\";\nexport * from \"./extension/rtcpFeedback\";\nexport * from \"./extension/rtpExtension\";\nexport * from \"./parameters\";\nexport * from \"./router\";\nexport * from \"./rtpReceiver\";\nexport * from \"./rtpSender\";\nexport * from \"./rtpTransceiver\";\nexport * from \"./stats\";\nexport * from \"../transceiverManager\";\nexport * from \"./track\";\n"]}

package/lib/webrtc/src/media/router.d.ts

@@ -18,6 +18,8 @@ export declare class RtpRouter {
     private registerRtpReceiver;
     registerRtpReceiverBySsrc(transceiver: RTCRtpTransceiver, params: RTCRtpReceiveParameters): void;
     registerRtpReceiverByRid(transceiver: RTCRtpTransceiver, param: RTCRtpSimulcastParameters, params: RTCRtpReceiveParameters): void;
+    /**停止した transceiver の SSRC / RID 登録を解除する */
+    unregisterTransceiver(transceiver: RTCRtpTransceiver): void;
     routeRtp: (packet: RtpPacket) => void;
     routeRtcp: (packet: RtcpPacket) => void;
 }

package/lib/webrtc/src/media/router.js

@@ -2,6 +2,7 @@
 Object.defineProperty(exports, "__esModule", { value: true });
 exports.RtpRouter = void 0;
 const rtp_1 = require("../imports/rtp");
+const codecCompatibility_1 = require("./codecCompatibility");
 const rtpReceiver_1 = require("./rtpReceiver");
 const track_1 = require("./track");
 const log = (0, rtp_1.debug)("werift:packages/webrtc/src/media/router.ts");
@@ -35,8 +36,10 @@ class RtpRouter {
                 const rid = extensions[rtp_1.RTP_EXTENSION_URI.sdesRTPStreamID];
                 if (typeof rid === "string") {
                     rtpReceiver = this.ridTable[rid];
-                    rtpReceiver.latestRid = rid;
-                    rtpReceiver.handleRtpByRid(packet, rid, extensions);
+                    if (rtpReceiver) {
+                        rtpReceiver.latestRid = rid;
+                        rtpReceiver.handleRtpByRid(packet, rid, extensions);
+                    }
                 }
                 else if (rtpReceiver) {
                     rtpReceiver.handleRtpBySsrc(packet, extensions);
@@ -145,7 +148,7 @@ class RtpRouter {
                 kind: transceiver.kind,
                 id: transceiver.sender.trackId,
                 remote: true,
-                codec: params.codecs[i],
+                codec: (0, codecCompatibility_1.deliveredTrackCodec)(transceiver.kind, params.codecs[i], params.codecs),
             }));
             if (encode.rtx) {
                 this.registerRtpReceiver(transceiver.receiver, encode.rtx.ssrc);
@@ -168,6 +171,23 @@ class RtpRouter {
         }));
         this.ridTable[param.rid] = transceiver.receiver;
     }
+    /**停止した transceiver の SSRC / RID 登録を解除する */
+    unregisterTransceiver(transceiver) {
+        const owners = [
+            transceiver.sender,
+            transceiver.receiver,
+        ];
+        for (const [ssrc, owner] of Object.entries(this.ssrcTable)) {
+            if (owners.includes(owner)) {
+                delete this.ssrcTable[Number(ssrc)];
+            }
+        }
+        for (const [rid, owner] of Object.entries(this.ridTable)) {
+            if (owners.includes(owner)) {
+                delete this.ridTable[rid];
+            }
+        }
+    }
 }
 exports.RtpRouter = RtpRouter;
 //# sourceMappingURL=router.js.map

package/lib/webrtc/src/media/router.js.map

@@ -1 +1 @@
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{\n  type Extensions,\n  RTP_EXTENSION_URI,\n  ReceiverEstimatedMaxBitrate,\n  type RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpRrPacket,\n  RtcpSourceDescriptionPacket,\n  RtcpSrPacket,\n  RtcpTransportLayerFeedback,\n  type RtpPacket,\n  debug,\n  rtpHeaderExtensionsParser,\n} from \"../imports/rtp\";\nimport type {\n  RTCRtpReceiveParameters,\n  RTCRtpSimulcastParameters,\n} from \"./parameters\";\nimport { RTCRtpReceiver } from \"./rtpReceiver\";\nimport type { RTCRtpSender } from \"./rtpSender\";\nimport type { RTCRtpTransceiver } from \"./rtpTransceiver\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/router.ts\");\n\nexport class RtpRouter {\n  ssrcTable: { [ssrc: number]: RTCRtpReceiver | RTCRtpSender } = {};\n  ridTable: { [rid: string]: RTCRtpReceiver | RTCRtpSender } = {};\n  extIdUriMap: { [id: number]: string } = {};\n\n  constructor() {}\n\n  registerRtpSender(sender: RTCRtpSender) {\n    this.ssrcTable[sender.ssrc] = sender;\n  }\n\n  private registerRtpReceiver(receiver: RTCRtpReceiver, ssrc: number) {\n    log(\"registerRtpReceiver\", ssrc);\n    this.ssrcTable[ssrc] = receiver;\n  }\n\n  registerRtpReceiverBySsrc(\n    transceiver: RTCRtpTransceiver,\n    params: RTCRtpReceiveParameters,\n  ) {\n    log(\"registerRtpReceiverBySsrc\", params);\n\n    params.encodings\n      .filter((e) => e.ssrc != undefined) // todo fix\n      .forEach((encode, i) => {\n        this.registerRtpReceiver(transceiver.receiver, encode.ssrc);\n        transceiver.addTrack(\n          new MediaStreamTrack({\n            ssrc: encode.ssrc,\n            kind: transceiver.kind,\n            id: transceiver.sender.trackId,\n            remote: true,\n            codec: params.codecs[i],\n          }),\n        );\n        if (encode.rtx) {\n          this.registerRtpReceiver(transceiver.receiver, encode.rtx.ssrc);\n        }\n      });\n\n    params.headerExtensions.forEach((extension) => {\n      this.extIdUriMap[extension.id] = extension.uri;\n    });\n  }\n\n  registerRtpReceiverByRid(\n    transceiver: RTCRtpTransceiver,\n    param: RTCRtpSimulcastParameters,\n    params: RTCRtpReceiveParameters,\n  ) {\n    // サイマルキャスト利用時のRTXをサポートしていないのでcodecs/encodingsは常に一つ\n    const [codec] = params.codecs;\n\n    log(\"registerRtpReceiverByRid\", param);\n    transceiver.addTrack(\n      new MediaStreamTrack({\n        rid: param.rid,\n        kind: transceiver.kind,\n        id: transceiver.sender.trackId,\n        remote: true,\n        codec,\n      }),\n    );\n    this.ridTable[param.rid] = transceiver.receiver;\n  }\n\n  routeRtp = (packet: RtpPacket) => {\n    const extensions: Extensions = rtpHeaderExtensionsParser(\n      packet.header.extensions,\n      this.extIdUriMap,\n    );\n\n    let rtpReceiver: RTCRtpReceiver | undefined = this.ssrcTable[\n      packet.header.ssrc\n    ] as RTCRtpReceiver;\n\n    const rid = extensions[RTP_EXTENSION_URI.sdesRTPStreamID];\n    if (typeof rid === \"string\") {\n      rtpReceiver = this.ridTable[rid] as RTCRtpReceiver;\n      rtpReceiver.latestRid = rid;\n      rtpReceiver.handleRtpByRid(packet, rid, extensions);\n    } else if (rtpReceiver) {\n      rtpReceiver.handleRtpBySsrc(packet, extensions);\n    } else {\n      // simulcast after send receiver report\n      rtpReceiver = Object.values(this.ridTable)\n        .filter((r): r is RTCRtpReceiver => r instanceof RTCRtpReceiver)\n        .find((r) => r.trackBySSRC[packet.header.ssrc]);\n      if (rtpReceiver) {\n        log(\"simulcast register receiver by ssrc\", packet.header.ssrc);\n        this.registerRtpReceiver(rtpReceiver, packet.header.ssrc);\n        rtpReceiver.handleRtpBySsrc(packet, extensions);\n      } else {\n        // bug\n      }\n    }\n\n    if (!rtpReceiver) {\n      log(\"ssrcReceiver not found\");\n      return;\n    }\n\n    const sdesMid = extensions[RTP_EXTENSION_URI.sdesMid];\n    if (typeof sdesMid === \"string\") {\n      rtpReceiver.sdesMid = sdesMid;\n    }\n\n    const repairedRid = extensions[\n      RTP_EXTENSION_URI.repairedRtpStreamId\n    ] as string;\n    if (typeof repairedRid === \"string\") {\n      rtpReceiver.latestRepairedRid = repairedRid;\n    }\n  };\n\n  routeRtcp = (packet: RtcpPacket) => {\n    const recipients: (RTCRtpReceiver | RTCRtpSender)[] = [];\n\n    switch (packet.type) {\n      case RtcpSrPacket.type:\n        {\n          packet = packet as RtcpSrPacket;\n          recipients.push(this.ssrcTable[packet.ssrc]);\n        }\n        break;\n      case RtcpRrPacket.type:\n        {\n          packet = packet as RtcpRrPacket;\n          packet.reports.forEach((report) => {\n            recipients.push(this.ssrcTable[report.ssrc]);\n          });\n        }\n        break;\n      case RtcpSourceDescriptionPacket.type:\n        {\n          const sdes = packet as RtcpSourceDescriptionPacket;\n          // log(\"sdes\", JSON.stringify(sdes.chunks));\n        }\n        break;\n      case RtcpTransportLayerFeedback.type:\n        {\n          const rtpfb = packet as RtcpTransportLayerFeedback;\n          if (rtpfb.feedback) {\n            recipients.push(this.ssrcTable[rtpfb.feedback.mediaSourceSsrc]);\n          }\n        }\n        break;\n      case RtcpPayloadSpecificFeedback.type:\n        {\n          const psfb = packet as RtcpPayloadSpecificFeedback;\n          switch (psfb.feedback.count) {\n            case ReceiverEstimatedMaxBitrate.count:\n              {\n                const remb = psfb.feedback as ReceiverEstimatedMaxBitrate;\n                recipients.push(this.ssrcTable[remb.ssrcFeedbacks[0]]);\n              }\n              break;\n            default:\n              recipients.push(\n                this.ssrcTable[psfb.feedback.senderSsrc] ||\n                  this.ssrcTable[psfb.feedback.mediaSsrc],\n              );\n          }\n        }\n        break;\n    }\n    recipients\n      .filter((v) => v) // todo simulcast\n      .forEach((recipient) => recipient.handleRtcpPacket(packet));\n  };\n}\n"]}
+{"version":3,"file":"router.js","sourceRoot":"","sources":["../../../../src/media/router.ts"],"names":[],"mappings":";;;AAAA,wCAawB;AACxB,6DAA2D;AAK3D,+CAA+C;AAG/C,mCAA2C;AAE3C,MAAM,GAAG,GAAG,IAAA,WAAK,EAAC,4CAA4C,CAAC,CAAC;AAEhE,MAAa,SAAS;IAKpB;QAJA;;;;mBAA+D,EAAE;WAAC;QAClE;;;;mBAA6D,EAAE;WAAC;QAChE;;;;mBAAwC,EAAE;WAAC;QAqF3C;;;;mBAAW,CAAC,MAAiB,EAAE,EAAE;gBAC/B,MAAM,UAAU,GAAe,IAAA,+BAAyB,EACtD,MAAM,CAAC,MAAM,CAAC,UAAU,EACxB,IAAI,CAAC,WAAW,CACjB,CAAC;gBAEF,IAAI,WAAW,GAA+B,IAAI,CAAC,SAAS,CAC1D,MAAM,CAAC,MAAM,CAAC,IAAI,CACD,CAAC;gBAEpB,MAAM,GAAG,GAAG,UAAU,CAAC,uBAAiB,CAAC,eAAe,CAAC,CAAC;gBAC1D,IAAI,OAAO,GAAG,KAAK,QAAQ,EAAE,CAAC;oBAC5B,WAAW,GAAG,IAAI,CAAC,QAAQ,CAAC,GAAG,CAA+B,CAAC;oBAC/D,IAAI,WAAW,EAAE,CAAC;wBAChB,WAAW,CAAC,SAAS,GAAG,GAAG,CAAC;wBAC5B,WAAW,CAAC,cAAc,CAAC,MAAM,EAAE,GAAG,EAAE,UAAU,CAAC,CAAC;oBACtD,CAAC;gBACH,CAAC;qBAAM,IAAI,WAAW,EAAE,CAAC;oBACvB,WAAW,CAAC,eAAe,CAAC,MAAM,EAAE,UAAU,CAAC,CAAC;gBAClD,CAAC;qBAAM,CAAC;oBACN,uCAAuC;oBACvC,WAAW,GAAG,MAAM,CAAC,MAAM,CAAC,IAAI,CAAC,QAAQ,CAAC;yBACvC,MAAM,CAAC,CAAC,CAAC,EAAuB,EAAE,CAAC,CAAC,YAAY,4BAAc,CAAC;yBAC/D,IAAI,CAAC,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,CAAC,WAAW,CAAC,MAAM,CAAC,MAAM,CAAC,IAAI,CAAC,CAAC,CAAC;oBAClD,IAAI,WAAW,EAAE,CAAC;wBAChB,GAAG,CAAC,qCAAqC,EAAE,MAAM,CAAC,MAAM,CAAC,IAAI,CAAC,CAAC;wBAC/D,IAAI,CAAC,mBAAmB,CAAC,WAAW,EAAE,MAAM,CAAC,MAAM,CAAC,IAAI,CAAC,CAAC;wBAC1D,WAAW,CAAC,eAAe,CAAC,MAAM,EAAE,UAAU,CAAC,CAAC;oBAClD,CAAC;yBAAM,CAAC;wBACN,MAAM;oBACR,CAAC;gBACH,CAAC;gBAED,IAAI,CAAC,WAAW,EAAE,CAAC;oBACjB,GAAG,CAAC,wBAAwB,CAAC,CAAC;oBAC9B,OAAO;gBACT,CAAC;gBAED,MAAM,OAAO,GAAG,UAAU,CAAC,uBAAiB,CAAC,OAAO,CAAC,CAAC;gBACtD,IAAI,OAAO,OAAO,KAAK,QAAQ,EAAE,CAAC;oBAChC,WAAW,CAAC,OAAO,GAAG,OAAO,CAAC;gBAChC,CAAC;gBAED,MAAM,WAAW,GAAG,UAAU,CAC5B,uBAAiB,CAAC,mBAAmB,CAC5B,CAAC;gBACZ,IAAI,OAAO,WAAW,KAAK,QAAQ,EAAE,CAAC;oBACpC,WAAW,CAAC,iBAAiB,GAAG,WAAW,CAAC;gBAC9C,CAAC;YACH,CAAC;WAAC;QAEF;;;;mBAAY,CAAC,MAAkB,EAAE,EAAE;gBACjC,MAAM,UAAU,GAAsC,EAAE,CAAC;gBAEzD,QAAQ,MAAM,CAAC,IAAI,EAAE,CAAC;oBACpB,KAAK,kBAAY,CAAC,IAAI;wBACpB,CAAC;4BACC,MAAM,GAAG,MAAsB,CAAC;4BAChC,UAAU,CAAC,IAAI,CAAC,IAAI,CAAC,SAAS,CAAC,MAAM,CAAC,IAAI,CAAC,CAAC,CAAC;wBAC/C,CAAC;wBACD,MAAM;oBACR,KAAK,kBAAY,CAAC,IAAI;wBACpB,CAAC;4BACC,MAAM,GAAG,MAAsB,CAAC;4BAChC,MAAM,CAAC,OAAO,CAAC,OAAO,CAAC,CAAC,MAAM,EAAE,EAAE;gCAChC,UAAU,CAAC,IAAI,CAAC,IAAI,CAAC,SAAS,CAAC,MAAM,CAAC,IAAI,CAAC,CAAC,CAAC;4BAC/C,CAAC,CAAC,CAAC;wBACL,CAAC;wBACD,MAAM;oBACR,KAAK,iCAA2B,CAAC,IAAI;wBACnC,CAAC;4BACC,MAAM,IAAI,GAAG,MAAqC,CAAC;4BACnD,4CAA4C;wBAC9C,CAAC;wBACD,MAAM;oBACR,KAAK,gCAA0B,CAAC,IAAI;wBAClC,CAAC;4BACC,MAAM,KAAK,GAAG,MAAoC,CAAC;4BACnD,IAAI,KAAK,CAAC,QAAQ,EAAE,CAAC;gCACnB,UAAU,CAAC,IAAI,CAAC,IAAI,CAAC,SAAS,CAAC,KAAK,CAAC,QAAQ,CAAC,eAAe,CAAC,CAAC,CAAC;4BAClE,CAAC;wBACH,CAAC;wBACD,MAAM;oBACR,KAAK,iCAA2B,CAAC,IAAI;wBACnC,CAAC;4BACC,MAAM,IAAI,GAAG,MAAqC,CAAC;4BACnD,QAAQ,IAAI,CAAC,QAAQ,CAAC,KAAK,EAAE,CAAC;gCAC5B,KAAK,iCAA2B,CAAC,KAAK;oCACpC,CAAC;wCACC,MAAM,IAAI,GAAG,IAAI,CAAC,QAAuC,CAAC;wCAC1D,UAAU,CAAC,IAAI,CAAC,IAAI,CAAC,SAAS,CAAC,IAAI,CAAC,aAAa,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC;oCACzD,CAAC;oCACD,MAAM;gCACR;oCACE,UAAU,CAAC,IAAI,CACb,IAAI,CAAC,SAAS,CAAC,IAAI,CAAC,QAAQ,CAAC,UAAU,CAAC;wCACtC,IAAI,CAAC,SAAS,CAAC,IAAI,CAAC,QAAQ,CAAC,SAAS,CAAC,CAC1C,CAAC;4BACN,CAAC;wBACH,CAAC;wBACD,MAAM;gBACV,CAAC;gBACD,UAAU;qBACP,MAAM,CAAC,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,CAAC,CAAC,iBAAiB;qBAClC,OAAO,CAAC,CAAC,SAAS,EAAE,EAAE,CAAC,SAAS,CAAC,gBAAgB,CAAC,MAAM,CAAC,CAAC,CAAC;YAChE,CAAC;WAAC;IA5La,CAAC;IAEhB,iBAAiB,CAAC,MAAoB;QACpC,IAAI,CAAC,SAAS,CAAC,MAAM,CAAC,IAAI,CAAC,GAAG,MAAM,CAAC;IACvC,CAAC;IAEO,mBAAmB,CAAC,QAAwB,EAAE,IAAY;QAChE,GAAG,CAAC,qBAAqB,EAAE,IAAI,CAAC,CAAC;QACjC,IAAI,CAAC,SAAS,CAAC,IAAI,CAAC,GAAG,QAAQ,CAAC;IAClC,CAAC;IAED,yBAAyB,CACvB,WAA8B,EAC9B,MAA+B;QAE/B,GAAG,CAAC,2BAA2B,EAAE,MAAM,CAAC,CAAC;QAEzC,MAAM,CAAC,SAAS;aACb,MAAM,CAAC,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,CAAC,IAAI,IAAI,SAAS,CAAC,CAAC,WAAW;aAC9C,OAAO,CAAC,CAAC,MAAM,EAAE,CAAC,EAAE,EAAE;YACrB,IAAI,CAAC,mBAAmB,CAAC,WAAW,CAAC,QAAQ,EAAE,MAAM,CAAC,IAAI,CAAC,CAAC;YAC5D,WAAW,CAAC,QAAQ,CAClB,IAAI,wBAAgB,CAAC;gBACnB,IAAI,EAAE,MAAM,CAAC,IAAI;gBACjB,IAAI,EAAE,WAAW,CAAC,IAAI;gBACtB,EAAE,EAAE,WAAW,CAAC,MAAM,CAAC,OAAO;gBAC9B,MAAM,EAAE,IAAI;gBACZ,KAAK,EAAE,IAAA,wCAAmB,EACxB,WAAW,CAAC,IAAI,EAChB,MAAM,CAAC,MAAM,CAAC,CAAC,CAAC,EAChB,MAAM,CAAC,MAAM,CACd;aACF,CAAC,CACH,CAAC;YACF,IAAI,MAAM,CAAC,GAAG,EAAE,CAAC;gBACf,IAAI,CAAC,mBAAmB,CAAC,WAAW,CAAC,QAAQ,EAAE,MAAM,CAAC,GAAG,CAAC,IAAI,CAAC,CAAC;YAClE,CAAC;QACH,CAAC,CAAC,CAAC;QAEL,MAAM,CAAC,gBAAgB,CAAC,OAAO,CAAC,CAAC,SAAS,EAAE,EAAE;YAC5C,IAAI,CAAC,WAAW,CAAC,SAAS,CAAC,EAAE,CAAC,GAAG,SAAS,CAAC,GAAG,CAAC;QACjD,CAAC,CAAC,CAAC;IACL,CAAC;IAED,wBAAwB,CACtB,WAA8B,EAC9B,KAAgC,EAChC,MAA+B;QAE/B,mDAAmD;QACnD,MAAM,CAAC,KAAK,CAAC,GAAG,MAAM,CAAC,MAAM,CAAC;QAE9B,GAAG,CAAC,0BAA0B,EAAE,KAAK,CAAC,CAAC;QACvC,WAAW,CAAC,QAAQ,CAClB,IAAI,wBAAgB,CAAC;YACnB,GAAG,EAAE,KAAK,CAAC,GAAG;YACd,IAAI,EAAE,WAAW,CAAC,IAAI;YACtB,EAAE,EAAE,WAAW,CAAC,MAAM,CAAC,OAAO;YAC9B,MAAM,EAAE,IAAI;YACZ,KAAK;SACN,CAAC,CACH,CAAC;QACF,IAAI,CAAC,QAAQ,CAAC,KAAK,CAAC,GAAG,CAAC,GAAG,WAAW,CAAC,QAAQ,CAAC;IAClD,CAAC;IAED,2CAA2C;IAC3C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{\n  type Extensions,\n  RTP_EXTENSION_URI,\n  ReceiverEstimatedMaxBitrate,\n  type RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpRrPacket,\n  RtcpSourceDescriptionPacket,\n  RtcpSrPacket,\n  RtcpTransportLayerFeedback,\n  type RtpPacket,\n  debug,\n  rtpHeaderExtensionsParser,\n} from \"../imports/rtp\";\nimport { deliveredTrackCodec } from \"./codecCompatibility\";\nimport type {\n  RTCRtpReceiveParameters,\n  RTCRtpSimulcastParameters,\n} from \"./parameters\";\nimport { RTCRtpReceiver } from \"./rtpReceiver\";\nimport type { RTCRtpSender } from \"./rtpSender\";\nimport type { RTCRtpTransceiver } from \"./rtpTransceiver\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/router.ts\");\n\nexport class RtpRouter {\n  ssrcTable: { [ssrc: number]: RTCRtpReceiver | RTCRtpSender } = {};\n  ridTable: { [rid: string]: RTCRtpReceiver | RTCRtpSender } = {};\n  extIdUriMap: { [id: number]: string } = {};\n\n  constructor() {}\n\n  registerRtpSender(sender: RTCRtpSender) {\n    this.ssrcTable[sender.ssrc] = sender;\n  }\n\n  private registerRtpReceiver(receiver: RTCRtpReceiver, ssrc: number) {\n    log(\"registerRtpReceiver\", ssrc);\n    this.ssrcTable[ssrc] = receiver;\n  }\n\n  registerRtpReceiverBySsrc(\n    transceiver: RTCRtpTransceiver,\n    params: RTCRtpReceiveParameters,\n  ) {\n    log(\"registerRtpReceiverBySsrc\", params);\n\n    params.encodings\n      .filter((e) => e.ssrc != undefined) // todo fix\n      .forEach((encode, i) => {\n        this.registerRtpReceiver(transceiver.receiver, encode.ssrc);\n        transceiver.addTrack(\n          new MediaStreamTrack({\n            ssrc: encode.ssrc,\n            kind: transceiver.kind,\n            id: transceiver.sender.trackId,\n            remote: true,\n            codec: deliveredTrackCodec(\n              transceiver.kind,\n              params.codecs[i],\n              params.codecs,\n            ),\n          }),\n        );\n        if (encode.rtx) {\n          this.registerRtpReceiver(transceiver.receiver, encode.rtx.ssrc);\n        }\n      });\n\n    params.headerExtensions.forEach((extension) => {\n      this.extIdUriMap[extension.id] = extension.uri;\n    });\n  }\n\n  registerRtpReceiverByRid(\n    transceiver: RTCRtpTransceiver,\n    param: RTCRtpSimulcastParameters,\n    params: RTCRtpReceiveParameters,\n  ) {\n    // サイマルキャスト利用時のRTXをサポートしていないのでcodecs/encodingsは常に一つ\n    const [codec] = params.codecs;\n\n    log(\"registerRtpReceiverByRid\", param);\n    transceiver.addTrack(\n      new MediaStreamTrack({\n        rid: param.rid,\n        kind: transceiver.kind,\n        id: transceiver.sender.trackId,\n        remote: true,\n        codec,\n      }),\n    );\n    this.ridTable[param.rid] = transceiver.receiver;\n  }\n\n  /**停止した transceiver の SSRC / RID 登録を解除する */\n  unregisterTransceiver(transceiver: RTCRtpTransceiver) {\n    const owners: (RTCRtpReceiver | RTCRtpSender)[] = [\n      transceiver.sender,\n      transceiver.receiver,\n    ];\n    for (const [ssrc, owner] of Object.entries(this.ssrcTable)) {\n      if (owners.includes(owner)) {\n        delete this.ssrcTable[Number(ssrc)];\n      }\n    }\n    for (const [rid, owner] of Object.entries(this.ridTable)) {\n      if (owners.includes(owner)) {\n        delete this.ridTable[rid];\n      }\n    }\n  }\n\n  routeRtp = (packet: RtpPacket) => {\n    const extensions: Extensions = rtpHeaderExtensionsParser(\n      packet.header.extensions,\n      this.extIdUriMap,\n    );\n\n    let rtpReceiver: RTCRtpReceiver | undefined = this.ssrcTable[\n      packet.header.ssrc\n    ] as RTCRtpReceiver;\n\n    const rid = extensions[RTP_EXTENSION_URI.sdesRTPStreamID];\n    if (typeof rid === \"string\") {\n      rtpReceiver = this.ridTable[rid] as RTCRtpReceiver | undefined;\n      if (rtpReceiver) {\n        rtpReceiver.latestRid = rid;\n        rtpReceiver.handleRtpByRid(packet, rid, extensions);\n      }\n    } else if (rtpReceiver) {\n      rtpReceiver.handleRtpBySsrc(packet, extensions);\n    } else {\n      // simulcast after send receiver report\n      rtpReceiver = Object.values(this.ridTable)\n        .filter((r): r is RTCRtpReceiver => r instanceof RTCRtpReceiver)\n        .find((r) => r.trackBySSRC[packet.header.ssrc]);\n      if (rtpReceiver) {\n        log(\"simulcast register receiver by ssrc\", packet.header.ssrc);\n        this.registerRtpReceiver(rtpReceiver, packet.header.ssrc);\n        rtpReceiver.handleRtpBySsrc(packet, extensions);\n      } else {\n        // bug\n      }\n    }\n\n    if (!rtpReceiver) {\n      log(\"ssrcReceiver not found\");\n      return;\n    }\n\n    const sdesMid = extensions[RTP_EXTENSION_URI.sdesMid];\n    if (typeof sdesMid === \"string\") {\n      rtpReceiver.sdesMid = sdesMid;\n    }\n\n    const repairedRid = extensions[\n      RTP_EXTENSION_URI.repairedRtpStreamId\n    ] as string;\n    if (typeof repairedRid === \"string\") {\n      rtpReceiver.latestRepairedRid = repairedRid;\n    }\n  };\n\n  routeRtcp = (packet: RtcpPacket) => {\n    const recipients: (RTCRtpReceiver | RTCRtpSender)[] = [];\n\n    switch (packet.type) {\n      case RtcpSrPacket.type:\n        {\n          packet = packet as RtcpSrPacket;\n          recipients.push(this.ssrcTable[packet.ssrc]);\n        }\n        break;\n      case RtcpRrPacket.type:\n        {\n          packet = packet as RtcpRrPacket;\n          packet.reports.forEach((report) => {\n            recipients.push(this.ssrcTable[report.ssrc]);\n          });\n        }\n        break;\n      case RtcpSourceDescriptionPacket.type:\n        {\n          const sdes = packet as RtcpSourceDescriptionPacket;\n          // log(\"sdes\", JSON.stringify(sdes.chunks));\n        }\n        break;\n      case RtcpTransportLayerFeedback.type:\n        {\n          const rtpfb = packet as RtcpTransportLayerFeedback;\n          if (rtpfb.feedback) {\n            recipients.push(this.ssrcTable[rtpfb.feedback.mediaSourceSsrc]);\n          }\n        }\n        break;\n      case RtcpPayloadSpecificFeedback.type:\n        {\n          const psfb = packet as RtcpPayloadSpecificFeedback;\n          switch (psfb.feedback.count) {\n            case ReceiverEstimatedMaxBitrate.count:\n              {\n                const remb = psfb.feedback as ReceiverEstimatedMaxBitrate;\n                recipients.push(this.ssrcTable[remb.ssrcFeedbacks[0]]);\n              }\n              break;\n            default:\n              recipients.push(\n                this.ssrcTable[psfb.feedback.senderSsrc] ||\n                  this.ssrcTable[psfb.feedback.mediaSsrc],\n              );\n          }\n        }\n        break;\n    }\n    recipients\n      .filter((v) => v) // todo simulcast\n      .forEach((recipient) => recipient.handleRtcpPacket(packet));\n  };\n}\n"]}

package/lib/webrtc/src/media/rtpReceiver.d.ts

@@ -64,12 +64,24 @@ export declare class RTCRtpReceiver {
     get twccEnabled(): import("./parameters").RTCPFB | undefined;
     get pliEnabled(): import("./parameters").RTCPFB | undefined;
     prepareReceive(params: RTCRtpReceiveParameters): void;
+    /**
+     * negotiated codec と receiver の実処理状態を再同期する。
+     * codec map と RTX mapping を置換し、既存 remote track の codec
+     * metadata も新しい先頭 codec に更新する。
+     * 既存の receiverTWCC は破棄し、新しい codec が transport-cc を持ち
+     * remote SSRC が分かる場合だけ再生成する。
+     * setCodecPreferences() 後の createAnswer() 再解決や remote からの
+     * 再交渉では、除外された payload type の RTP を受け付けないようにする。
+     */
+    resyncCodecs(params: RTCRtpReceiveParameters, mediaSourceSsrc?: number): void;
     /**
      * setup TWCC if supported
      */
     setupTWCC(mediaSourceSsrc: number): void;
     addTrack(track: MediaStreamTrack): boolean;
     stop(): void;
+    /**transceiver の停止確定時に remote track を ended にする */
+    endTracks(): void;
     runRtcp(): Promise<void>;
     private getInboundRtpStatsId;
     private getRemoteOutboundRtpStatsId;

package/lib/webrtc/src/media/rtpReceiver.js

@@ -6,6 +6,7 @@ const promises_1 = require("timers/promises");
 const common_1 = require("../imports/common");
 const rtp_1 = require("../imports/rtp");
 const utils_1 = require("../utils");
+const codecCompatibility_1 = require("./codecCompatibility");
 const nack_1 = require("./receiver/nack");
 const receiverTwcc_1 = require("./receiver/receiverTwcc");
 const statistics_1 = require("./receiver/statistics");
@@ -289,6 +290,34 @@ class RTCRtpReceiver {
             }
         });
     }
+    /**
+     * negotiated codec と receiver の実処理状態を再同期する。
+     * codec map と RTX mapping を置換し、既存 remote track の codec
+     * metadata も新しい先頭 codec に更新する。
+     * 既存の receiverTWCC は破棄し、新しい codec が transport-cc を持ち
+     * remote SSRC が分かる場合だけ再生成する。
+     * setCodecPreferences() 後の createAnswer() 再解決や remote からの
+     * 再交渉では、除外された payload type の RTP を受け付けないようにする。
+     */
+    resyncCodecs(params, mediaSourceSsrc) {
+        for (const key of Object.keys(this.codecs)) {
+            delete this.codecs[Number(key)];
+        }
+        for (const key of Object.keys(this.ssrcByRtx)) {
+            delete this.ssrcByRtx[Number(key)];
+        }
+        this.prepareReceive(params);
+        const codec = (0, codecCompatibility_1.deliveredTrackCodec)(this.kind, params.codecs[0], params.codecs);
+        if (codec) {
+            for (const track of this.tracks) {
+                track.codec = codec;
+            }
+        }
+        this.receiverTWCC = undefined;
+        if (mediaSourceSsrc != undefined) {
+            this.setupTWCC(mediaSourceSsrc);
+        }
+    }
     /**
      * setup TWCC if supported
      */
@@ -326,6 +355,12 @@ class RTCRtpReceiver {
             this.receiverTWCC.twccRunning = false;
         this.nack.close();
     }
+    /**transceiver の停止確定時に remote track を ended にする */
+    endTracks() {
+        for (const track of [...this.tracks, this.defaultTrack]) {
+            track.stop();
+        }
+    }
     async runRtcp() {
         if (this.rtcpRunning || this.stopped)
             return;

package/lib/webrtc/src/media/rtpReceiver.js.map

@@ -1 +1 @@
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{ randomUUID } from \"crypto\";\nimport { setTimeout } from \"timers/promises\";\nimport { Event, int } from \"../imports/common\";\n\nimport {\n  type Extensions,\n  PictureLossIndication,\n  RTP_EXTENSION_URI,\n  Red,\n  RedHandler,\n  type RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpReceiverInfo,\n  RtcpRrPacket,\n  RtcpSrPacket,\n  type RtpPacket,\n  type TransportWideCCPayload,\n  debug,\n  unwrapRtx,\n} from \"../imports/rtp\";\nimport type { PeerConfig } from \"../peerConnection\";\nimport type { RTCDtlsTransport } from \"../transport/dtls\";\nimport type { Kind } from \"../types/domain\";\nimport { compactNtp, ntpTimeToEpochMs, timestampSeconds } from \"../utils\";\nimport type {\n  RTCRtpCodecParameters,\n  RTCRtpReceiveParameters,\n} from \"./parameters\";\nimport { NackHandler } from \"./receiver/nack\";\nimport { ReceiverTWCC } from \"./receiver/receiverTwcc\";\nimport { StreamStatistics } from \"./receiver/statistics\";\n\nimport { codecParametersFromString } from \"../sdp\";\nimport { usePLI, useTWCC } from \"./extension/rtcpFeedback\";\nimport {\n  type RTCCodecStats,\n  type RTCInboundRtpStreamStats,\n  type RTCRemoteOutboundRtpStreamStats,\n  type RTCStats,\n  type RTCStatsReport,\n  buildStatsReport,\n  generateCodecStatsId,\n  generateStatsId,\n  getStatsTimestamp,\n} from \"./stats\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/rtpReceiver.ts\");\n\nexport class RTCRtpReceiver {\n  private readonly codecs: { [pt: number]: RTCRtpCodecParameters } = {};\n  private readonly defaultTrack: MediaStreamTrack;\n  private get codecArray() {\n    return Object.values(this.codecs).sort(\n      (a, b) => a.payloadType - b.payloadType,\n    );\n  }\n  private readonly ssrcByRtx: { [rtxSsrc: number]: number } = {};\n  private readonly nack = new NackHandler(this);\n  private readonly audioRedHandler = new RedHandler();\n\n  readonly type = \"receiver\";\n  readonly uuid = randomUUID().toString();\n  readonly tracks: MediaStreamTrack[] = [];\n  readonly trackBySSRC: { [ssrc: string]: MediaStreamTrack } = {};\n  readonly trackByRID: { [rid: string]: MediaStreamTrack } = {};\n  /**last sender Report Timestamp\n   * compactNtp\n   */\n  readonly lastSRtimestamp: { [ssrc: number]: number } = {};\n  /**seconds */\n  readonly receiveLastSRTimestamp: { [ssrc: number]: number } = {};\n  readonly onPacketLost = this.nack.onPacketLost;\n  readonly onRtcp = new Event<[RtcpPacket]>();\n\n  dtlsTransport!: RTCDtlsTransport;\n  sdesMid?: string;\n  latestRid?: string;\n  latestRepairedRid?: string;\n\n  receiverTWCC?: ReceiverTWCC;\n  stopped = false;\n  remoteStreamId?: string;\n  remoteStreamIds: string[] = [];\n  remoteTrackId?: string;\n\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n  private remoteStreams: { [ssrc: number]: StreamStatistics } = {};\n  private senderReportsReceivedBySsrc: { [ssrc: number]: number } = {};\n  private remoteTimestampsBySsrc: { [ssrc: number]: number } = {};\n  private remotePacketCountBySsrc: { [ssrc: number]: number } = {};\n  private remoteOctetCountBySsrc: { [ssrc: number]: number } = {};\n  private nackCountBySsrc: { [ssrc: number]: number } = {};\n  private pliCountBySsrc: { [ssrc: number]: number } = {};\n\n  constructor(\n    readonly config: PeerConfig,\n    public kind: Kind,\n    public rtcpSsrc: number,\n  ) {\n    this.defaultTrack = new MediaStreamTrack({ kind, remote: true });\n    this.onPacketLost.subscribe((nack) => {\n      this.nackCountBySsrc[nack.mediaSourceSsrc] =\n        (this.nackCountBySsrc[nack.mediaSourceSsrc] ?? 0) + 1;\n    });\n  }\n\n  get transport() {\n    return this.dtlsTransport ?? null;\n  }\n\n  setDtlsTransport(dtls: RTCDtlsTransport) {\n    this.dtlsTransport = dtls;\n  }\n\n  // todo fix\n  get track() {\n    return this.tracks[0] ?? this.defaultTrack;\n  }\n\n  get nackEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find((f) => f.type === \"nack\");\n  }\n\n  get twccEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find(\n      (f) => f.type === useTWCC().type,\n    );\n  }\n\n  get pliEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find(\n      (f) => f.type === usePLI().type,\n    );\n  }\n\n  prepareReceive(params: RTCRtpReceiveParameters) {\n    params.codecs.forEach((c) => {\n      this.codecs[c.payloadType] = c;\n    });\n    params.encodings.forEach((e) => {\n      if (e.rtx) {\n        this.ssrcByRtx[e.rtx.ssrc] = e.ssrc;\n      }\n    });\n  }\n\n  /**\n   * setup TWCC if supported\n   */\n  setupTWCC(mediaSourceSsrc: number) {\n    if (this.twccEnabled && !this.receiverTWCC) {\n      this.receiverTWCC = new ReceiverTWCC(\n        this.dtlsTransport,\n        this.rtcpSsrc,\n        mediaSourceSsrc,\n      );\n    }\n  }\n\n  addTrack(track: MediaStreamTrack) {\n    const exist = this.tracks.find((t) => {\n      if (t.rid) {\n        return t.rid === track.rid;\n      }\n      if (t.ssrc) {\n        return t.ssrc === track.ssrc;\n      }\n    });\n    if (exist) {\n      return false;\n    }\n    this.tracks.push(track);\n    if (track.ssrc) {\n      this.trackBySSRC[track.ssrc] = track;\n    }\n    if (track.rid) {\n      this.trackByRID[track.rid] = track;\n    }\n    return true;\n  }\n\n  stop() {\n    this.stopped = true;\n    this.rtcpRunning = false;\n    this.rtcpCancel.abort();\n\n    if (this.receiverTWCC) this.receiverTWCC.twccRunning = false;\n    this.nack.close();\n  }\n\n  async runRtcp() {\n    if (this.rtcpRunning || this.stopped) return;\n    this.rtcpRunning = true;\n\n    try {\n      while (this.rtcpRunning) {\n        await setTimeout(500 + Math.random() * 1000, undefined, {\n          signal: this.rtcpCancel.signal,\n        });\n\n        const reports = Object.entries(this.remoteStreams).map(\n          ([ssrc, stream]) => {\n            let lastSRtimestamp = 0,\n              delaySinceLastSR = 0;\n            if (this.lastSRtimestamp[ssrc]) {\n              lastSRtimestamp = this.lastSRtimestamp[ssrc];\n              const delaySeconds =\n                timestampSeconds() - this.receiveLastSRTimestamp[ssrc];\n              if (delaySeconds > 0 && delaySeconds < 65536) {\n                delaySinceLastSR = int(delaySeconds * 65536);\n              }\n            }\n\n            return new RtcpReceiverInfo({\n              ssrc: Number(ssrc),\n              fractionLost: stream.fraction_lost,\n              packetsLost: stream.packets_lost,\n              highestSequence: stream.max_seq,\n              jitter: stream.jitter,\n              lsr: lastSRtimestamp,\n              dlsr: delaySinceLastSR,\n            });\n          },\n        );\n\n        const packet = new RtcpRrPacket({ ssrc: this.rtcpSsrc, reports });\n\n        try {\n          if (this.config.debug.receiverReportDelay) {\n            await setTimeout(this.config.debug.receiverReportDelay);\n          }\n          await this.dtlsTransport.sendRtcp([packet]);\n        } catch (error) {\n          log(\"sendRtcp failed\", error);\n          await setTimeout(500 + Math.random() * 1000);\n        }\n      }\n    } catch (error) {}\n  }\n\n  private getInboundRtpStatsId(track: MediaStreamTrack) {\n    return generateStatsId(\"inbound-rtp\", track.id ?? track.uuid);\n  }\n\n  private getRemoteOutboundRtpStatsId(track: MediaStreamTrack) {\n    return generateStatsId(\"remote-outbound-rtp\", track.id ?? track.uuid);\n  }\n\n  getStatsRootIds(selector?: MediaStreamTrack) {\n    return this.tracks\n      .filter((track) => (!selector ? true : track === selector))\n      .filter((track) => track.ssrc)\n      .map((track) => this.getInboundRtpStatsId(track));\n  }\n\n  collectStats(timestamp: number): RTCStats[] {\n    const stats: RTCStats[] = [];\n    const transportId = this.dtlsTransport\n      ? generateStatsId(\"transport\", this.dtlsTransport.id)\n      : undefined;\n    const activeCodec = this.codecArray[0];\n    const emittedCodecIds = new Set<string>();\n\n    // Collect stats for each track\n    for (const track of this.tracks) {\n      if (!track.ssrc) continue;\n\n      const streamStats = this.remoteStreams[track.ssrc];\n\n      // Inbound RTP stats\n      const hasRemoteTimestamp =\n        this.remoteTimestampsBySsrc[track.ssrc] !== undefined;\n      const remoteId =\n        this.lastSRtimestamp[track.ssrc] !== undefined || hasRemoteTimestamp\n          ? this.getRemoteOutboundRtpStatsId(track)\n          : undefined;\n      const codecId =\n        activeCodec && transportId\n          ? generateCodecStatsId(\n              transportId,\n              activeCodec.payloadType,\n              track.id ?? track.uuid,\n            )\n          : undefined;\n\n      if (\n        activeCodec &&\n        transportId &&\n        codecId &&\n        !emittedCodecIds.has(codecId)\n      ) {\n        emittedCodecIds.add(codecId);\n        const codecStats: RTCCodecStats = {\n          type: \"codec\",\n          id: codecId,\n          timestamp,\n          payloadType: activeCodec.payloadType,\n          transportId,\n          mimeType: activeCodec.mimeType,\n          clockRate: activeCodec.clockRate,\n          channels: activeCodec.channels,\n          sdpFmtpLine: activeCodec.parameters,\n        };\n        stats.push(codecStats);\n      }\n\n      const inboundRtpStats: RTCInboundRtpStreamStats = {\n        type: \"inbound-rtp\",\n        id: this.getInboundRtpStatsId(track),\n        timestamp,\n        ssrc: track.ssrc,\n        kind: this.kind,\n        transportId,\n        codecId,\n        mid: this.sdesMid,\n        trackIdentifier: track.id ?? track.uuid,\n        packetsReceived: streamStats?.packets_received ?? 0,\n        bytesReceived: streamStats?.bytesReceived ?? 0,\n        headerBytesReceived: streamStats?.headerBytesReceived ?? 0,\n        packetsLost: streamStats?.packets_lost ?? 0,\n        jitter: streamStats?.clockRate\n          ? streamStats.jitter / streamStats.clockRate\n          : undefined,\n        lastPacketReceivedTimestamp: streamStats?.lastPacketReceivedTimestamp,\n        remoteId,\n        nackCount: this.nackCountBySsrc[track.ssrc] || undefined,\n        pliCount: this.pliCountBySsrc[track.ssrc] || undefined,\n      };\n      stats.push(inboundRtpStats);\n\n      // Remote outbound RTP stats (if we have SR info)\n      if (remoteId) {\n        const remoteOutboundStats: RTCRemoteOutboundRtpStreamStats = {\n          type: \"remote-outbound-rtp\",\n          id: this.getRemoteOutboundRtpStatsId(track),\n          timestamp,\n          ssrc: track.ssrc,\n          kind: this.kind,\n          transportId,\n          codecId: inboundRtpStats.codecId,\n          localId: inboundRtpStats.id,\n          remoteTimestamp: this.remoteTimestampsBySsrc[track.ssrc],\n          reportsSent: this.senderReportsReceivedBySsrc[track.ssrc] ?? 0,\n          packetsSent: this.remotePacketCountBySsrc[track.ssrc],\n          bytesSent: this.remoteOctetCountBySsrc[track.ssrc],\n        };\n        stats.push(remoteOutboundStats);\n      }\n    }\n\n    return stats;\n  }\n\n  async getStats(): Promise<RTCStatsReport> {\n    const timestamp = getStatsTimestamp();\n    const stats = this.collectStats(timestamp);\n\n    if (this.dtlsTransport) {\n      stats.push(...(await this.dtlsTransport.getStats(timestamp)));\n    }\n\n    return buildStatsReport(stats, this.getStatsRootIds());\n  }\n\n  async sendRtcpPLI(mediaSsrc: number) {\n    if (!this.pliEnabled) {\n      log(\"pli not supported\", { mediaSsrc });\n      return;\n    }\n\n    if (this.stopped) {\n      return;\n    }\n\n    log(\"sendRtcpPLI\", { mediaSsrc });\n\n    const packet = new RtcpPayloadSpecificFeedback({\n      feedback: new PictureLossIndication({\n        senderSsrc: this.rtcpSsrc,\n        mediaSsrc,\n      }),\n    });\n    try {\n      this.pliCountBySsrc[mediaSsrc] =\n        (this.pliCountBySsrc[mediaSsrc] ?? 0) + 1;\n      await this.dtlsTransport.sendRtcp([packet]);\n    } catch (error) {\n      log(error);\n    }\n  }\n\n  handleRtcpPacket(packet: RtcpPacket) {\n    switch (packet.type) {\n      case RtcpSrPacket.type:\n        {\n          const sr = packet as RtcpSrPacket;\n          this.lastSRtimestamp[sr.ssrc] = compactNtp(\n            sr.senderInfo.ntpTimestamp,\n          );\n          this.receiveLastSRTimestamp[sr.ssrc] = timestampSeconds();\n          this.senderReportsReceivedBySsrc[sr.ssrc] =\n            (this.senderReportsReceivedBySsrc[sr.ssrc] ?? 0) + 1;\n          this.remoteTimestampsBySsrc[sr.ssrc] = ntpTimeToEpochMs(\n            sr.senderInfo.ntpTimestamp,\n          );\n          this.remotePacketCountBySsrc[sr.ssrc] = sr.senderInfo.packetCount;\n          this.remoteOctetCountBySsrc[sr.ssrc] = sr.senderInfo.octetCount;\n\n          const track = this.trackBySSRC[packet.ssrc];\n          if (track) {\n            track.onReceiveRtcp.execute(packet);\n          }\n        }\n        break;\n    }\n    this.onRtcp.execute(packet);\n  }\n\n  handleRtpBySsrc = (packet: RtpPacket, extensions: Extensions) => {\n    const track = this.trackBySSRC[packet.header.ssrc];\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  handleRtpByRid = (packet: RtpPacket, rid: string, extensions: Extensions) => {\n    const track = this.trackByRID[rid];\n    if (!this.trackBySSRC[packet.header.ssrc]) {\n      this.trackBySSRC[packet.header.ssrc] = track;\n    }\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  private handleRTP(\n    packet: RtpPacket,\n    extensions: Extensions,\n    track?: MediaStreamTrack,\n  ) {\n    if (this.stopped) {\n      return;\n    }\n\n    const codec = this.codecs[packet.header.payloadType];\n    if (!codec) {\n      // log(\"unknown codec \" + packet.header.payloadType);\n      return;\n    }\n\n    this.remoteStreams[packet.header.ssrc] =\n      this.remoteStreams[packet.header.ssrc] ??\n      new StreamStatistics(codec.clockRate);\n    this.remoteStreams[packet.header.ssrc].add(packet);\n\n    if (this.receiverTWCC) {\n      const transportSequenceNumber = extensions[\n        RTP_EXTENSION_URI.transportWideCC\n      ] as TransportWideCCPayload;\n\n      if (!transportSequenceNumber == undefined) {\n        throw new Error(\"undefined\");\n      }\n\n      this.receiverTWCC.handleTWCC(transportSequenceNumber);\n    } else if (this.twccEnabled) {\n      this.setupTWCC(packet.header.ssrc);\n    }\n\n    if (codec.name.toLowerCase() === \"rtx\") {\n      const originalSsrc = this.ssrcByRtx[packet.header.ssrc];\n      const codecParams = codecParametersFromString(codec.parameters ?? \"\");\n      const rtxCodec = this.codecs[codecParams[\"apt\"]];\n      if (packet.payload.length < 2) return;\n\n      packet = unwrapRtx(packet, rtxCodec.payloadType, originalSsrc);\n      track = this.trackBySSRC[originalSsrc];\n    }\n\n    let red: Red | undefined;\n    if (codec.name.toLowerCase() === \"red\") {\n      red = Red.deSerialize(packet.payload);\n      if (\n        !Object.keys(this.codecs).includes(\n          red.header.fields[0].blockPT.toString(),\n        )\n      ) {\n        return;\n      }\n    }\n\n    if (track?.kind === \"video\" && this.nackEnabled) {\n      this.nack.addPacket(packet);\n    }\n\n    if (track) {\n      if (red) {\n        if (track.kind === \"audio\") {\n          const payloads = this.audioRedHandler.push(red, packet);\n          for (const packet of payloads) {\n            track.onReceiveRtp.execute(packet.clone(), extensions);\n          }\n        } else {\n        }\n      } else {\n        track.onReceiveRtp.execute(packet.clone(), extensions);\n      }\n    }\n\n    this.runRtcp();\n  }\n}\n"]}
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{ randomUUID } from \"crypto\";\nimport { setTimeout } from \"timers/promises\";\nimport { Event, int } from \"../imports/common\";\n\nimport {\n  type Extensions,\n  PictureLossIndication,\n  RTP_EXTENSION_URI,\n  Red,\n  RedHandler,\n  type RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpReceiverInfo,\n  RtcpRrPacket,\n  RtcpSrPacket,\n  type RtpPacket,\n  type TransportWideCCPayload,\n  debug,\n  unwrapRtx,\n} from \"../imports/rtp\";\nimport type { PeerConfig } from \"../peerConnection\";\nimport type { RTCDtlsTransport } from \"../transport/dtls\";\nimport type { Kind } from \"../types/domain\";\nimport { compactNtp, ntpTimeToEpochMs, timestampSeconds } from \"../utils\";\nimport { deliveredTrackCodec } from \"./codecCompatibility\";\nimport type {\n  RTCRtpCodecParameters,\n  RTCRtpReceiveParameters,\n} from \"./parameters\";\nimport { NackHandler } from \"./receiver/nack\";\nimport { ReceiverTWCC } from \"./receiver/receiverTwcc\";\nimport { StreamStatistics } from \"./receiver/statistics\";\n\nimport { codecParametersFromString } from \"../sdp\";\nimport { usePLI, useTWCC } from \"./extension/rtcpFeedback\";\nimport {\n  type RTCCodecStats,\n  type RTCInboundRtpStreamStats,\n  type RTCRemoteOutboundRtpStreamStats,\n  type RTCStats,\n  type RTCStatsReport,\n  buildStatsReport,\n  generateCodecStatsId,\n  generateStatsId,\n  getStatsTimestamp,\n} from \"./stats\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/rtpReceiver.ts\");\n\nexport class RTCRtpReceiver {\n  private readonly codecs: { [pt: number]: RTCRtpCodecParameters } = {};\n  private readonly defaultTrack: MediaStreamTrack;\n  private get codecArray() {\n    return Object.values(this.codecs).sort(\n      (a, b) => a.payloadType - b.payloadType,\n    );\n  }\n  private readonly ssrcByRtx: { [rtxSsrc: number]: number } = {};\n  private readonly nack = new NackHandler(this);\n  private readonly audioRedHandler = new RedHandler();\n\n  readonly type = \"receiver\";\n  readonly uuid = randomUUID().toString();\n  readonly tracks: MediaStreamTrack[] = [];\n  readonly trackBySSRC: { [ssrc: string]: MediaStreamTrack } = {};\n  readonly trackByRID: { [rid: string]: MediaStreamTrack } = {};\n  /**last sender Report Timestamp\n   * compactNtp\n   */\n  readonly lastSRtimestamp: { [ssrc: number]: number } = {};\n  /**seconds */\n  readonly receiveLastSRTimestamp: { [ssrc: number]: number } = {};\n  readonly onPacketLost = this.nack.onPacketLost;\n  readonly onRtcp = new Event<[RtcpPacket]>();\n\n  dtlsTransport!: RTCDtlsTransport;\n  sdesMid?: string;\n  latestRid?: string;\n  latestRepairedRid?: string;\n\n  receiverTWCC?: ReceiverTWCC;\n  stopped = false;\n  remoteStreamId?: string;\n  remoteStreamIds: string[] = [];\n  remoteTrackId?: string;\n\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n  private remoteStreams: { [ssrc: number]: StreamStatistics } = {};\n  private senderReportsReceivedBySsrc: { [ssrc: number]: number } = {};\n  private remoteTimestampsBySsrc: { [ssrc: number]: number } = {};\n  private remotePacketCountBySsrc: { [ssrc: number]: number } = {};\n  private remoteOctetCountBySsrc: { [ssrc: number]: number } = {};\n  private nackCountBySsrc: { [ssrc: number]: number } = {};\n  private pliCountBySsrc: { [ssrc: number]: number } = {};\n\n  constructor(\n    readonly config: PeerConfig,\n    public kind: Kind,\n    public rtcpSsrc: number,\n  ) {\n    this.defaultTrack = new MediaStreamTrack({ kind, remote: true });\n    this.onPacketLost.subscribe((nack) => {\n      this.nackCountBySsrc[nack.mediaSourceSsrc] =\n        (this.nackCountBySsrc[nack.mediaSourceSsrc] ?? 0) + 1;\n    });\n  }\n\n  get transport() {\n    return this.dtlsTransport ?? null;\n  }\n\n  setDtlsTransport(dtls: RTCDtlsTransport) {\n    this.dtlsTransport = dtls;\n  }\n\n  // todo fix\n  get track() {\n    return this.tracks[0] ?? this.defaultTrack;\n  }\n\n  get nackEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find((f) => f.type === \"nack\");\n  }\n\n  get twccEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find(\n      (f) => f.type === useTWCC().type,\n    );\n  }\n\n  get pliEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find(\n      (f) => f.type === usePLI().type,\n    );\n  }\n\n  prepareReceive(params: RTCRtpReceiveParameters) {\n    params.codecs.forEach((c) => {\n      this.codecs[c.payloadType] = c;\n    });\n    params.encodings.forEach((e) => {\n      if (e.rtx) {\n        this.ssrcByRtx[e.rtx.ssrc] = e.ssrc;\n      }\n    });\n  }\n\n  /**\n   * negotiated codec と receiver の実処理状態を再同期する。\n   * codec map と RTX mapping を置換し、既存 remote track の codec\n   * metadata も新しい先頭 codec に更新する。\n   * 既存の receiverTWCC は破棄し、新しい codec が transport-cc を持ち\n   * remote SSRC が分かる場合だけ再生成する。\n   * setCodecPreferences() 後の createAnswer() 再解決や remote からの\n   * 再交渉では、除外された payload type の RTP を受け付けないようにする。\n   */\n  resyncCodecs(params: RTCRtpReceiveParameters, mediaSourceSsrc?: number) {\n    for (const key of Object.keys(this.codecs)) {\n      delete this.codecs[Number(key)];\n    }\n    for (const key of Object.keys(this.ssrcByRtx)) {\n      delete this.ssrcByRtx[Number(key)];\n    }\n    this.prepareReceive(params);\n    const codec = deliveredTrackCodec(\n      this.kind,\n      params.codecs[0],\n      params.codecs,\n    );\n    if (codec) {\n      for (const track of this.tracks) {\n        track.codec = codec;\n      }\n    }\n    this.receiverTWCC = undefined;\n    if (mediaSourceSsrc != undefined) {\n      this.setupTWCC(mediaSourceSsrc);\n    }\n  }\n\n  /**\n   * setup TWCC if supported\n   */\n  setupTWCC(mediaSourceSsrc: number) {\n    if (this.twccEnabled && !this.receiverTWCC) {\n      this.receiverTWCC = new ReceiverTWCC(\n        this.dtlsTransport,\n        this.rtcpSsrc,\n        mediaSourceSsrc,\n      );\n    }\n  }\n\n  addTrack(track: MediaStreamTrack) {\n    const exist = this.tracks.find((t) => {\n      if (t.rid) {\n        return t.rid === track.rid;\n      }\n      if (t.ssrc) {\n        return t.ssrc === track.ssrc;\n      }\n    });\n    if (exist) {\n      return false;\n    }\n    this.tracks.push(track);\n    if (track.ssrc) {\n      this.trackBySSRC[track.ssrc] = track;\n    }\n    if (track.rid) {\n      this.trackByRID[track.rid] = track;\n    }\n    return true;\n  }\n\n  stop() {\n    this.stopped = true;\n    this.rtcpRunning = false;\n    this.rtcpCancel.abort();\n\n    if (this.receiverTWCC) this.receiverTWCC.twccRunning = false;\n    this.nack.close();\n  }\n\n  /**transceiver の停止確定時に remote track を ended にする */\n  endTracks() {\n    for (const track of [...this.tracks, this.defaultTrack]) {\n      track.stop();\n    }\n  }\n\n  async runRtcp() {\n    if (this.rtcpRunning || this.stopped) return;\n    this.rtcpRunning = true;\n\n    try {\n      while (this.rtcpRunning) {\n        await setTimeout(500 + Math.random() * 1000, undefined, {\n          signal: this.rtcpCancel.signal,\n        });\n\n        const reports = Object.entries(this.remoteStreams).map(\n          ([ssrc, stream]) => {\n            let lastSRtimestamp = 0,\n              delaySinceLastSR = 0;\n            if (this.lastSRtimestamp[ssrc]) {\n              lastSRtimestamp = this.lastSRtimestamp[ssrc];\n              const delaySeconds =\n                timestampSeconds() - this.receiveLastSRTimestamp[ssrc];\n              if (delaySeconds > 0 && delaySeconds < 65536) {\n                delaySinceLastSR = int(delaySeconds * 65536);\n              }\n            }\n\n            return new RtcpReceiverInfo({\n              ssrc: Number(ssrc),\n              fractionLost: stream.fraction_lost,\n              packetsLost: stream.packets_lost,\n              highestSequence: stream.max_seq,\n              jitter: stream.jitter,\n              lsr: lastSRtimestamp,\n              dlsr: delaySinceLastSR,\n            });\n          },\n        );\n\n        const packet = new RtcpRrPacket({ ssrc: this.rtcpSsrc, reports });\n\n        try {\n          if (this.config.debug.receiverReportDelay) {\n            await setTimeout(this.config.debug.receiverReportDelay);\n          }\n          await this.dtlsTransport.sendRtcp([packet]);\n        } catch (error) {\n          log(\"sendRtcp failed\", error);\n          await setTimeout(500 + Math.random() * 1000);\n        }\n      }\n    } catch (error) {}\n  }\n\n  private getInboundRtpStatsId(track: MediaStreamTrack) {\n    return generateStatsId(\"inbound-rtp\", track.id ?? track.uuid);\n  }\n\n  private getRemoteOutboundRtpStatsId(track: MediaStreamTrack) {\n    return generateStatsId(\"remote-outbound-rtp\", track.id ?? track.uuid);\n  }\n\n  getStatsRootIds(selector?: MediaStreamTrack) {\n    return this.tracks\n      .filter((track) => (!selector ? true : track === selector))\n      .filter((track) => track.ssrc)\n      .map((track) => this.getInboundRtpStatsId(track));\n  }\n\n  collectStats(timestamp: number): RTCStats[] {\n    const stats: RTCStats[] = [];\n    const transportId = this.dtlsTransport\n      ? generateStatsId(\"transport\", this.dtlsTransport.id)\n      : undefined;\n    const activeCodec = this.codecArray[0];\n    const emittedCodecIds = new Set<string>();\n\n    // Collect stats for each track\n    for (const track of this.tracks) {\n      if (!track.ssrc) continue;\n\n      const streamStats = this.remoteStreams[track.ssrc];\n\n      // Inbound RTP stats\n      const hasRemoteTimestamp =\n        this.remoteTimestampsBySsrc[track.ssrc] !== undefined;\n      const remoteId =\n        this.lastSRtimestamp[track.ssrc] !== undefined || hasRemoteTimestamp\n          ? this.getRemoteOutboundRtpStatsId(track)\n          : undefined;\n      const codecId =\n        activeCodec && transportId\n          ? generateCodecStatsId(\n              transportId,\n              activeCodec.payloadType,\n              track.id ?? track.uuid,\n            )\n          : undefined;\n\n      if (\n        activeCodec &&\n        transportId &&\n        codecId &&\n        !emittedCodecIds.has(codecId)\n      ) {\n        emittedCodecIds.add(codecId);\n        const codecStats: RTCCodecStats = {\n          type: \"codec\",\n          id: codecId,\n          timestamp,\n          payloadType: activeCodec.payloadType,\n          transportId,\n          mimeType: activeCodec.mimeType,\n          clockRate: activeCodec.clockRate,\n          channels: activeCodec.channels,\n          sdpFmtpLine: activeCodec.parameters,\n        };\n        stats.push(codecStats);\n      }\n\n      const inboundRtpStats: RTCInboundRtpStreamStats = {\n        type: \"inbound-rtp\",\n        id: this.getInboundRtpStatsId(track),\n        timestamp,\n        ssrc: track.ssrc,\n        kind: this.kind,\n        transportId,\n        codecId,\n        mid: this.sdesMid,\n        trackIdentifier: track.id ?? track.uuid,\n        packetsReceived: streamStats?.packets_received ?? 0,\n        bytesReceived: streamStats?.bytesReceived ?? 0,\n        headerBytesReceived: streamStats?.headerBytesReceived ?? 0,\n        packetsLost: streamStats?.packets_lost ?? 0,\n        jitter: streamStats?.clockRate\n          ? streamStats.jitter / streamStats.clockRate\n          : undefined,\n        lastPacketReceivedTimestamp: streamStats?.lastPacketReceivedTimestamp,\n        remoteId,\n        nackCount: this.nackCountBySsrc[track.ssrc] || undefined,\n        pliCount: this.pliCountBySsrc[track.ssrc] || undefined,\n      };\n      stats.push(inboundRtpStats);\n\n      // Remote outbound RTP stats (if we have SR info)\n      if (remoteId) {\n        const remoteOutboundStats: RTCRemoteOutboundRtpStreamStats = {\n          type: \"remote-outbound-rtp\",\n          id: this.getRemoteOutboundRtpStatsId(track),\n          timestamp,\n          ssrc: track.ssrc,\n          kind: this.kind,\n          transportId,\n          codecId: inboundRtpStats.codecId,\n          localId: inboundRtpStats.id,\n          remoteTimestamp: this.remoteTimestampsBySsrc[track.ssrc],\n          reportsSent: this.senderReportsReceivedBySsrc[track.ssrc] ?? 0,\n          packetsSent: this.remotePacketCountBySsrc[track.ssrc],\n          bytesSent: this.remoteOctetCountBySsrc[track.ssrc],\n        };\n        stats.push(remoteOutboundStats);\n      }\n    }\n\n    return stats;\n  }\n\n  async getStats(): Promise<RTCStatsReport> {\n    const timestamp = getStatsTimestamp();\n    const stats = this.collectStats(timestamp);\n\n    if (this.dtlsTransport) {\n      stats.push(...(await this.dtlsTransport.getStats(timestamp)));\n    }\n\n    return buildStatsReport(stats, this.getStatsRootIds());\n  }\n\n  async sendRtcpPLI(mediaSsrc: number) {\n    if (!this.pliEnabled) {\n      log(\"pli not supported\", { mediaSsrc });\n      return;\n    }\n\n    if (this.stopped) {\n      return;\n    }\n\n    log(\"sendRtcpPLI\", { mediaSsrc });\n\n    const packet = new RtcpPayloadSpecificFeedback({\n      feedback: new PictureLossIndication({\n        senderSsrc: this.rtcpSsrc,\n        mediaSsrc,\n      }),\n    });\n    try {\n      this.pliCountBySsrc[mediaSsrc] =\n        (this.pliCountBySsrc[mediaSsrc] ?? 0) + 1;\n      await this.dtlsTransport.sendRtcp([packet]);\n    } catch (error) {\n      log(error);\n    }\n  }\n\n  handleRtcpPacket(packet: RtcpPacket) {\n    switch (packet.type) {\n      case RtcpSrPacket.type:\n        {\n          const sr = packet as RtcpSrPacket;\n          this.lastSRtimestamp[sr.ssrc] = compactNtp(\n            sr.senderInfo.ntpTimestamp,\n          );\n          this.receiveLastSRTimestamp[sr.ssrc] = timestampSeconds();\n          this.senderReportsReceivedBySsrc[sr.ssrc] =\n            (this.senderReportsReceivedBySsrc[sr.ssrc] ?? 0) + 1;\n          this.remoteTimestampsBySsrc[sr.ssrc] = ntpTimeToEpochMs(\n            sr.senderInfo.ntpTimestamp,\n          );\n          this.remotePacketCountBySsrc[sr.ssrc] = sr.senderInfo.packetCount;\n          this.remoteOctetCountBySsrc[sr.ssrc] = sr.senderInfo.octetCount;\n\n          const track = this.trackBySSRC[packet.ssrc];\n          if (track) {\n            track.onReceiveRtcp.execute(packet);\n          }\n        }\n        break;\n    }\n    this.onRtcp.execute(packet);\n  }\n\n  handleRtpBySsrc = (packet: RtpPacket, extensions: Extensions) => {\n    const track = this.trackBySSRC[packet.header.ssrc];\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  handleRtpByRid = (packet: RtpPacket, rid: string, extensions: Extensions) => {\n    const track = this.trackByRID[rid];\n    if (!this.trackBySSRC[packet.header.ssrc]) {\n      this.trackBySSRC[packet.header.ssrc] = track;\n    }\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  private handleRTP(\n    packet: RtpPacket,\n    extensions: Extensions,\n    track?: MediaStreamTrack,\n  ) {\n    if (this.stopped) {\n      return;\n    }\n\n    const codec = this.codecs[packet.header.payloadType];\n    if (!codec) {\n      // log(\"unknown codec \" + packet.header.payloadType);\n      return;\n    }\n\n    this.remoteStreams[packet.header.ssrc] =\n      this.remoteStreams[packet.header.ssrc] ??\n      new StreamStatistics(codec.clockRate);\n    this.remoteStreams[packet.header.ssrc].add(packet);\n\n    if (this.receiverTWCC) {\n      const transportSequenceNumber = extensions[\n        RTP_EXTENSION_URI.transportWideCC\n      ] as TransportWideCCPayload;\n\n      if (!transportSequenceNumber == undefined) {\n        throw new Error(\"undefined\");\n      }\n\n      this.receiverTWCC.handleTWCC(transportSequenceNumber);\n    } else if (this.twccEnabled) {\n      this.setupTWCC(packet.header.ssrc);\n    }\n\n    if (codec.name.toLowerCase() === \"rtx\") {\n      const originalSsrc = this.ssrcByRtx[packet.header.ssrc];\n      const codecParams = codecParametersFromString(codec.parameters ?? \"\");\n      const rtxCodec = this.codecs[codecParams[\"apt\"]];\n      if (packet.payload.length < 2) return;\n\n      packet = unwrapRtx(packet, rtxCodec.payloadType, originalSsrc);\n      track = this.trackBySSRC[originalSsrc];\n    }\n\n    let red: Red | undefined;\n    if (codec.name.toLowerCase() === \"red\") {\n      red = Red.deSerialize(packet.payload);\n      if (\n        !Object.keys(this.codecs).includes(\n          red.header.fields[0].blockPT.toString(),\n        )\n      ) {\n        return;\n      }\n    }\n\n    if (track?.kind === \"video\" && this.nackEnabled) {\n      this.nack.addPacket(packet);\n    }\n\n    if (track) {\n      if (red) {\n        if (track.kind === \"audio\") {\n          const payloads = this.audioRedHandler.push(red, packet);\n          for (const packet of payloads) {\n            track.onReceiveRtp.execute(packet.clone(), extensions);\n          }\n        } else {\n        }\n      } else {\n        track.onReceiveRtp.execute(packet.clone(), extensions);\n      }\n    }\n\n    this.runRtcp();\n  }\n}\n"]}