added more logs

client.py

@@ -6,6 +6,11 @@ import wave
 import websockets
 import requests
 import ssl
+import sys
+
+logging.basicConfig(level=logging.DEBUG, format='%(asctime)s %(levelname)s: %(message)s',
+                    handlers=[logging.StreamHandler(sys.stdout)], force=True)
+logger = logging.getLogger(__name__)
 
 # Parameters for reading and sending the audio
 AUDIO_FILE_URL = "https://raw.githubusercontent.com/AshDavid12/runpod-serverless-forked/main/test_hebrew.wav"  # Use WAV file
@@ -35,6 +40,7 @@ def convert_to_mono_16k(audio_file_path):
 
 
 async def send_audio(websocket):
+    print(f"hi")
     buffer_size = 1024 * 16  # Send smaller chunks (16KB) for real-time processing
     logging.info("Converting the audio to mono and 16kHz.")
 
@@ -60,7 +66,7 @@ async def send_audio(websocket):
         for i in range(0, len(raw_data), buffer_size):
             pcm_chunk = raw_data[i:i + buffer_size]
             await websocket.send(pcm_chunk)  # Send raw PCM data chunk
-            logging.info(f"Sent PCM chunk of size {len(pcm_chunk)} bytes.")
+            #logging.info(f"Sent PCM chunk of size {len(pcm_chunk)} bytes.")
             await asyncio.sleep(0.01)  # Simulate real-time sending
 
         logging.info("Completed sending all audio data.")

infer.py

@@ -225,6 +225,8 @@ async def websocket_transcribe(websocket: WebSocket):
 
         # A buffer to store raw PCM audio data
         pcm_audio_buffer = bytearray()
+        logging.info("im here, is it failing?.")
+
 
         # Metadata for the incoming PCM data (sample rate, channels, and sample width should be consistent)
         sample_rate = 16000  # 16kHz
@@ -236,7 +238,7 @@ async def websocket_transcribe(websocket: WebSocket):
         if not os.path.exists(tmp_directory):
             logging.info(f"Creating /tmp directory: {tmp_directory}")
             os.makedirs(tmp_directory)
-
+        logging.info("im here, is it failing?2.")
         while True:
             try:
                 # Receive the next chunk of PCM audio data