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🔨 refactor whisper_live according to flake8

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This commit is contained in:
makaveli10
2024-02-09 13:45:13 +05:30
parent b4abe95fc6
commit 9fbff47126
13 changed files with 549 additions and 540 deletions
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whisper_live/client.py
+57 -124
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@@ -2,68 +2,14 @@ import os
import wave
import numpy as np
import scipy
import ffmpeg
import pyaudio
import threading
import textwrap
import json
import websocket
import uuid
import time
def format_time(s):
"""Convert seconds (float) to SRT time format."""
hours = int(s // 3600)
minutes = int((s % 3600) // 60)
seconds = int(s % 60)
milliseconds = int((s - int(s)) * 1000)
return f"{hours:02}:{minutes:02}:{seconds:02},{milliseconds:03}"
def create_srt_file(segments, output_file):
with open(output_file, 'w', encoding='utf-8') as srt_file:
segment_number = 1
for segment in segments:
start_time = format_time(float(segment['start']))
end_time = format_time(float(segment['end']))
text = segment['text']
srt_file.write(f"{segment_number}\n")
srt_file.write(f"{start_time} --> {end_time}\n")
srt_file.write(f"{text}\n\n")
segment_number += 1
def resample(file: str, sr: int = 16000):
"""
# https://github.com/openai/whisper/blob/7858aa9c08d98f75575035ecd6481f462d66ca27/whisper/audio.py#L22
Open an audio file and read as mono waveform, resampling as necessary,
save the resampled audio
Args:
file (str): The audio file to open
sr (int): The sample rate to resample the audio if necessary
Returns:
resampled_file (str): The resampled audio file
"""
try:
# This launches a subprocess to decode audio while down-mixing and resampling as necessary.
# Requires the ffmpeg CLI and `ffmpeg-python` package to be installed.
out, _ = (
ffmpeg.input(file, threads=0)
.output("-", format="s16le", acodec="pcm_s16le", ac=1, ar=sr)
.run(cmd=["ffmpeg", "-nostdin"], capture_stdout=True, capture_stderr=True)
)
except ffmpeg.Error as e:
raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e
np_buffer = np.frombuffer(out, dtype=np.int16)
resampled_file = f"{file.split('.')[0]}_resampled.wav"
scipy.io.wavfile.write(resampled_file, sr, np_buffer.astype(np.int16))
return resampled_file
import ffmpeg
import whisper_live.utils as utils
class Client:
@@ -150,10 +96,40 @@ class Client:
self.transcript = []
print("[INFO]: * recording")
def handle_status_messages(self, message_data):
"""Handles server status messages."""
status = message_data["status"]
if status == "WAIT":
self.waiting = True
print(f"[INFO]: Server is full. Estimated wait time {round(message_data['message'])} minutes.")
elif status == "ERROR":
print(f"Message from Server: {message_data['message']}")
self.server_error = True
elif status == "WARNING":
print(f"Message from Server: {message_data['message']}")
def process_segments(self, segments):
"""Processes transcript segments."""
text = []
for i, seg in enumerate(segments):
if not text or text[-1] != seg["text"]:
text.append(seg["text"])
if i == len(segments) - 1:
self.last_segment = seg
elif (self.server_backend == "faster_whisper" and
(not self.transcript or
float(seg['start']) >= float(self.transcript[-1]['end']))):
self.transcript.append(seg)
# Truncate to last 3 entries for brevity.
text = text[-3:]
# utils.clear_screen()
utils.print_transcript(text)
def on_message(self, ws, message):
"""
Callback function called when a message is received from the server.
It updates various attributes of the client based on the received message, including
recording status, language detection, and server messages. If a disconnect message
is received, it sets the recording status to False.
@@ -171,14 +147,7 @@ class Client:
return
if "status" in message.keys():
if message["status"] == "WAIT":
self.waiting = True
print(
f"[INFO]:Server is full. Estimated wait time {round(message['message'])} minutes."
)
elif message["status"] == "ERROR":
print(f"Message from Server: {message['message']}")
self.server_error = True
self.handle_status_messages(message)
return
if "message" in message.keys() and message["message"] == "DISCONNECT":
@@ -199,38 +168,8 @@ class Client:
)
return
if "segments" not in message.keys():
return
message = message["segments"]
text = []
n_segments = len(message)
if n_segments:
for i, seg in enumerate(message):
if text and text[-1] == seg["text"]:
# already got it
continue
text.append(seg["text"])
if i == n_segments-1:
self.last_segment = seg
elif self.server_backend == "faster_whisper":
if not len(self.transcript) or float(seg['start']) >= float(self.transcript[-1]['end']):
self.transcript.append(seg)
# keep only last 3
if len(text) > 3:
text = text[-3:]
wrapper = textwrap.TextWrapper(width=60)
word_list = wrapper.wrap(text="".join(text))
# Print each line.
if os.name == "nt":
os.system("cls")
else:
os.system("clear")
for element in word_list:
print(element)
if "segments" in message.keys():
self.process_segments(message["segments"])
def on_error(self, ws, error):
print(f"[ERROR] WebSocket Error: {error}")
@@ -246,7 +185,7 @@ class Client:
def on_open(self, ws):
"""
Callback function called when the WebSocket connection is successfully opened.
Sends an initial configuration message to the server, including client UID,
language selection, and task type.
@@ -270,8 +209,8 @@ class Client:
def bytes_to_float_array(audio_bytes):
"""
Convert audio data from bytes to a NumPy float array.
It assumes that the audio data is in 16-bit PCM format. The audio data is normalized to
It assumes that the audio data is in 16-bit PCM format. The audio data is normalized to
have values between -1 and 1.
Args:
@@ -299,10 +238,10 @@ class Client:
def play_file(self, filename):
"""
Play an audio file and send it to the server for processing.
Reads an audio file, plays it through the audio output, and simultaneously sends
the audio data to the server for processing. It uses PyAudio to create an audio
stream for playback. The audio data is read from the file in chunks, converted to
the audio data to the server for processing. It uses PyAudio to create an audio
stream for playback. The audio data is read from the file in chunks, converted to
floating-point format, and sent to the server using WebSocket communication.
This method is typically used when you want to process pre-recorded audio and send it
to the server in real-time.
@@ -310,7 +249,7 @@ class Client:
Args:
filename (str): The path to the audio file to be played and sent to the server.
"""
# read audio and create pyaudio stream
with wave.open(filename, "rb") as wavfile:
self.stream = self.p.open(
@@ -356,7 +295,7 @@ class Client:
"""
Close the WebSocket connection and join the WebSocket thread.
First attempts to close the WebSocket connection using `self.client_socket.close()`. After
First attempts to close the WebSocket connection using `self.client_socket.close()`. After
closing the connection, it joins the WebSocket thread to ensure proper termination.
"""
@@ -383,7 +322,7 @@ class Client:
"""
Write audio frames to a WAV file.
The WAV file is created or overwritten with the specified name. The audio frames should be
The WAV file is created or overwritten with the specified name. The audio frames should be
in the correct format and match the specified channel, sample width, and sample rate.
Args:
@@ -433,7 +372,6 @@ class Client:
print("[INFO]: HLS stream processing finished.")
def record(self, out_file="output_recording.wav"):
"""
Record audio data from the input stream and save it to a WAV file.
@@ -444,11 +382,12 @@ class Client:
Audio data is saved in chunks to the "chunks" directory. Each chunk is saved as a separate WAV file.
The recording will continue until the specified duration is reached or until the `RECORDING` flag is set to `False`.
The recording process can be interrupted by sending a KeyboardInterrupt (e.g., pressing Ctrl+C). After recording,
The recording process can be interrupted by sending a KeyboardInterrupt (e.g., pressing Ctrl+C). After recording,
the method combines all the saved audio chunks into the specified `out_file`.
Args:
out_file (str, optional): The name of the output WAV file to save the entire recording. Default is "output_recording.wav".
out_file (str, optional): The name of the output WAV file to save the entire recording.
Default is "output_recording.wav".
"""
n_audio_file = 0
@@ -458,7 +397,7 @@ class Client:
for _ in range(0, int(self.rate / self.chunk * self.record_seconds)):
if not self.recording:
break
data = self.stream.read(self.chunk, exception_on_overflow = False)
data = self.stream.read(self.chunk, exception_on_overflow=False)
self.frames += data
audio_array = Client.bytes_to_float_array(data)
@@ -498,8 +437,8 @@ class Client:
def write_output_recording(self, n_audio_file, out_file):
"""
Combine and save recorded audio chunks into a single WAV file.
The individual audio chunk files are expected to be located in the "chunks" directory. Reads each chunk
The individual audio chunk files are expected to be located in the "chunks" directory. Reads each chunk
file, appends its audio data to the final recording, and then deletes the chunk file. After combining
and saving, the final recording is stored in the specified `out_file`.
@@ -532,7 +471,7 @@ class Client:
def write_srt_file(self, output_path="output.srt"):
self.transcript.append(self.last_segment)
create_srt_file(self.transcript, output_path)
utils.create_srt_file(self.transcript, output_path)
class TranscriptionClient:
@@ -558,13 +497,7 @@ class TranscriptionClient:
transcription_client()
```
"""
def __init__(self,
host,
port,
lang=None,
translate=False,
model="small",
):
def __init__(self, host, port, lang=None, translate=False, model="small"):
self.client = Client(host, port, lang, translate, model)
def __call__(self, audio=None, hls_url=None):
@@ -572,12 +505,12 @@ class TranscriptionClient:
Start the transcription process.
Initiates the transcription process by connecting to the server via a WebSocket. It waits for the server
to be ready to receive audio data and then sends audio for transcription. If an audio file is provided, it
to be ready to receive audio data and then sends audio for transcription. If an audio file is provided, it
will be played and streamed to the server; otherwise, it will perform live recording.
Args:
audio (str, optional): Path to an audio file for transcription. Default is None, which triggers live recording.
"""
print("[INFO]: Waiting for server ready ...")
while not self.client.recording:
@@ -589,7 +522,7 @@ class TranscriptionClient:
if hls_url is not None:
self.client.process_hls_stream(hls_url)
elif audio is not None:
resampled_file = resample(audio)
resampled_file = utils.resample(audio)
self.client.play_file(resampled_file)
else:
self.client.record()
self.client.record()