dont let the thread die

arguments_classes/module_arguments.py

@@ -11,7 +11,7 @@ class ModuleArguments:
     mode: Optional[str] = field(
         default="socket",
         metadata={
-            "help": "The mode to run the pipeline in. Either 'local' or 'socket'. Default is 'socket'."
+            "help": "The mode to run the pipeline in. Either 'local', 'socket', or 'none'. Default is 'socket'."
         },
     )
     local_mac_optimal_settings: bool = field(

audio_streaming_client.py

@@ -2,20 +2,30 @@
 from queue import Queue
 import sounddevice as sd
 import numpy as np
-import requests
-import base64
 import time
 from dataclasses import dataclass, field
 import websocket
-import threading
 import ssl
 
+
 @dataclass
 class AudioStreamingClientArguments:
-    sample_rate: int = field(default=16000, metadata={"help": "Audio sample rate in Hz. Default is 16000."})
-    chunk_size: int = field(default=512, metadata={"help": "The size of audio chunks in samples. Default is 512."})
-    api_url: str = field(default="https://yxfmjcvuzgi123sw.us-east-1.aws.endpoints.huggingface.cloud", metadata={"help": "The URL of the API endpoint."})
-    auth_token: str = field(default="your_auth_token", metadata={"help": "Authentication token for the API."})
+    sample_rate: int = field(
+        default=16000, metadata={"help": "Audio sample rate in Hz. Default is 16000."}
+    )
+    chunk_size: int = field(
+        default=512,
+        metadata={"help": "The size of audio chunks in samples. Default is 512."},
+    )
+    api_url: str = field(
+        default="https://yxfmjcvuzgi123sw.us-east-1.aws.endpoints.huggingface.cloud",
+        metadata={"help": "The URL of the API endpoint."},
+    )
+    auth_token: str = field(
+        default="your_auth_token",
+        metadata={"help": "Authentication token for the API."},
+    )
+
 
 class AudioStreamingClient:
     def __init__(self, args: AudioStreamingClientArguments):
@@ -27,9 +37,11 @@ def __init__(self, args: AudioStreamingClientArguments):
         self.headers = {
             "Accept": "application/json",
             "Authorization": f"Bearer {self.args.auth_token}",
-            "Content-Type": "application/json"
+            "Content-Type": "application/json",
         }
-        self.session_state = "idle"  # Possible states: idle, sending, processing, waiting
+        self.session_state = (
+            "idle"  # Possible states: idle, sending, processing, waiting
+        )
         self.ws_ready = threading.Event()
 
     def start(self):
@@ -43,12 +55,14 @@ def start(self):
             on_open=self.on_open,
             on_message=self.on_message,
             on_error=self.on_error,
-            on_close=self.on_close
+            on_close=self.on_close,
         )
 
-        ws_thread = threading.Thread(target=self.ws.run_forever, kwargs={'sslopt': {"cert_reqs": ssl.CERT_NONE}})
-        ws_thread.start()
-
+        self.ws_thread = threading.Thread(
+            target=self.ws.run_forever, kwargs={"sslopt": {"cert_reqs": ssl.CERT_NONE}}
+        )
+        self.ws_thread.start()
+
         # Wait for the WebSocket to be ready
         self.ws_ready.wait()
         self.start_audio_streaming()
@@ -57,17 +71,25 @@ def start_audio_streaming(self):
         self.send_thread = threading.Thread(target=self.send_audio)
         self.play_thread = threading.Thread(target=self.play_audio)
 
-        with sd.InputStream(samplerate=self.args.sample_rate, channels=1, dtype='int16', callback=self.audio_input_callback, blocksize=self.args.chunk_size):
+        with sd.InputStream(
+            samplerate=self.args.sample_rate,
+            channels=1,
+            dtype="int16",
+            callback=self.audio_input_callback,
+            blocksize=self.args.chunk_size,
+        ):
             self.send_thread.start()
             self.play_thread.start()
+            input("Press Enter to stop streaming...")
+            self.on_shutdown()
 
     def on_open(self, ws):
         print("WebSocket connection opened.")
         self.ws_ready.set()  # Signal that the WebSocket is ready
 
     def on_message(self, ws, message):
         # message is bytes
-        if message == b'DONE':
+        if message == b"DONE":
             print("listen")
             self.session_state = "listen"
         else:
@@ -97,7 +119,7 @@ def send_audio(self):
                 if self.session_state != "processing":
                     self.ws.send(chunk.tobytes(), opcode=websocket.ABNF.OPCODE_BINARY)
                 else:
-                    self.ws.send([], opcode=websocket.ABNF.OPCODE_BINARY)  # handshake 
+                    self.ws.send([], opcode=websocket.ABNF.OPCODE_BINARY)  # handshake
             time.sleep(0.01)
 
     def audio_input_callback(self, indata, frames, time, status):
@@ -106,33 +128,57 @@ def audio_input_callback(self, indata, frames, time, status):
     def audio_out_callback(self, outdata, frames, time, status):
         if not self.recv_queue.empty():
             chunk = self.recv_queue.get()
-            
+
             # Ensure chunk is int16 and clip to valid range
             chunk_int16 = np.clip(chunk, -32768, 32767).astype(np.int16)
-            
+
             if len(chunk_int16) < len(outdata):
-                outdata[:len(chunk_int16), 0] = chunk_int16
-                outdata[len(chunk_int16):] = 0
+                outdata[: len(chunk_int16), 0] = chunk_int16
+                outdata[len(chunk_int16) :] = 0
             else:
-                outdata[:, 0] = chunk_int16[:len(outdata)]
+                outdata[:, 0] = chunk_int16[: len(outdata)]
         else:
             outdata[:] = 0
 
     def play_audio(self):
-        with sd.OutputStream(samplerate=self.args.sample_rate, channels=1, dtype='int16', callback=self.audio_out_callback, blocksize=self.args.chunk_size):
+        with sd.OutputStream(
+            samplerate=self.args.sample_rate,
+            channels=1,
+            dtype="int16",
+            callback=self.audio_out_callback,
+            blocksize=self.args.chunk_size,
+        ):
             while not self.stop_event.is_set():
                 time.sleep(0.1)
 
+
 if __name__ == "__main__":
     import argparse
 
     parser = argparse.ArgumentParser(description="Audio Streaming Client")
-    parser.add_argument("--sample_rate", type=int, default=16000, help="Audio sample rate in Hz. Default is 16000.")
-    parser.add_argument("--chunk_size", type=int, default=1024, help="The size of audio chunks in samples. Default is 1024.")
-    parser.add_argument("--api_url", type=str, required=True, help="The URL of the API endpoint.")
-    parser.add_argument("--auth_token", type=str, required=True, help="Authentication token for the API.")
+    parser.add_argument(
+        "--sample_rate",
+        type=int,
+        default=16000,
+        help="Audio sample rate in Hz. Default is 16000.",
+    )
+    parser.add_argument(
+        "--chunk_size",
+        type=int,
+        default=1024,
+        help="The size of audio chunks in samples. Default is 1024.",
+    )
+    parser.add_argument(
+        "--api_url", type=str, required=True, help="The URL of the API endpoint."
+    )
+    parser.add_argument(
+        "--auth_token",
+        type=str,
+        required=True,
+        help="Authentication token for the API.",
+    )
 
     args = parser.parse_args()
     client_args = AudioStreamingClientArguments(**vars(args))
     client = AudioStreamingClient(client_args)
-    client.start()
+    client.start()

requirements.txt

@@ -8,4 +8,5 @@ funasr>=1.1.6
 faster-whisper>=1.0.3
 modelscope>=1.17.1
 deepfilternet>=0.5.6
-openai>=1.40.1
+openai>=1.40.1
+websocket-client>=1.8.0

requirements_mac.txt

@@ -10,4 +10,5 @@ funasr>=1.1.6
 faster-whisper>=1.0.3
 modelscope>=1.17.1
 deepfilternet>=0.5.6
-openai>=1.40.1
+openai>=1.40.1
+websocket-client>=1.8.0