werift 0.20.1 → 0.21.0

package/lib/rtp/src/extra/processor/webmCallback.d.ts

@@ -1,13 +1,4 @@
-import type { SupportedCodec } from "../container/webm";
-import { WebmBase, type WebmInput, type WebmOption, type WebmOutput } from "./webm";
-export interface WebmTrack {
-    width?: number;
-    height?: number;
-    kind: "audio" | "video";
-    codec: SupportedCodec;
-    clockRate: number;
-    trackNumber: number;
-}
+import { WebmBase, type WebmInput, type WebmOption, type WebmOutput, type WebmTrack } from "./webm";
 export declare class WebmCallback extends WebmBase {
     private cb?;
     private queue;

package/lib/rtp/src/extra/processor/webmCallback.js.map

@@ -1 +1 @@
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+{"version":3,"file":"webmCallback.js","sourceRoot":"","sources":["../../../../../../rtp/src/extra/processor/webmCallback.ts"],"names":[],"mappings":";;;AAAA,0CAAqD;AAErD,6BAAqC;AACrC,iCASgB;AAEhB,MAAa,YAAa,SAAQ,eAAQ;IAGxC,YAAY,MAAmB,EAAE,UAAsB,EAAE;QACvD,KAAK,CACH,MAAM,EACN,KAAK,EAAE,MAAM,EAAE,EAAE;YACf,MAAM,EAAE,GAAG,IAAI,CAAC,EAAE,CAAC;YACnB,IAAI,EAAE,EAAE,CAAC;gBACP,MAAM,IAAI,CAAC,KAAK,CAAC,IAAI,CAAC,GAAG,EAAE,CAAC,EAAE,CAAC,MAAM,CAAC,CAAC,CAAC;YAC1C,CAAC;QACH,CAAC,EACD,OAAO,CACR,CAAC;QAZI;;;;;WAA0C;QAC1C;;;;mBAAQ,IAAI,gBAAY,EAAE;WAAC;QAcnC;;;;mBAAO,CAAC,EAAwC,EAAE,EAAE;gBAClD,IAAI,CAAC,EAAE,GAAG,EAAE,CAAC;gBACb,IAAI,CAAC,KAAK,EAAE,CAAC;YACf,CAAC;WAAC;QAEF;;;;mBAAa,CAAC,KAAgB,EAAE,EAAE;gBAChC,IAAI,CAAC,iBAAiB,CAAC,KAAK,CAAC,CAAC;YAChC,CAAC;WAAC;QACF;;;;mBAAa,CAAC,KAAgB,EAAE,EAAE;gBAChC,IAAI,CAAC,iBAAiB,CAAC,KAAK,CAAC,CAAC;YAChC,CAAC;WAAC;QAEF;;;;mBAAU,GAAG,EAAE;gBACb,IAAI,CAAC,EAAE,GAAG,SAAS,CAAC;gBACpB,IAAI,CAAC,KAAK,CAAC,MAAM,EAAE,CAAC;YACtB,CAAC;WAAC;IAjBF,CAAC;CAkBF;AAhCD,oCAgCC;AAEM,MAAM,gBAAgB,GAAG,CAAC,IAAY,EAAE,EAAE,CAAC,KAAK,EAAE,KAAiB,EAAE,EAAE;IAC5E,IAAI,KAAK,CAAC,UAAU,EAAE,CAAC;QACrB,MAAM,IAAA,qBAAU,EAAC,IAAI,EAAE,KAAK,CAAC,UAAU,CAAC,CAAC;IAC3C,CAAC;SAAM,IAAI,KAAK,CAAC,GAAG,EAAE,CAAC;QACrB,MAAM,EAAE,eAAe,EAAE,GAAG,KAAK,CAAC,GAAG,CAAC;QACtC,MAAM,OAAO,GAAG,MAAM,IAAA,eAAI,EAAC,IAAI,EAAE,IAAI,CAAC,CAAC;QAEvC,eAAe;QACf,MAAM,OAAO,CAAC,KAAK,CACjB,eAAe,EACf,CAAC,EACD,eAAe,CAAC,MAAM,EACtB,uBAAgB,CACjB,CAAC;QAEF,WAAW;QACX,MAAM,IAAI,GAAG,MAAM,IAAA,eAAI,EAAC,IAAI,CAAC,CAAC;QAC9B,MAAM,MAAM,GAAG,IAAA,yBAAkB,EAAC,IAAI,CAAC,IAAI,CAAC,CAAC;QAC7C,MAAM,OAAO,CAAC,KAAK,CAAC,MAAM,EAAE,CAAC,EAAE,MAAM,CAAC,MAAM,EAAE,0BAAmB,CAAC,CAAC;QAEnE,MAAM,OAAO,CAAC,KAAK,EAAE,CAAC;IACxB,CAAC;AACH,CAAC,CAAC;AAtBW,QAAA,gBAAgB,oBAsB3B","sourcesContent":["import { appendFile, open, stat } from \"fs/promises\";\n\nimport { PromiseQueue } from \"../..\";\nimport {\n  DurationPosition,\n  SegmentSizePosition,\n  WebmBase,\n  type WebmInput,\n  type WebmOption,\n  type WebmOutput,\n  type WebmTrack,\n  replaceSegmentSize,\n} from \"./webm\";\n\nexport class WebmCallback extends WebmBase {\n  private cb?: (input: WebmOutput) => Promise<void>;\n  private queue = new PromiseQueue();\n  constructor(tracks: WebmTrack[], options: WebmOption = {}) {\n    super(\n      tracks,\n      async (output) => {\n        const cb = this.cb;\n        if (cb) {\n          await this.queue.push(() => cb(output));\n        }\n      },\n      options,\n    );\n  }\n\n  pipe = (cb: (input: WebmOutput) => Promise<void>) => {\n    this.cb = cb;\n    this.start();\n  };\n\n  inputAudio = (input: WebmInput) => {\n    this.processAudioInput(input);\n  };\n  inputVideo = (input: WebmInput) => {\n    this.processVideoInput(input);\n  };\n\n  destroy = () => {\n    this.cb = undefined;\n    this.queue.cancel();\n  };\n}\n\nexport const saveToFileSystem = (path: string) => async (value: WebmOutput) => {\n  if (value.saveToFile) {\n    await appendFile(path, value.saveToFile);\n  } else if (value.eol) {\n    const { durationElement } = value.eol;\n    const handler = await open(path, \"r+\");\n\n    // set duration\n    await handler.write(\n      durationElement,\n      0,\n      durationElement.length,\n      DurationPosition,\n    );\n\n    // set size\n    const meta = await stat(path);\n    const resize = replaceSegmentSize(meta.size);\n    await handler.write(resize, 0, resize.length, SegmentSizePosition);\n\n    await handler.close();\n  }\n};\n"]}

package/lib/rtp/src/rtp/headerExtension.d.ts

@@ -7,17 +7,58 @@ export declare const RTP_EXTENSION_URI: {
     readonly absSendTime: "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
     readonly dependencyDescriptor: "https://aomediacodec.github.io/av1-rtp-spec/#dependency-descriptor-rtp-header-extension";
     readonly audioLevelIndication: "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
+    readonly videoOrientation: "urn:3gpp:video-orientation";
 };
 export type TransportWideCCPayload = number;
 export type AudioLevelIndicationPayload = {
     v: boolean;
     level: number;
 };
+export interface videoOrientationPayload {
+    /**
+     Camera: indicates the direction of the camera used for this video stream. It can be used by the MTSI client in
+    receiver to e.g. display the received video differently depending on the source camera.
+    0: Front-facing camera, facing the user. If camera direction is unknown by the sending MTSI client in the terminal
+    then this is the default value used.
+    1: Back-facing camera, facing away from the user.
+   */
+    c: number;
+    /**
+     F = Flip: indicates a horizontal (left-right flip) mirror operation on the video as sent on the link.
+    0: No flip operation. If the sending MTSI client in terminal does not know if a horizontal mirror operation is
+    necessary, then this is the default value used.
+    1: Horizontal flip operation
+     */
+    f: number;
+    /**
+     R1, R0 = Rotation: indicates the rotation of the video as transmitted on the link. The receiver should rotate the video to
+    compensate that rotation. E.g. a 90° Counter Clockwise rotation should be compensated by the receiver with a 90°
+    Clockwise rotation prior to displaying.
+  
+      +----+----+-----------------------------------------------+------------------------------+
+    | R1 | R0 | Rotation of the video as sent on the link     | Rotation on the receiver      |
+    |    |    |                                               | before display                |
+    +----+----+-----------------------------------------------+------------------------------+
+    |  0 |  0 | 0° rotation                                   | None                         |
+    +----+----+-----------------------------------------------+------------------------------+
+    |  0 |  1 | 90° Counter Clockwise (CCW) rotation or 270°  | 90° Clockwise (CW) rotation  |
+    |    |    | Clockwise (CW) rotation                       |                              |
+    +----+----+-----------------------------------------------+------------------------------+
+    |  1 |  0 | 180° CCW rotation or 180° CW rotation         | 180° CW rotation             |
+    +----+----+-----------------------------------------------+------------------------------+
+    |  1 |  1 | 270° CCW rotation or 90° CW rotation          | 90° CCW rotation             |
+    +----+----+-----------------------------------------------+------------------------------+
+  
+     */
+    r1: number;
+    r0: number;
+}
+export interface Extensions {
+    [uri: string]: any;
+}
 export declare function rtpHeaderExtensionsParser(extensions: Extension[], extIdUriMap: {
     [id: number]: string;
-}): {
-    [uri: string]: any;
-};
+}): Extensions;
 export declare function serializeSdesMid(id: string): Buffer;
 export declare function serializeSdesRTPStreamID(id: string): Buffer;
 export declare function serializeRepairedRtpStreamId(id: string): Buffer;
@@ -28,3 +69,4 @@ export declare function deserializeString(buf: Buffer): string;
 export declare function deserializeUint16BE(buf: Buffer): number;
 export declare function deserializeAbsSendTime(buf: Buffer): any;
 export declare function deserializeAudioLevelIndication(buf: Buffer): AudioLevelIndicationPayload;
+export declare function deserializeVideoOrientation(payload: Buffer): videoOrientationPayload;

package/lib/rtp/src/rtp/headerExtension.js

@@ -12,6 +12,7 @@ exports.deserializeString = deserializeString;
 exports.deserializeUint16BE = deserializeUint16BE;
 exports.deserializeAbsSendTime = deserializeAbsSendTime;
 exports.deserializeAudioLevelIndication = deserializeAudioLevelIndication;
+exports.deserializeVideoOrientation = deserializeVideoOrientation;
 const src_1 = require("../../../common/src");
 exports.RTP_EXTENSION_URI = {
     sdesMid: "urn:ietf:params:rtp-hdrext:sdes:mid",
@@ -21,6 +22,7 @@ exports.RTP_EXTENSION_URI = {
     absSendTime: "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",
     dependencyDescriptor: "https://aomediacodec.github.io/av1-rtp-spec/#dependency-descriptor-rtp-header-extension",
     audioLevelIndication: "urn:ietf:params:rtp-hdrext:ssrc-audio-level",
+    videoOrientation: "urn:3gpp:video-orientation",
 };
 function rtpHeaderExtensionsParser(extensions, extIdUriMap) {
     return extensions
@@ -47,6 +49,8 @@ function rtpHeaderExtensionsParser(extensions, extIdUriMap) {
                     value: deserializeAudioLevelIndication(extension.payload),
                 };
             }
+            case exports.RTP_EXTENSION_URI.videoOrientation:
+                return { uri, value: deserializeVideoOrientation(extension.payload) };
             default:
                 return { uri, value: extension.payload };
         }
@@ -98,4 +102,15 @@ function deserializeAudioLevelIndication(buf) {
     };
     return value;
 }
+function deserializeVideoOrientation(payload) {
+    const stream = new src_1.BitStream(payload);
+    stream.readBits(4);
+    const value = {
+        c: stream.readBits(1),
+        f: stream.readBits(1),
+        r1: stream.readBits(1),
+        r0: stream.readBits(1),
+    };
+    return value;
+}
 //# sourceMappingURL=headerExtension.js.map

package/lib/rtp/src/rtp/headerExtension.js.map

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level: number };\n\nexport function rtpHeaderExtensionsParser(\n  extensions: Extension[],\n  extIdUriMap: { [id: number]: string },\n): { [uri: string]: any } {\n  return extensions\n    .map((extension) => {\n      const uri = extIdUriMap[extension.id];\n      if (!uri) {\n        return { uri: \"unknown\", value: extension.payload };\n      }\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: deserializeString(extension.payload) };\n        case RTP_EXTENSION_URI.transportWideCC:\n          return { uri, value: deserializeUint16BE(extension.payload) };\n        case RTP_EXTENSION_URI.absSendTime:\n          return {\n            uri,\n            value: deserializeAbsSendTime(extension.payload),\n          };\n        case RTP_EXTENSION_URI.audioLevelIndication: {\n          return {\n            uri,\n            value: deserializeAudioLevelIndication(extension.payload),\n          };\n        }\n        default:\n          return { uri, value: extension.payload };\n      }\n    })\n    .reduce((acc: { [uri: string]: any }, cur) => {\n      if (cur) acc[cur.uri] = cur.value;\n      return acc;\n    }, {});\n}\n\nexport function serializeSdesMid(id: string) {\n  return Buffer.from(id);\n}\n\nexport function serializeSdesRTPStreamID(id: string) {\n  return Buffer.from(id);\n}\n\nexport function serializeRepairedRtpStreamId(id: string) {\n  return Buffer.from(id);\n}\n\nexport function serializeTransportWideCC(transportSequenceNumber: number) {\n  return bufferWriter([2], [transportSequenceNumber]);\n}\n\nexport function serializeAbsSendTime(ntpTime: bigint) {\n  const buf = Buffer.alloc(3);\n  const time = (ntpTime >> 14n) & 0x00ffffffn;\n  buf.writeUIntBE(Number(time), 0, 3);\n  return buf;\n}\n\nexport function serializeAudioLevelIndication(level: number) {\n  const stream = new BitStream(Buffer.alloc(1));\n  stream.writeBits(1, 1);\n  stream.writeBits(7, level);\n\n  return stream.uint8Array;\n}\n\nexport function deserializeString(buf: Buffer) {\n  return buf.toString();\n}\n\nexport function deserializeUint16BE(buf: Buffer) {\n  return buf.readUInt16BE();\n}\n\nexport function deserializeAbsSendTime(buf: Buffer) {\n  return bufferReader(buf, [3])[0];\n}\n\nexport function deserializeAudioLevelIndication(buf: Buffer) {\n  const stream = new BitStream(buf);\n  const value: AudioLevelIndicationPayload = {\n    v: stream.readBits(1) === 1,\n    level: stream.readBits(7),\n  };\n  return value;\n}\n"]}
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It can be used by the MTSI client in\n  receiver to e.g. display the received video differently depending on the source camera.\n  0: Front-facing camera, facing the user. If camera direction is unknown by the sending MTSI client in the terminal\n  then this is the default value used.\n  1: Back-facing camera, facing away from the user. \n */\n  c: number;\n  /**\n   F = Flip: indicates a horizontal (left-right flip) mirror operation on the video as sent on the link.\n  0: No flip operation. If the sending MTSI client in terminal does not know if a horizontal mirror operation is\n  necessary, then this is the default value used.\n  1: Horizontal flip operation \n   */\n  f: number;\n  /**\n   R1, R0 = Rotation: indicates the rotation of the video as transmitted on the link. The receiver should rotate the video to\n  compensate that rotation. E.g. a 90° Counter Clockwise rotation should be compensated by the receiver with a 90°\n  Clockwise rotation prior to displaying. \n\n    +----+----+-----------------------------------------------+------------------------------+\n  | R1 | R0 | Rotation of the video as sent on the link     | Rotation on the receiver      |\n  |    |    |                                               | before display                |\n  +----+----+-----------------------------------------------+------------------------------+\n  |  0 |  0 | 0° rotation                                   | None                         |\n  +----+----+-----------------------------------------------+------------------------------+\n  |  0 |  1 | 90° Counter Clockwise (CCW) rotation or 270°  | 90° Clockwise (CW) rotation  |\n  |    |    | Clockwise (CW) rotation                       |                              |\n  +----+----+-----------------------------------------------+------------------------------+\n  |  1 |  0 | 180° CCW rotation or 180° CW rotation         | 180° CW rotation             |\n  +----+----+-----------------------------------------------+------------------------------+\n  |  1 |  1 | 270° CCW rotation or 90° CW rotation          | 90° CCW rotation             |\n  +----+----+-----------------------------------------------+------------------------------+\n\n   */\n  r1: number;\n  r0: number;\n}\n\nexport interface Extensions {\n  [uri: string]: any;\n}\n\nexport function rtpHeaderExtensionsParser(\n  extensions: Extension[],\n  extIdUriMap: { [id: number]: string },\n): Extensions {\n  return extensions\n    .map((extension) => {\n      const uri = extIdUriMap[extension.id];\n      if (!uri) {\n        return { uri: \"unknown\", value: extension.payload };\n      }\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: deserializeString(extension.payload) };\n        case RTP_EXTENSION_URI.transportWideCC:\n          return { uri, value: deserializeUint16BE(extension.payload) };\n        case RTP_EXTENSION_URI.absSendTime:\n          return {\n            uri,\n            value: deserializeAbsSendTime(extension.payload),\n          };\n        case RTP_EXTENSION_URI.audioLevelIndication: {\n          return {\n            uri,\n            value: deserializeAudioLevelIndication(extension.payload),\n          };\n        }\n        case RTP_EXTENSION_URI.videoOrientation:\n          return { uri, value: deserializeVideoOrientation(extension.payload) };\n        default:\n          return { uri, value: extension.payload };\n      }\n    })\n    .reduce((acc: { [uri: string]: any }, cur) => {\n      if (cur) acc[cur.uri] = cur.value;\n      return acc;\n    }, {});\n}\n\nexport function serializeSdesMid(id: string) {\n  return Buffer.from(id);\n}\n\nexport function serializeSdesRTPStreamID(id: string) {\n  return Buffer.from(id);\n}\n\nexport function serializeRepairedRtpStreamId(id: string) {\n  return Buffer.from(id);\n}\n\nexport function serializeTransportWideCC(transportSequenceNumber: number) {\n  return bufferWriter([2], [transportSequenceNumber]);\n}\n\nexport function serializeAbsSendTime(ntpTime: bigint) {\n  const buf = Buffer.alloc(3);\n  const time = (ntpTime >> 14n) & 0x00ffffffn;\n  buf.writeUIntBE(Number(time), 0, 3);\n  return buf;\n}\n\nexport function serializeAudioLevelIndication(level: number) {\n  const stream = new BitStream(Buffer.alloc(1));\n  stream.writeBits(1, 1);\n  stream.writeBits(7, level);\n\n  return stream.uint8Array;\n}\n\nexport function deserializeString(buf: Buffer) {\n  return buf.toString();\n}\n\nexport function deserializeUint16BE(buf: Buffer) {\n  return buf.readUInt16BE();\n}\n\nexport function deserializeAbsSendTime(buf: Buffer) {\n  return bufferReader(buf, [3])[0];\n}\n\nexport function deserializeAudioLevelIndication(buf: Buffer) {\n  const stream = new BitStream(buf);\n  const value: AudioLevelIndicationPayload = {\n    v: stream.readBits(1) === 1,\n    level: stream.readBits(7),\n  };\n  return value;\n}\n\nexport function deserializeVideoOrientation(payload: Buffer) {\n  const stream = new BitStream(payload);\n  stream.readBits(4);\n  const value: videoOrientationPayload = {\n    c: stream.readBits(1),\n    f: stream.readBits(1),\n    r1: stream.readBits(1),\n    r0: stream.readBits(1),\n  };\n  return value;\n}\n"]}

package/lib/sctp/src/index.d.ts

@@ -1,3 +1,3 @@
 export { SCTP_STATE, WEBRTC_PPID } from "./const";
 export { SCTP } from "./sctp";
-export { Transport } from "./transport";
+export type { Transport } from "./transport";

package/lib/sctp/src/index.js.map

@@ -1 +1 @@
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+{"version":3,"file":"index.js","sourceRoot":"","sources":["../../../../sctp/src/index.ts"],"names":[],"mappings":";;;AAAA,iCAAkD;AAAzC,mGAAA,UAAU,OAAA;AAAE,oGAAA,WAAW,OAAA;AAChC,+BAA8B;AAArB,4FAAA,IAAI,OAAA","sourcesContent":["export { SCTP_STATE, WEBRTC_PPID } from \"./const\";\nexport { SCTP } from \"./sctp\";\nexport type { Transport } from \"./transport\";\n"]}

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

@@ -1,4 +1,4 @@
-import { HashAlgorithms, SignatureAlgorithm } from "../../dtls/src/cipher/const";
+import { type HashAlgorithms, SignatureAlgorithm } from "../../dtls/src/cipher/const";
 import type { DtlsRole } from "./transport/dtls";
 export declare const DATA_CHANNEL_ACK = 2;
 export declare const DATA_CHANNEL_OPEN = 3;
@@ -33,4 +33,4 @@ export declare const SRTP_PROFILE: {
 };
 export declare const SenderDirections: string[];
 export declare const ReceiverDirection: string[];
-export { HashAlgorithms, SignatureAlgorithm };
+export { type HashAlgorithms, SignatureAlgorithm };

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,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 type { 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"]}
+{"version":3,"file":"const.js","sourceRoot":"","sources":["../../../src/const.ts"],"names":[],"mappings":";;;AAAA,uDAGqC;AA8DP,mGA/D5B,0BAAkB,OA+D4B;AA7DhD,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  type HashAlgorithms,\n  SignatureAlgorithm,\n} from \"../../dtls/src/cipher/const\";\nimport {\n  ProtectionProfileAeadAes128Gcm,\n  ProtectionProfileAes128CmHmacSha1_80,\n} from \"../../dtls/src/context/srtp\";\nimport type { 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 { type HashAlgorithms, SignatureAlgorithm };\n"]}

package/lib/webrtc/src/media/extension/rtpExtension.d.ts

@@ -6,3 +6,4 @@ export declare function useTransportWideCC(): RTCRtpHeaderExtensionParameters;
 export declare function useAbsSendTime(): RTCRtpHeaderExtensionParameters;
 export declare function useDependencyDescriptor(): RTCRtpHeaderExtensionParameters;
 export declare function useAudioLevelIndication(): RTCRtpHeaderExtensionParameters;
+export declare function useVideoOrientation(): RTCRtpHeaderExtensionParameters;

package/lib/webrtc/src/media/extension/rtpExtension.js

@@ -7,6 +7,7 @@ exports.useTransportWideCC = useTransportWideCC;
 exports.useAbsSendTime = useAbsSendTime;
 exports.useDependencyDescriptor = useDependencyDescriptor;
 exports.useAudioLevelIndication = useAudioLevelIndication;
+exports.useVideoOrientation = useVideoOrientation;
 const src_1 = require("../../../../rtp/src");
 const parameters_1 = require("../parameters");
 function useSdesMid() {
@@ -44,4 +45,9 @@ function useAudioLevelIndication() {
         uri: src_1.RTP_EXTENSION_URI.audioLevelIndication,
     });
 }
+function useVideoOrientation() {
+    return new parameters_1.RTCRtpHeaderExtensionParameters({
+        uri: src_1.RTP_EXTENSION_URI.videoOrientation,
+    });
+}
 //# sourceMappingURL=rtpExtension.js.map

package/lib/webrtc/src/media/extension/rtpExtension.js.map

@@ -1 +1 @@
-{"version":3,"file":"rtpExtension.js","sourceRoot":"","sources":["../../../../../src/media/extension/rtpExtension.ts"],"names":[],"mappings":";;AAGA,gCAIC;AAED,gDAIC;AAED,wDAIC;AAED,gDAIC;AAED,wCAIC;AAED,0DAIC;AAED,0DAIC;AA3CD,6CAAwD;AACxD,8CAAgE;AAEhE,SAAgB,UAAU;IACxB,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,OAAO;KAC/B,CAAC,CAAC;AACL,CAAC;AAED,SAAgB,kBAAkB;IAChC,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,eAAe;KACvC,CAAC,CAAC;AACL,CAAC;AAED,SAAgB,sBAAsB;IACpC,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,mBAAmB;KAC3C,CAAC,CAAC;AACL,CAAC;AAED,SAAgB,kBAAkB;IAChC,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,eAAe;KACvC,CAAC,CAAC;AACL,CAAC;AAED,SAAgB,cAAc;IAC5B,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,WAAW;KACnC,CAAC,CAAC;AACL,CAAC;AAED,SAAgB,uBAAuB;IACrC,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,oBAAoB;KAC5C,CAAC,CAAC;AACL,CAAC;AAED,SAAgB,uBAAuB;IACrC,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,oBAAoB;KAC5C,CAAC,CAAC;AACL,CAAC","sourcesContent":["import { RTP_EXTENSION_URI } from \"../../../../rtp/src\";\nimport { RTCRtpHeaderExtensionParameters } from \"../parameters\";\n\nexport function useSdesMid() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.sdesMid,\n  });\n}\n\nexport function useSdesRTPStreamId() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.sdesRTPStreamID,\n  });\n}\n\nexport function useRepairedRtpStreamId() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.repairedRtpStreamId,\n  });\n}\n\nexport function useTransportWideCC() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.transportWideCC,\n  });\n}\n\nexport function useAbsSendTime() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.absSendTime,\n  });\n}\n\nexport function useDependencyDescriptor() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.dependencyDescriptor,\n  });\n}\n\nexport function useAudioLevelIndication() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.audioLevelIndication,\n  });\n}\n"]}
+{"version":3,"file":"rtpExtension.js","sourceRoot":"","sources":["../../../../../src/media/extension/rtpExtension.ts"],"names":[],"mappings":";;AAGA,gCAIC;AAED,gDAIC;AAED,wDAIC;AAED,gDAIC;AAED,wCAIC;AAED,0DAIC;AAED,0DAIC;AAED,kDAIC;AAjDD,6CAAwD;AACxD,8CAAgE;AAEhE,SAAgB,UAAU;IACxB,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,OAAO;KAC/B,CAAC,CAAC;AACL,CAAC;AAED,SAAgB,kBAAkB;IAChC,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,eAAe;KACvC,CAAC,CAAC;AACL,CAAC;AAED,SAAgB,sBAAsB;IACpC,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,mBAAmB;KAC3C,CAAC,CAAC;AACL,CAAC;AAED,SAAgB,kBAAkB;IAChC,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,eAAe;KACvC,CAAC,CAAC;AACL,CAAC;AAED,SAAgB,cAAc;IAC5B,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,WAAW;KACnC,CAAC,CAAC;AACL,CAAC;AAED,SAAgB,uBAAuB;IACrC,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,oBAAoB;KAC5C,CAAC,CAAC;AACL,CAAC;AAED,SAAgB,uBAAuB;IACrC,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,oBAAoB;KAC5C,CAAC,CAAC;AACL,CAAC;AAED,SAAgB,mBAAmB;IACjC,OAAO,IAAI,4CAA+B,CAAC;QACzC,GAAG,EAAE,uBAAiB,CAAC,gBAAgB;KACxC,CAAC,CAAC;AACL,CAAC","sourcesContent":["import { RTP_EXTENSION_URI } from \"../../../../rtp/src\";\nimport { RTCRtpHeaderExtensionParameters } from \"../parameters\";\n\nexport function useSdesMid() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.sdesMid,\n  });\n}\n\nexport function useSdesRTPStreamId() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.sdesRTPStreamID,\n  });\n}\n\nexport function useRepairedRtpStreamId() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.repairedRtpStreamId,\n  });\n}\n\nexport function useTransportWideCC() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.transportWideCC,\n  });\n}\n\nexport function useAbsSendTime() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.absSendTime,\n  });\n}\n\nexport function useDependencyDescriptor() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.dependencyDescriptor,\n  });\n}\n\nexport function useAudioLevelIndication() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.audioLevelIndication,\n  });\n}\n\nexport function useVideoOrientation() {\n  return new RTCRtpHeaderExtensionParameters({\n    uri: RTP_EXTENSION_URI.videoOrientation,\n  });\n}\n"]}

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

@@ -3,9 +3,6 @@ import type { RTCRtpReceiveParameters, RTCRtpSimulcastParameters } from "./param
 import { RTCRtpReceiver } from "./rtpReceiver";
 import type { RTCRtpSender } from "./rtpSender";
 import type { RTCRtpTransceiver } from "./rtpTransceiver";
-export type Extensions = {
-    [uri: string]: number | string;
-};
 export declare class RtpRouter {
     ssrcTable: {
         [ssrc: number]: RTCRtpReceiver | RTCRtpSender;

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

@@ -1 +1 @@
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debug from \"debug\";\n\nimport {\n  RTP_EXTENSION_URI,\n  ReceiverEstimatedMaxBitrate,\n  type RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpRrPacket,\n  RtcpSourceDescriptionPacket,\n  RtcpSrPacket,\n  RtcpTransportLayerFeedback,\n  type RtpPacket,\n  rtpHeaderExtensionsParser,\n} from \"../../../rtp/src\";\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 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  routeRtp = (packet: RtpPacket) => {\n    const 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(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 {\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  rtpHeaderExtensionsParser,\n} from \"../../../rtp/src\";\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(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

@@ -1,11 +1,10 @@
 import { Event } from "../imports/common";
 import { type PeerConfig } from "..";
-import { type RtcpPacket, type RtpPacket } from "../../../rtp/src";
+import { type Extensions, type RtcpPacket, type RtpPacket } from "../../../rtp/src";
 import type { RTCDtlsTransport } from "../transport/dtls";
 import type { Kind } from "../types/domain";
 import type { RTCRtpReceiveParameters } from "./parameters";
 import { ReceiverTWCC } from "./receiver/receiverTwcc";
-import type { Extensions } from "./router";
 import type { MediaStreamTrack } from "./track";
 export declare class RTCRtpReceiver {
     readonly config: PeerConfig;

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

@@ -399,14 +399,14 @@ class RTCRtpReceiver {
                 if (track.kind === "audio") {
                     const payloads = this.audioRedHandler.push(red, packet);
                     for (const packet of payloads) {
-                        track.onReceiveRtp.execute(packet.clone());
+                        track.onReceiveRtp.execute(packet.clone(), extensions);
                     }
                 }
                 else {
                 }
             }
             else {
-                track.onReceiveRtp.execute(packet.clone());
+                track.onReceiveRtp.execute(packet.clone(), extensions);
             }
         }
         this.runRtcp();

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

@@ -1 +1 @@
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{ debug } from \"debug\";\nimport { setTimeout } from \"timers/promises\";\nimport { v4 as uuid } from \"uuid\";\nimport { Event } from \"../imports/common\";\n\nimport {\n  type PeerConfig,\n  codecParametersFromString,\n  usePLI,\n  useTWCC,\n} from \"..\";\nimport { int } from \"../../../common/src\";\nimport {\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  unwrapRtx,\n} from \"../../../rtp/src\";\nimport type { RTCDtlsTransport } from \"../transport/dtls\";\nimport type { Kind } from \"../types/domain\";\nimport { compactNtp, 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\";\nimport type { Extensions } from \"./router\";\nimport type { 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\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      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 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());\n          }\n        } else {\n        }\n      } else {\n        track.onReceiveRtp.execute(packet.clone());\n      }\n    }\n\n    this.runRtcp();\n  }\n}\n"]}
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{ debug } from \"debug\";\nimport { setTimeout } from \"timers/promises\";\nimport { v4 as uuid } from \"uuid\";\nimport { Event } from \"../imports/common\";\n\nimport {\n  type PeerConfig,\n  codecParametersFromString,\n  usePLI,\n  useTWCC,\n} from \"..\";\nimport { int } from \"../../../common/src\";\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  unwrapRtx,\n} from \"../../../rtp/src\";\nimport type { RTCDtlsTransport } from \"../transport/dtls\";\nimport type { Kind } from \"../types/domain\";\nimport { compactNtp, 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 type { 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\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      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 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"]}

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

@@ -1,5 +1,5 @@
 import { Event } from "../imports/common";
-import { type RtcpPacket, type RtpHeader, RtpPacket } from "../../../rtp/src";
+import { type Extensions, type RtcpPacket, type RtpHeader, RtpPacket } from "../../../rtp/src";
 import { EventTarget } from "../helper";
 import type { Kind } from "../types/domain";
 import type { RTCRtpCodecParameters } from "./parameters";
@@ -18,7 +18,7 @@ export declare class MediaStreamTrack extends EventTarget {
     codec?: RTCRtpCodecParameters;
     /**todo impl */
     enabled: boolean;
-    readonly onReceiveRtp: Event<[RtpPacket]>;
+    readonly onReceiveRtp: Event<[RtpPacket, (Extensions | undefined)?]>;
     readonly onReceiveRtcp: Event<[RtcpPacket]>;
     readonly onSourceChanged: Event<[Pick<RtpHeader, "sequenceNumber" | "timestamp">]>;
     stopped: boolean;

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

@@ -1 +1 @@
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+{"version":3,"file":"track.js","sourceRoot":"","sources":["../../../../src/media/track.ts"],"names":[],"mappings":";;;AAAA,+BAA0B;AAC1B,8CAA0C;AAE1C,0CAK0B;AAC1B,sCAAwC;AAIxC,MAAa,gBAAiB,SAAQ,oBAAW;IAyB/C,YACE,KAAiE;QAEjE,KAAK,EAAE,CAAC;QA3BD;;;;mBAAO,IAAA,SAAE,GAAE;WAAC;QACrB,mBAAmB;QACnB;;;;;WAAkB;QAClB;;;;mBAAS,KAAK;WAAC;QACf;;;;;WAAc;QACd;;;;;WAAY;QACZ;;;;;WAAY;QACZ,eAAe;QACf;;;;;WAAc;QACd;;;;;WAAa;QACb;;;;;WAAmB;QACnB;;;;;WAA8B;QAC9B,eAAe;QACf;;;;mBAAU,IAAI;WAAC;QAEN;;;;mBAAe,IAAI,cAAK,EAA4B;WAAC;QACrD;;;;mBAAgB,IAAI,cAAK,EAAgB;WAAC;QAC1C;;;;mBAAkB,IAAI,cAAK,EAEjC;WAAC;QAEJ;;;;mBAAU,KAAK;WAAC;QAChB;;;;mBAAQ,IAAI;WAAC;QAgBb;;;;mBAAO,GAAG,EAAE;gBACV,IAAI,CAAC,OAAO,GAAG,IAAI,CAAC;gBACpB,IAAI,CAAC,KAAK,GAAG,IAAI,CAAC;gBAClB,IAAI,CAAC,YAAY,CAAC,QAAQ,EAAE,CAAC;gBAC7B,IAAI,CAAC,IAAI,CAAC,OAAO,CAAC,CAAC;YACrB,CAAC;WAAC;QAEF;;;;mBAAW,CAAC,GAAuB,EAAE,EAAE;gBACrC,IAAI,IAAI,CAAC,MAAM,EAAE,CAAC;oBAChB,MAAM,IAAI,KAAK,CAAC,qBAAqB,CAAC,CAAC;gBACzC,CAAC;gBACD,IAAI,IAAI,CAAC,OAAO,EAAE,CAAC;oBACjB,OAAO;gBACT,CAAC;gBAED,MAAM,MAAM,GAAG,MAAM,CAAC,QAAQ,CAAC,GAAG,CAAC,CAAC,CAAC,CAAC,eAAS,CAAC,WAAW,CAAC,GAAG,CAAC,CAAC,CAAC,CAAC,GAAG,CAAC;gBACvE,MAAM,CAAC,MAAM,CAAC,WAAW;oBACvB,IAAI,CAAC,KAAK,EAAE,WAAW,IAAI,MAAM,CAAC,MAAM,CAAC,WAAW,CAAC;gBACvD,IAAI,CAAC,YAAY,CAAC,OAAO,CAAC,MAAM,CAAC,CAAC;YACpC,CAAC;WAAC;QA7BA,MAAM,CAAC,MAAM,CAAC,IAAI,EAAE,KAAK,CAAC,CAAC;QAE3B,IAAI,CAAC,YAAY,CAAC,SAAS,CAAC,CAAC,GAAG,EAAE,EAAE;YAClC,IAAI,CAAC,KAAK,GAAG,KAAK,CAAC;YACnB,IAAI,CAAC,MAAM,GAAG,GAAG,CAAC,MAAM,CAAC;QAC3B,CAAC,CAAC,CAAC;QAEH,IAAI,CAAC,KAAK,GAAG,GAAG,IAAI,CAAC,MAAM,CAAC,CAAC,CAAC,QAAQ,CAAC,CAAC,CAAC,OAAO,IAAI,IAAI,CAAC,IAAI,EAAE,CAAC;IAClE,CAAC;CAsBF;AA3DD,4CA2DC;AAED,MAAa,WAAW;IAItB,YAAY,KAAgD;QAH5D;;;;;WAAY;QACZ;;;;mBAA6B,EAAE;WAAC;QAG9B,IAAI,KAAK,CAAC,OAAO,CAAC,KAAK,CAAC,EAAE,CAAC;YACzB,IAAI,CAAC,MAAM,GAAG,KAAK,CAAC;QACtB,CAAC;aAAM,CAAC;YACN,MAAM,CAAC,MAAM,CAAC,IAAI,EAAE,KAAK,CAAC,CAAC;QAC7B,CAAC;QACD,IAAI,CAAC,EAAE,KAAP,IAAI,CAAC,EAAE,GAAK,IAAA,SAAE,GAAE,EAAC;IACnB,CAAC;IAED,QAAQ,CAAC,KAAuB;QAC9B,KAAK,CAAC,QAAQ,GAAG,IAAI,CAAC,EAAE,CAAC;QACzB,IAAI,CAAC,MAAM,CAAC,IAAI,CAAC,KAAK,CAAC,CAAC;IAC1B,CAAC;IAED,SAAS;QACP,OAAO,IAAI,CAAC,MAAM,CAAC;IACrB,CAAC;IAED,cAAc;QACZ,OAAO,IAAI,CAAC,MAAM,CAAC,MAAM,CAAC,CAAC,KAAK,EAAE,EAAE,CAAC,KAAK,CAAC,IAAI,KAAK,OAAO,CAAC,CAAC;IAC/D,CAAC;IAED,cAAc;QACZ,OAAO,IAAI,CAAC,MAAM,CAAC,MAAM,CAAC,CAAC,KAAK,EAAE,EAAE,CAAC,KAAK,CAAC,IAAI,KAAK,OAAO,CAAC,CAAC;IAC/D,CAAC;CACF;AA7BD,kCA6BC","sourcesContent":["import { v4 } from \"uuid\";\nimport { Event } from \"../imports/common\";\n\nimport {\n  type Extensions,\n  type RtcpPacket,\n  type RtpHeader,\n  RtpPacket,\n} from \"../../../rtp/src\";\nimport { EventTarget } from \"../helper\";\nimport type { Kind } from \"../types/domain\";\nimport type { RTCRtpCodecParameters } from \"./parameters\";\n\nexport class MediaStreamTrack extends EventTarget {\n  readonly uuid = v4();\n  /**MediaStream ID*/\n  streamId?: string;\n  remote = false;\n  label: string;\n  kind!: Kind;\n  id?: string;\n  /**mediaSsrc */\n  ssrc?: number;\n  rid?: string;\n  header?: RtpHeader;\n  codec?: RTCRtpCodecParameters;\n  /**todo impl */\n  enabled = true;\n\n  readonly onReceiveRtp = new Event<[RtpPacket, Extensions?]>();\n  readonly onReceiveRtcp = new Event<[RtcpPacket]>();\n  readonly onSourceChanged = new Event<\n    [Pick<RtpHeader, \"sequenceNumber\" | \"timestamp\">]\n  >();\n\n  stopped = false;\n  muted = true;\n\n  constructor(\n    props: Partial<MediaStreamTrack> & Pick<MediaStreamTrack, \"kind\">,\n  ) {\n    super();\n    Object.assign(this, props);\n\n    this.onReceiveRtp.subscribe((rtp) => {\n      this.muted = false;\n      this.header = rtp.header;\n    });\n\n    this.label = `${this.remote ? \"remote\" : \"local\"} ${this.kind}`;\n  }\n\n  stop = () => {\n    this.stopped = true;\n    this.muted = true;\n    this.onReceiveRtp.complete();\n    this.emit(\"ended\");\n  };\n\n  writeRtp = (rtp: RtpPacket | Buffer) => {\n    if (this.remote) {\n      throw new Error(\"this is remoteTrack\");\n    }\n    if (this.stopped) {\n      return;\n    }\n\n    const packet = Buffer.isBuffer(rtp) ? RtpPacket.deSerialize(rtp) : rtp;\n    packet.header.payloadType =\n      this.codec?.payloadType ?? packet.header.payloadType;\n    this.onReceiveRtp.execute(packet);\n  };\n}\n\nexport class MediaStream {\n  id!: string;\n  tracks: MediaStreamTrack[] = [];\n\n  constructor(props: Partial<MediaStream> | MediaStreamTrack[]) {\n    if (Array.isArray(props)) {\n      this.tracks = props;\n    } else {\n      Object.assign(this, props);\n    }\n    this.id ??= v4();\n  }\n\n  addTrack(track: MediaStreamTrack) {\n    track.streamId = this.id;\n    this.tracks.push(track);\n  }\n\n  getTracks() {\n    return this.tracks;\n  }\n\n  getAudioTracks() {\n    return this.tracks.filter((track) => track.kind === \"audio\");\n  }\n\n  getVideoTracks() {\n    return this.tracks.filter((track) => track.kind === \"video\");\n  }\n}\n"]}

package/lib/webrtc/src/nonstandard/recorder/index.d.ts

@@ -41,6 +41,7 @@ export declare class MediaRecorder {
 export interface MediaRecorderOptions {
     width: number;
     height: number;
+    roll: number;
     jitterBufferLatency: number;
     jitterBufferSize: number;
     disableLipSync: boolean;

package/lib/webrtc/src/nonstandard/recorder/index.js.map

@@ -1 +1 @@
-{"version":3,"file":"index.js","sourceRoot":"","sources":["../../../../../src/nonstandard/recorder/index.ts"],"names":[],"mappings":";;;AAAA,iDAA6C;AAG7C,wCAA4C;AAI5C,MAAa,aAAa;IAOxB,YACS,KAoBJ;QApBH;;;;mBAAO,KAAK;WAoBT;QA3BL;;;;;WAAoB;QACpB;;;;;WAAa;QACb;;;;mBAA6B,EAAE;WAAC;QAChC;;;;mBAAU,KAAK;WAAC;QAChB;;;;mBAAU,IAAI,cAAK,EAAW;WAAC;QAyB7B,IAAI,CAAC,MAAM,GAAG,KAAK,CAAC,MAAM,IAAI,IAAI,CAAC,MAAM,CAAC;QAE1C,MAAM,EAAE,IAAI,EAAE,MAAM,EAAE,GAAG,KAAK,CAAC;QAE/B,IAAI,IAAI,EAAE,CAAC;YACT,IAAI,CAAC,GAAG,GAAG,IAAI,CAAC,KAAK,CAAC,GAAG,CAAC,CAAC,KAAK,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC;YACxC,IAAI,CAAC,MAAM,GAAG,CAAC,GAAG,EAAE;gBAClB,QAAQ,IAAI,CAAC,GAAG,EAAE,CAAC;oBACjB,KAAK,MAAM;wBACT,OAAO,IAAI,kBAAW,CAAC;4BACrB,GAAG,KAAK;4BACR,IAAI,EAAE,IAAK;4BACX,MAAM,EAAE,MAAO;yBAChB,CAAC,CAAC;oBACL;wBACE,MAAM,IAAI,KAAK,EAAE,CAAC;gBACtB,CAAC;YACH,CAAC,CAAC,EAAE,CAAC;QACP,CAAC;aAAM,CAAC;YACN,IAAI,CAAC,MAAM,GAAG,IAAI,kBAAW,CAAC;gBAC5B,GAAG,KAAK;gBACR,IAAI,EAAE,IAAK;gBACX,MAAM,EAAE,MAAO;aAChB,CAAC,CAAC;QACL,CAAC;QAED,IAAI,IAAI,CAAC,MAAM,CAAC,MAAM,GAAG,CAAC,EAAE,CAAC;YAC3B,IAAI,CAAC,KAAK,CAAC,WAAW,GAAG,IAAI,CAAC,MAAM,CAAC,MAAM,CAAC;YAC5C,IAAI,CAAC,KAAK,EAAE,CAAC,KAAK,CAAC,CAAC,KAAK,EAAE,EAAE;gBAC3B,IAAI,CAAC,OAAO,CAAC,OAAO,CAAC,KAAK,CAAC,CAAC;YAC9B,CAAC,CAAC,CAAC;QACL,CAAC;IACH,CAAC;IAED,KAAK,CAAC,QAAQ,CAAC,KAAuB;QACpC,IAAI,CAAC,MAAM,CAAC,IAAI,CAAC,KAAK,CAAC,CAAC;QACxB,MAAM,IAAI,CAAC,KAAK,EAAE,CAAC;IACrB,CAAC;IAEO,KAAK,CAAC,KAAK;QACjB,IACE,IAAI,CAAC,MAAM,CAAC,MAAM,KAAK,IAAI,CAAC,KAAK,CAAC,WAAW;YAC7C,IAAI,CAAC,OAAO,KAAK,KAAK,EACtB,CAAC;YACD,IAAI,CAAC,OAAO,GAAG,IAAI,CAAC;YACpB,MAAM,IAAI,CAAC,MAAM,CAAC,KAAK,CAAC,IAAI,CAAC,MAAM,CAAC,CAAC;QACvC,CAAC;IACH,CAAC;IAED,KAAK,CAAC,IAAI;QACR,MAAM,IAAI,CAAC,MAAM,CAAC,IAAI,EAAE,CAAC;IAC3B,CAAC;CACF;AAlFD,sCAkFC","sourcesContent":["import { Event } from \"../../imports/common\";\nimport type { MediaStreamTrack } from \"../../media/track\";\nimport type { MediaWriter, StreamEvent } from \"./writer\";\nimport { WebmFactory } from \"./writer/webm\";\n\nexport type { StreamEvent };\n\nexport class MediaRecorder {\n  writer: MediaWriter;\n  ext?: string;\n  tracks: MediaStreamTrack[] = [];\n  started = false;\n  onError = new Event<[Error]>();\n\n  constructor(\n    public props: Partial<MediaRecorderOptions> &\n      (\n        | {\n            numOfTracks: number;\n            tracks?: MediaStreamTrack[];\n          }\n        | {\n            numOfTracks?: number;\n            tracks: MediaStreamTrack[];\n          }\n      ) &\n      (\n        | {\n            path: string;\n            stream?: StreamEvent;\n          }\n        | {\n            path?: string;\n            stream: StreamEvent;\n          }\n      ),\n  ) {\n    this.tracks = props.tracks ?? this.tracks;\n\n    const { path, stream } = props;\n\n    if (path) {\n      this.ext = path.split(\".\").slice(-1)[0];\n      this.writer = (() => {\n        switch (this.ext) {\n          case \"webm\":\n            return new WebmFactory({\n              ...props,\n              path: path!,\n              stream: stream!,\n            });\n          default:\n            throw new Error();\n        }\n      })();\n    } else {\n      this.writer = new WebmFactory({\n        ...props,\n        path: path!,\n        stream: stream!,\n      });\n    }\n\n    if (this.tracks.length > 0) {\n      this.props.numOfTracks = this.tracks.length;\n      this.start().catch((error) => {\n        this.onError.execute(error);\n      });\n    }\n  }\n\n  async addTrack(track: MediaStreamTrack) {\n    this.tracks.push(track);\n    await this.start();\n  }\n\n  private async start() {\n    if (\n      this.tracks.length === this.props.numOfTracks &&\n      this.started === false\n    ) {\n      this.started = true;\n      await this.writer.start(this.tracks);\n    }\n  }\n\n  async stop() {\n    await this.writer.stop();\n  }\n}\n\nexport interface MediaRecorderOptions {\n  width: number;\n  height: number;\n  jitterBufferLatency: number;\n  jitterBufferSize: number;\n  disableLipSync: boolean;\n  waitForKeyframe: boolean;\n  defaultDuration: number;\n  tracks: MediaStreamTrack[];\n  disableNtp: boolean;\n}\n"]}
+{"version":3,"file":"index.js","sourceRoot":"","sources":["../../../../../src/nonstandard/recorder/index.ts"],"names":[],"mappings":";;;AAAA,iDAA6C;AAG7C,wCAA4C;AAI5C,MAAa,aAAa;IAOxB,YACS,KAoBJ;QApBH;;;;mBAAO,KAAK;WAoBT;QA3BL;;;;;WAAoB;QACpB;;;;;WAAa;QACb;;;;mBAA6B,EAAE;WAAC;QAChC;;;;mBAAU,KAAK;WAAC;QAChB;;;;mBAAU,IAAI,cAAK,EAAW;WAAC;QAyB7B,IAAI,CAAC,MAAM,GAAG,KAAK,CAAC,MAAM,IAAI,IAAI,CAAC,MAAM,CAAC;QAE1C,MAAM,EAAE,IAAI,EAAE,MAAM,EAAE,GAAG,KAAK,CAAC;QAE/B,IAAI,IAAI,EAAE,CAAC;YACT,IAAI,CAAC,GAAG,GAAG,IAAI,CAAC,KAAK,CAAC,GAAG,CAAC,CAAC,KAAK,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC;YACxC,IAAI,CAAC,MAAM,GAAG,CAAC,GAAG,EAAE;gBAClB,QAAQ,IAAI,CAAC,GAAG,EAAE,CAAC;oBACjB,KAAK,MAAM;wBACT,OAAO,IAAI,kBAAW,CAAC;4BACrB,GAAG,KAAK;4BACR,IAAI,EAAE,IAAK;4BACX,MAAM,EAAE,MAAO;yBAChB,CAAC,CAAC;oBACL;wBACE,MAAM,IAAI,KAAK,EAAE,CAAC;gBACtB,CAAC;YACH,CAAC,CAAC,EAAE,CAAC;QACP,CAAC;aAAM,CAAC;YACN,IAAI,CAAC,MAAM,GAAG,IAAI,kBAAW,CAAC;gBAC5B,GAAG,KAAK;gBACR,IAAI,EAAE,IAAK;gBACX,MAAM,EAAE,MAAO;aAChB,CAAC,CAAC;QACL,CAAC;QAED,IAAI,IAAI,CAAC,MAAM,CAAC,MAAM,GAAG,CAAC,EAAE,CAAC;YAC3B,IAAI,CAAC,KAAK,CAAC,WAAW,GAAG,IAAI,CAAC,MAAM,CAAC,MAAM,CAAC;YAC5C,IAAI,CAAC,KAAK,EAAE,CAAC,KAAK,CAAC,CAAC,KAAK,EAAE,EAAE;gBAC3B,IAAI,CAAC,OAAO,CAAC,OAAO,CAAC,KAAK,CAAC,CAAC;YAC9B,CAAC,CAAC,CAAC;QACL,CAAC;IACH,CAAC;IAED,KAAK,CAAC,QAAQ,CAAC,KAAuB;QACpC,IAAI,CAAC,MAAM,CAAC,IAAI,CAAC,KAAK,CAAC,CAAC;QACxB,MAAM,IAAI,CAAC,KAAK,EAAE,CAAC;IACrB,CAAC;IAEO,KAAK,CAAC,KAAK;QACjB,IACE,IAAI,CAAC,MAAM,CAAC,MAAM,KAAK,IAAI,CAAC,KAAK,CAAC,WAAW;YAC7C,IAAI,CAAC,OAAO,KAAK,KAAK,EACtB,CAAC;YACD,IAAI,CAAC,OAAO,GAAG,IAAI,CAAC;YACpB,MAAM,IAAI,CAAC,MAAM,CAAC,KAAK,CAAC,IAAI,CAAC,MAAM,CAAC,CAAC;QACvC,CAAC;IACH,CAAC;IAED,KAAK,CAAC,IAAI;QACR,MAAM,IAAI,CAAC,MAAM,CAAC,IAAI,EAAE,CAAC;IAC3B,CAAC;CACF;AAlFD,sCAkFC","sourcesContent":["import { Event } from \"../../imports/common\";\nimport type { MediaStreamTrack } from \"../../media/track\";\nimport type { MediaWriter, StreamEvent } from \"./writer\";\nimport { WebmFactory } from \"./writer/webm\";\n\nexport type { StreamEvent };\n\nexport class MediaRecorder {\n  writer: MediaWriter;\n  ext?: string;\n  tracks: MediaStreamTrack[] = [];\n  started = false;\n  onError = new Event<[Error]>();\n\n  constructor(\n    public props: Partial<MediaRecorderOptions> &\n      (\n        | {\n            numOfTracks: number;\n            tracks?: MediaStreamTrack[];\n          }\n        | {\n            numOfTracks?: number;\n            tracks: MediaStreamTrack[];\n          }\n      ) &\n      (\n        | {\n            path: string;\n            stream?: StreamEvent;\n          }\n        | {\n            path?: string;\n            stream: StreamEvent;\n          }\n      ),\n  ) {\n    this.tracks = props.tracks ?? this.tracks;\n\n    const { path, stream } = props;\n\n    if (path) {\n      this.ext = path.split(\".\").slice(-1)[0];\n      this.writer = (() => {\n        switch (this.ext) {\n          case \"webm\":\n            return new WebmFactory({\n              ...props,\n              path: path!,\n              stream: stream!,\n            });\n          default:\n            throw new Error();\n        }\n      })();\n    } else {\n      this.writer = new WebmFactory({\n        ...props,\n        path: path!,\n        stream: stream!,\n      });\n    }\n\n    if (this.tracks.length > 0) {\n      this.props.numOfTracks = this.tracks.length;\n      this.start().catch((error) => {\n        this.onError.execute(error);\n      });\n    }\n  }\n\n  async addTrack(track: MediaStreamTrack) {\n    this.tracks.push(track);\n    await this.start();\n  }\n\n  private async start() {\n    if (\n      this.tracks.length === this.props.numOfTracks &&\n      this.started === false\n    ) {\n      this.started = true;\n      await this.writer.start(this.tracks);\n    }\n  }\n\n  async stop() {\n    await this.writer.stop();\n  }\n}\n\nexport interface MediaRecorderOptions {\n  width: number;\n  height: number;\n  roll: number;\n  jitterBufferLatency: number;\n  jitterBufferSize: number;\n  disableLipSync: boolean;\n  waitForKeyframe: boolean;\n  defaultDuration: number;\n  tracks: MediaStreamTrack[];\n  disableNtp: boolean;\n}\n"]}

package/lib/webrtc/src/nonstandard/recorder/writer/webm.js

@@ -55,6 +55,7 @@ class WebmFactory extends _1.MediaWriter {
                     trackNumber,
                     width: this.props.width ?? 640,
                     height: this.props.height ?? 360,
+                    roll: this.props.roll,
                     payloadType,
                     track,
                 };

package/lib/webrtc/src/nonstandard/recorder/writer/webm.js.map

@@ -1 +1 @@
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{ unlink } from \"fs/promises\";\nimport { EventDisposer } from \"../../../imports/common\";\n\nimport { MediaWriter } from \".\";\nimport { type MediaStreamTrack, WeriftError } from \"../../..\";\nimport {\n  DepacketizeCallback,\n  JitterBufferCallback,\n  LipsyncCallback,\n  NtpTimeCallback,\n  RtcpSourceCallback,\n  RtpSourceCallback,\n  RtpTimeCallback,\n  type SupportedCodec,\n  WebmCallback,\n  saveToFileSystem,\n} from \"../../../../../rtp/src/extra\";\n\nconst sourcePath = \"packages/webrtc/src/nonstandard/recorder/writer/webm.ts\";\n\nexport class WebmFactory extends MediaWriter {\n  rtpSources: RtpSourceCallback[] = [];\n\n  unSubscribers = new EventDisposer();\n\n  async start(tracks: MediaStreamTrack[]) {\n    if (this.props.path) {\n      await unlink(this.props.path).catch((e) => e);\n    }\n\n    const inputTracks = tracks.map((track, i) => {\n      const trackNumber = i + 1;\n      const payloadType = track.codec!.payloadType;\n\n      if (track.kind === \"video\") {\n        const codec = ((): SupportedCodec => {\n          switch (track.codec?.name.toLowerCase() as SupportedVideoCodec) {\n            case \"vp8\":\n              return \"VP8\";\n            case \"vp9\":\n              return \"VP9\";\n            case \"h264\":\n              return \"MPEG4/ISO/AVC\";\n            case \"av1x\":\n              return \"AV1\";\n            default:\n              throw new WeriftError({\n                message: \"unsupported codec\",\n                payload: { track, path: sourcePath },\n              });\n          }\n        })();\n        return {\n          kind: \"video\" as const,\n          codec,\n          clockRate: 90000,\n          trackNumber,\n          width: this.props.width ?? 640,\n          height: this.props.height ?? 360,\n          payloadType,\n          track,\n        };\n      } else {\n        return {\n          kind: \"audio\" as const,\n          codec: \"OPUS\" as const,\n          clockRate: 48000,\n          trackNumber,\n          payloadType,\n          track,\n        };\n      }\n    });\n\n    const webm = new WebmCallback(inputTracks, {\n      duration: this.props.defaultDuration ?? 1000 * 60 * 60 * 24,\n    });\n    const lipsync = new LipsyncCallback();\n\n    if (inputTracks.length === 1 || this.props.disableNtp) {\n      this.props.disableLipSync = true;\n    }\n\n    this.rtpSources = inputTracks.map(({ track, clockRate, codec }) => {\n      const rtpSource = new RtpSourceCallback();\n      const rtcpSource = new RtcpSourceCallback();\n      track.onReceiveRtp\n        .subscribe((rtp) => {\n          rtpSource.input(rtp.clone());\n        })\n        .disposer(this.unSubscribers);\n      track.onReceiveRtcp\n        .subscribe((rtcp) => {\n          rtcpSource.input(rtcp);\n        })\n        .disposer(this.unSubscribers);\n      const time = this.props.disableNtp\n        ? new RtpTimeCallback(clockRate)\n        : new NtpTimeCallback(clockRate);\n\n      if (track.kind === \"video\") {\n        const depacketizer = new DepacketizeCallback(codec, {\n          isFinalPacketInSequence: (h) => h.marker,\n        });\n        const jitterBuffer = new JitterBufferCallback(clockRate, {\n          bufferSize: this.props.jitterBufferSize,\n          latency: this.props.jitterBufferLatency,\n        });\n\n        rtpSource.pipe(jitterBuffer.input);\n        rtcpSource.pipe(time.input);\n\n        jitterBuffer.pipe(time.input);\n        time.pipe(depacketizer.input);\n        if (this.props.disableLipSync) {\n          depacketizer.pipe(webm.inputVideo);\n        } else {\n          depacketizer.pipe(lipsync.inputVideo);\n          lipsync.pipeVideo(webm.inputVideo);\n        }\n      } else {\n        const depacketizer = new DepacketizeCallback(codec);\n\n        rtpSource.pipe(time.input);\n        rtcpSource.pipe(time.input);\n\n        time.pipe(depacketizer.input);\n        if (this.props.disableLipSync) {\n          depacketizer.pipe(webm.inputAudio);\n        } else {\n          depacketizer.pipe(lipsync.inputAudio);\n          lipsync.pipeAudio(webm.inputAudio);\n        }\n      }\n\n      return rtpSource;\n    });\n    if (this.props.path) {\n      webm.pipe(saveToFileSystem(this.props.path));\n    } else if (this.props.stream) {\n      webm.pipe(async (o) => {\n        this.props.stream.execute(o);\n      });\n    }\n  }\n\n  async stop() {\n    await Promise.all(this.rtpSources.map((r) => r.stop()));\n    this.unSubscribers.dispose();\n  }\n}\n\nconst supportedVideoCodecs = [\"h264\", \"vp8\", \"vp9\", \"av1x\"] as const;\ntype SupportedVideoCodec = (typeof supportedVideoCodecs)[number];\n"]}
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{ unlink } from \"fs/promises\";\nimport { EventDisposer } from \"../../../imports/common\";\n\nimport { MediaWriter } from \".\";\nimport { type MediaStreamTrack, WeriftError } from \"../../..\";\nimport {\n  DepacketizeCallback,\n  JitterBufferCallback,\n  LipsyncCallback,\n  NtpTimeCallback,\n  RtcpSourceCallback,\n  RtpSourceCallback,\n  RtpTimeCallback,\n  type SupportedCodec,\n  WebmCallback,\n  type WebmTrack,\n  saveToFileSystem,\n} from \"../../../../../rtp/src/extra\";\n\nconst sourcePath = \"packages/webrtc/src/nonstandard/recorder/writer/webm.ts\";\n\nexport class WebmFactory extends MediaWriter {\n  rtpSources: RtpSourceCallback[] = [];\n\n  unSubscribers = new EventDisposer();\n\n  async start(tracks: MediaStreamTrack[]) {\n    if (this.props.path) {\n      await unlink(this.props.path).catch((e) => e);\n    }\n\n    const inputTracks: (WebmTrack & { track: MediaStreamTrack })[] = tracks.map(\n      (track, i) => {\n        const trackNumber = i + 1;\n        const payloadType = track.codec!.payloadType;\n\n        if (track.kind === \"video\") {\n          const codec = ((): SupportedCodec => {\n            switch (track.codec?.name.toLowerCase() as SupportedVideoCodec) {\n              case \"vp8\":\n                return \"VP8\";\n              case \"vp9\":\n                return \"VP9\";\n              case \"h264\":\n                return \"MPEG4/ISO/AVC\";\n              case \"av1x\":\n                return \"AV1\";\n              default:\n                throw new WeriftError({\n                  message: \"unsupported codec\",\n                  payload: { track, path: sourcePath },\n                });\n            }\n          })();\n          return {\n            kind: \"video\" as const,\n            codec,\n            clockRate: 90000,\n            trackNumber,\n            width: this.props.width ?? 640,\n            height: this.props.height ?? 360,\n            roll: this.props.roll,\n            payloadType,\n            track,\n          };\n        } else {\n          return {\n            kind: \"audio\" as const,\n            codec: \"OPUS\" as const,\n            clockRate: 48000,\n            trackNumber,\n            payloadType,\n            track,\n          };\n        }\n      },\n    );\n\n    const webm = new WebmCallback(inputTracks, {\n      duration: this.props.defaultDuration ?? 1000 * 60 * 60 * 24,\n    });\n    const lipsync = new LipsyncCallback();\n\n    if (inputTracks.length === 1 || this.props.disableNtp) {\n      this.props.disableLipSync = true;\n    }\n\n    this.rtpSources = inputTracks.map(({ track, clockRate, codec }) => {\n      const rtpSource = new RtpSourceCallback();\n      const rtcpSource = new RtcpSourceCallback();\n      track.onReceiveRtp\n        .subscribe((rtp) => {\n          rtpSource.input(rtp.clone());\n        })\n        .disposer(this.unSubscribers);\n      track.onReceiveRtcp\n        .subscribe((rtcp) => {\n          rtcpSource.input(rtcp);\n        })\n        .disposer(this.unSubscribers);\n      const time = this.props.disableNtp\n        ? new RtpTimeCallback(clockRate)\n        : new NtpTimeCallback(clockRate);\n\n      if (track.kind === \"video\") {\n        const depacketizer = new DepacketizeCallback(codec, {\n          isFinalPacketInSequence: (h) => h.marker,\n        });\n        const jitterBuffer = new JitterBufferCallback(clockRate, {\n          bufferSize: this.props.jitterBufferSize,\n          latency: this.props.jitterBufferLatency,\n        });\n\n        rtpSource.pipe(jitterBuffer.input);\n        rtcpSource.pipe(time.input);\n\n        jitterBuffer.pipe(time.input);\n        time.pipe(depacketizer.input);\n        if (this.props.disableLipSync) {\n          depacketizer.pipe(webm.inputVideo);\n        } else {\n          depacketizer.pipe(lipsync.inputVideo);\n          lipsync.pipeVideo(webm.inputVideo);\n        }\n      } else {\n        const depacketizer = new DepacketizeCallback(codec);\n\n        rtpSource.pipe(time.input);\n        rtcpSource.pipe(time.input);\n\n        time.pipe(depacketizer.input);\n        if (this.props.disableLipSync) {\n          depacketizer.pipe(webm.inputAudio);\n        } else {\n          depacketizer.pipe(lipsync.inputAudio);\n          lipsync.pipeAudio(webm.inputAudio);\n        }\n      }\n\n      return rtpSource;\n    });\n    if (this.props.path) {\n      webm.pipe(saveToFileSystem(this.props.path));\n    } else if (this.props.stream) {\n      webm.pipe(async (o) => {\n        this.props.stream.execute(o);\n      });\n    }\n  }\n\n  async stop() {\n    await Promise.all(this.rtpSources.map((r) => r.stop()));\n    this.unSubscribers.dispose();\n  }\n}\n\nconst supportedVideoCodecs = [\"h264\", \"vp8\", \"vp9\", \"av1x\"] as const;\ntype SupportedVideoCodec = (typeof supportedVideoCodecs)[number];\n"]}