Update websocket_handler.py

websocket_handler.py

@@ -608,14 +608,17 @@ async def generate_and_stream_tts(
             "to": "playing_audio"
         })
         
-        # Convert audio to base64 for transmission
+        # Convert entire audio to base64 for transmission
+        import base64
         audio_base64 = base64.b64encode(audio_data).decode('utf-8')
         
         # Stream audio in chunks
-        chunk_size = session.audio_chunk_size
+        chunk_size = 16384  # Larger chunk size for base64
         total_length = len(audio_base64)
         total_chunks = (total_length + chunk_size - 1) // chunk_size
         
+        log_info(f"Streaming TTS audio: {len(audio_data)} bytes as {total_length} base64 chars in {total_chunks} chunks")
+        
         for i in range(0, total_length, chunk_size):
             # Check for cancellation
             if asyncio.current_task().cancelled():
@@ -630,7 +633,7 @@ async def generate_and_stream_tts(
                 "chunk_index": chunk_index,
                 "total_chunks": total_chunks,
                 "is_last": chunk_index == total_chunks - 1,
-                "mime_type": "audio/mpeg"  # MP3 format için
+                "mime_type": "audio/mpeg"
             })
             
             # Small delay to prevent overwhelming the client