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from fastapi import FastAPI, HTTPException
from fastapi.responses import JSONResponse, FileResponse
from pydantic import BaseModel
import numpy as np
import io
import soundfile as sf
import base64
import logging
import torch
import librosa
from transformers import Wav2Vec2ForCTC, AutoProcessor
from pathlib import Path
from moviepy.editor import VideoFileClip
import magic # For MIME type detection
import tempfile
import os
# Import functions from other modules
from asr import transcribe, ASR_LANGUAGES
from tts import synthesize, TTS_LANGUAGES
from lid import identify
from asr import ASR_SAMPLING_RATE, transcribe
# Configure logging
logging.basicConfig(level=logging.INFO)
logger = logging.getLogger(__name__)
app = FastAPI(title="MMS: Scaling Speech Technology to 1000+ languages")
# Define request models
class AudioRequest(BaseModel):
audio: str # Base64 encoded audio or video data
language: str
class TTSRequest(BaseModel):
text: str
language: str
speed: float
def detect_mime_type(input_bytes):
mime = magic.Magic(mime=True)
return mime.from_buffer(input_bytes)
def extract_audio(input_bytes):
mime_type = detect_mime_type(input_bytes)
if mime_type.startswith('audio/'):
return sf.read(io.BytesIO(input_bytes))
elif mime_type.startswith('video/'):
with tempfile.NamedTemporaryFile(delete=False, suffix='.webm') as temp_file:
temp_file.write(input_bytes)
temp_file_path = temp_file.name
try:
video = VideoFileClip(temp_file_path)
audio = video.audio
audio_array = audio.to_soundarray()
sample_rate = audio.fps
video.close()
return audio_array, sample_rate
finally:
os.unlink(temp_file_path)
else:
raise ValueError(f"Unsupported MIME type: {mime_type}")
@app.post("/transcribe")
async def transcribe_audio(request: AudioRequest):
try:
input_bytes = base64.b64decode(request.audio)
audio_array, sample_rate = extract_audio(input_bytes)
# Convert to mono if stereo
if len(audio_array.shape) > 1:
audio_array = audio_array.mean(axis=1)
# Ensure audio_array is float32
audio_array = audio_array.astype(np.float32)
# Resample if necessary
if sample_rate != ASR_SAMPLING_RATE:
audio_array = librosa.resample(audio_array, orig_sr=sample_rate, target_sr=ASR_SAMPLING_RATE)
result = transcribe(audio_array, request.language)
return JSONResponse(content={"transcription": result})
except Exception as e:
logger.error(f"Error in transcribe_audio: {str(e)}")
raise HTTPException(status_code=500, detail=f"An error occurred: {str(e)}")
@app.post("/synthesize")
async def synthesize_speech(request: TTSRequest):
try:
audio, filtered_text = synthesize(request.text, request.language, request.speed)
# Convert numpy array to bytes
buffer = io.BytesIO()
sf.write(buffer, audio, 22050, format='wav')
buffer.seek(0)
return FileResponse(
buffer,
media_type="audio/wav",
headers={"Content-Disposition": "attachment; filename=synthesized_audio.wav"}
)
except Exception as e:
logger.error(f"Error in synthesize_speech: {str(e)}")
raise HTTPException(status_code=500, detail=f"An error occurred: {str(e)}")
@app.post("/identify")
async def identify_language(request: AudioRequest):
try:
input_bytes = base64.b64decode(request.audio)
audio_array, sample_rate = extract_audio(input_bytes)
result = identify(audio_array)
return JSONResponse(content={"language_identification": result})
except Exception as e:
logger.error(f"Error in identify_language: {str(e)}")
raise HTTPException(status_code=500, detail=f"An error occurred: {str(e)}")
@app.get("/asr_languages")
async def get_asr_languages():
try:
return JSONResponse(content=ASR_LANGUAGES)
except Exception as e:
logger.error(f"Error in get_asr_languages: {str(e)}")
raise HTTPException(status_code=500, detail=f"An error occurred: {str(e)}")
@app.get("/tts_languages")
async def get_tts_languages():
try:
return JSONResponse(content=TTS_LANGUAGES)
except Exception as e:
logger.error(f"Error in get_tts_languages: {str(e)}")
raise HTTPException(status_code=500, detail=f"An error occurred: {str(e)}")