@livekit/rtc-node 0.12.1 → 0.13.0

package/dist/audio_frame.cjs

@@ -22,7 +22,6 @@ __export(audio_frame_exports, {
   combineAudioFrames: () => combineAudioFrames
 });
 module.exports = __toCommonJS(audio_frame_exports);
-var import_protobuf = require("@bufbuild/protobuf");
 var import_ffi_client = require("./ffi_client.cjs");
 var import_audio_frame_pb = require("./proto/audio_frame_pb.cjs");
 class AudioFrame {
@@ -56,7 +55,7 @@ class AudioFrame {
   }
   /** @internal */
   protoInfo() {
-    return (0, import_protobuf.create)(import_audio_frame_pb.AudioFrameBufferInfoSchema, {
+    return new import_audio_frame_pb.AudioFrameBufferInfo({
       dataPtr: import_ffi_client.FfiClient.instance.retrievePtr(new Uint8Array(this.data.buffer)),
       sampleRate: this.sampleRate,
       numChannels: this.channels,
@@ -68,6 +67,7 @@ const combineAudioFrames = (buffer) => {
   if (!Array.isArray(buffer)) {
     return buffer;
   }
+  buffer = buffer;
   if (buffer.length === 0) {
     throw new Error("buffer is empty");
   }

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 { create } from '@bufbuild/protobuf';\nimport { FfiClient, FfiHandle } from './ffi_client.js';\nimport type { OwnedAudioFrameBuffer } from './proto/audio_frame_pb.js';\nimport { type AudioFrameBufferInfo, AudioFrameBufferInfoSchema } from './proto/audio_frame_pb.js';\n\nexport class AudioFrame {\n  data: Int16Array;\n  sampleRate: number;\n  channels: number;\n  samplesPerChannel: number;\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(data: Int16Array, sampleRate: number, channels: number, samplesPerChannel: number) {\n    this.data = data;\n    this.sampleRate = sampleRate;\n    this.channels = channels;\n    this.samplesPerChannel = samplesPerChannel;\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 create(AudioFrameBufferInfoSchema, {\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\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 as AudioFrame;\n  }\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,sBAAuB;AACvB,wBAAqC;AAErC,4BAAsE;AAE/D,MAAM,WAAW;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA,EAWtB,YAAY,MAAkB,YAAoB,UAAkB,mBAA2B;AAC7F,SAAK,OAAO;AACZ,SAAK,aAAa;AAClB,SAAK,WAAW;AAChB,SAAK,oBAAoB;AAAA,EAC3B;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,cAAc,KAAK,oBAAoB;AACxD,UAAM,OAAO,4BAAU,SAAS,WAAW,KAAK,SAAS,GAAG;AAC5D,QAAI,4BAAU,MAAM,OAAQ,EAAE,EAAE,QAAQ;AACxC,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,eAAO,wBAAO,kDAA4B;AAAA,MACxC,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;AACF;AAWO,MAAM,qBAAqB,CAAC,WAAkD;AACnF,MAAI,CAAC,MAAM,QAAQ,MAAM,GAAG;AAC1B,WAAO;AAAA,EACT;AAEA,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 { 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  // 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(data: Int16Array, sampleRate: number, channels: number, samplesPerChannel: number) {\n    this.data = data;\n    this.sampleRate = sampleRate;\n    this.channels = channels;\n    this.samplesPerChannel = samplesPerChannel;\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\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,EAWtB,YAAY,MAAkB,YAAoB,UAAkB,mBAA2B;AAC7F,SAAK,OAAO;AACZ,SAAK,aAAa;AAClB,SAAK,WAAW;AAChB,SAAK,oBAAoB;AAAA,EAC3B;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;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":[]}

package/dist/audio_frame.d.cts

@@ -0,0 +1,31 @@
+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';
+
+declare class AudioFrame {
+    data: Int16Array;
+    sampleRate: number;
+    channels: number;
+    samplesPerChannel: number;
+    constructor(data: Int16Array, sampleRate: number, channels: number, samplesPerChannel: number);
+    static create(sampleRate: number, channels: number, samplesPerChannel: number): AudioFrame;
+    /** @internal */
+    static fromOwnedInfo(owned: OwnedAudioFrameBuffer): AudioFrame;
+    /** @internal */
+    protoInfo(): AudioFrameBufferInfo;
+}
+/**
+ * Combines one or more `rtc.AudioFrame` objects into a single `rtc.AudioFrame`.
+ *
+ * This function concatenates the audio data from multiple frames, ensuring that all frames have
+ * the same sample rate and number of channels. It efficiently merges the data by preallocating the
+ * necessary memory and copying the frame data without unnecessary reallocations.
+ *
+ * @param buffer - a single AudioFrame or list thereof
+ */
+declare const combineAudioFrames: (buffer: AudioFrame | AudioFrame[]) => AudioFrame;
+
+export { AudioFrame, combineAudioFrames };

package/dist/audio_frame.d.ts

@@ -1,6 +1,11 @@
-import type { OwnedAudioFrameBuffer } from './proto/audio_frame_pb.js';
-import { type AudioFrameBufferInfo } from './proto/audio_frame_pb.js';
-export declare class AudioFrame {
+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';
+
+declare class AudioFrame {
     data: Int16Array;
     sampleRate: number;
     channels: number;
@@ -21,5 +26,6 @@ export declare class AudioFrame {
  *
  * @param buffer - a single AudioFrame or list thereof
  */
-export declare const combineAudioFrames: (buffer: AudioFrame | AudioFrame[]) => AudioFrame;
-//# sourceMappingURL=audio_frame.d.ts.map
+declare const combineAudioFrames: (buffer: AudioFrame | AudioFrame[]) => AudioFrame;
+
+export { AudioFrame, combineAudioFrames };

package/dist/audio_frame.d.ts.map

@@ -1 +1 @@
-{"version":3,"file":"audio_frame.d.ts","sourceRoot":"","sources":["../src/audio_frame.ts"],"names":[],"mappings":"AAKA,OAAO,KAAK,EAAE,qBAAqB,EAAE,MAAM,2BAA2B,CAAC;AACvE,OAAO,EAAE,KAAK,oBAAoB,EAA8B,MAAM,2BAA2B,CAAC;AAElG,qBAAa,UAAU;IACrB,IAAI,EAAE,UAAU,CAAC;IACjB,UAAU,EAAE,MAAM,CAAC;IACnB,QAAQ,EAAE,MAAM,CAAC;IACjB,iBAAiB,EAAE,MAAM,CAAC;gBAOd,IAAI,EAAE,UAAU,EAAE,UAAU,EAAE,MAAM,EAAE,QAAQ,EAAE,MAAM,EAAE,iBAAiB,EAAE,MAAM;IAO7F,MAAM,CAAC,MAAM,CAAC,UAAU,EAAE,MAAM,EAAE,QAAQ,EAAE,MAAM,EAAE,iBAAiB,EAAE,MAAM,GAAG,UAAU;IAK1F,gBAAgB;IAChB,MAAM,CAAC,aAAa,CAAC,KAAK,EAAE,qBAAqB,GAAG,UAAU;IAa9D,gBAAgB;IAChB,SAAS,IAAI,oBAAoB;CAQlC;AAED;;;;;;;;GAQG;AACH,eAAO,MAAM,kBAAkB,WAAY,UAAU,GAAG,UAAU,EAAE,KAAG,UA2BtE,CAAC"}
+{"version":3,"file":"audio_frame.d.ts","sourceRoot":"","sources":["../src/audio_frame.ts"],"names":[],"mappings":"AAIA,OAAO,KAAK,EAAE,qBAAqB,EAAE,MAAM,2BAA2B,CAAC;AACvE,OAAO,EAAE,oBAAoB,EAAE,MAAM,2BAA2B,CAAC;AAEjE,qBAAa,UAAU;IACrB,IAAI,EAAE,UAAU,CAAC;IACjB,UAAU,EAAE,MAAM,CAAC;IACnB,QAAQ,EAAE,MAAM,CAAC;IACjB,iBAAiB,EAAE,MAAM,CAAC;gBAOd,IAAI,EAAE,UAAU,EAAE,UAAU,EAAE,MAAM,EAAE,QAAQ,EAAE,MAAM,EAAE,iBAAiB,EAAE,MAAM;IAO7F,MAAM,CAAC,MAAM,CAAC,UAAU,EAAE,MAAM,EAAE,QAAQ,EAAE,MAAM,EAAE,iBAAiB,EAAE,MAAM,GAAG,UAAU;IAK1F,gBAAgB;IAChB,MAAM,CAAC,aAAa,CAAC,KAAK,EAAE,qBAAqB,GAAG,UAAU;IAa9D,gBAAgB;IAChB,SAAS,IAAI,oBAAoB;CAQlC;AAED;;;;;;;;GAQG;AACH,eAAO,MAAM,kBAAkB,WAAY,UAAU,GAAG,UAAU,EAAE,KAAG,UA4BtE,CAAC"}

package/dist/audio_frame.js

@@ -1,6 +1,5 @@
-import { create } from "@bufbuild/protobuf";
 import { FfiClient, FfiHandle } from "./ffi_client.js";
-import { AudioFrameBufferInfoSchema } from "./proto/audio_frame_pb.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
@@ -32,7 +31,7 @@ class AudioFrame {
   }
   /** @internal */
   protoInfo() {
-    return create(AudioFrameBufferInfoSchema, {
+    return new AudioFrameBufferInfo({
       dataPtr: FfiClient.instance.retrievePtr(new Uint8Array(this.data.buffer)),
       sampleRate: this.sampleRate,
       numChannels: this.channels,
@@ -44,6 +43,7 @@ const combineAudioFrames = (buffer) => {
   if (!Array.isArray(buffer)) {
     return buffer;
   }
+  buffer = buffer;
   if (buffer.length === 0) {
     throw new Error("buffer is empty");
   }

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 { create } from '@bufbuild/protobuf';\nimport { FfiClient, FfiHandle } from './ffi_client.js';\nimport type { OwnedAudioFrameBuffer } from './proto/audio_frame_pb.js';\nimport { type AudioFrameBufferInfo, AudioFrameBufferInfoSchema } from './proto/audio_frame_pb.js';\n\nexport class AudioFrame {\n  data: Int16Array;\n  sampleRate: number;\n  channels: number;\n  samplesPerChannel: number;\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(data: Int16Array, sampleRate: number, channels: number, samplesPerChannel: number) {\n    this.data = data;\n    this.sampleRate = sampleRate;\n    this.channels = channels;\n    this.samplesPerChannel = samplesPerChannel;\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 create(AudioFrameBufferInfoSchema, {\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\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 as AudioFrame;\n  }\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,cAAc;AACvB,SAAS,WAAW,iBAAiB;AAErC,SAAoC,kCAAkC;AAE/D,MAAM,WAAW;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA,EAWtB,YAAY,MAAkB,YAAoB,UAAkB,mBAA2B;AAC7F,SAAK,OAAO;AACZ,SAAK,aAAa;AAClB,SAAK,WAAW;AAChB,SAAK,oBAAoB;AAAA,EAC3B;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,cAAc,KAAK,oBAAoB;AACxD,UAAM,OAAO,UAAU,SAAS,WAAW,KAAK,SAAS,GAAG;AAC5D,QAAI,UAAU,MAAM,OAAQ,EAAE,EAAE,QAAQ;AACxC,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,OAAO,4BAA4B;AAAA,MACxC,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;AACF;AAWO,MAAM,qBAAqB,CAAC,WAAkD;AACnF,MAAI,CAAC,MAAM,QAAQ,MAAM,GAAG;AAC1B,WAAO;AAAA,EACT;AAEA,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 { 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  // 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(data: Int16Array, sampleRate: number, channels: number, samplesPerChannel: number) {\n    this.data = data;\n    this.sampleRate = sampleRate;\n    this.channels = channels;\n    this.samplesPerChannel = samplesPerChannel;\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\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,EAWtB,YAAY,MAAkB,YAAoB,UAAkB,mBAA2B;AAC7F,SAAK,OAAO;AACZ,SAAK,aAAa;AAClB,SAAK,WAAW;AAChB,SAAK,oBAAoB;AAAA,EAC3B;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;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":[]}

package/dist/audio_resampler.cjs

@@ -22,7 +22,6 @@ __export(audio_resampler_exports, {
   AudioResamplerQuality: () => AudioResamplerQuality
 });
 module.exports = __toCommonJS(audio_resampler_exports);
-var import_protobuf = require("@bufbuild/protobuf");
 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");
@@ -52,7 +51,7 @@ class AudioResampler {
     this.#inputRate = inputRate;
     this.#outputRate = outputRate;
     this.#channels = channels;
-    const req = (0, import_protobuf.create)(import_audio_frame_pb.NewSoxResamplerRequestSchema, {
+    const req = new import_audio_frame_pb.NewSoxResamplerRequest({
       inputRate,
       outputRate,
       numChannels: channels,
@@ -67,10 +66,14 @@ class AudioResampler {
         value: req
       }
     });
-    if (res.message.case !== "resampler") {
-      throw new Error(res.message.value ?? "Unknown Error");
+    switch (res.message.case) {
+      case "resampler":
+        this.#ffiHandle = new import_ffi_client.FfiHandle(res.message.value.handle.id);
+        break;
+      case "error":
+      default:
+        throw new Error(res.message.value);
     }
-    this.#ffiHandle = new import_ffi_client.FfiHandle(res.message.value.handle.id);
   }
   /**
    * Push audio data into the resampler and retrieve any available resampled data.
@@ -84,7 +87,7 @@ class AudioResampler {
    * be empty if no output data is available yet.
    */
   push(data) {
-    const req = (0, import_protobuf.create)(import_audio_frame_pb.PushSoxResamplerRequestSchema, {
+    const req = new import_audio_frame_pb.PushSoxResamplerRequest({
       resamplerHandle: this.#ffiHandle.handle,
       dataPtr: data.protoInfo().dataPtr,
       size: data.data.byteLength
@@ -119,7 +122,7 @@ class AudioResampler {
    * internal buffers are processed and all resampled data is output.
    */
   flush() {
-    const req = (0, import_protobuf.create)(import_audio_frame_pb.FlushSoxResamplerRequestSchema, {
+    const req = new import_audio_frame_pb.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 { create } from '@bufbuild/protobuf';\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  FlushSoxResamplerRequestSchema,\n  NewSoxResamplerRequestSchema,\n  PushSoxResamplerRequestSchema,\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 = create(NewSoxResamplerRequestSchema, {\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    if (res.message.case !== 'resampler') {\n      throw new Error(res.message.value ?? 'Unknown Error');\n    }\n\n    this.#ffiHandle = new FfiHandle(res.message.value.handle!.id);\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 = create(PushSoxResamplerRequestSchema, {\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 = create(FlushSoxResamplerRequestSchema, {\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,sBAAuB;AACvB,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,UAAM,wBAAO,oDAA8B;AAAA,MAC/C;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,QAAI,IAAI,QAAQ,SAAS,aAAa;AACpC,YAAM,IAAI,MAAM,IAAI,QAAQ,SAAS,eAAe;AAAA,IACtD;AAEA,SAAK,aAAa,IAAI,4BAAU,IAAI,QAAQ,MAAM,OAAQ,EAAE;AAAA,EAC9D;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA,EAaA,KAAK,MAAgC;AACnC,UAAM,UAAM,wBAAO,qDAA+B;AAAA,MAChD,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,IAAI;AACxE,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,UAAM,wBAAO,sDAAgC;AAAA,MACjD,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,IAAI;AACxE,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 { 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  /**\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;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"]}

package/dist/audio_resampler.d.cts

@@ -0,0 +1,59 @@
+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';
+
+/**
+ * Resampler quality. Higher quality settings result in better audio quality but require more
+ * processing power.
+ */
+declare enum AudioResamplerQuality {
+    QUICK = 0,
+    LOW = 1,
+    MEDIUM = 2,
+    HIGH = 3,
+    VERY_HIGH = 4
+}
+/**
+ * AudioResampler provides functionality to resample audio data from an input sample rate to
+ * an output sample rate using the Sox resampling library. It supports multiple channels and
+ * configurable resampling quality.
+ */
+declare class AudioResampler {
+    #private;
+    /**
+     * Initializes a new AudioResampler.
+     *
+     * @param inputRate - The sample rate of the input audio data (in Hz).
+     * @param outputRate - The desired sample rate of the output audio data (in Hz).
+     * @param channels - The number of audio channels (e.g., 1 for mono, 2 for stereo). Defaults to 1.
+     * @param quality - The quality setting for the resampler. Defaults to
+     * `AudioResamplerQuality.MEDIUM`.
+     */
+    constructor(inputRate: number, outputRate: number, channels?: number, quality?: AudioResamplerQuality);
+    /**
+     * Push audio data into the resampler and retrieve any available resampled data.
+     *
+     * This method accepts audio data, resamples it according to the configured input and output rates,
+     * and returns any resampled data that is available after processing the input.
+     *
+     * @param data - The audio frame to resample
+     *
+     * @returns A list of {@link AudioFrame} objects containing the resampled audio data. The list may
+     * be empty if no output data is available yet.
+     */
+    push(data: AudioFrame): AudioFrame[];
+    /**
+     * Flush any remaining audio data through the resampler and retrieve the resampled data.
+     *
+     * @remarks
+     * This method should be called when no more input data will be provided to ensure that all
+     * internal buffers are processed and all resampled data is output.
+     */
+    flush(): AudioFrame[];
+}
+
+export { AudioResampler, AudioResamplerQuality };

package/dist/audio_resampler.d.ts

@@ -1,9 +1,16 @@
 import { AudioFrame } from './audio_frame.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';
+
 /**
  * Resampler quality. Higher quality settings result in better audio quality but require more
  * processing power.
  */
-export declare enum AudioResamplerQuality {
+declare enum AudioResamplerQuality {
     QUICK = 0,
     LOW = 1,
     MEDIUM = 2,
@@ -15,7 +22,7 @@ export declare enum AudioResamplerQuality {
  * an output sample rate using the Sox resampling library. It supports multiple channels and
  * configurable resampling quality.
  */
-export declare class AudioResampler {
+declare class AudioResampler {
     #private;
     /**
      * Initializes a new AudioResampler.
@@ -48,4 +55,5 @@ export declare class AudioResampler {
      */
     flush(): AudioFrame[];
 }
-//# sourceMappingURL=audio_resampler.d.ts.map
+
+export { AudioResampler, AudioResamplerQuality };

package/dist/audio_resampler.d.ts.map

@@ -1 +1 @@
-{"version":3,"file":"audio_resampler.d.ts","sourceRoot":"","sources":["../src/audio_resampler.ts"],"names":[],"mappings":"AAIA,OAAO,EAAE,UAAU,EAAE,MAAM,kBAAkB,CAAC;AAe9C;;;GAGG;AACH,oBAAY,qBAAqB;IAC/B,KAAK,IAAsC;IAC3C,GAAG,IAAoC;IACvC,MAAM,IAAuC;IAC7C,IAAI,IAAqC;IACzC,SAAS,IAAyC;CACnD;AAED;;;;GAIG;AACH,qBAAa,cAAc;;IAMzB;;;;;;;;OAQG;gBAED,SAAS,EAAE,MAAM,EACjB,UAAU,EAAE,MAAM,EAClB,QAAQ,SAAI,EACZ,OAAO,wBAA+B;IA8BxC;;;;;;;;;;OAUG;IACH,IAAI,CAAC,IAAI,EAAE,UAAU,GAAG,UAAU,EAAE;IAiCpC;;;;;;OAMG;IACH,KAAK,IAAI,UAAU,EAAE;CA8BtB"}
+{"version":3,"file":"audio_resampler.d.ts","sourceRoot":"","sources":["../src/audio_resampler.ts"],"names":[],"mappings":"AAGA,OAAO,EAAE,UAAU,EAAE,MAAM,kBAAkB,CAAC;AAe9C;;;GAGG;AACH,oBAAY,qBAAqB;IAC/B,KAAK,IAAsC;IAC3C,GAAG,IAAoC;IACvC,MAAM,IAAuC;IAC7C,IAAI,IAAqC;IACzC,SAAS,IAAyC;CACnD;AAED;;;;GAIG;AACH,qBAAa,cAAc;;IAMzB;;;;;;;;OAQG;gBAED,SAAS,EAAE,MAAM,EACjB,UAAU,EAAE,MAAM,EAClB,QAAQ,SAAI,EACZ,OAAO,wBAA+B;IAiCxC;;;;;;;;;;OAUG;IACH,IAAI,CAAC,IAAI,EAAE,UAAU,GAAG,UAAU,EAAE;IAiCpC;;;;;;OAMG;IACH,KAAK,IAAI,UAAU,EAAE;CA8BtB"}

package/dist/audio_resampler.js

@@ -1,10 +1,9 @@
-import { create } from "@bufbuild/protobuf";
 import { AudioFrame } from "./audio_frame.js";
 import { FfiClient, FfiHandle } from "./ffi_client.js";
 import {
-  FlushSoxResamplerRequestSchema,
-  NewSoxResamplerRequestSchema,
-  PushSoxResamplerRequestSchema,
+  FlushSoxResamplerRequest,
+  NewSoxResamplerRequest,
+  PushSoxResamplerRequest,
   SoxQualityRecipe,
   SoxResamplerDataType
 } from "./proto/audio_frame_pb.js";
@@ -34,7 +33,7 @@ class AudioResampler {
     this.#inputRate = inputRate;
     this.#outputRate = outputRate;
     this.#channels = channels;
-    const req = create(NewSoxResamplerRequestSchema, {
+    const req = new NewSoxResamplerRequest({
       inputRate,
       outputRate,
       numChannels: channels,
@@ -49,10 +48,14 @@ class AudioResampler {
         value: req
       }
     });
-    if (res.message.case !== "resampler") {
-      throw new Error(res.message.value ?? "Unknown Error");
+    switch (res.message.case) {
+      case "resampler":
+        this.#ffiHandle = new FfiHandle(res.message.value.handle.id);
+        break;
+      case "error":
+      default:
+        throw new Error(res.message.value);
     }
-    this.#ffiHandle = new FfiHandle(res.message.value.handle.id);
   }
   /**
    * Push audio data into the resampler and retrieve any available resampled data.
@@ -66,7 +69,7 @@ class AudioResampler {
    * be empty if no output data is available yet.
    */
   push(data) {
-    const req = create(PushSoxResamplerRequestSchema, {
+    const req = new PushSoxResamplerRequest({
       resamplerHandle: this.#ffiHandle.handle,
       dataPtr: data.protoInfo().dataPtr,
       size: data.data.byteLength
@@ -101,7 +104,7 @@ class AudioResampler {
    * internal buffers are processed and all resampled data is output.
    */
   flush() {
-    const req = create(FlushSoxResamplerRequestSchema, {
+    const req = new FlushSoxResamplerRequest({
       resamplerHandle: this.#ffiHandle.handle
     });
     const res = FfiClient.instance.request({

package/dist/audio_resampler.js.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 { create } from '@bufbuild/protobuf';\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  FlushSoxResamplerRequestSchema,\n  NewSoxResamplerRequestSchema,\n  PushSoxResamplerRequestSchema,\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 = create(NewSoxResamplerRequestSchema, {\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    if (res.message.case !== 'resampler') {\n      throw new Error(res.message.value ?? 'Unknown Error');\n    }\n\n    this.#ffiHandle = new FfiHandle(res.message.value.handle!.id);\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 = create(PushSoxResamplerRequestSchema, {\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 = create(FlushSoxResamplerRequestSchema, {\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":"AAGA,SAAS,cAAc;AACvB,SAAS,kBAAkB;AAC3B,SAAS,WAAW,iBAAiB;AAMrC;AAAA,EACE;AAAA,EACA;AAAA,EACA;AAAA,EACA;AAAA,EACA;AAAA,OACK;AAMA,IAAK,yBAAL,CAAKA,2BAAL;AACL,EAAAA,8CAAA,WAAQ,iBAAiB,sBAAzB;AACA,EAAAA,8CAAA,SAAM,iBAAiB,oBAAvB;AACA,EAAAA,8CAAA,YAAS,iBAAiB,uBAA1B;AACA,EAAAA,8CAAA,UAAO,iBAAiB,qBAAxB;AACA,EAAAA,8CAAA,eAAY,iBAAiB,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,OAAO,8BAA8B;AAAA,MAC/C;AAAA,MACA;AAAA,MACA,aAAa;AAAA,MACb,eAAe;AAAA,MACf,eAAe,qBAAqB;AAAA,MACpC,gBAAgB,qBAAqB;AAAA,MACrC,OAAO;AAAA,IACT,CAAC;AAED,UAAM,MAAM,UAAU,SAAS,QAAiC;AAAA,MAC9D,SAAS;AAAA,QACP,MAAM;AAAA,QACN,OAAO;AAAA,MACT;AAAA,IACF,CAAC;AAED,QAAI,IAAI,QAAQ,SAAS,aAAa;AACpC,YAAM,IAAI,MAAM,IAAI,QAAQ,SAAS,eAAe;AAAA,IACtD;AAEA,SAAK,aAAa,IAAI,UAAU,IAAI,QAAQ,MAAM,OAAQ,EAAE;AAAA,EAC9D;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA,EAaA,KAAK,MAAgC;AACnC,UAAM,MAAM,OAAO,+BAA+B;AAAA,MAChD,iBAAiB,KAAK,WAAW;AAAA,MACjC,SAAS,KAAK,UAAU,EAAE;AAAA,MAC1B,MAAM,KAAK,KAAK;AAAA,IAClB,CAAC;AAED,UAAM,MAAM,UAAU,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,UAAU,SAAS,WAAW,IAAI,WAAW,IAAI,IAAI;AACxE,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,OAAO,gCAAgC;AAAA,MACjD,iBAAiB,KAAK,WAAW;AAAA,IACnC,CAAC;AAED,UAAM,MAAM,UAAU,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,UAAU,SAAS,WAAW,IAAI,WAAW,IAAI,IAAI;AACxE,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 { 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  /**\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":"AAGA,SAAS,kBAAkB;AAC3B,SAAS,WAAW,iBAAiB;AAMrC;AAAA,EACE;AAAA,EACA;AAAA,EACA;AAAA,EACA;AAAA,EACA;AAAA,OACK;AAMA,IAAK,yBAAL,CAAKA,2BAAL;AACL,EAAAA,8CAAA,WAAQ,iBAAiB,sBAAzB;AACA,EAAAA,8CAAA,SAAM,iBAAiB,oBAAvB;AACA,EAAAA,8CAAA,YAAS,iBAAiB,uBAA1B;AACA,EAAAA,8CAAA,UAAO,iBAAiB,qBAAxB;AACA,EAAAA,8CAAA,eAAY,iBAAiB,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,uBAAuB;AAAA,MACrC;AAAA,MACA;AAAA,MACA,aAAa;AAAA,MACb,eAAe;AAAA,MACf,eAAe,qBAAqB;AAAA,MACpC,gBAAgB,qBAAqB;AAAA,MACrC,OAAO;AAAA,IACT,CAAC;AAED,UAAM,MAAM,UAAU,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,UAAU,IAAI,QAAQ,MAAM,OAAQ,EAAG;AAC7D;AAAA,MACF,KAAK;AAAA,MACL;AACE,cAAM,IAAI,MAAM,IAAI,QAAQ,KAAK;AAAA,IACrC;AAAA,EACF;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA,EAaA,KAAK,MAAgC;AACnC,UAAM,MAAM,IAAI,wBAAwB;AAAA,MACtC,iBAAiB,KAAK,WAAW;AAAA,MACjC,SAAS,KAAK,UAAU,EAAE;AAAA,MAC1B,MAAM,KAAK,KAAK;AAAA,IAClB,CAAC;AAED,UAAM,MAAM,UAAU,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,UAAU,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,yBAAyB;AAAA,MACvC,iBAAiB,KAAK,WAAW;AAAA,IACnC,CAAC;AAED,UAAM,MAAM,UAAU,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,UAAU,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_source.cjs

@@ -21,7 +21,6 @@ __export(audio_source_exports, {
   AudioSource: () => AudioSource
 });
 module.exports = __toCommonJS(audio_source_exports);
-var import_protobuf = require("@bufbuild/protobuf");
 var import_ffi_client = require("./ffi_client.cjs");
 var import_native = require("./napi/native.cjs");
 var import_audio_frame_pb = require("./proto/audio_frame_pb.cjs");
@@ -38,7 +37,7 @@ class AudioSource {
     this.queueSize = queueSize;
     this.lastCapture = 0;
     this.currentQueueSize = 0;
-    const req = (0, import_protobuf.create)(import_audio_frame_pb.NewAudioSourceRequestSchema, {
+    const req = new import_audio_frame_pb.NewAudioSourceRequest({
       type: import_audio_frame_pb.AudioSourceType.AUDIO_SOURCE_NATIVE,
       sampleRate,
       numChannels,
@@ -60,7 +59,7 @@ class AudioSource {
     );
   }
   clearQueue() {
-    const req = (0, import_protobuf.create)(import_audio_frame_pb.ClearAudioBufferRequestSchema, {
+    const req = new import_audio_frame_pb.ClearAudioBufferRequest({
       sourceHandle: this.ffiHandle.handle
     });
     import_ffi_client.FfiClient.instance.request({
@@ -94,7 +93,7 @@ class AudioSource {
       clearTimeout(this.timeout);
     }
     this.timeout = setTimeout(this.release, this.currentQueueSize - 50);
-    const req = (0, import_protobuf.create)(import_audio_frame_pb.CaptureAudioFrameRequestSchema, {
+    const req = new import_audio_frame_pb.CaptureAudioFrameRequest({
       sourceHandle: this.ffiHandle.handle,
       buffer: frame.protoInfo()
     });
@@ -108,6 +107,9 @@ class AudioSource {
       throw new Error(cb.error);
     }
   }
+  async close() {
+    this.ffiHandle.dispose();
+  }
 }
 // Annotate the CommonJS export names for ESM import in node:
 0 && (module.exports = {

package/dist/audio_source.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/audio_source.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { create } from '@bufbuild/protobuf';\nimport type { AudioFrame } from './audio_frame.js';\nimport { FfiClient } from './ffi_client.js';\nimport { FfiHandle } from './napi/native.js';\nimport type {\n  AudioSourceInfo,\n  CaptureAudioFrameCallback,\n  CaptureAudioFrameResponse,\n  ClearAudioBufferResponse,\n  NewAudioSourceResponse,\n} from './proto/audio_frame_pb.js';\nimport {\n  AudioSourceType,\n  CaptureAudioFrameRequestSchema,\n  ClearAudioBufferRequestSchema,\n  NewAudioSourceRequestSchema,\n} from './proto/audio_frame_pb.js';\n\nexport class AudioSource {\n  /** @internal */\n  info: AudioSourceInfo;\n  /** @internal */\n  ffiHandle: FfiHandle;\n  /** @internal */\n  lastCapture: number;\n  /** @internal */\n  currentQueueSize: number;\n  /** @internal */\n  release = () => {};\n  promise = this.newPromise();\n  /** @internal */\n  timeout?: ReturnType<typeof setTimeout> = undefined;\n\n  sampleRate: number;\n  numChannels: number;\n  queueSize: number;\n\n  constructor(sampleRate: number, numChannels: number, queueSize = 1000) {\n    this.sampleRate = sampleRate;\n    this.numChannels = numChannels;\n    this.queueSize = queueSize;\n\n    this.lastCapture = 0;\n    this.currentQueueSize = 0;\n\n    const req = create(NewAudioSourceRequestSchema, {\n      type: AudioSourceType.AUDIO_SOURCE_NATIVE,\n      sampleRate: sampleRate,\n      numChannels: numChannels,\n      queueSizeMs: queueSize,\n    });\n\n    const res = FfiClient.instance.request<NewAudioSourceResponse>({\n      message: {\n        case: 'newAudioSource',\n        value: req,\n      },\n    });\n\n    this.info = res.source!.info!;\n    this.ffiHandle = new FfiHandle(res.source!.handle!.id);\n  }\n\n  get queuedDuration(): number {\n    return Math.max(\n      this.currentQueueSize - Number(process.hrtime.bigint() / BigInt(1000000)) + this.lastCapture,\n      0,\n    );\n  }\n\n  clearQueue() {\n    const req = create(ClearAudioBufferRequestSchema, {\n      sourceHandle: this.ffiHandle.handle,\n    });\n\n    FfiClient.instance.request<ClearAudioBufferResponse>({\n      message: {\n        case: 'clearAudioBuffer',\n        value: req,\n      },\n    });\n\n    this.release();\n  }\n\n  /** @internal */\n  async newPromise() {\n    return new Promise<void>((resolve) => {\n      this.release = resolve;\n    });\n  }\n\n  async waitForPlayout() {\n    return this.promise.then(() => {\n      this.lastCapture = 0;\n      this.currentQueueSize = 0;\n      this.promise = this.newPromise();\n    });\n  }\n\n  async captureFrame(frame: AudioFrame) {\n    const now = Number(process.hrtime.bigint() / BigInt(1000000));\n    const elapsed = this.lastCapture === 0 ? 0 : now - this.lastCapture;\n    const frameDurationMs = (frame.samplesPerChannel / frame.sampleRate) * 1000;\n    this.currentQueueSize += frameDurationMs - elapsed;\n\n    this.lastCapture = now;\n\n    if (this.timeout) {\n      clearTimeout(this.timeout);\n    }\n\n    // remove 50ms to account for processing time\n    // (e.g. using wait_for_playout for very small chunks)\n    this.timeout = setTimeout(this.release, this.currentQueueSize - 50);\n\n    const req = create(CaptureAudioFrameRequestSchema, {\n      sourceHandle: this.ffiHandle.handle,\n      buffer: frame.protoInfo(),\n    });\n\n    const res = FfiClient.instance.request<CaptureAudioFrameResponse>({\n      message: { case: 'captureAudioFrame', value: req },\n    });\n\n    const cb = await FfiClient.instance.waitFor<CaptureAudioFrameCallback>((ev) => {\n      return ev.message.case == 'captureAudioFrame' && ev.message.value.asyncId == res.asyncId;\n    });\n\n    if (cb.error) {\n      throw new Error(cb.error);\n    }\n  }\n}\n"],"mappings":";;;;;;;;;;;;;;;;;;AAAA;AAAA;AAAA;AAAA;AAAA;AAGA,sBAAuB;AAEvB,wBAA0B;AAC1B,oBAA0B;AAQ1B,4BAKO;AAEA,MAAM,YAAY;AAAA,EAmBvB,YAAY,YAAoB,aAAqB,YAAY,KAAM;AATvE;AAAA,mBAAU,MAAM;AAAA,IAAC;AACjB,mBAAU,KAAK,WAAW;AAE1B;AAAA,mBAA0C;AAOxC,SAAK,aAAa;AAClB,SAAK,cAAc;AACnB,SAAK,YAAY;AAEjB,SAAK,cAAc;AACnB,SAAK,mBAAmB;AAExB,UAAM,UAAM,wBAAO,mDAA6B;AAAA,MAC9C,MAAM,sCAAgB;AAAA,MACtB;AAAA,MACA;AAAA,MACA,aAAa;AAAA,IACf,CAAC;AAED,UAAM,MAAM,4BAAU,SAAS,QAAgC;AAAA,MAC7D,SAAS;AAAA,QACP,MAAM;AAAA,QACN,OAAO;AAAA,MACT;AAAA,IACF,CAAC;AAED,SAAK,OAAO,IAAI,OAAQ;AACxB,SAAK,YAAY,IAAI,wBAAU,IAAI,OAAQ,OAAQ,EAAE;AAAA,EACvD;AAAA,EAEA,IAAI,iBAAyB;AAC3B,WAAO,KAAK;AAAA,MACV,KAAK,mBAAmB,OAAO,QAAQ,OAAO,OAAO,IAAI,OAAO,GAAO,CAAC,IAAI,KAAK;AAAA,MACjF;AAAA,IACF;AAAA,EACF;AAAA,EAEA,aAAa;AACX,UAAM,UAAM,wBAAO,qDAA+B;AAAA,MAChD,cAAc,KAAK,UAAU;AAAA,IAC/B,CAAC;AAED,gCAAU,SAAS,QAAkC;AAAA,MACnD,SAAS;AAAA,QACP,MAAM;AAAA,QACN,OAAO;AAAA,MACT;AAAA,IACF,CAAC;AAED,SAAK,QAAQ;AAAA,EACf;AAAA;AAAA,EAGA,MAAM,aAAa;AACjB,WAAO,IAAI,QAAc,CAAC,YAAY;AACpC,WAAK,UAAU;AAAA,IACjB,CAAC;AAAA,EACH;AAAA,EAEA,MAAM,iBAAiB;AACrB,WAAO,KAAK,QAAQ,KAAK,MAAM;AAC7B,WAAK,cAAc;AACnB,WAAK,mBAAmB;AACxB,WAAK,UAAU,KAAK,WAAW;AAAA,IACjC,CAAC;AAAA,EACH;AAAA,EAEA,MAAM,aAAa,OAAmB;AACpC,UAAM,MAAM,OAAO,QAAQ,OAAO,OAAO,IAAI,OAAO,GAAO,CAAC;AAC5D,UAAM,UAAU,KAAK,gBAAgB,IAAI,IAAI,MAAM,KAAK;AACxD,UAAM,kBAAmB,MAAM,oBAAoB,MAAM,aAAc;AACvE,SAAK,oBAAoB,kBAAkB;AAE3C,SAAK,cAAc;AAEnB,QAAI,KAAK,SAAS;AAChB,mBAAa,KAAK,OAAO;AAAA,IAC3B;AAIA,SAAK,UAAU,WAAW,KAAK,SAAS,KAAK,mBAAmB,EAAE;AAElE,UAAM,UAAM,wBAAO,sDAAgC;AAAA,MACjD,cAAc,KAAK,UAAU;AAAA,MAC7B,QAAQ,MAAM,UAAU;AAAA,IAC1B,CAAC;AAED,UAAM,MAAM,4BAAU,SAAS,QAAmC;AAAA,MAChE,SAAS,EAAE,MAAM,qBAAqB,OAAO,IAAI;AAAA,IACnD,CAAC;AAED,UAAM,KAAK,MAAM,4BAAU,SAAS,QAAmC,CAAC,OAAO;AAC7E,aAAO,GAAG,QAAQ,QAAQ,uBAAuB,GAAG,QAAQ,MAAM,WAAW,IAAI;AAAA,IACnF,CAAC;AAED,QAAI,GAAG,OAAO;AACZ,YAAM,IAAI,MAAM,GAAG,KAAK;AAAA,IAC1B;AAAA,EACF;AACF;","names":[]}
+{"version":3,"sources":["../src/audio_source.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport type { AudioFrame } from './audio_frame.js';\nimport { FfiClient } from './ffi_client.js';\nimport { FfiHandle } from './napi/native.js';\nimport type {\n  AudioSourceInfo,\n  CaptureAudioFrameCallback,\n  CaptureAudioFrameResponse,\n  ClearAudioBufferResponse,\n  NewAudioSourceResponse,\n} from './proto/audio_frame_pb.js';\nimport {\n  AudioSourceType,\n  CaptureAudioFrameRequest,\n  ClearAudioBufferRequest,\n  NewAudioSourceRequest,\n} from './proto/audio_frame_pb.js';\n\nexport class AudioSource {\n  /** @internal */\n  info: AudioSourceInfo;\n  /** @internal */\n  ffiHandle: FfiHandle;\n  /** @internal */\n  lastCapture: number;\n  /** @internal */\n  currentQueueSize: number;\n  /** @internal */\n  release = () => {};\n  promise = this.newPromise();\n  /** @internal */\n  timeout?: ReturnType<typeof setTimeout> = undefined;\n\n  sampleRate: number;\n  numChannels: number;\n  queueSize: number;\n\n  constructor(sampleRate: number, numChannels: number, queueSize = 1000) {\n    this.sampleRate = sampleRate;\n    this.numChannels = numChannels;\n    this.queueSize = queueSize;\n\n    this.lastCapture = 0;\n    this.currentQueueSize = 0;\n\n    const req = new NewAudioSourceRequest({\n      type: AudioSourceType.AUDIO_SOURCE_NATIVE,\n      sampleRate: sampleRate,\n      numChannels: numChannels,\n      queueSizeMs: queueSize,\n    });\n\n    const res = FfiClient.instance.request<NewAudioSourceResponse>({\n      message: {\n        case: 'newAudioSource',\n        value: req,\n      },\n    });\n\n    this.info = res.source!.info!;\n    this.ffiHandle = new FfiHandle(res.source!.handle!.id!);\n  }\n\n  get queuedDuration(): number {\n    return Math.max(\n      this.currentQueueSize - Number(process.hrtime.bigint() / BigInt(1000000)) + this.lastCapture,\n      0,\n    );\n  }\n\n  clearQueue() {\n    const req = new ClearAudioBufferRequest({\n      sourceHandle: this.ffiHandle.handle,\n    });\n\n    FfiClient.instance.request<ClearAudioBufferResponse>({\n      message: {\n        case: 'clearAudioBuffer',\n        value: req,\n      },\n    });\n\n    this.release();\n  }\n\n  /** @internal */\n  async newPromise() {\n    return new Promise<void>((resolve) => {\n      this.release = resolve;\n    });\n  }\n\n  async waitForPlayout() {\n    return this.promise.then(() => {\n      this.lastCapture = 0;\n      this.currentQueueSize = 0;\n      this.promise = this.newPromise();\n    });\n  }\n\n  async captureFrame(frame: AudioFrame) {\n    const now = Number(process.hrtime.bigint() / BigInt(1000000));\n    const elapsed = this.lastCapture === 0 ? 0 : now - this.lastCapture;\n    const frameDurationMs = (frame.samplesPerChannel / frame.sampleRate) * 1000;\n    this.currentQueueSize += frameDurationMs - elapsed;\n\n    this.lastCapture = now;\n\n    if (this.timeout) {\n      clearTimeout(this.timeout);\n    }\n\n    // remove 50ms to account for processing time\n    // (e.g. using wait_for_playout for very small chunks)\n    this.timeout = setTimeout(this.release, this.currentQueueSize - 50);\n\n    const req = new CaptureAudioFrameRequest({\n      sourceHandle: this.ffiHandle.handle,\n      buffer: frame.protoInfo(),\n    });\n\n    const res = FfiClient.instance.request<CaptureAudioFrameResponse>({\n      message: { case: 'captureAudioFrame', value: req },\n    });\n\n    const cb = await FfiClient.instance.waitFor<CaptureAudioFrameCallback>((ev) => {\n      return ev.message.case == 'captureAudioFrame' && ev.message.value.asyncId == res.asyncId;\n    });\n\n    if (cb.error) {\n      throw new Error(cb.error);\n    }\n  }\n\n  async close() {\n    this.ffiHandle.dispose();\n  }\n}\n"],"mappings":";;;;;;;;;;;;;;;;;;AAAA;AAAA;AAAA;AAAA;AAAA;AAIA,wBAA0B;AAC1B,oBAA0B;AAQ1B,4BAKO;AAEA,MAAM,YAAY;AAAA,EAmBvB,YAAY,YAAoB,aAAqB,YAAY,KAAM;AATvE;AAAA,mBAAU,MAAM;AAAA,IAAC;AACjB,mBAAU,KAAK,WAAW;AAE1B;AAAA,mBAA0C;AAOxC,SAAK,aAAa;AAClB,SAAK,cAAc;AACnB,SAAK,YAAY;AAEjB,SAAK,cAAc;AACnB,SAAK,mBAAmB;AAExB,UAAM,MAAM,IAAI,4CAAsB;AAAA,MACpC,MAAM,sCAAgB;AAAA,MACtB;AAAA,MACA;AAAA,MACA,aAAa;AAAA,IACf,CAAC;AAED,UAAM,MAAM,4BAAU,SAAS,QAAgC;AAAA,MAC7D,SAAS;AAAA,QACP,MAAM;AAAA,QACN,OAAO;AAAA,MACT;AAAA,IACF,CAAC;AAED,SAAK,OAAO,IAAI,OAAQ;AACxB,SAAK,YAAY,IAAI,wBAAU,IAAI,OAAQ,OAAQ,EAAG;AAAA,EACxD;AAAA,EAEA,IAAI,iBAAyB;AAC3B,WAAO,KAAK;AAAA,MACV,KAAK,mBAAmB,OAAO,QAAQ,OAAO,OAAO,IAAI,OAAO,GAAO,CAAC,IAAI,KAAK;AAAA,MACjF;AAAA,IACF;AAAA,EACF;AAAA,EAEA,aAAa;AACX,UAAM,MAAM,IAAI,8CAAwB;AAAA,MACtC,cAAc,KAAK,UAAU;AAAA,IAC/B,CAAC;AAED,gCAAU,SAAS,QAAkC;AAAA,MACnD,SAAS;AAAA,QACP,MAAM;AAAA,QACN,OAAO;AAAA,MACT;AAAA,IACF,CAAC;AAED,SAAK,QAAQ;AAAA,EACf;AAAA;AAAA,EAGA,MAAM,aAAa;AACjB,WAAO,IAAI,QAAc,CAAC,YAAY;AACpC,WAAK,UAAU;AAAA,IACjB,CAAC;AAAA,EACH;AAAA,EAEA,MAAM,iBAAiB;AACrB,WAAO,KAAK,QAAQ,KAAK,MAAM;AAC7B,WAAK,cAAc;AACnB,WAAK,mBAAmB;AACxB,WAAK,UAAU,KAAK,WAAW;AAAA,IACjC,CAAC;AAAA,EACH;AAAA,EAEA,MAAM,aAAa,OAAmB;AACpC,UAAM,MAAM,OAAO,QAAQ,OAAO,OAAO,IAAI,OAAO,GAAO,CAAC;AAC5D,UAAM,UAAU,KAAK,gBAAgB,IAAI,IAAI,MAAM,KAAK;AACxD,UAAM,kBAAmB,MAAM,oBAAoB,MAAM,aAAc;AACvE,SAAK,oBAAoB,kBAAkB;AAE3C,SAAK,cAAc;AAEnB,QAAI,KAAK,SAAS;AAChB,mBAAa,KAAK,OAAO;AAAA,IAC3B;AAIA,SAAK,UAAU,WAAW,KAAK,SAAS,KAAK,mBAAmB,EAAE;AAElE,UAAM,MAAM,IAAI,+CAAyB;AAAA,MACvC,cAAc,KAAK,UAAU;AAAA,MAC7B,QAAQ,MAAM,UAAU;AAAA,IAC1B,CAAC;AAED,UAAM,MAAM,4BAAU,SAAS,QAAmC;AAAA,MAChE,SAAS,EAAE,MAAM,qBAAqB,OAAO,IAAI;AAAA,IACnD,CAAC;AAED,UAAM,KAAK,MAAM,4BAAU,SAAS,QAAmC,CAAC,OAAO;AAC7E,aAAO,GAAG,QAAQ,QAAQ,uBAAuB,GAAG,QAAQ,MAAM,WAAW,IAAI;AAAA,IACnF,CAAC;AAED,QAAI,GAAG,OAAO;AACZ,YAAM,IAAI,MAAM,GAAG,KAAK;AAAA,IAC1B;AAAA,EACF;AAAA,EAEA,MAAM,QAAQ;AACZ,SAAK,UAAU,QAAQ;AAAA,EACzB;AACF;","names":[]}

package/dist/audio_source.d.cts

@@ -0,0 +1,37 @@
+import { AudioFrame } from './audio_frame.cjs';
+import { FfiHandle } from './napi/native.d.cjs';
+import { AudioSourceInfo } 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';
+
+declare class AudioSource {
+    /** @internal */
+    info: AudioSourceInfo;
+    /** @internal */
+    ffiHandle: FfiHandle;
+    /** @internal */
+    lastCapture: number;
+    /** @internal */
+    currentQueueSize: number;
+    /** @internal */
+    release: () => void;
+    promise: Promise<void>;
+    /** @internal */
+    timeout?: ReturnType<typeof setTimeout>;
+    sampleRate: number;
+    numChannels: number;
+    queueSize: number;
+    constructor(sampleRate: number, numChannels: number, queueSize?: number);
+    get queuedDuration(): number;
+    clearQueue(): void;
+    /** @internal */
+    newPromise(): Promise<void>;
+    waitForPlayout(): Promise<void>;
+    captureFrame(frame: AudioFrame): Promise<void>;
+    close(): Promise<void>;
+}
+
+export { AudioSource };

package/dist/audio_source.d.ts

@@ -1,7 +1,13 @@
-import type { AudioFrame } from './audio_frame.js';
-import { FfiHandle } from './napi/native.js';
-import type { AudioSourceInfo } from './proto/audio_frame_pb.js';
-export declare class AudioSource {
+import { AudioFrame } from './audio_frame.js';
+import { FfiHandle } from './napi/native.d.js';
+import { AudioSourceInfo } 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';
+
+declare class AudioSource {
     /** @internal */
     info: AudioSourceInfo;
     /** @internal */
@@ -25,5 +31,7 @@ export declare class AudioSource {
     newPromise(): Promise<void>;
     waitForPlayout(): Promise<void>;
     captureFrame(frame: AudioFrame): Promise<void>;
+    close(): Promise<void>;
 }
-//# sourceMappingURL=audio_source.d.ts.map
+
+export { AudioSource };

package/dist/audio_source.d.ts.map

@@ -1 +1 @@
-{"version":3,"file":"audio_source.d.ts","sourceRoot":"","sources":["../src/audio_source.ts"],"names":[],"mappings":"AAIA,OAAO,KAAK,EAAE,UAAU,EAAE,MAAM,kBAAkB,CAAC;AAEnD,OAAO,EAAE,SAAS,EAAE,MAAM,kBAAkB,CAAC;AAC7C,OAAO,KAAK,EACV,eAAe,EAKhB,MAAM,2BAA2B,CAAC;AAQnC,qBAAa,WAAW;IACtB,gBAAgB;IAChB,IAAI,EAAE,eAAe,CAAC;IACtB,gBAAgB;IAChB,SAAS,EAAE,SAAS,CAAC;IACrB,gBAAgB;IAChB,WAAW,EAAE,MAAM,CAAC;IACpB,gBAAgB;IAChB,gBAAgB,EAAE,MAAM,CAAC;IACzB,gBAAgB;IAChB,OAAO,aAAY;IACnB,OAAO,gBAAqB;IAC5B,gBAAgB;IAChB,OAAO,CAAC,EAAE,UAAU,CAAC,OAAO,UAAU,CAAC,CAAa;IAEpD,UAAU,EAAE,MAAM,CAAC;IACnB,WAAW,EAAE,MAAM,CAAC;IACpB,SAAS,EAAE,MAAM,CAAC;gBAEN,UAAU,EAAE,MAAM,EAAE,WAAW,EAAE,MAAM,EAAE,SAAS,SAAO;IA0BrE,IAAI,cAAc,IAAI,MAAM,CAK3B;IAED,UAAU;IAeV,gBAAgB;IACV,UAAU;IAMV,cAAc;IAQd,YAAY,CAAC,KAAK,EAAE,UAAU;CAiCrC"}
+{"version":3,"file":"audio_source.d.ts","sourceRoot":"","sources":["../src/audio_source.ts"],"names":[],"mappings":"AAGA,OAAO,KAAK,EAAE,UAAU,EAAE,MAAM,kBAAkB,CAAC;AAEnD,OAAO,EAAE,SAAS,EAAE,MAAM,kBAAkB,CAAC;AAC7C,OAAO,KAAK,EACV,eAAe,EAKhB,MAAM,2BAA2B,CAAC;AAQnC,qBAAa,WAAW;IACtB,gBAAgB;IAChB,IAAI,EAAE,eAAe,CAAC;IACtB,gBAAgB;IAChB,SAAS,EAAE,SAAS,CAAC;IACrB,gBAAgB;IAChB,WAAW,EAAE,MAAM,CAAC;IACpB,gBAAgB;IAChB,gBAAgB,EAAE,MAAM,CAAC;IACzB,gBAAgB;IAChB,OAAO,aAAY;IACnB,OAAO,gBAAqB;IAC5B,gBAAgB;IAChB,OAAO,CAAC,EAAE,UAAU,CAAC,OAAO,UAAU,CAAC,CAAa;IAEpD,UAAU,EAAE,MAAM,CAAC;IACnB,WAAW,EAAE,MAAM,CAAC;IACpB,SAAS,EAAE,MAAM,CAAC;gBAEN,UAAU,EAAE,MAAM,EAAE,WAAW,EAAE,MAAM,EAAE,SAAS,SAAO;IA0BrE,IAAI,cAAc,IAAI,MAAM,CAK3B;IAED,UAAU;IAeV,gBAAgB;IACV,UAAU;IAMV,cAAc;IAQd,YAAY,CAAC,KAAK,EAAE,UAAU;IAkC9B,KAAK;CAGZ"}