@livekit/rtc-node 0.11.0 → 0.12.0

package/dist/audio_frame.cjs

@@ -0,0 +1,94 @@
+"use strict";
+var __defProp = Object.defineProperty;
+var __getOwnPropDesc = Object.getOwnPropertyDescriptor;
+var __getOwnPropNames = Object.getOwnPropertyNames;
+var __hasOwnProp = Object.prototype.hasOwnProperty;
+var __export = (target, all) => {
+  for (var name in all)
+    __defProp(target, name, { get: all[name], enumerable: true });
+};
+var __copyProps = (to, from, except, desc) => {
+  if (from && typeof from === "object" || typeof from === "function") {
+    for (let key of __getOwnPropNames(from))
+      if (!__hasOwnProp.call(to, key) && key !== except)
+        __defProp(to, key, { get: () => from[key], enumerable: !(desc = __getOwnPropDesc(from, key)) || desc.enumerable });
+  }
+  return to;
+};
+var __toCommonJS = (mod) => __copyProps(__defProp({}, "__esModule", { value: true }), mod);
+var audio_frame_exports = {};
+__export(audio_frame_exports, {
+  AudioFrame: () => AudioFrame,
+  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 {
+  // 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
+  // noise being added to the end.
+  // it is recommended to use buffer.subarray instead.
+  // XXX(nbsp): add this when writing proper docs
+  constructor(data, sampleRate, channels, samplesPerChannel) {
+    this.data = data;
+    this.sampleRate = sampleRate;
+    this.channels = channels;
+    this.samplesPerChannel = samplesPerChannel;
+  }
+  static create(sampleRate, channels, samplesPerChannel) {
+    const data = new Int16Array(channels * samplesPerChannel);
+    return new AudioFrame(data, sampleRate, channels, samplesPerChannel);
+  }
+  /** @internal */
+  static fromOwnedInfo(owned) {
+    const info = owned.info;
+    const len = info.numChannels * info.samplesPerChannel * 2;
+    const data = import_ffi_client.FfiClient.instance.copyBuffer(info.dataPtr, len);
+    new import_ffi_client.FfiHandle(owned.handle.id).dispose();
+    return new AudioFrame(
+      new Int16Array(data.buffer),
+      info.sampleRate,
+      info.numChannels,
+      info.samplesPerChannel
+    );
+  }
+  /** @internal */
+  protoInfo() {
+    return (0, import_protobuf.create)(import_audio_frame_pb.AudioFrameBufferInfoSchema, {
+      dataPtr: import_ffi_client.FfiClient.instance.retrievePtr(new Uint8Array(this.data.buffer)),
+      sampleRate: this.sampleRate,
+      numChannels: this.channels,
+      samplesPerChannel: this.samplesPerChannel
+    });
+  }
+}
+const combineAudioFrames = (buffer) => {
+  if (!Array.isArray(buffer)) {
+    return buffer;
+  }
+  if (buffer.length === 0) {
+    throw new Error("buffer is empty");
+  }
+  const sampleRate = buffer[0].sampleRate;
+  const channels = buffer[0].channels;
+  let totalSamplesPerChannel = 0;
+  for (const frame of buffer) {
+    if (frame.sampleRate != sampleRate) {
+      throw new Error(`sample rate mismatch: expected ${sampleRate}, got ${frame.sampleRate}`);
+    }
+    if (frame.channels != channels) {
+      throw new Error(`channel mismatch: expected ${channels}, got ${frame.channels}`);
+    }
+    totalSamplesPerChannel += frame.samplesPerChannel;
+  }
+  const data = new Int16Array(buffer.map((x) => [...x.data]).flat());
+  return new AudioFrame(data, sampleRate, channels, totalSamplesPerChannel);
+};
+// Annotate the CommonJS export names for ESM import in node:
+0 && (module.exports = {
+  AudioFrame,
+  combineAudioFrames
+});
+//# sourceMappingURL=audio_frame.cjs.map

package/dist/audio_frame.cjs.map

@@ -0,0 +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":[]}

package/dist/audio_frame.js

@@ -1,72 +1,69 @@
-// SPDX-FileCopyrightText: 2024 LiveKit, Inc.
-//
-// SPDX-License-Identifier: Apache-2.0
-import { create } from '@bufbuild/protobuf';
-import { FfiClient, FfiHandle } from './ffi_client.js';
-import { AudioFrameBufferInfoSchema } from './proto/audio_frame_pb.js';
-export 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
-    // noise being added to the end.
-    // it is recommended to use buffer.subarray instead.
-    // XXX(nbsp): add this when writing proper docs
-    constructor(data, sampleRate, channels, samplesPerChannel) {
-        this.data = data;
-        this.sampleRate = sampleRate;
-        this.channels = channels;
-        this.samplesPerChannel = samplesPerChannel;
-    }
-    static create(sampleRate, channels, samplesPerChannel) {
-        const data = new Int16Array(channels * samplesPerChannel);
-        return new AudioFrame(data, sampleRate, channels, samplesPerChannel);
-    }
-    /** @internal */
-    static fromOwnedInfo(owned) {
-        const info = owned.info;
-        const len = info.numChannels * info.samplesPerChannel * 2; // c_int16
-        const data = FfiClient.instance.copyBuffer(info.dataPtr, len);
-        new FfiHandle(owned.handle.id).dispose();
-        return new AudioFrame(new Int16Array(data.buffer), info.sampleRate, info.numChannels, info.samplesPerChannel);
-    }
-    /** @internal */
-    protoInfo() {
-        return create(AudioFrameBufferInfoSchema, {
-            dataPtr: FfiClient.instance.retrievePtr(new Uint8Array(this.data.buffer)),
-            sampleRate: this.sampleRate,
-            numChannels: this.channels,
-            samplesPerChannel: this.samplesPerChannel,
-        });
-    }
+import { create } from "@bufbuild/protobuf";
+import { FfiClient, FfiHandle } from "./ffi_client.js";
+import { AudioFrameBufferInfoSchema } 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
+  // noise being added to the end.
+  // it is recommended to use buffer.subarray instead.
+  // XXX(nbsp): add this when writing proper docs
+  constructor(data, sampleRate, channels, samplesPerChannel) {
+    this.data = data;
+    this.sampleRate = sampleRate;
+    this.channels = channels;
+    this.samplesPerChannel = samplesPerChannel;
+  }
+  static create(sampleRate, channels, samplesPerChannel) {
+    const data = new Int16Array(channels * samplesPerChannel);
+    return new AudioFrame(data, sampleRate, channels, samplesPerChannel);
+  }
+  /** @internal */
+  static fromOwnedInfo(owned) {
+    const info = owned.info;
+    const len = info.numChannels * info.samplesPerChannel * 2;
+    const data = FfiClient.instance.copyBuffer(info.dataPtr, len);
+    new FfiHandle(owned.handle.id).dispose();
+    return new AudioFrame(
+      new Int16Array(data.buffer),
+      info.sampleRate,
+      info.numChannels,
+      info.samplesPerChannel
+    );
+  }
+  /** @internal */
+  protoInfo() {
+    return create(AudioFrameBufferInfoSchema, {
+      dataPtr: FfiClient.instance.retrievePtr(new Uint8Array(this.data.buffer)),
+      sampleRate: this.sampleRate,
+      numChannels: this.channels,
+      samplesPerChannel: this.samplesPerChannel
+    });
+  }
 }
-/**
- * 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
- */
-export const combineAudioFrames = (buffer) => {
-    if (!Array.isArray(buffer)) {
-        return buffer;
-    }
-    if (buffer.length === 0) {
-        throw new Error('buffer is empty');
+const combineAudioFrames = (buffer) => {
+  if (!Array.isArray(buffer)) {
+    return buffer;
+  }
+  if (buffer.length === 0) {
+    throw new Error("buffer is empty");
+  }
+  const sampleRate = buffer[0].sampleRate;
+  const channels = buffer[0].channels;
+  let totalSamplesPerChannel = 0;
+  for (const frame of buffer) {
+    if (frame.sampleRate != sampleRate) {
+      throw new Error(`sample rate mismatch: expected ${sampleRate}, got ${frame.sampleRate}`);
     }
-    const sampleRate = buffer[0].sampleRate;
-    const channels = buffer[0].channels;
-    let totalSamplesPerChannel = 0;
-    for (const frame of buffer) {
-        if (frame.sampleRate != sampleRate) {
-            throw new Error(`sample rate mismatch: expected ${sampleRate}, got ${frame.sampleRate}`);
-        }
-        if (frame.channels != channels) {
-            throw new Error(`channel mismatch: expected ${channels}, got ${frame.channels}`);
-        }
-        totalSamplesPerChannel += frame.samplesPerChannel;
+    if (frame.channels != channels) {
+      throw new Error(`channel mismatch: expected ${channels}, got ${frame.channels}`);
     }
-    const data = new Int16Array(buffer.map((x) => [...x.data]).flat());
-    return new AudioFrame(data, sampleRate, channels, totalSamplesPerChannel);
+    totalSamplesPerChannel += frame.samplesPerChannel;
+  }
+  const data = new Int16Array(buffer.map((x) => [...x.data]).flat());
+  return new AudioFrame(data, sampleRate, channels, totalSamplesPerChannel);
+};
+export {
+  AudioFrame,
+  combineAudioFrames
 };
 //# sourceMappingURL=audio_frame.js.map

package/dist/audio_frame.js.map

@@ -1 +1 @@
-{"version":3,"file":"audio_frame.js","sourceRoot":"","sources":["../src/audio_frame.ts"],"names":[],"mappings":"AAAA,6CAA6C;AAC7C,EAAE;AACF,sCAAsC;AACtC,OAAO,EAAE,MAAM,EAAE,MAAM,oBAAoB,CAAC;AAC5C,OAAO,EAAE,SAAS,EAAE,SAAS,EAAE,MAAM,iBAAiB,CAAC;AAEvD,OAAO,EAA6B,0BAA0B,EAAE,MAAM,2BAA2B,CAAC;AAElG,MAAM,OAAO,UAAU;IAMrB,gFAAgF;IAChF,6FAA6F;IAC7F,gCAAgC;IAChC,oDAAoD;IACpD,+CAA+C;IAC/C,YAAY,IAAgB,EAAE,UAAkB,EAAE,QAAgB,EAAE,iBAAyB;QAC3F,IAAI,CAAC,IAAI,GAAG,IAAI,CAAC;QACjB,IAAI,CAAC,UAAU,GAAG,UAAU,CAAC;QAC7B,IAAI,CAAC,QAAQ,GAAG,QAAQ,CAAC;QACzB,IAAI,CAAC,iBAAiB,GAAG,iBAAiB,CAAC;IAC7C,CAAC;IAED,MAAM,CAAC,MAAM,CAAC,UAAkB,EAAE,QAAgB,EAAE,iBAAyB;QAC3E,MAAM,IAAI,GAAG,IAAI,UAAU,CAAC,QAAQ,GAAG,iBAAiB,CAAC,CAAC;QAC1D,OAAO,IAAI,UAAU,CAAC,IAAI,EAAE,UAAU,EAAE,QAAQ,EAAE,iBAAiB,CAAC,CAAC;IACvE,CAAC;IAED,gBAAgB;IAChB,MAAM,CAAC,aAAa,CAAC,KAA4B;QAC/C,MAAM,IAAI,GAAG,KAAK,CAAC,IAAK,CAAC;QACzB,MAAM,GAAG,GAAG,IAAI,CAAC,WAAW,GAAG,IAAI,CAAC,iBAAiB,GAAG,CAAC,CAAC,CAAC,UAAU;QACrE,MAAM,IAAI,GAAG,SAAS,CAAC,QAAQ,CAAC,UAAU,CAAC,IAAI,CAAC,OAAO,EAAE,GAAG,CAAC,CAAC;QAC9D,IAAI,SAAS,CAAC,KAAK,CAAC,MAAO,CAAC,EAAE,CAAC,CAAC,OAAO,EAAE,CAAC;QAC1C,OAAO,IAAI,UAAU,CACnB,IAAI,UAAU,CAAC,IAAI,CAAC,MAAM,CAAC,EAC3B,IAAI,CAAC,UAAU,EACf,IAAI,CAAC,WAAW,EAChB,IAAI,CAAC,iBAAiB,CACvB,CAAC;IACJ,CAAC;IAED,gBAAgB;IAChB,SAAS;QACP,OAAO,MAAM,CAAC,0BAA0B,EAAE;YACxC,OAAO,EAAE,SAAS,CAAC,QAAQ,CAAC,WAAW,CAAC,IAAI,UAAU,CAAC,IAAI,CAAC,IAAI,CAAC,MAAM,CAAC,CAAC;YACzE,UAAU,EAAE,IAAI,CAAC,UAAU;YAC3B,WAAW,EAAE,IAAI,CAAC,QAAQ;YAC1B,iBAAiB,EAAE,IAAI,CAAC,iBAAiB;SAC1C,CAAC,CAAC;IACL,CAAC;CACF;AAED;;;;;;;;GAQG;AACH,MAAM,CAAC,MAAM,kBAAkB,GAAG,CAAC,MAAiC,EAAc,EAAE;IAClF,IAAI,CAAC,KAAK,CAAC,OAAO,CAAC,MAAM,CAAC,EAAE,CAAC;QAC3B,OAAO,MAAoB,CAAC;IAC9B,CAAC;IAED,IAAI,MAAM,CAAC,MAAM,KAAK,CAAC,EAAE,CAAC;QACxB,MAAM,IAAI,KAAK,CAAC,iBAAiB,CAAC,CAAC;IACrC,CAAC;IAED,MAAM,UAAU,GAAG,MAAM,CAAC,CAAC,CAAC,CAAC,UAAU,CAAC;IACxC,MAAM,QAAQ,GAAG,MAAM,CAAC,CAAC,CAAC,CAAC,QAAQ,CAAC;IAEpC,IAAI,sBAAsB,GAAG,CAAC,CAAC;IAC/B,KAAK,MAAM,KAAK,IAAI,MAAM,EAAE,CAAC;QAC3B,IAAI,KAAK,CAAC,UAAU,IAAI,UAAU,EAAE,CAAC;YACnC,MAAM,IAAI,KAAK,CAAC,kCAAkC,UAAU,SAAS,KAAK,CAAC,UAAU,EAAE,CAAC,CAAC;QAC3F,CAAC;QAED,IAAI,KAAK,CAAC,QAAQ,IAAI,QAAQ,EAAE,CAAC;YAC/B,MAAM,IAAI,KAAK,CAAC,8BAA8B,QAAQ,SAAS,KAAK,CAAC,QAAQ,EAAE,CAAC,CAAC;QACnF,CAAC;QAED,sBAAsB,IAAI,KAAK,CAAC,iBAAiB,CAAC;IACpD,CAAC;IAED,MAAM,IAAI,GAAG,IAAI,UAAU,CAAC,MAAM,CAAC,GAAG,CAAC,CAAC,CAAC,EAAE,EAAE,CAAC,CAAC,GAAG,CAAC,CAAC,IAAI,CAAC,CAAC,CAAC,IAAI,EAAE,CAAC,CAAC;IACnE,OAAO,IAAI,UAAU,CAAC,IAAI,EAAE,UAAU,EAAE,QAAQ,EAAE,sBAAsB,CAAC,CAAC;AAC5E,CAAC,CAAC"}
+{"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":[]}

package/dist/audio_resampler.cjs

@@ -0,0 +1,153 @@
+"use strict";
+var __defProp = Object.defineProperty;
+var __getOwnPropDesc = Object.getOwnPropertyDescriptor;
+var __getOwnPropNames = Object.getOwnPropertyNames;
+var __hasOwnProp = Object.prototype.hasOwnProperty;
+var __export = (target, all) => {
+  for (var name in all)
+    __defProp(target, name, { get: all[name], enumerable: true });
+};
+var __copyProps = (to, from, except, desc) => {
+  if (from && typeof from === "object" || typeof from === "function") {
+    for (let key of __getOwnPropNames(from))
+      if (!__hasOwnProp.call(to, key) && key !== except)
+        __defProp(to, key, { get: () => from[key], enumerable: !(desc = __getOwnPropDesc(from, key)) || desc.enumerable });
+  }
+  return to;
+};
+var __toCommonJS = (mod) => __copyProps(__defProp({}, "__esModule", { value: true }), mod);
+var audio_resampler_exports = {};
+__export(audio_resampler_exports, {
+  AudioResampler: () => AudioResampler,
+  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");
+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";
+  return AudioResamplerQuality2;
+})(AudioResamplerQuality || {});
+class AudioResampler {
+  #inputRate;
+  #outputRate;
+  #channels;
+  #ffiHandle;
+  /**
+   * 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, outputRate, channels = 1, quality = AudioResamplerQuality.MEDIUM) {
+    this.#inputRate = inputRate;
+    this.#outputRate = outputRate;
+    this.#channels = channels;
+    const req = (0, import_protobuf.create)(import_audio_frame_pb.NewSoxResamplerRequestSchema, {
+      inputRate,
+      outputRate,
+      numChannels: channels,
+      qualityRecipe: quality,
+      inputDataType: import_audio_frame_pb.SoxResamplerDataType.SOXR_DATATYPE_INT16I,
+      outputDataType: import_audio_frame_pb.SoxResamplerDataType.SOXR_DATATYPE_INT16I,
+      flags: 0
+    });
+    const res = import_ffi_client.FfiClient.instance.request({
+      message: {
+        case: "newSoxResampler",
+        value: req
+      }
+    });
+    if (res.message.case !== "resampler") {
+      throw new Error(res.message.value ?? "Unknown Error");
+    }
+    this.#ffiHandle = new import_ffi_client.FfiHandle(res.message.value.handle.id);
+  }
+  /**
+   * 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) {
+    const req = (0, import_protobuf.create)(import_audio_frame_pb.PushSoxResamplerRequestSchema, {
+      resamplerHandle: this.#ffiHandle.handle,
+      dataPtr: data.protoInfo().dataPtr,
+      size: data.data.byteLength
+    });
+    const res = import_ffi_client.FfiClient.instance.request({
+      message: {
+        case: "pushSoxResampler",
+        value: req
+      }
+    });
+    if (res.error) {
+      throw new Error(res.error);
+    }
+    if (!res.outputPtr) {
+      return [];
+    }
+    const outputData = import_ffi_client.FfiClient.instance.copyBuffer(res.outputPtr, res.size);
+    return [
+      new import_audio_frame.AudioFrame(
+        new Int16Array(outputData.buffer),
+        this.#outputRate,
+        this.#channels,
+        Math.trunc(outputData.length / this.#channels / 2)
+      )
+    ];
+  }
+  /**
+   * 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() {
+    const req = (0, import_protobuf.create)(import_audio_frame_pb.FlushSoxResamplerRequestSchema, {
+      resamplerHandle: this.#ffiHandle.handle
+    });
+    const res = import_ffi_client.FfiClient.instance.request({
+      message: {
+        case: "flushSoxResampler",
+        value: req
+      }
+    });
+    if (res.error) {
+      throw new Error(res.error);
+    }
+    if (!res.outputPtr) {
+      return [];
+    }
+    const outputData = import_ffi_client.FfiClient.instance.copyBuffer(res.outputPtr, res.size);
+    return [
+      new import_audio_frame.AudioFrame(
+        new Int16Array(outputData.buffer),
+        this.#outputRate,
+        this.#channels,
+        Math.trunc(outputData.length / this.#channels / 2)
+      )
+    ];
+  }
+}
+// Annotate the CommonJS export names for ESM import in node:
+0 && (module.exports = {
+  AudioResampler,
+  AudioResamplerQuality
+});
+//# sourceMappingURL=audio_resampler.cjs.map

package/dist/audio_resampler.cjs.map

@@ -0,0 +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"]}