werift 0.16.0 → 0.16.2

package/lib/rtp/src/processor/base.d.ts

@@ -1,18 +0,0 @@
-import Event from "rx.mini";
-import { RtcpPacket, RtpPacket } from "..";
-export declare abstract class Pipeline {
-    protected children?: Pipeline | Output;
-    private disposer?;
-    constructor(streams?: {
-        rtpStream?: Event<[RtpPacket]>;
-        rtcpStream?: Event<[RtcpPacket]>;
-    });
-    pipe(children: Pipeline | Output): Pipeline | Output;
-    pushRtpPackets(packets: RtpPacket[]): void;
-    pushRtcpPackets(packets: RtcpPacket[]): void;
-    stop(): void;
-}
-export declare abstract class Output {
-    pushRtpPackets?(packets: RtpPacket[]): void;
-    pushRtcpPackets?(packets: RtcpPacket[]): void;
-}

package/lib/rtp/src/processor/base.js

@@ -1,39 +0,0 @@
-"use strict";
-Object.defineProperty(exports, "__esModule", { value: true });
-exports.Output = exports.Pipeline = void 0;
-class Pipeline {
-    constructor(streams) {
-        const disposers = [];
-        {
-            const { unSubscribe } = streams?.rtpStream?.subscribe?.((packet) => {
-                this.pushRtpPackets([packet]);
-            }) ?? {};
-            disposers.push(unSubscribe);
-        }
-        {
-            const { unSubscribe } = streams?.rtcpStream?.subscribe?.((packet) => {
-                this.pushRtcpPackets([packet]);
-            }) ?? {};
-            disposers.push(unSubscribe);
-        }
-        this.disposer = () => {
-            disposers.forEach((d) => d?.());
-        };
-    }
-    pipe(children) {
-        this.children = children;
-        return this;
-    }
-    pushRtpPackets(packets) { }
-    pushRtcpPackets(packets) { }
-    stop() {
-        this.disposer?.();
-    }
-}
-exports.Pipeline = Pipeline;
-class Output {
-    pushRtpPackets(packets) { }
-    pushRtcpPackets(packets) { }
-}
-exports.Output = Output;
-//# sourceMappingURL=base.js.map

package/lib/rtp/src/processor/base.js.map

@@ -1 +0,0 @@
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package/lib/rtp/src/processor/lipsync.d.ts

@@ -1,30 +0,0 @@
-import Event from "rx.mini";
-import { RtpPacket } from "..";
-import { RtcpPacket } from "../rtcp/rtcp";
-import { Pipeline } from "./base";
-export declare class LipSync extends Pipeline {
-    clockRate: number;
-    mismatch: number;
-    baseNtpTimestamp?: bigint;
-    baseRtpTimestamp?: number;
-    private rtpPackets;
-    constructor(clockRate: number, mismatch: number, streams?: {
-        rtpStream?: Event<[RtpPacket]>;
-        rtcpStream?: Event<[RtcpPacket]>;
-    });
-    pushRtcpPackets(packets: RtcpPacket[]): void;
-    private srReceived;
-    pushRtpPackets(packets: RtpPacket[]): void;
-    private calcNtpTime;
-}
-export declare const ntpTime2Time: (ntp: bigint) => number;
-/**4294967295 */
-export declare const Max32bit: number;
-export interface BufferResolve {
-    packets: {
-        packet: RtpPacket;
-        offset: number;
-    }[];
-    /**NTP seconds */
-    startAtNtpTime: number;
-}

package/lib/rtp/src/processor/lipsync.js

@@ -1,92 +0,0 @@
-"use strict";
-Object.defineProperty(exports, "__esModule", { value: true });
-exports.Max32bit = exports.ntpTime2Time = exports.LipSync = void 0;
-const src_1 = require("../../../common/src");
-const __1 = require("..");
-const base_1 = require("./base");
-// #### リップシンク
-// ```
-//         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
-//        +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-// sender |              NTP timestamp, most significant word             |
-// info   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//        |             NTP timestamp, least significant word             |
-//        +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//        |                         RTP timestamp                         |
-//        +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//        |                     sender's packet count                     |
-//        +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//        |                      sender's octet count                     |
-//        +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
-// ```
-// - RTCP-SRにはNTP timestampとRTCP-SRを送った時点で最後に送ったRTPパケットのtimestampが載っている
-// - RTP timestampは開始時点の値がランダムなので、Audio,VideoのRTPパケットの同期にはそのままでは使えない
-//     - ある時点のRTP timestampのNTP timestampがRTCP-SRでわかるので、任意のRTP timestampに対応するNTP timestampを計算できる
-// - Audio,VideoのRTPパケットの同期にはRTP timestampをNTP timestampに変換して行えば良い
-// WIP
-// todo impl
-class LipSync extends base_1.Pipeline {
-    constructor(clockRate, mismatch, streams) {
-        super(streams);
-        this.clockRate = clockRate;
-        this.mismatch = mismatch;
-        this.rtpPackets = {};
-    }
-    pushRtcpPackets(packets) {
-        packets.forEach((sr) => {
-            if (sr instanceof __1.RtcpSrPacket) {
-                this.srReceived(sr);
-            }
-        });
-        this.children?.pushRtcpPackets?.(packets);
-    }
-    srReceived(sr) {
-        const { ntpTimestamp, rtpTimestamp } = sr.senderInfo;
-        this.baseNtpTimestamp = ntpTimestamp;
-        this.baseRtpTimestamp = rtpTimestamp;
-    }
-    pushRtpPackets(packets) {
-        packets.forEach((p) => {
-            this.rtpPackets[p.header.payloadType] =
-                this.rtpPackets[p.header.payloadType] ?? [];
-            this.rtpPackets[p.header.payloadType].push(p);
-        });
-        if (Object.keys(this.rtpPackets).length === 2) {
-            const [a, b] = Object.values(this.rtpPackets);
-            const lastA = this.calcNtpTime(a.at(-1).header.timestamp);
-            const lastB = this.calcNtpTime(b.at(-1).header.timestamp);
-            if (lastA == undefined || lastB == undefined) {
-                this.children?.pushRtpPackets?.(packets);
-            }
-            else {
-                //
-            }
-        }
-    }
-    calcNtpTime(rtpTimestamp) {
-        if (!this.baseRtpTimestamp || !this.baseNtpTimestamp) {
-            return;
-        }
-        // base rtpTimestamp is rollover
-        if (rtpTimestamp - this.baseRtpTimestamp > exports.Max32bit - this.clockRate * 60) {
-            this.baseRtpTimestamp += exports.Max32bit;
-        }
-        // target rtpTimestamp is rollover
-        else if (rtpTimestamp + (exports.Max32bit - this.clockRate * 60) - this.baseRtpTimestamp <
-            0) {
-            rtpTimestamp += exports.Max32bit;
-        }
-        const elapsed = (rtpTimestamp - this.baseRtpTimestamp) / this.clockRate;
-        return (0, exports.ntpTime2Time)(this.baseNtpTimestamp) + elapsed;
-    }
-}
-exports.LipSync = LipSync;
-const ntpTime2Time = (ntp) => {
-    const [ntpSec, ntpMsec] = (0, src_1.bufferReader)((0, src_1.bufferWriter)([8], [ntp]), [4, 4]);
-    return Number(`${ntpSec}.${ntpMsec}`);
-};
-exports.ntpTime2Time = ntpTime2Time;
-/**4294967295 */
-exports.Max32bit = Number((0x01n << 32n) - 1n);
-//# sourceMappingURL=lipsync.js.map

package/lib/rtp/src/processor/lipsync.js.map

@@ -1 +0,0 @@
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RTCP-SRにはNTP timestampとRTCP-SRを送った時点で最後に送ったRTPパケットのtimestampが載っている\n// - RTP timestampは開始時点の値がランダムなので、Audio,VideoのRTPパケットの同期にはそのままでは使えない\n//     - ある時点のRTP timestampのNTP timestampがRTCP-SRでわかるので、任意のRTP timestampに対応するNTP timestampを計算できる\n// - Audio,VideoのRTPパケットの同期にはRTP timestampをNTP timestampに変換して行えば良い\n\n// WIP\n// todo impl\nexport class LipSync extends Pipeline {\n  baseNtpTimestamp?: bigint;\n  baseRtpTimestamp?: number;\n  private rtpPackets: { [pt: number]: RtpPacket[] } = {};\n\n  constructor(\n    public clockRate: number,\n    public mismatch: number,\n    streams?: {\n      rtpStream?: Event<[RtpPacket]>;\n      rtcpStream?: Event<[RtcpPacket]>;\n    }\n  ) {\n    super(streams);\n  }\n\n  pushRtcpPackets(packets: RtcpPacket[]) {\n    packets.forEach((sr) => {\n      if (sr instanceof RtcpSrPacket) {\n        this.srReceived(sr);\n      }\n    });\n    this.children?.pushRtcpPackets?.(packets);\n  }\n\n  private srReceived(sr: RtcpSrPacket) {\n    const { ntpTimestamp, rtpTimestamp } = sr.senderInfo;\n    this.baseNtpTimestamp = ntpTimestamp;\n    this.baseRtpTimestamp = rtpTimestamp;\n  }\n\n  pushRtpPackets(packets: RtpPacket[]) {\n    packets.forEach((p) => {\n      this.rtpPackets[p.header.payloadType] =\n        this.rtpPackets[p.header.payloadType] ?? [];\n      this.rtpPackets[p.header.payloadType].push(p);\n    });\n    if (Object.keys(this.rtpPackets).length === 2) {\n      const [a, b] = Object.values(this.rtpPackets);\n      const lastA = this.calcNtpTime(a.at(-1)!.header.timestamp);\n      const lastB = this.calcNtpTime(b.at(-1)!.header.timestamp);\n\n      if (lastA == undefined || lastB == undefined) {\n        this.children?.pushRtpPackets?.(packets);\n      } else {\n        //\n      }\n    }\n  }\n\n  private calcNtpTime(rtpTimestamp: number) {\n    if (!this.baseRtpTimestamp || !this.baseNtpTimestamp) {\n      return;\n    }\n\n    // base rtpTimestamp is rollover\n    if (rtpTimestamp - this.baseRtpTimestamp > Max32bit - this.clockRate * 60) {\n      this.baseRtpTimestamp += Max32bit;\n    }\n    // target rtpTimestamp is rollover\n    else if (\n      rtpTimestamp + (Max32bit - this.clockRate * 60) - this.baseRtpTimestamp <\n      0\n    ) {\n      rtpTimestamp += Max32bit;\n    }\n\n    const elapsed = (rtpTimestamp - this.baseRtpTimestamp) / this.clockRate;\n\n    return ntpTime2Time(this.baseNtpTimestamp) + elapsed;\n  }\n}\n\nexport const ntpTime2Time = (ntp: bigint) => {\n  const [ntpSec, ntpMsec] = bufferReader(bufferWriter([8], [ntp]), [4, 4]);\n\n  return Number(`${ntpSec}.${ntpMsec}`);\n};\n\n/**4294967295 */\nexport const Max32bit = Number((0x01n << 32n) - 1n);\n\nexport interface BufferResolve {\n  packets: {\n    packet: RtpPacket;\n    offset: number;\n  }[];\n  /**NTP seconds */\n  startAtNtpTime: number;\n}\n"]}

package/lib/rtp/src/processor/sampleBuilder.d.ts

@@ -1,13 +0,0 @@
-import Event from "rx.mini";
-import { RtcpPacket, RtpHeader, RtpPacket } from "..";
-import { Pipeline } from "./base";
-export declare class SampleBuilder extends Pipeline {
-    private isFinalPacketInSequence;
-    private buffering;
-    constructor(isFinalPacketInSequence: (header: RtpHeader) => boolean, streams?: {
-        rtpStream?: Event<[RtpPacket]>;
-        rtcpStream?: Event<[RtcpPacket]>;
-    });
-    pushRtpPackets(incoming: RtpPacket[]): void;
-    pushRtcpPackets(packets: RtcpPacket[]): void;
-}

package/lib/rtp/src/processor/sampleBuilder.js

@@ -1,33 +0,0 @@
-"use strict";
-Object.defineProperty(exports, "__esModule", { value: true });
-exports.SampleBuilder = void 0;
-const helper_1 = require("../helper");
-const base_1 = require("./base");
-class SampleBuilder extends base_1.Pipeline {
-    constructor(isFinalPacketInSequence, streams) {
-        super(streams);
-        this.isFinalPacketInSequence = isFinalPacketInSequence;
-        this.buffering = [];
-    }
-    pushRtpPackets(incoming) {
-        this.buffering = [...this.buffering, ...incoming];
-        let tail;
-        for (const [i, p] of (0, helper_1.enumerate)(this.buffering)) {
-            if (this.isFinalPacketInSequence(p.header)) {
-                tail = i;
-                break;
-            }
-        }
-        if (tail == undefined) {
-            return;
-        }
-        const packets = this.buffering.slice(0, tail + 1);
-        this.buffering = this.buffering.slice(tail + 1);
-        this.children?.pushRtpPackets?.(packets);
-    }
-    pushRtcpPackets(packets) {
-        this.children?.pushRtcpPackets?.(packets);
-    }
-}
-exports.SampleBuilder = SampleBuilder;
-//# sourceMappingURL=sampleBuilder.js.map

package/lib/rtp/src/processor/sampleBuilder.js.map

@@ -1 +0,0 @@
-{"version":3,"file":"sampleBuilder.js","sourceRoot":"","sources":["../../../../../rtp/src/processor/sampleBuilder.ts"],"names":[],"mappings":";;;AAGA,sCAAsC;AACtC,iCAAkC;AAElC,MAAa,aAAc,SAAQ,eAAQ;IAGzC,YACU,uBAAuD,EAC/D,OAGC;QAED,KAAK,CAAC,OAAO,CAAC,CAAC;QANP,4BAAuB,GAAvB,uBAAuB,CAAgC;QAHzD,cAAS,GAAgB,EAAE,CAAC;IAUpC,CAAC;IAED,cAAc,CAAC,QAAqB;QAClC,IAAI,CAAC,SAAS,GAAG,CAAC,GAAG,IAAI,CAAC,SAAS,EAAE,GAAG,QAAQ,CAAC,CAAC;QAElD,IAAI,IAAwB,CAAC;QAC7B,KAAK,MAAM,CAAC,CAAC,EAAE,CAAC,CAAC,IAAI,IAAA,kBAAS,EAAC,IAAI,CAAC,SAAS,CAAC,EAAE;YAC9C,IAAI,IAAI,CAAC,uBAAuB,CAAC,CAAC,CAAC,MAAM,CAAC,EAAE;gBAC1C,IAAI,GAAG,CAAC,CAAC;gBACT,MAAM;aACP;SACF;QACD,IAAI,IAAI,IAAI,SAAS,EAAE;YACrB,OAAO;SACR;QAED,MAAM,OAAO,GAAG,IAAI,CAAC,SAAS,CAAC,KAAK,CAAC,CAAC,EAAE,IAAI,GAAG,CAAC,CAAC,CAAC;QAClD,IAAI,CAAC,SAAS,GAAG,IAAI,CAAC,SAAS,CAAC,KAAK,CAAC,IAAI,GAAG,CAAC,CAAC,CAAC;QAEhD,IAAI,CAAC,QAAQ,EAAE,cAAc,EAAE,CAAC,OAAO,CAAC,CAAC;IAC3C,CAAC;IAED,eAAe,CAAC,OAAqB;QACnC,IAAI,CAAC,QAAQ,EAAE,eAAe,EAAE,CAAC,OAAO,CAAC,CAAC;IAC5C,CAAC;CACF;AApCD,sCAoCC","sourcesContent":["import Event from \"rx.mini\";\n\nimport { RtcpPacket, RtpHeader, RtpPacket } from \"..\";\nimport { enumerate } from \"../helper\";\nimport { Pipeline } from \"./base\";\n\nexport class SampleBuilder extends Pipeline {\n  private buffering: RtpPacket[] = [];\n\n  constructor(\n    private isFinalPacketInSequence: (header: RtpHeader) => boolean,\n    streams?: {\n      rtpStream?: Event<[RtpPacket]>;\n      rtcpStream?: Event<[RtcpPacket]>;\n    }\n  ) {\n    super(streams);\n  }\n\n  pushRtpPackets(incoming: RtpPacket[]) {\n    this.buffering = [...this.buffering, ...incoming];\n\n    let tail: number | undefined;\n    for (const [i, p] of enumerate(this.buffering)) {\n      if (this.isFinalPacketInSequence(p.header)) {\n        tail = i;\n        break;\n      }\n    }\n    if (tail == undefined) {\n      return;\n    }\n\n    const packets = this.buffering.slice(0, tail + 1);\n    this.buffering = this.buffering.slice(tail + 1);\n\n    this.children?.pushRtpPackets?.(packets);\n  }\n\n  pushRtcpPackets(packets: RtcpPacket[]) {\n    this.children?.pushRtcpPackets?.(packets);\n  }\n}\n"]}

package/lib/rtp/src/processor_v2/depacketizer.d.ts

@@ -1,17 +0,0 @@
-/// <reference types="node" />
-/// <reference types="node" />
-import { TransformStream } from "stream/web";
-import { RtpHeader } from "..";
-import { RtpOutput } from "./source/rtp";
-export declare type DepacketizerInput = RtpOutput;
-export interface DepacketizerOutput {
-    frame?: {
-        data: Buffer;
-        isKeyframe: boolean;
-        timestamp: number;
-    };
-    eol?: boolean;
-}
-export declare const depacketizeTransformer: (isFinalPacketInSequence: (header: RtpHeader) => boolean, codec: string, options?: {
-    waitForKeyframe?: boolean | undefined;
-} | undefined) => TransformStream<RtpOutput, DepacketizerOutput>;

package/lib/rtp/src/processor_v2/depacketizer.js

@@ -1,84 +0,0 @@
-"use strict";
-var __importDefault = (this && this.__importDefault) || function (mod) {
-    return (mod && mod.__esModule) ? mod : { "default": mod };
-};
-Object.defineProperty(exports, "__esModule", { value: true });
-exports.depacketizeTransformer = void 0;
-const debug_1 = __importDefault(require("debug"));
-const web_1 = require("stream/web");
-const __1 = require("..");
-const codec_1 = require("../codec");
-const helper_1 = require("../helper");
-const srcPath = `werift-rtp : packages/rtp/src/processor_v2/depacketizer.ts`;
-const log = (0, debug_1.default)(srcPath);
-const depacketizeTransformer = (...args) => new DepacketizeTransformer(...args).transform;
-exports.depacketizeTransformer = depacketizeTransformer;
-class DepacketizeTransformer {
-    constructor(isFinalPacketInSequence, codec, options = {}) {
-        this.isFinalPacketInSequence = isFinalPacketInSequence;
-        this.buffering = [];
-        this.packetLostHappened = false;
-        this.transform = new web_1.TransformStream({
-            transform: (input, output) => {
-                if (!input.rtp) {
-                    if (input.eol) {
-                        output.enqueue({ eol: true });
-                    }
-                    return;
-                }
-                const packets = this.onRtp(input.rtp);
-                if (packets) {
-                    try {
-                        const { data, isKeyframe } = (0, codec_1.dePacketizeRtpPackets)(codec, packets);
-                        if (this.packetLostHappened) {
-                            if (isKeyframe) {
-                                this.packetLostHappened = false;
-                            }
-                            else if (options.waitForKeyframe) {
-                                return;
-                            }
-                        }
-                        output.enqueue({
-                            frame: {
-                                data,
-                                isKeyframe,
-                                timestamp: packets[0].header.timestamp,
-                            },
-                        });
-                    }
-                    catch (error) { }
-                }
-            },
-        });
-    }
-    onRtp(rtp) {
-        const { sequenceNumber } = rtp.header;
-        if (this.lastSeqNum != undefined) {
-            const expect = (0, __1.uint16Add)(this.lastSeqNum, 1);
-            if (sequenceNumber < expect) {
-                return;
-            }
-            if (sequenceNumber > expect) {
-                log("packet lost happened", { expect, sequenceNumber });
-                this.packetLostHappened = true;
-                this.buffering = [];
-            }
-        }
-        this.buffering.push(rtp);
-        this.lastSeqNum = sequenceNumber;
-        let finalPacket;
-        for (const [i, p] of (0, helper_1.enumerate)(this.buffering)) {
-            if (this.isFinalPacketInSequence(p.header)) {
-                finalPacket = i;
-                break;
-            }
-        }
-        if (finalPacket == undefined) {
-            return;
-        }
-        const packets = this.buffering;
-        this.buffering = [];
-        return packets;
-    }
-}
-//# sourceMappingURL=depacketizer.js.map

package/lib/rtp/src/processor_v2/depacketizer.js.map

@@ -1 +0,0 @@
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isKeyframe: boolean; timestamp: number };\n  eol?: boolean;\n}\n\nexport const depacketizeTransformer = (\n  ...args: ConstructorParameters<typeof DepacketizeTransformer>\n) => new DepacketizeTransformer(...args).transform;\n\nclass DepacketizeTransformer {\n  private buffering: RtpPacket[] = [];\n\n  transform: TransformStream<DepacketizerInput, DepacketizerOutput>;\n  packetLostHappened = false;\n\n  constructor(\n    private isFinalPacketInSequence: (header: RtpHeader) => boolean,\n    codec: string,\n    options: { waitForKeyframe?: boolean } = {}\n  ) {\n    this.transform = new TransformStream({\n      transform: (input, output) => {\n        if (!input.rtp) {\n          if (input.eol) {\n            output.enqueue({ eol: true });\n          }\n          return;\n        }\n\n        const packets = this.onRtp(input.rtp);\n        if (packets) {\n          try {\n            const { data, isKeyframe } = dePacketizeRtpPackets(codec, packets);\n\n            if (this.packetLostHappened) {\n              if (isKeyframe) {\n                this.packetLostHappened = false;\n              } else if (options.waitForKeyframe) {\n                return;\n              }\n            }\n\n            output.enqueue({\n              frame: {\n                data,\n                isKeyframe,\n                timestamp: packets[0].header.timestamp,\n              },\n            });\n          } catch (error) {}\n        }\n      },\n    });\n  }\n\n  lastSeqNum?: number;\n  private onRtp(rtp: RtpPacket) {\n    const { sequenceNumber } = rtp.header;\n    if (this.lastSeqNum != undefined) {\n      const expect = uint16Add(this.lastSeqNum, 1);\n      if (sequenceNumber < expect) {\n        return;\n      }\n      if (sequenceNumber > expect) {\n        log(\"packet lost happened\", { expect, sequenceNumber });\n        this.packetLostHappened = true;\n        this.buffering = [];\n      }\n    }\n\n    this.buffering.push(rtp);\n    this.lastSeqNum = sequenceNumber;\n\n    let finalPacket: number | undefined;\n    for (const [i, p] of enumerate(this.buffering)) {\n      if (this.isFinalPacketInSequence(p.header)) {\n        finalPacket = i;\n        break;\n      }\n    }\n    if (finalPacket == undefined) {\n      return;\n    }\n\n    const packets = this.buffering;\n    this.buffering = [];\n\n    return packets;\n  }\n}\n"]}

package/lib/rtp/src/processor_v2/index.d.ts

@@ -1,4 +0,0 @@
-export * from "./depacketizer";
-export * from "./jitterBuffer";
-export * from "./source";
-export * from "./webmLive";

package/lib/rtp/src/processor_v2/index.js.map

@@ -1 +0,0 @@
-{"version":3,"file":"index.js","sourceRoot":"","sources":["../../../../../rtp/src/processor_v2/index.ts"],"names":[],"mappings":";;;;;;;;;;;;;;;;AAAA,iDAA+B;AAC/B,iDAA+B;AAC/B,2CAAyB;AACzB,6CAA2B","sourcesContent":["export * from \"./depacketizer\";\nexport * from \"./jitterBuffer\";\nexport * from \"./source\";\nexport * from \"./webmLive\";\n"]}

package/lib/rtp/src/processor_v2/jitterBuffer.d.ts

@@ -1,33 +0,0 @@
-/// <reference types="node" />
-import { TransformStream } from "stream/web";
-import { RtpPacket } from "..";
-import { RtpOutput } from "./source/rtp";
-export declare type JitterBufferInput = RtpOutput;
-export interface JitterBufferOutput extends RtpOutput {
-    isPacketLost?: {
-        from: number;
-        to: number;
-    };
-}
-export declare const jitterBufferTransformer: (clockRate: number, options?: Partial<JitterBufferOptions> | undefined) => TransformStream<RtpOutput, JitterBufferOutput>;
-export declare class JitterBufferTransformer {
-    clockRate: number;
-    transform: TransformStream<JitterBufferInput, JitterBufferOutput>;
-    options: JitterBufferOptions;
-    presentSeqNum?: number;
-    rtpBuffer: {
-        [sequenceNumber: number]: RtpPacket;
-    };
-    private get expectNextSeqNum();
-    constructor(clockRate: number, options?: Partial<JitterBufferOptions>);
-    private processRtp;
-    private pushRtpBuffer;
-    private resolveBuffer;
-    private sortAndClearBuffer;
-    private disposeTimeoutPackets;
-}
-export interface JitterBufferOptions {
-    /**milliseconds */
-    latency: number;
-    bufferSize: number;
-}

package/lib/rtp/src/processor_v2/jitterBuffer.js

@@ -1,154 +0,0 @@
-"use strict";
-var __importDefault = (this && this.__importDefault) || function (mod) {
-    return (mod && mod.__esModule) ? mod : { "default": mod };
-};
-Object.defineProperty(exports, "__esModule", { value: true });
-exports.JitterBufferTransformer = exports.jitterBufferTransformer = void 0;
-const debug_1 = __importDefault(require("debug"));
-const web_1 = require("stream/web");
-const __1 = require("..");
-const srcPath = `werift-rtp : packages/rtp/src/processor_v2/jitterBuffer.ts`;
-const log = (0, debug_1.default)(srcPath);
-const jitterBufferTransformer = (...args) => new JitterBufferTransformer(...args).transform;
-exports.jitterBufferTransformer = jitterBufferTransformer;
-class JitterBufferTransformer {
-    constructor(clockRate, options = {}) {
-        this.clockRate = clockRate;
-        this.rtpBuffer = {};
-        this.options = {
-            latency: options.latency ?? 200,
-            bufferSize: options.bufferSize ?? 10000,
-        };
-        this.transform = new web_1.TransformStream({
-            transform: (input, output) => {
-                if (!input.rtp) {
-                    if (input.eol) {
-                        const packets = this.sortAndClearBuffer(this.rtpBuffer);
-                        packets.forEach((rtp) => output.enqueue({ rtp }));
-                        output.enqueue({ eol: true });
-                    }
-                    return;
-                }
-                const { packets, timeoutSeqNum } = this.processRtp(input.rtp);
-                if (timeoutSeqNum != undefined) {
-                    const isPacketLost = {
-                        from: this.expectNextSeqNum,
-                        to: timeoutSeqNum,
-                    };
-                    this.presentSeqNum = input.rtp.header.sequenceNumber;
-                    output.enqueue({ isPacketLost });
-                    if (packets) {
-                        [...packets, input.rtp].forEach((rtp) => output.enqueue({ rtp }));
-                    }
-                }
-                else {
-                    if (packets) {
-                        packets.forEach((rtp) => output.enqueue({ rtp }));
-                    }
-                }
-            },
-        });
-    }
-    get expectNextSeqNum() {
-        return (0, __1.uint16Add)(this.presentSeqNum, 1);
-    }
-    processRtp(rtp) {
-        const { sequenceNumber, timestamp } = rtp.header;
-        if (this.presentSeqNum == undefined) {
-            this.presentSeqNum = sequenceNumber;
-            return { packets: [rtp] };
-        }
-        if (sequenceNumber <= this.presentSeqNum) {
-            return { nothing: undefined };
-        }
-        if (sequenceNumber === this.expectNextSeqNum) {
-            this.presentSeqNum = sequenceNumber;
-            const rtpBuffer = this.resolveBuffer((0, __1.uint16Add)(sequenceNumber, 1));
-            this.presentSeqNum =
-                rtpBuffer.at(-1)?.header.sequenceNumber ?? this.presentSeqNum;
-            return { packets: [rtp, ...rtpBuffer] };
-        }
-        this.pushRtpBuffer(rtp);
-        const { latestTimeoutSeqNum, sorted } = this.disposeTimeoutPackets(timestamp);
-        if (latestTimeoutSeqNum) {
-            return { timeoutSeqNum: latestTimeoutSeqNum, packets: sorted };
-        }
-        else {
-            return { nothing: undefined };
-        }
-    }
-    pushRtpBuffer(rtp) {
-        if (Object.values(this.rtpBuffer).length > this.options.bufferSize) {
-            return;
-        }
-        this.rtpBuffer[rtp.header.sequenceNumber] = rtp;
-    }
-    resolveBuffer(seqNumFrom) {
-        const resolve = [];
-        for (let index = seqNumFrom;; index = (0, __1.uint16Add)(index, 1)) {
-            const rtp = this.rtpBuffer[index];
-            if (rtp) {
-                resolve.push(rtp);
-                delete this.rtpBuffer[index];
-            }
-            else {
-                break;
-            }
-        }
-        return resolve;
-    }
-    sortAndClearBuffer(rtpBuffer) {
-        const buffer = [];
-        for (let index = this.presentSeqNum ?? 0;; index = (0, __1.uint16Add)(index, 1)) {
-            const rtp = rtpBuffer[index];
-            if (rtp) {
-                buffer.push(rtp);
-                delete rtpBuffer[index];
-            }
-            if (Object.values(rtpBuffer).length === 0) {
-                break;
-            }
-        }
-        return buffer;
-    }
-    disposeTimeoutPackets(baseTimestamp) {
-        let latestTimeoutSeqNum;
-        const packets = Object.values(this.rtpBuffer)
-            .map((rtp) => {
-            const { timestamp, sequenceNumber } = rtp.header;
-            if ((0, __1.uint32Gt)(timestamp, baseTimestamp)) {
-                log("gap", { timestamp, baseTimestamp });
-                return;
-            }
-            const elapsedSec = (0, __1.uint32Add)(baseTimestamp, -timestamp) / this.clockRate;
-            if (elapsedSec * 1000 > this.options.latency) {
-                log("timeout packet", {
-                    sequenceNumber,
-                    elapsedSec,
-                    baseTimestamp,
-                    timestamp,
-                });
-                if (latestTimeoutSeqNum == undefined) {
-                    latestTimeoutSeqNum = sequenceNumber;
-                }
-                // 現在のSeqNumとの差が最も大きいSeqNumを探す
-                if ((0, __1.uint16Add)(sequenceNumber, -this.presentSeqNum) >
-                    (0, __1.uint16Add)(latestTimeoutSeqNum, -this.presentSeqNum)) {
-                    latestTimeoutSeqNum = sequenceNumber;
-                }
-                const packet = this.rtpBuffer[sequenceNumber];
-                delete this.rtpBuffer[sequenceNumber];
-                return packet;
-            }
-        })
-            .flatMap((p) => p)
-            .filter((p) => p);
-        const sorted = this.sortAndClearBuffer(packets.reduce((acc, cur) => {
-            acc[cur.header.sequenceNumber] = cur;
-            return acc;
-        }, {}));
-        return { latestTimeoutSeqNum, sorted };
-    }
-}
-exports.JitterBufferTransformer = JitterBufferTransformer;
-//# sourceMappingURL=jitterBuffer.js.map

package/lib/rtp/src/processor_v2/jitterBuffer.js.map

@@ -1 +0,0 @@
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debug from \"debug\";\nimport { TransformStream } from \"stream/web\";\n\nimport {\n  RequireAtLeastOne,\n  RtpPacket,\n  uint16Add,\n  uint32Add,\n  uint32Gt,\n} from \"..\";\nimport { RtpOutput } from \"./source/rtp\";\n\nconst srcPath = `werift-rtp : packages/rtp/src/processor_v2/jitterBuffer.ts`;\nconst log = debug(srcPath);\n\nexport type JitterBufferInput = RtpOutput;\n\nexport interface JitterBufferOutput extends RtpOutput {\n  isPacketLost?: { from: number; to: number };\n}\n\nexport const jitterBufferTransformer = (\n  ...args: ConstructorParameters<typeof JitterBufferTransformer>\n) => new JitterBufferTransformer(...args).transform;\n\nexport class JitterBufferTransformer {\n  transform: TransformStream<JitterBufferInput, JitterBufferOutput>;\n  options: JitterBufferOptions;\n  presentSeqNum?: number;\n  rtpBuffer: { [sequenceNumber: number]: RtpPacket } = {};\n\n  private get expectNextSeqNum() {\n    return uint16Add(this.presentSeqNum!, 1);\n  }\n\n  constructor(\n    public clockRate: number,\n    options: Partial<JitterBufferOptions> = {}\n  ) {\n    this.options = {\n      latency: options.latency ?? 200,\n      bufferSize: options.bufferSize ?? 10000,\n    };\n\n    this.transform = new TransformStream({\n      transform: (input, output) => {\n        if (!input.rtp) {\n          if (input.eol) {\n            const packets = this.sortAndClearBuffer(this.rtpBuffer);\n            packets.forEach((rtp) => output.enqueue({ rtp }));\n            output.enqueue({ eol: true });\n          }\n          return;\n        }\n\n        const { packets, timeoutSeqNum } = this.processRtp(input.rtp);\n\n        if (timeoutSeqNum != undefined) {\n          const isPacketLost = {\n            from: this.expectNextSeqNum,\n            to: timeoutSeqNum,\n          };\n          this.presentSeqNum = input.rtp.header.sequenceNumber;\n          output.enqueue({ isPacketLost });\n          if (packets) {\n            [...packets, input.rtp].forEach((rtp) => output.enqueue({ rtp }));\n          }\n        } else {\n          if (packets) {\n            packets.forEach((rtp) => output.enqueue({ rtp }));\n          }\n        }\n      },\n    });\n  }\n\n  private processRtp(rtp: RtpPacket): RequireAtLeastOne<{\n    packets: RtpPacket[];\n    timeoutSeqNum: number;\n    nothing: undefined;\n  }> {\n    const { sequenceNumber, timestamp } = rtp.header;\n\n    if (this.presentSeqNum == undefined) {\n      this.presentSeqNum = sequenceNumber;\n      return { packets: [rtp] };\n    }\n\n    if (sequenceNumber <= this.presentSeqNum) {\n      return { nothing: undefined };\n    }\n\n    if (sequenceNumber === this.expectNextSeqNum) {\n      this.presentSeqNum = sequenceNumber;\n\n      const rtpBuffer = this.resolveBuffer(uint16Add(sequenceNumber, 1));\n      this.presentSeqNum =\n        rtpBuffer.at(-1)?.header.sequenceNumber ?? this.presentSeqNum;\n\n      return { packets: [rtp, ...rtpBuffer] };\n    }\n\n    this.pushRtpBuffer(rtp);\n\n    const { latestTimeoutSeqNum, sorted } =\n      this.disposeTimeoutPackets(timestamp);\n\n    if (latestTimeoutSeqNum) {\n      return { timeoutSeqNum: latestTimeoutSeqNum, packets: sorted };\n    } else {\n      return { nothing: undefined };\n    }\n  }\n\n  private pushRtpBuffer(rtp: RtpPacket) {\n    if (Object.values(this.rtpBuffer).length > this.options.bufferSize) {\n      return;\n    }\n\n    this.rtpBuffer[rtp.header.sequenceNumber] = rtp;\n  }\n\n  private resolveBuffer(seqNumFrom: number) {\n    const resolve: RtpPacket[] = [];\n\n    for (let index = seqNumFrom; ; index = uint16Add(index, 1)) {\n      const rtp = this.rtpBuffer[index];\n      if (rtp) {\n        resolve.push(rtp);\n        delete this.rtpBuffer[index];\n      } else {\n        break;\n      }\n    }\n    return resolve;\n  }\n\n  private sortAndClearBuffer(rtpBuffer: {\n    [sequenceNumber: number]: RtpPacket;\n  }) {\n    const buffer: RtpPacket[] = [];\n    for (let index = this.presentSeqNum ?? 0; ; index = uint16Add(index, 1)) {\n      const rtp = rtpBuffer[index];\n      if (rtp) {\n        buffer.push(rtp);\n        delete rtpBuffer[index];\n      }\n      if (Object.values(rtpBuffer).length === 0) {\n        break;\n      }\n    }\n    return buffer;\n  }\n\n  private disposeTimeoutPackets(baseTimestamp: number) {\n    let latestTimeoutSeqNum: number | undefined;\n\n    const packets = Object.values(this.rtpBuffer)\n      .map((rtp) => {\n        const { timestamp, sequenceNumber } = rtp.header;\n\n        if (uint32Gt(timestamp, baseTimestamp)) {\n          log(\"gap\", { timestamp, baseTimestamp });\n          return;\n        }\n\n        const elapsedSec =\n          uint32Add(baseTimestamp, -timestamp) / this.clockRate;\n\n        if (elapsedSec * 1000 > this.options.latency) {\n          log(\"timeout packet\", {\n            sequenceNumber,\n            elapsedSec,\n            baseTimestamp,\n            timestamp,\n          });\n\n          if (latestTimeoutSeqNum == undefined) {\n            latestTimeoutSeqNum = sequenceNumber;\n          }\n          // 現在のSeqNumとの差が最も大きいSeqNumを探す\n          if (\n            uint16Add(sequenceNumber, -this.presentSeqNum!) >\n            uint16Add(latestTimeoutSeqNum, -this.presentSeqNum!)\n          ) {\n            latestTimeoutSeqNum = sequenceNumber;\n          }\n\n          const packet = this.rtpBuffer[sequenceNumber];\n          delete this.rtpBuffer[sequenceNumber];\n          return packet;\n        }\n      })\n      .flatMap((p): RtpPacket => p as RtpPacket)\n      .filter((p) => p);\n\n    const sorted = this.sortAndClearBuffer(\n      packets.reduce((acc, cur) => {\n        acc[cur.header.sequenceNumber] = cur;\n        return acc;\n      }, {})\n    );\n\n    return { latestTimeoutSeqNum, sorted };\n  }\n}\n\nexport interface JitterBufferOptions {\n  /**milliseconds */\n  latency: number;\n  bufferSize: number;\n}\n"]}

package/lib/rtp/src/processor_v2/source/base.d.ts

@@ -1,8 +0,0 @@
-/// <reference types="node" />
-import { ReadableStream, ReadableStreamController } from "stream/web";
-export declare class SourceStream<T> {
-    readable: ReadableStream<T>;
-    write: (chunk: T) => void;
-    protected controller: ReadableStreamController<T>;
-    constructor();
-}

package/lib/rtp/src/processor_v2/source/base.js

@@ -1,16 +0,0 @@
-"use strict";
-Object.defineProperty(exports, "__esModule", { value: true });
-exports.SourceStream = void 0;
-const web_1 = require("stream/web");
-class SourceStream {
-    constructor() {
-        this.readable = new web_1.ReadableStream({
-            start: (controller) => {
-                this.controller = controller;
-                this.write = (chunk) => controller.enqueue(chunk);
-            },
-        });
-    }
-}
-exports.SourceStream = SourceStream;
-//# sourceMappingURL=base.js.map

package/lib/rtp/src/processor_v2/source/base.js.map

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package/lib/rtp/src/processor_v2/source/index.js.map

@@ -1 +0,0 @@
-{"version":3,"file":"index.js","sourceRoot":"","sources":["../../../../../../rtp/src/processor_v2/source/index.ts"],"names":[],"mappings":";;;AAAA,+BAAwC;AAE/B,gGAFA,qBAAe,OAEA","sourcesContent":["import { RtpSourceStream } from \"./rtp\";\n\nexport { RtpSourceStream };\n"]}