Refactor ASR plugin options and improve audio processing

- Renamed `hideTimeout` to `autoHideTimeout` for clarity.
- Updated the default options destructuring for better readability.
- Removed unnecessary comments and cleaned up the audio processing logic.
- Ensured that the audio buffer is processed correctly before sending to the `onAudioChunk` callback.

packages/artplayer-plugin-asr/src/index.js

@@ -1,7 +1,13 @@
 import style from 'bundle-text:./style.less';
 
 export default function artplayerPluginAsr(option = {}) {
-    const { hideTimeout = 10000, length = 3, interval = 100, sampleRate = 16000, onAudioChunk = () => null } = option;
+    const {
+        length = 3,
+        interval = 100,
+        sampleRate = 16000,
+        autoHideTimeout = 10000,
+        onAudioChunk = () => null,
+    } = option;
 
     return (art) => {
         let started = false;
@@ -39,7 +45,7 @@ export default function artplayerPluginAsr(option = {}) {
         function append(subtitle) {
             if (typeof subtitle !== 'string') return;
             clearTimeout(hideTimer);
-            hideTimer = setTimeout(hide, hideTimeout);
+            hideTimer = setTimeout(hide, autoHideTimeout);
             $asr.style.display = '';
             $asr.innerHTML = splitByPunctuation(subtitle)
                 .map((line) => `<div class="art-asr-line">${line}</div>`)
@@ -134,21 +140,15 @@ export default function artplayerPluginAsr(option = {}) {
                     workletLoaded = true;
                 }
 
-                // 创建输出音频的 GainNode（正常播放）
                 gainNode = audioCtx.createGain();
                 gainNode.gain.value = 1;
-
                 recorderNode = new AudioWorkletNode(audioCtx, 'recorder-processor');
                 recorderNode.port.onmessage = (event) => {
                     bufferChunks.push(new Float32Array(event.data));
                 };
-
-                // 分流：识别和播放
                 audioSource.connect(recorderNode);
                 audioSource.connect(gainNode);
-
-                gainNode.connect(audioCtx.destination); // 正常声音输出
-                // recorderNode 不连接 destination，避免输出干扰
+                gainNode.connect(audioCtx.destination);
 
                 const CHUNK_SAMPLES = sampleRate * (interval / 1000);
                 timer = setInterval(async () => {
@@ -166,8 +166,8 @@ export default function artplayerPluginAsr(option = {}) {
                     if (accumulated.length < CHUNK_SAMPLES) return;
 
                     const chunkToSend = accumulated.slice(0, CHUNK_SAMPLES);
-                    const pcm = floatTo16BitPCM(chunkToSend);
-                    const subtitle = await onAudioChunk(pcm);
+                    const buffer = floatTo16BitPCM(chunkToSend);
+                    const subtitle = await onAudioChunk(buffer);
                     append(subtitle);
                 }, interval);