ruff ruff

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,
  )
 
- self.ws_thread = threading.Thread(target=self.ws.run_forever, kwargs={'sslopt': {"cert_reqs": ssl.CERT_NONE}})
+ 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()
@@ -57,7 +71,13 @@ 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...")
@@ -69,7 +89,7 @@ def on_open(self, ws):
 
  def on_message(self, ws, message):
  # message is bytes
- if message == b'DONE':
+ if message == b"DONE":
  print("listen")
  self.session_state = "listen"
  else:
@@ -99,7 +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):
@@ -108,33 +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()