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from __future__ import annotations | |
import asyncio | |
from collections import OrderedDict | |
from io import BytesIO | |
import time | |
from typing import TYPE_CHECKING, Annotated, Literal | |
from fastapi import ( | |
FastAPI, | |
Form, | |
HTTPException, | |
Path, | |
Query, | |
Response, | |
UploadFile, | |
WebSocket, | |
WebSocketDisconnect, | |
) | |
from fastapi.responses import StreamingResponse | |
from fastapi.websockets import WebSocketState | |
from faster_whisper import WhisperModel | |
from faster_whisper.vad import VadOptions, get_speech_timestamps | |
import gradio as gr | |
import huggingface_hub | |
from pydantic import AfterValidator | |
from faster_whisper_server import utils | |
from faster_whisper_server.asr import FasterWhisperASR | |
from faster_whisper_server.audio import AudioStream, audio_samples_from_file | |
from faster_whisper_server.config import ( | |
SAMPLES_PER_SECOND, | |
Language, | |
ResponseFormat, | |
Task, | |
config, | |
) | |
from faster_whisper_server.gradio_app import create_gradio_demo | |
from faster_whisper_server.logger import logger | |
from faster_whisper_server.server_models import ( | |
ModelObject, | |
TranscriptionJsonResponse, | |
TranscriptionVerboseJsonResponse, | |
) | |
from faster_whisper_server.transcriber import audio_transcriber | |
if TYPE_CHECKING: | |
from collections.abc import Generator, Iterable | |
from faster_whisper.transcribe import Segment, TranscriptionInfo | |
from huggingface_hub.hf_api import ModelInfo | |
loaded_models: OrderedDict[str, WhisperModel] = OrderedDict() | |
def load_model(model_name: str) -> WhisperModel: | |
if model_name in loaded_models: | |
logger.debug(f"{model_name} model already loaded") | |
return loaded_models[model_name] | |
if len(loaded_models) >= config.max_models: | |
oldest_model_name = next(iter(loaded_models)) | |
logger.info(f"Max models ({config.max_models}) reached. Unloading the oldest model: {oldest_model_name}") | |
del loaded_models[oldest_model_name] | |
logger.debug(f"Loading {model_name}...") | |
start = time.perf_counter() | |
# NOTE: will raise an exception if the model name isn't valid | |
whisper = WhisperModel( | |
model_name, | |
device=config.whisper.inference_device, | |
compute_type=config.whisper.compute_type, | |
) | |
logger.info( | |
f"Loaded {model_name} loaded in {time.perf_counter() - start:.2f} seconds. {config.whisper.inference_device}({config.whisper.compute_type}) will be used for inference." # noqa: E501 | |
) | |
loaded_models[model_name] = whisper | |
return whisper | |
app = FastAPI() | |
def health() -> Response: | |
return Response(status_code=200, content="OK") | |
def get_models() -> list[ModelObject]: | |
models = huggingface_hub.list_models(library="ctranslate2", tags="automatic-speech-recognition") | |
models = [ | |
ModelObject( | |
id=model.id, | |
created=int(model.created_at.timestamp()), | |
object_="model", | |
owned_by=model.id.split("/")[0], | |
) | |
for model in models | |
if model.created_at is not None | |
] | |
return models | |
# NOTE: `examples` doesn't work https://github.com/tiangolo/fastapi/discussions/10537 | |
def get_model( | |
model_name: Annotated[str, Path(example="Systran/faster-distil-whisper-large-v3")], | |
) -> ModelObject: | |
models = list( | |
huggingface_hub.list_models(model_name=model_name, library="ctranslate2", tags="automatic-speech-recognition") | |
) | |
if len(models) == 0: | |
raise HTTPException(status_code=404, detail="Model doesn't exists") | |
exact_match: ModelInfo | None = None | |
for model in models: | |
if model.id == model_name: | |
exact_match = model | |
break | |
if exact_match is None: | |
raise HTTPException( | |
status_code=404, | |
detail=f"Model doesn't exists. Possible matches: {", ".join([model.id for model in models])}", | |
) | |
assert exact_match.created_at is not None | |
return ModelObject( | |
id=exact_match.id, | |
created=int(exact_match.created_at.timestamp()), | |
object_="model", | |
owned_by=exact_match.id.split("/")[0], | |
) | |
def segments_to_response( | |
segments: Iterable[Segment], | |
transcription_info: TranscriptionInfo, | |
response_format: ResponseFormat, | |
) -> str | TranscriptionJsonResponse | TranscriptionVerboseJsonResponse: | |
segments = list(segments) | |
if response_format == ResponseFormat.TEXT: # noqa: RET503 | |
return utils.segments_text(segments) | |
elif response_format == ResponseFormat.JSON: | |
return TranscriptionJsonResponse.from_segments(segments) | |
elif response_format == ResponseFormat.VERBOSE_JSON: | |
return TranscriptionVerboseJsonResponse.from_segments(segments, transcription_info) | |
def format_as_sse(data: str) -> str: | |
return f"data: {data}\n\n" | |
def segments_to_streaming_response( | |
segments: Iterable[Segment], | |
transcription_info: TranscriptionInfo, | |
response_format: ResponseFormat, | |
) -> StreamingResponse: | |
def segment_responses() -> Generator[str, None, None]: | |
for segment in segments: | |
if response_format == ResponseFormat.TEXT: | |
data = segment.text | |
elif response_format == ResponseFormat.JSON: | |
data = TranscriptionJsonResponse.from_segments([segment]).model_dump_json() | |
elif response_format == ResponseFormat.VERBOSE_JSON: | |
data = TranscriptionVerboseJsonResponse.from_segment(segment, transcription_info).model_dump_json() | |
yield format_as_sse(data) | |
return StreamingResponse(segment_responses(), media_type="text/event-stream") | |
def handle_default_openai_model(model_name: str) -> str: | |
"""Exists because some callers may not be able override the default("whisper-1") model name. | |
For example, https://github.com/open-webui/open-webui/issues/2248#issuecomment-2162997623. | |
""" | |
if model_name == "whisper-1": | |
logger.info(f"{model_name} is not a valid model name. Using {config.whisper.model} instead.") | |
return config.whisper.model | |
return model_name | |
ModelName = Annotated[str, AfterValidator(handle_default_openai_model)] | |
def translate_file( | |
file: Annotated[UploadFile, Form()], | |
model: Annotated[ModelName, Form()] = config.whisper.model, | |
prompt: Annotated[str | None, Form()] = None, | |
response_format: Annotated[ResponseFormat, Form()] = config.default_response_format, | |
temperature: Annotated[float, Form()] = 0.0, | |
stream: Annotated[bool, Form()] = False, | |
) -> str | TranscriptionJsonResponse | TranscriptionVerboseJsonResponse | StreamingResponse: | |
whisper = load_model(model) | |
segments, transcription_info = whisper.transcribe( | |
file.file, | |
task=Task.TRANSLATE, | |
initial_prompt=prompt, | |
temperature=temperature, | |
vad_filter=True, | |
) | |
if stream: | |
return segments_to_streaming_response(segments, transcription_info, response_format) | |
else: | |
return segments_to_response(segments, transcription_info, response_format) | |
# https://platform.openai.com/docs/api-reference/audio/createTranscription | |
# https://github.com/openai/openai-openapi/blob/master/openapi.yaml#L8915 | |
def transcribe_file( | |
file: Annotated[UploadFile, Form()], | |
model: Annotated[ModelName, Form()] = config.whisper.model, | |
language: Annotated[Language | None, Form()] = config.default_language, | |
prompt: Annotated[str | None, Form()] = None, | |
response_format: Annotated[ResponseFormat, Form()] = config.default_response_format, | |
temperature: Annotated[float, Form()] = 0.0, | |
timestamp_granularities: Annotated[ | |
list[Literal["segment", "word"]], | |
Form(alias="timestamp_granularities[]"), | |
] = ["segment"], | |
stream: Annotated[bool, Form()] = False, | |
) -> str | TranscriptionJsonResponse | TranscriptionVerboseJsonResponse | StreamingResponse: | |
whisper = load_model(model) | |
segments, transcription_info = whisper.transcribe( | |
file.file, | |
task=Task.TRANSCRIBE, | |
language=language, | |
initial_prompt=prompt, | |
word_timestamps="word" in timestamp_granularities, | |
temperature=temperature, | |
vad_filter=True, | |
) | |
if stream: | |
return segments_to_streaming_response(segments, transcription_info, response_format) | |
else: | |
return segments_to_response(segments, transcription_info, response_format) | |
async def audio_receiver(ws: WebSocket, audio_stream: AudioStream) -> None: | |
try: | |
while True: | |
bytes_ = await asyncio.wait_for(ws.receive_bytes(), timeout=config.max_no_data_seconds) | |
logger.debug(f"Received {len(bytes_)} bytes of audio data") | |
audio_samples = audio_samples_from_file(BytesIO(bytes_)) | |
audio_stream.extend(audio_samples) | |
if audio_stream.duration - config.inactivity_window_seconds >= 0: | |
audio = audio_stream.after(audio_stream.duration - config.inactivity_window_seconds) | |
vad_opts = VadOptions(min_silence_duration_ms=500, speech_pad_ms=0) | |
# NOTE: This is a synchronous operation that runs every time new data is received. | |
# This shouldn't be an issue unless data is being received in tiny chunks or the user's machine is a potato. # noqa: E501 | |
timestamps = get_speech_timestamps(audio.data, vad_opts) | |
if len(timestamps) == 0: | |
logger.info(f"No speech detected in the last {config.inactivity_window_seconds} seconds.") | |
break | |
elif ( | |
# last speech end time | |
config.inactivity_window_seconds - timestamps[-1]["end"] / SAMPLES_PER_SECOND | |
>= config.max_inactivity_seconds | |
): | |
logger.info(f"Not enough speech in the last {config.inactivity_window_seconds} seconds.") | |
break | |
except TimeoutError: | |
logger.info(f"No data received in {config.max_no_data_seconds} seconds. Closing the connection.") | |
except WebSocketDisconnect as e: | |
logger.info(f"Client disconnected: {e}") | |
audio_stream.close() | |
async def transcribe_stream( | |
ws: WebSocket, | |
model: Annotated[ModelName, Query()] = config.whisper.model, | |
language: Annotated[Language | None, Query()] = config.default_language, | |
response_format: Annotated[ResponseFormat, Query()] = config.default_response_format, | |
temperature: Annotated[float, Query()] = 0.0, | |
) -> None: | |
await ws.accept() | |
transcribe_opts = { | |
"language": language, | |
"temperature": temperature, | |
"vad_filter": True, | |
"condition_on_previous_text": False, | |
} | |
whisper = load_model(model) | |
asr = FasterWhisperASR(whisper, **transcribe_opts) | |
audio_stream = AudioStream() | |
async with asyncio.TaskGroup() as tg: | |
tg.create_task(audio_receiver(ws, audio_stream)) | |
async for transcription in audio_transcriber(asr, audio_stream): | |
logger.debug(f"Sending transcription: {transcription.text}") | |
if ws.client_state == WebSocketState.DISCONNECTED: | |
break | |
if response_format == ResponseFormat.TEXT: | |
await ws.send_text(transcription.text) | |
elif response_format == ResponseFormat.JSON: | |
await ws.send_json(TranscriptionJsonResponse.from_transcription(transcription).model_dump()) | |
elif response_format == ResponseFormat.VERBOSE_JSON: | |
await ws.send_json(TranscriptionVerboseJsonResponse.from_transcription(transcription).model_dump()) | |
if ws.client_state != WebSocketState.DISCONNECTED: | |
logger.info("Closing the connection.") | |
await ws.close() | |
app = gr.mount_gradio_app(app, create_gradio_demo(config), path="/") | |