werift 0.17.0 → 0.17.2

package/lib/{webrtc/src/media/receiver/red.js → rtp/src/rtp/red/handler.js}

@@ -1,6 +1,6 @@
 "use strict";
 Object.defineProperty(exports, "__esModule", { value: true });
-exports.AudioRedHandler = void 0;
+exports.RedHandler = void 0;
 const __1 = require("../..");
 // 0                   1                    2                   3
 // 0 1 2 3 4 5 6 7 8 9 0 1 2 3  4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
@@ -11,29 +11,29 @@ const __1 = require("../..");
 // +-+-+-+-+-+-+-+-+
 // |0|   Block PT  |
 // +-+-+-+-+-+-+-+-+
-class AudioRedHandler {
+class RedHandler {
     constructor() {
         this.size = 150;
         this.sequenceNumbers = [];
     }
-    push(red, rtp) {
+    push(red, base) {
         const packets = [];
         red.blocks.forEach(({ blockPT, timestampOffset, block }, i) => {
-            const sequenceNumber = (0, __1.uint16Add)(rtp.header.sequenceNumber, -(red.blocks.length - (i + 1)));
+            const sequenceNumber = (0, __1.uint16Add)(base.header.sequenceNumber, -(red.blocks.length - (i + 1)));
             if (timestampOffset) {
                 packets.push(new __1.RtpPacket(new __1.RtpHeader({
-                    timestamp: (0, __1.uint32Add)(rtp.header.timestamp, -timestampOffset),
+                    timestamp: (0, __1.uint32Add)(base.header.timestamp, -timestampOffset),
                     payloadType: blockPT,
-                    ssrc: rtp.header.ssrc,
+                    ssrc: base.header.ssrc,
                     sequenceNumber,
                     marker: true,
                 }), block));
             }
             else {
                 packets.push(new __1.RtpPacket(new __1.RtpHeader({
-                    timestamp: rtp.header.timestamp,
+                    timestamp: base.header.timestamp,
                     payloadType: blockPT,
-                    ssrc: rtp.header.ssrc,
+                    ssrc: base.header.ssrc,
                     sequenceNumber,
                     marker: true,
                 }), block));
@@ -56,5 +56,5 @@ class AudioRedHandler {
         return filtered;
     }
 }
-exports.AudioRedHandler = AudioRedHandler;
-//# sourceMappingURL=red.js.map
+exports.RedHandler = RedHandler;
+//# sourceMappingURL=handler.js.map

package/lib/rtp/src/rtp/red/handler.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"handler.js","sourceRoot":"","sources":["../../../../../../rtp/src/rtp/red/handler.ts"],"names":[],"mappings":";;;AAAA,6BAAwE;AAExE,iEAAiE;AACjE,mEAAmE;AACnE,oEAAoE;AACpE,oEAAoE;AACpE,oEAAoE;AAEpE,kBAAkB;AAClB,oBAAoB;AACpB,oBAAoB;AACpB,oBAAoB;AAEpB,MAAa,UAAU;IAAvB;QACmB,SAAI,GAAG,GAAG,CAAC;QACpB,oBAAe,GAAa,EAAE,CAAC;IAsDzC,CAAC;IApDC,IAAI,CAAC,GAAQ,EAAE,IAAe;QAC5B,MAAM,OAAO,GAAgB,EAAE,CAAC;QAEhC,GAAG,CAAC,MAAM,CAAC,OAAO,CAAC,CAAC,EAAE,OAAO,EAAE,eAAe,EAAE,KAAK,EAAE,EAAE,CAAC,EAAE,EAAE;YAC5D,MAAM,cAAc,GAAG,IAAA,aAAS,EAC9B,IAAI,CAAC,MAAM,CAAC,cAAc,EAC1B,CAAC,CAAC,GAAG,CAAC,MAAM,CAAC,MAAM,GAAG,CAAC,CAAC,GAAG,CAAC,CAAC,CAAC,CAC/B,CAAC;YACF,IAAI,eAAe,EAAE;gBACnB,OAAO,CAAC,IAAI,CACV,IAAI,aAAS,CACX,IAAI,aAAS,CAAC;oBACZ,SAAS,EAAE,IAAA,aAAS,EAAC,IAAI,CAAC,MAAM,CAAC,SAAS,EAAE,CAAC,eAAe,CAAC;oBAC7D,WAAW,EAAE,OAAO;oBACpB,IAAI,EAAE,IAAI,CAAC,MAAM,CAAC,IAAI;oBACtB,cAAc;oBACd,MAAM,EAAE,IAAI;iBACb,CAAC,EACF,KAAK,CACN,CACF,CAAC;aACH;iBAAM;gBACL,OAAO,CAAC,IAAI,CACV,IAAI,aAAS,CACX,IAAI,aAAS,CAAC;oBACZ,SAAS,EAAE,IAAI,CAAC,MAAM,CAAC,SAAS;oBAChC,WAAW,EAAE,OAAO;oBACpB,IAAI,EAAE,IAAI,CAAC,MAAM,CAAC,IAAI;oBACtB,cAAc;oBACd,MAAM,EAAE,IAAI;iBACb,CAAC,EACF,KAAK,CACN,CACF,CAAC;aACH;QACH,CAAC,CAAC,CAAC;QAEH,MAAM,QAAQ,GAAG,OAAO,CAAC,MAAM,CAAC,CAAC,CAAC,EAAE,EAAE;YACpC,YAAY;YACZ,IAAI,IAAI,CAAC,eAAe,CAAC,QAAQ,CAAC,CAAC,CAAC,MAAM,CAAC,cAAc,CAAC,EAAE;gBAC1D,OAAO,KAAK,CAAC;aACd;iBAAM;gBACL,kBAAkB;gBAClB,IAAI,IAAI,CAAC,eAAe,CAAC,MAAM,GAAG,IAAI,CAAC,IAAI,EAAE;oBAC3C,IAAI,CAAC,eAAe,CAAC,KAAK,EAAE,CAAC;iBAC9B;gBACD,IAAI,CAAC,eAAe,CAAC,IAAI,CAAC,CAAC,CAAC,MAAM,CAAC,cAAc,CAAC,CAAC;gBACnD,OAAO,IAAI,CAAC;aACb;QACH,CAAC,CAAC,CAAC;QACH,OAAO,QAAQ,CAAC;IAClB,CAAC;CACF;AAxDD,gCAwDC","sourcesContent":["import { Red, RtpHeader, RtpPacket, uint16Add, uint32Add } from \"../..\";\n\n// 0                   1                    2                   3\n// 0 1 2 3 4 5 6 7 8 9 0 1 2 3  4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1\n// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+\n// |F|   block PT  |  timestamp offset         |   block length    |\n// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+\n\n// 0 1 2 3 4 5 6 7\n// +-+-+-+-+-+-+-+-+\n// |0|   Block PT  |\n// +-+-+-+-+-+-+-+-+\n\nexport class RedHandler {\n  private readonly size = 150;\n  private sequenceNumbers: number[] = [];\n\n  push(red: Red, base: RtpPacket) {\n    const packets: RtpPacket[] = [];\n\n    red.blocks.forEach(({ blockPT, timestampOffset, block }, i) => {\n      const sequenceNumber = uint16Add(\n        base.header.sequenceNumber,\n        -(red.blocks.length - (i + 1))\n      );\n      if (timestampOffset) {\n        packets.push(\n          new RtpPacket(\n            new RtpHeader({\n              timestamp: uint32Add(base.header.timestamp, -timestampOffset),\n              payloadType: blockPT,\n              ssrc: base.header.ssrc,\n              sequenceNumber,\n              marker: true,\n            }),\n            block\n          )\n        );\n      } else {\n        packets.push(\n          new RtpPacket(\n            new RtpHeader({\n              timestamp: base.header.timestamp,\n              payloadType: blockPT,\n              ssrc: base.header.ssrc,\n              sequenceNumber,\n              marker: true,\n            }),\n            block\n          )\n        );\n      }\n    });\n\n    const filtered = packets.filter((p) => {\n      // duplicate\n      if (this.sequenceNumbers.includes(p.header.sequenceNumber)) {\n        return false;\n      } else {\n        // buffer overflow\n        if (this.sequenceNumbers.length > this.size) {\n          this.sequenceNumbers.shift();\n        }\n        this.sequenceNumbers.push(p.header.sequenceNumber);\n        return true;\n      }\n    });\n    return filtered;\n  }\n}\n"]}

package/lib/webrtc/src/const.d.ts

@@ -32,5 +32,5 @@ export declare const SRTP_PROFILE: {
     readonly SRTP_AEAD_AES_128_GCM: 7;
 };
 export declare const SenderDirections: string[];
-export declare const NotSenderDirections: string[];
+export declare const ReceiverDirection: string[];
 export { HashAlgorithms, SignatureAlgorithm };

package/lib/webrtc/src/const.js

@@ -1,6 +1,6 @@
 "use strict";
 Object.defineProperty(exports, "__esModule", { value: true });
-exports.SignatureAlgorithm = exports.NotSenderDirections = exports.SenderDirections = exports.SRTP_PROFILE = exports.SSRC_INFO_ATTRS = exports.FMTP_INT_PARAMETERS = exports.DTLS_SETUP_ROLE = exports.DTLS_ROLE_SETUP = exports.DIRECTIONS = exports.MEDIA_KINDS = exports.DISCARD_PORT = exports.DISCARD_HOST = exports.WEBRTC_BINARY_EMPTY = exports.WEBRTC_STRING_EMPTY = exports.WEBRTC_BINARY = exports.WEBRTC_STRING = exports.WEBRTC_DCEP = exports.DATA_CHANNEL_PARTIAL_RELIABLE_TIMED_UNORDERED = exports.DATA_CHANNEL_PARTIAL_RELIABLE_REXMIT_UNORDERED = exports.DATA_CHANNEL_RELIABLE_UNORDERED = exports.DATA_CHANNEL_PARTIAL_RELIABLE_TIMED = exports.DATA_CHANNEL_PARTIAL_RELIABLE_REXMIT = exports.DATA_CHANNEL_RELIABLE = exports.DATA_CHANNEL_OPEN = exports.DATA_CHANNEL_ACK = void 0;
+exports.SignatureAlgorithm = exports.ReceiverDirection = exports.SenderDirections = exports.SRTP_PROFILE = exports.SSRC_INFO_ATTRS = exports.FMTP_INT_PARAMETERS = exports.DTLS_SETUP_ROLE = exports.DTLS_ROLE_SETUP = exports.DIRECTIONS = exports.MEDIA_KINDS = exports.DISCARD_PORT = exports.DISCARD_HOST = exports.WEBRTC_BINARY_EMPTY = exports.WEBRTC_STRING_EMPTY = exports.WEBRTC_BINARY = exports.WEBRTC_STRING = exports.WEBRTC_DCEP = exports.DATA_CHANNEL_PARTIAL_RELIABLE_TIMED_UNORDERED = exports.DATA_CHANNEL_PARTIAL_RELIABLE_REXMIT_UNORDERED = exports.DATA_CHANNEL_RELIABLE_UNORDERED = exports.DATA_CHANNEL_PARTIAL_RELIABLE_TIMED = exports.DATA_CHANNEL_PARTIAL_RELIABLE_REXMIT = exports.DATA_CHANNEL_RELIABLE = exports.DATA_CHANNEL_OPEN = exports.DATA_CHANNEL_ACK = void 0;
 const const_1 = require("../../dtls/src/cipher/const");
 Object.defineProperty(exports, "SignatureAlgorithm", { enumerable: true, get: function () { return const_1.SignatureAlgorithm; } });
 const srtp_1 = require("../../dtls/src/context/srtp");
@@ -48,5 +48,5 @@ exports.SRTP_PROFILE = {
     SRTP_AEAD_AES_128_GCM: srtp_1.ProtectionProfileAeadAes128Gcm,
 };
 exports.SenderDirections = ["sendonly", "sendrecv"];
-exports.NotSenderDirections = ["inactive", "recvonly"];
+exports.ReceiverDirection = ["sendrecv", "recvonly"];
 //# sourceMappingURL=const.js.map

package/lib/webrtc/src/const.js.map

@@ -1 +1 @@
-{"version":3,"file":"const.js","sourceRoot":"","sources":["../../../src/const.ts"],"names":[],"mappings":";;;AAAA,uDAGqC;AA8DZ,mGA/DvB,0BAAkB,OA+DuB;AA7D3C,sDAGqC;AAGrC,gCAAgC;AACnB,QAAA,gBAAgB,GAAG,CAAC,CAAC;AACrB,QAAA,iBAAiB,GAAG,CAAC,CAAC;AAEnC,kCAAkC;AACrB,QAAA,qBAAqB,GAAG,IAAI,CAAC;AAC7B,QAAA,oCAAoC,GAAG,IAAI,CAAC;AAC5C,QAAA,mCAAmC,GAAG,IAAI,CAAC;AAC3C,QAAA,+BAA+B,GAAG,IAAI,CAAC;AACvC,QAAA,8CAA8C,GAAG,IAAI,CAAC;AACtD,QAAA,6CAA6C,GAAG,IAAI,CAAC;AAErD,QAAA,WAAW,GAAG,EAAE,CAAC;AACjB,QAAA,aAAa,GAAG,EAAE,CAAC;AACnB,QAAA,aAAa,GAAG,EAAE,CAAC;AACnB,QAAA,mBAAmB,GAAG,EAAE,CAAC;AACzB,QAAA,mBAAmB,GAAG,EAAE,CAAC;AAEzB,QAAA,YAAY,GAAG,SAAS,CAAC;AACzB,QAAA,YAAY,GAAG,CAAC,CAAC;AACjB,QAAA,WAAW,GAAG,CAAC,OAAO,EAAE,OAAO,CAAC,CAAC;AAEjC,QAAA,UAAU,GAAG,CAAC,UAAU,EAAE,UAAU,EAAE,UAAU,EAAE,UAAU,CAAC,CAAC;AAC9D,QAAA,eAAe,GAAG;IAC7B,IAAI,EAAE,SAAS;IACf,MAAM,EAAE,QAAQ;IAChB,MAAM,EAAE,SAAS;CAClB,CAAC;AACW,QAAA,eAAe,GAAgC,MAAM,CAAC,IAAI,CACrE,uBAAe,CAChB,CAAC,MAAM,CAAC,CAAC,GAAG,EAAE,GAAG,EAAE,EAAE;IACpB,MAAM,GAAG,GAAI,uBAAuB,CAAC,GAAG,CAAC,CAAC;IAC1C,GAAG,CAAC,GAAG,CAAC,GAAG,GAAG,CAAC;IACf,OAAO,GAAG,CAAC;AACb,CAAC,EAAE,EAAS,CAAC,CAAC;AACD,QAAA,mBAAmB,GAAG;IACjC,KAAK;IACL,QAAQ;IACR,QAAQ;IACR,iBAAiB;IACjB,UAAU;IACV,QAAQ;IACR,cAAc;CACf,CAAC;AAEW,QAAA,eAAe,GAAG,CAAC,OAAO,EAAE,MAAM,EAAE,SAAS,EAAE,OAAO,CAAC,CAAC;AAExD,QAAA,YAAY,GAAG;IAC1B,2BAA2B,EAAE,2CAAoC;IACjE,qBAAqB,EAAE,qCAA8B;CAC7C,CAAC;AAEE,QAAA,gBAAgB,GAAG,CAAC,UAAU,EAAE,UAAU,CAAC,CAAC;AAC5C,QAAA,mBAAmB,GAAG,CAAC,UAAU,EAAE,UAAU,CAAC,CAAC","sourcesContent":["import {\n  HashAlgorithms,\n  SignatureAlgorithm,\n} from \"../../dtls/src/cipher/const\";\nimport {\n  ProtectionProfileAeadAes128Gcm,\n  ProtectionProfileAes128CmHmacSha1_80,\n} from \"../../dtls/src/context/srtp\";\nimport { DtlsRole } from \"./transport/dtls\";\n\n// data channel export constants\nexport const DATA_CHANNEL_ACK = 2;\nexport const DATA_CHANNEL_OPEN = 3;\n\n// 5.1.  DATA_CHANNEL_OPEN Message\nexport const DATA_CHANNEL_RELIABLE = 0x00;\nexport const DATA_CHANNEL_PARTIAL_RELIABLE_REXMIT = 0x01;\nexport const DATA_CHANNEL_PARTIAL_RELIABLE_TIMED = 0x02;\nexport const DATA_CHANNEL_RELIABLE_UNORDERED = 0x80;\nexport const DATA_CHANNEL_PARTIAL_RELIABLE_REXMIT_UNORDERED = 0x81;\nexport const DATA_CHANNEL_PARTIAL_RELIABLE_TIMED_UNORDERED = 0x82;\n\nexport const WEBRTC_DCEP = 50;\nexport const WEBRTC_STRING = 51;\nexport const WEBRTC_BINARY = 53;\nexport const WEBRTC_STRING_EMPTY = 56;\nexport const WEBRTC_BINARY_EMPTY = 57;\n\nexport const DISCARD_HOST = \"0.0.0.0\";\nexport const DISCARD_PORT = 9;\nexport const MEDIA_KINDS = [\"audio\", \"video\"];\n\nexport const DIRECTIONS = [\"inactive\", \"sendonly\", \"recvonly\", \"sendrecv\"];\nexport const DTLS_ROLE_SETUP = {\n  auto: \"actpass\",\n  client: \"active\",\n  server: \"passive\",\n};\nexport const DTLS_SETUP_ROLE: { [key: string]: DtlsRole } = Object.keys(\n  DTLS_ROLE_SETUP\n).reduce((acc, cur) => {\n  const key = (DTLS_ROLE_SETUP as any)[cur];\n  acc[key] = cur;\n  return acc;\n}, {} as any);\nexport const FMTP_INT_PARAMETERS = [\n  \"apt\",\n  \"max-fr\",\n  \"max-fs\",\n  \"maxplaybackrate\",\n  \"minptime\",\n  \"stereo\",\n  \"useinbandfec\",\n];\n\nexport const SSRC_INFO_ATTRS = [\"cname\", \"msid\", \"mslabel\", \"label\"];\n\nexport const SRTP_PROFILE = {\n  SRTP_AES128_CM_HMAC_SHA1_80: ProtectionProfileAes128CmHmacSha1_80,\n  SRTP_AEAD_AES_128_GCM: ProtectionProfileAeadAes128Gcm,\n} as const;\n\nexport const SenderDirections = [\"sendonly\", \"sendrecv\"];\nexport const NotSenderDirections = [\"inactive\", \"recvonly\"];\n\nexport { HashAlgorithms, SignatureAlgorithm };\n"]}
+{"version":3,"file":"const.js","sourceRoot":"","sources":["../../../src/const.ts"],"names":[],"mappings":";;;AAAA,uDAGqC;AA8DZ,mGA/DvB,0BAAkB,OA+DuB;AA7D3C,sDAGqC;AAGrC,gCAAgC;AACnB,QAAA,gBAAgB,GAAG,CAAC,CAAC;AACrB,QAAA,iBAAiB,GAAG,CAAC,CAAC;AAEnC,kCAAkC;AACrB,QAAA,qBAAqB,GAAG,IAAI,CAAC;AAC7B,QAAA,oCAAoC,GAAG,IAAI,CAAC;AAC5C,QAAA,mCAAmC,GAAG,IAAI,CAAC;AAC3C,QAAA,+BAA+B,GAAG,IAAI,CAAC;AACvC,QAAA,8CAA8C,GAAG,IAAI,CAAC;AACtD,QAAA,6CAA6C,GAAG,IAAI,CAAC;AAErD,QAAA,WAAW,GAAG,EAAE,CAAC;AACjB,QAAA,aAAa,GAAG,EAAE,CAAC;AACnB,QAAA,aAAa,GAAG,EAAE,CAAC;AACnB,QAAA,mBAAmB,GAAG,EAAE,CAAC;AACzB,QAAA,mBAAmB,GAAG,EAAE,CAAC;AAEzB,QAAA,YAAY,GAAG,SAAS,CAAC;AACzB,QAAA,YAAY,GAAG,CAAC,CAAC;AACjB,QAAA,WAAW,GAAG,CAAC,OAAO,EAAE,OAAO,CAAC,CAAC;AAEjC,QAAA,UAAU,GAAG,CAAC,UAAU,EAAE,UAAU,EAAE,UAAU,EAAE,UAAU,CAAC,CAAC;AAC9D,QAAA,eAAe,GAAG;IAC7B,IAAI,EAAE,SAAS;IACf,MAAM,EAAE,QAAQ;IAChB,MAAM,EAAE,SAAS;CAClB,CAAC;AACW,QAAA,eAAe,GAAgC,MAAM,CAAC,IAAI,CACrE,uBAAe,CAChB,CAAC,MAAM,CAAC,CAAC,GAAG,EAAE,GAAG,EAAE,EAAE;IACpB,MAAM,GAAG,GAAI,uBAAuB,CAAC,GAAG,CAAC,CAAC;IAC1C,GAAG,CAAC,GAAG,CAAC,GAAG,GAAG,CAAC;IACf,OAAO,GAAG,CAAC;AACb,CAAC,EAAE,EAAS,CAAC,CAAC;AACD,QAAA,mBAAmB,GAAG;IACjC,KAAK;IACL,QAAQ;IACR,QAAQ;IACR,iBAAiB;IACjB,UAAU;IACV,QAAQ;IACR,cAAc;CACf,CAAC;AAEW,QAAA,eAAe,GAAG,CAAC,OAAO,EAAE,MAAM,EAAE,SAAS,EAAE,OAAO,CAAC,CAAC;AAExD,QAAA,YAAY,GAAG;IAC1B,2BAA2B,EAAE,2CAAoC;IACjE,qBAAqB,EAAE,qCAA8B;CAC7C,CAAC;AAEE,QAAA,gBAAgB,GAAG,CAAC,UAAU,EAAE,UAAU,CAAC,CAAC;AAC5C,QAAA,iBAAiB,GAAG,CAAC,UAAU,EAAE,UAAU,CAAC,CAAC","sourcesContent":["import {\n  HashAlgorithms,\n  SignatureAlgorithm,\n} from \"../../dtls/src/cipher/const\";\nimport {\n  ProtectionProfileAeadAes128Gcm,\n  ProtectionProfileAes128CmHmacSha1_80,\n} from \"../../dtls/src/context/srtp\";\nimport { DtlsRole } from \"./transport/dtls\";\n\n// data channel export constants\nexport const DATA_CHANNEL_ACK = 2;\nexport const DATA_CHANNEL_OPEN = 3;\n\n// 5.1.  DATA_CHANNEL_OPEN Message\nexport const DATA_CHANNEL_RELIABLE = 0x00;\nexport const DATA_CHANNEL_PARTIAL_RELIABLE_REXMIT = 0x01;\nexport const DATA_CHANNEL_PARTIAL_RELIABLE_TIMED = 0x02;\nexport const DATA_CHANNEL_RELIABLE_UNORDERED = 0x80;\nexport const DATA_CHANNEL_PARTIAL_RELIABLE_REXMIT_UNORDERED = 0x81;\nexport const DATA_CHANNEL_PARTIAL_RELIABLE_TIMED_UNORDERED = 0x82;\n\nexport const WEBRTC_DCEP = 50;\nexport const WEBRTC_STRING = 51;\nexport const WEBRTC_BINARY = 53;\nexport const WEBRTC_STRING_EMPTY = 56;\nexport const WEBRTC_BINARY_EMPTY = 57;\n\nexport const DISCARD_HOST = \"0.0.0.0\";\nexport const DISCARD_PORT = 9;\nexport const MEDIA_KINDS = [\"audio\", \"video\"];\n\nexport const DIRECTIONS = [\"inactive\", \"sendonly\", \"recvonly\", \"sendrecv\"];\nexport const DTLS_ROLE_SETUP = {\n  auto: \"actpass\",\n  client: \"active\",\n  server: \"passive\",\n};\nexport const DTLS_SETUP_ROLE: { [key: string]: DtlsRole } = Object.keys(\n  DTLS_ROLE_SETUP\n).reduce((acc, cur) => {\n  const key = (DTLS_ROLE_SETUP as any)[cur];\n  acc[key] = cur;\n  return acc;\n}, {} as any);\nexport const FMTP_INT_PARAMETERS = [\n  \"apt\",\n  \"max-fr\",\n  \"max-fs\",\n  \"maxplaybackrate\",\n  \"minptime\",\n  \"stereo\",\n  \"useinbandfec\",\n];\n\nexport const SSRC_INFO_ATTRS = [\"cname\", \"msid\", \"mslabel\", \"label\"];\n\nexport const SRTP_PROFILE = {\n  SRTP_AES128_CM_HMAC_SHA1_80: ProtectionProfileAes128CmHmacSha1_80,\n  SRTP_AEAD_AES_128_GCM: ProtectionProfileAeadAes128Gcm,\n} as const;\n\nexport const SenderDirections = [\"sendonly\", \"sendrecv\"];\nexport const ReceiverDirection = [\"sendrecv\", \"recvonly\"];\n\nexport { HashAlgorithms, SignatureAlgorithm };\n"]}

package/lib/webrtc/src/media/receiver/nack.d.ts

@@ -10,6 +10,7 @@ export declare class NackHandler {
     mediaSourceSsrc?: number;
     retryCount: number;
     constructor(receiver: RTCRtpReceiver);
+    private start;
     get lostSeqNumbers(): number[];
     getLost(seq: number): number;
     setLost(seq: number, count: number): void;

package/lib/webrtc/src/media/receiver/nack.js

@@ -16,9 +16,36 @@ class NackHandler {
         this.receiver = receiver;
         this.newEstSeqNum = 0;
         this._lost = {};
-        this.nackLoop = setInterval(() => this.sendNack(), 20);
         this.onPacketLost = new rx_mini_1.default();
         this.retryCount = 10;
+        this.sendNack = () => new Promise((r, f) => {
+            if (this.lostSeqNumbers.length > 0 && this.mediaSourceSsrc) {
+                const nack = new src_2.GenericNack({
+                    senderSsrc: this.receiver.rtcpSsrc,
+                    mediaSourceSsrc: this.mediaSourceSsrc,
+                    lost: this.lostSeqNumbers,
+                });
+                const rtcp = new src_2.RtcpTransportLayerFeedback({
+                    feedback: nack,
+                });
+                this.receiver.dtlsTransport.sendRtcp([rtcp]).then(r).catch(f);
+                this.updateRetryCount();
+                this.onPacketLost.execute(nack);
+            }
+        });
+    }
+    start() {
+        if (this.nackLoop) {
+            return;
+        }
+        this.nackLoop = setInterval(async () => {
+            try {
+                await this.sendNack();
+            }
+            catch (error) {
+                log("failed to send nack", error);
+            }
+        }, 20);
     }
     get lostSeqNumbers() {
         return Object.keys(this._lost).map(Number).sort();
@@ -33,6 +60,7 @@ class NackHandler {
         delete this._lost[sequenceNumber];
     }
     addPacket(packet) {
+        this.start();
         const { sequenceNumber, ssrc } = packet.header;
         this.mediaSourceSsrc = ssrc;
         if (this.newEstSeqNum === 0) {
@@ -79,21 +107,6 @@ class NackHandler {
             }
         });
     }
-    sendNack() {
-        if (this.lostSeqNumbers.length > 0 && this.mediaSourceSsrc) {
-            const nack = new src_2.GenericNack({
-                senderSsrc: this.receiver.rtcpSsrc,
-                mediaSourceSsrc: this.mediaSourceSsrc,
-                lost: this.lostSeqNumbers,
-            });
-            const rtcp = new src_2.RtcpTransportLayerFeedback({
-                feedback: nack,
-            });
-            this.receiver.dtlsTransport.sendRtcp([rtcp]);
-            this.updateRetryCount();
-            this.onPacketLost.execute(nack);
-        }
-    }
 }
 exports.NackHandler = NackHandler;
 //# sourceMappingURL=nack.js.map

package/lib/webrtc/src/media/receiver/nack.js.map

@@ -1 +1 @@
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debug from \"debug\";\nimport range from \"lodash/range\";\nimport Event from \"rx.mini\";\n\nimport { uint16Add } from \"../../../../common/src\";\nimport {\n  GenericNack,\n  RtcpTransportLayerFeedback,\n  RtpPacket,\n} from \"../../../../rtp/src\";\nimport { RTCRtpReceiver } from \"../rtpReceiver\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/receiver/nack.ts\");\n\nconst LOST_SIZE = 30 * 5;\n\nexport class NackHandler {\n  private newEstSeqNum = 0;\n  private _lost: { [seqNum: number]: number } = {};\n  private nackLoop = setInterval(() => this.sendNack(), 20);\n\n  readonly onPacketLost = new Event<[GenericNack]>();\n  mediaSourceSsrc?: number;\n  retryCount = 10;\n\n  constructor(private receiver: RTCRtpReceiver) {}\n\n  get lostSeqNumbers() {\n    return Object.keys(this._lost).map(Number).sort();\n  }\n\n  getLost(seq: number) {\n    return this._lost[seq];\n  }\n\n  setLost(seq: number, count: number) {\n    this._lost[seq] = count;\n  }\n\n  removeLost(sequenceNumber: number) {\n    delete this._lost[sequenceNumber];\n  }\n\n  addPacket(packet: RtpPacket) {\n    const { sequenceNumber, ssrc } = packet.header;\n    this.mediaSourceSsrc = ssrc;\n\n    if (this.newEstSeqNum === 0) {\n      this.newEstSeqNum = sequenceNumber;\n      return;\n    }\n\n    if (this.getLost(sequenceNumber)) {\n      this.removeLost(sequenceNumber);\n      return;\n    }\n\n    if (sequenceNumber === uint16Add(this.newEstSeqNum, 1)) {\n      this.newEstSeqNum = sequenceNumber;\n    } else if (sequenceNumber > uint16Add(this.newEstSeqNum, 1)) {\n      // packet lost detected\n      range(uint16Add(this.newEstSeqNum, 1), sequenceNumber).forEach((seq) => {\n        this.setLost(seq, 1);\n      });\n      this.receiver.sendRtcpPLI(this.mediaSourceSsrc);\n\n      this.newEstSeqNum = sequenceNumber;\n      this.pruneLost();\n    }\n  }\n\n  private pruneLost() {\n    if (this.lostSeqNumbers.length > LOST_SIZE) {\n      this._lost = Object.entries(this._lost)\n        .slice(-LOST_SIZE)\n        .reduce((acc, [key, v]) => {\n          acc[key] = v;\n          return acc;\n        }, {} as { [seqNum: number]: number });\n    }\n  }\n\n  close() {\n    clearInterval(this.nackLoop);\n    this._lost = {};\n  }\n\n  private updateRetryCount() {\n    this.lostSeqNumbers.forEach((seq) => {\n      const count = this._lost[seq]++;\n      if (count > this.retryCount) {\n        this.removeLost(seq);\n        return seq;\n      }\n    });\n  }\n\n  private sendNack() {\n    if (this.lostSeqNumbers.length > 0 && this.mediaSourceSsrc) {\n      const nack = new GenericNack({\n        senderSsrc: this.receiver.rtcpSsrc,\n        mediaSourceSsrc: this.mediaSourceSsrc,\n        lost: this.lostSeqNumbers,\n      });\n      const rtcp = new RtcpTransportLayerFeedback({\n        feedback: nack,\n      });\n      this.receiver.dtlsTransport.sendRtcp([rtcp]);\n\n      this.updateRetryCount();\n      this.onPacketLost.execute(nack);\n    }\n  }\n}\n"]}
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debug from \"debug\";\nimport range from \"lodash/range\";\nimport Event from \"rx.mini\";\n\nimport { uint16Add } from \"../../../../common/src\";\nimport {\n  GenericNack,\n  RtcpTransportLayerFeedback,\n  RtpPacket,\n} from \"../../../../rtp/src\";\nimport { RTCRtpReceiver } from \"../rtpReceiver\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/receiver/nack.ts\");\n\nconst LOST_SIZE = 30 * 5;\n\nexport class NackHandler {\n  private newEstSeqNum = 0;\n  private _lost: { [seqNum: number]: number } = {};\n  private nackLoop: any;\n\n  readonly onPacketLost = new Event<[GenericNack]>();\n  mediaSourceSsrc?: number;\n  retryCount = 10;\n\n  constructor(private receiver: RTCRtpReceiver) {}\n\n  private start() {\n    if (this.nackLoop) {\n      return;\n    }\n    this.nackLoop = setInterval(async () => {\n      try {\n        await this.sendNack();\n      } catch (error) {\n        log(\"failed to send nack\", error);\n      }\n    }, 20);\n  }\n\n  get lostSeqNumbers() {\n    return Object.keys(this._lost).map(Number).sort();\n  }\n\n  getLost(seq: number) {\n    return this._lost[seq];\n  }\n\n  setLost(seq: number, count: number) {\n    this._lost[seq] = count;\n  }\n\n  removeLost(sequenceNumber: number) {\n    delete this._lost[sequenceNumber];\n  }\n\n  addPacket(packet: RtpPacket) {\n    this.start();\n\n    const { sequenceNumber, ssrc } = packet.header;\n    this.mediaSourceSsrc = ssrc;\n\n    if (this.newEstSeqNum === 0) {\n      this.newEstSeqNum = sequenceNumber;\n      return;\n    }\n\n    if (this.getLost(sequenceNumber)) {\n      this.removeLost(sequenceNumber);\n      return;\n    }\n\n    if (sequenceNumber === uint16Add(this.newEstSeqNum, 1)) {\n      this.newEstSeqNum = sequenceNumber;\n    } else if (sequenceNumber > uint16Add(this.newEstSeqNum, 1)) {\n      // packet lost detected\n      range(uint16Add(this.newEstSeqNum, 1), sequenceNumber).forEach((seq) => {\n        this.setLost(seq, 1);\n      });\n      this.receiver.sendRtcpPLI(this.mediaSourceSsrc);\n\n      this.newEstSeqNum = sequenceNumber;\n      this.pruneLost();\n    }\n  }\n\n  private pruneLost() {\n    if (this.lostSeqNumbers.length > LOST_SIZE) {\n      this._lost = Object.entries(this._lost)\n        .slice(-LOST_SIZE)\n        .reduce((acc, [key, v]) => {\n          acc[key] = v;\n          return acc;\n        }, {} as { [seqNum: number]: number });\n    }\n  }\n\n  close() {\n    clearInterval(this.nackLoop);\n    this._lost = {};\n  }\n\n  private updateRetryCount() {\n    this.lostSeqNumbers.forEach((seq) => {\n      const count = this._lost[seq]++;\n      if (count > this.retryCount) {\n        this.removeLost(seq);\n        return seq;\n      }\n    });\n  }\n\n  private sendNack = () =>\n    new Promise((r, f) => {\n      if (this.lostSeqNumbers.length > 0 && this.mediaSourceSsrc) {\n        const nack = new GenericNack({\n          senderSsrc: this.receiver.rtcpSsrc,\n          mediaSourceSsrc: this.mediaSourceSsrc,\n          lost: this.lostSeqNumbers,\n        });\n        const rtcp = new RtcpTransportLayerFeedback({\n          feedback: nack,\n        });\n        this.receiver.dtlsTransport.sendRtcp([rtcp]).then(r).catch(f);\n\n        this.updateRetryCount();\n        this.onPacketLost.execute(nack);\n      }\n    });\n}\n"]}

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

@@ -18,38 +18,41 @@ class RtpRouter {
         this.extIdUriMap = {};
         this.routeRtp = (packet) => {
             const extensions = RtpRouter.rtpHeaderExtensionsParser(packet.header.extensions, this.extIdUriMap);
-            let ssrcReceiver = this.ssrcTable[packet.header.ssrc];
+            let rtpReceiver = this.ssrcTable[packet.header.ssrc];
             const rid = extensions[rtpExtension_1.RTP_EXTENSION_URI.sdesRTPStreamID];
             if (typeof rid === "string") {
-                ssrcReceiver = this.ridTable[rid];
-                ssrcReceiver.latestRid = rid;
-                ssrcReceiver.handleRtpByRid(packet, rid, extensions);
+                rtpReceiver = this.ridTable[rid];
+                rtpReceiver.latestRid = rid;
+                rtpReceiver.handleRtpByRid(packet, rid, extensions);
             }
-            else if (ssrcReceiver) {
-                ssrcReceiver.handleRtpBySsrc(packet, extensions);
+            else if (rtpReceiver) {
+                rtpReceiver.handleRtpBySsrc(packet, extensions);
             }
             else {
                 // simulcast after send receiver report
-                ssrcReceiver = Object.values(this.ridTable)
+                rtpReceiver = Object.values(this.ridTable)
                     .filter((r) => r instanceof rtpReceiver_1.RTCRtpReceiver)
                     .find((r) => r.trackBySSRC[packet.header.ssrc]);
-                if (ssrcReceiver) {
+                if (rtpReceiver) {
                     log("simulcast register receiver by ssrc", packet.header.ssrc);
-                    this.registerRtpReceiver(ssrcReceiver, packet.header.ssrc);
-                    ssrcReceiver.handleRtpBySsrc(packet, extensions);
+                    this.registerRtpReceiver(rtpReceiver, packet.header.ssrc);
+                    rtpReceiver.handleRtpBySsrc(packet, extensions);
+                }
+                else {
+                    // bug
                 }
             }
-            if (!ssrcReceiver) {
+            if (!rtpReceiver) {
                 log("ssrcReceiver not found");
                 return;
             }
             const sdesMid = extensions[rtpExtension_1.RTP_EXTENSION_URI.sdesMid];
             if (typeof sdesMid === "string") {
-                ssrcReceiver.sdesMid = sdesMid;
+                rtpReceiver.sdesMid = sdesMid;
             }
             const repairedRid = extensions[rtpExtension_1.RTP_EXTENSION_URI.repairedRtpStreamId];
             if (typeof repairedRid === "string") {
-                ssrcReceiver.latestRepairedRid = repairedRid;
+                rtpReceiver.latestRepairedRid = repairedRid;
             }
         };
         this.routeRtcp = (packet) => {
@@ -106,6 +109,7 @@ class RtpRouter {
         this.ssrcTable[sender.ssrc] = sender;
     }
     registerRtpReceiver(receiver, ssrc) {
+        log("registerRtpReceiver", ssrc);
         this.ssrcTable[ssrc] = receiver;
     }
     registerRtpReceiverBySsrc(transceiver, params) {

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

@@ -1 +1 @@
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debug from \"debug\";\n\nimport { bufferReader } from \"../../../common/src\";\nimport {\n  Extension,\n  ReceiverEstimatedMaxBitrate,\n  RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpRrPacket,\n  RtcpSourceDescriptionPacket,\n  RtcpSrPacket,\n  RtcpTransportLayerFeedback,\n  RtpPacket,\n} from \"../../../rtp/src\";\nimport { RTP_EXTENSION_URI } from \"./extension/rtpExtension\";\nimport {\n  RTCRtpReceiveParameters,\n  RTCRtpSimulcastParameters,\n} from \"./parameters\";\nimport { RTCRtpReceiver } from \"./rtpReceiver\";\nimport { RTCRtpSender } from \"./rtpSender\";\nimport { RTCRtpTransceiver } from \"./rtpTransceiver\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/router.ts\");\n\nexport type Extensions = { [uri: string]: number | string };\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    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  static rtpHeaderExtensionsParser(\n    extensions: Extension[],\n    extIdUriMap: { [id: number]: string }\n  ): Extensions {\n    return extensions\n      .map((extension) => {\n        const uri = extIdUriMap[extension.id];\n        switch (uri) {\n          case RTP_EXTENSION_URI.sdesMid:\n          case RTP_EXTENSION_URI.sdesRTPStreamID:\n          case RTP_EXTENSION_URI.repairedRtpStreamId:\n            return { uri, value: extension.payload.toString() };\n          case RTP_EXTENSION_URI.transportWideCC:\n            return { uri, value: extension.payload.readUInt16BE() };\n          case RTP_EXTENSION_URI.absSendTime:\n            return {\n              uri,\n              value: bufferReader(extension.payload, [3])[0],\n            };\n          default:\n            return { uri, value: 0 };\n        }\n      })\n      .reduce((acc: { [uri: string]: any }, cur) => {\n        if (cur) acc[cur.uri] = cur.value;\n        return acc;\n      }, {});\n  }\n\n  routeRtp = (packet: RtpPacket) => {\n    const extensions = RtpRouter.rtpHeaderExtensionsParser(\n      packet.header.extensions,\n      this.extIdUriMap\n    );\n\n    let ssrcReceiver: 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      ssrcReceiver = this.ridTable[rid] as RTCRtpReceiver;\n      ssrcReceiver.latestRid = rid;\n      ssrcReceiver.handleRtpByRid(packet, rid, extensions);\n    } else if (ssrcReceiver) {\n      ssrcReceiver.handleRtpBySsrc(packet, extensions);\n    } else {\n      // simulcast after send receiver report\n      ssrcReceiver = Object.values(this.ridTable)\n        .filter((r): r is RTCRtpReceiver => r instanceof RTCRtpReceiver)\n        .find((r) => r.trackBySSRC[packet.header.ssrc]);\n      if (ssrcReceiver) {\n        log(\"simulcast register receiver by ssrc\", packet.header.ssrc);\n        this.registerRtpReceiver(ssrcReceiver, packet.header.ssrc);\n        ssrcReceiver.handleRtpBySsrc(packet, extensions);\n      }\n    }\n\n    if (!ssrcReceiver) {\n      log(\"ssrcReceiver not found\");\n      return;\n    }\n\n    const sdesMid = extensions[RTP_EXTENSION_URI.sdesMid];\n    if (typeof sdesMid === \"string\") {\n      ssrcReceiver.sdesMid = sdesMid;\n    }\n\n    const repairedRid = extensions[\n      RTP_EXTENSION_URI.repairedRtpStreamId\n    ] as string;\n    if (typeof repairedRid === \"string\") {\n      ssrcReceiver.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              const remb = psfb.feedback as ReceiverEstimatedMaxBitrate;\n              recipients.push(this.ssrcTable[remb.ssrcFeedbacks[0]]);\n              break;\n            default:\n              recipients.push(this.ssrcTable[psfb.feedback.senderSsrc]);\n          }\n        }\n        break;\n    }\n    recipients\n      .filter((v) => v) // todo simulcast\n      .forEach((recipient) => recipient.handleRtcpPacket(packet));\n  };\n}\n"]}
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debug from \"debug\";\n\nimport { bufferReader } from \"../../../common/src\";\nimport {\n  Extension,\n  ReceiverEstimatedMaxBitrate,\n  RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpRrPacket,\n  RtcpSourceDescriptionPacket,\n  RtcpSrPacket,\n  RtcpTransportLayerFeedback,\n  RtpPacket,\n} from \"../../../rtp/src\";\nimport { RTP_EXTENSION_URI } from \"./extension/rtpExtension\";\nimport {\n  RTCRtpReceiveParameters,\n  RTCRtpSimulcastParameters,\n} from \"./parameters\";\nimport { RTCRtpReceiver } from \"./rtpReceiver\";\nimport { RTCRtpSender } from \"./rtpSender\";\nimport { RTCRtpTransceiver } from \"./rtpTransceiver\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/router.ts\");\n\nexport type Extensions = { [uri: string]: number | string };\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  static rtpHeaderExtensionsParser(\n    extensions: Extension[],\n    extIdUriMap: { [id: number]: string }\n  ): Extensions {\n    return extensions\n      .map((extension) => {\n        const uri = extIdUriMap[extension.id];\n        switch (uri) {\n          case RTP_EXTENSION_URI.sdesMid:\n          case RTP_EXTENSION_URI.sdesRTPStreamID:\n          case RTP_EXTENSION_URI.repairedRtpStreamId:\n            return { uri, value: extension.payload.toString() };\n          case RTP_EXTENSION_URI.transportWideCC:\n            return { uri, value: extension.payload.readUInt16BE() };\n          case RTP_EXTENSION_URI.absSendTime:\n            return {\n              uri,\n              value: bufferReader(extension.payload, [3])[0],\n            };\n          default:\n            return { uri, value: 0 };\n        }\n      })\n      .reduce((acc: { [uri: string]: any }, cur) => {\n        if (cur) acc[cur.uri] = cur.value;\n        return acc;\n      }, {});\n  }\n\n  routeRtp = (packet: RtpPacket) => {\n    const extensions = RtpRouter.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              const remb = psfb.feedback as ReceiverEstimatedMaxBitrate;\n              recipients.push(this.ssrcTable[remb.ssrcFeedbacks[0]]);\n              break;\n            default:\n              recipients.push(this.ssrcTable[psfb.feedback.senderSsrc]);\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

@@ -15,7 +15,7 @@ export declare class RTCRtpReceiver {
     private get codecArray();
     private readonly ssrcByRtx;
     private readonly nack;
-    private readonly redHandler;
+    private readonly audioRedHandler;
     readonly type = "receiver";
     readonly uuid: string;
     readonly tracks: MediaStreamTrack[];

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

@@ -16,7 +16,6 @@ const utils_1 = require("../utils");
 const rtpExtension_1 = require("./extension/rtpExtension");
 const nack_1 = require("./receiver/nack");
 const receiverTwcc_1 = require("./receiver/receiverTwcc");
-const red_1 = require("./receiver/red");
 const statistics_1 = require("./receiver/statistics");
 const log = (0, debug_1.debug)("werift:packages/webrtc/src/media/rtpReceiver.ts");
 class RTCRtpReceiver {
@@ -27,7 +26,7 @@ class RTCRtpReceiver {
         this.codecs = {};
         this.ssrcByRtx = {};
         this.nack = new nack_1.NackHandler(this);
-        this.redHandler = new red_1.AudioRedHandler();
+        this.audioRedHandler = new src_2.RedHandler();
         this.type = "receiver";
         this.uuid = (0, uuid_1.v4)();
         this.tracks = [];
@@ -96,18 +95,23 @@ class RTCRtpReceiver {
     }
     addTrack(track) {
         const exist = this.tracks.find((t) => {
-            if (t.rid)
+            if (t.rid) {
                 return t.rid === track.rid;
-            if (t.ssrc)
+            }
+            if (t.ssrc) {
                 return t.ssrc === track.ssrc;
+            }
         });
-        if (exist)
+        if (exist) {
             return false;
+        }
         this.tracks.push(track);
-        if (track.ssrc)
+        if (track.ssrc) {
             this.trackBySSRC[track.ssrc] = track;
-        if (track.rid)
+        }
+        if (track.rid) {
             this.trackByRID[track.rid] = track;
+        }
         return true;
     }
     stop() {
@@ -199,8 +203,9 @@ class RTCRtpReceiver {
         this.onRtcp.execute(packet);
     }
     handleRTP(packet, extensions, track) {
-        if (this.stopped)
+        if (this.stopped) {
             return;
+        }
         const codec = this.codecs[packet.header.payloadType];
         if (!codec) {
             // log("unknown codec " + packet.header.payloadType);
@@ -242,7 +247,7 @@ class RTCRtpReceiver {
         if (track) {
             if (red) {
                 if (track.kind === "audio") {
-                    const payloads = this.redHandler.push(red, packet);
+                    const payloads = this.audioRedHandler.push(red, packet);
                     for (const packet of payloads) {
                         track.onReceiveRtp.execute(packet.clone());
                     }

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

@@ -1 +1 @@
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{ debug } from \"debug\";\nimport { jspack } from \"jspack\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport { v4 as uuid } from \"uuid\";\n\nimport { int } from \"../../../common/src\";\nimport {\n  PictureLossIndication,\n  Red,\n  RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpReceiverInfo,\n  RtcpRrPacket,\n  RtcpSrPacket,\n  RtpHeader,\n  RtpPacket,\n} from \"../../../rtp/src\";\nimport { codecParametersFromString, PeerConfig, usePLI, useTWCC } from \"..\";\nimport { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { compactNtp, timestampSeconds } from \"../utils\";\nimport { RTP_EXTENSION_URI } from \"./extension/rtpExtension\";\nimport { RTCRtpCodecParameters, RTCRtpReceiveParameters } from \"./parameters\";\nimport { NackHandler } from \"./receiver/nack\";\nimport { ReceiverTWCC } from \"./receiver/receiverTwcc\";\nimport { AudioRedHandler } from \"./receiver/red\";\nimport { StreamStatistics } from \"./receiver/statistics\";\nimport { Extensions } from \"./router\";\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 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 redHandler = new AudioRedHandler();\n\n  readonly type = \"receiver\";\n  readonly uuid = uuid();\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  remoteTrackId?: string;\n\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n  private remoteStreams: { [ssrc: number]: StreamStatistics } = {};\n\n  constructor(\n    readonly config: PeerConfig,\n    public kind: Kind,\n    public rtcpSsrc: number\n  ) {}\n\n  setDtlsTransport(dtls: RTCDtlsTransport) {\n    this.dtlsTransport = dtls;\n  }\n\n  // todo fix\n  get track() {\n    return this.tracks[0];\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) return t.rid === track.rid;\n      if (t.ssrc) return t.ssrc === track.ssrc;\n    });\n    if (exist) return false;\n    this.tracks.push(track);\n    if (track.ssrc) this.trackBySSRC[track.ssrc] = track;\n    if (track.rid) this.trackByRID[track.rid] = track;\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  /**todo impl */\n  getStats() {}\n\n  async sendRtcpPLI(mediaSsrc: number) {\n    if (!this.pliEnabled) {\n      log(\"pli not supported\", { mediaSsrc });\n      return;\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      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\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) return;\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 number;\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.redHandler.push(red, packet);\n          for (const packet of payloads) {\n            track.onReceiveRtp.execute(packet.clone());\n          }\n        } else {\n        }\n      } else {\n        track.onReceiveRtp.execute(packet.clone());\n      }\n    }\n\n    this.runRtcp();\n  }\n}\n\nexport function unwrapRtx(rtx: RtpPacket, payloadType: number, ssrc: number) {\n  const packet = new RtpPacket(\n    new RtpHeader({\n      payloadType,\n      marker: rtx.header.marker,\n      sequenceNumber: jspack.Unpack(\"!H\", rtx.payload.subarray(0, 2))[0],\n      timestamp: rtx.header.timestamp,\n      ssrc,\n    }),\n    rtx.payload.subarray(2)\n  );\n  return packet;\n}\n"]}
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{ debug } from \"debug\";\nimport { jspack } from \"jspack\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport { v4 as uuid } from \"uuid\";\n\nimport { int } from \"../../../common/src\";\nimport {\n  PictureLossIndication,\n  Red,\n  RedHandler,\n  RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpReceiverInfo,\n  RtcpRrPacket,\n  RtcpSrPacket,\n  RtpHeader,\n  RtpPacket,\n} from \"../../../rtp/src\";\nimport { codecParametersFromString, PeerConfig, usePLI, useTWCC } from \"..\";\nimport { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { compactNtp, timestampSeconds } from \"../utils\";\nimport { RTP_EXTENSION_URI } from \"./extension/rtpExtension\";\nimport { RTCRtpCodecParameters, RTCRtpReceiveParameters } from \"./parameters\";\nimport { NackHandler } from \"./receiver/nack\";\nimport { ReceiverTWCC } from \"./receiver/receiverTwcc\";\nimport { StreamStatistics } from \"./receiver/statistics\";\nimport { Extensions } from \"./router\";\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 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 = uuid();\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  remoteTrackId?: string;\n\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n  private remoteStreams: { [ssrc: number]: StreamStatistics } = {};\n\n  constructor(\n    readonly config: PeerConfig,\n    public kind: Kind,\n    public rtcpSsrc: number\n  ) {}\n\n  setDtlsTransport(dtls: RTCDtlsTransport) {\n    this.dtlsTransport = dtls;\n  }\n\n  // todo fix\n  get track() {\n    return this.tracks[0];\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  /**todo impl */\n  getStats() {}\n\n  async sendRtcpPLI(mediaSsrc: number) {\n    if (!this.pliEnabled) {\n      log(\"pli not supported\", { mediaSsrc });\n      return;\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      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\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 number;\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());\n          }\n        } else {\n        }\n      } else {\n        track.onReceiveRtp.execute(packet.clone());\n      }\n    }\n\n    this.runRtcp();\n  }\n}\n\nexport function unwrapRtx(rtx: RtpPacket, payloadType: number, ssrc: number) {\n  const packet = new RtpPacket(\n    new RtpHeader({\n      payloadType,\n      marker: rtx.header.marker,\n      sequenceNumber: jspack.Unpack(\"!H\", rtx.payload.subarray(0, 2))[0],\n      timestamp: rtx.header.timestamp,\n      ssrc,\n    }),\n    rtx.payload.subarray(2)\n  );\n  return packet;\n}\n"]}

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

@@ -10,16 +10,14 @@ export declare class RTCRtpTransceiver {
     receiver: RTCRtpReceiver;
     sender: RTCRtpSender;
     /**RFC 8829 4.2.4.  direction the transceiver was initialized with */
-    direction: Direction;
-    readonly uuid: string;
+    private _direction;
+    readonly id: string;
     readonly onTrack: Event<[MediaStreamTrack, RTCRtpTransceiver]>;
     mid?: string;
     mLineIndex?: number;
+    /**should not be reused because it has been used for sending before. */
     usedForSender: boolean;
     private _currentDirection?;
-    set currentDirection(direction: Direction | "stopped" | undefined);
-    /**RFC 8829 4.2.5. last negotiated direction */
-    get currentDirection(): Direction | "stopped" | undefined;
     offerDirection: Direction;
     _codecs: RTCRtpCodecParameters[];
     set codecs(codecs: RTCRtpCodecParameters[]);
@@ -30,8 +28,14 @@ export declare class RTCRtpTransceiver {
     stopped: boolean;
     constructor(kind: Kind, dtlsTransport: RTCDtlsTransport, receiver: RTCRtpReceiver, sender: RTCRtpSender, 
     /**RFC 8829 4.2.4.  direction the transceiver was initialized with */
-    direction: Direction);
+    _direction: Direction);
     get dtlsTransport(): RTCDtlsTransport;
+    /**RFC 8829 4.2.4. setDirectionに渡された最後の値を示します */
+    get direction(): "inactive" | "sendonly" | "recvonly" | "sendrecv";
+    setDirection(direction: Direction): void;
+    /**RFC 8829 4.2.5. last negotiated direction */
+    get currentDirection(): Direction | undefined;
+    setCurrentDirection(direction: Direction | undefined): void;
     setDtlsTransport(dtls: RTCDtlsTransport): void;
     get msid(): string;
     addTrack(track: MediaStreamTrack): void;

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

@@ -33,13 +33,14 @@ const const_1 = require("../const");
 class RTCRtpTransceiver {
     constructor(kind, dtlsTransport, receiver, sender, 
     /**RFC 8829 4.2.4.  direction the transceiver was initialized with */
-    direction) {
+    _direction) {
         this.kind = kind;
         this.receiver = receiver;
         this.sender = sender;
-        this.direction = direction;
-        this.uuid = uuid.v4();
+        this._direction = _direction;
+        this.id = uuid.v4();
         this.onTrack = new rx_mini_1.default();
+        /**should not be reused because it has been used for sending before. */
         this.usedForSender = false;
         this._codecs = [];
         this.headerExtensions = [];
@@ -48,16 +49,6 @@ class RTCRtpTransceiver {
         this.stopped = false;
         this.setDtlsTransport(dtlsTransport);
     }
-    set currentDirection(direction) {
-        this._currentDirection = direction;
-        if (const_1.SenderDirections.includes(this._currentDirection || "")) {
-            this.usedForSender = true;
-        }
-    }
-    /**RFC 8829 4.2.5. last negotiated direction */
-    get currentDirection() {
-        return this._currentDirection;
-    }
     set codecs(codecs) {
         this._codecs = codecs;
     }
@@ -67,6 +58,23 @@ class RTCRtpTransceiver {
     get dtlsTransport() {
         return this.receiver.dtlsTransport;
     }
+    /**RFC 8829 4.2.4. setDirectionに渡された最後の値を示します */
+    get direction() {
+        return this._direction;
+    }
+    setDirection(direction) {
+        this._direction = direction;
+        if (const_1.SenderDirections.includes(this._currentDirection ?? "")) {
+            this.usedForSender = true;
+        }
+    }
+    /**RFC 8829 4.2.5. last negotiated direction */
+    get currentDirection() {
+        return this._currentDirection;
+    }
+    setCurrentDirection(direction) {
+        this._currentDirection = direction;
+    }
     setDtlsTransport(dtls) {
         this.receiver.setDtlsTransport(dtls);
         this.sender.setDtlsTransport(dtls);
@@ -83,8 +91,9 @@ class RTCRtpTransceiver {
     // todo impl
     // https://www.w3.org/TR/webrtc/#methods-8
     stop() {
-        if (this.stopping)
+        if (this.stopping) {
             return;
+        }
         // todo Stop sending and receiving with transceiver.
         this.stopping = true;
     }