werift 0.16.0 → 0.16.1

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

@@ -1 +1 @@
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{ BitWriter, getBit } from \"../../../common/src\";\n\nexport type Extension = { id: number; payload: Buffer };\n\nexport const ExtensionProfiles = {\n  OneByte: 0xbede, // 48862\n  TwoByte: 0x1000, // 4096\n} as const;\ntype ExtensionProfile =\n  typeof ExtensionProfiles[keyof typeof ExtensionProfiles];\n\nconst seqNumOffset = 2;\nconst timestampOffset = 4;\nconst ssrcOffset = 8;\nconst csrcOffset = 12;\nconst csrcSize = 4;\n\n/*\n *  0                   1                   2                   3\n *  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1\n * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+\n * |V=2|P|X|  CC   |M|     PT      |       sequence number         |\n * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+\n * |                           timestamp                           |\n * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+\n * |           synchronization source (SSRC) identifier            |\n * +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+\n * |            contributing source (CSRC) identifiers             |\n * |                             ....                              |\n * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+\n */\n\nexport class RtpHeader {\n  version: number = 2;\n  padding: boolean = false;\n  paddingSize: number = 0;\n  extension: boolean = false;\n  marker: boolean = false;\n  payloadOffset: number = 0;\n  payloadType: number = 0;\n  /**16bit */\n  sequenceNumber: number = 0;\n  /**32bit milliseconds/1000 */\n  timestamp: number = 0;\n  ssrc: number = 0;\n  csrcLength = 0;\n  csrc: number[] = [];\n  extensionProfile: ExtensionProfile = ExtensionProfiles.OneByte;\n  /**deserialize only */\n  extensionLength?: number;\n  extensions: Extension[] = [];\n  constructor(props: Partial<RtpHeader> = {}) {\n    Object.assign(this, props);\n  }\n\n  static deSerialize(rawPacket: Buffer) {\n    const h = new RtpHeader();\n    let currOffset = 0;\n    const v_p_x_cc = rawPacket[currOffset++];\n    h.version = getBit(v_p_x_cc, 0, 2);\n    h.padding = getBit(v_p_x_cc, 2) > 0;\n    h.extension = getBit(v_p_x_cc, 3) > 0;\n    h.csrcLength = getBit(v_p_x_cc, 4, 4);\n    h.csrc = [...Array(h.csrcLength)].map(() => {\n      const csrc = rawPacket.readUInt32BE(currOffset);\n      currOffset += 4;\n      return csrc;\n    });\n    currOffset += csrcOffset - 1;\n\n    const m_pt = rawPacket[1];\n    h.marker = getBit(m_pt, 0) > 0;\n    h.payloadType = getBit(m_pt, 1, 7);\n\n    h.sequenceNumber = rawPacket.readUInt16BE(seqNumOffset);\n    h.timestamp = rawPacket.readUInt32BE(timestampOffset);\n    h.ssrc = rawPacket.readUInt32BE(ssrcOffset);\n\n    for (let i = 0; i < h.csrc.length; i++) {\n      const offset = csrcOffset + i * csrcSize;\n      h.csrc[i] = rawPacket.subarray(offset).readUInt32BE();\n    }\n    if (h.extension) {\n      h.extensionProfile = rawPacket\n        .subarray(currOffset)\n        .readUInt16BE() as ExtensionProfile;\n      currOffset += 2;\n      const extensionLength = rawPacket.subarray(currOffset).readUInt16BE() * 4;\n      h.extensionLength = extensionLength;\n      currOffset += 2;\n\n      switch (h.extensionProfile) {\n        // RFC 8285 RTP One Byte Header Extension\n        case ExtensionProfiles.OneByte:\n          {\n            const end = currOffset + extensionLength;\n            while (currOffset < end) {\n              if (rawPacket[currOffset] === 0x00) {\n                currOffset++;\n                continue;\n              }\n\n              const extId = rawPacket[currOffset] >> 4;\n              const len =\n                (rawPacket[currOffset] & (rawPacket[currOffset] ^ 0xf0)) + 1; // and not &^\n              currOffset++;\n              if (extId === 0xf) {\n                break;\n              }\n              const extension: Extension = {\n                id: extId,\n                payload: rawPacket.subarray(currOffset, currOffset + len),\n              };\n              h.extensions = [...h.extensions, extension];\n              currOffset += len;\n            }\n          }\n          break;\n        // RFC 8285 RTP Two Byte Header Extension\n        case ExtensionProfiles.TwoByte:\n          {\n            const end = currOffset + extensionLength;\n            while (currOffset < end) {\n              if (rawPacket[currOffset] === 0x00) {\n                currOffset++;\n                continue;\n              }\n              const extId = rawPacket[currOffset];\n              currOffset++;\n              const len = rawPacket[currOffset];\n              currOffset++;\n\n              const extension: Extension = {\n                id: extId,\n                payload: rawPacket.subarray(currOffset, currOffset + len),\n              };\n              h.extensions = [...h.extensions, extension];\n              currOffset += len;\n            }\n          }\n          break;\n        default:\n          {\n            const extension: Extension = {\n              id: 0,\n              payload: rawPacket.subarray(\n                currOffset,\n                currOffset + extensionLength\n              ),\n            };\n            h.extensions = [...h.extensions, extension];\n            currOffset += h.extensions[0].payload.length;\n          }\n          break;\n      }\n    }\n    h.payloadOffset = currOffset;\n    if (h.padding) {\n      h.paddingSize = rawPacket[rawPacket.length - 1];\n    }\n\n    return h;\n  }\n\n  get serializeSize() {\n    const { csrc, extensionProfile, extensions } = this;\n\n    let size = 12 + csrc.length * csrcSize;\n\n    if (extensions.length > 0 || this.extension === true) {\n      let extSize = 4;\n      switch (extensionProfile) {\n        case ExtensionProfiles.OneByte:\n          for (const extension of extensions) {\n            extSize += 1 + extension.payload.length;\n          }\n          break;\n        case ExtensionProfiles.TwoByte:\n          for (const extension of extensions) {\n            extSize += 2 + extension.payload.length;\n          }\n          break;\n        default:\n          extSize += extensions[0].payload.length;\n      }\n      size += Math.floor((extSize + 3) / 4) * 4;\n    }\n\n    return size;\n  }\n\n  serialize(size: number) {\n    const buf = Buffer.alloc(size);\n    let offset = 0;\n\n    const v_p_x_cc = new BitWriter(8);\n    v_p_x_cc.set(2, 0, this.version);\n    if (this.padding) v_p_x_cc.set(1, 2, 1);\n    if (this.extensions.length > 0) this.extension = true;\n    if (this.extension) v_p_x_cc.set(1, 3, 1);\n    v_p_x_cc.set(4, 4, this.csrc.length);\n    buf.writeUInt8(v_p_x_cc.value, offset++);\n\n    const m_pt = new BitWriter(8);\n    if (this.marker) m_pt.set(1, 0, 1);\n    m_pt.set(7, 1, this.payloadType);\n    buf.writeUInt8(m_pt.value, offset++);\n\n    buf.writeUInt16BE(this.sequenceNumber, seqNumOffset);\n    offset += 2;\n    buf.writeUInt32BE(this.timestamp, timestampOffset);\n    offset += 4;\n    buf.writeUInt32BE(this.ssrc, ssrcOffset);\n    offset += 4;\n\n    for (const csrc of this.csrc) {\n      buf.writeUInt32BE(csrc, offset);\n      offset += 4;\n    }\n\n    if (this.extension) {\n      const extHeaderPos = offset;\n      buf.writeUInt16BE(this.extensionProfile, offset);\n      offset += 4;\n      const startExtensionsPos = offset;\n\n      switch (this.extensionProfile) {\n        case ExtensionProfiles.OneByte:\n          for (const extension of this.extensions) {\n            buf.writeUInt8(\n              (extension.id << 4) | (extension.payload.length - 1),\n              offset++\n            );\n            extension.payload.copy(buf, offset);\n            offset += extension.payload.length;\n          }\n          break;\n        case ExtensionProfiles.TwoByte: // 1バイトで収まらなくなった歴史的経緯\n          for (const extension of this.extensions) {\n            buf.writeUInt8(extension.id, offset++);\n            buf.writeUInt8(extension.payload.length, offset++);\n            extension.payload.copy(buf, offset);\n            offset += extension.payload.length;\n          }\n          break;\n        default:\n          const extLen = this.extensions[0].payload.length;\n          if (extLen % 4 != 0) {\n            throw new Error();\n          }\n          this.extensions[0].payload.copy(buf, offset);\n          offset += extLen;\n      }\n\n      const extSize = offset - startExtensionsPos;\n      const roundedExtSize = Math.trunc((extSize + 3) / 4) * 4;\n\n      buf.writeUInt16BE(Math.trunc(roundedExtSize / 4), extHeaderPos + 2);\n      // padding 4 bytes boundaries\n      for (let i = 0; i < roundedExtSize - extSize; i++) {\n        buf.writeUInt8(0, offset);\n        offset++;\n      }\n    }\n    this.payloadOffset = offset;\n    return buf;\n  }\n}\n\nexport class RtpPacket {\n  constructor(public header: RtpHeader, public payload: Buffer) {}\n\n  get serializeSize() {\n    return this.header.serializeSize + this.payload.length;\n  }\n\n  clone() {\n    return new RtpPacket(new RtpHeader({ ...this.header }), this.payload);\n  }\n\n  serialize() {\n    let buf = this.header.serialize(\n      this.header.serializeSize + this.payload.length\n    );\n    const { payloadOffset } = this.header;\n    this.payload.copy(buf, payloadOffset);\n    if (this.header.padding) {\n      const padding = Buffer.alloc(this.header.paddingSize);\n      padding.writeUInt8(this.header.paddingSize, this.header.paddingSize - 1);\n      buf = Buffer.concat([buf, padding]);\n    }\n\n    return buf;\n  }\n\n  static deSerialize(buf: Buffer) {\n    const header = RtpHeader.deSerialize(buf);\n    const p = new RtpPacket(\n      header,\n      buf.subarray(header.payloadOffset, buf.length - header.paddingSize)\n    );\n    return p;\n  }\n}\n"]}
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{ BitWriter, getBit } from \"../../../common/src\";\n\nexport type Extension = { id: number; payload: Buffer };\n\nexport const ExtensionProfiles = {\n  OneByte: 0xbede, // 48862\n  TwoByte: 0x1000, // 4096\n} as const;\ntype ExtensionProfile =\n  typeof ExtensionProfiles[keyof typeof ExtensionProfiles];\n\nconst seqNumOffset = 2;\nconst timestampOffset = 4;\nconst ssrcOffset = 8;\nconst csrcOffset = 12;\nconst csrcSize = 4;\n\n/*\n *  0                   1                   2                   3\n *  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1\n * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+\n * |V=2|P|X|  CC   |M|     PT      |       sequence number         |\n * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+\n * |                           timestamp                           |\n * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+\n * |           synchronization source (SSRC) identifier            |\n * +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+\n * |            contributing source (CSRC) identifiers             |\n * |                             ....                              |\n * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+\n */\n\nexport class RtpHeader {\n  version: number = 2;\n  padding: boolean = false;\n  paddingSize: number = 0;\n  extension: boolean = false;\n  marker: boolean = false;\n  payloadOffset: number = 0;\n  payloadType: number = 0;\n  /**16bit */\n  sequenceNumber: number = 0;\n  /**32bit milliseconds/1000 */\n  timestamp: number = 0;\n  ssrc: number = 0;\n  csrcLength = 0;\n  csrc: number[] = [];\n  extensionProfile: ExtensionProfile = ExtensionProfiles.OneByte;\n  /**deserialize only */\n  extensionLength?: number;\n  extensions: Extension[] = [];\n  constructor(props: Partial<RtpHeader> = {}) {\n    Object.assign(this, props);\n  }\n\n  static deSerialize(rawPacket: Buffer) {\n    const h = new RtpHeader();\n    let currOffset = 0;\n    const v_p_x_cc = rawPacket[currOffset++];\n    h.version = getBit(v_p_x_cc, 0, 2);\n    h.padding = getBit(v_p_x_cc, 2) > 0;\n    h.extension = getBit(v_p_x_cc, 3) > 0;\n    h.csrcLength = getBit(v_p_x_cc, 4, 4);\n    h.csrc = [...Array(h.csrcLength)].map(() => {\n      const csrc = rawPacket.readUInt32BE(currOffset);\n      currOffset += 4;\n      return csrc;\n    });\n    currOffset += csrcOffset - 1;\n\n    const m_pt = rawPacket[1];\n    h.marker = getBit(m_pt, 0) > 0;\n    h.payloadType = getBit(m_pt, 1, 7);\n\n    h.sequenceNumber = rawPacket.readUInt16BE(seqNumOffset);\n    h.timestamp = rawPacket.readUInt32BE(timestampOffset);\n    h.ssrc = rawPacket.readUInt32BE(ssrcOffset);\n\n    for (let i = 0; i < h.csrc.length; i++) {\n      const offset = csrcOffset + i * csrcSize;\n      h.csrc[i] = rawPacket.subarray(offset).readUInt32BE();\n    }\n    if (h.extension) {\n      h.extensionProfile = rawPacket\n        .subarray(currOffset)\n        .readUInt16BE() as ExtensionProfile;\n      currOffset += 2;\n      const extensionLength = rawPacket.subarray(currOffset).readUInt16BE() * 4;\n      h.extensionLength = extensionLength;\n      currOffset += 2;\n\n      switch (h.extensionProfile) {\n        // RFC 8285 RTP One Byte Header Extension\n        case ExtensionProfiles.OneByte:\n          {\n            const end = currOffset + extensionLength;\n            while (currOffset < end) {\n              if (rawPacket[currOffset] === 0x00) {\n                currOffset++;\n                continue;\n              }\n\n              const extId = rawPacket[currOffset] >> 4;\n              const len =\n                (rawPacket[currOffset] & (rawPacket[currOffset] ^ 0xf0)) + 1; // and not &^\n              currOffset++;\n              if (extId === 0xf) {\n                break;\n              }\n              const extension: Extension = {\n                id: extId,\n                payload: rawPacket.subarray(currOffset, currOffset + len),\n              };\n              h.extensions = [...h.extensions, extension];\n              currOffset += len;\n            }\n          }\n          break;\n        // RFC 8285 RTP Two Byte Header Extension\n        case ExtensionProfiles.TwoByte:\n          {\n            const end = currOffset + extensionLength;\n            while (currOffset < end) {\n              if (rawPacket[currOffset] === 0x00) {\n                currOffset++;\n                continue;\n              }\n              const extId = rawPacket[currOffset];\n              currOffset++;\n              const len = rawPacket[currOffset];\n              currOffset++;\n\n              const extension: Extension = {\n                id: extId,\n                payload: rawPacket.subarray(currOffset, currOffset + len),\n              };\n              h.extensions = [...h.extensions, extension];\n              currOffset += len;\n            }\n          }\n          break;\n        default:\n          {\n            const extension: Extension = {\n              id: 0,\n              payload: rawPacket.subarray(\n                currOffset,\n                currOffset + extensionLength\n              ),\n            };\n            h.extensions = [...h.extensions, extension];\n            currOffset += h.extensions[0].payload.length;\n          }\n          break;\n      }\n    }\n    h.payloadOffset = currOffset;\n    if (h.padding) {\n      h.paddingSize = rawPacket[rawPacket.length - 1];\n    }\n\n    return h;\n  }\n\n  get serializeSize() {\n    const { csrc, extensionProfile, extensions } = this;\n\n    let size = 12 + csrc.length * csrcSize;\n\n    if (extensions.length > 0 || this.extension === true) {\n      let extSize = 4;\n      switch (extensionProfile) {\n        case ExtensionProfiles.OneByte:\n          for (const extension of extensions) {\n            extSize += 1 + extension.payload.length;\n          }\n          break;\n        case ExtensionProfiles.TwoByte:\n          for (const extension of extensions) {\n            extSize += 2 + extension.payload.length;\n          }\n          break;\n        default:\n          extSize += extensions[0].payload.length;\n      }\n      size += Math.floor((extSize + 3) / 4) * 4;\n    }\n\n    return size;\n  }\n\n  serialize(size: number) {\n    const buf = Buffer.alloc(size);\n    let offset = 0;\n\n    const v_p_x_cc = new BitWriter(8);\n    v_p_x_cc.set(2, 0, this.version);\n    if (this.padding) v_p_x_cc.set(1, 2, 1);\n    if (this.extensions.length > 0) this.extension = true;\n    if (this.extension) v_p_x_cc.set(1, 3, 1);\n    v_p_x_cc.set(4, 4, this.csrc.length);\n    buf.writeUInt8(v_p_x_cc.value, offset++);\n\n    const m_pt = new BitWriter(8);\n    if (this.marker) m_pt.set(1, 0, 1);\n    m_pt.set(7, 1, this.payloadType);\n    buf.writeUInt8(m_pt.value, offset++);\n\n    buf.writeUInt16BE(this.sequenceNumber, seqNumOffset);\n    offset += 2;\n    buf.writeUInt32BE(this.timestamp, timestampOffset);\n    offset += 4;\n    buf.writeUInt32BE(this.ssrc, ssrcOffset);\n    offset += 4;\n\n    for (const csrc of this.csrc) {\n      buf.writeUInt32BE(csrc, offset);\n      offset += 4;\n    }\n\n    if (this.extension) {\n      const extHeaderPos = offset;\n      buf.writeUInt16BE(this.extensionProfile, offset);\n      offset += 4;\n      const startExtensionsPos = offset;\n\n      switch (this.extensionProfile) {\n        case ExtensionProfiles.OneByte:\n          for (const extension of this.extensions) {\n            buf.writeUInt8(\n              (extension.id << 4) | (extension.payload.length - 1),\n              offset++\n            );\n            extension.payload.copy(buf, offset);\n            offset += extension.payload.length;\n          }\n          break;\n        case ExtensionProfiles.TwoByte: // 1バイトで収まらなくなった歴史的経緯\n          for (const extension of this.extensions) {\n            buf.writeUInt8(extension.id, offset++);\n            buf.writeUInt8(extension.payload.length, offset++);\n            extension.payload.copy(buf, offset);\n            offset += extension.payload.length;\n          }\n          break;\n        default:\n          const extLen = this.extensions[0].payload.length;\n          if (extLen % 4 != 0) {\n            throw new Error();\n          }\n          this.extensions[0].payload.copy(buf, offset);\n          offset += extLen;\n      }\n\n      const extSize = offset - startExtensionsPos;\n      const roundedExtSize = Math.trunc((extSize + 3) / 4) * 4;\n\n      buf.writeUInt16BE(Math.trunc(roundedExtSize / 4), extHeaderPos + 2);\n      // padding 4 bytes boundaries\n      for (let i = 0; i < roundedExtSize - extSize; i++) {\n        buf.writeUInt8(0, offset);\n        offset++;\n      }\n    }\n    this.payloadOffset = offset;\n    return buf;\n  }\n}\n\nexport class RtpPacket {\n  constructor(public header: RtpHeader, public payload: Buffer) {}\n\n  get serializeSize() {\n    return this.header.serializeSize + this.payload.length;\n  }\n\n  clone() {\n    return new RtpPacket(new RtpHeader({ ...this.header }), this.payload);\n  }\n\n  serialize() {\n    let buf = this.header.serialize(\n      this.header.serializeSize + this.payload.length\n    );\n    const { payloadOffset } = this.header;\n    this.payload.copy(buf, payloadOffset);\n    if (this.header.padding) {\n      const padding = Buffer.alloc(this.header.paddingSize);\n      padding.writeUInt8(this.header.paddingSize, this.header.paddingSize - 1);\n      buf = Buffer.concat([buf, padding]);\n    }\n\n    return buf;\n  }\n\n  static deSerialize(buf: Buffer) {\n    const header = RtpHeader.deSerialize(buf);\n    const p = new RtpPacket(\n      header,\n      buf.subarray(header.payloadOffset, buf.length - header.paddingSize)\n    );\n    return p;\n  }\n\n  clear() {\n    this.payload = null as any;\n  }\n}\n"]}

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

@@ -42,7 +42,6 @@ export declare class RTCRtpReceiver {
     latestRid?: string;
     latestRepairedRid?: string;
     receiverTWCC?: ReceiverTWCC;
-    supportTWCC: boolean;
     stopped: boolean;
     remoteStreamId?: string;
     remoteTrackId?: string;
@@ -53,11 +52,13 @@ export declare class RTCRtpReceiver {
     setDtlsTransport(dtls: RTCDtlsTransport): void;
     get track(): MediaStreamTrack;
     get nackEnabled(): import("./parameters").RTCPFB | undefined;
+    get twccEnabled(): import("./parameters").RTCPFB | undefined;
+    get pliEnabled(): import("./parameters").RTCPFB | undefined;
     prepareReceive(params: RTCRtpReceiveParameters): void;
     /**
      * setup TWCC if supported
      */
-    setupTWCC(mediaSourceSsrc?: number): void;
+    setupTWCC(mediaSourceSsrc: number): void;
     addTrack(track: MediaStreamTrack): boolean;
     stop(): void;
     runRtcp(): Promise<void>;

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

@@ -41,7 +41,6 @@ class RTCRtpReceiver {
         this.receiveLastSRTimestamp = {};
         this.onPacketLost = this.nack.onPacketLost;
         this.onRtcp = new rx_mini_1.default();
-        this.supportTWCC = false;
         this.stopped = false;
         this.rtcpRunning = false;
         this.rtcpCancel = new AbortController();
@@ -69,7 +68,13 @@ class RTCRtpReceiver {
         return this.tracks[0];
     }
     get nackEnabled() {
-        return this.codecArray[0].rtcpFeedback.find((f) => f.type === "nack");
+        return this.codecArray[0]?.rtcpFeedback.find((f) => f.type === "nack");
+    }
+    get twccEnabled() {
+        return this.codecArray[0]?.rtcpFeedback.find((f) => f.type === (0, __1.useTWCC)().type);
+    }
+    get pliEnabled() {
+        return this.codecArray[0]?.rtcpFeedback.find((f) => f.type === (0, __1.usePLI)().type);
     }
     prepareReceive(params) {
         params.codecs.forEach((c) => {
@@ -85,9 +90,7 @@ class RTCRtpReceiver {
      * setup TWCC if supported
      */
     setupTWCC(mediaSourceSsrc) {
-        this.supportTWCC = !!Object.values(this.codecs).find((codec) => codec.rtcpFeedback.find((v) => v.type === "transport-cc"));
-        log("twcc support", this.supportTWCC);
-        if (this.supportTWCC && mediaSourceSsrc) {
+        if (this.twccEnabled && !this.receiverTWCC) {
             this.receiverTWCC = new receiverTwcc_1.ReceiverTWCC(this.dtlsTransport, this.rtcpSsrc, mediaSourceSsrc);
         }
     }
@@ -161,6 +164,11 @@ class RTCRtpReceiver {
     /**todo impl */
     getStats() { }
     async sendRtcpPLI(mediaSsrc) {
+        if (!this.pliEnabled) {
+            log("pli not supported", { mediaSsrc });
+            return;
+        }
+        log("sendRtcpPLI", { mediaSsrc });
         const packet = new src_2.RtcpPayloadSpecificFeedback({
             feedback: new src_2.PictureLossIndication({
                 senderSsrc: this.rtcpSsrc,
@@ -209,7 +217,7 @@ class RTCRtpReceiver {
             }
             this.receiverTWCC.handleTWCC(transportSequenceNumber);
         }
-        else if (this.supportTWCC) {
+        else if (this.twccEnabled) {
             this.setupTWCC(packet.header.ssrc);
         }
         if (codec.name.toLowerCase() === "rtx") {

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

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{ debug } from \"debug\";\nimport { jspack } from \"jspack\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport { v4 as uuid } from \"uuid\";\n\nimport { int } from \"../../../common/src\";\nimport {\n  PictureLossIndication,\n  Red,\n  RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpReceiverInfo,\n  RtcpRrPacket,\n  RtcpSrPacket,\n  RtpHeader,\n  RtpPacket,\n} from \"../../../rtp/src\";\nimport { codecParametersFromString, PeerConfig } from \"..\";\nimport { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { compactNtp, timestampSeconds } from \"../utils\";\nimport { RTP_EXTENSION_URI } from \"./extension/rtpExtension\";\nimport { RTCRtpCodecParameters, RTCRtpReceiveParameters } from \"./parameters\";\nimport { NackHandler } from \"./receiver/nack\";\nimport { ReceiverTWCC } from \"./receiver/receiverTwcc\";\nimport { AudioRedHandler } from \"./receiver/red\";\nimport { StreamStatistics } from \"./receiver/statistics\";\nimport { Extensions } from \"./router\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/rtpReceiver.ts\");\n\nexport class RTCRtpReceiver {\n  private readonly codecs: { [pt: number]: RTCRtpCodecParameters } = {};\n  private get codecArray() {\n    return Object.values(this.codecs).sort(\n      (a, b) => a.payloadType - b.payloadType\n    );\n  }\n  private readonly ssrcByRtx: { [rtxSsrc: number]: number } = {};\n  private readonly nack = new NackHandler(this);\n  private readonly redHandler = new AudioRedHandler();\n\n  readonly type = \"receiver\";\n  readonly uuid = uuid();\n  readonly tracks: MediaStreamTrack[] = [];\n  readonly trackBySSRC: { [ssrc: string]: MediaStreamTrack } = {};\n  readonly trackByRID: { [rid: string]: MediaStreamTrack } = {};\n  /**last sender Report Timestamp\n   * compactNtp\n   */\n  readonly lastSRtimestamp: { [ssrc: number]: number } = {};\n  /**seconds */\n  readonly receiveLastSRTimestamp: { [ssrc: number]: number } = {};\n  readonly onPacketLost = this.nack.onPacketLost;\n  readonly onRtcp = new Event<[RtcpPacket]>();\n\n  dtlsTransport!: RTCDtlsTransport;\n  sdesMid?: string;\n  latestRid?: string;\n  latestRepairedRid?: string;\n\n  receiverTWCC?: ReceiverTWCC;\n  supportTWCC = false;\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  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    this.supportTWCC = !!Object.values(this.codecs).find((codec) =>\n      codec.rtcpFeedback.find((v) => v.type === \"transport-cc\")\n    );\n    log(\"twcc support\", this.supportTWCC);\n\n    if (this.supportTWCC && mediaSourceSsrc) {\n      this.receiverTWCC = new ReceiverTWCC(\n        this.dtlsTransport,\n        this.rtcpSsrc,\n        mediaSourceSsrc\n      );\n    }\n  }\n\n  addTrack(track: MediaStreamTrack) {\n    const exist = this.tracks.find((t) => {\n      if (t.rid) return t.rid === track.rid;\n      if (t.ssrc) return t.ssrc === track.ssrc;\n    });\n    if (exist) return false;\n    this.tracks.push(track);\n    if (track.ssrc) this.trackBySSRC[track.ssrc] = track;\n    if (track.rid) this.trackByRID[track.rid] = track;\n    return true;\n  }\n\n  stop() {\n    this.stopped = true;\n    this.rtcpRunning = false;\n    this.rtcpCancel.abort();\n\n    if (this.receiverTWCC) this.receiverTWCC.twccRunning = false;\n    this.nack.close();\n  }\n\n  async runRtcp() {\n    if (this.rtcpRunning || this.stopped) return;\n    this.rtcpRunning = true;\n\n    try {\n      while (this.rtcpRunning) {\n        await setTimeout(500 + Math.random() * 1000, undefined, {\n          signal: this.rtcpCancel.signal,\n        });\n\n        const reports = Object.entries(this.remoteStreams).map(\n          ([ssrc, stream]) => {\n            let lastSRtimestamp = 0,\n              delaySinceLastSR = 0;\n            if (this.lastSRtimestamp[ssrc]) {\n              lastSRtimestamp = this.lastSRtimestamp[ssrc];\n              const delaySeconds =\n                timestampSeconds() - this.receiveLastSRTimestamp[ssrc];\n              if (delaySeconds > 0 && delaySeconds < 65536) {\n                delaySinceLastSR = int(delaySeconds * 65536);\n              }\n            }\n\n            return new RtcpReceiverInfo({\n              ssrc: Number(ssrc),\n              fractionLost: stream.fraction_lost,\n              packetsLost: stream.packets_lost,\n              highestSequence: stream.max_seq,\n              jitter: stream.jitter,\n              lsr: lastSRtimestamp,\n              dlsr: delaySinceLastSR,\n            });\n          }\n        );\n\n        const packet = new RtcpRrPacket({ ssrc: this.rtcpSsrc, reports });\n\n        try {\n          if (this.config.debug.receiverReportDelay) {\n            await setTimeout(this.config.debug.receiverReportDelay);\n          }\n          await this.dtlsTransport.sendRtcp([packet]);\n        } catch (error) {\n          log(\"sendRtcp failed\", error);\n          await setTimeout(500 + Math.random() * 1000);\n        }\n      }\n    } catch (error) {}\n  }\n\n  /**todo impl */\n  getStats() {}\n\n  async sendRtcpPLI(mediaSsrc: number) {\n    const packet = new RtcpPayloadSpecificFeedback({\n      feedback: new PictureLossIndication({\n        senderSsrc: this.rtcpSsrc,\n        mediaSsrc,\n      }),\n    });\n    try {\n      await this.dtlsTransport.sendRtcp([packet]);\n    } catch (error) {\n      log(error);\n    }\n  }\n\n  handleRtcpPacket(packet: RtcpPacket) {\n    switch (packet.type) {\n      case RtcpSrPacket.type:\n        {\n          const sr = packet as RtcpSrPacket;\n          this.lastSRtimestamp[sr.ssrc] = compactNtp(\n            sr.senderInfo.ntpTimestamp\n          );\n          this.receiveLastSRTimestamp[sr.ssrc] = timestampSeconds();\n\n          const track = this.trackBySSRC[packet.ssrc];\n          if (track) {\n            track.onReceiveRtcp.execute(packet);\n          }\n        }\n        break;\n    }\n    this.onRtcp.execute(packet);\n  }\n\n  handleRtpBySsrc = (packet: RtpPacket, extensions: Extensions) => {\n    const track = this.trackBySSRC[packet.header.ssrc];\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  handleRtpByRid = (packet: RtpPacket, rid: string, extensions: Extensions) => {\n    const track = this.trackByRID[rid];\n    if (!this.trackBySSRC[packet.header.ssrc]) {\n      this.trackBySSRC[packet.header.ssrc] = track;\n    }\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  private handleRTP(\n    packet: RtpPacket,\n    extensions: Extensions,\n    track?: MediaStreamTrack\n  ) {\n    if (this.stopped) return;\n\n    const codec = this.codecs[packet.header.payloadType];\n    if (!codec) {\n      // log(\"unknown codec \" + packet.header.payloadType);\n      return;\n    }\n\n    this.remoteStreams[packet.header.ssrc] =\n      this.remoteStreams[packet.header.ssrc] ??\n      new StreamStatistics(codec.clockRate);\n    this.remoteStreams[packet.header.ssrc].add(packet);\n\n    if (this.receiverTWCC) {\n      const transportSequenceNumber = extensions[\n        RTP_EXTENSION_URI.transportWideCC\n      ] as number;\n      if (!transportSequenceNumber == undefined) {\n        throw new Error(\"undefined\");\n      }\n\n      this.receiverTWCC.handleTWCC(transportSequenceNumber);\n    } else if (this.supportTWCC) {\n      this.setupTWCC(packet.header.ssrc);\n    }\n\n    if (codec.name.toLowerCase() === \"rtx\") {\n      const originalSsrc = this.ssrcByRtx[packet.header.ssrc];\n      const codecParams = codecParametersFromString(codec.parameters ?? \"\");\n      const rtxCodec = this.codecs[codecParams[\"apt\"]];\n      if (packet.payload.length < 2) return;\n\n      packet = unwrapRtx(packet, rtxCodec.payloadType, originalSsrc);\n      track = this.trackBySSRC[originalSsrc];\n    }\n\n    let red: Red | undefined;\n    if (codec.name.toLowerCase() === \"red\") {\n      red = Red.deSerialize(packet.payload);\n      if (\n        !Object.keys(this.codecs).includes(\n          red.header.fields[0].blockPT.toString()\n        )\n      ) {\n        return;\n      }\n    }\n\n    if (track?.kind === \"video\" && this.nackEnabled) {\n      this.nack.addPacket(packet);\n    }\n\n    if (track) {\n      if (red) {\n        if (track.kind === \"audio\") {\n          const payloads = this.redHandler.push(red, packet);\n          for (const packet of payloads) {\n            track.onReceiveRtp.execute(packet.clone());\n          }\n        } else {\n        }\n      } else {\n        track.onReceiveRtp.execute(packet.clone());\n      }\n    }\n\n    this.runRtcp();\n  }\n}\n\nexport function unwrapRtx(rtx: RtpPacket, payloadType: number, ssrc: number) {\n  const packet = new RtpPacket(\n    new RtpHeader({\n      payloadType,\n      marker: rtx.header.marker,\n      sequenceNumber: jspack.Unpack(\"!H\", rtx.payload.subarray(0, 2))[0],\n      timestamp: rtx.header.timestamp,\n      ssrc,\n    }),\n    rtx.payload.subarray(2)\n  );\n  return packet;\n}\n"]}
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{ debug } from \"debug\";\nimport { jspack } from \"jspack\";\nimport Event from \"rx.mini\";\nimport { setTimeout } from \"timers/promises\";\nimport { v4 as uuid } from \"uuid\";\n\nimport { int } from \"../../../common/src\";\nimport {\n  PictureLossIndication,\n  Red,\n  RtcpPacket,\n  RtcpPayloadSpecificFeedback,\n  RtcpReceiverInfo,\n  RtcpRrPacket,\n  RtcpSrPacket,\n  RtpHeader,\n  RtpPacket,\n} from \"../../../rtp/src\";\nimport { codecParametersFromString, PeerConfig, usePLI, useTWCC } from \"..\";\nimport { RTCDtlsTransport } from \"../transport/dtls\";\nimport { Kind } from \"../types/domain\";\nimport { compactNtp, timestampSeconds } from \"../utils\";\nimport { RTP_EXTENSION_URI } from \"./extension/rtpExtension\";\nimport { RTCRtpCodecParameters, RTCRtpReceiveParameters } from \"./parameters\";\nimport { NackHandler } from \"./receiver/nack\";\nimport { ReceiverTWCC } from \"./receiver/receiverTwcc\";\nimport { AudioRedHandler } from \"./receiver/red\";\nimport { StreamStatistics } from \"./receiver/statistics\";\nimport { Extensions } from \"./router\";\nimport { MediaStreamTrack } from \"./track\";\n\nconst log = debug(\"werift:packages/webrtc/src/media/rtpReceiver.ts\");\n\nexport class RTCRtpReceiver {\n  private readonly codecs: { [pt: number]: RTCRtpCodecParameters } = {};\n  private get codecArray() {\n    return Object.values(this.codecs).sort(\n      (a, b) => a.payloadType - b.payloadType\n    );\n  }\n  private readonly ssrcByRtx: { [rtxSsrc: number]: number } = {};\n  private readonly nack = new NackHandler(this);\n  private readonly redHandler = new AudioRedHandler();\n\n  readonly type = \"receiver\";\n  readonly uuid = uuid();\n  readonly tracks: MediaStreamTrack[] = [];\n  readonly trackBySSRC: { [ssrc: string]: MediaStreamTrack } = {};\n  readonly trackByRID: { [rid: string]: MediaStreamTrack } = {};\n  /**last sender Report Timestamp\n   * compactNtp\n   */\n  readonly lastSRtimestamp: { [ssrc: number]: number } = {};\n  /**seconds */\n  readonly receiveLastSRTimestamp: { [ssrc: number]: number } = {};\n  readonly onPacketLost = this.nack.onPacketLost;\n  readonly onRtcp = new Event<[RtcpPacket]>();\n\n  dtlsTransport!: RTCDtlsTransport;\n  sdesMid?: string;\n  latestRid?: string;\n  latestRepairedRid?: string;\n\n  receiverTWCC?: ReceiverTWCC;\n  stopped = false;\n  remoteStreamId?: string;\n  remoteTrackId?: string;\n\n  rtcpRunning = false;\n  private rtcpCancel = new AbortController();\n  private remoteStreams: { [ssrc: number]: StreamStatistics } = {};\n\n  constructor(\n    readonly config: PeerConfig,\n    public kind: Kind,\n    public rtcpSsrc: number\n  ) {}\n\n  setDtlsTransport(dtls: RTCDtlsTransport) {\n    this.dtlsTransport = dtls;\n  }\n\n  // todo fix\n  get track() {\n    return this.tracks[0];\n  }\n\n  get nackEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find((f) => f.type === \"nack\");\n  }\n\n  get twccEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find(\n      (f) => f.type === useTWCC().type\n    );\n  }\n\n  get pliEnabled() {\n    return this.codecArray[0]?.rtcpFeedback.find(\n      (f) => f.type === usePLI().type\n    );\n  }\n\n  prepareReceive(params: RTCRtpReceiveParameters) {\n    params.codecs.forEach((c) => {\n      this.codecs[c.payloadType] = c;\n    });\n    params.encodings.forEach((e) => {\n      if (e.rtx) {\n        this.ssrcByRtx[e.rtx.ssrc] = e.ssrc;\n      }\n    });\n  }\n\n  /**\n   * setup TWCC if supported\n   */\n  setupTWCC(mediaSourceSsrc: number) {\n    if (this.twccEnabled && !this.receiverTWCC) {\n      this.receiverTWCC = new ReceiverTWCC(\n        this.dtlsTransport,\n        this.rtcpSsrc,\n        mediaSourceSsrc\n      );\n    }\n  }\n\n  addTrack(track: MediaStreamTrack) {\n    const exist = this.tracks.find((t) => {\n      if (t.rid) return t.rid === track.rid;\n      if (t.ssrc) return t.ssrc === track.ssrc;\n    });\n    if (exist) return false;\n    this.tracks.push(track);\n    if (track.ssrc) this.trackBySSRC[track.ssrc] = track;\n    if (track.rid) this.trackByRID[track.rid] = track;\n    return true;\n  }\n\n  stop() {\n    this.stopped = true;\n    this.rtcpRunning = false;\n    this.rtcpCancel.abort();\n\n    if (this.receiverTWCC) this.receiverTWCC.twccRunning = false;\n    this.nack.close();\n  }\n\n  async runRtcp() {\n    if (this.rtcpRunning || this.stopped) return;\n    this.rtcpRunning = true;\n\n    try {\n      while (this.rtcpRunning) {\n        await setTimeout(500 + Math.random() * 1000, undefined, {\n          signal: this.rtcpCancel.signal,\n        });\n\n        const reports = Object.entries(this.remoteStreams).map(\n          ([ssrc, stream]) => {\n            let lastSRtimestamp = 0,\n              delaySinceLastSR = 0;\n            if (this.lastSRtimestamp[ssrc]) {\n              lastSRtimestamp = this.lastSRtimestamp[ssrc];\n              const delaySeconds =\n                timestampSeconds() - this.receiveLastSRTimestamp[ssrc];\n              if (delaySeconds > 0 && delaySeconds < 65536) {\n                delaySinceLastSR = int(delaySeconds * 65536);\n              }\n            }\n\n            return new RtcpReceiverInfo({\n              ssrc: Number(ssrc),\n              fractionLost: stream.fraction_lost,\n              packetsLost: stream.packets_lost,\n              highestSequence: stream.max_seq,\n              jitter: stream.jitter,\n              lsr: lastSRtimestamp,\n              dlsr: delaySinceLastSR,\n            });\n          }\n        );\n\n        const packet = new RtcpRrPacket({ ssrc: this.rtcpSsrc, reports });\n\n        try {\n          if (this.config.debug.receiverReportDelay) {\n            await setTimeout(this.config.debug.receiverReportDelay);\n          }\n          await this.dtlsTransport.sendRtcp([packet]);\n        } catch (error) {\n          log(\"sendRtcp failed\", error);\n          await setTimeout(500 + Math.random() * 1000);\n        }\n      }\n    } catch (error) {}\n  }\n\n  /**todo impl */\n  getStats() {}\n\n  async sendRtcpPLI(mediaSsrc: number) {\n    if (!this.pliEnabled) {\n      log(\"pli not supported\", { mediaSsrc });\n      return;\n    }\n    log(\"sendRtcpPLI\", { mediaSsrc });\n\n    const packet = new RtcpPayloadSpecificFeedback({\n      feedback: new PictureLossIndication({\n        senderSsrc: this.rtcpSsrc,\n        mediaSsrc,\n      }),\n    });\n    try {\n      await this.dtlsTransport.sendRtcp([packet]);\n    } catch (error) {\n      log(error);\n    }\n  }\n\n  handleRtcpPacket(packet: RtcpPacket) {\n    switch (packet.type) {\n      case RtcpSrPacket.type:\n        {\n          const sr = packet as RtcpSrPacket;\n          this.lastSRtimestamp[sr.ssrc] = compactNtp(\n            sr.senderInfo.ntpTimestamp\n          );\n          this.receiveLastSRTimestamp[sr.ssrc] = timestampSeconds();\n\n          const track = this.trackBySSRC[packet.ssrc];\n          if (track) {\n            track.onReceiveRtcp.execute(packet);\n          }\n        }\n        break;\n    }\n    this.onRtcp.execute(packet);\n  }\n\n  handleRtpBySsrc = (packet: RtpPacket, extensions: Extensions) => {\n    const track = this.trackBySSRC[packet.header.ssrc];\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  handleRtpByRid = (packet: RtpPacket, rid: string, extensions: Extensions) => {\n    const track = this.trackByRID[rid];\n    if (!this.trackBySSRC[packet.header.ssrc]) {\n      this.trackBySSRC[packet.header.ssrc] = track;\n    }\n\n    this.handleRTP(packet, extensions, track);\n  };\n\n  private handleRTP(\n    packet: RtpPacket,\n    extensions: Extensions,\n    track?: MediaStreamTrack\n  ) {\n    if (this.stopped) return;\n\n    const codec = this.codecs[packet.header.payloadType];\n    if (!codec) {\n      // log(\"unknown codec \" + packet.header.payloadType);\n      return;\n    }\n\n    this.remoteStreams[packet.header.ssrc] =\n      this.remoteStreams[packet.header.ssrc] ??\n      new StreamStatistics(codec.clockRate);\n    this.remoteStreams[packet.header.ssrc].add(packet);\n\n    if (this.receiverTWCC) {\n      const transportSequenceNumber = extensions[\n        RTP_EXTENSION_URI.transportWideCC\n      ] as number;\n      if (!transportSequenceNumber == undefined) {\n        throw new Error(\"undefined\");\n      }\n\n      this.receiverTWCC.handleTWCC(transportSequenceNumber);\n    } else if (this.twccEnabled) {\n      this.setupTWCC(packet.header.ssrc);\n    }\n\n    if (codec.name.toLowerCase() === \"rtx\") {\n      const originalSsrc = this.ssrcByRtx[packet.header.ssrc];\n      const codecParams = codecParametersFromString(codec.parameters ?? \"\");\n      const rtxCodec = this.codecs[codecParams[\"apt\"]];\n      if (packet.payload.length < 2) return;\n\n      packet = unwrapRtx(packet, rtxCodec.payloadType, originalSsrc);\n      track = this.trackBySSRC[originalSsrc];\n    }\n\n    let red: Red | undefined;\n    if (codec.name.toLowerCase() === \"red\") {\n      red = Red.deSerialize(packet.payload);\n      if (\n        !Object.keys(this.codecs).includes(\n          red.header.fields[0].blockPT.toString()\n        )\n      ) {\n        return;\n      }\n    }\n\n    if (track?.kind === \"video\" && this.nackEnabled) {\n      this.nack.addPacket(packet);\n    }\n\n    if (track) {\n      if (red) {\n        if (track.kind === \"audio\") {\n          const payloads = this.redHandler.push(red, packet);\n          for (const packet of payloads) {\n            track.onReceiveRtp.execute(packet.clone());\n          }\n        } else {\n        }\n      } else {\n        track.onReceiveRtp.execute(packet.clone());\n      }\n    }\n\n    this.runRtcp();\n  }\n}\n\nexport function unwrapRtx(rtx: RtpPacket, payloadType: number, ssrc: number) {\n  const packet = new RtpPacket(\n    new RtpHeader({\n      payloadType,\n      marker: rtx.header.marker,\n      sequenceNumber: jspack.Unpack(\"!H\", rtx.payload.subarray(0, 2))[0],\n      timestamp: rtx.header.timestamp,\n      ssrc,\n    }),\n    rtx.payload.subarray(2)\n  );\n  return packet;\n}\n"]}

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

@@ -55,11 +55,12 @@ class WebmFactory extends _1.MediaWriter {
                 };
             }
         });
-        const webm = new __1.WebmLiveSink(inputTracks, {
+        const webm = new __1.WebmStream(inputTracks, {
             duration: this.options.defaultDuration ?? 1000 * 60 * 60 * 24,
         });
         this.rtpSources = inputTracks.map(({ track, clockRate, codec }) => {
-            const rtpSource = new __1.RtpSourceStream(track.onReceiveRtp);
+            const rtpSource = new __1.RtpSourceStream();
+            track.onReceiveRtp.subscribe((r) => rtpSource.push(r));
             const jitterBuffer = (0, __1.jitterBufferTransformer)(clockRate, {
                 latency: this.options.jitterBufferLatency,
                 bufferSize: this.options.jitterBufferSize,
@@ -67,15 +68,16 @@ class WebmFactory extends _1.MediaWriter {
             if (track.kind === "video") {
                 rtpSource.readable
                     .pipeThrough(jitterBuffer)
-                    .pipeThrough((0, __1.depacketizeTransformer)((h) => h.marker, codec, {
+                    .pipeThrough((0, __1.depacketizeTransformer)(codec, {
                     waitForKeyframe: this.options.waitForKeyframe,
+                    isFinalPacketInSequence: (h) => h.marker,
                 }))
                     .pipeTo(webm.videoStream);
             }
             else {
                 rtpSource.readable
                     .pipeThrough(jitterBuffer)
-                    .pipeThrough((0, __1.depacketizeTransformer)(() => true, codec))
+                    .pipeThrough((0, __1.depacketizeTransformer)(codec))
                     .pipeTo(webm.audioStream);
             }
             return rtpSource;
@@ -84,8 +86,8 @@ class WebmFactory extends _1.MediaWriter {
         const readChunk = async ({ value, done, }) => {
             if (done)
                 return;
-            if (value.packet) {
-                await (0, promises_1.appendFile)(this.path, value.packet);
+            if (value.saveToFile) {
+                await (0, promises_1.appendFile)(this.path, value.saveToFile);
             }
             else if (value.eol) {
                 const { durationElement } = value.eol;

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

@@ -1 +1 @@
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{ appendFile, open, unlink } from \"fs/promises\";\nimport { ReadableStreamDefaultReadResult } from \"stream/web\";\n\nimport { SupportedCodec } from \"../../../../../rtp/src/container/webm\";\nimport {\n  depacketizeTransformer,\n  jitterBufferTransformer,\n  MediaStreamTrack,\n  RtpSourceStream,\n  WebmLiveOutput,\n  WebmLiveSink,\n  WeriftError,\n} from \"../../..\";\nimport { MediaWriter } from \".\";\n\nconst sourcePath = \"packages/webrtc/src/nonstandard/recorder/writer/webm.ts\";\n\nexport class WebmFactory extends MediaWriter {\n  rtpSources: RtpSourceStream[] = [];\n\n  async start(tracks: MediaStreamTrack[]) {\n    await unlink(this.path).catch((e) => e);\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.options.width,\n          height: this.options.height,\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 WebmLiveSink(inputTracks, {\n      duration: this.options.defaultDuration ?? 1000 * 60 * 60 * 24,\n    });\n\n    this.rtpSources = inputTracks.map(({ track, clockRate, codec }) => {\n      const rtpSource = new RtpSourceStream(track.onReceiveRtp);\n\n      const jitterBuffer = jitterBufferTransformer(clockRate, {\n        latency: this.options.jitterBufferLatency,\n        bufferSize: this.options.jitterBufferSize,\n      });\n\n      if (track.kind === \"video\") {\n        rtpSource.readable\n          .pipeThrough(jitterBuffer)\n          .pipeThrough(\n            depacketizeTransformer((h) => h.marker, codec, {\n              waitForKeyframe: this.options.waitForKeyframe,\n            })\n          )\n          .pipeTo(webm.videoStream);\n      } else {\n        rtpSource.readable\n          .pipeThrough(jitterBuffer)\n          .pipeThrough(depacketizeTransformer(() => true, codec))\n          .pipeTo(webm.audioStream);\n      }\n\n      return rtpSource;\n    });\n\n    const reader = webm.webmStream.getReader();\n    const readChunk = async ({\n      value,\n      done,\n    }: ReadableStreamDefaultReadResult<WebmLiveOutput>) => {\n      if (done) return;\n\n      if (value.packet) {\n        await appendFile(this.path, value.packet);\n      } else if (value.eol) {\n        const { durationElement } = value.eol;\n        const handler = await open(this.path, \"r+\");\n        await handler.write(durationElement, 0, durationElement.length, 83);\n        await handler.close();\n      }\n      reader.read().then(readChunk);\n    };\n    reader.read().then(readChunk);\n  }\n\n  async stop() {\n    await Promise.all(this.rtpSources.map((r) => r.stop()));\n  }\n}\n\nconst supportedVideoCodecs = [\"h264\", \"vp8\", \"vp9\", \"av1x\"] as const;\ntype SupportedVideoCodec = typeof supportedVideoCodecs[number];\n"]}
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{ appendFile, open, unlink } from \"fs/promises\";\nimport { ReadableStreamDefaultReadResult } from \"stream/web\";\n\nimport { SupportedCodec } from \"../../../../../rtp/src/container/webm\";\nimport {\n  depacketizeTransformer,\n  jitterBufferTransformer,\n  MediaStreamTrack,\n  RtpSourceStream,\n  WebmLiveOutput,\n  WebmStream,\n  WeriftError,\n} from \"../../..\";\nimport { MediaWriter } from \".\";\n\nconst sourcePath = \"packages/webrtc/src/nonstandard/recorder/writer/webm.ts\";\n\nexport class WebmFactory extends MediaWriter {\n  rtpSources: RtpSourceStream[] = [];\n\n  async start(tracks: MediaStreamTrack[]) {\n    await unlink(this.path).catch((e) => e);\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.options.width,\n          height: this.options.height,\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 WebmStream(inputTracks, {\n      duration: this.options.defaultDuration ?? 1000 * 60 * 60 * 24,\n    });\n\n    this.rtpSources = inputTracks.map(({ track, clockRate, codec }) => {\n      const rtpSource = new RtpSourceStream();\n      track.onReceiveRtp.subscribe((r) => rtpSource.push(r));\n\n      const jitterBuffer = jitterBufferTransformer(clockRate, {\n        latency: this.options.jitterBufferLatency,\n        bufferSize: this.options.jitterBufferSize,\n      });\n\n      if (track.kind === \"video\") {\n        rtpSource.readable\n          .pipeThrough(jitterBuffer)\n          .pipeThrough(\n            depacketizeTransformer(codec, {\n              waitForKeyframe: this.options.waitForKeyframe,\n              isFinalPacketInSequence: (h) => h.marker,\n            })\n          )\n          .pipeTo(webm.videoStream);\n      } else {\n        rtpSource.readable\n          .pipeThrough(jitterBuffer)\n          .pipeThrough(depacketizeTransformer(codec))\n          .pipeTo(webm.audioStream);\n      }\n\n      return rtpSource;\n    });\n\n    const reader = webm.webmStream.getReader();\n    const readChunk = async ({\n      value,\n      done,\n    }: ReadableStreamDefaultReadResult<WebmLiveOutput>) => {\n      if (done) return;\n\n      if (value.saveToFile) {\n        await appendFile(this.path, value.saveToFile);\n      } else if (value.eol) {\n        const { durationElement } = value.eol;\n        const handler = await open(this.path, \"r+\");\n        await handler.write(durationElement, 0, durationElement.length, 83);\n        await handler.close();\n      }\n      reader.read().then(readChunk);\n    };\n    reader.read().then(readChunk);\n  }\n\n  async stop() {\n    await Promise.all(this.rtpSources.map((r) => r.stop()));\n  }\n}\n\nconst supportedVideoCodecs = [\"h264\", \"vp8\", \"vp9\", \"av1x\"] as const;\ntype SupportedVideoCodec = typeof supportedVideoCodecs[number];\n"]}

package/lib/webrtc/src/peerConnection.js

@@ -768,7 +768,9 @@ class RTCPeerConnection extends helper_1.EventTarget {
                 }));
             }
         }
-        transceiver.receiver.setupTWCC(remoteMedia.ssrc[0]?.ssrc);
+        if (remoteMedia.ssrc[0]?.ssrc) {
+            transceiver.receiver.setupTWCC(remoteMedia.ssrc[0].ssrc);
+        }
     }
     setRemoteSCTP(remoteMedia, sctpTransport, mLineIndex) {
         // # configure sctp