ttp-agent-sdk 2.48.5 → 2.48.8

package/dist/agent-widget.dev.js

@@ -12047,7 +12047,7 @@ var TextChatSDK = /*#__PURE__*/function (_EventEmitter) {
 
       // SDK build time for debugging
       if (true) {
-        helloMessage.lastBuildTime = "2026-08-20T16:29:04.205Z";
+        helloMessage.lastBuildTime = "2026-08-20T21:32:52.088Z";
       }
       try {
         this.ws.send(JSON.stringify(helloMessage));
@@ -21357,8 +21357,8 @@ var VoiceSDK = _v2_VoiceSDK_js__WEBPACK_IMPORTED_MODULE_0__["default"];
 
 
 // Version - injected at build time from package.json via webpack DefinePlugin
-var VERSION = "2.48.5";
-var BUILD_TIME = "2026-08-20T16:29:04.205Z";
+var VERSION = "2.48.8";
+var BUILD_TIME = "2026-08-20T21:32:52.088Z";
 console.log("%c TTP Agent SDK v".concat(VERSION, " (").concat(BUILD_TIME, ") "), 'background: #4f46e5; color: white; font-size: 12px; font-weight: bold; padding: 2px 6px; border-radius: 4px;');
 
 // Named exports
@@ -25569,6 +25569,49 @@ function _setPrototypeOf(t, e) { return _setPrototypeOf = Object.setPrototypeOf
 // so creating a new AudioContext immediately after closing the old one can fail silently.
 var _sharedPlayerContext = null;
 var _sharedPlayerSampleRate = null;
+function _sinc(x) {
+  if (x === 0) return 1;
+  var px = Math.PI * x;
+  return Math.sin(px) / px;
+}
+function _blackman(x) {
+  var n = (x + 1) / 2;
+  return 0.42 - 0.5 * Math.cos(2 * Math.PI * n) + 0.08 * Math.cos(4 * Math.PI * n);
+}
+
+/**
+ * Windowed-sinc resampler (16 taps, anti-aliased on downsample). Used only when
+ * the HTML-audio hop ended up on a low-rate context (e.g. Bluetooth hands-free
+ * caps output at 16 kHz): the MediaStream → <audio> hop is not reliable unless
+ * the stream honestly runs at context.sampleRate, so TTS PCM is resampled
+ * before createBuffer — the same invariant ElevenLabs keeps in its
+ * audioConcatProcessor worklet.
+ */
+function resampleFloat32ToRate(input, fromRate, toRate) {
+  if (fromRate === toRate || !input || input.length === 0) return input;
+  var ratio = fromRate / toRate;
+  var outLength = Math.max(1, Math.round(input.length / ratio));
+  var out = new Float32Array(outLength);
+  var TAPS = 16;
+  var HALF = TAPS / 2;
+  var cutoff = 0.92 * Math.min(1, toRate / fromRate);
+  for (var i = 0; i < outLength; i++) {
+    var pos = i * ratio;
+    var center = Math.floor(pos);
+    var sum = 0;
+    var wsum = 0;
+    for (var t = -HALF + 1; t <= HALF; t++) {
+      var idx = center + t;
+      if (idx < 0 || idx >= input.length) continue;
+      var x = pos - idx;
+      var w = _blackman(t / HALF) * _sinc(x * cutoff) * cutoff;
+      sum += input[idx] * w;
+      wsum += w;
+    }
+    out[i] = wsum !== 0 ? sum / wsum : 0;
+  }
+  return out;
+}
 
 /**
 
@@ -25620,6 +25663,7 @@ var AudioPlayer = /*#__PURE__*/function (_EventEmitter) {
     _this._mediaStreamDest = null;
     _this._htmlAudioRouteActive = false;
     _this._htmlAudioPlayOk = false;
+    _this._loggedHopResample = false;
     // Greeting (and other bursts) fire many playChunk()s without await.
     // Serialize context creation so they cannot open two AudioContexts
     // at different rates in the middle of a sentence.
@@ -25915,7 +25959,7 @@ var AudioPlayer = /*#__PURE__*/function (_EventEmitter) {
     key: "prepareChunk",
     value: (function () {
       var _prepareChunk = _asyncToGenerator(/*#__PURE__*/_regenerator().m(function _callee2(pcmData) {
-        var _this$outputFormat2, processedData, padded, int16Array, float32Array, NORMALIZATION, length, i, audioDataSampleRate, audioBuffer, actualDuration, _t;
+        var _this$outputFormat2, processedData, padded, int16Array, float32Array, NORMALIZATION, length, i, audioDataSampleRate, playback, audioBuffer, actualDuration, _t;
         return _regenerator().w(function (_context2) {
           while (1) switch (_context2.p = _context2.n) {
             case 0:
@@ -25964,11 +26008,12 @@ var AudioPlayer = /*#__PURE__*/function (_EventEmitter) {
 
               // Create audio buffer with the ACTUAL sample rate of the audio data
               audioDataSampleRate = ((_this$outputFormat2 = this.outputFormat) === null || _this$outputFormat2 === void 0 ? void 0 : _this$outputFormat2.sampleRate) || this.audioContext.sampleRate;
+              playback = this._honestRateSamples(float32Array, audioDataSampleRate);
               audioBuffer = this.audioContext.createBuffer(1,
               // mono
 
-              float32Array.length, audioDataSampleRate);
-              audioBuffer.getChannelData(0).set(float32Array);
+              playback.samples.length, playback.sampleRate);
+              audioBuffer.getChannelData(0).set(playback.samples);
 
               // Wall-clock duration is sampleCount / TTS rate (audioBuffer.duration).
               // Dividing by contextSampleRate when the context is 48 kHz and TTS is
@@ -26335,7 +26380,7 @@ var AudioPlayer = /*#__PURE__*/function (_EventEmitter) {
     value: (function () {
       var _processPcmQueue = _asyncToGenerator(/*#__PURE__*/_regenerator().m(function _callee4() {
         var _this5 = this;
-        var _this$outputFormat3, _this$outputFormat4, pcmData, processedData, padded, int16Array, float32Array, NORMALIZATION, length, i, audioDataSampleRate, contextSampleRate, audioBuffer, source, currentTime, actualDuration, startTime, _t3;
+        var _this$outputFormat3, _this$outputFormat4, pcmData, processedData, padded, int16Array, float32Array, NORMALIZATION, length, i, audioDataSampleRate, contextSampleRate, playback, audioBuffer, source, currentTime, actualDuration, startTime, _t3;
         return _regenerator().w(function (_context5) {
           while (1) switch (_context5.p = _context5.n) {
             case 0:
@@ -26402,10 +26447,11 @@ var AudioPlayer = /*#__PURE__*/function (_EventEmitter) {
               } else if (this.scheduledBuffers < 3) {
                 console.log("\uD83C\uDFB5 AudioPlayer: Creating buffer at ".concat(audioDataSampleRate, "Hz (AudioContext: ").concat(contextSampleRate, "Hz)"));
               }
+              playback = this._honestRateSamples(float32Array, audioDataSampleRate);
               audioBuffer = this.audioContext.createBuffer(1,
               // mono
-              float32Array.length, audioDataSampleRate);
-              audioBuffer.getChannelData(0).set(float32Array);
+              playback.samples.length, playback.sampleRate);
+              audioBuffer.getChannelData(0).set(playback.samples);
 
               // Schedule playback with seamless timing
               source = this.audioContext.createBufferSource();
@@ -27061,6 +27107,34 @@ var AudioPlayer = /*#__PURE__*/function (_EventEmitter) {
     value: function _isHardwarePlaybackRate(rate) {
       return typeof rate === 'number' && rate >= 44100;
     }
+
+    /**
+     * HTML-hop mismatch guard. The MediaStreamDestination → <audio> hop is only
+     * reliable when the context runs at a hardware rate; if Chrome still gave a
+     * low-rate context (e.g. Bluetooth hands-free output), resample the TTS audio
+     * to the context rate ourselves so the stream honestly runs at
+     * context.sampleRate. Destination playback is untouched — Web Audio resamples
+     * correctly on that path.
+     */
+  }, {
+    key: "_honestRateSamples",
+    value: function _honestRateSamples(float32Array, audioDataSampleRate) {
+      var contextRate = this.audioContext.sampleRate;
+      if (this._htmlHopNeedsHardwareRate() && !this._isHardwarePlaybackRate(contextRate) && audioDataSampleRate !== contextRate) {
+        if (!this._loggedHopResample) {
+          this._loggedHopResample = true;
+          console.warn("\u26A0\uFE0F AudioPlayer: HTML hop on a ".concat(contextRate, "Hz context \u2014 resampling TTS ").concat(audioDataSampleRate, "Hz \u2192 ").concat(contextRate, "Hz in the player"));
+        }
+        return {
+          samples: resampleFloat32ToRate(float32Array, audioDataSampleRate, contextRate),
+          sampleRate: contextRate
+        };
+      }
+      return {
+        samples: float32Array,
+        sampleRate: audioDataSampleRate
+      };
+    }
   }, {
     key: "getHtmlAudioPlaybackInfo",
     value: function getHtmlAudioPlaybackInfo() {
@@ -27214,9 +27288,12 @@ var AudioPlayer = /*#__PURE__*/function (_EventEmitter) {
           while (1) switch (_context9.p = _context9.n) {
             case 0:
               ttsSampleRate = this._ttsSampleRate();
-              useHardwareRate = this._htmlHopNeedsHardwareRate(); // Destination path: match TTS (24 kHz). HTML last-hop: omit so the
-              // context (and MediaStream) run at the device rate — usually 48 kHz.
-              desiredSampleRate = useHardwareRate ? null : ttsSampleRate;
+              useHardwareRate = this._htmlHopNeedsHardwareRate(); // Destination path: match TTS (24 kHz). HTML last-hop: request the standard
+              // hardware rate explicitly. Omitting it lets Chrome inherit whatever rate is
+              // already active on the page (the widget opens the 16 kHz mic context before
+              // the player), and the MediaStream → <audio> hop then mis-clocks — heard as
+              // chipmunk-fast or half-speed speech.
+              desiredSampleRate = useHardwareRate ? 48000 : ttsSampleRate;
               if (!useHardwareRate && ![24000, 44100, 48000].includes(desiredSampleRate)) {
                 console.log("\u2139\uFE0F AudioPlayer: Backend requested ".concat(desiredSampleRate, "Hz, but browser may resample"));
                 console.log("   Consider requesting 24000Hz, 44100Hz, or 48000Hz from backend to reduce resampling");
@@ -27296,7 +27373,7 @@ var AudioPlayer = /*#__PURE__*/function (_EventEmitter) {
               setupAfterResume();
               return _context9.a(2);
             case 8:
-              console.log("\uD83C\uDFB5 AudioPlayer: Creating AudioContext (tts=".concat(ttsSampleRate, "Hz, htmlHop=").concat(useHardwareRate, ", requested=").concat(desiredSampleRate || 'hardware default', ")"));
+              console.log("\uD83C\uDFB5 AudioPlayer: Creating AudioContext (tts=".concat(ttsSampleRate, "Hz, htmlHop=").concat(useHardwareRate, ", requested=").concat(desiredSampleRate, "Hz)"));
 
               // Close old shared context if sample rate changed
               if (_sharedPlayerContext && _sharedPlayerContext.state !== 'closed') {
@@ -27321,7 +27398,9 @@ var AudioPlayer = /*#__PURE__*/function (_EventEmitter) {
                 _sharedPlayerSampleRate = this.audioContext.sampleRate;
                 console.log("\u2705 AudioContext created at ".concat(this.audioContext.sampleRate, "Hz (tts=").concat(ttsSampleRate, "Hz, htmlHop=").concat(useHardwareRate, ")"));
                 if (useHardwareRate && !this._isHardwarePlaybackRate(this.audioContext.sampleRate)) {
-                  console.warn("\u26A0\uFE0F AudioPlayer: HTML hop expected \u226544.1 kHz context, got ".concat(this.audioContext.sampleRate, "Hz"));
+                  // The hop stays on; _honestRateSamples resamples TTS to the context rate
+                  // per chunk so the MediaStream honestly runs at context.sampleRate.
+                  console.warn("\u26A0\uFE0F AudioPlayer: HTML hop got ".concat(this.audioContext.sampleRate, "Hz (<44.1 kHz) \u2014 resampling TTS to the context rate in the player"));
                 } else if (!useHardwareRate && Math.abs(this.audioContext.sampleRate - desiredSampleRate) > 100) {
                   console.warn("\u26A0\uFE0F AudioPlayer: Browser sample rate ".concat(this.audioContext.sampleRate, "Hz \u2260 requested ").concat(desiredSampleRate, "Hz; Web Audio will resample"));
                 }
@@ -29170,7 +29249,7 @@ var VoiceSDK_v2 = /*#__PURE__*/function (_EventEmitter) {
         // iOS in-app webviews lack "Safari" in the UA (real Safari always has it).
         var webview = isAndroid && (/\bwv\b/.test(ua) || /Version\/[\d.]+.*Chrome/.test(ua)) || isIos && !/Safari/i.test(ua) || false;
         var env = {
-          sdkVersion:  true ? "2.48.5" : 0,
+          sdkVersion:  true ? "2.48.8" : 0,
           ua: ua,
           platform: (uaData === null || uaData === void 0 ? void 0 : uaData.platform) || navigator.platform || '',
           mobile: (_uaData$mobile = uaData === null || uaData === void 0 ? void 0 : uaData.mobile) !== null && _uaData$mobile !== void 0 ? _uaData$mobile : isAndroid || isIos,
@@ -29185,7 +29264,7 @@ var VoiceSDK_v2 = /*#__PURE__*/function (_EventEmitter) {
       } catch (e) {
         console.warn('⚠️ VoiceSDK v2: Failed to build client env:', e);
         return {
-          sdkVersion:  true ? "2.48.5" : 0
+          sdkVersion:  true ? "2.48.8" : 0
         };
       }
     }
@@ -29327,7 +29406,7 @@ var VoiceSDK_v2 = /*#__PURE__*/function (_EventEmitter) {
 
               // Include SDK build time for debugging
               if (true) {
-                helloMessage.lastBuildTime = "2026-08-20T16:29:04.205Z";
+                helloMessage.lastBuildTime = "2026-08-20T21:32:52.088Z";
               }
 
               // Client environment (device/browser/webview) for backend logs + Langfuse metadata
@@ -35377,7 +35456,7 @@ var TTPChatWidget = /*#__PURE__*/function () {
         return;
       }
       this._ensureAboutStyles();
-      var version =  true ? "2.48.5" : 0;
+      var version =  true ? "2.48.8" : 0;
       var convId = this._getLastConversationId();
       var t = function t(k, fb) {
         try {