@amaster.ai/asr-client 1.0.0-beta.33 → 1.0.0-beta.7

package/dist/index.cjs

@@ -29,16 +29,11 @@ module.exports = __toCommonJS(index_exports);
 // src/asr-client.ts
 var ASR_PATH = "/api/proxy/builtin/platform/qwen-asr-realtime/api-ws/v1/realtime";
 async function listen(onTranscript) {
-  if (typeof navigator === "undefined" || !navigator.mediaDevices?.getUserMedia) {
-    throw new Error("Microphone not supported. Requires HTTPS and browser with MediaDevices API.");
-  }
   const ws = new WebSocket(ASR_PATH);
   let mediaStream = null;
   let audioContext = null;
   let processor = null;
-  let isRunning = true;
   const stop = () => {
-    isRunning = false;
     if (mediaStream) {
       mediaStream.getTracks().forEach((t) => t.stop());
       mediaStream = null;
@@ -52,7 +47,8 @@ async function listen(onTranscript) {
       audioContext = null;
     }
     if (ws.readyState === WebSocket.OPEN) {
-      ws.send(JSON.stringify({ event_id: `event_${Date.now()}`, type: "session.finish" }));
+      ws.send(JSON.stringify({ type: "input_audio_buffer.commit" }));
+      ws.close();
     }
   };
   return new Promise((resolve, reject) => {
@@ -67,7 +63,7 @@ async function listen(onTranscript) {
           const source = audioContext.createMediaStreamSource(mediaStream);
           processor = audioContext.createScriptProcessor(4096, 1, 1);
           processor.onaudioprocess = (e) => {
-            if (!isRunning || ws.readyState !== WebSocket.OPEN) return;
+            if (ws.readyState !== WebSocket.OPEN) return;
             const input = e.inputBuffer.getChannelData(0);
             const pcm = new Int16Array(input.length);
             for (let i = 0; i < input.length; i++) {
@@ -87,16 +83,10 @@ async function listen(onTranscript) {
         }
       }
       if (data.type === "conversation.item.input_audio_transcription.text") {
-        onTranscript(data.text || data.stash || "", false);
+        onTranscript(data.text || "", false);
       }
       if (data.type === "conversation.item.input_audio_transcription.completed") {
-        onTranscript(data.transcript || data.text || "", true);
-      }
-      if (data.type === "session.finished") {
-        if (data.transcript) {
-          onTranscript(data.transcript, true);
-        }
-        ws.close(1e3, "ASR finished");
+        onTranscript(data.text || data.transcript || "", true);
       }
       if (data.type === "error") {
         stop();

package/dist/index.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/index.ts","../src/asr-client.ts"],"sourcesContent":["export { listen } from './asr-client';\nexport type { ASRClient, ASRClientConfig } from './asr-client';\nexport { createASRClient, createASRClient as createAsrClient } from './asr-client';\n","/**\n * ASR Realtime WebSocket Client\n */\n\nconst ASR_PATH = '/api/proxy/builtin/platform/qwen-asr-realtime/api-ws/v1/realtime';\n\n/**\n * Simple ASR: start listening and get transcript\n * @returns stop function\n * @example\n * const stop = await listen((text, isFinal) => console.log(text))\n * // later: stop()\n */\nexport async function listen(\n  onTranscript: (text: string, isFinal: boolean) => void\n): Promise<() => void> {\n  // Check browser support first\n  if (typeof navigator === 'undefined' || !navigator.mediaDevices?.getUserMedia) {\n    throw new Error('Microphone not supported. Requires HTTPS and browser with MediaDevices API.');\n  }\n\n  const ws = new WebSocket(ASR_PATH);\n  let mediaStream: MediaStream | null = null;\n  let audioContext: AudioContext | null = null;\n  let processor: ScriptProcessorNode | null = null;\n  let isRunning = true;\n\n  const stop = () => {\n    isRunning = false;\n    if (mediaStream) { mediaStream.getTracks().forEach(t => t.stop()); mediaStream = null; }\n    if (processor) { processor.disconnect(); processor = null; }\n    if (audioContext) { audioContext.close(); audioContext = null; }\n    if (ws.readyState === WebSocket.OPEN) {\n      // Send session.finish to get final result\n      ws.send(JSON.stringify({ event_id: `event_${Date.now()}`, type: 'session.finish' }));\n    }\n  };\n\n  return new Promise((resolve, reject) => {\n    ws.onmessage = async (event) => {\n      const data = JSON.parse(event.data);\n\n      // Start recording immediately on session.created (use default config)\n      if (data.type === 'session.created') {\n        try {\n          mediaStream = await navigator.mediaDevices.getUserMedia({\n            audio: { sampleRate: 16000, channelCount: 1, echoCancellation: true }\n          });\n          audioContext = new AudioContext({ sampleRate: 16000 });\n          const source = audioContext.createMediaStreamSource(mediaStream);\n          processor = audioContext.createScriptProcessor(4096, 1, 1);\n\n          processor.onaudioprocess = (e) => {\n            if (!isRunning || ws.readyState !== WebSocket.OPEN) return;\n            const input = e.inputBuffer.getChannelData(0);\n            const pcm = new Int16Array(input.length);\n            for (let i = 0; i < input.length; i++) {\n              const s = Math.max(-1, Math.min(1, input[i]!));\n              pcm[i] = s < 0 ? s * 32768 : s * 32767;\n            }\n            const bytes = new Uint8Array(pcm.buffer);\n            let binary = '';\n            for (let i = 0; i < bytes.length; i++) binary += String.fromCharCode(bytes[i]!);\n            ws.send(JSON.stringify({ type: 'input_audio_buffer.append', audio: btoa(binary) }));\n          };\n\n          source.connect(processor);\n          processor.connect(audioContext.destination);\n          resolve(stop);\n        } catch (err) {\n          reject(err);\n        }\n      }\n\n      // Interim result\n      if (data.type === 'conversation.item.input_audio_transcription.text') {\n        onTranscript(data.text || data.stash || '', false);\n      }\n\n      // Sentence complete (VAD detected pause)\n      if (data.type === 'conversation.item.input_audio_transcription.completed') {\n        onTranscript(data.transcript || data.text || '', true);\n      }\n\n      // Final result after session.finish\n      if (data.type === 'session.finished') {\n        if (data.transcript) {\n          onTranscript(data.transcript, true);\n        }\n        ws.close(1000, 'ASR finished');\n      }\n\n      if (data.type === 'error') {\n        stop();\n        reject(new Error(data.error?.message || 'ASR error'));\n      }\n    };\n\n    ws.onerror = () => reject(new Error('ASR connection failed'));\n  });\n}\n\nexport interface ASRClientConfig {\n  /** Audio format, default 'pcm16' */\n  audioFormat?: 'pcm16' | 'g711a' | 'g711u';\n  /** Sample rate, default 16000 */\n  sampleRate?: number;\n  /** Called when connection is ready */\n  onReady?: () => void;\n  /** Called when speech is detected */\n  onSpeechStart?: () => void;\n  /** Called when speech stops */\n  onSpeechEnd?: () => void;\n  /** Called on transcript result */\n  onTranscript?: (text: string, isFinal: boolean) => void;\n  /** Called on error */\n  onError?: (error: Error) => void;\n}\n\nexport interface ASRClient {\n  /** Connect to ASR service */\n  connect(): Promise<void>;\n  /** Start recording from microphone */\n  startRecording(): Promise<void>;\n  /** Stop recording */\n  stopRecording(): void;\n  /** Close connection */\n  close(): void;\n}\n\nexport function createASRClient(config: ASRClientConfig): ASRClient {\n  const {\n    // audioFormat = 'pcm16',\n    sampleRate = 16000,\n    onReady,\n    onSpeechStart,\n    onSpeechEnd,\n    onTranscript,\n    onError,\n  } = config;\n\n  let ws: WebSocket | null = null;\n  let mediaStream: MediaStream | null = null;\n  let audioContext: AudioContext | null = null;\n  let processor: ScriptProcessorNode | null = null;\n\n  async function connect(): Promise<void> {\n    return new Promise((resolve, reject) => {\n      ws = new WebSocket(ASR_PATH);\n\n      ws.onopen = () => {};\n\n      ws.onmessage = (event) => {\n        const data = JSON.parse(event.data);\n\n        if (data.type === 'session.created') {\n          onReady?.();\n          resolve();\n        }\n\n        if (data.type === 'input_audio_buffer.speech_started') {\n          onSpeechStart?.();\n        }\n\n        if (data.type === 'input_audio_buffer.speech_stopped') {\n          onSpeechEnd?.();\n        }\n\n        if (data.type === 'conversation.item.input_audio_transcription.text') {\n          onTranscript?.(data.text || '', false);\n        }\n\n        if (data.type === 'conversation.item.input_audio_transcription.completed') {\n          onTranscript?.(data.text || data.transcript || '', true);\n        }\n\n        if (data.type === 'error') {\n          const err = new Error(data.error?.message || 'Unknown error');\n          onError?.(err);\n          reject(err);\n        }\n      };\n\n      ws.onerror = () => {\n        const err = new Error('WebSocket connection error');\n        onError?.(err);\n        reject(err);\n      };\n\n      ws.onclose = () => {\n        ws = null;\n      };\n    });\n  }\n\n  async function startRecording(): Promise<void> {\n    if (typeof window === 'undefined') {\n      throw new Error('Recording only supported in browser');\n    }\n\n    if (!ws || ws.readyState !== WebSocket.OPEN) {\n      throw new Error('WebSocket not connected');\n    }\n\n    try {\n      mediaStream = await navigator.mediaDevices.getUserMedia({\n        audio: {\n          sampleRate,\n          channelCount: 1,\n          echoCancellation: true,\n          noiseSuppression: true,\n        },\n      });\n\n      audioContext = new AudioContext({ sampleRate });\n      const source = audioContext.createMediaStreamSource(mediaStream);\n      processor = audioContext.createScriptProcessor(4096, 1, 1);\n\n      processor.onaudioprocess = (e) => {\n        if (!ws || ws.readyState !== WebSocket.OPEN) return;\n\n        const inputData = e.inputBuffer.getChannelData(0);\n        const inputLen = inputData.length;\n\n        const pcm = new Int16Array(inputLen);\n        for (let i = 0; i < inputLen; i++) {\n          const s = Math.max(-1, Math.min(1, inputData[i]!));\n          pcm[i] = s < 0 ? s * 32768 : s * 32767;\n        }\n\n        const bytes = new Uint8Array(pcm.buffer);\n        const len = bytes.length;\n        let binary = '';\n        for (let i = 0; i < len; i++) {\n          binary += String.fromCharCode(bytes[i]!);\n        }\n        const base64 = btoa(binary);\n\n        ws.send(JSON.stringify({\n          type: 'input_audio_buffer.append',\n          audio: base64,\n        }));\n      };\n\n      source.connect(processor);\n      processor.connect(audioContext.destination);\n    } catch (err) {\n      onError?.(err as Error);\n      throw err;\n    }\n  }\n\n  function stopRecording() {\n    if (mediaStream) {\n      mediaStream.getTracks().forEach(track => track.stop());\n      mediaStream = null;\n    }\n\n    if (processor) {\n      processor.disconnect();\n      processor = null;\n    }\n\n    if (audioContext) {\n      audioContext.close();\n      audioContext = null;\n    }\n\n    if (ws && ws.readyState === WebSocket.OPEN) {\n      ws.send(JSON.stringify({ type: 'input_audio_buffer.commit' }));\n    }\n  }\n\n  function close() {\n    stopRecording();\n    if (ws) {\n      ws.close();\n      ws = null;\n  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+{"version":3,"sources":["../src/index.ts","../src/asr-client.ts"],"sourcesContent":["export { listen } from './asr-client';\nexport type { ASRClient, ASRClientConfig } from './asr-client';\nexport { createASRClient, createASRClient as createAsrClient } from './asr-client';\n","/**\r\n * ASR Realtime WebSocket Client\r\n */\r\n\r\nconst ASR_PATH = '/api/proxy/builtin/platform/qwen-asr-realtime/api-ws/v1/realtime';\r\n\r\n/**\r\n * Simple ASR: start listening and get transcript\r\n * @returns stop function\r\n * @example\r\n * const stop = await listen((text, isFinal) => console.log(text))\r\n * // later: stop()\r\n */\r\nexport async function listen(\r\n  onTranscript: (text: string, isFinal: boolean) => void\r\n): Promise<() => void> {\r\n  const ws = new WebSocket(ASR_PATH);\r\n  let mediaStream: MediaStream | null = null;\r\n  let audioContext: AudioContext | null = null;\r\n  let processor: ScriptProcessorNode | null = null;\r\n\r\n  const stop = () => {\r\n    if (mediaStream) { mediaStream.getTracks().forEach(t => t.stop()); mediaStream = null; }\r\n    if (processor) { processor.disconnect(); processor = null; }\r\n    if (audioContext) { audioContext.close(); audioContext = null; }\r\n    if (ws.readyState === WebSocket.OPEN) {\r\n      ws.send(JSON.stringify({ type: 'input_audio_buffer.commit' }));\r\n      ws.close();\r\n    }\r\n  };\r\n\r\n  return new Promise((resolve, reject) => {\r\n    ws.onmessage = async (event) => {\r\n      const data = JSON.parse(event.data);\r\n\r\n      if (data.type === 'session.created') {\r\n        // Start recording\r\n        try {\r\n          mediaStream = await navigator.mediaDevices.getUserMedia({\r\n            audio: { sampleRate: 16000, channelCount: 1, echoCancellation: true }\r\n          });\r\n          audioContext = new AudioContext({ sampleRate: 16000 });\r\n          const source = audioContext.createMediaStreamSource(mediaStream);\r\n          processor = audioContext.createScriptProcessor(4096, 1, 1);\r\n\r\n          processor.onaudioprocess = (e) => {\r\n            if (ws.readyState !== WebSocket.OPEN) return;\r\n            const input = e.inputBuffer.getChannelData(0);\r\n            const pcm = new Int16Array(input.length);\r\n            for (let i = 0; i < input.length; i++) {\r\n              const s = Math.max(-1, Math.min(1, input[i]!));\r\n              pcm[i] = s < 0 ? s * 32768 : s * 32767;\r\n            }\r\n            const bytes = new Uint8Array(pcm.buffer);\r\n            let binary = '';\r\n            for (let i = 0; i < bytes.length; i++) binary += String.fromCharCode(bytes[i]!);\r\n            ws.send(JSON.stringify({ type: 'input_audio_buffer.append', audio: btoa(binary) }));\r\n          };\r\n\r\n          source.connect(processor);\r\n          processor.connect(audioContext.destination);\r\n          resolve(stop);\r\n        } catch (err) {\r\n          reject(err);\r\n        }\r\n      }\r\n\r\n      if (data.type === 'conversation.item.input_audio_transcription.text') {\r\n        onTranscript(data.text || '', false);\r\n      }\r\n\r\n      if (data.type === 'conversation.item.input_audio_transcription.completed') {\r\n        onTranscript(data.text || data.transcript || '', true);\r\n      }\r\n\r\n      if (data.type === 'error') {\r\n        stop();\r\n        reject(new Error(data.error?.message || 'ASR error'));\r\n      }\r\n    };\r\n\r\n    ws.onerror = () => reject(new Error('ASR connection failed'));\r\n  });\r\n}\r\n\r\nexport interface ASRClientConfig {\r\n  /** Audio format, default 'pcm16' */\r\n  audioFormat?: 'pcm16' | 'g711a' | 'g711u';\r\n  /** Sample rate, default 16000 */\r\n  sampleRate?: number;\r\n  /** Called when connection is ready */\r\n  onReady?: () => void;\r\n  /** Called when speech is detected */\r\n  onSpeechStart?: () => void;\r\n  /** Called when speech stops */\r\n  onSpeechEnd?: () => void;\r\n  /** Called on transcript result */\r\n  onTranscript?: (text: string, isFinal: boolean) => void;\r\n  /** Called on error */\r\n  onError?: (error: Error) => void;\r\n}\r\n\r\nexport interface ASRClient {\r\n  /** Connect to ASR service */\r\n  connect(): Promise<void>;\r\n  /** Start recording from microphone */\r\n  startRecording(): Promise<void>;\r\n  /** Stop recording */\r\n  stopRecording(): void;\r\n  /** Close connection */\r\n  close(): void;\r\n}\r\n\r\nexport function createASRClient(config: ASRClientConfig): ASRClient {\r\n  const {\r\n    // audioFormat = 'pcm16',\r\n    sampleRate = 16000,\r\n    onReady,\r\n    onSpeechStart,\r\n    onSpeechEnd,\r\n    onTranscript,\r\n    onError,\r\n  } = config;\r\n\r\n  let ws: WebSocket | null = null;\r\n  let mediaStream: MediaStream | null = null;\r\n  let audioContext: AudioContext | null = null;\r\n  let processor: ScriptProcessorNode | null = null;\r\n\r\n  async function connect(): Promise<void> {\r\n    return new Promise((resolve, reject) => {\r\n      ws = new WebSocket(ASR_PATH);\r\n\r\n      ws.onopen = () => {};\r\n\r\n      ws.onmessage = (event) => {\r\n        const data = JSON.parse(event.data);\r\n\r\n        if (data.type === 'session.created') {\r\n          onReady?.();\r\n          resolve();\r\n        }\r\n\r\n        if (data.type === 'input_audio_buffer.speech_started') {\r\n          onSpeechStart?.();\r\n        }\r\n\r\n        if (data.type === 'input_audio_buffer.speech_stopped') {\r\n          onSpeechEnd?.();\r\n        }\r\n\r\n        if (data.type === 'conversation.item.input_audio_transcription.text') {\r\n          onTranscript?.(data.text || '', false);\r\n        }\r\n\r\n        if (data.type === 'conversation.item.input_audio_transcription.completed') {\r\n          onTranscript?.(data.text || data.transcript || '', true);\r\n        }\r\n\r\n        if (data.type === 'error') {\r\n          const err = new Error(data.error?.message || 'Unknown error');\r\n          onError?.(err);\r\n          reject(err);\r\n        }\r\n      };\r\n\r\n      ws.onerror = () => {\r\n        const err = new Error('WebSocket connection error');\r\n        onError?.(err);\r\n        reject(err);\r\n      };\r\n\r\n      ws.onclose = () => {\r\n        ws = null;\r\n      };\r\n    });\r\n  }\r\n\r\n  async function startRecording(): Promise<void> {\r\n    if (typeof window === 'undefined') {\r\n      throw new Error('Recording only supported in browser');\r\n    }\r\n\r\n    if (!ws || ws.readyState !== WebSocket.OPEN) {\r\n      throw new Error('WebSocket not connected');\r\n    }\r\n\r\n    try {\r\n      mediaStream = await navigator.mediaDevices.getUserMedia({\r\n        audio: {\r\n          sampleRate,\r\n          channelCount: 1,\r\n          echoCancellation: true,\r\n          noiseSuppression: true,\r\n        },\r\n      });\r\n\r\n      audioContext = new AudioContext({ sampleRate });\r\n      const source = audioContext.createMediaStreamSource(mediaStream);\r\n      processor = audioContext.createScriptProcessor(4096, 1, 1);\r\n\r\n      processor.onaudioprocess = (e) => {\r\n        if (!ws || ws.readyState !== WebSocket.OPEN) return;\r\n\r\n        const inputData = e.inputBuffer.getChannelData(0);\r\n        const inputLen = inputData.length;\r\n\r\n        const pcm = new Int16Array(inputLen);\r\n        for (let i = 0; i < inputLen; i++) {\r\n          const s = Math.max(-1, Math.min(1, inputData[i]!));\r\n          pcm[i] = s < 0 ? s * 32768 : s * 32767;\r\n        }\r\n\r\n        const bytes = new Uint8Array(pcm.buffer);\r\n        const len = bytes.length;\r\n        let binary = '';\r\n        for (let i = 0; i < len; i++) {\r\n          binary += String.fromCharCode(bytes[i]!);\r\n        }\r\n        const base64 = btoa(binary);\r\n\r\n        ws.send(JSON.stringify({\r\n          type: 'input_audio_buffer.append',\r\n          audio: base64,\r\n        }));\r\n      };\r\n\r\n      source.connect(processor);\r\n      processor.connect(audioContext.destination);\r\n    } catch (err) {\r\n      onError?.(err as Error);\r\n      throw err;\r\n    }\r\n  }\r\n\r\n  function stopRecording() {\r\n    if (mediaStream) {\r\n      mediaStream.getTracks().forEach(track => track.stop());\r\n      mediaStream = null;\r\n    }\r\n\r\n    if (processor) {\r\n      processor.disconnect();\r\n      processor = null;\r\n    }\r\n\r\n    if (audioContext) {\r\n      audioContext.close();\r\n      audioContext = null;\r\n    }\r\n\r\n    if (ws && ws.readyState === WebSocket.OPEN) {\r\n      ws.send(JSON.stringify({ type: 'input_audio_buffer.commit' }));\r\n    }\r\n  }\r\n\r\n  function close() {\r\n    stopRecording();\r\n    if (ws) {\r\n      ws.close();\r\n      ws = null;\r\n    }\r\n  }\r\n\r\n  return {\r\n    connect,\r\n    startRecording,\r\n    stopRecording,\r\n    close,\r\n  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package/dist/index.js

@@ -1,16 +1,11 @@
 // src/asr-client.ts
 var ASR_PATH = "/api/proxy/builtin/platform/qwen-asr-realtime/api-ws/v1/realtime";
 async function listen(onTranscript) {
-  if (typeof navigator === "undefined" || !navigator.mediaDevices?.getUserMedia) {
-    throw new Error("Microphone not supported. Requires HTTPS and browser with MediaDevices API.");
-  }
   const ws = new WebSocket(ASR_PATH);
   let mediaStream = null;
   let audioContext = null;
   let processor = null;
-  let isRunning = true;
   const stop = () => {
-    isRunning = false;
     if (mediaStream) {
       mediaStream.getTracks().forEach((t) => t.stop());
       mediaStream = null;
@@ -24,7 +19,8 @@ async function listen(onTranscript) {
       audioContext = null;
     }
     if (ws.readyState === WebSocket.OPEN) {
-      ws.send(JSON.stringify({ event_id: `event_${Date.now()}`, type: "session.finish" }));
+      ws.send(JSON.stringify({ type: "input_audio_buffer.commit" }));
+      ws.close();
     }
   };
   return new Promise((resolve, reject) => {
@@ -39,7 +35,7 @@ async function listen(onTranscript) {
           const source = audioContext.createMediaStreamSource(mediaStream);
           processor = audioContext.createScriptProcessor(4096, 1, 1);
           processor.onaudioprocess = (e) => {
-            if (!isRunning || ws.readyState !== WebSocket.OPEN) return;
+            if (ws.readyState !== WebSocket.OPEN) return;
             const input = e.inputBuffer.getChannelData(0);
             const pcm = new Int16Array(input.length);
             for (let i = 0; i < input.length; i++) {
@@ -59,16 +55,10 @@ async function listen(onTranscript) {
         }
       }
       if (data.type === "conversation.item.input_audio_transcription.text") {
-        onTranscript(data.text || data.stash || "", false);
+        onTranscript(data.text || "", false);
       }
       if (data.type === "conversation.item.input_audio_transcription.completed") {
-        onTranscript(data.transcript || data.text || "", true);
-      }
-      if (data.type === "session.finished") {
-        if (data.transcript) {
-          onTranscript(data.transcript, true);
-        }
-        ws.close(1e3, "ASR finished");
+        onTranscript(data.text || data.transcript || "", true);
       }
       if (data.type === "error") {
         stop();

package/dist/index.js.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/asr-client.ts"],"sourcesContent":["/**\n * ASR Realtime WebSocket Client\n */\n\nconst ASR_PATH = '/api/proxy/builtin/platform/qwen-asr-realtime/api-ws/v1/realtime';\n\n/**\n * Simple ASR: start listening and get transcript\n * @returns stop function\n * @example\n * const stop = await listen((text, isFinal) => console.log(text))\n * // later: stop()\n */\nexport async function listen(\n  onTranscript: (text: string, isFinal: boolean) => void\n): Promise<() => void> {\n  // Check browser support first\n  if (typeof navigator === 'undefined' || !navigator.mediaDevices?.getUserMedia) {\n    throw new Error('Microphone not supported. Requires HTTPS and browser with MediaDevices API.');\n  }\n\n  const ws = new WebSocket(ASR_PATH);\n  let mediaStream: MediaStream | null = null;\n  let audioContext: AudioContext | null = null;\n  let processor: ScriptProcessorNode | null = null;\n  let isRunning = true;\n\n  const stop = () => {\n    isRunning = false;\n    if (mediaStream) { mediaStream.getTracks().forEach(t => t.stop()); mediaStream = null; }\n    if (processor) { processor.disconnect(); processor = null; }\n    if (audioContext) { audioContext.close(); audioContext = null; }\n    if (ws.readyState === WebSocket.OPEN) {\n      // Send session.finish to get final result\n      ws.send(JSON.stringify({ event_id: `event_${Date.now()}`, type: 'session.finish' }));\n    }\n  };\n\n  return new Promise((resolve, reject) => {\n    ws.onmessage = async (event) => {\n      const data = JSON.parse(event.data);\n\n      // Start recording immediately on session.created (use default config)\n      if (data.type === 'session.created') {\n        try {\n          mediaStream = await navigator.mediaDevices.getUserMedia({\n            audio: { sampleRate: 16000, channelCount: 1, echoCancellation: true }\n          });\n          audioContext = new AudioContext({ sampleRate: 16000 });\n          const source = audioContext.createMediaStreamSource(mediaStream);\n          processor = audioContext.createScriptProcessor(4096, 1, 1);\n\n          processor.onaudioprocess = (e) => {\n            if (!isRunning || ws.readyState !== WebSocket.OPEN) return;\n            const input = e.inputBuffer.getChannelData(0);\n            const pcm = new Int16Array(input.length);\n            for (let i = 0; i < input.length; i++) {\n              const s = Math.max(-1, Math.min(1, input[i]!));\n              pcm[i] = s < 0 ? s * 32768 : s * 32767;\n            }\n            const bytes = new Uint8Array(pcm.buffer);\n            let binary = '';\n            for (let i = 0; i < bytes.length; i++) binary += String.fromCharCode(bytes[i]!);\n            ws.send(JSON.stringify({ type: 'input_audio_buffer.append', audio: btoa(binary) }));\n          };\n\n          source.connect(processor);\n          processor.connect(audioContext.destination);\n          resolve(stop);\n        } catch (err) {\n          reject(err);\n        }\n      }\n\n      // Interim result\n      if (data.type === 'conversation.item.input_audio_transcription.text') {\n        onTranscript(data.text || data.stash || '', false);\n      }\n\n      // Sentence complete (VAD detected pause)\n      if (data.type === 'conversation.item.input_audio_transcription.completed') {\n        onTranscript(data.transcript || data.text || '', true);\n      }\n\n      // Final result after session.finish\n      if (data.type === 'session.finished') {\n        if (data.transcript) {\n          onTranscript(data.transcript, true);\n        }\n        ws.close(1000, 'ASR finished');\n      }\n\n      if (data.type === 'error') {\n        stop();\n        reject(new Error(data.error?.message || 'ASR error'));\n      }\n    };\n\n    ws.onerror = () => reject(new Error('ASR connection failed'));\n  });\n}\n\nexport interface ASRClientConfig {\n  /** Audio format, default 'pcm16' */\n  audioFormat?: 'pcm16' | 'g711a' | 'g711u';\n  /** Sample rate, default 16000 */\n  sampleRate?: number;\n  /** Called when connection is ready */\n  onReady?: () => void;\n  /** Called when speech is detected */\n  onSpeechStart?: () => void;\n  /** Called when speech stops */\n  onSpeechEnd?: () => void;\n  /** Called on transcript result */\n  onTranscript?: (text: string, isFinal: boolean) => void;\n  /** Called on error */\n  onError?: (error: Error) => void;\n}\n\nexport interface ASRClient {\n  /** Connect to ASR service */\n  connect(): Promise<void>;\n  /** Start recording from microphone */\n  startRecording(): Promise<void>;\n  /** Stop recording */\n  stopRecording(): void;\n  /** Close connection */\n  close(): 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 }\n\n        if (data.type === 'conversation.item.input_audio_transcription.text') {\n          onTranscript?.(data.text || '', false);\n        }\n\n        if (data.type === 'conversation.item.input_audio_transcription.completed') {\n          onTranscript?.(data.text || data.transcript || '', true);\n        }\n\n        if (data.type === 'error') {\n          const err = new Error(data.error?.message || 'Unknown error');\n          onError?.(err);\n          reject(err);\n        }\n      };\n\n      ws.onerror = () => {\n        const err = new Error('WebSocket connection error');\n        onError?.(err);\n        reject(err);\n      };\n\n      ws.onclose = () => {\n        ws = null;\n      };\n    });\n  }\n\n  async function startRecording(): Promise<void> {\n    if (typeof window === 'undefined') {\n      throw new Error('Recording only supported in browser');\n    }\n\n    if (!ws || ws.readyState !== WebSocket.OPEN) {\n      throw new Error('WebSocket not connected');\n    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microphone */\r\n  startRecording(): Promise<void>;\r\n  /** Stop recording */\r\n  stopRecording(): void;\r\n  /** Close connection */\r\n  close(): void;\r\n}\r\n\r\nexport function createASRClient(config: ASRClientConfig): ASRClient {\r\n  const {\r\n    // audioFormat = 'pcm16',\r\n    sampleRate = 16000,\r\n    onReady,\r\n    onSpeechStart,\r\n    onSpeechEnd,\r\n    onTranscript,\r\n    onError,\r\n  } = config;\r\n\r\n  let ws: WebSocket | null = null;\r\n  let mediaStream: MediaStream | null = null;\r\n  let audioContext: AudioContext | null = null;\r\n  let processor: ScriptProcessorNode | null = null;\r\n\r\n  async function connect(): Promise<void> {\r\n    return new Promise((resolve, reject) => {\r\n      ws = new WebSocket(ASR_PATH);\r\n\r\n      ws.onopen = () => {};\r\n\r\n      ws.onmessage = (event) => {\r\n        const data = JSON.parse(event.data);\r\n\r\n        if (data.type === 'session.created') {\r\n          onReady?.();\r\n          resolve();\r\n        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package/package.json

@@ -1,45 +1,45 @@
-{
-  "name": "@amaster.ai/asr-client",
-  "version": "1.0.0-beta.33",
-  "description": "Qwen ASR Realtime WebSocket client with microphone recording",
-  "type": "module",
-  "main": "./dist/index.cjs",
-  "module": "./dist/index.js",
-  "types": "./dist/index.d.ts",
-  "exports": {
-    ".": {
-      "types": "./dist/index.d.ts",
-      "import": "./dist/index.js",
-      "require": "./dist/index.cjs"
-    }
-  },
-  "files": [
-    "dist",
-    "README.md"
-  ],
-  "keywords": [
-    "asr",
-    "speech-to-text",
-    "qwen",
-    "realtime",
-    "websocket",
-    "audio",
-    "speech-recognition"
-  ],
-  "author": "Amaster Team",
-  "license": "MIT",
-  "publishConfig": {
-    "access": "public",
-    "registry": "https://registry.npmjs.org/"
-  },
-  "devDependencies": {
-    "tsup": "^8.3.5",
-    "typescript": "~5.7.2"
-  },
-  "scripts": {
-    "build": "tsup",
-    "dev": "tsup --watch",
-    "clean": "rm -rf dist *.tsbuildinfo",
-    "type-check": "tsc --noEmit"
-  }
-}
+{
+  "name": "@amaster.ai/asr-client",
+  "version": "1.0.0-beta.7",
+  "description": "Qwen ASR Realtime WebSocket client with microphone recording",
+  "type": "module",
+  "main": "./dist/index.cjs",
+  "module": "./dist/index.js",
+  "types": "./dist/index.d.ts",
+  "exports": {
+    ".": {
+      "types": "./dist/index.d.ts",
+      "import": "./dist/index.js",
+      "require": "./dist/index.cjs"
+    }
+  },
+  "files": [
+    "dist",
+    "README.md"
+  ],
+  "scripts": {
+    "build": "tsup",
+    "dev": "tsup --watch",
+    "clean": "rm -rf dist *.tsbuildinfo",
+    "type-check": "tsc --noEmit"
+  },
+  "keywords": [
+    "asr",
+    "speech-to-text",
+    "qwen",
+    "realtime",
+    "websocket",
+    "audio",
+    "speech-recognition"
+  ],
+  "author": "Amaster Team",
+  "license": "MIT",
+  "publishConfig": {
+    "access": "public",
+    "registry": "https://registry.npmjs.org/"
+  },
+  "devDependencies": {
+    "tsup": "^8.3.5",
+    "typescript": "~5.7.2"
+  }
+}

package/LICENSE

@@ -1,21 +0,0 @@
-MIT License
-
-Copyright (c) 2026 Amaster Team
-
-Permission is hereby granted, free of charge, to any person obtaining a copy
-of this software and associated documentation files (the "Software"), to deal
-in the Software without restriction, including without limitation the rights
-to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
-copies of the Software, and to permit persons to whom the Software is
-furnished to do so, subject to the following conditions:
-
-The above copyright notice and this permission notice shall be included in all
-copies or substantial portions of the Software.
-
-THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
-IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
-FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
-AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
-LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
-OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
-SOFTWARE.