remove line that was not allowing page to render

Author: a-dinoto

fern/pages/05-guides/cookbooks/streaming-stt/turn_detection_improvement_using_async.mdx

@@ -5,7 +5,6 @@ title: "Determine Optimal Turn Detection Settings from Historical Audio Analysis
 This guide shows how to analyze utterance gaps from multiple pre-recorded audio files to automatically determine optimal turn detection settings for real-time streaming transcription. It processes an entire folder, aggregates gap statistics across all recordings, and configures the WebSocket with parameters tailored to your specific conversation patterns.
 
 ## Quickstart
-
 ```python
 import requests
 import time
@@ -575,7 +574,7 @@ if __name__ == "__main__":
     main()
 ```
 
-## Step-by-step guide
+## Step-By-Step Guide
 
 Before we begin, make sure you have an AssemblyAI account and an API key. You can [sign up](https://assemblyai.com/dashboard/signup) and get your API key from your dashboard.
 
@@ -586,7 +585,8 @@ pip install requests pyaudio websocket-client
 ```
 
 2. **Configuration and Global Variables**
-Sets up API credentials, file paths, audio parameters (16kHz sample rate, mono channel), and initializes global variables for managing WebSocket connections and audio streaming threads.
+
+Set up API credentials, file paths, audio parameters (16kHz sample rate, mono channel), and initialize global variables for managing WebSocket connections and audio streaming threads.
 
 ```python
 import requests
@@ -624,7 +624,8 @@ OPTIMIZED_CONFIG = {}
 ```
 
 3. **Define get_audio_files() Function**
-Scans a specified folder for audio/video files with supported extensions and returns a sorted list of file paths for batch processing.
+
+This function scans a specified folder for audio/video files with supported extensions and returns a sorted list of file paths for batch processing.
 
 ```python
 def get_audio_files(folder_path):
@@ -650,7 +651,8 @@ def get_audio_files(folder_path):
 ```
 
 4. **Define `analyze_single_file()` Function**
-Uploads an audio file to AssemblyAI, requests transcription with speaker labels enabled, polls until completion, then calculates gap statistics between utterances (average, median, min, max) and saves the transcript JSON.
+
+This function uploads an audio file to AssemblyAI, requests transcription with speaker labels enabled, polls until completion, then calculates gap statistics between utterances (average, median, min, max) and saves the transcript JSON.
 
 ```python
 def analyze_single_file(audio_file, api_key, file_index, total_files):
@@ -762,7 +764,8 @@ def analyze_single_file(audio_file, api_key, file_index, total_files):
 ```
 
 5. **Define `analyze_multiple_files()` Function**
-Orchestrates the analysis of all files in a folder by calling `analyze_single_file()` for each, aggregates all gap data across files, calculates overall statistics, displays per-file breakdowns, and saves a comprehensive summary JSON.
+
+This function orchestrates the analysis of all files in a folder by calling `analyze_single_file()` for each, aggregates all gap data across files, calculates overall statistics, displays per-file breakdowns, and saves a comprehensive summary JSON.
 
 ```python
 def analyze_multiple_files(folder_path, api_key):
@@ -879,7 +882,8 @@ def analyze_multiple_files(folder_path, api_key):
 ```
 
 6. **Define `determine_streaming_config()` Function**
-Takes aggregated gap statistics and selects one of three preset configurations (Aggressive <500ms, Balanced 500-1000ms, Conservative >1000ms) with optimized turn detection parameters for different conversation styles.
+
+This function takes aggregated gap statistics and selects one of three preset configurations with optimized turn detection parameters for different conversation styles.
 
 ```python
 def determine_streaming_config(aggregated_stats):
@@ -944,7 +948,8 @@ def determine_streaming_config(aggregated_stats):
 ```
 
 7. **Create WebSocket Event Handlers (`on_open`, `on_message`, `on_error`, `on_close`)**
-Manage the real-time streaming connection lifecycle: `on_open` starts the audio streaming thread, `on_message` processes transcription results (partial and final turns), and the close/error handlers clean up resources.
+
+These functions manage the real-time streaming connection lifecycle: `on_open` starts the audio streaming thread, `on_message` processes transcription results (partial and final turns), and the close/error handlers clean up resources.
 
 ```python
 def on_open(ws):
@@ -1031,7 +1036,8 @@ def on_close(ws, close_status_code, close_msg):
 ```
 
 8. **Define `run_streaming()` Function**
-Initializes PyAudio to capture microphone input, establishes a WebSocket connection with the optimized configuration parameters, and streams audio in real-time while displaying transcription results until the user stops with Ctrl+C.
+
+This function initializes PyAudio to capture microphone input, establishes a WebSocket connection with the optimized configuration parameters, and streams audio in real-time while displaying transcription results until the user stops with Ctrl+C.
 
 ```python
 def run_streaming(config):
@@ -1133,7 +1139,8 @@ def run_streaming(config):
 ```
 
 9. **Define `main()` Workflow**
-Executes the three-step process: analyze all audio files in the folder, determine the best streaming configuration based on aggregated utterance gaps, then launch real-time streaming with the optimized settings.
+
+Execute the three-step process: analyze all audio files in the folder, determine the best streaming configuration based on aggregated utterance gaps, then launch real-time streaming with the optimized settings.
 
 ```python
 def main():