diff --git a/dist/index.html b/dist/index.html
index fd35d3d..0c7d14c 100644
--- a/dist/index.html
+++ b/dist/index.html
@@ -256,12 +256,15 @@ async function transcribe(blob) {
   if (!GROQ_API_KEY) return { text: '', error: 'no_key' };
   const tStart = performance.now();
 
+  // Обрезаем 0.2с в начале (щелчок)
+  const sendBlob = await trimStartBlob(blob);
+
   const form = new FormData();
-  form.append('file', blob, 'audio.webm');
+  form.append('file', sendBlob, sendBlob.type === 'audio/wav' ? 'audio.wav' : 'audio.webm');
   form.append('model', 'whisper-large-v3');
   form.append('language', 'it');
 
-  console.log('[HTTP] blob ' + (blob.size/1024).toFixed(1) + 'KB → multipart POST');
+  console.log('[HTTP] blob ' + (sendBlob.size/1024).toFixed(1) + 'KB → multipart POST');
 
   try {
     const r = await fetch('https://lang.kube5s.ru/openai/v1/audio/transcriptions', {
@@ -623,15 +626,30 @@ function encodeWAV(samples, sampleRate) {
 
 // Обрезает первые cutSec секунд, возвращает WAV Blob (для Groq)
 async function trimStartBlob(blob, cutSec = 0.2) {
-  const ctx = new OfflineAudioContext(1, 1, 44100);
-  const ab = await blob.arrayBuffer();
-  const buf = await ctx.decodeAudioData(ab);
-  const sr = buf.sampleRate;
-  const data = buf.getChannelData(0);
-  const cutSamples = Math.floor(sr * cutSec);
-  const trimmed = data.slice(cutSamples);
-  const wav = encodeWAV(trimmed, sr);
-  return new Blob([wav], { type: 'audio/wav' });
+  try {
+    const ab = await blob.arrayBuffer();
+    const actx = new AudioContext();
+    const rawBuf = await actx.decodeAudioData(ab);
+    const sr = rawBuf.sampleRate;
+    const cutSamples = Math.floor(sr * cutSec);
+    const newLen = rawBuf.length - cutSamples;
+    if (newLen < sr * 0.05) { actx.close(); return blob; }
+
+    // Render trimmed via OfflineAudioContext (надёжнее ручного slice)
+    const offline = new OfflineAudioContext(1, newLen, sr);
+    const src = offline.createBufferSource();
+    src.buffer = rawBuf;
+    src.connect(offline.destination);
+    src.start(0, cutSec / sr);
+    const rendered = await offline.startRendering();
+    actx.close();
+
+    const wav = encodeWAV(rendered.getChannelData(0), sr);
+    return new Blob([wav], { type: 'audio/wav' });
+  } catch(e) {
+    console.log('[trimStartBlob] fallback to raw:', e);
+    return blob;
+  }
 }
 
 </script>
@@ -761,7 +779,7 @@ async function doCompare(blob, originalText, duration) {
 <script>
 // ---------- Глобальные переменные и инициализация ----------
 const GROQ_API_KEY = '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';
-const VERSION = 'v81';
+const VERSION = 'v82';
 
 let mediaRecorder, audioChunks = [], userAudioBlob = null, ttsText = '';
 let isRecording = false, isAnalyzing = false, micAnalyser = null, micAnimId = null;
diff --git a/js/audio.js b/js/audio.js
index 6661997..5c320b1 100644
--- a/js/audio.js
+++ b/js/audio.js
@@ -321,13 +321,28 @@ function encodeWAV(samples, sampleRate) {
 
 // Обрезает первые cutSec секунд, возвращает WAV Blob (для Groq)
 async function trimStartBlob(blob, cutSec = 0.2) {
-  const ctx = new OfflineAudioContext(1, 1, 44100);
-  const ab = await blob.arrayBuffer();
-  const buf = await ctx.decodeAudioData(ab);
-  const sr = buf.sampleRate;
-  const data = buf.getChannelData(0);
-  const cutSamples = Math.floor(sr * cutSec);
-  const trimmed = data.slice(cutSamples);
-  const wav = encodeWAV(trimmed, sr);
-  return new Blob([wav], { type: 'audio/wav' });
+  try {
+    const ab = await blob.arrayBuffer();
+    const actx = new AudioContext();
+    const rawBuf = await actx.decodeAudioData(ab);
+    const sr = rawBuf.sampleRate;
+    const cutSamples = Math.floor(sr * cutSec);
+    const newLen = rawBuf.length - cutSamples;
+    if (newLen < sr * 0.05) { actx.close(); return blob; }
+
+    // Render trimmed via OfflineAudioContext (надёжнее ручного slice)
+    const offline = new OfflineAudioContext(1, newLen, sr);
+    const src = offline.createBufferSource();
+    src.buffer = rawBuf;
+    src.connect(offline.destination);
+    src.start(0, cutSec / sr);
+    const rendered = await offline.startRendering();
+    actx.close();
+
+    const wav = encodeWAV(rendered.getChannelData(0), sr);
+    return new Blob([wav], { type: 'audio/wav' });
+  } catch(e) {
+    console.log('[trimStartBlob] fallback to raw:', e);
+    return blob;
+  }
 }
diff --git a/js/main.js b/js/main.js
index 3acf50d..db09a4e 100644
--- a/js/main.js
+++ b/js/main.js
@@ -1,6 +1,6 @@
 // ---------- Глобальные переменные и инициализация ----------
 const GROQ_API_KEY = '';
-const VERSION = 'v81';
+const VERSION = 'v82';
 
 let mediaRecorder, audioChunks = [], userAudioBlob = null, ttsText = '';
 let isRecording = false, isAnalyzing = false, micAnalyser = null, micAnimId = null;
diff --git a/js/transcribe.js b/js/transcribe.js
index f5313db..3adcdb1 100644
--- a/js/transcribe.js
+++ b/js/transcribe.js
@@ -4,12 +4,15 @@ async function transcribe(blob) {
   if (!GROQ_API_KEY) return { text: '', error: 'no_key' };
   const tStart = performance.now();
 
+  // Обрезаем 0.2с в начале (щелчок)
+  const sendBlob = await trimStartBlob(blob);
+
   const form = new FormData();
-  form.append('file', blob, 'audio.webm');
+  form.append('file', sendBlob, sendBlob.type === 'audio/wav' ? 'audio.wav' : 'audio.webm');
   form.append('model', 'whisper-large-v3');
   form.append('language', 'it');
 
-  console.log('[HTTP] blob ' + (blob.size/1024).toFixed(1) + 'KB → multipart POST');
+  console.log('[HTTP] blob ' + (sendBlob.size/1024).toFixed(1) + 'KB → multipart POST');
 
   try {
     const r = await fetch('https://lang.kube5s.ru/openai/v1/audio/transcriptions', {