piixishstream 0.0.0-stage → 1.0.2

package/dist/client/voice/CodecPayloadType.js

@@ -0,0 +1,60 @@
+export const CodecPayloadType = {
+    opus: {
+        name: "opus",
+        type: "audio",
+        clockRate: 48000,
+        priority: 1000,
+        payload_type: 120,
+    },
+    H264: {
+        name: "H264",
+        type: "video",
+        clockRate: 90000,
+        priority: 1000,
+        payload_type: 101,
+        rtx_payload_type: 102,
+        encode: true,
+        decode: true,
+    },
+    H265: {
+        name: "H265",
+        type: "video",
+        clockRate: 90000,
+        priority: 1000,
+        payload_type: 103,
+        rtx_payload_type: 104,
+        encode: true,
+        decode: true,
+    },
+    VP8: {
+        name: "VP8",
+        type: "video",
+        clockRate: 90000,
+        priority: 1000,
+        payload_type: 105,
+        rtx_payload_type: 106,
+        encode: true,
+        decode: true,
+    },
+    VP9: {
+        name: "VP9",
+        type: "video",
+        clockRate: 90000,
+        priority: 1000,
+        payload_type: 107,
+        rtx_payload_type: 108,
+        encode: true,
+        decode: true,
+    },
+    AV1: {
+        name: "AV1",
+        type: "video",
+        clockRate: 90000,
+        priority: 1000,
+        payload_type: 109,
+        rtx_payload_type: 110,
+        encode: true,
+        decode: true,
+    },
+};
+//# sourceMappingURL=CodecPayloadType.js.map

package/dist/client/voice/CodecPayloadType.js.map

@@ -0,0 +1 @@
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package/dist/client/voice/StreamConnection.d.ts

@@ -0,0 +1,11 @@
+import { BaseMediaConnection } from "./BaseMediaConnection.js";
+export declare class StreamConnection extends BaseMediaConnection {
+    private _streamKey;
+    private _serverId;
+    setSpeaking(speaking: boolean): void;
+    get daveChannelId(): string;
+    get serverId(): string | null;
+    set serverId(id: string);
+    get streamKey(): string | null;
+    set streamKey(value: string);
+}

package/dist/client/voice/StreamConnection.js

@@ -0,0 +1,34 @@
+import { VoiceOpCodes } from "../voice/VoiceOpCodes.js";
+import { BaseMediaConnection } from "./BaseMediaConnection.js";
+export class StreamConnection extends BaseMediaConnection {
+    _streamKey = null;
+    _serverId = null;
+    setSpeaking(speaking) {
+        if (!this.webRtcParams)
+            throw new Error("WebRTC connection not ready");
+        this.sendOpcode(VoiceOpCodes.SPEAKING, {
+            delay: 0,
+            speaking: speaking ? 2 : 0,
+            ssrc: this.webRtcParams.audioSsrc,
+        });
+    }
+    get daveChannelId() {
+        if (this._serverId === null)
+            throw new Error("Server ID not set (this shouldn't happen)");
+        const channelId = BigInt(this._serverId) - 1n;
+        return channelId.toString();
+    }
+    get serverId() {
+        return this._serverId;
+    }
+    set serverId(id) {
+        this._serverId = id;
+    }
+    get streamKey() {
+        return this._streamKey;
+    }
+    set streamKey(value) {
+        this._streamKey = value;
+    }
+}
+//# sourceMappingURL=StreamConnection.js.map

package/dist/client/voice/StreamConnection.js.map

@@ -0,0 +1 @@
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package/dist/client/voice/VoiceConnection.d.ts

@@ -0,0 +1,8 @@
+import { BaseMediaConnection } from "./BaseMediaConnection.js";
+import type { StreamConnection } from "./StreamConnection.js";
+export declare class VoiceConnection extends BaseMediaConnection {
+    streamConnection?: StreamConnection;
+    get daveChannelId(): string;
+    get serverId(): string;
+    stop(): void;
+}

package/dist/client/voice/VoiceConnection.js

@@ -0,0 +1,15 @@
+import { BaseMediaConnection } from "./BaseMediaConnection.js";
+export class VoiceConnection extends BaseMediaConnection {
+    streamConnection;
+    get daveChannelId() {
+        return this.channelId;
+    }
+    get serverId() {
+        return this.guildId ?? this.channelId; // for guild vc it is the guild id, for dm voice it is the channel id
+    }
+    stop() {
+        super.stop();
+        this.streamConnection?.stop();
+    }
+}
+//# sourceMappingURL=VoiceConnection.js.map

package/dist/client/voice/VoiceConnection.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"VoiceConnection.js","sourceRoot":"","sources":["../../../src/client/voice/VoiceConnection.ts"],"names":[],"mappings":"AAAA,OAAO,EAAE,mBAAmB,EAAE,MAAM,0BAA0B,CAAC;AAG/D,MAAM,OAAO,eAAgB,SAAQ,mBAAmB;IAC/C,gBAAgB,CAAoB;IAE3C,IAAoB,aAAa;QAC/B,OAAO,IAAI,CAAC,SAAS,CAAC;IACxB,CAAC;IAED,IAAoB,QAAQ;QAC1B,OAAO,IAAI,CAAC,OAAO,IAAI,IAAI,CAAC,SAAS,CAAC,CAAC,qEAAqE;IAC9G,CAAC;IAEe,IAAI;QAClB,KAAK,CAAC,IAAI,EAAE,CAAC;QACb,IAAI,CAAC,gBAAgB,EAAE,IAAI,EAAE,CAAC;IAChC,CAAC;CACF"}

package/dist/client/voice/VoiceMessageTypes.d.ts

@@ -0,0 +1,178 @@
+import type { VoiceOpCodes } from "./VoiceOpCodes.js";
+import type { SupportedEncryptionModes } from "../../utils.js";
+type StreamInfo = {
+    active: boolean;
+    quality: number;
+    rid: string;
+    ssrc: number;
+    rtx_ssrc: number;
+    /**
+     * always "video" from what I observed
+     */
+    type: string;
+};
+type SimulcastInfo = {
+    type: string;
+    rid: string;
+    quality: number;
+};
+type CodecPayloadType = {
+    name: string;
+    type: "audio";
+    priority: number;
+    payload_type: number;
+} | {
+    name: string;
+    type: "video";
+    priority: number;
+    payload_type: number;
+    rtx_payload_type: number;
+    encode: boolean;
+    decode: boolean;
+};
+export declare namespace Message {
+    type Identify = {
+        server_id: string;
+        user_id: string;
+        session_id: string;
+        token: string;
+        video: boolean;
+        streams: SimulcastInfo[];
+        max_dave_protocol_version?: number;
+    };
+    type Resume = {
+        server_id: string;
+        session_id: string;
+        token: string;
+        seq_ack: number;
+    };
+    type Heartbeat = {
+        t: number;
+        seq_ack?: number;
+    };
+    type SelectProtocol = {
+        protocol: "udp";
+        codecs: CodecPayloadType[];
+        data: {
+            address: string;
+            port: number;
+            mode: SupportedEncryptionModes;
+        };
+    } | {
+        protocol: "webrtc";
+        codecs: CodecPayloadType[];
+        data: string;
+        sdp: string;
+        rtc_connection_id: string;
+    };
+    type Video = {
+        audio_ssrc: number;
+        video_ssrc: number;
+        rtx_ssrc: number;
+        streams: {
+            type: "video";
+            rid: string;
+            ssrc: number;
+            active: boolean;
+            quality: number;
+            rtx_ssrc: number;
+            max_bitrate: number;
+            max_framerate: number;
+            max_resolution: {
+                type: "fixed";
+                width: number;
+                height: number;
+            };
+        }[];
+    };
+    type Hello = {
+        heartbeat_interval: number;
+    };
+    type Ready = {
+        ssrc: number;
+        ip: string;
+        port: number;
+        modes: SupportedEncryptionModes[];
+        experiments: string[];
+        streams: StreamInfo[];
+    };
+    type Speaking = {
+        speaking: 0 | 1 | 2;
+        delay: number;
+        ssrc: number;
+    };
+    type SelectProtocolAck = {
+        audio_codec: string;
+        video_codec: string;
+        dave_protocol_version: number;
+    } & ({
+        secret_key: number[];
+        mode: string;
+    } | {
+        media_session_id: number;
+        sdp: string;
+    });
+    type HeartbeatAck = {
+        t: number;
+    };
+    type ClientsConnect = {
+        user_ids: string[];
+    };
+    type ClientDisconnect = {
+        user_id: string;
+    };
+    type DavePrepareTransition = {
+        transition_id: number;
+        protocol_version: number;
+    };
+    type DaveExecuteTransition = {
+        transition_id: number;
+    };
+    type DaveTransitionReady = {
+        transition_id: number;
+    };
+    type DavePrepareEpoch = {
+        epoch: number;
+        protocol_version: number;
+    };
+    type MlsInvalidCommitWelcome = {
+        transition_id: number;
+    };
+}
+export declare namespace GatewayResponse {
+    type Generic<Op extends VoiceOpCodes, T extends Record<string, unknown> | null> = {
+        op: Op;
+        d: T;
+        seq?: number;
+    };
+    export type Hello = Generic<VoiceOpCodes.HELLO, Message.Hello>;
+    export type Ready = Generic<VoiceOpCodes.READY, Message.Ready>;
+    export type Resumed = Generic<VoiceOpCodes.RESUMED, null>;
+    export type Speaking = Generic<VoiceOpCodes.SPEAKING, Message.Speaking>;
+    export type SelectProtocolAck = Generic<VoiceOpCodes.SELECT_PROTOCOL_ACK, Message.SelectProtocolAck>;
+    export type HeartbeatAck = Generic<VoiceOpCodes.HEARTBEAT_ACK, Message.HeartbeatAck>;
+    export type ClientsConnect = Generic<VoiceOpCodes.CLIENTS_CONNECT, Message.ClientsConnect>;
+    export type ClientDisconnect = Generic<VoiceOpCodes.CLIENT_DISCONNECT, Message.ClientDisconnect>;
+    export type DavePrepareTransition = Generic<VoiceOpCodes.DAVE_PREPARE_TRANSITION, Message.DavePrepareTransition>;
+    export type DaveExecuteTransition = Generic<VoiceOpCodes.DAVE_EXECUTE_TRANSITION, Message.DaveExecuteTransition>;
+    export type DavePrepareEpoch = Generic<VoiceOpCodes.DAVE_PREPARE_EPOCH, Message.DavePrepareEpoch>;
+    export {};
+}
+export type GatewayResponse = GatewayResponse.Hello | GatewayResponse.Ready | GatewayResponse.Resumed | GatewayResponse.Speaking | GatewayResponse.SelectProtocolAck | GatewayResponse.HeartbeatAck | GatewayResponse.ClientsConnect | GatewayResponse.ClientDisconnect | GatewayResponse.DavePrepareTransition | GatewayResponse.DaveExecuteTransition | GatewayResponse.DavePrepareEpoch;
+export declare namespace GatewayRequest {
+    type Generic<Op extends VoiceOpCodes, T extends Record<string, unknown> | null> = {
+        op: Op;
+        d: T;
+    };
+    export type Identify = Generic<VoiceOpCodes.IDENTIFY, Message.Identify>;
+    export type Resume = Generic<VoiceOpCodes.RESUME, Message.Resume>;
+    export type Heartbeat = Generic<VoiceOpCodes.HEARTBEAT, Message.Heartbeat>;
+    export type SelectProtocol = Generic<VoiceOpCodes.SELECT_PROTOCOL, Message.SelectProtocol>;
+    export type Video = Generic<VoiceOpCodes.VIDEO, Message.Video>;
+    export type Speaking = Generic<VoiceOpCodes.SPEAKING, Message.Speaking>;
+    export type DaveTransitionReady = Generic<VoiceOpCodes.DAVE_TRANSITION_READY, Message.DaveTransitionReady>;
+    export type MlsInvalidCommitWelcome = Generic<VoiceOpCodes.MLS_INVALID_COMMIT_WELCOME, Message.MlsInvalidCommitWelcome>;
+    export {};
+}
+export type GatewayRequest = GatewayRequest.Identify | GatewayRequest.Resume | GatewayRequest.Heartbeat | GatewayRequest.SelectProtocol | GatewayRequest.Video | GatewayRequest.Speaking | GatewayRequest.DaveTransitionReady | GatewayRequest.MlsInvalidCommitWelcome;
+export {};

package/dist/client/voice/VoiceMessageTypes.js

@@ -0,0 +1,2 @@
+export {};
+//# sourceMappingURL=VoiceMessageTypes.js.map

package/dist/client/voice/VoiceMessageTypes.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"VoiceMessageTypes.js","sourceRoot":"","sources":["../../../src/client/voice/VoiceMessageTypes.ts"],"names":[],"mappings":""}

package/dist/client/voice/VoiceOpCodes.d.ts

@@ -0,0 +1,35 @@
+export declare enum VoiceOpCodes {
+    IDENTIFY = 0,
+    SELECT_PROTOCOL = 1,
+    READY = 2,
+    HEARTBEAT = 3,
+    SELECT_PROTOCOL_ACK = 4,
+    SPEAKING = 5,
+    HEARTBEAT_ACK = 6,
+    RESUME = 7,
+    HELLO = 8,
+    RESUMED = 9,
+    CLIENTS_CONNECT = 11,
+    VIDEO = 12,
+    CLIENT_DISCONNECT = 13,
+    SESSION_UPDATE = 14,
+    MEDIA_SINK_WANTS = 15,
+    VOICE_BACKEND_VERSION = 16,
+    CHANNEL_OPTIONS_UPDATE = 17,
+    FLAGS = 18,
+    SPEED_TEST = 19,
+    PLATFORM = 20,
+    DAVE_PREPARE_TRANSITION = 21,
+    DAVE_EXECUTE_TRANSITION = 22,
+    DAVE_TRANSITION_READY = 23,
+    DAVE_PREPARE_EPOCH = 24,
+    MLS_INVALID_COMMIT_WELCOME = 31
+}
+export declare enum VoiceOpCodesBinary {
+    MLS_EXTERNAL_SENDER = 25,
+    MLS_KEY_PACKAGE = 26,
+    MLS_PROPOSALS = 27,
+    MLS_COMMIT_WELCOME = 28,
+    MLS_ANNOUNCE_COMMIT_TRANSITION = 29,
+    MLS_WELCOME = 30
+}

package/dist/client/voice/VoiceOpCodes.js

@@ -0,0 +1,38 @@
+export var VoiceOpCodes;
+(function (VoiceOpCodes) {
+    VoiceOpCodes[VoiceOpCodes["IDENTIFY"] = 0] = "IDENTIFY";
+    VoiceOpCodes[VoiceOpCodes["SELECT_PROTOCOL"] = 1] = "SELECT_PROTOCOL";
+    VoiceOpCodes[VoiceOpCodes["READY"] = 2] = "READY";
+    VoiceOpCodes[VoiceOpCodes["HEARTBEAT"] = 3] = "HEARTBEAT";
+    VoiceOpCodes[VoiceOpCodes["SELECT_PROTOCOL_ACK"] = 4] = "SELECT_PROTOCOL_ACK";
+    VoiceOpCodes[VoiceOpCodes["SPEAKING"] = 5] = "SPEAKING";
+    VoiceOpCodes[VoiceOpCodes["HEARTBEAT_ACK"] = 6] = "HEARTBEAT_ACK";
+    VoiceOpCodes[VoiceOpCodes["RESUME"] = 7] = "RESUME";
+    VoiceOpCodes[VoiceOpCodes["HELLO"] = 8] = "HELLO";
+    VoiceOpCodes[VoiceOpCodes["RESUMED"] = 9] = "RESUMED";
+    VoiceOpCodes[VoiceOpCodes["CLIENTS_CONNECT"] = 11] = "CLIENTS_CONNECT";
+    VoiceOpCodes[VoiceOpCodes["VIDEO"] = 12] = "VIDEO";
+    VoiceOpCodes[VoiceOpCodes["CLIENT_DISCONNECT"] = 13] = "CLIENT_DISCONNECT";
+    VoiceOpCodes[VoiceOpCodes["SESSION_UPDATE"] = 14] = "SESSION_UPDATE";
+    VoiceOpCodes[VoiceOpCodes["MEDIA_SINK_WANTS"] = 15] = "MEDIA_SINK_WANTS";
+    VoiceOpCodes[VoiceOpCodes["VOICE_BACKEND_VERSION"] = 16] = "VOICE_BACKEND_VERSION";
+    VoiceOpCodes[VoiceOpCodes["CHANNEL_OPTIONS_UPDATE"] = 17] = "CHANNEL_OPTIONS_UPDATE";
+    VoiceOpCodes[VoiceOpCodes["FLAGS"] = 18] = "FLAGS";
+    VoiceOpCodes[VoiceOpCodes["SPEED_TEST"] = 19] = "SPEED_TEST";
+    VoiceOpCodes[VoiceOpCodes["PLATFORM"] = 20] = "PLATFORM";
+    VoiceOpCodes[VoiceOpCodes["DAVE_PREPARE_TRANSITION"] = 21] = "DAVE_PREPARE_TRANSITION";
+    VoiceOpCodes[VoiceOpCodes["DAVE_EXECUTE_TRANSITION"] = 22] = "DAVE_EXECUTE_TRANSITION";
+    VoiceOpCodes[VoiceOpCodes["DAVE_TRANSITION_READY"] = 23] = "DAVE_TRANSITION_READY";
+    VoiceOpCodes[VoiceOpCodes["DAVE_PREPARE_EPOCH"] = 24] = "DAVE_PREPARE_EPOCH";
+    VoiceOpCodes[VoiceOpCodes["MLS_INVALID_COMMIT_WELCOME"] = 31] = "MLS_INVALID_COMMIT_WELCOME";
+})(VoiceOpCodes || (VoiceOpCodes = {}));
+export var VoiceOpCodesBinary;
+(function (VoiceOpCodesBinary) {
+    VoiceOpCodesBinary[VoiceOpCodesBinary["MLS_EXTERNAL_SENDER"] = 25] = "MLS_EXTERNAL_SENDER";
+    VoiceOpCodesBinary[VoiceOpCodesBinary["MLS_KEY_PACKAGE"] = 26] = "MLS_KEY_PACKAGE";
+    VoiceOpCodesBinary[VoiceOpCodesBinary["MLS_PROPOSALS"] = 27] = "MLS_PROPOSALS";
+    VoiceOpCodesBinary[VoiceOpCodesBinary["MLS_COMMIT_WELCOME"] = 28] = "MLS_COMMIT_WELCOME";
+    VoiceOpCodesBinary[VoiceOpCodesBinary["MLS_ANNOUNCE_COMMIT_TRANSITION"] = 29] = "MLS_ANNOUNCE_COMMIT_TRANSITION";
+    VoiceOpCodesBinary[VoiceOpCodesBinary["MLS_WELCOME"] = 30] = "MLS_WELCOME";
+})(VoiceOpCodesBinary || (VoiceOpCodesBinary = {}));
+//# sourceMappingURL=VoiceOpCodes.js.map

package/dist/client/voice/VoiceOpCodes.js.map

@@ -0,0 +1 @@
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package/dist/client/voice/WebRtcWrapper.d.ts

@@ -0,0 +1,23 @@
+import { PeerConnection } from "@lng2004/node-datachannel";
+import type { BaseMediaConnection } from "./BaseMediaConnection.js";
+export declare class WebRtcConnWrapper {
+    private _mediaConn;
+    private _webRtcConn?;
+    private _audioDef;
+    private _videoDef;
+    private _audioTrack?;
+    private _videoTrack?;
+    private _audioPacketizer?;
+    private _videoPacketizer?;
+    private _videoCodec?;
+    constructor(mediaConn: BaseMediaConnection);
+    initWebRtc(): PeerConnection;
+    private _setMediaHandler;
+    close(): void;
+    get webRtcConn(): PeerConnection | undefined;
+    get ready(): boolean;
+    get mediaConnection(): BaseMediaConnection;
+    sendAudioFrame(frame: Buffer, frametime: number): void;
+    sendVideoFrame(frame: Buffer, frametime: number): void;
+    setPacketizer(videoCodec: string): void;
+}

package/dist/client/voice/WebRtcWrapper.js

@@ -0,0 +1,159 @@
+import { PeerConnection, Audio, Video, PacingHandler, RtpPacketizer, H264RtpPacketizer, H265RtpPacketizer, AV1RtpPacketizer, RtpPacketizationConfig, RtcpNackResponder, RtcpSrReporter, } from "@lng2004/node-datachannel";
+import { CodecPayloadType } from "./CodecPayloadType.js";
+import { normalizeVideoCodec } from "../../utils.js";
+import { splitNalu, H264Helpers, H264NalUnitTypes, startCode3, } from "../processing/AnnexBHelper.js";
+import { rewriteSPSVUI } from "../processing/SPSVUIRewriter.js";
+export class WebRtcConnWrapper {
+    _mediaConn;
+    _webRtcConn;
+    _audioDef;
+    _videoDef;
+    _audioTrack;
+    _videoTrack;
+    _audioPacketizer;
+    _videoPacketizer;
+    _videoCodec;
+    constructor(mediaConn) {
+        this._mediaConn = mediaConn;
+        this._audioDef = new Audio("0", "SendRecv");
+        this._videoDef = new Video("1", "SendRecv");
+        this._audioDef.addOpusCodec(CodecPayloadType.opus.payload_type);
+        for (const { name, payload_type, rtx_payload_type, clockRate, } of Object.values(CodecPayloadType).filter((el) => el.type === "video")) {
+            switch (name) {
+                case "H264":
+                    this._videoDef.addH264Codec(payload_type);
+                    break;
+                case "H265":
+                    this._videoDef.addH265Codec(payload_type);
+                    break;
+                case "VP8":
+                    this._videoDef.addVP8Codec(payload_type);
+                    break;
+                case "VP9":
+                    this._videoDef.addVP9Codec(payload_type);
+                    break;
+                case "AV1":
+                    this._videoDef.addAV1Codec(payload_type);
+                    break;
+            }
+            this._videoDef.addRTXCodec(rtx_payload_type, payload_type, clockRate);
+        }
+    }
+    initWebRtc() {
+        this._webRtcConn = new PeerConnection("", {
+            iceServers: ["stun:stun.l.google.com:19302"],
+        });
+        this._audioTrack = this._webRtcConn.addTrack(this._audioDef);
+        this._videoTrack = this._webRtcConn.addTrack(this._videoDef);
+        this._setMediaHandler();
+        return this._webRtcConn;
+    }
+    _setMediaHandler() {
+        if (this._audioPacketizer)
+            this._audioTrack?.setMediaHandler(this._audioPacketizer);
+        if (this._videoPacketizer)
+            this._videoTrack?.setMediaHandler(this._videoPacketizer);
+    }
+    close() {
+        this._webRtcConn?.close();
+    }
+    get webRtcConn() {
+        return this._webRtcConn;
+    }
+    get ready() {
+        return this._webRtcConn?.state() === "connected";
+    }
+    get mediaConnection() {
+        return this._mediaConn;
+    }
+    sendAudioFrame(frame, frametime) {
+        if (!this.ready)
+            return;
+        if (!this._audioPacketizer)
+            return;
+        const { rtpConfig } = this._audioPacketizer;
+        const { clockRate } = rtpConfig;
+        if (this.mediaConnection.daveReady)
+            frame = this.mediaConnection.daveSession.encryptOpus(frame);
+        this._audioTrack?.sendMessageBinary(frame);
+        rtpConfig.timestamp += Math.round((frametime * clockRate) / 1000);
+    }
+    sendVideoFrame(frame, frametime) {
+        if (!this.ready)
+            return;
+        if (!this._videoPacketizer)
+            return;
+        const { rtpConfig } = this._videoPacketizer;
+        const { clockRate } = rtpConfig;
+        if (this._videoCodec === "H264") {
+            let spsRewritten = false;
+            const nalus = splitNalu(frame).map((el) => {
+                if (H264Helpers.getUnitType(el) === H264NalUnitTypes.SPS) {
+                    spsRewritten = true;
+                    return rewriteSPSVUI(el);
+                }
+                return el;
+            });
+            if (spsRewritten)
+                frame = Buffer.concat(nalus.flatMap((el) => [startCode3, el]));
+        }
+        if (this.mediaConnection.daveReady) {
+            let daveCodec = 0 /* Codec.UNKNOWN */;
+            switch (this._videoCodec) {
+                case "H264":
+                    daveCodec = 4 /* Codec.H264 */;
+                    break;
+                case "H265":
+                    daveCodec = 5 /* Codec.H265 */;
+                    break;
+                case "VP8":
+                    daveCodec = 2 /* Codec.VP8 */;
+                    break;
+                case "VP9":
+                    daveCodec = 3 /* Codec.VP9 */;
+                    break;
+                case "AV1":
+                    daveCodec = 6 /* Codec.AV1 */;
+                    break;
+            }
+            frame = this.mediaConnection.daveSession.encrypt(1 /* MediaType.VIDEO */, daveCodec, frame);
+        }
+        this._videoTrack?.sendMessageBinary(frame);
+        rtpConfig.timestamp += Math.round((frametime * clockRate) / 1000);
+    }
+    setPacketizer(videoCodec) {
+        if (!this.mediaConnection.webRtcParams)
+            throw new Error("WebRTC connection not ready");
+        const { audioSsrc, videoSsrc } = this.mediaConnection.webRtcParams;
+        const rtpConfigAudio = new RtpPacketizationConfig(audioSsrc, "", CodecPayloadType.opus.payload_type, CodecPayloadType.opus.clockRate);
+        rtpConfigAudio.playoutDelayId = 5;
+        rtpConfigAudio.playoutDelayMin = 0;
+        rtpConfigAudio.playoutDelayMax = 1;
+        this._audioPacketizer = new RtpPacketizer(rtpConfigAudio);
+        this._audioPacketizer.addToChain(new RtcpSrReporter(rtpConfigAudio));
+        this._audioPacketizer.addToChain(new RtcpNackResponder());
+        this._videoCodec = normalizeVideoCodec(videoCodec);
+        const rtpConfigVideo = new RtpPacketizationConfig(videoSsrc, "", CodecPayloadType[this._videoCodec].payload_type, CodecPayloadType[this._videoCodec].clockRate);
+        rtpConfigVideo.playoutDelayId = 5;
+        rtpConfigVideo.playoutDelayMin = 0;
+        rtpConfigVideo.playoutDelayMax = 10;
+        switch (this._videoCodec) {
+            case "H264":
+                this._videoPacketizer = new H264RtpPacketizer("StartSequence", rtpConfigVideo);
+                break;
+            case "H265":
+                this._videoPacketizer = new H265RtpPacketizer("StartSequence", rtpConfigVideo);
+                break;
+            case "AV1":
+                this._videoPacketizer = new AV1RtpPacketizer("Obu", rtpConfigVideo);
+                break;
+            default:
+                throw new Error(`Packetizer not implemented for ${this._videoCodec}`);
+        }
+        this._videoPacketizer.addToChain(new RtcpSrReporter(rtpConfigVideo));
+        this._videoPacketizer.addToChain(new RtcpNackResponder());
+        // this._videoPacketizer.addToChain(new PacingHandler(25 * 1000 * 1000, 1));
+        this._setMediaHandler();
+    }
+}
+//# sourceMappingURL=WebRtcWrapper.js.map

package/dist/client/voice/WebRtcWrapper.js.map

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package/dist/client/voice/index.d.ts

@@ -0,0 +1,4 @@
+export * from "./VoiceConnection.js";
+export * from "./VoiceOpCodes.js";
+export * from "./StreamConnection.js";
+export * from "./BaseMediaConnection.js";

package/dist/client/voice/index.js

@@ -0,0 +1,6 @@
+export * from "./VoiceConnection.js";
+export * from "./VoiceOpCodes.js";
+// export * from './MediaUdp.js';
+export * from "./StreamConnection.js";
+export * from "./BaseMediaConnection.js";
+//# sourceMappingURL=index.js.map

package/dist/client/voice/index.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"index.js","sourceRoot":"","sources":["../../../src/client/voice/index.ts"],"names":[],"mappings":"AAAA,cAAc,sBAAsB,CAAC;AACrC,cAAc,mBAAmB,CAAC;AAClC,iCAAiC;AACjC,cAAc,uBAAuB,CAAC;AACtC,cAAc,0BAA0B,CAAC"}

package/dist/index.d.ts

@@ -0,0 +1,3 @@
+export * from "./client/index.js";
+export * from "./media/index.js";
+export * as Utils from "./utils.js";

package/dist/index.js

@@ -0,0 +1,4 @@
+export * from "./client/index.js";
+export * from "./media/index.js";
+export * as Utils from "./utils.js";
+//# sourceMappingURL=index.js.map

package/dist/index.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"index.js","sourceRoot":"","sources":["../src/index.ts"],"names":[],"mappings":"AAAA,cAAc,mBAAmB,CAAC;AAClC,cAAc,kBAAkB,CAAC;AACjC,OAAO,KAAK,KAAK,MAAM,YAAY,CAAC"}

package/dist/media/AudioStream.d.ts

@@ -0,0 +1,7 @@
+import { BaseMediaStream } from "./BaseMediaStream.js";
+import type { WebRtcConnWrapper } from "../client/voice/WebRtcWrapper.js";
+export declare class AudioStream extends BaseMediaStream {
+    private _conn;
+    constructor(conn: WebRtcConnWrapper, noSleep?: boolean);
+    protected _sendFrame(frame: Buffer, frametime: number): Promise<void>;
+}

package/dist/media/AudioStream.js

@@ -0,0 +1,12 @@
+import { BaseMediaStream } from "./BaseMediaStream.js";
+export class AudioStream extends BaseMediaStream {
+    _conn;
+    constructor(conn, noSleep = false) {
+        super("audio", noSleep);
+        this._conn = conn;
+    }
+    async _sendFrame(frame, frametime) {
+        this._conn.sendAudioFrame(frame, frametime);
+    }
+}
+//# sourceMappingURL=AudioStream.js.map