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import os
import logging
import json
import gradio as gr
import numpy as np
from dotenv import load_dotenv
from fastapi import FastAPI
from fastapi.staticfiles import StaticFiles
from fastapi.responses import StreamingResponse, HTMLResponse
from fastrtc import (
AdditionalOutputs,
ReplyOnPause,
Stream,
AlgoOptions,
SileroVadOptions,
audio_to_bytes,
)
from utils.logger_config import setup_logging
from utils.device import get_device
from utils.turn_server import get_rtc_credentials
from utils.model import initialize_whisper_model
load_dotenv()
setup_logging()
logger = logging.getLogger(__name__)
UI_MODE = os.getenv("UI_MODE", "fastapi").lower() # gradio | fastapi
UI_TYPE = os.getenv("UI_TYPE", "base").lower() # base | screen
APP_MODE = os.getenv("APP_MODE", "local").lower() # local | deployed
TURN_PROVIDER = os.getenv("TURN_PROVIDER", "hf-cloudflare") # hf-cloudflare | cloudflare | twilio
MODEL_ID = os.getenv("MODEL_ID", "openai/whisper-large-v3-turbo")
LANGUAGE = os.getenv("LANGUAGE", "english")
logger.info(f"""
--------------------------------------
Configuration (environment variables):
- UI_MODE: {UI_MODE}
- UI_TYPE: {UI_TYPE}
- APP_MODE: {APP_MODE}
- TURN_PROVIDER: {TURN_PROVIDER}
- MODEL_ID: {MODEL_ID}
- LANGUAGE: {LANGUAGE}
--------------------------------------
""")
transcribe_pipeline = initialize_whisper_model(
model_id=MODEL_ID,
try_compile=True, # Set to False to disable trying to compile the model
try_use_flash_attention=True, # Set to False to disable trying to use flash attention
device=get_device(force_cpu=False) # Set to False to use GPU if available
)
async def transcribe(audio: tuple[int, np.ndarray]):
sample_rate, audio_array = audio
logger.info(f"Sample rate: {sample_rate}Hz, Shape: {audio_array.shape}")
outputs = transcribe_pipeline(
audio_to_bytes(audio),
chunk_length_s=5,
batch_size=1,
generate_kwargs={
'task': 'transcribe',
'language': LANGUAGE,
},
#return_timestamps="word"
)
yield AdditionalOutputs(outputs["text"].strip())
logger.info("Initializing FastRTC stream")
stream = Stream(
handler=ReplyOnPause(
transcribe,
algo_options=AlgoOptions(
# Duration in seconds of audio chunks passed to the VAD model (default 0.6)
audio_chunk_duration=0.5,
# If the chunk has more than started_talking_threshold seconds of speech, the user started talking (default 0.2)
started_talking_threshold=0.1,
# If, after the user started speaking, there is a chunk with less than speech_threshold seconds of speech, the user stopped speaking. (default 0.1)
speech_threshold=0.1,
# Max duration of speech chunks before the handler is triggered, even if a pause is not detected by the VAD model. (default -inf)
max_continuous_speech_s=15
),
model_options=SileroVadOptions(
# Threshold for what is considered speech (default 0.5)
threshold=0.5,
# Final speech chunks shorter min_speech_duration_ms are thrown out (default 250)
min_speech_duration_ms=250,
# Max duration of speech chunks, longer will be split at the timestamp of the last silence that lasts more than 100ms (if any) or just before max_speech_duration_s (default float('inf')) (used internally in the VAD algorithm to split the audio that's passed to the algorithm)
max_speech_duration_s=10,
# Wait for ms at the end of each speech chunk before separating it (default 2000)
min_silence_duration_ms=400,
# Chunk size for VAD model. Can be 512, 1024, 1536 for 16k s.r. (default 1024)
window_size_samples=1024,
# Final speech chunks are padded by speech_pad_ms each side (default 400)
speech_pad_ms=200,
),
),
# send-receive: bidirectional streaming (default)
# send: client to server only
# receive: server to client only
modality="audio",
mode="send",
additional_outputs=[
gr.Textbox(label="Transcript"),
],
additional_outputs_handler=lambda current, new: current + " " + new,
rtc_configuration=get_rtc_credentials(provider=TURN_PROVIDER) if APP_MODE == "deployed" else None,
)
app = FastAPI()
app.mount("/static", StaticFiles(directory="static"), name="static")
stream.mount(app)
@app.get("/")
async def index():
if UI_TYPE == "base":
html_content = open("static/index.html").read()
elif UI_TYPE == "screen":
html_content = open("static/index-screen.html").read()
rtc_configuration = await get_rtc_credentials(provider=TURN_PROVIDER) if APP_MODE == "deployed" else None
logger.info(f"RTC configuration: {rtc_configuration}")
html_content = html_content.replace("__INJECTED_RTC_CONFIG__", json.dumps(rtc_configuration))
return HTMLResponse(content=html_content)
@app.get("/transcript")
def _(webrtc_id: str):
logger.debug(f"New transcript stream request for webrtc_id: {webrtc_id}")
async def output_stream():
try:
async for output in stream.output_stream(webrtc_id):
transcript = output.args[0]
logger.debug(f"Sending transcript for {webrtc_id}: {transcript[:50]}...")
yield f"event: output\ndata: {transcript}\n\n"
except Exception as e:
logger.error(f"Error in transcript stream for {webrtc_id}: {str(e)}")
raise
return StreamingResponse(output_stream(), media_type="text/event-stream")
if __name__ == "__main__":
server_name = os.getenv("SERVER_NAME", "localhost")
port = os.getenv("PORT", 7860)
if UI_MODE == "gradio":
logger.info("Launching Gradio UI")
stream.ui.launch(
server_port=port,
server_name=server_name,
ssl_verify=False,
debug=True
)
else:
import uvicorn
logger.info("Launching FastAPI server")
uvicorn.run(app, host=server_name, port=port)