@livekit/rtc-node 0.13.24 → 0.13.25

package/dist/audio_filter.cjs

@@ -21,11 +21,11 @@ __export(audio_filter_exports, {
   AudioFilter: () => AudioFilter
 });
 module.exports = __toCommonJS(audio_filter_exports);
+var import_rtc_ffi_bindings = require("@livekit/rtc-ffi-bindings");
 var import_ffi_client = require("./ffi_client.cjs");
-var import_audio_frame_pb = require("./proto/audio_frame_pb.cjs");
 class AudioFilter {
   constructor(moduleId, path, dependencies = []) {
-    const req = new import_audio_frame_pb.LoadAudioFilterPluginRequest({
+    const req = new import_rtc_ffi_bindings.LoadAudioFilterPluginRequest({
       moduleId,
       pluginPath: path,
       dependencies

package/dist/audio_filter.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/audio_filter.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { FfiClient } from './ffi_client.js';\nimport type { LoadAudioFilterPluginResponse } from './proto/audio_frame_pb.js';\nimport { LoadAudioFilterPluginRequest } from './proto/audio_frame_pb.js';\n\nexport class AudioFilter {\n  constructor(moduleId: string, path: string, dependencies: string[] = []) {\n    const req = new LoadAudioFilterPluginRequest({\n      moduleId,\n      pluginPath: path,\n      dependencies,\n    });\n\n    const res = FfiClient.instance.request<LoadAudioFilterPluginResponse>({\n      message: {\n        case: 'loadAudioFilterPlugin',\n        value: req,\n      },\n    });\n\n    if (res.error) {\n      throw new Error(`Failed to initialize audio filter: ${res.error}`);\n    }\n  }\n}\n"],"mappings":";;;;;;;;;;;;;;;;;;AAAA;AAAA;AAAA;AAAA;AAAA;AAGA,wBAA0B;AAE1B,4BAA6C;AAEtC,MAAM,YAAY;AAAA,EACvB,YAAY,UAAkB,MAAc,eAAyB,CAAC,GAAG;AACvE,UAAM,MAAM,IAAI,mDAA6B;AAAA,MAC3C;AAAA,MACA,YAAY;AAAA,MACZ;AAAA,IACF,CAAC;AAED,UAAM,MAAM,4BAAU,SAAS,QAAuC;AAAA,MACpE,SAAS;AAAA,QACP,MAAM;AAAA,QACN,OAAO;AAAA,MACT;AAAA,IACF,CAAC;AAED,QAAI,IAAI,OAAO;AACb,YAAM,IAAI,MAAM,sCAAsC,IAAI,KAAK,EAAE;AAAA,IACnE;AAAA,EACF;AACF;","names":[]}
+{"version":3,"sources":["../src/audio_filter.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport type { LoadAudioFilterPluginResponse } from '@livekit/rtc-ffi-bindings';\nimport { LoadAudioFilterPluginRequest } from '@livekit/rtc-ffi-bindings';\nimport { FfiClient } from './ffi_client.js';\n\nexport class AudioFilter {\n  constructor(moduleId: string, path: string, dependencies: string[] = []) {\n    const req = new LoadAudioFilterPluginRequest({\n      moduleId,\n      pluginPath: path,\n      dependencies,\n    });\n\n    const res = FfiClient.instance.request<LoadAudioFilterPluginResponse>({\n      message: {\n        case: 'loadAudioFilterPlugin',\n        value: req,\n      },\n    });\n\n    if (res.error) {\n      throw new Error(`Failed to initialize audio filter: ${res.error}`);\n    }\n  }\n}\n"],"mappings":";;;;;;;;;;;;;;;;;;AAAA;AAAA;AAAA;AAAA;AAAA;AAIA,8BAA6C;AAC7C,wBAA0B;AAEnB,MAAM,YAAY;AAAA,EACvB,YAAY,UAAkB,MAAc,eAAyB,CAAC,GAAG;AACvE,UAAM,MAAM,IAAI,qDAA6B;AAAA,MAC3C;AAAA,MACA,YAAY;AAAA,MACZ;AAAA,IACF,CAAC;AAED,UAAM,MAAM,4BAAU,SAAS,QAAuC;AAAA,MACpE,SAAS;AAAA,QACP,MAAM;AAAA,QACN,OAAO;AAAA,MACT;AAAA,IACF,CAAC;AAED,QAAI,IAAI,OAAO;AACb,YAAM,IAAI,MAAM,sCAAsC,IAAI,KAAK,EAAE;AAAA,IACnE;AAAA,EACF;AACF;","names":[]}

package/dist/audio_filter.js

@@ -1,5 +1,5 @@
+import { LoadAudioFilterPluginRequest } from "@livekit/rtc-ffi-bindings";
 import { FfiClient } from "./ffi_client.js";
-import { LoadAudioFilterPluginRequest } from "./proto/audio_frame_pb.js";
 class AudioFilter {
   constructor(moduleId, path, dependencies = []) {
     const req = new LoadAudioFilterPluginRequest({

package/dist/audio_filter.js.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/audio_filter.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { FfiClient } from './ffi_client.js';\nimport type { LoadAudioFilterPluginResponse } from './proto/audio_frame_pb.js';\nimport { LoadAudioFilterPluginRequest } from './proto/audio_frame_pb.js';\n\nexport class AudioFilter {\n  constructor(moduleId: string, path: string, dependencies: string[] = []) {\n    const req = new LoadAudioFilterPluginRequest({\n      moduleId,\n      pluginPath: path,\n      dependencies,\n    });\n\n    const res = FfiClient.instance.request<LoadAudioFilterPluginResponse>({\n      message: {\n        case: 'loadAudioFilterPlugin',\n        value: req,\n      },\n    });\n\n    if (res.error) {\n      throw new Error(`Failed to initialize audio filter: ${res.error}`);\n    }\n  }\n}\n"],"mappings":"AAGA,SAAS,iBAAiB;AAE1B,SAAS,oCAAoC;AAEtC,MAAM,YAAY;AAAA,EACvB,YAAY,UAAkB,MAAc,eAAyB,CAAC,GAAG;AACvE,UAAM,MAAM,IAAI,6BAA6B;AAAA,MAC3C;AAAA,MACA,YAAY;AAAA,MACZ;AAAA,IACF,CAAC;AAED,UAAM,MAAM,UAAU,SAAS,QAAuC;AAAA,MACpE,SAAS;AAAA,QACP,MAAM;AAAA,QACN,OAAO;AAAA,MACT;AAAA,IACF,CAAC;AAED,QAAI,IAAI,OAAO;AACb,YAAM,IAAI,MAAM,sCAAsC,IAAI,KAAK,EAAE;AAAA,IACnE;AAAA,EACF;AACF;","names":[]}
+{"version":3,"sources":["../src/audio_filter.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport type { LoadAudioFilterPluginResponse } from '@livekit/rtc-ffi-bindings';\nimport { LoadAudioFilterPluginRequest } from '@livekit/rtc-ffi-bindings';\nimport { FfiClient } from './ffi_client.js';\n\nexport class AudioFilter {\n  constructor(moduleId: string, path: string, dependencies: string[] = []) {\n    const req = new LoadAudioFilterPluginRequest({\n      moduleId,\n      pluginPath: path,\n      dependencies,\n    });\n\n    const res = FfiClient.instance.request<LoadAudioFilterPluginResponse>({\n      message: {\n        case: 'loadAudioFilterPlugin',\n        value: req,\n      },\n    });\n\n    if (res.error) {\n      throw new Error(`Failed to initialize audio filter: ${res.error}`);\n    }\n  }\n}\n"],"mappings":"AAIA,SAAS,oCAAoC;AAC7C,SAAS,iBAAiB;AAEnB,MAAM,YAAY;AAAA,EACvB,YAAY,UAAkB,MAAc,eAAyB,CAAC,GAAG;AACvE,UAAM,MAAM,IAAI,6BAA6B;AAAA,MAC3C;AAAA,MACA,YAAY;AAAA,MACZ;AAAA,IACF,CAAC;AAED,UAAM,MAAM,UAAU,SAAS,QAAuC;AAAA,MACpE,SAAS;AAAA,QACP,MAAM;AAAA,QACN,OAAO;AAAA,MACT;AAAA,IACF,CAAC;AAED,QAAI,IAAI,OAAO;AACb,YAAM,IAAI,MAAM,sCAAsC,IAAI,KAAK,EAAE;AAAA,IACnE;AAAA,EACF;AACF;","names":[]}

package/dist/audio_frame.cjs

@@ -22,8 +22,8 @@ __export(audio_frame_exports, {
   combineAudioFrames: () => combineAudioFrames
 });
 module.exports = __toCommonJS(audio_frame_exports);
+var import_rtc_ffi_bindings = require("@livekit/rtc-ffi-bindings");
 var import_ffi_client = require("./ffi_client.cjs");
-var import_audio_frame_pb = require("./proto/audio_frame_pb.cjs");
 class AudioFrame {
   // note: if converting from Uint8Array to Int16Array, *do not* use buffer.slice!
   // it is marked unstable by Node and can cause undefined behaviour, such as massive chunks of
@@ -37,9 +37,9 @@ class AudioFrame {
     this.samplesPerChannel = samplesPerChannel;
     this._userdata = userdata;
   }
-  static create(sampleRate, channels, samplesPerChannel) {
+  static create(sampleRate, channels, samplesPerChannel, userdata) {
     const data = new Int16Array(channels * samplesPerChannel);
-    return new AudioFrame(data, sampleRate, channels, samplesPerChannel);
+    return new AudioFrame(data, sampleRate, channels, samplesPerChannel, userdata);
   }
   /** @internal */
   static fromOwnedInfo(owned) {
@@ -56,7 +56,7 @@ class AudioFrame {
   }
   /** @internal */
   protoInfo() {
-    return new import_audio_frame_pb.AudioFrameBufferInfo({
+    return new import_rtc_ffi_bindings.AudioFrameBufferInfo({
       dataPtr: import_ffi_client.FfiClient.instance.retrievePtr(new Uint8Array(this.data.buffer)),
       sampleRate: this.sampleRate,
       numChannels: this.channels,
@@ -89,7 +89,11 @@ const combineAudioFrames = (buffer) => {
     totalSamplesPerChannel += frame.samplesPerChannel;
   }
   const data = new Int16Array(buffer.map((x) => [...x.data]).flat());
-  return new AudioFrame(data, sampleRate, channels, totalSamplesPerChannel);
+  const mergedUserdata = {};
+  for (const frame of buffer) {
+    Object.assign(mergedUserdata, frame.userdata);
+  }
+  return new AudioFrame(data, sampleRate, channels, totalSamplesPerChannel, mergedUserdata);
 };
 // Annotate the CommonJS export names for ESM import in node:
 0 && (module.exports = {

package/dist/audio_frame.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/audio_frame.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { FfiClient, FfiHandle } from './ffi_client.js';\nimport type { OwnedAudioFrameBuffer } from './proto/audio_frame_pb.js';\nimport { AudioFrameBufferInfo } from './proto/audio_frame_pb.js';\n\nexport class AudioFrame {\n  data: Int16Array;\n  sampleRate: number;\n  channels: number;\n  samplesPerChannel: number;\n\n  private _userdata: Record<string, unknown>;\n\n  // note: if converting from Uint8Array to Int16Array, *do not* use buffer.slice!\n  // it is marked unstable by Node and can cause undefined behaviour, such as massive chunks of\n  // noise being added to the end.\n  // it is recommended to use buffer.subarray instead.\n  // XXX(nbsp): add this when writing proper docs\n  constructor(\n    data: Int16Array,\n    sampleRate: number,\n    channels: number,\n    samplesPerChannel: number,\n    userdata: Record<string, unknown> = {},\n  ) {\n    this.data = data;\n    this.sampleRate = sampleRate;\n    this.channels = channels;\n    this.samplesPerChannel = samplesPerChannel;\n    this._userdata = userdata;\n  }\n\n  static create(sampleRate: number, channels: number, samplesPerChannel: number): AudioFrame {\n    const data = new Int16Array(channels * samplesPerChannel);\n    return new AudioFrame(data, sampleRate, channels, samplesPerChannel);\n  }\n\n  /** @internal */\n  static fromOwnedInfo(owned: OwnedAudioFrameBuffer): AudioFrame {\n    const info = owned.info!;\n    const len = info.numChannels! * info.samplesPerChannel! * 2; // c_int16\n    const data = FfiClient.instance.copyBuffer(info.dataPtr!, len);\n    new FfiHandle(owned.handle!.id!).dispose();\n    return new AudioFrame(\n      new Int16Array(data.buffer),\n      info.sampleRate!,\n      info.numChannels!,\n      info.samplesPerChannel!,\n    );\n  }\n\n  /** @internal */\n  protoInfo(): AudioFrameBufferInfo {\n    return new AudioFrameBufferInfo({\n      dataPtr: FfiClient.instance.retrievePtr(new Uint8Array(this.data.buffer)),\n      sampleRate: this.sampleRate,\n      numChannels: this.channels,\n      samplesPerChannel: this.samplesPerChannel,\n    });\n  }\n\n  /** Returns the user data associated with the audio frame. */\n  get userdata() {\n    return this._userdata;\n  }\n}\n\n/**\n * Combines one or more `rtc.AudioFrame` objects into a single `rtc.AudioFrame`.\n *\n * This function concatenates the audio data from multiple frames, ensuring that all frames have\n * the same sample rate and number of channels. It efficiently merges the data by preallocating the\n * necessary memory and copying the frame data without unnecessary reallocations.\n *\n * @param buffer - a single AudioFrame or list thereof\n */\nexport const combineAudioFrames = (buffer: AudioFrame | AudioFrame[]): AudioFrame => {\n  if (!Array.isArray(buffer)) {\n    return buffer;\n  }\n  buffer = buffer as AudioFrame[];\n\n  if (buffer.length === 0) {\n    throw new Error('buffer is empty');\n  }\n\n  const sampleRate = buffer[0]!.sampleRate;\n  const channels = buffer[0]!.channels;\n\n  let totalSamplesPerChannel = 0;\n  for (const frame of buffer) {\n    if (frame.sampleRate != sampleRate) {\n      throw new Error(`sample rate mismatch: expected ${sampleRate}, got ${frame.sampleRate}`);\n    }\n\n    if (frame.channels != channels) {\n      throw new Error(`channel mismatch: expected ${channels}, got ${frame.channels}`);\n    }\n\n    totalSamplesPerChannel += frame.samplesPerChannel;\n  }\n\n  const data = new Int16Array(buffer.map((x) => [...x.data]).flat());\n  return new AudioFrame(data, sampleRate, channels, totalSamplesPerChannel);\n};\n"],"mappings":";;;;;;;;;;;;;;;;;;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAGA,wBAAqC;AAErC,4BAAqC;AAE9B,MAAM,WAAW;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA,EAatB,YACE,MACA,YACA,UACA,mBACA,WAAoC,CAAC,GACrC;AACA,SAAK,OAAO;AACZ,SAAK,aAAa;AAClB,SAAK,WAAW;AAChB,SAAK,oBAAoB;AACzB,SAAK,YAAY;AAAA,EACnB;AAAA,EAEA,OAAO,OAAO,YAAoB,UAAkB,mBAAuC;AACzF,UAAM,OAAO,IAAI,WAAW,WAAW,iBAAiB;AACxD,WAAO,IAAI,WAAW,MAAM,YAAY,UAAU,iBAAiB;AAAA,EACrE;AAAA;AAAA,EAGA,OAAO,cAAc,OAA0C;AAC7D,UAAM,OAAO,MAAM;AACnB,UAAM,MAAM,KAAK,cAAe,KAAK,oBAAqB;AAC1D,UAAM,OAAO,4BAAU,SAAS,WAAW,KAAK,SAAU,GAAG;AAC7D,QAAI,4BAAU,MAAM,OAAQ,EAAG,EAAE,QAAQ;AACzC,WAAO,IAAI;AAAA,MACT,IAAI,WAAW,KAAK,MAAM;AAAA,MAC1B,KAAK;AAAA,MACL,KAAK;AAAA,MACL,KAAK;AAAA,IACP;AAAA,EACF;AAAA;AAAA,EAGA,YAAkC;AAChC,WAAO,IAAI,2CAAqB;AAAA,MAC9B,SAAS,4BAAU,SAAS,YAAY,IAAI,WAAW,KAAK,KAAK,MAAM,CAAC;AAAA,MACxE,YAAY,KAAK;AAAA,MACjB,aAAa,KAAK;AAAA,MAClB,mBAAmB,KAAK;AAAA,IAC1B,CAAC;AAAA,EACH;AAAA;AAAA,EAGA,IAAI,WAAW;AACb,WAAO,KAAK;AAAA,EACd;AACF;AAWO,MAAM,qBAAqB,CAAC,WAAkD;AACnF,MAAI,CAAC,MAAM,QAAQ,MAAM,GAAG;AAC1B,WAAO;AAAA,EACT;AACA,WAAS;AAET,MAAI,OAAO,WAAW,GAAG;AACvB,UAAM,IAAI,MAAM,iBAAiB;AAAA,EACnC;AAEA,QAAM,aAAa,OAAO,CAAC,EAAG;AAC9B,QAAM,WAAW,OAAO,CAAC,EAAG;AAE5B,MAAI,yBAAyB;AAC7B,aAAW,SAAS,QAAQ;AAC1B,QAAI,MAAM,cAAc,YAAY;AAClC,YAAM,IAAI,MAAM,kCAAkC,UAAU,SAAS,MAAM,UAAU,EAAE;AAAA,IACzF;AAEA,QAAI,MAAM,YAAY,UAAU;AAC9B,YAAM,IAAI,MAAM,8BAA8B,QAAQ,SAAS,MAAM,QAAQ,EAAE;AAAA,IACjF;AAEA,8BAA0B,MAAM;AAAA,EAClC;AAEA,QAAM,OAAO,IAAI,WAAW,OAAO,IAAI,CAAC,MAAM,CAAC,GAAG,EAAE,IAAI,CAAC,EAAE,KAAK,CAAC;AACjE,SAAO,IAAI,WAAW,MAAM,YAAY,UAAU,sBAAsB;AAC1E;","names":[]}
+{"version":3,"sources":["../src/audio_frame.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport type { OwnedAudioFrameBuffer } from '@livekit/rtc-ffi-bindings';\nimport { AudioFrameBufferInfo } from '@livekit/rtc-ffi-bindings';\nimport { FfiClient, FfiHandle } from './ffi_client.js';\n\nexport class AudioFrame {\n  data: Int16Array;\n  sampleRate: number;\n  channels: number;\n  samplesPerChannel: number;\n\n  private _userdata: Record<string, unknown>;\n\n  // note: if converting from Uint8Array to Int16Array, *do not* use buffer.slice!\n  // it is marked unstable by Node and can cause undefined behaviour, such as massive chunks of\n  // noise being added to the end.\n  // it is recommended to use buffer.subarray instead.\n  // XXX(nbsp): add this when writing proper docs\n  constructor(\n    data: Int16Array,\n    sampleRate: number,\n    channels: number,\n    samplesPerChannel: number,\n    userdata: Record<string, unknown> = {},\n  ) {\n    this.data = data;\n    this.sampleRate = sampleRate;\n    this.channels = channels;\n    this.samplesPerChannel = samplesPerChannel;\n    this._userdata = userdata;\n  }\n\n  static create(\n    sampleRate: number,\n    channels: number,\n    samplesPerChannel: number,\n    userdata?: Record<string, unknown>,\n  ): AudioFrame {\n    const data = new Int16Array(channels * samplesPerChannel);\n    return new AudioFrame(data, sampleRate, channels, samplesPerChannel, userdata);\n  }\n\n  /** @internal */\n  static fromOwnedInfo(owned: OwnedAudioFrameBuffer): AudioFrame {\n    const info = owned.info!;\n    const len = info.numChannels! * info.samplesPerChannel! * 2; // c_int16\n    const data = FfiClient.instance.copyBuffer(info.dataPtr!, len);\n    new FfiHandle(owned.handle!.id!).dispose();\n    return new AudioFrame(\n      new Int16Array(data.buffer),\n      info.sampleRate!,\n      info.numChannels!,\n      info.samplesPerChannel!,\n    );\n  }\n\n  /** @internal */\n  protoInfo(): AudioFrameBufferInfo {\n    return new AudioFrameBufferInfo({\n      dataPtr: FfiClient.instance.retrievePtr(new Uint8Array(this.data.buffer)),\n      sampleRate: this.sampleRate,\n      numChannels: this.channels,\n      samplesPerChannel: this.samplesPerChannel,\n    });\n  }\n\n  /** Returns the user data associated with the audio frame. */\n  get userdata() {\n    return this._userdata;\n  }\n}\n\n/**\n * Combines one or more `rtc.AudioFrame` objects into a single `rtc.AudioFrame`.\n *\n * This function concatenates the audio data from multiple frames, ensuring that all frames have\n * the same sample rate and number of channels. It efficiently merges the data by preallocating the\n * necessary memory and copying the frame data without unnecessary reallocations.\n *\n * @param buffer - a single AudioFrame or list thereof\n */\nexport const combineAudioFrames = (buffer: AudioFrame | AudioFrame[]): AudioFrame => {\n  if (!Array.isArray(buffer)) {\n    return buffer;\n  }\n  buffer = buffer as AudioFrame[];\n\n  if (buffer.length === 0) {\n    throw new Error('buffer is empty');\n  }\n\n  const sampleRate = buffer[0]!.sampleRate;\n  const channels = buffer[0]!.channels;\n\n  let totalSamplesPerChannel = 0;\n  for (const frame of buffer) {\n    if (frame.sampleRate != sampleRate) {\n      throw new Error(`sample rate mismatch: expected ${sampleRate}, got ${frame.sampleRate}`);\n    }\n\n    if (frame.channels != channels) {\n      throw new Error(`channel mismatch: expected ${channels}, got ${frame.channels}`);\n    }\n\n    totalSamplesPerChannel += frame.samplesPerChannel;\n  }\n\n  const data = new Int16Array(buffer.map((x) => [...x.data]).flat());\n\n  // Merge userdata from all frames\n  const mergedUserdata: Record<string, unknown> = {};\n  for (const frame of buffer) {\n    Object.assign(mergedUserdata, frame.userdata);\n  }\n\n  return new AudioFrame(data, sampleRate, channels, totalSamplesPerChannel, mergedUserdata);\n};\n"],"mappings":";;;;;;;;;;;;;;;;;;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAIA,8BAAqC;AACrC,wBAAqC;AAE9B,MAAM,WAAW;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA,EAatB,YACE,MACA,YACA,UACA,mBACA,WAAoC,CAAC,GACrC;AACA,SAAK,OAAO;AACZ,SAAK,aAAa;AAClB,SAAK,WAAW;AAChB,SAAK,oBAAoB;AACzB,SAAK,YAAY;AAAA,EACnB;AAAA,EAEA,OAAO,OACL,YACA,UACA,mBACA,UACY;AACZ,UAAM,OAAO,IAAI,WAAW,WAAW,iBAAiB;AACxD,WAAO,IAAI,WAAW,MAAM,YAAY,UAAU,mBAAmB,QAAQ;AAAA,EAC/E;AAAA;AAAA,EAGA,OAAO,cAAc,OAA0C;AAC7D,UAAM,OAAO,MAAM;AACnB,UAAM,MAAM,KAAK,cAAe,KAAK,oBAAqB;AAC1D,UAAM,OAAO,4BAAU,SAAS,WAAW,KAAK,SAAU,GAAG;AAC7D,QAAI,4BAAU,MAAM,OAAQ,EAAG,EAAE,QAAQ;AACzC,WAAO,IAAI;AAAA,MACT,IAAI,WAAW,KAAK,MAAM;AAAA,MAC1B,KAAK;AAAA,MACL,KAAK;AAAA,MACL,KAAK;AAAA,IACP;AAAA,EACF;AAAA;AAAA,EAGA,YAAkC;AAChC,WAAO,IAAI,6CAAqB;AAAA,MAC9B,SAAS,4BAAU,SAAS,YAAY,IAAI,WAAW,KAAK,KAAK,MAAM,CAAC;AAAA,MACxE,YAAY,KAAK;AAAA,MACjB,aAAa,KAAK;AAAA,MAClB,mBAAmB,KAAK;AAAA,IAC1B,CAAC;AAAA,EACH;AAAA;AAAA,EAGA,IAAI,WAAW;AACb,WAAO,KAAK;AAAA,EACd;AACF;AAWO,MAAM,qBAAqB,CAAC,WAAkD;AACnF,MAAI,CAAC,MAAM,QAAQ,MAAM,GAAG;AAC1B,WAAO;AAAA,EACT;AACA,WAAS;AAET,MAAI,OAAO,WAAW,GAAG;AACvB,UAAM,IAAI,MAAM,iBAAiB;AAAA,EACnC;AAEA,QAAM,aAAa,OAAO,CAAC,EAAG;AAC9B,QAAM,WAAW,OAAO,CAAC,EAAG;AAE5B,MAAI,yBAAyB;AAC7B,aAAW,SAAS,QAAQ;AAC1B,QAAI,MAAM,cAAc,YAAY;AAClC,YAAM,IAAI,MAAM,kCAAkC,UAAU,SAAS,MAAM,UAAU,EAAE;AAAA,IACzF;AAEA,QAAI,MAAM,YAAY,UAAU;AAC9B,YAAM,IAAI,MAAM,8BAA8B,QAAQ,SAAS,MAAM,QAAQ,EAAE;AAAA,IACjF;AAEA,8BAA0B,MAAM;AAAA,EAClC;AAEA,QAAM,OAAO,IAAI,WAAW,OAAO,IAAI,CAAC,MAAM,CAAC,GAAG,EAAE,IAAI,CAAC,EAAE,KAAK,CAAC;AAGjE,QAAM,iBAA0C,CAAC;AACjD,aAAW,SAAS,QAAQ;AAC1B,WAAO,OAAO,gBAAgB,MAAM,QAAQ;AAAA,EAC9C;AAEA,SAAO,IAAI,WAAW,MAAM,YAAY,UAAU,wBAAwB,cAAc;AAC1F;","names":[]}

package/dist/audio_frame.d.cts

@@ -1,9 +1,4 @@
-import { OwnedAudioFrameBuffer, AudioFrameBufferInfo } from './proto/audio_frame_pb.cjs';
-import '@bufbuild/protobuf';
-import './proto/track_pb.cjs';
-import './proto/stats_pb.cjs';
-import './proto/e2ee_pb.cjs';
-import './proto/handle_pb.cjs';
+import { OwnedAudioFrameBuffer, AudioFrameBufferInfo } from '@livekit/rtc-ffi-bindings';
 
 declare class AudioFrame {
     data: Int16Array;
@@ -12,7 +7,7 @@ declare class AudioFrame {
     samplesPerChannel: number;
     private _userdata;
     constructor(data: Int16Array, sampleRate: number, channels: number, samplesPerChannel: number, userdata?: Record<string, unknown>);
-    static create(sampleRate: number, channels: number, samplesPerChannel: number): AudioFrame;
+    static create(sampleRate: number, channels: number, samplesPerChannel: number, userdata?: Record<string, unknown>): AudioFrame;
     /** @internal */
     static fromOwnedInfo(owned: OwnedAudioFrameBuffer): AudioFrame;
     /** @internal */

package/dist/audio_frame.d.ts

@@ -1,9 +1,4 @@
-import { OwnedAudioFrameBuffer, AudioFrameBufferInfo } from './proto/audio_frame_pb.js';
-import '@bufbuild/protobuf';
-import './proto/track_pb.js';
-import './proto/stats_pb.js';
-import './proto/e2ee_pb.js';
-import './proto/handle_pb.js';
+import { OwnedAudioFrameBuffer, AudioFrameBufferInfo } from '@livekit/rtc-ffi-bindings';
 
 declare class AudioFrame {
     data: Int16Array;
@@ -12,7 +7,7 @@ declare class AudioFrame {
     samplesPerChannel: number;
     private _userdata;
     constructor(data: Int16Array, sampleRate: number, channels: number, samplesPerChannel: number, userdata?: Record<string, unknown>);
-    static create(sampleRate: number, channels: number, samplesPerChannel: number): AudioFrame;
+    static create(sampleRate: number, channels: number, samplesPerChannel: number, userdata?: Record<string, unknown>): AudioFrame;
     /** @internal */
     static fromOwnedInfo(owned: OwnedAudioFrameBuffer): AudioFrame;
     /** @internal */

package/dist/audio_frame.d.ts.map

@@ -1 +1 @@
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package/dist/audio_frame.js

@@ -1,5 +1,5 @@
+import { AudioFrameBufferInfo } from "@livekit/rtc-ffi-bindings";
 import { FfiClient, FfiHandle } from "./ffi_client.js";
-import { AudioFrameBufferInfo } from "./proto/audio_frame_pb.js";
 class AudioFrame {
   // note: if converting from Uint8Array to Int16Array, *do not* use buffer.slice!
   // it is marked unstable by Node and can cause undefined behaviour, such as massive chunks of
@@ -13,9 +13,9 @@ class AudioFrame {
     this.samplesPerChannel = samplesPerChannel;
     this._userdata = userdata;
   }
-  static create(sampleRate, channels, samplesPerChannel) {
+  static create(sampleRate, channels, samplesPerChannel, userdata) {
     const data = new Int16Array(channels * samplesPerChannel);
-    return new AudioFrame(data, sampleRate, channels, samplesPerChannel);
+    return new AudioFrame(data, sampleRate, channels, samplesPerChannel, userdata);
   }
   /** @internal */
   static fromOwnedInfo(owned) {
@@ -65,7 +65,11 @@ const combineAudioFrames = (buffer) => {
     totalSamplesPerChannel += frame.samplesPerChannel;
   }
   const data = new Int16Array(buffer.map((x) => [...x.data]).flat());
-  return new AudioFrame(data, sampleRate, channels, totalSamplesPerChannel);
+  const mergedUserdata = {};
+  for (const frame of buffer) {
+    Object.assign(mergedUserdata, frame.userdata);
+  }
+  return new AudioFrame(data, sampleRate, channels, totalSamplesPerChannel, mergedUserdata);
 };
 export {
   AudioFrame,

package/dist/audio_frame.js.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/audio_frame.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { FfiClient, FfiHandle } from './ffi_client.js';\nimport type { OwnedAudioFrameBuffer } from './proto/audio_frame_pb.js';\nimport { AudioFrameBufferInfo } from './proto/audio_frame_pb.js';\n\nexport class AudioFrame {\n  data: Int16Array;\n  sampleRate: number;\n  channels: number;\n  samplesPerChannel: number;\n\n  private _userdata: Record<string, unknown>;\n\n  // note: if converting from Uint8Array to Int16Array, *do not* use buffer.slice!\n  // it is marked unstable by Node and can cause undefined behaviour, such as massive chunks of\n  // noise being added to the end.\n  // it is recommended to use buffer.subarray instead.\n  // XXX(nbsp): add this when writing proper docs\n  constructor(\n    data: Int16Array,\n    sampleRate: number,\n    channels: number,\n    samplesPerChannel: number,\n    userdata: Record<string, unknown> = {},\n  ) {\n    this.data = data;\n    this.sampleRate = sampleRate;\n    this.channels = channels;\n    this.samplesPerChannel = samplesPerChannel;\n    this._userdata = userdata;\n  }\n\n  static create(sampleRate: number, channels: number, samplesPerChannel: number): AudioFrame {\n    const data = new Int16Array(channels * samplesPerChannel);\n    return new AudioFrame(data, sampleRate, channels, samplesPerChannel);\n  }\n\n  /** @internal */\n  static fromOwnedInfo(owned: OwnedAudioFrameBuffer): AudioFrame {\n    const info = owned.info!;\n    const len = info.numChannels! * info.samplesPerChannel! * 2; // c_int16\n    const data = FfiClient.instance.copyBuffer(info.dataPtr!, len);\n    new FfiHandle(owned.handle!.id!).dispose();\n    return new AudioFrame(\n      new Int16Array(data.buffer),\n      info.sampleRate!,\n      info.numChannels!,\n      info.samplesPerChannel!,\n    );\n  }\n\n  /** @internal */\n  protoInfo(): AudioFrameBufferInfo {\n    return new AudioFrameBufferInfo({\n      dataPtr: FfiClient.instance.retrievePtr(new Uint8Array(this.data.buffer)),\n      sampleRate: this.sampleRate,\n      numChannels: this.channels,\n      samplesPerChannel: this.samplesPerChannel,\n    });\n  }\n\n  /** Returns the user data associated with the audio frame. */\n  get userdata() {\n    return this._userdata;\n  }\n}\n\n/**\n * Combines one or more `rtc.AudioFrame` objects into a single `rtc.AudioFrame`.\n *\n * This function concatenates the audio data from multiple frames, ensuring that all frames have\n * the same sample rate and number of channels. It efficiently merges the data by preallocating the\n * necessary memory and copying the frame data without unnecessary reallocations.\n *\n * @param buffer - a single AudioFrame or list thereof\n */\nexport const combineAudioFrames = (buffer: AudioFrame | AudioFrame[]): AudioFrame => {\n  if (!Array.isArray(buffer)) {\n    return buffer;\n  }\n  buffer = buffer as AudioFrame[];\n\n  if (buffer.length === 0) {\n    throw new Error('buffer is empty');\n  }\n\n  const sampleRate = buffer[0]!.sampleRate;\n  const channels = buffer[0]!.channels;\n\n  let totalSamplesPerChannel = 0;\n  for (const frame of buffer) {\n    if (frame.sampleRate != sampleRate) {\n      throw new Error(`sample rate mismatch: expected ${sampleRate}, got ${frame.sampleRate}`);\n    }\n\n    if (frame.channels != channels) {\n      throw new Error(`channel mismatch: expected ${channels}, got ${frame.channels}`);\n    }\n\n    totalSamplesPerChannel += frame.samplesPerChannel;\n  }\n\n  const data = new Int16Array(buffer.map((x) => [...x.data]).flat());\n  return new AudioFrame(data, sampleRate, channels, totalSamplesPerChannel);\n};\n"],"mappings":"AAGA,SAAS,WAAW,iBAAiB;AAErC,SAAS,4BAA4B;AAE9B,MAAM,WAAW;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA,EAatB,YACE,MACA,YACA,UACA,mBACA,WAAoC,CAAC,GACrC;AACA,SAAK,OAAO;AACZ,SAAK,aAAa;AAClB,SAAK,WAAW;AAChB,SAAK,oBAAoB;AACzB,SAAK,YAAY;AAAA,EACnB;AAAA,EAEA,OAAO,OAAO,YAAoB,UAAkB,mBAAuC;AACzF,UAAM,OAAO,IAAI,WAAW,WAAW,iBAAiB;AACxD,WAAO,IAAI,WAAW,MAAM,YAAY,UAAU,iBAAiB;AAAA,EACrE;AAAA;AAAA,EAGA,OAAO,cAAc,OAA0C;AAC7D,UAAM,OAAO,MAAM;AACnB,UAAM,MAAM,KAAK,cAAe,KAAK,oBAAqB;AAC1D,UAAM,OAAO,UAAU,SAAS,WAAW,KAAK,SAAU,GAAG;AAC7D,QAAI,UAAU,MAAM,OAAQ,EAAG,EAAE,QAAQ;AACzC,WAAO,IAAI;AAAA,MACT,IAAI,WAAW,KAAK,MAAM;AAAA,MAC1B,KAAK;AAAA,MACL,KAAK;AAAA,MACL,KAAK;AAAA,IACP;AAAA,EACF;AAAA;AAAA,EAGA,YAAkC;AAChC,WAAO,IAAI,qBAAqB;AAAA,MAC9B,SAAS,UAAU,SAAS,YAAY,IAAI,WAAW,KAAK,KAAK,MAAM,CAAC;AAAA,MACxE,YAAY,KAAK;AAAA,MACjB,aAAa,KAAK;AAAA,MAClB,mBAAmB,KAAK;AAAA,IAC1B,CAAC;AAAA,EACH;AAAA;AAAA,EAGA,IAAI,WAAW;AACb,WAAO,KAAK;AAAA,EACd;AACF;AAWO,MAAM,qBAAqB,CAAC,WAAkD;AACnF,MAAI,CAAC,MAAM,QAAQ,MAAM,GAAG;AAC1B,WAAO;AAAA,EACT;AACA,WAAS;AAET,MAAI,OAAO,WAAW,GAAG;AACvB,UAAM,IAAI,MAAM,iBAAiB;AAAA,EACnC;AAEA,QAAM,aAAa,OAAO,CAAC,EAAG;AAC9B,QAAM,WAAW,OAAO,CAAC,EAAG;AAE5B,MAAI,yBAAyB;AAC7B,aAAW,SAAS,QAAQ;AAC1B,QAAI,MAAM,cAAc,YAAY;AAClC,YAAM,IAAI,MAAM,kCAAkC,UAAU,SAAS,MAAM,UAAU,EAAE;AAAA,IACzF;AAEA,QAAI,MAAM,YAAY,UAAU;AAC9B,YAAM,IAAI,MAAM,8BAA8B,QAAQ,SAAS,MAAM,QAAQ,EAAE;AAAA,IACjF;AAEA,8BAA0B,MAAM;AAAA,EAClC;AAEA,QAAM,OAAO,IAAI,WAAW,OAAO,IAAI,CAAC,MAAM,CAAC,GAAG,EAAE,IAAI,CAAC,EAAE,KAAK,CAAC;AACjE,SAAO,IAAI,WAAW,MAAM,YAAY,UAAU,sBAAsB;AAC1E;","names":[]}
+{"version":3,"sources":["../src/audio_frame.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport type { OwnedAudioFrameBuffer } from '@livekit/rtc-ffi-bindings';\nimport { AudioFrameBufferInfo } from '@livekit/rtc-ffi-bindings';\nimport { FfiClient, FfiHandle } from './ffi_client.js';\n\nexport class AudioFrame {\n  data: Int16Array;\n  sampleRate: number;\n  channels: number;\n  samplesPerChannel: number;\n\n  private _userdata: Record<string, unknown>;\n\n  // note: if converting from Uint8Array to Int16Array, *do not* use buffer.slice!\n  // it is marked unstable by Node and can cause undefined behaviour, such as massive chunks of\n  // noise being added to the end.\n  // it is recommended to use buffer.subarray instead.\n  // XXX(nbsp): add this when writing proper docs\n  constructor(\n    data: Int16Array,\n    sampleRate: number,\n    channels: number,\n    samplesPerChannel: number,\n    userdata: Record<string, unknown> = {},\n  ) {\n    this.data = data;\n    this.sampleRate = sampleRate;\n    this.channels = channels;\n    this.samplesPerChannel = samplesPerChannel;\n    this._userdata = userdata;\n  }\n\n  static create(\n    sampleRate: number,\n    channels: number,\n    samplesPerChannel: number,\n    userdata?: Record<string, unknown>,\n  ): AudioFrame {\n    const data = new Int16Array(channels * samplesPerChannel);\n    return new AudioFrame(data, sampleRate, channels, samplesPerChannel, userdata);\n  }\n\n  /** @internal */\n  static fromOwnedInfo(owned: OwnedAudioFrameBuffer): AudioFrame {\n    const info = owned.info!;\n    const len = info.numChannels! * info.samplesPerChannel! * 2; // c_int16\n    const data = FfiClient.instance.copyBuffer(info.dataPtr!, len);\n    new FfiHandle(owned.handle!.id!).dispose();\n    return new AudioFrame(\n      new Int16Array(data.buffer),\n      info.sampleRate!,\n      info.numChannels!,\n      info.samplesPerChannel!,\n    );\n  }\n\n  /** @internal */\n  protoInfo(): AudioFrameBufferInfo {\n    return new AudioFrameBufferInfo({\n      dataPtr: FfiClient.instance.retrievePtr(new Uint8Array(this.data.buffer)),\n      sampleRate: this.sampleRate,\n      numChannels: this.channels,\n      samplesPerChannel: this.samplesPerChannel,\n    });\n  }\n\n  /** Returns the user data associated with the audio frame. */\n  get userdata() {\n    return this._userdata;\n  }\n}\n\n/**\n * Combines one or more `rtc.AudioFrame` objects into a single `rtc.AudioFrame`.\n *\n * This function concatenates the audio data from multiple frames, ensuring that all frames have\n * the same sample rate and number of channels. 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package/dist/audio_mixer.cjs

@@ -152,7 +152,7 @@ class AudioMixer {
       const removals = [];
       for (const result of results) {
         if (result.status !== "fulfilled") {
-          import_log.log.warn("AudioMixer: Stream contribution failed:", result.reason);
+          import_log.log.warn({ reason: result.reason }, "AudioMixer: Stream contribution failed");
           continue;
         }
         const contrib = result.value;

package/dist/audio_mixer.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/audio_mixer.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2025 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { AsyncQueue } from './async_queue.js';\nimport { AudioFrame } from './audio_frame.js';\nimport { log } from './log.js';\n\n// Re-export AsyncQueue for backward compatibility\nexport { AsyncQueue } from './async_queue.js';\n\n// Define types for async iteration (since lib: es2015 doesn't include them)\ntype AudioStream = {\n  [Symbol.asyncIterator](): {\n    next(): Promise<IteratorResult<AudioFrame>>;\n  };\n};\n\ninterface Contribution {\n  stream: AudioStream;\n  data: Int16Array;\n  buffer: Int16Array;\n  hadData: boolean;\n  exhausted: boolean;\n}\n\nexport interface AudioMixerOptions {\n  /**\n   * The size of the audio block (in samples) for mixing.\n   * If not provided, defaults to sampleRate / 10 (100ms).\n   */\n  blocksize?: number;\n\n  /**\n   * The maximum wait time in milliseconds for each stream to provide\n   * audio data before timing out. Defaults to 100 ms.\n   */\n  streamTimeoutMs?: number;\n\n  /**\n   * The maximum number of mixed frames to store in the output queue.\n   * Defaults to 100.\n   */\n  capacity?: number;\n}\n\n/**\n * AudioMixer combines multiple async audio streams into a single output stream.\n *\n * The mixer accepts multiple async audio streams and mixes them into a single output stream.\n * Each output frame is generated with a fixed chunk size determined by the blocksize (in samples).\n * If blocksize is not provided (or 0), it defaults to 100ms.\n *\n * Each input stream is processed in parallel, accumulating audio data until at least one chunk\n * of samples is available. If an input stream does not provide data within the specified timeout,\n * a warning is logged. The mixer can be closed immediately\n * (dropping unconsumed frames) or allowed to flush remaining data using endInput().\n *\n * @example\n * ```typescript\n * const mixer = new AudioMixer(48000, 2);\n * mixer.addStream(stream1);\n * mixer.addStream(stream2);\n *\n * for await (const frame of mixer) {\n *   // Process mixed audio frame\n * }\n * ```\n */\nexport class AudioMixer {\n  private streams: Set<AudioStream>;\n  private buffers: Map<AudioStream, Int16Array>;\n  private streamIterators: Map<AudioStream, { next(): Promise<IteratorResult<AudioFrame>> }>;\n  private sampleRate: number;\n  private numChannels: number;\n  private chunkSize: number;\n  private streamTimeoutMs: number;\n  private queue: AsyncQueue<AudioFrame>;\n  private streamSignal: AsyncQueue<void>; // Signals when streams are added\n  private ending: boolean;\n  private mixerTask?: Promise<void>;\n  private closed: boolean;\n\n  /**\n   * Initialize the AudioMixer.\n   *\n   * @param sampleRate - The audio sample rate in Hz.\n   * @param numChannels - The number of audio channels.\n   * @param options - Optional configuration for the mixer.\n   */\n  constructor(sampleRate: number, numChannels: number, options: AudioMixerOptions = {}) {\n    this.streams = new Set();\n    this.buffers = new Map();\n    this.streamIterators = new Map();\n    this.sampleRate = sampleRate;\n    this.numChannels = numChannels;\n    this.chunkSize =\n      options.blocksize && options.blocksize > 0 ? options.blocksize : Math.floor(sampleRate / 10);\n    this.streamTimeoutMs = options.streamTimeoutMs ?? 100;\n    this.queue = new AsyncQueue<AudioFrame>(options.capacity ?? 100);\n    this.streamSignal = new AsyncQueue<void>(1); // there should only be one mixer\n    this.ending = false;\n    this.closed = false;\n\n    // Start the mixer task\n    this.mixerTask = this.mixer();\n  }\n\n  /**\n   * Add an audio stream to the mixer.\n   *\n   * The stream is added to the internal set of streams and an empty buffer is initialized for it,\n   * if not already present.\n   *\n   * @param stream - An async iterable that produces AudioFrame objects.\n   * @throws Error if the mixer has been closed.\n   */\n  addStream(stream: AudioStream): void {\n    if (this.ending) {\n      throw new Error('Cannot add stream after mixer has been closed');\n    }\n\n    this.streams.add(stream);\n    if (!this.buffers.has(stream)) {\n      this.buffers.set(stream, new Int16Array(0));\n    }\n\n    // Signal that a stream was added (non-blocking)\n    this.streamSignal.put(undefined).catch(() => {\n      // Ignore errors if signal queue is closed\n    });\n  }\n\n  /**\n   * Remove an audio stream from the mixer.\n   *\n   * This method removes the specified stream and its associated buffer from the mixer.\n   *\n   * @param stream - The audio stream to remove.\n   */\n  removeStream(stream: AudioStream): void {\n    this.streams.delete(stream);\n    this.buffers.delete(stream);\n    this.streamIterators.delete(stream);\n  }\n\n  /**\n   * Returns an async iterator for the mixed audio frames.\n   */\n  [Symbol.asyncIterator]() {\n    return {\n      next: async (): Promise<IteratorResult<AudioFrame>> => {\n        const frame = await this.getNextFrame();\n        if (frame === null) {\n          return { done: true, value: undefined };\n        }\n        return { done: false, value: frame };\n      },\n    };\n  }\n\n  /**\n   * Immediately stop mixing and close the mixer.\n   *\n   * This stops the mixing task, and any unconsumed output in the queue may be dropped.\n   */\n  async aclose(): Promise<void> {\n    if (this.closed) {\n      return;\n    }\n    this.closed = true;\n    this.ending = true;\n\n    // Close both queues to wake up any waiting operations\n    this.streamSignal.close();\n    this.queue.close();\n\n    await this.mixerTask;\n  }\n\n  /**\n   * Signal that no more streams will be added.\n   *\n   * This method marks the mixer as closed so that it flushes any remaining buffered output before ending.\n   * Note that existing streams will still be processed until exhausted.\n   */\n  endInput(): void {\n    this.ending = true;\n  }\n\n  private async getNextFrame(): Promise<AudioFrame | null> {\n    while (true) {\n      // Try to get an item from the queue (non-blocking)\n      const frame = this.queue.get();\n\n      if (frame !== undefined) {\n        return frame;\n      }\n\n      // Check if mixer is closed or ending\n      if (this.queue.closed || (this.ending && this.streams.size === 0)) {\n        return null;\n      }\n\n      // Queue is empty but mixer is still running - wait for an item to be added\n      await this.queue.waitForItem();\n    }\n  }\n\n  private async mixer(): Promise<void> {\n    // Main mixing loop that continuously processes streams and produces output frames\n    while (true) {\n      // If we're in ending mode and there are no more streams, exit\n      if (this.ending && this.streams.size === 0) {\n        break;\n      }\n\n      if (this.streams.size === 0) {\n        // Wait for a stream to be added (signal queue will have an item)\n        await this.streamSignal.waitForItem();\n        // Consume the signal\n        this.streamSignal.get();\n        continue;\n      }\n\n      // Process all streams in parallel\n      const streamArray = Array.from(this.streams);\n      const promises = streamArray.map((stream) => this.getContribution(stream));\n      const results = await Promise.all(\n        promises.map((p) =>\n          p\n            .then((value) => ({ status: 'fulfilled' as const, value }))\n            .catch((reason) => ({ status: 'rejected' as const, reason })),\n        ),\n      );\n\n      const contributions: Int16Array[] = [];\n      let anyData = false;\n      const removals: AudioStream[] = [];\n\n      for (const result of results) {\n        if (result.status !== 'fulfilled') {\n          log.warn('AudioMixer: Stream contribution failed:', result.reason);\n          continue;\n        }\n\n        const contrib = result.value;\n        contributions.push(contrib.data);\n        this.buffers.set(contrib.stream, contrib.buffer);\n\n        if (contrib.hadData) {\n          anyData = true;\n        }\n\n        // Mark exhausted streams with no remaining buffer for removal\n        if (contrib.exhausted && contrib.buffer.length === 0) {\n          removals.push(contrib.stream);\n        }\n      }\n\n      // Remove exhausted streams\n      for (const stream of removals) {\n        this.removeStream(stream);\n      }\n\n      // If all streams are exhausted, end the mixer automatically.\n      // This keeps `for await...of` consumers from hanging indefinitely when inputs complete.\n      if (!this.ending && removals.length > 0 && this.streams.size === 0) {\n        this.ending = true;\n      }\n\n      if (!anyData) {\n        // No data available from any stream, wait briefly before trying again\n        await this.sleep(1);\n        continue;\n      }\n\n      // Mix the audio data\n      const mixed = this.mixAudio(contributions);\n      const frame = new AudioFrame(mixed, this.sampleRate, this.numChannels, this.chunkSize);\n\n      if (this.closed) {\n        break;\n      }\n\n      try {\n        // Add mixed frame to output queue\n        await this.queue.put(frame);\n      } catch {\n        // Queue closed while trying to add frame\n        break;\n      }\n    }\n\n    // Close the queue to signal end of stream\n    this.queue.close();\n  }\n\n  private async getContribution(stream: AudioStream): Promise<Contribution> {\n    let buf = this.buffers.get(stream) ?? new Int16Array(0);\n    const initialBufferLength = buf.length;\n    let exhausted = false;\n    let receivedDataInThisCall = false;\n\n    // Get or create iterator for this stream\n    let iterator = this.streamIterators.get(stream);\n    if (!iterator) {\n      iterator = stream[Symbol.asyncIterator]();\n      this.streamIterators.set(stream, iterator);\n    }\n\n    // Accumulate data until we have at least chunkSize samples\n    while (buf.length < this.chunkSize * this.numChannels && !exhausted && !this.closed) {\n      try {\n        const result = await Promise.race([iterator.next(), this.timeout(this.streamTimeoutMs)]);\n\n        if (result === 'timeout') {\n          console.warn(`AudioMixer: stream timeout after ${this.streamTimeoutMs}ms`);\n          break;\n        }\n\n        if (result.done) {\n          exhausted = true;\n          break;\n        }\n\n        const frame = result.value;\n        const newData = frame.data;\n\n        // Mark that we received data in this call\n        receivedDataInThisCall = true;\n\n        // Concatenate buffers\n        if (buf.length === 0) {\n          buf = newData;\n        } else {\n          const combined = new Int16Array(buf.length + newData.length);\n          combined.set(buf);\n          combined.set(newData, buf.length);\n     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+{"version":3,"sources":["../src/audio_mixer.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2025 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { AsyncQueue } from './async_queue.js';\nimport { AudioFrame } from './audio_frame.js';\nimport { log } from './log.js';\n\n// Re-export AsyncQueue for backward compatibility\nexport { AsyncQueue } from './async_queue.js';\n\n// Define types for async iteration (since lib: es2015 doesn't include them)\ntype AudioStream = {\n  [Symbol.asyncIterator](): {\n    next(): Promise<IteratorResult<AudioFrame>>;\n  };\n};\n\ninterface Contribution {\n  stream: AudioStream;\n  data: Int16Array;\n  buffer: Int16Array;\n  hadData: boolean;\n  exhausted: boolean;\n}\n\nexport interface AudioMixerOptions {\n  /**\n   * The size of the audio block (in samples) for mixing.\n   * If not provided, defaults to sampleRate / 10 (100ms).\n   */\n  blocksize?: number;\n\n  /**\n   * The maximum wait time in milliseconds for each stream to provide\n   * audio data before timing out. Defaults to 100 ms.\n   */\n  streamTimeoutMs?: number;\n\n  /**\n   * The maximum number of mixed frames to store in the output queue.\n   * Defaults to 100.\n   */\n  capacity?: number;\n}\n\n/**\n * AudioMixer combines multiple async audio streams into a single output stream.\n *\n * The mixer accepts multiple async audio streams and mixes them into a single output stream.\n * Each output frame is generated with a fixed chunk size determined by the blocksize (in samples).\n * If blocksize is not provided (or 0), it defaults to 100ms.\n *\n * Each input stream is processed in parallel, accumulating audio data until at least one chunk\n * of samples is available. If an input stream does not provide data within the specified timeout,\n * a warning is logged. The mixer can be closed immediately\n * (dropping unconsumed frames) or allowed to flush remaining data using endInput().\n *\n * @example\n * ```typescript\n * const mixer = new AudioMixer(48000, 2);\n * mixer.addStream(stream1);\n * mixer.addStream(stream2);\n *\n * for await (const frame of mixer) {\n *   // Process mixed audio frame\n * }\n * ```\n */\nexport class AudioMixer {\n  private streams: Set<AudioStream>;\n  private buffers: Map<AudioStream, Int16Array>;\n  private streamIterators: Map<AudioStream, { next(): Promise<IteratorResult<AudioFrame>> }>;\n  private sampleRate: number;\n  private numChannels: number;\n  private chunkSize: number;\n  private streamTimeoutMs: number;\n  private queue: AsyncQueue<AudioFrame>;\n  private streamSignal: AsyncQueue<void>; // Signals when streams are added\n  private ending: boolean;\n  private mixerTask?: Promise<void>;\n  private closed: boolean;\n\n  /**\n   * Initialize the AudioMixer.\n   *\n   * @param sampleRate - The audio sample rate in Hz.\n   * @param numChannels - The number of audio channels.\n   * @param options - Optional configuration for the mixer.\n   */\n  constructor(sampleRate: number, numChannels: number, options: AudioMixerOptions = {}) {\n    this.streams = new Set();\n    this.buffers = new Map();\n    this.streamIterators = new Map();\n    this.sampleRate = sampleRate;\n    this.numChannels = numChannels;\n    this.chunkSize =\n      options.blocksize && options.blocksize > 0 ? options.blocksize : Math.floor(sampleRate / 10);\n    this.streamTimeoutMs = options.streamTimeoutMs ?? 100;\n    this.queue = new AsyncQueue<AudioFrame>(options.capacity ?? 100);\n    this.streamSignal = new AsyncQueue<void>(1); // there should only be one mixer\n    this.ending = false;\n    this.closed = false;\n\n    // Start the mixer task\n    this.mixerTask = this.mixer();\n  }\n\n  /**\n   * Add an audio stream to the mixer.\n   *\n   * The stream is added to the internal set of streams and an empty buffer is initialized for it,\n   * if not already present.\n   *\n   * @param stream - An async iterable that produces AudioFrame objects.\n   * @throws Error if the mixer has been closed.\n   */\n  addStream(stream: AudioStream): void {\n    if (this.ending) {\n      throw new Error('Cannot add stream after mixer has been closed');\n    }\n\n    this.streams.add(stream);\n    if (!this.buffers.has(stream)) {\n      this.buffers.set(stream, new Int16Array(0));\n    }\n\n    // Signal that a stream was added (non-blocking)\n    this.streamSignal.put(undefined).catch(() => {\n      // Ignore errors if signal queue is closed\n    });\n  }\n\n  /**\n   * Remove an audio stream from the mixer.\n   *\n   * This method removes the specified stream and its associated buffer from the mixer.\n   *\n   * @param stream - The audio stream to remove.\n   */\n  removeStream(stream: AudioStream): void {\n    this.streams.delete(stream);\n    this.buffers.delete(stream);\n    this.streamIterators.delete(stream);\n  }\n\n  /**\n   * Returns an async iterator for the mixed audio frames.\n   */\n  [Symbol.asyncIterator]() {\n    return {\n      next: async (): Promise<IteratorResult<AudioFrame>> => {\n        const frame = await this.getNextFrame();\n        if (frame === null) {\n          return { done: true, value: undefined };\n        }\n        return { done: false, value: frame };\n      },\n    };\n  }\n\n  /**\n   * Immediately stop mixing and close the mixer.\n   *\n   * This stops the mixing task, and any unconsumed output in the queue may be dropped.\n   */\n  async aclose(): Promise<void> {\n    if (this.closed) {\n      return;\n    }\n    this.closed = true;\n    this.ending = true;\n\n    // Close both queues to wake up any waiting operations\n    this.streamSignal.close();\n    this.queue.close();\n\n    await this.mixerTask;\n  }\n\n  /**\n   * Signal that no more streams will be added.\n   *\n   * This method marks the mixer as closed so that it flushes any remaining buffered output before ending.\n   * Note that existing streams will still be processed until exhausted.\n   */\n  endInput(): void {\n    this.ending = true;\n  }\n\n  private async getNextFrame(): Promise<AudioFrame | null> {\n    while (true) {\n      // Try to get an item from the queue (non-blocking)\n      const frame = this.queue.get();\n\n      if (frame !== undefined) {\n        return frame;\n      }\n\n      // Check if mixer is closed or ending\n      if (this.queue.closed || (this.ending && this.streams.size === 0)) {\n        return null;\n      }\n\n      // Queue is empty but mixer is still running - wait for an item to be added\n      await this.queue.waitForItem();\n    }\n  }\n\n  private async mixer(): Promise<void> {\n    // Main mixing loop that continuously processes streams and produces output frames\n    while (true) {\n      // If we're in ending mode and there are no more streams, exit\n      if (this.ending && this.streams.size === 0) {\n        break;\n      }\n\n      if (this.streams.size === 0) {\n        // Wait for a stream to be added (signal queue will have an item)\n        await this.streamSignal.waitForItem();\n        // Consume the signal\n        this.streamSignal.get();\n        continue;\n      }\n\n      // Process all streams in parallel\n      const streamArray = Array.from(this.streams);\n      const promises = streamArray.map((stream) => this.getContribution(stream));\n      const results = await Promise.all(\n        promises.map((p) =>\n          p\n            .then((value) => ({ status: 'fulfilled' as const, value }))\n            .catch((reason) => ({ status: 'rejected' as const, reason })),\n        ),\n      );\n\n      const contributions: Int16Array[] = [];\n      let anyData = false;\n      const removals: AudioStream[] = [];\n\n      for (const result of results) {\n        if (result.status !== 'fulfilled') {\n          log.warn({ reason: result.reason }, 'AudioMixer: Stream contribution failed');\n          continue;\n        }\n\n        const contrib = result.value;\n        contributions.push(contrib.data);\n        this.buffers.set(contrib.stream, contrib.buffer);\n\n        if (contrib.hadData) {\n          anyData = true;\n        }\n\n        // Mark exhausted streams with no remaining buffer for removal\n        if (contrib.exhausted && contrib.buffer.length === 0) {\n          removals.push(contrib.stream);\n        }\n      }\n\n      // Remove exhausted streams\n      for (const stream of removals) {\n        this.removeStream(stream);\n      }\n\n      // If all streams are exhausted, end the mixer automatically.\n      // This keeps `for await...of` consumers from hanging indefinitely when inputs complete.\n      if (!this.ending && removals.length > 0 && this.streams.size === 0) {\n        this.ending = true;\n      }\n\n      if (!anyData) {\n        // No data available from any stream, wait briefly before trying again\n        await this.sleep(1);\n        continue;\n      }\n\n      // Mix the audio data\n      const mixed = this.mixAudio(contributions);\n      const frame = new AudioFrame(mixed, this.sampleRate, this.numChannels, this.chunkSize);\n\n      if (this.closed) {\n        break;\n      }\n\n      try {\n        // Add mixed frame to output queue\n        await this.queue.put(frame);\n      } catch {\n        // Queue closed while trying to add frame\n        break;\n      }\n    }\n\n    // Close the queue to signal end of stream\n    this.queue.close();\n  }\n\n  private async getContribution(stream: AudioStream): Promise<Contribution> {\n    let buf = this.buffers.get(stream) ?? new Int16Array(0);\n    const initialBufferLength = buf.length;\n    let exhausted = false;\n    let receivedDataInThisCall = false;\n\n    // Get or create iterator for this stream\n    let iterator = this.streamIterators.get(stream);\n    if (!iterator) {\n      iterator = stream[Symbol.asyncIterator]();\n      this.streamIterators.set(stream, iterator);\n    }\n\n    // Accumulate data until we have at least chunkSize samples\n    while (buf.length < this.chunkSize * this.numChannels && !exhausted && !this.closed) {\n      try {\n        const result = await Promise.race([iterator.next(), this.timeout(this.streamTimeoutMs)]);\n\n        if (result === 'timeout') {\n          console.warn(`AudioMixer: stream timeout after ${this.streamTimeoutMs}ms`);\n          break;\n        }\n\n        if (result.done) {\n          exhausted = true;\n          break;\n        }\n\n        const frame = result.value;\n        const newData = frame.data;\n\n        // Mark that we received data in this call\n        receivedDataInThisCall = true;\n\n        // Concatenate buffers\n        if (buf.length === 0) {\n          buf = newData;\n        } else {\n          const combined = new Int16Array(buf.length + newData.length);\n          combined.set(buf);\n          combined.set(newData, buf.length);\n     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    buffer: buf,\n      hadData,\n      exhausted,\n    };\n  }\n\n  private mixAudio(contributions: Int16Array[]): Int16Array {\n    if (contributions.length === 0) {\n      return new Int16Array(this.chunkSize * this.numChannels);\n    }\n\n    const length = this.chunkSize * this.numChannels;\n    // Use a wider accumulator to avoid int16 overflow while summing.\n    const acc = new Int32Array(length);\n\n    // Sum all contributions\n    for (const contrib of contributions) {\n      for (let i = 0; i < length; i++) {\n        const val = contrib[i];\n        if (val !== undefined) {\n          acc[i] = (acc[i] ?? 0) + val;\n        }\n      }\n    }\n\n    // Clip to Int16 range\n    const mixed = new Int16Array(length);\n    for (let i = 0; i < length; i++) {\n      const val = acc[i] ?? 0;\n      if (val > 32767) {\n        mixed[i] = 32767;\n      } else if (val < -32768) {\n        mixed[i] = -32768;\n      } else {\n        mixed[i] = val;\n      }\n    }\n\n    return 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package/dist/audio_mixer.d.cts

@@ -1,11 +1,6 @@
 import { AudioFrame } from './audio_frame.cjs';
 export { AsyncQueue } from './async_queue.cjs';
-import './proto/audio_frame_pb.cjs';
-import '@bufbuild/protobuf';
-import './proto/track_pb.cjs';
-import './proto/stats_pb.cjs';
-import './proto/e2ee_pb.cjs';
-import './proto/handle_pb.cjs';
+import '@livekit/rtc-ffi-bindings';
 
 type AudioStream = {
     [Symbol.asyncIterator](): {

package/dist/audio_mixer.d.ts

@@ -1,11 +1,6 @@
 import { AudioFrame } from './audio_frame.js';
 export { AsyncQueue } from './async_queue.js';
-import './proto/audio_frame_pb.js';
-import '@bufbuild/protobuf';
-import './proto/track_pb.js';
-import './proto/stats_pb.js';
-import './proto/e2ee_pb.js';
-import './proto/handle_pb.js';
+import '@livekit/rtc-ffi-bindings';
 
 type AudioStream = {
     [Symbol.asyncIterator](): {

package/dist/audio_mixer.js

@@ -128,7 +128,7 @@ class AudioMixer {
       const removals = [];
       for (const result of results) {
         if (result.status !== "fulfilled") {
-          log.warn("AudioMixer: Stream contribution failed:", result.reason);
+          log.warn({ reason: result.reason }, "AudioMixer: Stream contribution failed");
           continue;
         }
         const contrib = result.value;

package/dist/audio_mixer.js.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/audio_mixer.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2025 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { AsyncQueue } from './async_queue.js';\nimport { AudioFrame } from './audio_frame.js';\nimport { log } from './log.js';\n\n// Re-export AsyncQueue for backward compatibility\nexport { AsyncQueue } from './async_queue.js';\n\n// Define types for async iteration (since lib: es2015 doesn't include them)\ntype AudioStream = {\n  [Symbol.asyncIterator](): {\n    next(): Promise<IteratorResult<AudioFrame>>;\n  };\n};\n\ninterface Contribution {\n  stream: AudioStream;\n  data: Int16Array;\n  buffer: Int16Array;\n  hadData: boolean;\n  exhausted: boolean;\n}\n\nexport interface AudioMixerOptions {\n  /**\n   * The size of the audio block (in samples) for mixing.\n   * If not provided, defaults to sampleRate / 10 (100ms).\n   */\n  blocksize?: number;\n\n  /**\n   * The maximum wait time in milliseconds for each stream to provide\n   * audio data before timing out. Defaults to 100 ms.\n   */\n  streamTimeoutMs?: number;\n\n  /**\n   * The maximum number of mixed frames to store in the output queue.\n   * Defaults to 100.\n   */\n  capacity?: number;\n}\n\n/**\n * AudioMixer combines multiple async audio streams into a single output stream.\n *\n * The mixer accepts multiple async audio streams and mixes them into a single output stream.\n * Each output frame is generated with a fixed chunk size determined by the blocksize (in samples).\n * If blocksize is not provided (or 0), it defaults to 100ms.\n *\n * Each input stream is processed in parallel, accumulating audio data until at least one chunk\n * of samples is available. If an input stream does not provide data within the specified timeout,\n * a warning is logged. The mixer can be closed immediately\n * (dropping unconsumed frames) or allowed to flush remaining data using endInput().\n *\n * @example\n * ```typescript\n * const mixer = new AudioMixer(48000, 2);\n * mixer.addStream(stream1);\n * mixer.addStream(stream2);\n *\n * for await (const frame of mixer) {\n *   // Process mixed audio frame\n * }\n * ```\n */\nexport class AudioMixer {\n  private streams: Set<AudioStream>;\n  private buffers: Map<AudioStream, Int16Array>;\n  private streamIterators: Map<AudioStream, { next(): Promise<IteratorResult<AudioFrame>> }>;\n  private sampleRate: number;\n  private numChannels: number;\n  private chunkSize: number;\n  private streamTimeoutMs: number;\n  private queue: AsyncQueue<AudioFrame>;\n  private streamSignal: AsyncQueue<void>; // Signals when streams are added\n  private ending: boolean;\n  private mixerTask?: Promise<void>;\n  private closed: boolean;\n\n  /**\n   * Initialize the AudioMixer.\n   *\n   * @param sampleRate - The audio sample rate in Hz.\n   * @param numChannels - The number of audio channels.\n   * @param options - Optional configuration for the mixer.\n   */\n  constructor(sampleRate: number, numChannels: number, options: AudioMixerOptions = {}) {\n    this.streams = new Set();\n    this.buffers = new Map();\n    this.streamIterators = new Map();\n    this.sampleRate = sampleRate;\n    this.numChannels = numChannels;\n    this.chunkSize =\n      options.blocksize && options.blocksize > 0 ? options.blocksize : Math.floor(sampleRate / 10);\n    this.streamTimeoutMs = options.streamTimeoutMs ?? 100;\n    this.queue = new AsyncQueue<AudioFrame>(options.capacity ?? 100);\n    this.streamSignal = new AsyncQueue<void>(1); // there should only be one mixer\n    this.ending = false;\n    this.closed = false;\n\n    // Start the mixer task\n    this.mixerTask = this.mixer();\n  }\n\n  /**\n   * Add an audio stream to the mixer.\n   *\n   * The stream is added to the internal set of streams and an empty buffer is initialized for it,\n   * if not already present.\n   *\n   * @param stream - An async iterable that produces AudioFrame objects.\n   * @throws Error if the mixer has been closed.\n   */\n  addStream(stream: AudioStream): void {\n    if (this.ending) {\n      throw new Error('Cannot add stream after mixer has been closed');\n    }\n\n    this.streams.add(stream);\n    if (!this.buffers.has(stream)) {\n      this.buffers.set(stream, new Int16Array(0));\n    }\n\n    // Signal that a stream was added (non-blocking)\n    this.streamSignal.put(undefined).catch(() => {\n      // Ignore errors if signal queue is closed\n    });\n  }\n\n  /**\n   * Remove an audio stream from the mixer.\n   *\n   * This method removes the specified stream and its associated buffer from the mixer.\n   *\n   * @param stream - The audio stream to remove.\n   */\n  removeStream(stream: AudioStream): void {\n    this.streams.delete(stream);\n    this.buffers.delete(stream);\n    this.streamIterators.delete(stream);\n  }\n\n  /**\n   * Returns an async iterator for the mixed audio frames.\n   */\n  [Symbol.asyncIterator]() {\n    return {\n      next: async (): Promise<IteratorResult<AudioFrame>> => {\n        const frame = await this.getNextFrame();\n        if (frame === null) {\n          return { done: true, value: undefined };\n        }\n        return { done: false, value: frame };\n      },\n    };\n  }\n\n  /**\n   * Immediately stop mixing and close the mixer.\n   *\n   * This stops the mixing task, and any unconsumed output in the queue may be dropped.\n   */\n  async aclose(): Promise<void> {\n    if (this.closed) {\n      return;\n    }\n    this.closed = true;\n    this.ending = true;\n\n    // Close both queues to wake up any waiting operations\n    this.streamSignal.close();\n    this.queue.close();\n\n    await this.mixerTask;\n  }\n\n  /**\n   * Signal that no more streams will be added.\n   *\n   * This method marks the mixer as closed so that it flushes any remaining buffered output before ending.\n   * Note that existing streams will still be processed until exhausted.\n   */\n  endInput(): void {\n    this.ending = true;\n  }\n\n  private async getNextFrame(): Promise<AudioFrame | null> {\n    while (true) {\n      // Try to get an item from the queue (non-blocking)\n      const frame = this.queue.get();\n\n      if (frame !== undefined) {\n        return frame;\n      }\n\n      // Check if mixer is closed or ending\n      if (this.queue.closed || (this.ending && this.streams.size === 0)) {\n        return null;\n      }\n\n      // Queue is empty but mixer is still running - wait for an item to be added\n      await this.queue.waitForItem();\n    }\n  }\n\n  private async mixer(): Promise<void> {\n    // Main mixing loop that continuously processes streams and produces output frames\n    while (true) {\n      // If we're in ending mode and there are no more streams, exit\n      if (this.ending && this.streams.size === 0) {\n        break;\n      }\n\n      if (this.streams.size === 0) {\n        // Wait for a stream to be added (signal queue will have an item)\n        await this.streamSignal.waitForItem();\n        // Consume the signal\n        this.streamSignal.get();\n        continue;\n      }\n\n      // Process all streams in parallel\n      const streamArray = Array.from(this.streams);\n      const promises = streamArray.map((stream) => this.getContribution(stream));\n      const results = await Promise.all(\n        promises.map((p) =>\n          p\n            .then((value) => ({ status: 'fulfilled' as const, value }))\n            .catch((reason) => ({ status: 'rejected' as const, reason })),\n        ),\n      );\n\n      const contributions: Int16Array[] = [];\n      let anyData = false;\n      const removals: AudioStream[] = [];\n\n      for (const result of results) {\n        if (result.status !== 'fulfilled') {\n          log.warn('AudioMixer: Stream contribution failed:', result.reason);\n          continue;\n        }\n\n        const contrib = result.value;\n        contributions.push(contrib.data);\n        this.buffers.set(contrib.stream, contrib.buffer);\n\n        if (contrib.hadData) {\n          anyData = true;\n        }\n\n        // Mark exhausted streams with no remaining buffer for removal\n        if (contrib.exhausted && contrib.buffer.length === 0) {\n          removals.push(contrib.stream);\n        }\n      }\n\n      // Remove exhausted streams\n      for (const stream of removals) {\n        this.removeStream(stream);\n      }\n\n      // If all streams are exhausted, end the mixer automatically.\n      // This keeps `for await...of` consumers from hanging indefinitely when inputs complete.\n      if (!this.ending && removals.length > 0 && this.streams.size === 0) {\n        this.ending = true;\n      }\n\n      if (!anyData) {\n        // No data available from any stream, wait briefly before trying again\n        await this.sleep(1);\n        continue;\n      }\n\n      // Mix the audio data\n      const mixed = this.mixAudio(contributions);\n      const frame = new AudioFrame(mixed, this.sampleRate, this.numChannels, this.chunkSize);\n\n      if (this.closed) {\n        break;\n      }\n\n      try {\n        // Add mixed frame to output queue\n        await this.queue.put(frame);\n      } catch {\n        // Queue closed while trying to add frame\n        break;\n      }\n    }\n\n    // Close the queue to signal end of stream\n    this.queue.close();\n  }\n\n  private async getContribution(stream: AudioStream): Promise<Contribution> {\n    let buf = this.buffers.get(stream) ?? new Int16Array(0);\n    const initialBufferLength = buf.length;\n    let exhausted = false;\n    let receivedDataInThisCall = false;\n\n    // Get or create iterator for this stream\n    let iterator = this.streamIterators.get(stream);\n    if (!iterator) {\n      iterator = stream[Symbol.asyncIterator]();\n      this.streamIterators.set(stream, iterator);\n    }\n\n    // Accumulate data until we have at least chunkSize samples\n    while (buf.length < this.chunkSize * this.numChannels && !exhausted && !this.closed) {\n      try {\n        const result = await Promise.race([iterator.next(), this.timeout(this.streamTimeoutMs)]);\n\n        if (result === 'timeout') {\n          console.warn(`AudioMixer: stream timeout after ${this.streamTimeoutMs}ms`);\n          break;\n        }\n\n        if (result.done) {\n          exhausted = true;\n          break;\n        }\n\n        const frame = result.value;\n        const newData = frame.data;\n\n        // Mark that we received data in this call\n        receivedDataInThisCall = true;\n\n        // Concatenate buffers\n        if (buf.length === 0) {\n          buf = newData;\n        } else {\n          const combined = new Int16Array(buf.length + newData.length);\n          combined.set(buf);\n          combined.set(newData, buf.length);\n     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+{"version":3,"sources":["../src/audio_mixer.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2025 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { AsyncQueue } from './async_queue.js';\nimport { AudioFrame } from './audio_frame.js';\nimport { log } from './log.js';\n\n// Re-export AsyncQueue for backward compatibility\nexport { AsyncQueue } from './async_queue.js';\n\n// Define types for async iteration (since lib: es2015 doesn't include them)\ntype AudioStream = {\n  [Symbol.asyncIterator](): {\n    next(): Promise<IteratorResult<AudioFrame>>;\n  };\n};\n\ninterface Contribution {\n  stream: AudioStream;\n  data: Int16Array;\n  buffer: Int16Array;\n  hadData: boolean;\n  exhausted: boolean;\n}\n\nexport interface AudioMixerOptions {\n  /**\n   * The size of the audio block (in samples) for mixing.\n   * If not provided, defaults to sampleRate / 10 (100ms).\n   */\n  blocksize?: number;\n\n  /**\n   * The maximum wait time in milliseconds for each stream to provide\n   * audio data before timing out. Defaults to 100 ms.\n   */\n  streamTimeoutMs?: number;\n\n  /**\n   * The maximum number of mixed frames to store in the output queue.\n   * Defaults to 100.\n   */\n  capacity?: number;\n}\n\n/**\n * AudioMixer combines multiple async audio streams into a single output stream.\n *\n * The mixer accepts multiple async audio streams and mixes them into a single output stream.\n * Each output frame is generated with a fixed chunk size determined by the blocksize (in samples).\n * If blocksize is not provided (or 0), it defaults to 100ms.\n *\n * Each input stream is processed in parallel, accumulating audio data until at least one chunk\n * of samples is available. If an input stream does not provide data within the specified timeout,\n * a warning is logged. The mixer can be closed immediately\n * (dropping unconsumed frames) or allowed to flush remaining data using endInput().\n *\n * @example\n * ```typescript\n * const mixer = new AudioMixer(48000, 2);\n * mixer.addStream(stream1);\n * mixer.addStream(stream2);\n *\n * for await (const frame of mixer) {\n *   // Process mixed audio frame\n * }\n * ```\n */\nexport class AudioMixer {\n  private streams: Set<AudioStream>;\n  private buffers: Map<AudioStream, Int16Array>;\n  private streamIterators: Map<AudioStream, { next(): Promise<IteratorResult<AudioFrame>> }>;\n  private sampleRate: number;\n  private numChannels: number;\n  private chunkSize: number;\n  private streamTimeoutMs: number;\n  private queue: AsyncQueue<AudioFrame>;\n  private streamSignal: AsyncQueue<void>; // Signals when streams are added\n  private ending: boolean;\n  private mixerTask?: Promise<void>;\n  private closed: boolean;\n\n  /**\n   * Initialize the AudioMixer.\n   *\n   * @param sampleRate - The audio sample rate in Hz.\n   * @param numChannels - The number of audio channels.\n   * @param options - Optional configuration for the mixer.\n   */\n  constructor(sampleRate: number, numChannels: number, options: AudioMixerOptions = {}) {\n    this.streams = new Set();\n    this.buffers = new Map();\n    this.streamIterators = new Map();\n    this.sampleRate = sampleRate;\n    this.numChannels = numChannels;\n    this.chunkSize =\n      options.blocksize && options.blocksize > 0 ? options.blocksize : Math.floor(sampleRate / 10);\n    this.streamTimeoutMs = options.streamTimeoutMs ?? 100;\n    this.queue = new AsyncQueue<AudioFrame>(options.capacity ?? 100);\n    this.streamSignal = new AsyncQueue<void>(1); // there should only be one mixer\n    this.ending = false;\n    this.closed = false;\n\n    // Start the mixer task\n    this.mixerTask = this.mixer();\n  }\n\n  /**\n   * Add an audio stream to the mixer.\n   *\n   * The stream is added to the internal set of streams and an empty buffer is initialized for it,\n   * if not already present.\n   *\n   * @param stream - An async iterable that produces AudioFrame objects.\n   * @throws Error if the mixer has been closed.\n   */\n  addStream(stream: AudioStream): void {\n    if (this.ending) {\n      throw new Error('Cannot add stream after mixer has been closed');\n    }\n\n    this.streams.add(stream);\n    if (!this.buffers.has(stream)) {\n      this.buffers.set(stream, new Int16Array(0));\n    }\n\n    // Signal that a stream was added (non-blocking)\n    this.streamSignal.put(undefined).catch(() => {\n      // Ignore errors if signal queue is closed\n    });\n  }\n\n  /**\n   * Remove an audio stream from the mixer.\n   *\n   * This method removes the specified stream and its associated buffer from the mixer.\n   *\n   * @param stream - The audio stream to remove.\n   */\n  removeStream(stream: AudioStream): void {\n    this.streams.delete(stream);\n    this.buffers.delete(stream);\n    this.streamIterators.delete(stream);\n  }\n\n  /**\n   * Returns an async iterator for the mixed audio frames.\n   */\n  [Symbol.asyncIterator]() {\n    return {\n      next: async (): Promise<IteratorResult<AudioFrame>> => {\n        const frame = await this.getNextFrame();\n        if (frame === null) {\n          return { done: true, value: undefined };\n        }\n        return { done: false, value: frame };\n      },\n    };\n  }\n\n  /**\n   * Immediately stop mixing and close the mixer.\n   *\n   * This stops the mixing task, and any unconsumed output in the queue may be dropped.\n   */\n  async aclose(): Promise<void> {\n    if (this.closed) {\n      return;\n    }\n    this.closed = true;\n    this.ending = true;\n\n    // Close both queues to wake up any waiting operations\n    this.streamSignal.close();\n    this.queue.close();\n\n    await this.mixerTask;\n  }\n\n  /**\n   * Signal that no more streams will be added.\n   *\n   * This method marks the mixer as closed so that it flushes any remaining buffered output before ending.\n   * Note that existing streams will still be processed until exhausted.\n   */\n  endInput(): void {\n    this.ending = true;\n  }\n\n  private async getNextFrame(): Promise<AudioFrame | null> {\n    while (true) {\n      // Try to get an item from the queue (non-blocking)\n      const frame = this.queue.get();\n\n      if (frame !== undefined) {\n        return frame;\n      }\n\n      // Check if mixer is closed or ending\n      if (this.queue.closed || (this.ending && this.streams.size === 0)) {\n        return null;\n      }\n\n      // Queue is empty but mixer is still running - wait for an item to be added\n      await this.queue.waitForItem();\n    }\n  }\n\n  private async mixer(): Promise<void> {\n    // Main mixing loop that continuously processes streams and produces output frames\n    while (true) {\n      // If we're in ending mode and there are no more streams, exit\n      if (this.ending && this.streams.size === 0) {\n        break;\n      }\n\n      if (this.streams.size === 0) {\n        // Wait for a stream to be added (signal queue will have an item)\n        await this.streamSignal.waitForItem();\n        // Consume the signal\n        this.streamSignal.get();\n        continue;\n      }\n\n      // Process all streams in parallel\n      const streamArray = Array.from(this.streams);\n      const promises = streamArray.map((stream) => this.getContribution(stream));\n      const results = await Promise.all(\n        promises.map((p) =>\n          p\n            .then((value) => ({ status: 'fulfilled' as const, value }))\n            .catch((reason) => ({ status: 'rejected' as const, reason })),\n        ),\n      );\n\n      const contributions: Int16Array[] = [];\n      let anyData = false;\n      const removals: AudioStream[] = [];\n\n      for (const result of results) {\n        if (result.status !== 'fulfilled') {\n          log.warn({ reason: result.reason }, 'AudioMixer: Stream contribution failed');\n          continue;\n        }\n\n        const contrib = result.value;\n        contributions.push(contrib.data);\n        this.buffers.set(contrib.stream, contrib.buffer);\n\n        if (contrib.hadData) {\n          anyData = true;\n        }\n\n        // Mark exhausted streams with no remaining buffer for removal\n        if (contrib.exhausted && contrib.buffer.length === 0) {\n          removals.push(contrib.stream);\n        }\n      }\n\n      // Remove exhausted streams\n      for (const stream of removals) {\n        this.removeStream(stream);\n      }\n\n      // If all streams are exhausted, end the mixer automatically.\n      // This keeps `for await...of` consumers from hanging indefinitely when inputs complete.\n      if (!this.ending && removals.length > 0 && this.streams.size === 0) {\n        this.ending = true;\n      }\n\n      if (!anyData) {\n        // No data available from any stream, wait briefly before trying again\n        await this.sleep(1);\n        continue;\n      }\n\n      // Mix the audio data\n      const mixed = this.mixAudio(contributions);\n      const frame = new AudioFrame(mixed, this.sampleRate, this.numChannels, this.chunkSize);\n\n      if (this.closed) {\n        break;\n      }\n\n      try {\n        // Add mixed frame to output queue\n        await this.queue.put(frame);\n      } catch {\n        // Queue closed while trying to add frame\n        break;\n      }\n    }\n\n    // Close the queue to signal end of stream\n    this.queue.close();\n  }\n\n  private async getContribution(stream: AudioStream): Promise<Contribution> {\n    let buf = this.buffers.get(stream) ?? new Int16Array(0);\n    const initialBufferLength = buf.length;\n    let exhausted = false;\n    let receivedDataInThisCall = false;\n\n    // Get or create iterator for this stream\n    let iterator = this.streamIterators.get(stream);\n    if (!iterator) {\n      iterator = stream[Symbol.asyncIterator]();\n      this.streamIterators.set(stream, iterator);\n    }\n\n    // Accumulate data until we have at least chunkSize samples\n    while (buf.length < this.chunkSize * this.numChannels && !exhausted && !this.closed) {\n      try {\n        const result = await Promise.race([iterator.next(), this.timeout(this.streamTimeoutMs)]);\n\n        if (result === 'timeout') {\n          console.warn(`AudioMixer: stream timeout after ${this.streamTimeoutMs}ms`);\n          break;\n        }\n\n        if (result.done) {\n          exhausted = true;\n          break;\n        }\n\n        const frame = result.value;\n        const newData = frame.data;\n\n        // Mark that we received data in this call\n        receivedDataInThisCall = true;\n\n        // Concatenate buffers\n        if (buf.length === 0) {\n          buf = newData;\n        } else {\n          const combined = new Int16Array(buf.length + newData.length);\n          combined.set(buf);\n          combined.set(newData, buf.length);\n          buf = combined;\n        }\n      } catch (error) {\n        console.error(`AudioMixer: Error reading from stream:`, error);\n        exhausted = true;\n        break;\n      }\n    }\n\n    // Extract contribution and update buffer\n    let contrib: Int16Array;\n    const samplesNeeded = this.chunkSize * this.numChannels;\n\n    if (buf.length >= samplesNeeded) {\n      // Extract the needed samples and keep the remainder in the buffer\n      contrib = buf.subarray(0, samplesNeeded);\n      buf = buf.subarray(samplesNeeded);\n    } else {\n      // Pad with zeros if we don't have enough data\n      const padded = new Int16Array(samplesNeeded);\n      padded.set(buf);\n      contrib = padded;\n      buf = new Int16Array(0);\n    }\n\n    // hadData means: we had data at start OR we received data during this call OR we have data remaining\n    const hadData = initialBufferLength > 0 || receivedDataInThisCall || buf.length > 0;\n\n    return {\n      stream,\n      data: contrib,\n      buffer: buf,\n      hadData,\n      exhausted,\n    };\n  }\n\n  private mixAudio(contributions: Int16Array[]): Int16Array {\n    if (contributions.length === 0) {\n      return new Int16Array(this.chunkSize * this.numChannels);\n    }\n\n    const length = this.chunkSize * this.numChannels;\n    // Use a wider accumulator to avoid int16 overflow while summing.\n    const acc = new Int32Array(length);\n\n    // Sum all contributions\n    for (const contrib of contributions) {\n      for (let i = 0; i < length; i++) {\n        const val = contrib[i];\n        if (val !== undefined) {\n          acc[i] = (acc[i] ?? 0) + val;\n        }\n      }\n    }\n\n    // Clip to Int16 range\n    const mixed = new Int16Array(length);\n    for (let i = 0; i < length; i++) {\n      const val = acc[i] ?? 0;\n      if (val > 32767) {\n        mixed[i] = 32767;\n      } else if (val < -32768) {\n        mixed[i] = -32768;\n      } else {\n        mixed[i] = val;\n      }\n    }\n\n    return mixed;\n  }\n\n  private sleep(ms: number): Promise<void> {\n    return new Promise((resolve) => setTimeout(resolve, ms));\n  }\n\n  private timeout(ms: number): Promise<'timeout'> {\n    return new Promise((resolve) => setTimeout(() => resolve('timeout'), ms));\n  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package/dist/audio_resampler.cjs

@@ -22,15 +22,15 @@ __export(audio_resampler_exports, {
   AudioResamplerQuality: () => AudioResamplerQuality
 });
 module.exports = __toCommonJS(audio_resampler_exports);
+var import_rtc_ffi_bindings = require("@livekit/rtc-ffi-bindings");
 var import_audio_frame = require("./audio_frame.cjs");
 var import_ffi_client = require("./ffi_client.cjs");
-var import_audio_frame_pb = require("./proto/audio_frame_pb.cjs");
 var AudioResamplerQuality = ((AudioResamplerQuality2) => {
-  AudioResamplerQuality2[AudioResamplerQuality2["QUICK"] = import_audio_frame_pb.SoxQualityRecipe.SOXR_QUALITY_QUICK] = "QUICK";
-  AudioResamplerQuality2[AudioResamplerQuality2["LOW"] = import_audio_frame_pb.SoxQualityRecipe.SOXR_QUALITY_LOW] = "LOW";
-  AudioResamplerQuality2[AudioResamplerQuality2["MEDIUM"] = import_audio_frame_pb.SoxQualityRecipe.SOXR_QUALITY_MEDIUM] = "MEDIUM";
-  AudioResamplerQuality2[AudioResamplerQuality2["HIGH"] = import_audio_frame_pb.SoxQualityRecipe.SOXR_QUALITY_HIGH] = "HIGH";
-  AudioResamplerQuality2[AudioResamplerQuality2["VERY_HIGH"] = import_audio_frame_pb.SoxQualityRecipe.SOXR_QUALITY_VERYHIGH] = "VERY_HIGH";
+  AudioResamplerQuality2[AudioResamplerQuality2["QUICK"] = import_rtc_ffi_bindings.SoxQualityRecipe.SOXR_QUALITY_QUICK] = "QUICK";
+  AudioResamplerQuality2[AudioResamplerQuality2["LOW"] = import_rtc_ffi_bindings.SoxQualityRecipe.SOXR_QUALITY_LOW] = "LOW";
+  AudioResamplerQuality2[AudioResamplerQuality2["MEDIUM"] = import_rtc_ffi_bindings.SoxQualityRecipe.SOXR_QUALITY_MEDIUM] = "MEDIUM";
+  AudioResamplerQuality2[AudioResamplerQuality2["HIGH"] = import_rtc_ffi_bindings.SoxQualityRecipe.SOXR_QUALITY_HIGH] = "HIGH";
+  AudioResamplerQuality2[AudioResamplerQuality2["VERY_HIGH"] = import_rtc_ffi_bindings.SoxQualityRecipe.SOXR_QUALITY_VERYHIGH] = "VERY_HIGH";
   return AudioResamplerQuality2;
 })(AudioResamplerQuality || {});
 class AudioResampler {
@@ -51,13 +51,13 @@ class AudioResampler {
     this.#inputRate = inputRate;
     this.#outputRate = outputRate;
     this.#channels = channels;
-    const req = new import_audio_frame_pb.NewSoxResamplerRequest({
+    const req = new import_rtc_ffi_bindings.NewSoxResamplerRequest({
       inputRate,
       outputRate,
       numChannels: channels,
       qualityRecipe: quality,
-      inputDataType: import_audio_frame_pb.SoxResamplerDataType.SOXR_DATATYPE_INT16I,
-      outputDataType: import_audio_frame_pb.SoxResamplerDataType.SOXR_DATATYPE_INT16I,
+      inputDataType: import_rtc_ffi_bindings.SoxResamplerDataType.SOXR_DATATYPE_INT16I,
+      outputDataType: import_rtc_ffi_bindings.SoxResamplerDataType.SOXR_DATATYPE_INT16I,
       flags: 0
     });
     const res = import_ffi_client.FfiClient.instance.request({
@@ -84,6 +84,13 @@ class AudioResampler {
   get channels() {
     return this.#channels;
   }
+  /**
+   * Releases the underlying native resampler handle. Must be called when
+   * the resampler is no longer needed to avoid leaking the FD.
+   */
+  close() {
+    this.#ffiHandle.dispose();
+  }
   /**
    * Push audio data into the resampler and retrieve any available resampled data.
    *
@@ -96,7 +103,7 @@ class AudioResampler {
    * be empty if no output data is available yet.
    */
   push(data) {
-    const req = new import_audio_frame_pb.PushSoxResamplerRequest({
+    const req = new import_rtc_ffi_bindings.PushSoxResamplerRequest({
       resamplerHandle: this.#ffiHandle.handle,
       dataPtr: data.protoInfo().dataPtr,
       size: data.data.byteLength
@@ -119,7 +126,8 @@ class AudioResampler {
         new Int16Array(outputData.buffer),
         this.#outputRate,
         this.#channels,
-        Math.trunc(outputData.length / this.#channels / 2)
+        Math.trunc(outputData.length / this.#channels / 2),
+        data.userdata
       )
     ];
   }
@@ -131,7 +139,7 @@ class AudioResampler {
    * internal buffers are processed and all resampled data is output.
    */
   flush() {
-    const req = new import_audio_frame_pb.FlushSoxResamplerRequest({
+    const req = new import_rtc_ffi_bindings.FlushSoxResamplerRequest({
       resamplerHandle: this.#ffiHandle.handle
     });
     const res = import_ffi_client.FfiClient.instance.request({

package/dist/audio_resampler.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/audio_resampler.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { AudioFrame } from './audio_frame.js';\nimport { FfiClient, FfiHandle } from './ffi_client.js';\nimport type {\n  FlushSoxResamplerResponse,\n  NewSoxResamplerResponse,\n  PushSoxResamplerResponse,\n} from './proto/audio_frame_pb.js';\nimport {\n  FlushSoxResamplerRequest,\n  NewSoxResamplerRequest,\n  PushSoxResamplerRequest,\n  SoxQualityRecipe,\n  SoxResamplerDataType,\n} from './proto/audio_frame_pb.js';\n\n/**\n * Resampler quality. Higher quality settings result in better audio quality but require more\n * processing power.\n */\nexport enum AudioResamplerQuality {\n  QUICK = SoxQualityRecipe.SOXR_QUALITY_QUICK,\n  LOW = SoxQualityRecipe.SOXR_QUALITY_LOW,\n  MEDIUM = SoxQualityRecipe.SOXR_QUALITY_MEDIUM,\n  HIGH = SoxQualityRecipe.SOXR_QUALITY_HIGH,\n  VERY_HIGH = SoxQualityRecipe.SOXR_QUALITY_VERYHIGH,\n}\n\n/**\n * AudioResampler provides functionality to resample audio data from an input sample rate to\n * an output sample rate using the Sox resampling library. It supports multiple channels and\n * configurable resampling quality.\n */\nexport class AudioResampler {\n  #inputRate: number;\n  #outputRate: number;\n  #channels: number;\n  #ffiHandle: FfiHandle;\n\n  /**\n   * Initializes a new AudioResampler.\n   *\n   * @param inputRate - The sample rate of the input audio data (in Hz).\n   * @param outputRate - The desired sample rate of the output audio data (in Hz).\n   * @param channels - The number of audio channels (e.g., 1 for mono, 2 for stereo). Defaults to 1.\n   * @param quality - The quality setting for the resampler. Defaults to\n   * `AudioResamplerQuality.MEDIUM`.\n   */\n  constructor(\n    inputRate: number,\n    outputRate: number,\n    channels = 1,\n    quality = AudioResamplerQuality.MEDIUM,\n  ) {\n    this.#inputRate = inputRate;\n    this.#outputRate = outputRate;\n    this.#channels = channels;\n\n    const req = new NewSoxResamplerRequest({\n      inputRate,\n      outputRate,\n      numChannels: channels,\n      qualityRecipe: quality as number as SoxQualityRecipe,\n      inputDataType: SoxResamplerDataType.SOXR_DATATYPE_INT16I,\n      outputDataType: SoxResamplerDataType.SOXR_DATATYPE_INT16I,\n      flags: 0,\n    });\n\n    const res = FfiClient.instance.request<NewSoxResamplerResponse>({\n      message: {\n        case: 'newSoxResampler',\n        value: req,\n      },\n    });\n\n    switch (res.message.case) {\n      case 'resampler':\n        this.#ffiHandle = new FfiHandle(res.message.value.handle!.id!);\n        break;\n      case 'error':\n      default:\n        throw new Error(res.message.value);\n    }\n  }\n\n  get inputRate(): number {\n    return this.#inputRate;\n  }\n\n  get outputRate(): number {\n    return this.#outputRate;\n  }\n\n  get channels(): number {\n    return this.#channels;\n  }\n\n  /**\n   * Push audio data into the resampler and retrieve any available resampled data.\n   *\n   * This method accepts audio data, resamples it according to the configured input and output rates,\n   * and returns any resampled data that is available after processing the input.\n   *\n   * @param data - The audio frame to resample\n   *\n   * @returns A list of {@link AudioFrame} objects containing the resampled audio data. The list may\n   * be empty if no output data is available yet.\n   */\n  push(data: AudioFrame): AudioFrame[] {\n    const req = new PushSoxResamplerRequest({\n      resamplerHandle: this.#ffiHandle.handle,\n      dataPtr: data.protoInfo().dataPtr,\n      size: data.data.byteLength,\n    });\n\n    const res = FfiClient.instance.request<PushSoxResamplerResponse>({\n      message: {\n        case: 'pushSoxResampler',\n        value: req,\n      },\n    });\n\n    if (res.error) {\n      throw new Error(res.error);\n    }\n\n    if (!res.outputPtr) {\n      return [];\n    }\n\n    const outputData = FfiClient.instance.copyBuffer(res.outputPtr, res.size!);\n    return [\n      new AudioFrame(\n        new Int16Array(outputData.buffer),\n        this.#outputRate,\n        this.#channels,\n        Math.trunc(outputData.length / this.#channels / 2),\n      ),\n    ];\n  }\n\n  /**\n   * Flush any remaining audio data through the resampler and retrieve the resampled data.\n   *\n   * @remarks\n   * This method should be called when no more input data will be provided to ensure that all\n   * internal buffers are processed and all resampled data is output.\n   */\n  flush(): AudioFrame[] {\n    const req = new FlushSoxResamplerRequest({\n      resamplerHandle: this.#ffiHandle.handle,\n    });\n\n    const res = FfiClient.instance.request<FlushSoxResamplerResponse>({\n      message: {\n        case: 'flushSoxResampler',\n        value: req,\n      },\n    });\n\n    if (res.error) {\n      throw new Error(res.error);\n    }\n\n    if (!res.outputPtr) {\n      return [];\n    }\n\n    const outputData = FfiClient.instance.copyBuffer(res.outputPtr, res.size!);\n    return [\n      new AudioFrame(\n        new Int16Array(outputData.buffer),\n        this.#outputRate,\n        this.#channels,\n        Math.trunc(outputData.length / this.#channels / 2),\n      ),\n    ];\n  }\n}\n"],"mappings":";;;;;;;;;;;;;;;;;;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAGA,yBAA2B;AAC3B,wBAAqC;AAMrC,4BAMO;AAMA,IAAK,yBAAL,CAAKA,2BAAL;AACL,EAAAA,8CAAA,WAAQ,uCAAiB,sBAAzB;AACA,EAAAA,8CAAA,SAAM,uCAAiB,oBAAvB;AACA,EAAAA,8CAAA,YAAS,uCAAiB,uBAA1B;AACA,EAAAA,8CAAA,UAAO,uCAAiB,qBAAxB;AACA,EAAAA,8CAAA,eAAY,uCAAiB,yBAA7B;AALU,SAAAA;AAAA,GAAA;AAaL,MAAM,eAAe;AAAA,EAC1B;AAAA,EACA;AAAA,EACA;AAAA,EACA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA,EAWA,YACE,WACA,YACA,WAAW,GACX,UAAU,sBAAsB,QAChC;AACA,SAAK,aAAa;AAClB,SAAK,cAAc;AACnB,SAAK,YAAY;AAEjB,UAAM,MAAM,IAAI,6CAAuB;AAAA,MACrC;AAAA,MACA;AAAA,MACA,aAAa;AAAA,MACb,eAAe;AAAA,MACf,eAAe,2CAAqB;AAAA,MACpC,gBAAgB,2CAAqB;AAAA,MACrC,OAAO;AAAA,IACT,CAAC;AAED,UAAM,MAAM,4BAAU,SAAS,QAAiC;AAAA,MAC9D,SAAS;AAAA,QACP,MAAM;AAAA,QACN,OAAO;AAAA,MACT;AAAA,IACF,CAAC;AAED,YAAQ,IAAI,QAAQ,MAAM;AAAA,MACxB,KAAK;AACH,aAAK,aAAa,IAAI,4BAAU,IAAI,QAAQ,MAAM,OAAQ,EAAG;AAC7D;AAAA,MACF,KAAK;AAAA,MACL;AACE,cAAM,IAAI,MAAM,IAAI,QAAQ,KAAK;AAAA,IACrC;AAAA,EACF;AAAA,EAEA,IAAI,YAAoB;AACtB,WAAO,KAAK;AAAA,EACd;AAAA,EAEA,IAAI,aAAqB;AACvB,WAAO,KAAK;AAAA,EACd;AAAA,EAEA,IAAI,WAAmB;AACrB,WAAO,KAAK;AAAA,EACd;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA,EAaA,KAAK,MAAgC;AACnC,UAAM,MAAM,IAAI,8CAAwB;AAAA,MACtC,iBAAiB,KAAK,WAAW;AAAA,MACjC,SAAS,KAAK,UAAU,EAAE;AAAA,MAC1B,MAAM,KAAK,KAAK;AAAA,IAClB,CAAC;AAED,UAAM,MAAM,4BAAU,SAAS,QAAkC;AAAA,MAC/D,SAAS;AAAA,QACP,MAAM;AAAA,QACN,OAAO;AAAA,MACT;AAAA,IACF,CAAC;AAED,QAAI,IAAI,OAAO;AACb,YAAM,IAAI,MAAM,IAAI,KAAK;AAAA,IAC3B;AAEA,QAAI,CAAC,IAAI,WAAW;AAClB,aAAO,CAAC;AAAA,IACV;AAEA,UAAM,aAAa,4BAAU,SAAS,WAAW,IAAI,WAAW,IAAI,IAAK;AACzE,WAAO;AAAA,MACL,IAAI;AAAA,QACF,IAAI,WAAW,WAAW,MAAM;AAAA,QAChC,KAAK;AAAA,QACL,KAAK;AAAA,QACL,KAAK,MAAM,WAAW,SAAS,KAAK,YAAY,CAAC;AAAA,MACnD;AAAA,IACF;AAAA,EACF;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA,EASA,QAAsB;AACpB,UAAM,MAAM,IAAI,+CAAyB;AAAA,MACvC,iBAAiB,KAAK,WAAW;AAAA,IACnC,CAAC;AAED,UAAM,MAAM,4BAAU,SAAS,QAAmC;AAAA,MAChE,SAAS;AAAA,QACP,MAAM;AAAA,QACN,OAAO;AAAA,MACT;AAAA,IACF,CAAC;AAED,QAAI,IAAI,OAAO;AACb,YAAM,IAAI,MAAM,IAAI,KAAK;AAAA,IAC3B;AAEA,QAAI,CAAC,IAAI,WAAW;AAClB,aAAO,CAAC;AAAA,IACV;AAEA,UAAM,aAAa,4BAAU,SAAS,WAAW,IAAI,WAAW,IAAI,IAAK;AACzE,WAAO;AAAA,MACL,IAAI;AAAA,QACF,IAAI,WAAW,WAAW,MAAM;AAAA,QAChC,KAAK;AAAA,QACL,KAAK;AAAA,QACL,KAAK,MAAM,WAAW,SAAS,KAAK,YAAY,CAAC;AAAA,MACnD;AAAA,IACF;AAAA,EACF;AACF;","names":["AudioResamplerQuality"]}
+{"version":3,"sources":["../src/audio_resampler.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport type {\n  FlushSoxResamplerResponse,\n  NewSoxResamplerResponse,\n  PushSoxResamplerResponse,\n} from '@livekit/rtc-ffi-bindings';\nimport {\n  FlushSoxResamplerRequest,\n  NewSoxResamplerRequest,\n  PushSoxResamplerRequest,\n  SoxQualityRecipe,\n  SoxResamplerDataType,\n} from '@livekit/rtc-ffi-bindings';\nimport { AudioFrame } from './audio_frame.js';\nimport { FfiClient, FfiHandle } from './ffi_client.js';\n\n/**\n * Resampler quality. Higher quality settings result in better audio quality but require more\n * processing power.\n */\nexport enum AudioResamplerQuality {\n  QUICK = SoxQualityRecipe.SOXR_QUALITY_QUICK,\n  LOW = SoxQualityRecipe.SOXR_QUALITY_LOW,\n  MEDIUM = SoxQualityRecipe.SOXR_QUALITY_MEDIUM,\n  HIGH = SoxQualityRecipe.SOXR_QUALITY_HIGH,\n  VERY_HIGH = SoxQualityRecipe.SOXR_QUALITY_VERYHIGH,\n}\n\n/**\n * AudioResampler provides functionality to resample audio data from an input sample rate to\n * an output sample rate using the Sox resampling library. It supports multiple channels and\n * configurable resampling quality.\n */\nexport class AudioResampler {\n  #inputRate: number;\n  #outputRate: number;\n  #channels: number;\n  #ffiHandle: FfiHandle;\n\n  /**\n   * Initializes a new AudioResampler.\n   *\n   * @param inputRate - The sample rate of the input audio data (in Hz).\n   * @param outputRate - The desired sample rate of the output audio data (in Hz).\n   * @param channels - The number of audio channels (e.g., 1 for mono, 2 for stereo). Defaults to 1.\n   * @param quality - The quality setting for the resampler. Defaults to\n   * `AudioResamplerQuality.MEDIUM`.\n   */\n  constructor(\n    inputRate: number,\n    outputRate: number,\n    channels = 1,\n    quality = AudioResamplerQuality.MEDIUM,\n  ) {\n    this.#inputRate = inputRate;\n    this.#outputRate = outputRate;\n    this.#channels = channels;\n\n    const req = new NewSoxResamplerRequest({\n      inputRate,\n      outputRate,\n      numChannels: channels,\n      qualityRecipe: quality as number as SoxQualityRecipe,\n      inputDataType: SoxResamplerDataType.SOXR_DATATYPE_INT16I,\n      outputDataType: SoxResamplerDataType.SOXR_DATATYPE_INT16I,\n      flags: 0,\n    });\n\n    const res = FfiClient.instance.request<NewSoxResamplerResponse>({\n      message: {\n        case: 'newSoxResampler',\n        value: req,\n      },\n    });\n\n    switch (res.message.case) {\n      case 'resampler':\n        this.#ffiHandle = new FfiHandle(res.message.value.handle!.id!);\n        break;\n      case 'error':\n      default:\n        throw new Error(res.message.value);\n    }\n  }\n\n  get inputRate(): number {\n    return this.#inputRate;\n  }\n\n  get outputRate(): number {\n    return this.#outputRate;\n  }\n\n  get channels(): number {\n    return this.#channels;\n  }\n\n  /**\n   * Releases the underlying native resampler handle. Must be called when\n   * the resampler is no longer needed to avoid leaking the FD.\n   */\n  close() {\n    this.#ffiHandle.dispose();\n  }\n\n  /**\n   * Push audio data into the resampler and retrieve any available resampled data.\n   *\n   * This method accepts audio data, resamples it according to the configured input and output rates,\n   * and returns any resampled data that is available after processing the input.\n   *\n   * @param data - The audio frame to resample\n   *\n   * @returns A list of {@link AudioFrame} objects containing the resampled audio data. The list may\n   * be empty if no output data is available yet.\n   */\n  push(data: AudioFrame): AudioFrame[] {\n    const req = new PushSoxResamplerRequest({\n      resamplerHandle: this.#ffiHandle.handle,\n      dataPtr: data.protoInfo().dataPtr,\n      size: data.data.byteLength,\n    });\n\n    const res = FfiClient.instance.request<PushSoxResamplerResponse>({\n      message: {\n        case: 'pushSoxResampler',\n        value: req,\n      },\n    });\n\n    if (res.error) {\n      throw new Error(res.error);\n    }\n\n    if (!res.outputPtr) {\n      return [];\n    }\n\n    const outputData = FfiClient.instance.copyBuffer(res.outputPtr, res.size!);\n\n    return [\n      new AudioFrame(\n        new Int16Array(outputData.buffer),\n        this.#outputRate,\n        this.#channels,\n        Math.trunc(outputData.length / this.#channels / 2),\n        data.userdata,\n      ),\n    ];\n  }\n\n  /**\n   * Flush any remaining audio data through the resampler and retrieve the resampled data.\n   *\n   * @remarks\n   * This method should be called when no more input data will be provided to ensure that all\n   * internal buffers are processed and all resampled data is output.\n   */\n  flush(): AudioFrame[] {\n    const req = new FlushSoxResamplerRequest({\n      resamplerHandle: this.#ffiHandle.handle,\n    });\n\n    const res = FfiClient.instance.request<FlushSoxResamplerResponse>({\n      message: {\n        case: 'flushSoxResampler',\n        value: req,\n      },\n    });\n\n    if (res.error) {\n      throw new Error(res.error);\n    }\n\n    if (!res.outputPtr) {\n      return [];\n    }\n\n    const outputData = FfiClient.instance.copyBuffer(res.outputPtr, res.size!);\n    return [\n      new AudioFrame(\n        new Int16Array(outputData.buffer),\n        this.#outputRate,\n        this.#channels,\n        Math.trunc(outputData.length / this.#channels / 2),\n      ),\n    ];\n  }\n}\n"],"mappings":";;;;;;;;;;;;;;;;;;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAQA,8BAMO;AACP,yBAA2B;AAC3B,wBAAqC;AAM9B,IAAK,yBAAL,CAAKA,2BAAL;AACL,EAAAA,8CAAA,WAAQ,yCAAiB,sBAAzB;AACA,EAAAA,8CAAA,SAAM,yCAAiB,oBAAvB;AACA,EAAAA,8CAAA,YAAS,yCAAiB,uBAA1B;AACA,EAAAA,8CAAA,UAAO,yCAAiB,qBAAxB;AACA,EAAAA,8CAAA,eAAY,yCAAiB,yBAA7B;AALU,SAAAA;AAAA,GAAA;AAaL,MAAM,eAAe;AAAA,EAC1B;AAAA,EACA;AAAA,EACA;AAAA,EACA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA,EAWA,YACE,WACA,YACA,WAAW,GACX,UAAU,sBAAsB,QAChC;AACA,SAAK,aAAa;AAClB,SAAK,cAAc;AACnB,SAAK,YAAY;AAEjB,UAAM,MAAM,IAAI,+CAAuB;AAAA,MACrC;AAAA,MACA;AAAA,MACA,aAAa;AAAA,MACb,eAAe;AAAA,MACf,eAAe,6CAAqB;AAAA,MACpC,gBAAgB,6CAAqB;AAAA,MACrC,OAAO;AAAA,IACT,CAAC;AAED,UAAM,MAAM,4BAAU,SAAS,QAAiC;AAAA,MAC9D,SAAS;AAAA,QACP,MAAM;AAAA,QACN,OAAO;AAAA,MACT;AAAA,IACF,CAAC;AAED,YAAQ,IAAI,QAAQ,MAAM;AAAA,MACxB,KAAK;AACH,aAAK,aAAa,IAAI,4BAAU,IAAI,QAAQ,MAAM,OAAQ,EAAG;AAC7D;AAAA,MACF,KAAK;AAAA,MACL;AACE,cAAM,IAAI,MAAM,IAAI,QAAQ,KAAK;AAAA,IACrC;AAAA,EACF;AAAA,EAEA,IAAI,YAAoB;AACtB,WAAO,KAAK;AAAA,EACd;AAAA,EAEA,IAAI,aAAqB;AACvB,WAAO,KAAK;AAAA,EACd;AAAA,EAEA,IAAI,WAAmB;AACrB,WAAO,KAAK;AAAA,EACd;AAAA;AAAA;AAAA;AAAA;AAAA,EAMA,QAAQ;AACN,SAAK,WAAW,QAAQ;AAAA,EAC1B;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA,EAaA,KAAK,MAAgC;AACnC,UAAM,MAAM,IAAI,gDAAwB;AAAA,MACtC,iBAAiB,KAAK,WAAW;AAAA,MACjC,SAAS,KAAK,UAAU,EAAE;AAAA,MAC1B,MAAM,KAAK,KAAK;AAAA,IAClB,CAAC;AAED,UAAM,MAAM,4BAAU,SAAS,QAAkC;AAAA,MAC/D,SAAS;AAAA,QACP,MAAM;AAAA,QACN,OAAO;AAAA,MACT;AAAA,IACF,CAAC;AAED,QAAI,IAAI,OAAO;AACb,YAAM,IAAI,MAAM,IAAI,KAAK;AAAA,IAC3B;AAEA,QAAI,CAAC,IAAI,WAAW;AAClB,aAAO,CAAC;AAAA,IACV;AAEA,UAAM,aAAa,4BAAU,SAAS,WAAW,IAAI,WAAW,IAAI,IAAK;AAEzE,WAAO;AAAA,MACL,IAAI;AAAA,QACF,IAAI,WAAW,WAAW,MAAM;AAAA,QAChC,KAAK;AAAA,QACL,KAAK;AAAA,QACL,KAAK,MAAM,WAAW,SAAS,KAAK,YAAY,CAAC;AAAA,QACjD,KAAK;AAAA,MACP;AAAA,IACF;AAAA,EACF;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA,EASA,QAAsB;AACpB,UAAM,MAAM,IAAI,iDAAyB;AAAA,MACvC,iBAAiB,KAAK,WAAW;AAAA,IACnC,CAAC;AAED,UAAM,MAAM,4BAAU,SAAS,QAAmC;AAAA,MAChE,SAAS;AAAA,QACP,MAAM;AAAA,QACN,OAAO;AAAA,MACT;AAAA,IACF,CAAC;AAED,QAAI,IAAI,OAAO;AACb,YAAM,IAAI,MAAM,IAAI,KAAK;AAAA,IAC3B;AAEA,QAAI,CAAC,IAAI,WAAW;AAClB,aAAO,CAAC;AAAA,IACV;AAEA,UAAM,aAAa,4BAAU,SAAS,WAAW,IAAI,WAAW,IAAI,IAAK;AACzE,WAAO;AAAA,MACL,IAAI;AAAA,QACF,IAAI,WAAW,WAAW,MAAM;AAAA,QAChC,KAAK;AAAA,QACL,KAAK;AAAA,QACL,KAAK,MAAM,WAAW,SAAS,KAAK,YAAY,CAAC;AAAA,MACnD;AAAA,IACF;AAAA,EACF;AACF;","names":["AudioResamplerQuality"]}

package/dist/audio_resampler.d.cts

@@ -1,10 +1,5 @@
 import { AudioFrame } from './audio_frame.cjs';
-import './proto/audio_frame_pb.cjs';
-import '@bufbuild/protobuf';
-import './proto/track_pb.cjs';
-import './proto/stats_pb.cjs';
-import './proto/e2ee_pb.cjs';
-import './proto/handle_pb.cjs';
+import '@livekit/rtc-ffi-bindings';
 
 /**
  * Resampler quality. Higher quality settings result in better audio quality but require more
@@ -37,6 +32,11 @@ declare class AudioResampler {
     get inputRate(): number;
     get outputRate(): number;
     get channels(): number;
+    /**
+     * Releases the underlying native resampler handle. Must be called when
+     * the resampler is no longer needed to avoid leaking the FD.
+     */
+    close(): void;
     /**
      * Push audio data into the resampler and retrieve any available resampled data.
      *