@cartesia/cartesia-js 1.0.0-alpha.3 → 1.0.0

package/.turbo/turbo-build.log

@@ -8,21 +8,21 @@ $ tsup src/ --format cjs,esm --dts
 ESM dist/index.js          437.00 B
 ESM dist/chunk-FXPGR372.js 0 B
 ESM dist/lib/client.js     132.00 B
-ESM dist/react/index.js    6.22 KB
+ESM dist/react/index.js    6.35 KB
 ESM dist/tts/index.js      261.00 B
 ESM dist/lib/constants.js  183.00 B
 ESM dist/lib/index.js      322.00 B
-ESM dist/chunk-ISRU7PLL.js 353.00 B
+ESM dist/chunk-IEN4NCER.js 353.00 B
 ESM dist/chunk-SGXUEFII.js 1.35 KB
 ESM dist/react/utils.js    109.00 B
 ESM dist/chunk-3FL2SNIR.js 337.00 B
-ESM dist/chunk-VK7LBMVI.js 439.00 B
-ESM dist/chunk-IQAXBRHU.js 5.84 KB
+ESM dist/chunk-F4QWVJY3.js 439.00 B
+ESM dist/chunk-FN7BK4PS.js 6.11 KB
 ESM dist/chunk-PQ6CIPFW.js 4.02 KB
 ESM dist/chunk-PQ5EVEEH.js 841.00 B
 ESM dist/chunk-2BFEKY3F.js 366.00 B
 ESM dist/tts/player.js     143.00 B
-ESM dist/chunk-36JBKJUN.js 3.52 KB
+ESM dist/chunk-JYLAM6VU.js 3.76 KB
 ESM dist/chunk-RO7TY474.js 1.95 KB
 ESM dist/chunk-WIFMLPT5.js 2.27 KB
 ESM dist/types/index.js    31.00 B
@@ -30,32 +30,32 @@ $ tsup src/ --format cjs,esm --dts
 ESM dist/tts/source.js     112.00 B
 ESM dist/tts/utils.js      395.00 B
 ESM dist/tts/websocket.js  242.00 B
-ESM ⚡️ Build success in 97ms
-CJS dist/index.cjs         21.01 KB
+ESM ⚡️ Build success in 108ms
+CJS dist/index.cjs         21.48 KB
 CJS dist/lib/client.cjs    3.34 KB
 CJS dist/lib/constants.cjs 1.43 KB
-CJS dist/lib/index.cjs     17.18 KB
-CJS dist/react/index.cjs   27.24 KB
+CJS dist/lib/index.cjs     17.45 KB
+CJS dist/react/index.cjs   27.85 KB
 CJS dist/react/utils.cjs   1.80 KB
-CJS dist/tts/index.cjs     15.66 KB
-CJS dist/tts/player.cjs    6.66 KB
+CJS dist/tts/index.cjs     15.93 KB
+CJS dist/tts/player.cjs    6.87 KB
 CJS dist/tts/source.cjs    6.63 KB
 CJS dist/tts/utils.cjs     3.87 KB
-CJS dist/tts/websocket.cjs 15.34 KB
+CJS dist/tts/websocket.cjs 15.62 KB
 CJS dist/types/index.cjs   764.00 B
 CJS dist/voices/index.cjs  5.04 KB
-CJS ⚡️ Build success in 101ms
+CJS ⚡️ Build success in 111ms
 DTS Build start
-DTS ⚡️ Build success in 6609ms
+DTS ⚡️ Build success in 6545ms
 DTS dist/index.d.cts         509.00 B
 DTS dist/lib/constants.d.cts 564.00 B
 DTS dist/lib/index.d.cts     410.00 B
 DTS dist/react/index.d.cts   1018.00 B
 DTS dist/react/utils.d.cts   240.00 B
 DTS dist/tts/index.d.cts     471.00 B
-DTS dist/tts/player.d.cts    1.06 KB
+DTS dist/tts/player.d.cts    1.20 KB
 DTS dist/tts/utils.d.cts     2.56 KB
-DTS dist/tts/websocket.d.cts 2.39 KB
+DTS dist/tts/websocket.d.cts 2.66 KB
 DTS dist/tts/source.d.cts    2.17 KB
 DTS dist/voices/index.d.cts  399.00 B
 DTS dist/lib/client.d.cts    267.00 B
@@ -66,9 +66,9 @@ $ tsup src/ --format cjs,esm --dts
 DTS dist/react/index.d.ts    1016.00 B
 DTS dist/react/utils.d.ts    240.00 B
 DTS dist/tts/index.d.ts      467.00 B
-DTS dist/tts/player.d.ts     1.06 KB
+DTS dist/tts/player.d.ts     1.20 KB
 DTS dist/tts/utils.d.ts      2.55 KB
-DTS dist/tts/websocket.d.ts  2.39 KB
+DTS dist/tts/websocket.d.ts  2.65 KB
 DTS dist/tts/source.d.ts     2.16 KB
 DTS dist/voices/index.d.ts   397.00 B
 DTS dist/lib/client.d.ts     266.00 B

package/CHANGELOG.md

@@ -1,5 +1,28 @@
 # @cartesia/cartesia-js
 
+## 1.0.0
+
+### Major Changes
+
+- 3ee5bfc: Initial release of Cartesia client with voices and WebSocket support
+
+### Minor Changes
+
+- e49f73a: Stabilize audio playback in the browser to support play/pause functionality.
+
+### Patch Changes
+
+- c98a0c7: Fix typo in README
+- 38af01f: Fix how URLs are constructed, solving WebSocket connection failure
+- 8ecf940: Add provisional Node.js support
+- 585d2c9: Makes JS client compatible with the Cartesia Stable API (2024-06-10)
+
+## 1.0.0-alpha.4
+
+### Patch Changes
+
+- c98a0c7: Fix typo in README
+
 ## 1.0.0-alpha.3
 
 ### Patch Changes

package/README.md

@@ -1,4 +1,12 @@
-# JavaScript Client for Cartesia
+# Cartesia JavaScript Client
+
+![NPM Version](https://img.shields.io/npm/v/%40cartesia%2Fcartesia-js?logo=npm)
+[![Discord](https://badgen.net/badge/black/Cartesia/icon?icon=discord&label)](https://discord.gg/ZVxavqHB9X)
+
+This client provides convenient access to [Cartesia's TTS models](https://cartesia.ai/). Sonic is the fastest text-to-speech model around—it can generate a second of audio in just 650ms, and it can stream out the first audio chunk in just 135ms. Alongside Sonic, we also offer an extensive prebuilt voice library for a variety of use cases.
+
+The full API documentation can be found on docs.cartesia.ai.
+
 
 ## Installation
 
@@ -98,10 +106,10 @@ for await (const message of response.events('message')) {
 
 #### Playing audio in the browser
 
-(We currently only support playing audio in the browser. Support for other JS environments is coming soon.)
+(The `WebPlayer` class only supports playing audio in the browser.)
 
 ```js
-// If you're using the client in the browser, you can play the audio like this:
+// If you're using the client in the browser, you can control audio playback using our WebPlayer:
 import { WebPlayer } from "@cartesia/cartesia-js";
 
 console.log("Playing stream...");
@@ -118,7 +126,7 @@ console.log("Done playing.");
 
 ## React
 
-We export a React hook that simplifies the process of using the TTS API. The hook manages the WebSocket connection and provides a simple interface for buffering and playing audio.
+We export a React hook that simplifies the process of using the TTS API. The hook manages the WebSocket connection and provides a simple interface for buffering, playing, pausing and restarting audio.
 
 ```jsx
 import { useTTS } from '@cartesia/cartesia-js/react';
@@ -126,7 +134,7 @@ import { useTTS } from '@cartesia/cartesia-js/react';
 function TextToSpeech() {
 	const tts = useTTS({
 		apiKey: "your-api-key",
-		samplingRate: 44100,
+		sampleRate: 44100,
 	})
 
 	const [text, setText] = useState("");
@@ -134,7 +142,7 @@ function TextToSpeech() {
 	const handlePlay = async () => {
 		// Begin buffering the audio.
 		const response = await tts.buffer({
-			model: "upbeat-moon",
+			model_id: "upbeat-moon",
 			voice: {
 				mode: "embedding",
 				embedding: Array(192).fill(1.0),

package/dist/{chunk-VK7LBMVI.js → chunk-F4QWVJY3.js}

@@ -1,6 +1,6 @@
 import {
   WebSocket
-} from "./chunk-IQAXBRHU.js";
+} from "./chunk-FN7BK4PS.js";
 import {
   Client
 } from "./chunk-PQ5EVEEH.js";

package/dist/{chunk-IQAXBRHU.js → chunk-FN7BK4PS.js}

@@ -51,13 +51,15 @@ var WebSocket = class extends Client {
     __privateSet(this, _sampleRate, sampleRate);
   }
   /**
-   * Send a message over the WebSocket in order to start a stream.
+   * Send a message over the WebSocket to start a stream.
    *
    * @param inputs - Stream options.
    * @param options - Options for the stream.
    * @param options.timeout - The maximum time to wait for a chunk before cancelling the stream.
-   * If `0`, the stream will not time out.
+   *                          If set to `0`, the stream will not time out.
    * @returns A Source object that can be passed to a Player to play the audio.
+   * @returns An Emittery instance that emits messages from the WebSocket.
+   * @returns An abort function that can be called to cancel the stream.
    */
   send(inputs, { timeout = 0 } = {}) {
     var _a, _b, _c, _d;
@@ -128,7 +130,11 @@ var WebSocket = class extends Client {
         clearTimeout(timeoutId);
       }
     });
-    return __spreadValues({ source }, getEmitteryCallbacks(emitter));
+    return __spreadProps(__spreadValues({
+      source
+    }, getEmitteryCallbacks(emitter)), {
+      stop: streamCompleteController.abort.bind(streamCompleteController)
+    });
   }
   /**
    * Authenticate and connect to a Cartesia streaming WebSocket.

package/dist/{chunk-ISRU7PLL.js → chunk-IEN4NCER.js}

@@ -3,7 +3,7 @@ import {
 } from "./chunk-SGXUEFII.js";
 import {
   TTS
-} from "./chunk-VK7LBMVI.js";
+} from "./chunk-F4QWVJY3.js";
 import {
   Client
 } from "./chunk-PQ5EVEEH.js";

package/dist/{chunk-36JBKJUN.js → chunk-JYLAM6VU.js}

@@ -10,8 +10,7 @@ import {
 } from "./chunk-WIFMLPT5.js";
 
 // src/tts/player.ts
-import Emittery from "emittery";
-var _context, _startNextPlaybackAt, _bufferDuration, _emitter, _playBuffer, playBuffer_fn;
+var _context, _startNextPlaybackAt, _bufferDuration, _playBuffer, playBuffer_fn;
 var Player = class {
   /**
    * Create a new Player.
@@ -24,7 +23,6 @@ var Player = class {
     __privateAdd(this, _context, null);
     __privateAdd(this, _startNextPlaybackAt, 0);
     __privateAdd(this, _bufferDuration, void 0);
-    __privateAdd(this, _emitter, new Emittery());
     __privateSet(this, _bufferDuration, bufferDuration);
   }
   /**
@@ -46,7 +44,6 @@ var Player = class {
         const playableAudio = buffer.subarray(0, read);
         plays.push(__privateMethod(this, _playBuffer, playBuffer_fn).call(this, playableAudio, source.sampleRate));
         if (read < buffer.length) {
-          yield __privateGet(this, _emitter).emit("finish");
           break;
         }
       }
@@ -96,17 +93,33 @@ var Player = class {
       }
     });
   }
+  /**
+   * Stop the audio.
+   *
+   * @returns A promise that resolves when the audio has been stopped.
+   */
+  stop() {
+    return __async(this, null, function* () {
+      var _a;
+      if (!__privateGet(this, _context)) {
+        throw new Error("AudioContext not initialized.");
+      }
+      yield (_a = __privateGet(this, _context)) == null ? void 0 : _a.close();
+    });
+  }
 };
 _context = new WeakMap();
 _startNextPlaybackAt = new WeakMap();
 _bufferDuration = new WeakMap();
-_emitter = new WeakMap();
 _playBuffer = new WeakSet();
 playBuffer_fn = function(buf, sampleRate) {
   return __async(this, null, function* () {
     if (!__privateGet(this, _context)) {
       throw new Error("AudioContext not initialized.");
     }
+    if (buf.length === 0) {
+      return;
+    }
     const startAt = __privateGet(this, _startNextPlaybackAt);
     const duration = buf.length / sampleRate;
     __privateSet(this, _startNextPlaybackAt, duration + Math.max(__privateGet(this, _context).currentTime, __privateGet(this, _startNextPlaybackAt)));

package/dist/index.cjs

@@ -336,13 +336,15 @@ var WebSocket = class extends Client {
     __privateSet(this, _sampleRate2, sampleRate);
   }
   /**
-   * Send a message over the WebSocket in order to start a stream.
+   * Send a message over the WebSocket to start a stream.
    *
    * @param inputs - Stream options.
    * @param options - Options for the stream.
    * @param options.timeout - The maximum time to wait for a chunk before cancelling the stream.
-   * If `0`, the stream will not time out.
+   *                          If set to `0`, the stream will not time out.
    * @returns A Source object that can be passed to a Player to play the audio.
+   * @returns An Emittery instance that emits messages from the WebSocket.
+   * @returns An abort function that can be called to cancel the stream.
    */
   send(inputs, { timeout = 0 } = {}) {
     var _a, _b, _c, _d;
@@ -413,7 +415,11 @@ var WebSocket = class extends Client {
         clearTimeout(timeoutId);
       }
     });
-    return __spreadValues({ source }, getEmitteryCallbacks(emitter));
+    return __spreadProps(__spreadValues({
+      source
+    }, getEmitteryCallbacks(emitter)), {
+      stop: streamCompleteController.abort.bind(streamCompleteController)
+    });
   }
   /**
    * Authenticate and connect to a Cartesia streaming WebSocket.
@@ -564,8 +570,7 @@ var Cartesia = class extends Client {
 };
 
 // src/tts/player.ts
-var import_emittery3 = __toESM(require("emittery"), 1);
-var _context, _startNextPlaybackAt, _bufferDuration, _emitter2, _playBuffer, playBuffer_fn;
+var _context, _startNextPlaybackAt, _bufferDuration, _playBuffer, playBuffer_fn;
 var Player = class {
   /**
    * Create a new Player.
@@ -578,7 +583,6 @@ var Player = class {
     __privateAdd(this, _context, null);
     __privateAdd(this, _startNextPlaybackAt, 0);
     __privateAdd(this, _bufferDuration, void 0);
-    __privateAdd(this, _emitter2, new import_emittery3.default());
     __privateSet(this, _bufferDuration, bufferDuration);
   }
   /**
@@ -600,7 +604,6 @@ var Player = class {
         const playableAudio = buffer.subarray(0, read);
         plays.push(__privateMethod(this, _playBuffer, playBuffer_fn).call(this, playableAudio, source.sampleRate));
         if (read < buffer.length) {
-          yield __privateGet(this, _emitter2).emit("finish");
           break;
         }
       }
@@ -650,17 +653,33 @@ var Player = class {
       }
     });
   }
+  /**
+   * Stop the audio.
+   *
+   * @returns A promise that resolves when the audio has been stopped.
+   */
+  stop() {
+    return __async(this, null, function* () {
+      var _a;
+      if (!__privateGet(this, _context)) {
+        throw new Error("AudioContext not initialized.");
+      }
+      yield (_a = __privateGet(this, _context)) == null ? void 0 : _a.close();
+    });
+  }
 };
 _context = new WeakMap();
 _startNextPlaybackAt = new WeakMap();
 _bufferDuration = new WeakMap();
-_emitter2 = new WeakMap();
 _playBuffer = new WeakSet();
 playBuffer_fn = function(buf, sampleRate) {
   return __async(this, null, function* () {
     if (!__privateGet(this, _context)) {
       throw new Error("AudioContext not initialized.");
     }
+    if (buf.length === 0) {
+      return;
+    }
     const startAt = __privateGet(this, _startNextPlaybackAt);
     const duration = buf.length / sampleRate;
     __privateSet(this, _startNextPlaybackAt, duration + Math.max(__privateGet(this, _context).currentTime, __privateGet(this, _startNextPlaybackAt)));

package/dist/index.js

@@ -1,16 +1,16 @@
 import "./chunk-FXPGR372.js";
 import {
   Cartesia
-} from "./chunk-ISRU7PLL.js";
+} from "./chunk-IEN4NCER.js";
 import "./chunk-SGXUEFII.js";
-import "./chunk-VK7LBMVI.js";
-import "./chunk-IQAXBRHU.js";
+import "./chunk-F4QWVJY3.js";
+import "./chunk-FN7BK4PS.js";
 import "./chunk-PQ6CIPFW.js";
 import "./chunk-PQ5EVEEH.js";
 import "./chunk-2BFEKY3F.js";
 import {
   Player
-} from "./chunk-36JBKJUN.js";
+} from "./chunk-JYLAM6VU.js";
 import "./chunk-RO7TY474.js";
 import "./chunk-WIFMLPT5.js";
 export {

package/dist/lib/index.cjs

@@ -321,13 +321,15 @@ var WebSocket = class extends Client {
     __privateSet(this, _sampleRate2, sampleRate);
   }
   /**
-   * Send a message over the WebSocket in order to start a stream.
+   * Send a message over the WebSocket to start a stream.
    *
    * @param inputs - Stream options.
    * @param options - Options for the stream.
    * @param options.timeout - The maximum time to wait for a chunk before cancelling the stream.
-   * If `0`, the stream will not time out.
+   *                          If set to `0`, the stream will not time out.
    * @returns A Source object that can be passed to a Player to play the audio.
+   * @returns An Emittery instance that emits messages from the WebSocket.
+   * @returns An abort function that can be called to cancel the stream.
    */
   send(inputs, { timeout = 0 } = {}) {
     var _a, _b, _c, _d;
@@ -398,7 +400,11 @@ var WebSocket = class extends Client {
         clearTimeout(timeoutId);
       }
     });
-    return __spreadValues({ source }, getEmitteryCallbacks(emitter));
+    return __spreadProps(__spreadValues({
+      source
+    }, getEmitteryCallbacks(emitter)), {
+      stop: streamCompleteController.abort.bind(streamCompleteController)
+    });
   }
   /**
    * Authenticate and connect to a Cartesia streaming WebSocket.

package/dist/lib/index.js

@@ -1,9 +1,9 @@
 import {
   Cartesia
-} from "../chunk-ISRU7PLL.js";
+} from "../chunk-IEN4NCER.js";
 import "../chunk-SGXUEFII.js";
-import "../chunk-VK7LBMVI.js";
-import "../chunk-IQAXBRHU.js";
+import "../chunk-F4QWVJY3.js";
+import "../chunk-FN7BK4PS.js";
 import "../chunk-PQ6CIPFW.js";
 import "../chunk-PQ5EVEEH.js";
 import "../chunk-2BFEKY3F.js";

package/dist/react/index.cjs

@@ -335,13 +335,15 @@ var WebSocket = class extends Client {
     __privateSet(this, _sampleRate2, sampleRate);
   }
   /**
-   * Send a message over the WebSocket in order to start a stream.
+   * Send a message over the WebSocket to start a stream.
    *
    * @param inputs - Stream options.
    * @param options - Options for the stream.
    * @param options.timeout - The maximum time to wait for a chunk before cancelling the stream.
-   * If `0`, the stream will not time out.
+   *                          If set to `0`, the stream will not time out.
    * @returns A Source object that can be passed to a Player to play the audio.
+   * @returns An Emittery instance that emits messages from the WebSocket.
+   * @returns An abort function that can be called to cancel the stream.
    */
   send(inputs, { timeout = 0 } = {}) {
     var _a, _b, _c, _d;
@@ -412,7 +414,11 @@ var WebSocket = class extends Client {
         clearTimeout(timeoutId);
       }
     });
-    return __spreadValues({ source }, getEmitteryCallbacks(emitter));
+    return __spreadProps(__spreadValues({
+      source
+    }, getEmitteryCallbacks(emitter)), {
+      stop: streamCompleteController.abort.bind(streamCompleteController)
+    });
   }
   /**
    * Authenticate and connect to a Cartesia streaming WebSocket.
@@ -563,8 +569,7 @@ var Cartesia = class extends Client {
 };
 
 // src/tts/player.ts
-var import_emittery3 = __toESM(require("emittery"), 1);
-var _context, _startNextPlaybackAt, _bufferDuration, _emitter2, _playBuffer, playBuffer_fn;
+var _context, _startNextPlaybackAt, _bufferDuration, _playBuffer, playBuffer_fn;
 var Player = class {
   /**
    * Create a new Player.
@@ -577,7 +582,6 @@ var Player = class {
     __privateAdd(this, _context, null);
     __privateAdd(this, _startNextPlaybackAt, 0);
     __privateAdd(this, _bufferDuration, void 0);
-    __privateAdd(this, _emitter2, new import_emittery3.default());
     __privateSet(this, _bufferDuration, bufferDuration);
   }
   /**
@@ -599,7 +603,6 @@ var Player = class {
         const playableAudio = buffer.subarray(0, read);
         plays.push(__privateMethod(this, _playBuffer, playBuffer_fn).call(this, playableAudio, source.sampleRate));
         if (read < buffer.length) {
-          yield __privateGet(this, _emitter2).emit("finish");
           break;
         }
       }
@@ -649,17 +652,33 @@ var Player = class {
       }
     });
   }
+  /**
+   * Stop the audio.
+   *
+   * @returns A promise that resolves when the audio has been stopped.
+   */
+  stop() {
+    return __async(this, null, function* () {
+      var _a;
+      if (!__privateGet(this, _context)) {
+        throw new Error("AudioContext not initialized.");
+      }
+      yield (_a = __privateGet(this, _context)) == null ? void 0 : _a.close();
+    });
+  }
 };
 _context = new WeakMap();
 _startNextPlaybackAt = new WeakMap();
 _bufferDuration = new WeakMap();
-_emitter2 = new WeakMap();
 _playBuffer = new WeakSet();
 playBuffer_fn = function(buf, sampleRate) {
   return __async(this, null, function* () {
     if (!__privateGet(this, _context)) {
       throw new Error("AudioContext not initialized.");
     }
+    if (buf.length === 0) {
+      return;
+    }
     const startAt = __privateGet(this, _startNextPlaybackAt);
     const duration = buf.length / sampleRate;
     __privateSet(this, _startNextPlaybackAt, duration + Math.max(__privateGet(this, _context).currentTime, __privateGet(this, _startNextPlaybackAt)));
@@ -726,10 +745,11 @@ function useTTS({
   const [messages, setMessages] = (0, import_react.useState)([]);
   const buffer = (0, import_react.useCallback)(
     (options) => __async(this, null, function* () {
-      var _a2;
+      var _a2, _b2;
+      (_a2 = websocketReturn.current) == null ? void 0 : _a2.stop();
       setMessages([]);
       setBufferStatus("buffering");
-      websocketReturn.current = (_a2 = websocket == null ? void 0 : websocket.send(options)) != null ? _a2 : null;
+      websocketReturn.current = (_b2 = websocket == null ? void 0 : websocket.send(options)) != null ? _b2 : null;
       if (!websocketReturn.current) {
         return;
       }
@@ -812,6 +832,9 @@ function useTTS({
     if (playbackStatus === "playing" || !websocketReturn.current) {
       return;
     }
+    if (player.current) {
+      yield player.current.stop();
+    }
     setPlaybackStatus("playing");
     const unsubscribes = [];
     unsubscribes.push(

package/dist/react/index.js

@@ -1,18 +1,18 @@
 import {
   Cartesia
-} from "../chunk-ISRU7PLL.js";
+} from "../chunk-IEN4NCER.js";
 import "../chunk-SGXUEFII.js";
 import {
   pingServer
 } from "../chunk-3FL2SNIR.js";
-import "../chunk-VK7LBMVI.js";
-import "../chunk-IQAXBRHU.js";
+import "../chunk-F4QWVJY3.js";
+import "../chunk-FN7BK4PS.js";
 import "../chunk-PQ6CIPFW.js";
 import "../chunk-PQ5EVEEH.js";
 import "../chunk-2BFEKY3F.js";
 import {
   Player
-} from "../chunk-36JBKJUN.js";
+} from "../chunk-JYLAM6VU.js";
 import "../chunk-RO7TY474.js";
 import {
   __async
@@ -68,10 +68,11 @@ function useTTS({
   const [messages, setMessages] = useState([]);
   const buffer = useCallback(
     (options) => __async(this, null, function* () {
-      var _a2;
+      var _a2, _b2;
+      (_a2 = websocketReturn.current) == null ? void 0 : _a2.stop();
       setMessages([]);
       setBufferStatus("buffering");
-      websocketReturn.current = (_a2 = websocket == null ? void 0 : websocket.send(options)) != null ? _a2 : null;
+      websocketReturn.current = (_b2 = websocket == null ? void 0 : websocket.send(options)) != null ? _b2 : null;
       if (!websocketReturn.current) {
         return;
       }
@@ -154,6 +155,9 @@ function useTTS({
     if (playbackStatus === "playing" || !websocketReturn.current) {
       return;
     }
+    if (player.current) {
+      yield player.current.stop();
+    }
     setPlaybackStatus("playing");
     const unsubscribes = [];
     unsubscribes.push(

package/dist/tts/index.cjs

@@ -321,13 +321,15 @@ var WebSocket = class extends Client {
     __privateSet(this, _sampleRate2, sampleRate);
   }
   /**
-   * Send a message over the WebSocket in order to start a stream.
+   * Send a message over the WebSocket to start a stream.
    *
    * @param inputs - Stream options.
    * @param options - Options for the stream.
    * @param options.timeout - The maximum time to wait for a chunk before cancelling the stream.
-   * If `0`, the stream will not time out.
+   *                          If set to `0`, the stream will not time out.
    * @returns A Source object that can be passed to a Player to play the audio.
+   * @returns An Emittery instance that emits messages from the WebSocket.
+   * @returns An abort function that can be called to cancel the stream.
    */
   send(inputs, { timeout = 0 } = {}) {
     var _a, _b, _c, _d;
@@ -398,7 +400,11 @@ var WebSocket = class extends Client {
         clearTimeout(timeoutId);
       }
     });
-    return __spreadValues({ source }, getEmitteryCallbacks(emitter));
+    return __spreadProps(__spreadValues({
+      source
+    }, getEmitteryCallbacks(emitter)), {
+      stop: streamCompleteController.abort.bind(streamCompleteController)
+    });
   }
   /**
    * Authenticate and connect to a Cartesia streaming WebSocket.

package/dist/tts/index.js

@@ -1,7 +1,7 @@
 import {
   TTS
-} from "../chunk-VK7LBMVI.js";
-import "../chunk-IQAXBRHU.js";
+} from "../chunk-F4QWVJY3.js";
+import "../chunk-FN7BK4PS.js";
 import "../chunk-PQ6CIPFW.js";
 import "../chunk-PQ5EVEEH.js";
 import "../chunk-2BFEKY3F.js";

package/dist/tts/player.cjs

@@ -75,7 +75,6 @@ __export(player_exports, {
   default: () => Player
 });
 module.exports = __toCommonJS(player_exports);
-var import_emittery = __toESM(require("emittery"), 1);
 
 // src/tts/utils.ts
 var import_base64_js = __toESM(require("base64-js"), 1);
@@ -94,7 +93,7 @@ function playAudioBuffer(floats, context, startAt, sampleRate) {
 }
 
 // src/tts/player.ts
-var _context, _startNextPlaybackAt, _bufferDuration, _emitter, _playBuffer, playBuffer_fn;
+var _context, _startNextPlaybackAt, _bufferDuration, _playBuffer, playBuffer_fn;
 var Player = class {
   /**
    * Create a new Player.
@@ -107,7 +106,6 @@ var Player = class {
     __privateAdd(this, _context, null);
     __privateAdd(this, _startNextPlaybackAt, 0);
     __privateAdd(this, _bufferDuration, void 0);
-    __privateAdd(this, _emitter, new import_emittery.default());
     __privateSet(this, _bufferDuration, bufferDuration);
   }
   /**
@@ -129,7 +127,6 @@ var Player = class {
         const playableAudio = buffer.subarray(0, read);
         plays.push(__privateMethod(this, _playBuffer, playBuffer_fn).call(this, playableAudio, source.sampleRate));
         if (read < buffer.length) {
-          yield __privateGet(this, _emitter).emit("finish");
           break;
         }
       }
@@ -179,17 +176,33 @@ var Player = class {
       }
     });
   }
+  /**
+   * Stop the audio.
+   *
+   * @returns A promise that resolves when the audio has been stopped.
+   */
+  stop() {
+    return __async(this, null, function* () {
+      var _a;
+      if (!__privateGet(this, _context)) {
+        throw new Error("AudioContext not initialized.");
+      }
+      yield (_a = __privateGet(this, _context)) == null ? void 0 : _a.close();
+    });
+  }
 };
 _context = new WeakMap();
 _startNextPlaybackAt = new WeakMap();
 _bufferDuration = new WeakMap();
-_emitter = new WeakMap();
 _playBuffer = new WeakSet();
 playBuffer_fn = function(buf, sampleRate) {
   return __async(this, null, function* () {
     if (!__privateGet(this, _context)) {
       throw new Error("AudioContext not initialized.");
     }
+    if (buf.length === 0) {
+      return;
+    }
     const startAt = __privateGet(this, _startNextPlaybackAt);
     const duration = buf.length / sampleRate;
     __privateSet(this, _startNextPlaybackAt, duration + Math.max(__privateGet(this, _context).currentTime, __privateGet(this, _startNextPlaybackAt)));

package/dist/tts/player.d.cts

@@ -38,6 +38,12 @@ declare class Player {
      * @returns A promise that resolves when the audio has been toggled.
      */
     toggle(): Promise<void>;
+    /**
+     * Stop the audio.
+     *
+     * @returns A promise that resolves when the audio has been stopped.
+     */
+    stop(): Promise<void>;
 }
 
 export { Player as default };

package/dist/tts/player.d.ts

@@ -38,6 +38,12 @@ declare class Player {
      * @returns A promise that resolves when the audio has been toggled.
      */
     toggle(): Promise<void>;
+    /**
+     * Stop the audio.
+     *
+     * @returns A promise that resolves when the audio has been stopped.
+     */
+    stop(): Promise<void>;
 }
 
 export { Player as default };

package/dist/tts/player.js

@@ -1,6 +1,6 @@
 import {
   Player
-} from "../chunk-36JBKJUN.js";
+} from "../chunk-JYLAM6VU.js";
 import "../chunk-RO7TY474.js";
 import "../chunk-WIFMLPT5.js";
 export {

package/dist/tts/websocket.cjs

@@ -319,13 +319,15 @@ var WebSocket = class extends Client {
     __privateSet(this, _sampleRate2, sampleRate);
   }
   /**
-   * Send a message over the WebSocket in order to start a stream.
+   * Send a message over the WebSocket to start a stream.
    *
    * @param inputs - Stream options.
    * @param options - Options for the stream.
    * @param options.timeout - The maximum time to wait for a chunk before cancelling the stream.
-   * If `0`, the stream will not time out.
+   *                          If set to `0`, the stream will not time out.
    * @returns A Source object that can be passed to a Player to play the audio.
+   * @returns An Emittery instance that emits messages from the WebSocket.
+   * @returns An abort function that can be called to cancel the stream.
    */
   send(inputs, { timeout = 0 } = {}) {
     var _a, _b, _c, _d;
@@ -396,7 +398,11 @@ var WebSocket = class extends Client {
         clearTimeout(timeoutId);
       }
     });
-    return __spreadValues({ source }, getEmitteryCallbacks(emitter));
+    return __spreadProps(__spreadValues({
+      source
+    }, getEmitteryCallbacks(emitter)), {
+      stop: streamCompleteController.abort.bind(streamCompleteController)
+    });
   }
   /**
    * Authenticate and connect to a Cartesia streaming WebSocket.

package/dist/tts/websocket.d.cts

@@ -14,15 +14,21 @@ declare class WebSocket extends Client {
      */
     constructor({ sampleRate }: WebSocketOptions, ...args: ConstructorParameters<typeof Client>);
     /**
-     * Send a message over the WebSocket in order to start a stream.
+     * Send a message over the WebSocket to start a stream.
      *
      * @param inputs - Stream options.
      * @param options - Options for the stream.
      * @param options.timeout - The maximum time to wait for a chunk before cancelling the stream.
-     * If `0`, the stream will not time out.
+     *                          If set to `0`, the stream will not time out.
      * @returns A Source object that can be passed to a Player to play the audio.
+     * @returns An Emittery instance that emits messages from the WebSocket.
+     * @returns An abort function that can be called to cancel the stream.
      */
     send(inputs: StreamRequest["inputs"], { timeout }?: StreamRequest["options"]): {
+        stop: {
+            (reason?: any): void;
+            (reason?: any): void;
+        };
         on: <Name extends keyof emittery.OmnipresentEventData | "message">(eventName: Name | readonly Name[], listener: (eventData: ({
             message: string;
         } & emittery.OmnipresentEventData)[Name]) => void | Promise<void>) => emittery.UnsubscribeFunction;

package/dist/tts/websocket.d.ts

@@ -14,15 +14,21 @@ declare class WebSocket extends Client {
      */
     constructor({ sampleRate }: WebSocketOptions, ...args: ConstructorParameters<typeof Client>);
     /**
-     * Send a message over the WebSocket in order to start a stream.
+     * Send a message over the WebSocket to start a stream.
      *
      * @param inputs - Stream options.
      * @param options - Options for the stream.
      * @param options.timeout - The maximum time to wait for a chunk before cancelling the stream.
-     * If `0`, the stream will not time out.
+     *                          If set to `0`, the stream will not time out.
      * @returns A Source object that can be passed to a Player to play the audio.
+     * @returns An Emittery instance that emits messages from the WebSocket.
+     * @returns An abort function that can be called to cancel the stream.
      */
     send(inputs: StreamRequest["inputs"], { timeout }?: StreamRequest["options"]): {
+        stop: {
+            (reason?: any): void;
+            (reason?: any): void;
+        };
         on: <Name extends keyof emittery.OmnipresentEventData | "message">(eventName: Name | readonly Name[], listener: (eventData: ({
             message: string;
         } & emittery.OmnipresentEventData)[Name]) => void | Promise<void>) => emittery.UnsubscribeFunction;

package/dist/tts/websocket.js

@@ -1,6 +1,6 @@
 import {
   WebSocket
-} from "../chunk-IQAXBRHU.js";
+} from "../chunk-FN7BK4PS.js";
 import "../chunk-PQ6CIPFW.js";
 import "../chunk-PQ5EVEEH.js";
 import "../chunk-2BFEKY3F.js";

package/package.json

@@ -4,7 +4,7 @@
 		"name": "Cartesia",
 		"url": "https://cartesia.ai"
 	},
-	"version": "1.0.0-alpha.3",
+	"version": "1.0.0",
 	"description": "Client for the Cartesia API.",
 	"type": "module",
 	"module": "./dist/index.js",

package/src/react/index.ts

@@ -86,6 +86,8 @@ export function useTTS({
 
 	const buffer = useCallback(
 		async (options: object) => {
+			websocketReturn.current?.stop(); // Abort the previous request if it exists.
+
 			setMessages([]);
 			setBufferStatus("buffering");
 			websocketReturn.current = websocket?.send(options) ?? null;
@@ -174,6 +176,11 @@ export function useTTS({
 		if (playbackStatus === "playing" || !websocketReturn.current) {
 			return;
 		}
+		if (player.current) {
+			// Stop the current player if it exists.
+			await player.current.stop();
+		}
+
 		setPlaybackStatus("playing");
 
 		const unsubscribes = [];
@@ -225,9 +232,6 @@ export function useTTS({
 		});
 	}, []);
 
-	// TODO:
-	// - [] Access the play and buffer cursors.
-	// - [] Seek to a specific time.
 	return {
 		buffer,
 		play,

package/src/tts/player.ts

@@ -1,16 +1,10 @@
-import Emittery from "emittery";
 import type Source from "./source";
 import { playAudioBuffer } from "./utils";
 
-type PlayEventData = {
-	finish: never;
-};
-
 export default class Player {
 	#context: AudioContext | null = null;
 	#startNextPlaybackAt = 0;
 	#bufferDuration: number;
-	#emitter = new Emittery<PlayEventData>();
 
 	/**
 	 * Create a new Player.
@@ -26,6 +20,9 @@ export default class Player {
 		if (!this.#context) {
 			throw new Error("AudioContext not initialized.");
 		}
+		if (buf.length === 0) {
+			return;
+		}
 
 		const startAt = this.#startNextPlaybackAt;
 		const duration = buf.length / sampleRate;
@@ -60,11 +57,9 @@ export default class Player {
 
 			if (read < buffer.length) {
 				// No more audio to read.
-				await this.#emitter.emit("finish");
 				break;
 			}
 		}
-
 		await Promise.all(plays);
 	}
 
@@ -107,4 +102,16 @@ export default class Player {
 			await this.resume();
 		}
 	}
+
+	/**
+	 * Stop the audio.
+	 *
+	 * @returns A promise that resolves when the audio has been stopped.
+	 */
+	async stop() {
+		if (!this.#context) {
+			throw new Error("AudioContext not initialized.");
+		}
+		await this.#context?.close();
+	}
 }

package/src/tts/websocket.ts

@@ -37,13 +37,15 @@ export default class WebSocket extends Client {
 	}
 
 	/**
-	 * Send a message over the WebSocket in order to start a stream.
+	 * Send a message over the WebSocket to start a stream.
 	 *
 	 * @param inputs - Stream options.
 	 * @param options - Options for the stream.
 	 * @param options.timeout - The maximum time to wait for a chunk before cancelling the stream.
-	 * If `0`, the stream will not time out.
+	 *                          If set to `0`, the stream will not time out.
 	 * @returns A Source object that can be passed to a Player to play the audio.
+	 * @returns An Emittery instance that emits messages from the WebSocket.
+	 * @returns An abort function that can be called to cancel the stream.
 	 */
 	send(
 		inputs: StreamRequest["inputs"],
@@ -125,7 +127,11 @@ export default class WebSocket extends Client {
 			}
 		});
 
-		return { source, ...getEmitteryCallbacks(emitter) };
+		return {
+			source,
+			...getEmitteryCallbacks(emitter),
+			stop: streamCompleteController.abort.bind(streamCompleteController),
+		};
 	}
 
 	/**