add whisper_live module
This commit is contained in:
makaveli10
@@ -0,0 +1,275 @@
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import os
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import argparse
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import wave
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import numpy as np
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import scipy
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import ffmpeg
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import pyaudio
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import threading
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import textwrap
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import json
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import websocket
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def resample(file: str, sr: int = 16000):
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"""
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# https://github.com/openai/whisper/blob/7858aa9c08d98f75575035ecd6481f462d66ca27/whisper/audio.py#L22
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Open an audio file and read as mono waveform, resampling as necessary,
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save the resampled audio
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Parameters
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----------
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file: str
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The audio file to open
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sr: int
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The sample rate to resample the audio if necessary
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"""
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try:
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# This launches a subprocess to decode audio while down-mixing and resampling as necessary.
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# Requires the ffmpeg CLI and `ffmpeg-python` package to be installed.
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out, _ = (
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ffmpeg.input(file, threads=0)
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.output("-", format="s16le", acodec="pcm_s16le", ac=1, ar=sr)
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.run(cmd=["ffmpeg", "-nostdin"], capture_stdout=True, capture_stderr=True)
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)
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except ffmpeg.Error as e:
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raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e
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np_buffer = np.frombuffer(out, dtype=np.int16)
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resampled_file = f"{file.split('.')[0]}_resampled.wav"
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scipy.io.wavfile.write(resampled_file, sr, np_buffer.astype(np.int16))
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return resampled_file
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class Client:
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CHUNK = 1024
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FORMAT = pyaudio.paInt16
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CHANNELS = 1
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RATE = 16000
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RECORD_SECONDS = 60000
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START_RECORDING = False
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multilingual = False
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language = None
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task = "transcribe"
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def __init__(self, host=None, port=None, is_multilingual=False, lang=None, translate=False):
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Client.multilingual = is_multilingual
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Client.language = lang if is_multilingual else "en"
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if translate:
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Client.task = "translate"
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self.timestamp_offset = 0.0
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self.audio_bytes = None
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self.p = pyaudio.PyAudio()
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self.stream = self.p.open(format=self.FORMAT,
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channels=self.CHANNELS,
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rate=self.RATE,
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input=True,
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frames_per_buffer=self.CHUNK)
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# create websocket connection
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if host is not None and port is not None:
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socket_url = f"ws://{host}:{port}"
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self.client_socket = websocket.WebSocketApp(socket_url,
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on_open=Client.on_open,
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on_message=Client.on_message,
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on_error=Client.on_error,
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on_close=Client.on_close)
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else:
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print("[ERROR]: No host or port specified.")
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return
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# start websocket client in a thread
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self.ws_thread = threading.Thread(target=self.client_socket.run_forever)
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self.ws_thread.setDaemon(True)
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self.ws_thread.start()
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self.frames = b""
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print("[INFO]: * recording")
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@staticmethod
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def on_message(ws, message):
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message = json.loads(message)
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if message == "SERVER_READY":
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Client.START_RECORDING = True
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return
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if isinstance(message, dict):
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Client.language = message.get("language")
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lang_prob = message.get("language_prob")
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print(f"[INFO]: Server detected language {Client.language} with probability {lang_prob}")
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return
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text = []
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if len(message):
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for seg in message:
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if len(text):
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if text[-1] != seg["text"]:
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text.append(seg["text"])
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else:
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text.append(seg["text"])
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if len(text) > 3:
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text = text[-3:]
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wrapper = textwrap.TextWrapper(width=60)
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word_list = wrapper.wrap(text="".join(text))
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# Print each line.
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if os.name=='nt':
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os.system('cls')
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else:
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os.system('clear')
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for element in word_list:
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print(element)
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@staticmethod
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def on_error(ws, error):
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print(error)
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@staticmethod
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def on_close(ws, close_status_code, close_msg):
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print(f"[INFO]: Websocket connection closed.")
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@staticmethod
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def on_open(ws):
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print(Client.multilingual, Client.language, Client.task)
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print("[INFO]: Opened connection")
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ws.send(json.dumps({
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'multilingual': Client.multilingual,
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'language': Client.language,
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'task': Client.task
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}))
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@staticmethod
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def bytes_to_float_array(audio_bytes):
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raw_data = np.frombuffer(
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buffer=audio_bytes, dtype=np.int16
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)
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return raw_data.astype(np.float32) / 32768.0
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def send_packet_to_server(self, message):
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try:
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self.client_socket.send(message, websocket.ABNF.OPCODE_BINARY)
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except Exception as e:
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print(e)
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def play_file(self, filename):
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# read audio and create pyaudio stream
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self.wf = wave.open(filename, 'rb')
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self.stream = self.p.open(format=self.p.get_format_from_width(self.wf.getsampwidth()),
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channels=self.wf.getnchannels(),
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rate=self.wf.getframerate(),
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input=True,
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output=True,
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frames_per_buffer=self.CHUNK)
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try:
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while True:
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data = self.wf.readframes(self.CHUNK)
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if data==b'': break
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audio_array = Client.bytes_to_float_array(data)
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self.send_packet_to_server(audio_array.tobytes())
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self.stream.write(data)
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self.wf.close()
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self.stream.close()
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except KeyboardInterrupt:
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self.wf.close()
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self.stream.stop_stream()
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self.stream.close()
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self.p.terminate()
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self.close_websocket()
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print("[INFO]: Keyboard interrupt.")
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def close_websocket(self):
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try:
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self.client_socket.close() # Close the WebSocket connection
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except Exception as e:
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print("[ERROR]: Error closing WebSocket:", e)
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try:
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self.ws_thread.join() # Wait for the WebSocket thread to finish
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except Exception as e:
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print("[ERROR:] Error joining WebSocket thread:", e)
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def get_client_socket(self):
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return self.client_socket
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def write_audio_frames_to_file(self, frames, file_name):
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wf = wave.open(file_name, 'wb')
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wf.setnchannels(self.CHANNELS)
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wf.setsampwidth(2)
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wf.setframerate(self.RATE)
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wf.writeframes(frames)
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wf.close()
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def record(self, out_file="output_recording.wav"):
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n_audio_file = 0
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# create dir for saving audio chunks
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if not os.path.exists("chunks"):
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os.makedirs("chunks", exist_ok=True)
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try:
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for _ in range(0, int(self.RATE / self.CHUNK * self.RECORD_SECONDS)):
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data = self.stream.read(self.CHUNK)
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self.frames += data
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audio_array = Client.bytes_to_float_array(data)
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self.send_packet_to_server(audio_array.tobytes())
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# save frames if more than a minute
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if len(self.frames) > 60*self.RATE:
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t = threading.Thread(
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target=self.write_audio_frames_to_file,
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args=(self.frames[:], f"chunks/{n_audio_file}.wav", )
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)
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t.start()
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n_audio_file += 1
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self.frames = b""
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except KeyboardInterrupt:
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if len(self.frames):
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self.write_audio_frames_to_file(
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self.frames[:], f"chunks/{n_audio_file}.wav")
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n_audio_file += 1
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self.stream.stop_stream()
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self.stream.close()
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self.p.terminate()
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self.close_websocket()
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# combine all the audio files
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self.write_output_recording(n_audio_file, out_file)
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def write_output_recording(self, n_audio_file, out_file):
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input_files = [f"chunks/{i}.wav" for i in range(n_audio_file) if os.path.exists(f"chunks/{i}.wav")]
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wf = wave.open(out_file, 'wb')
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wf.setnchannels(self.CHANNELS)
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wf.setsampwidth(2)
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wf.setframerate(self.RATE)
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for in_file in input_files:
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w = wave.open(in_file, 'rb')
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while True:
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data = w.readframes(self.CHUNK)
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if data==b'': break
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wf.writeframes(data)
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w.close()
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# remove this file
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os.remove(in_file)
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wf.close()
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class TranscriptionClient:
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def __init__(self, host, port, is_multilingual=False, lang=None, translate=False):
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self.client = Client(host, port, is_multilingual, lang, translate)
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def __call__(self, audio=None):
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print("[INFO]: Waiting for server ready ...")
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while not Client.START_RECORDING:
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pass
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print("[INFO]: Server Ready!")
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if audio is not None:
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resampled_file = resample(audio)
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self.client.play_file(resampled_file)
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else:
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self.client.record()
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