Files
WhisperLive/whisper_live/backend/faster_whisper_backend.py
T
makaveli10 c1ac71ada0 Refactor 🔨
Signed-off-by: makaveli10 <vineet.suryan@collabora.com>
2025-03-24 16:48:33 +05:30

382 lines
16 KiB
Python

import json
import logging
import threading
import time
import torch
from whisper_live.transcriber.transcriber_faster_whisper import WhisperModel
from whisper_live.backend.base import ServeClientBase
class ServeClientFasterWhisper(ServeClientBase):
SINGLE_MODEL = None
SINGLE_MODEL_LOCK = threading.Lock()
def __init__(self, websocket, task="transcribe", device=None, language=None, client_uid=None, model="small.en",
initial_prompt=None, vad_parameters=None, use_vad=True, single_model=False):
"""
Initialize a ServeClient instance.
The Whisper model is initialized based on the client's language and device availability.
The transcription thread is started upon initialization. A "SERVER_READY" message is sent
to the client to indicate that the server is ready.
Args:
websocket (WebSocket): The WebSocket connection for the client.
task (str, optional): The task type, e.g., "transcribe." Defaults to "transcribe".
device (str, optional): The device type for Whisper, "cuda" or "cpu". Defaults to None.
language (str, optional): The language for transcription. Defaults to None.
client_uid (str, optional): A unique identifier for the client. Defaults to None.
model (str, optional): The whisper model size. Defaults to 'small.en'
initial_prompt (str, optional): Prompt for whisper inference. Defaults to None.
single_model (bool, optional): Whether to instantiate a new model for each client connection. Defaults to False.
"""
super().__init__(client_uid, websocket)
self.model_sizes = [
"tiny", "tiny.en", "base", "base.en", "small", "small.en",
"medium", "medium.en", "large-v2", "large-v3", "distil-small.en",
"distil-medium.en", "distil-large-v2", "distil-large-v3",
"large-v3-turbo", "turbo"
]
self.model_size_or_path = model
self.language = "en" if self.model_size_or_path.endswith("en") else language
self.task = task
self.initial_prompt = initial_prompt
self.vad_parameters = vad_parameters or {"onset": 0.5}
self.no_speech_thresh = 0.45
self.same_output_threshold = 10
self.end_time_for_same_output = None
device = "cuda" if torch.cuda.is_available() else "cpu"
if device == "cuda":
major, _ = torch.cuda.get_device_capability(device)
self.compute_type = "float16" if major >= 7 else "float32"
else:
self.compute_type = "int8"
if self.model_size_or_path is None:
return
logging.info(f"Using Device={device} with precision {self.compute_type}")
try:
if single_model:
if ServeClientFasterWhisper.SINGLE_MODEL is None:
self.create_model(device)
ServeClientFasterWhisper.SINGLE_MODEL = self.transcriber
else:
self.transcriber = ServeClientFasterWhisper.SINGLE_MODEL
else:
self.create_model(device)
except Exception as e:
logging.error(f"Failed to load model: {e}")
self.websocket.send(json.dumps({
"uid": self.client_uid,
"status": "ERROR",
"message": f"Failed to load model: {str(self.model_size_or_path)}"
}))
self.websocket.close()
return
self.use_vad = use_vad
# threading
self.trans_thread = threading.Thread(target=self.speech_to_text)
self.trans_thread.start()
self.websocket.send(
json.dumps(
{
"uid": self.client_uid,
"message": self.SERVER_READY,
"backend": "faster_whisper"
}
)
)
def create_model(self, device):
"""
Instantiates a new model, sets it as the transcriber.
"""
self.transcriber = WhisperModel(
self.model_size_or_path,
device=device,
compute_type=self.compute_type,
local_files_only=False,
)
def check_valid_model(self, model_size):
"""
Check if it's a valid whisper model size.
Args:
model_size (str): The name of the model size to check.
Returns:
str: The model size if valid, None otherwise.
"""
if model_size not in self.model_sizes:
self.websocket.send(
json.dumps(
{
"uid": self.client_uid,
"status": "ERROR",
"message": f"Invalid model size {model_size}. Available choices: {self.model_sizes}"
}
)
)
return None
return model_size
def set_language(self, info):
"""
Updates the language attribute based on the detected language information.
Args:
info (object): An object containing the detected language and its probability. This object
must have at least two attributes: `language`, a string indicating the detected
language, and `language_probability`, a float representing the confidence level
of the language detection.
"""
if info.language_probability > 0.5:
self.language = info.language
logging.info(f"Detected language {self.language} with probability {info.language_probability}")
self.websocket.send(json.dumps(
{"uid": self.client_uid, "language": self.language, "language_prob": info.language_probability}))
def transcribe_audio(self, input_sample):
"""
Transcribes the provided audio sample using the configured transcriber instance.
If the language has not been set, it updates the session's language based on the transcription
information.
Args:
input_sample (np.array): The audio chunk to be transcribed. This should be a NumPy
array representing the audio data.
Returns:
The transcription result from the transcriber. The exact format of this result
depends on the implementation of the `transcriber.transcribe` method but typically
includes the transcribed text.
"""
if ServeClientFasterWhisper.SINGLE_MODEL:
ServeClientFasterWhisper.SINGLE_MODEL_LOCK.acquire()
result, info = self.transcriber.transcribe(
input_sample,
initial_prompt=self.initial_prompt,
language=self.language,
task=self.task,
vad_filter=self.use_vad,
vad_parameters=self.vad_parameters if self.use_vad else None)
if ServeClientFasterWhisper.SINGLE_MODEL:
ServeClientFasterWhisper.SINGLE_MODEL_LOCK.release()
if self.language is None and info is not None:
self.set_language(info)
return result
def get_previous_output(self):
"""
Retrieves previously generated transcription outputs if no new transcription is available
from the current audio chunks.
Checks the time since the last transcription output and, if it is within a specified
threshold, returns the most recent segments of transcribed text. It also manages
adding a pause (blank segment) to indicate a significant gap in speech based on a defined
threshold.
Returns:
segments (list): A list of transcription segments. This may include the most recent
transcribed text segments or a blank segment to indicate a pause
in speech.
"""
segments = []
if self.t_start is None:
self.t_start = time.time()
if time.time() - self.t_start < self.show_prev_out_thresh:
segments = self.prepare_segments()
# add a blank if there is no speech for 3 seconds
if len(self.text) and self.text[-1] != '':
if time.time() - self.t_start > self.add_pause_thresh:
self.text.append('')
return segments
def handle_transcription_output(self, result, duration):
"""
Handle the transcription output, updating the transcript and sending data to the client.
Args:
result (str): The result from whisper inference i.e. the list of segments.
duration (float): Duration of the transcribed audio chunk.
"""
segments = []
if len(result):
self.t_start = None
last_segment = self.update_segments(result, duration)
segments = self.prepare_segments(last_segment)
else:
# show previous output if there is pause i.e. no output from whisper
segments = self.get_previous_output()
if len(segments):
self.send_transcription_to_client(segments)
def speech_to_text(self):
"""
Process an audio stream in an infinite loop, continuously transcribing the speech.
This method continuously receives audio frames, performs real-time transcription, and sends
transcribed segments to the client via a WebSocket connection.
If the client's language is not detected, it waits for 30 seconds of audio input to make a language prediction.
It utilizes the Whisper ASR model to transcribe the audio, continuously processing and streaming results. Segments
are sent to the client in real-time, and a history of segments is maintained to provide context.Pauses in speech
(no output from Whisper) are handled by showing the previous output for a set duration. A blank segment is added if
there is no speech for a specified duration to indicate a pause.
Raises:
Exception: If there is an issue with audio processing or WebSocket communication.
"""
while True:
if self.exit:
logging.info("Exiting speech to text thread")
break
if self.frames_np is None:
continue
self.clip_audio_if_no_valid_segment()
input_bytes, duration = self.get_audio_chunk_for_processing()
if duration < 1.0:
time.sleep(0.1) # wait for audio chunks to arrive
continue
try:
input_sample = input_bytes.copy()
result = self.transcribe_audio(input_sample)
if result is None or self.language is None:
self.timestamp_offset += duration
time.sleep(0.25) # wait for voice activity, result is None when no voice activity
continue
self.handle_transcription_output(result, duration)
except Exception as e:
logging.error(f"[ERROR]: Failed to transcribe audio chunk: {e}")
time.sleep(0.01)
def format_segment(self, start, end, text, completed=False):
"""
Formats a transcription segment with precise start and end times alongside the transcribed text.
Args:
start (float): The start time of the transcription segment in seconds.
end (float): The end time of the transcription segment in seconds.
text (str): The transcribed text corresponding to the segment.
Returns:
dict: A dictionary representing the formatted transcription segment, including
'start' and 'end' times as strings with three decimal places and the 'text'
of the transcription.
"""
return {
'start': "{:.3f}".format(start),
'end': "{:.3f}".format(end),
'text': text,
'completed': completed
}
def update_segments(self, segments, duration):
"""
Processes the segments from whisper. Appends all the segments to the list
except for the last segment assuming that it is incomplete.
Updates the ongoing transcript with transcribed segments, including their start and end times.
Complete segments are appended to the transcript in chronological order. Incomplete segments
(assumed to be the last one) are processed to identify repeated content. If the same incomplete
segment is seen multiple times, it updates the offset and appends the segment to the transcript.
A threshold is used to detect repeated content and ensure it is only included once in the transcript.
The timestamp offset is updated based on the duration of processed segments. The method returns the
last processed segment, allowing it to be sent to the client for real-time updates.
Args:
segments(dict) : dictionary of segments as returned by whisper
duration(float): duration of the current chunk
Returns:
dict or None: The last processed segment with its start time, end time, and transcribed text.
Returns None if there are no valid segments to process.
"""
offset = None
self.current_out = ''
last_segment = None
# process complete segments
if len(segments) > 1 and segments[-1].no_speech_prob <= self.no_speech_thresh:
for i, s in enumerate(segments[:-1]):
text_ = s.text
self.text.append(text_)
with self.lock:
start, end = self.timestamp_offset + s.start, self.timestamp_offset + min(duration, s.end)
if start >= end:
continue
if s.no_speech_prob > self.no_speech_thresh:
continue
self.transcript.append(self.format_segment(start, end, text_, completed=True))
offset = min(duration, s.end)
# only process the last segment if it satisfies the no_speech_thresh
if segments[-1].no_speech_prob <= self.no_speech_thresh:
self.current_out += segments[-1].text
with self.lock:
last_segment = self.format_segment(
self.timestamp_offset + segments[-1].start,
self.timestamp_offset + min(duration, segments[-1].end),
self.current_out,
completed=False
)
if self.current_out.strip() == self.prev_out.strip() and self.current_out != '':
self.same_output_count += 1
# if we remove the audio because of same output on the nth reptition we might remove the
# audio thats not yet transcribed so, capturing the time when it was repeated for the first time
if self.end_time_for_same_output is None:
self.end_time_for_same_output = segments[-1].end
time.sleep(0.1) # wait for some voice activity just in case there is an unitended pause from the speaker for better punctuations.
else:
self.same_output_count = 0
self.end_time_for_same_output = None
# if same incomplete segment is seen multiple times then update the offset
# and append the segment to the list
if self.same_output_count > self.same_output_threshold:
if not len(self.text) or self.text[-1].strip().lower() != self.current_out.strip().lower():
self.text.append(self.current_out)
with self.lock:
self.transcript.append(self.format_segment(
self.timestamp_offset,
self.timestamp_offset + min(duration, self.end_time_for_same_output),
self.current_out,
completed=True
))
self.current_out = ''
offset = min(duration, self.end_time_for_same_output)
self.same_output_count = 0
last_segment = None
self.end_time_for_same_output = None
else:
self.prev_out = self.current_out
# update offset
if offset is not None:
with self.lock:
self.timestamp_offset += offset
return last_segment