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import hashlib | |
import json | |
import logging | |
import os | |
import uuid | |
from functools import lru_cache | |
from pathlib import Path | |
from pydub import AudioSegment | |
from pydub.silence import split_on_silence | |
from concurrent.futures import ThreadPoolExecutor | |
import aiohttp | |
import aiofiles | |
import requests | |
import mimetypes | |
from fastapi import ( | |
Depends, | |
FastAPI, | |
File, | |
HTTPException, | |
Request, | |
UploadFile, | |
status, | |
APIRouter, | |
) | |
from fastapi.middleware.cors import CORSMiddleware | |
from fastapi.responses import FileResponse | |
from pydantic import BaseModel | |
from open_webui.utils.auth import get_admin_user, get_verified_user | |
from open_webui.config import ( | |
WHISPER_MODEL_AUTO_UPDATE, | |
WHISPER_MODEL_DIR, | |
CACHE_DIR, | |
WHISPER_LANGUAGE, | |
) | |
from open_webui.constants import ERROR_MESSAGES | |
from open_webui.env import ( | |
AIOHTTP_CLIENT_SESSION_SSL, | |
AIOHTTP_CLIENT_TIMEOUT, | |
ENV, | |
SRC_LOG_LEVELS, | |
DEVICE_TYPE, | |
ENABLE_FORWARD_USER_INFO_HEADERS, | |
) | |
router = APIRouter() | |
# Constants | |
MAX_FILE_SIZE_MB = 20 | |
MAX_FILE_SIZE = MAX_FILE_SIZE_MB * 1024 * 1024 # Convert MB to bytes | |
AZURE_MAX_FILE_SIZE_MB = 200 | |
AZURE_MAX_FILE_SIZE = AZURE_MAX_FILE_SIZE_MB * 1024 * 1024 # Convert MB to bytes | |
log = logging.getLogger(__name__) | |
log.setLevel(SRC_LOG_LEVELS["AUDIO"]) | |
SPEECH_CACHE_DIR = CACHE_DIR / "audio" / "speech" | |
SPEECH_CACHE_DIR.mkdir(parents=True, exist_ok=True) | |
########################################## | |
# | |
# Utility functions | |
# | |
########################################## | |
from pydub import AudioSegment | |
from pydub.utils import mediainfo | |
def is_audio_conversion_required(file_path): | |
""" | |
Check if the given audio file needs conversion to mp3. | |
""" | |
SUPPORTED_FORMATS = {"flac", "m4a", "mp3", "mp4", "mpeg", "wav", "webm"} | |
if not os.path.isfile(file_path): | |
log.error(f"File not found: {file_path}") | |
return False | |
try: | |
info = mediainfo(file_path) | |
codec_name = info.get("codec_name", "").lower() | |
codec_type = info.get("codec_type", "").lower() | |
codec_tag_string = info.get("codec_tag_string", "").lower() | |
if codec_name == "aac" and codec_type == "audio" and codec_tag_string == "mp4a": | |
# File is AAC/mp4a audio, recommend mp3 conversion | |
return True | |
# If the codec name or file extension is in the supported formats | |
if ( | |
codec_name in SUPPORTED_FORMATS | |
or os.path.splitext(file_path)[1][1:].lower() in SUPPORTED_FORMATS | |
): | |
return False # Already supported | |
return True | |
except Exception as e: | |
log.error(f"Error getting audio format: {e}") | |
return False | |
def convert_audio_to_mp3(file_path): | |
"""Convert audio file to mp3 format.""" | |
try: | |
output_path = os.path.splitext(file_path)[0] + ".mp3" | |
audio = AudioSegment.from_file(file_path) | |
audio.export(output_path, format="mp3") | |
log.info(f"Converted {file_path} to {output_path}") | |
return output_path | |
except Exception as e: | |
log.error(f"Error converting audio file: {e}") | |
return None | |
def set_faster_whisper_model(model: str, auto_update: bool = False): | |
whisper_model = None | |
if model: | |
from faster_whisper import WhisperModel | |
faster_whisper_kwargs = { | |
"model_size_or_path": model, | |
"device": DEVICE_TYPE if DEVICE_TYPE and DEVICE_TYPE == "cuda" else "cpu", | |
"compute_type": "int8", | |
"download_root": WHISPER_MODEL_DIR, | |
"local_files_only": not auto_update, | |
} | |
try: | |
whisper_model = WhisperModel(**faster_whisper_kwargs) | |
except Exception: | |
log.warning( | |
"WhisperModel initialization failed, attempting download with local_files_only=False" | |
) | |
faster_whisper_kwargs["local_files_only"] = False | |
whisper_model = WhisperModel(**faster_whisper_kwargs) | |
return whisper_model | |
########################################## | |
# | |
# Audio API | |
# | |
########################################## | |
class TTSConfigForm(BaseModel): | |
OPENAI_API_BASE_URL: str | |
OPENAI_API_KEY: str | |
API_KEY: str | |
ENGINE: str | |
MODEL: str | |
VOICE: str | |
SPLIT_ON: str | |
AZURE_SPEECH_REGION: str | |
AZURE_SPEECH_BASE_URL: str | |
AZURE_SPEECH_OUTPUT_FORMAT: str | |
class STTConfigForm(BaseModel): | |
OPENAI_API_BASE_URL: str | |
OPENAI_API_KEY: str | |
ENGINE: str | |
MODEL: str | |
WHISPER_MODEL: str | |
DEEPGRAM_API_KEY: str | |
AZURE_API_KEY: str | |
AZURE_REGION: str | |
AZURE_LOCALES: str | |
AZURE_BASE_URL: str | |
AZURE_MAX_SPEAKERS: str | |
class AudioConfigUpdateForm(BaseModel): | |
tts: TTSConfigForm | |
stt: STTConfigForm | |
async def get_audio_config(request: Request, user=Depends(get_admin_user)): | |
return { | |
"tts": { | |
"OPENAI_API_BASE_URL": request.app.state.config.TTS_OPENAI_API_BASE_URL, | |
"OPENAI_API_KEY": request.app.state.config.TTS_OPENAI_API_KEY, | |
"API_KEY": request.app.state.config.TTS_API_KEY, | |
"ENGINE": request.app.state.config.TTS_ENGINE, | |
"MODEL": request.app.state.config.TTS_MODEL, | |
"VOICE": request.app.state.config.TTS_VOICE, | |
"SPLIT_ON": request.app.state.config.TTS_SPLIT_ON, | |
"AZURE_SPEECH_REGION": request.app.state.config.TTS_AZURE_SPEECH_REGION, | |
"AZURE_SPEECH_BASE_URL": request.app.state.config.TTS_AZURE_SPEECH_BASE_URL, | |
"AZURE_SPEECH_OUTPUT_FORMAT": request.app.state.config.TTS_AZURE_SPEECH_OUTPUT_FORMAT, | |
}, | |
"stt": { | |
"OPENAI_API_BASE_URL": request.app.state.config.STT_OPENAI_API_BASE_URL, | |
"OPENAI_API_KEY": request.app.state.config.STT_OPENAI_API_KEY, | |
"ENGINE": request.app.state.config.STT_ENGINE, | |
"MODEL": request.app.state.config.STT_MODEL, | |
"WHISPER_MODEL": request.app.state.config.WHISPER_MODEL, | |
"DEEPGRAM_API_KEY": request.app.state.config.DEEPGRAM_API_KEY, | |
"AZURE_API_KEY": request.app.state.config.AUDIO_STT_AZURE_API_KEY, | |
"AZURE_REGION": request.app.state.config.AUDIO_STT_AZURE_REGION, | |
"AZURE_LOCALES": request.app.state.config.AUDIO_STT_AZURE_LOCALES, | |
"AZURE_BASE_URL": request.app.state.config.AUDIO_STT_AZURE_BASE_URL, | |
"AZURE_MAX_SPEAKERS": request.app.state.config.AUDIO_STT_AZURE_MAX_SPEAKERS, | |
}, | |
} | |
async def update_audio_config( | |
request: Request, form_data: AudioConfigUpdateForm, user=Depends(get_admin_user) | |
): | |
request.app.state.config.TTS_OPENAI_API_BASE_URL = form_data.tts.OPENAI_API_BASE_URL | |
request.app.state.config.TTS_OPENAI_API_KEY = form_data.tts.OPENAI_API_KEY | |
request.app.state.config.TTS_API_KEY = form_data.tts.API_KEY | |
request.app.state.config.TTS_ENGINE = form_data.tts.ENGINE | |
request.app.state.config.TTS_MODEL = form_data.tts.MODEL | |
request.app.state.config.TTS_VOICE = form_data.tts.VOICE | |
request.app.state.config.TTS_SPLIT_ON = form_data.tts.SPLIT_ON | |
request.app.state.config.TTS_AZURE_SPEECH_REGION = form_data.tts.AZURE_SPEECH_REGION | |
request.app.state.config.TTS_AZURE_SPEECH_BASE_URL = ( | |
form_data.tts.AZURE_SPEECH_BASE_URL | |
) | |
request.app.state.config.TTS_AZURE_SPEECH_OUTPUT_FORMAT = ( | |
form_data.tts.AZURE_SPEECH_OUTPUT_FORMAT | |
) | |
request.app.state.config.STT_OPENAI_API_BASE_URL = form_data.stt.OPENAI_API_BASE_URL | |
request.app.state.config.STT_OPENAI_API_KEY = form_data.stt.OPENAI_API_KEY | |
request.app.state.config.STT_ENGINE = form_data.stt.ENGINE | |
request.app.state.config.STT_MODEL = form_data.stt.MODEL | |
request.app.state.config.WHISPER_MODEL = form_data.stt.WHISPER_MODEL | |
request.app.state.config.DEEPGRAM_API_KEY = form_data.stt.DEEPGRAM_API_KEY | |
request.app.state.config.AUDIO_STT_AZURE_API_KEY = form_data.stt.AZURE_API_KEY | |
request.app.state.config.AUDIO_STT_AZURE_REGION = form_data.stt.AZURE_REGION | |
request.app.state.config.AUDIO_STT_AZURE_LOCALES = form_data.stt.AZURE_LOCALES | |
request.app.state.config.AUDIO_STT_AZURE_BASE_URL = form_data.stt.AZURE_BASE_URL | |
request.app.state.config.AUDIO_STT_AZURE_MAX_SPEAKERS = ( | |
form_data.stt.AZURE_MAX_SPEAKERS | |
) | |
if request.app.state.config.STT_ENGINE == "": | |
request.app.state.faster_whisper_model = set_faster_whisper_model( | |
form_data.stt.WHISPER_MODEL, WHISPER_MODEL_AUTO_UPDATE | |
) | |
return { | |
"tts": { | |
"OPENAI_API_BASE_URL": request.app.state.config.TTS_OPENAI_API_BASE_URL, | |
"OPENAI_API_KEY": request.app.state.config.TTS_OPENAI_API_KEY, | |
"API_KEY": request.app.state.config.TTS_API_KEY, | |
"ENGINE": request.app.state.config.TTS_ENGINE, | |
"MODEL": request.app.state.config.TTS_MODEL, | |
"VOICE": request.app.state.config.TTS_VOICE, | |
"SPLIT_ON": request.app.state.config.TTS_SPLIT_ON, | |
"AZURE_SPEECH_REGION": request.app.state.config.TTS_AZURE_SPEECH_REGION, | |
"AZURE_SPEECH_BASE_URL": request.app.state.config.TTS_AZURE_SPEECH_BASE_URL, | |
"AZURE_SPEECH_OUTPUT_FORMAT": request.app.state.config.TTS_AZURE_SPEECH_OUTPUT_FORMAT, | |
}, | |
"stt": { | |
"OPENAI_API_BASE_URL": request.app.state.config.STT_OPENAI_API_BASE_URL, | |
"OPENAI_API_KEY": request.app.state.config.STT_OPENAI_API_KEY, | |
"ENGINE": request.app.state.config.STT_ENGINE, | |
"MODEL": request.app.state.config.STT_MODEL, | |
"WHISPER_MODEL": request.app.state.config.WHISPER_MODEL, | |
"DEEPGRAM_API_KEY": request.app.state.config.DEEPGRAM_API_KEY, | |
"AZURE_API_KEY": request.app.state.config.AUDIO_STT_AZURE_API_KEY, | |
"AZURE_REGION": request.app.state.config.AUDIO_STT_AZURE_REGION, | |
"AZURE_LOCALES": request.app.state.config.AUDIO_STT_AZURE_LOCALES, | |
"AZURE_BASE_URL": request.app.state.config.AUDIO_STT_AZURE_BASE_URL, | |
"AZURE_MAX_SPEAKERS": request.app.state.config.AUDIO_STT_AZURE_MAX_SPEAKERS, | |
}, | |
} | |
def load_speech_pipeline(request): | |
from transformers import pipeline | |
from datasets import load_dataset | |
if request.app.state.speech_synthesiser is None: | |
request.app.state.speech_synthesiser = pipeline( | |
"text-to-speech", "microsoft/speecht5_tts" | |
) | |
if request.app.state.speech_speaker_embeddings_dataset is None: | |
request.app.state.speech_speaker_embeddings_dataset = load_dataset( | |
"Matthijs/cmu-arctic-xvectors", split="validation" | |
) | |
async def speech(request: Request, user=Depends(get_verified_user)): | |
body = await request.body() | |
name = hashlib.sha256( | |
body | |
+ str(request.app.state.config.TTS_ENGINE).encode("utf-8") | |
+ str(request.app.state.config.TTS_MODEL).encode("utf-8") | |
).hexdigest() | |
file_path = SPEECH_CACHE_DIR.joinpath(f"{name}.mp3") | |
file_body_path = SPEECH_CACHE_DIR.joinpath(f"{name}.json") | |
# Check if the file already exists in the cache | |
if file_path.is_file(): | |
return FileResponse(file_path) | |
payload = None | |
try: | |
payload = json.loads(body.decode("utf-8")) | |
except Exception as e: | |
log.exception(e) | |
raise HTTPException(status_code=400, detail="Invalid JSON payload") | |
if request.app.state.config.TTS_ENGINE == "openai": | |
payload["model"] = request.app.state.config.TTS_MODEL | |
try: | |
timeout = aiohttp.ClientTimeout(total=AIOHTTP_CLIENT_TIMEOUT) | |
async with aiohttp.ClientSession( | |
timeout=timeout, trust_env=True | |
) as session: | |
async with session.post( | |
url=f"{request.app.state.config.TTS_OPENAI_API_BASE_URL}/audio/speech", | |
json=payload, | |
headers={ | |
"Content-Type": "application/json", | |
"Authorization": f"Bearer {request.app.state.config.TTS_OPENAI_API_KEY}", | |
**( | |
{ | |
"X-OpenWebUI-User-Name": user.name, | |
"X-OpenWebUI-User-Id": user.id, | |
"X-OpenWebUI-User-Email": user.email, | |
"X-OpenWebUI-User-Role": user.role, | |
} | |
if ENABLE_FORWARD_USER_INFO_HEADERS | |
else {} | |
), | |
}, | |
ssl=AIOHTTP_CLIENT_SESSION_SSL, | |
) as r: | |
r.raise_for_status() | |
async with aiofiles.open(file_path, "wb") as f: | |
await f.write(await r.read()) | |
async with aiofiles.open(file_body_path, "w") as f: | |
await f.write(json.dumps(payload)) | |
return FileResponse(file_path) | |
except Exception as e: | |
log.exception(e) | |
detail = None | |
try: | |
if r.status != 200: | |
res = await r.json() | |
if "error" in res: | |
detail = f"External: {res['error'].get('message', '')}" | |
except Exception: | |
detail = f"External: {e}" | |
raise HTTPException( | |
status_code=getattr(r, "status", 500) if r else 500, | |
detail=detail if detail else "Open WebUI: Server Connection Error", | |
) | |
elif request.app.state.config.TTS_ENGINE == "elevenlabs": | |
voice_id = payload.get("voice", "") | |
if voice_id not in get_available_voices(request): | |
raise HTTPException( | |
status_code=400, | |
detail="Invalid voice id", | |
) | |
try: | |
timeout = aiohttp.ClientTimeout(total=AIOHTTP_CLIENT_TIMEOUT) | |
async with aiohttp.ClientSession( | |
timeout=timeout, trust_env=True | |
) as session: | |
async with session.post( | |
f"https://api.elevenlabs.io/v1/text-to-speech/{voice_id}", | |
json={ | |
"text": payload["input"], | |
"model_id": request.app.state.config.TTS_MODEL, | |
"voice_settings": {"stability": 0.5, "similarity_boost": 0.5}, | |
}, | |
headers={ | |
"Accept": "audio/mpeg", | |
"Content-Type": "application/json", | |
"xi-api-key": request.app.state.config.TTS_API_KEY, | |
}, | |
ssl=AIOHTTP_CLIENT_SESSION_SSL, | |
) as r: | |
r.raise_for_status() | |
async with aiofiles.open(file_path, "wb") as f: | |
await f.write(await r.read()) | |
async with aiofiles.open(file_body_path, "w") as f: | |
await f.write(json.dumps(payload)) | |
return FileResponse(file_path) | |
except Exception as e: | |
log.exception(e) | |
detail = None | |
try: | |
if r.status != 200: | |
res = await r.json() | |
if "error" in res: | |
detail = f"External: {res['error'].get('message', '')}" | |
except Exception: | |
detail = f"External: {e}" | |
raise HTTPException( | |
status_code=getattr(r, "status", 500) if r else 500, | |
detail=detail if detail else "Open WebUI: Server Connection Error", | |
) | |
elif request.app.state.config.TTS_ENGINE == "azure": | |
try: | |
payload = json.loads(body.decode("utf-8")) | |
except Exception as e: | |
log.exception(e) | |
raise HTTPException(status_code=400, detail="Invalid JSON payload") | |
region = request.app.state.config.TTS_AZURE_SPEECH_REGION or "eastus" | |
base_url = request.app.state.config.TTS_AZURE_SPEECH_BASE_URL | |
language = request.app.state.config.TTS_VOICE | |
locale = "-".join(request.app.state.config.TTS_VOICE.split("-")[:1]) | |
output_format = request.app.state.config.TTS_AZURE_SPEECH_OUTPUT_FORMAT | |
try: | |
data = f"""<speak version="1.0" xmlns="http://www.w3.org/2001/10/synthesis" xml:lang="{locale}"> | |
<voice name="{language}">{payload["input"]}</voice> | |
</speak>""" | |
timeout = aiohttp.ClientTimeout(total=AIOHTTP_CLIENT_TIMEOUT) | |
async with aiohttp.ClientSession( | |
timeout=timeout, trust_env=True | |
) as session: | |
async with session.post( | |
(base_url or f"https://{region}.tts.speech.microsoft.com") | |
+ "/cognitiveservices/v1", | |
headers={ | |
"Ocp-Apim-Subscription-Key": request.app.state.config.TTS_API_KEY, | |
"Content-Type": "application/ssml+xml", | |
"X-Microsoft-OutputFormat": output_format, | |
}, | |
data=data, | |
ssl=AIOHTTP_CLIENT_SESSION_SSL, | |
) as r: | |
r.raise_for_status() | |
async with aiofiles.open(file_path, "wb") as f: | |
await f.write(await r.read()) | |
async with aiofiles.open(file_body_path, "w") as f: | |
await f.write(json.dumps(payload)) | |
return FileResponse(file_path) | |
except Exception as e: | |
log.exception(e) | |
detail = None | |
try: | |
if r.status != 200: | |
res = await r.json() | |
if "error" in res: | |
detail = f"External: {res['error'].get('message', '')}" | |
except Exception: | |
detail = f"External: {e}" | |
raise HTTPException( | |
status_code=getattr(r, "status", 500) if r else 500, | |
detail=detail if detail else "Open WebUI: Server Connection Error", | |
) | |
elif request.app.state.config.TTS_ENGINE == "transformers": | |
payload = None | |
try: | |
payload = json.loads(body.decode("utf-8")) | |
except Exception as e: | |
log.exception(e) | |
raise HTTPException(status_code=400, detail="Invalid JSON payload") | |
import torch | |
import soundfile as sf | |
load_speech_pipeline(request) | |
embeddings_dataset = request.app.state.speech_speaker_embeddings_dataset | |
speaker_index = 6799 | |
try: | |
speaker_index = embeddings_dataset["filename"].index( | |
request.app.state.config.TTS_MODEL | |
) | |
except Exception: | |
pass | |
speaker_embedding = torch.tensor( | |
embeddings_dataset[speaker_index]["xvector"] | |
).unsqueeze(0) | |
speech = request.app.state.speech_synthesiser( | |
payload["input"], | |
forward_params={"speaker_embeddings": speaker_embedding}, | |
) | |
sf.write(file_path, speech["audio"], samplerate=speech["sampling_rate"]) | |
async with aiofiles.open(file_body_path, "w") as f: | |
await f.write(json.dumps(payload)) | |
return FileResponse(file_path) | |
def transcription_handler(request, file_path): | |
filename = os.path.basename(file_path) | |
file_dir = os.path.dirname(file_path) | |
id = filename.split(".")[0] | |
if request.app.state.config.STT_ENGINE == "": | |
if request.app.state.faster_whisper_model is None: | |
request.app.state.faster_whisper_model = set_faster_whisper_model( | |
request.app.state.config.WHISPER_MODEL | |
) | |
model = request.app.state.faster_whisper_model | |
segments, info = model.transcribe( | |
file_path, | |
beam_size=5, | |
vad_filter=request.app.state.config.WHISPER_VAD_FILTER, | |
language=WHISPER_LANGUAGE, | |
) | |
log.info( | |
"Detected language '%s' with probability %f" | |
% (info.language, info.language_probability) | |
) | |
transcript = "".join([segment.text for segment in list(segments)]) | |
data = {"text": transcript.strip()} | |
# save the transcript to a json file | |
transcript_file = f"{file_dir}/{id}.json" | |
with open(transcript_file, "w") as f: | |
json.dump(data, f) | |
log.debug(data) | |
return data | |
elif request.app.state.config.STT_ENGINE == "openai": | |
r = None | |
try: | |
r = requests.post( | |
url=f"{request.app.state.config.STT_OPENAI_API_BASE_URL}/audio/transcriptions", | |
headers={ | |
"Authorization": f"Bearer {request.app.state.config.STT_OPENAI_API_KEY}" | |
}, | |
files={"file": (filename, open(file_path, "rb"))}, | |
data={"model": request.app.state.config.STT_MODEL}, | |
) | |
r.raise_for_status() | |
data = r.json() | |
# save the transcript to a json file | |
transcript_file = f"{file_dir}/{id}.json" | |
with open(transcript_file, "w") as f: | |
json.dump(data, f) | |
return data | |
except Exception as e: | |
log.exception(e) | |
detail = None | |
if r is not None: | |
try: | |
res = r.json() | |
if "error" in res: | |
detail = f"External: {res['error'].get('message', '')}" | |
except Exception: | |
detail = f"External: {e}" | |
raise Exception(detail if detail else "Open WebUI: Server Connection Error") | |
elif request.app.state.config.STT_ENGINE == "deepgram": | |
try: | |
# Determine the MIME type of the file | |
mime, _ = mimetypes.guess_type(file_path) | |
if not mime: | |
mime = "audio/wav" # fallback to wav if undetectable | |
# Read the audio file | |
with open(file_path, "rb") as f: | |
file_data = f.read() | |
# Build headers and parameters | |
headers = { | |
"Authorization": f"Token {request.app.state.config.DEEPGRAM_API_KEY}", | |
"Content-Type": mime, | |
} | |
# Add model if specified | |
params = {} | |
if request.app.state.config.STT_MODEL: | |
params["model"] = request.app.state.config.STT_MODEL | |
# Make request to Deepgram API | |
r = requests.post( | |
"https://api.deepgram.com/v1/listen", | |
headers=headers, | |
params=params, | |
data=file_data, | |
) | |
r.raise_for_status() | |
response_data = r.json() | |
# Extract transcript from Deepgram response | |
try: | |
transcript = response_data["results"]["channels"][0]["alternatives"][ | |
0 | |
].get("transcript", "") | |
except (KeyError, IndexError) as e: | |
log.error(f"Malformed response from Deepgram: {str(e)}") | |
raise Exception( | |
"Failed to parse Deepgram response - unexpected response format" | |
) | |
data = {"text": transcript.strip()} | |
# Save transcript | |
transcript_file = f"{file_dir}/{id}.json" | |
with open(transcript_file, "w") as f: | |
json.dump(data, f) | |
return data | |
except Exception as e: | |
log.exception(e) | |
detail = None | |
if r is not None: | |
try: | |
res = r.json() | |
if "error" in res: | |
detail = f"External: {res['error'].get('message', '')}" | |
except Exception: | |
detail = f"External: {e}" | |
raise Exception(detail if detail else "Open WebUI: Server Connection Error") | |
elif request.app.state.config.STT_ENGINE == "azure": | |
# Check file exists and size | |
if not os.path.exists(file_path): | |
raise HTTPException(status_code=400, detail="Audio file not found") | |
# Check file size (Azure has a larger limit of 200MB) | |
file_size = os.path.getsize(file_path) | |
if file_size > AZURE_MAX_FILE_SIZE: | |
raise HTTPException( | |
status_code=400, | |
detail=f"File size exceeds Azure's limit of {AZURE_MAX_FILE_SIZE_MB}MB", | |
) | |
api_key = request.app.state.config.AUDIO_STT_AZURE_API_KEY | |
region = request.app.state.config.AUDIO_STT_AZURE_REGION or "eastus" | |
locales = request.app.state.config.AUDIO_STT_AZURE_LOCALES | |
base_url = request.app.state.config.AUDIO_STT_AZURE_BASE_URL | |
max_speakers = request.app.state.config.AUDIO_STT_AZURE_MAX_SPEAKERS or 3 | |
# IF NO LOCALES, USE DEFAULTS | |
if len(locales) < 2: | |
locales = [ | |
"en-US", | |
"es-ES", | |
"es-MX", | |
"fr-FR", | |
"hi-IN", | |
"it-IT", | |
"de-DE", | |
"en-GB", | |
"en-IN", | |
"ja-JP", | |
"ko-KR", | |
"pt-BR", | |
"zh-CN", | |
] | |
locales = ",".join(locales) | |
if not api_key or not region: | |
raise HTTPException( | |
status_code=400, | |
detail="Azure API key is required for Azure STT", | |
) | |
r = None | |
try: | |
# Prepare the request | |
data = { | |
"definition": json.dumps( | |
{ | |
"locales": locales.split(","), | |
"diarization": {"maxSpeakers": max_speakers, "enabled": True}, | |
} | |
if locales | |
else {} | |
) | |
} | |
url = ( | |
base_url or f"https://{region}.api.cognitive.microsoft.com" | |
) + "/speechtotext/transcriptions:transcribe?api-version=2024-11-15" | |
# Use context manager to ensure file is properly closed | |
with open(file_path, "rb") as audio_file: | |
r = requests.post( | |
url=url, | |
files={"audio": audio_file}, | |
data=data, | |
headers={ | |
"Ocp-Apim-Subscription-Key": api_key, | |
}, | |
) | |
r.raise_for_status() | |
response = r.json() | |
# Extract transcript from response | |
if not response.get("combinedPhrases"): | |
raise ValueError("No transcription found in response") | |
# Get the full transcript from combinedPhrases | |
transcript = response["combinedPhrases"][0].get("text", "").strip() | |
if not transcript: | |
raise ValueError("Empty transcript in response") | |
data = {"text": transcript} | |
# Save transcript to json file (consistent with other providers) | |
transcript_file = f"{file_dir}/{id}.json" | |
with open(transcript_file, "w") as f: | |
json.dump(data, f) | |
log.debug(data) | |
return data | |
except (KeyError, IndexError, ValueError) as e: | |
log.exception("Error parsing Azure response") | |
raise HTTPException( | |
status_code=500, | |
detail=f"Failed to parse Azure response: {str(e)}", | |
) | |
except requests.exceptions.RequestException as e: | |
log.exception(e) | |
detail = None | |
try: | |
if r is not None and r.status_code != 200: | |
res = r.json() | |
if "error" in res: | |
detail = f"External: {res['error'].get('message', '')}" | |
except Exception: | |
detail = f"External: {e}" | |
raise HTTPException( | |
status_code=getattr(r, "status_code", 500) if r else 500, | |
detail=detail if detail else "Open WebUI: Server Connection Error", | |
) | |
def transcribe(request: Request, file_path): | |
log.info(f"transcribe: {file_path}") | |
if is_audio_conversion_required(file_path): | |
file_path = convert_audio_to_mp3(file_path) | |
try: | |
file_path = compress_audio(file_path) | |
except Exception as e: | |
log.exception(e) | |
# Always produce a list of chunk paths (could be one entry if small) | |
try: | |
chunk_paths = split_audio(file_path, MAX_FILE_SIZE) | |
print(f"Chunk paths: {chunk_paths}") | |
except Exception as e: | |
log.exception(e) | |
raise HTTPException( | |
status_code=status.HTTP_400_BAD_REQUEST, | |
detail=ERROR_MESSAGES.DEFAULT(e), | |
) | |
results = [] | |
try: | |
with ThreadPoolExecutor() as executor: | |
# Submit tasks for each chunk_path | |
futures = [ | |
executor.submit(transcription_handler, request, chunk_path) | |
for chunk_path in chunk_paths | |
] | |
# Gather results as they complete | |
for future in futures: | |
try: | |
results.append(future.result()) | |
except Exception as transcribe_exc: | |
log.exception(f"Error transcribing chunk: {transcribe_exc}") | |
raise HTTPException( | |
status_code=status.HTTP_500_INTERNAL_SERVER_ERROR, | |
detail="Error during transcription.", | |
) | |
finally: | |
# Clean up only the temporary chunks, never the original file | |
for chunk_path in chunk_paths: | |
if chunk_path != file_path and os.path.isfile(chunk_path): | |
try: | |
os.remove(chunk_path) | |
except Exception: | |
pass | |
return { | |
"text": " ".join([result["text"] for result in results]), | |
} | |
def compress_audio(file_path): | |
if os.path.getsize(file_path) > MAX_FILE_SIZE: | |
id = os.path.splitext(os.path.basename(file_path))[ | |
0 | |
] # Handles names with multiple dots | |
file_dir = os.path.dirname(file_path) | |
audio = AudioSegment.from_file(file_path) | |
audio = audio.set_frame_rate(16000).set_channels(1) # Compress audio | |
compressed_path = os.path.join(file_dir, f"{id}_compressed.mp3") | |
audio.export(compressed_path, format="mp3", bitrate="32k") | |
# log.debug(f"Compressed audio to {compressed_path}") # Uncomment if log is defined | |
return compressed_path | |
else: | |
return file_path | |
def split_audio(file_path, max_bytes, format="mp3", bitrate="32k"): | |
""" | |
Splits audio into chunks not exceeding max_bytes. | |
Returns a list of chunk file paths. If audio fits, returns list with original path. | |
""" | |
file_size = os.path.getsize(file_path) | |
if file_size <= max_bytes: | |
return [file_path] # Nothing to split | |
audio = AudioSegment.from_file(file_path) | |
duration_ms = len(audio) | |
orig_size = file_size | |
approx_chunk_ms = max(int(duration_ms * (max_bytes / orig_size)) - 1000, 1000) | |
chunks = [] | |
start = 0 | |
i = 0 | |
base, _ = os.path.splitext(file_path) | |
while start < duration_ms: | |
end = min(start + approx_chunk_ms, duration_ms) | |
chunk = audio[start:end] | |
chunk_path = f"{base}_chunk_{i}.{format}" | |
chunk.export(chunk_path, format=format, bitrate=bitrate) | |
# Reduce chunk duration if still too large | |
while os.path.getsize(chunk_path) > max_bytes and (end - start) > 5000: | |
end = start + ((end - start) // 2) | |
chunk = audio[start:end] | |
chunk.export(chunk_path, format=format, bitrate=bitrate) | |
if os.path.getsize(chunk_path) > max_bytes: | |
os.remove(chunk_path) | |
raise Exception("Audio chunk cannot be reduced below max file size.") | |
chunks.append(chunk_path) | |
start = end | |
i += 1 | |
return chunks | |
def transcription( | |
request: Request, | |
file: UploadFile = File(...), | |
user=Depends(get_verified_user), | |
): | |
log.info(f"file.content_type: {file.content_type}") | |
SUPPORTED_CONTENT_TYPES = {"video/webm"} # Extend if you add more video types! | |
if not ( | |
file.content_type.startswith("audio/") | |
or file.content_type in SUPPORTED_CONTENT_TYPES | |
): | |
raise HTTPException( | |
status_code=status.HTTP_400_BAD_REQUEST, | |
detail=ERROR_MESSAGES.FILE_NOT_SUPPORTED, | |
) | |
try: | |
ext = file.filename.split(".")[-1] | |
id = uuid.uuid4() | |
filename = f"{id}.{ext}" | |
contents = file.file.read() | |
file_dir = f"{CACHE_DIR}/audio/transcriptions" | |
os.makedirs(file_dir, exist_ok=True) | |
file_path = f"{file_dir}/{filename}" | |
with open(file_path, "wb") as f: | |
f.write(contents) | |
try: | |
result = transcribe(request, file_path) | |
return { | |
**result, | |
"filename": os.path.basename(file_path), | |
} | |
except Exception as e: | |
log.exception(e) | |
raise HTTPException( | |
status_code=status.HTTP_400_BAD_REQUEST, | |
detail=ERROR_MESSAGES.DEFAULT(e), | |
) | |
except Exception as e: | |
log.exception(e) | |
raise HTTPException( | |
status_code=status.HTTP_400_BAD_REQUEST, | |
detail=ERROR_MESSAGES.DEFAULT(e), | |
) | |
def get_available_models(request: Request) -> list[dict]: | |
available_models = [] | |
if request.app.state.config.TTS_ENGINE == "openai": | |
# Use custom endpoint if not using the official OpenAI API URL | |
if not request.app.state.config.TTS_OPENAI_API_BASE_URL.startswith( | |
"https://api.openai.com" | |
): | |
try: | |
response = requests.get( | |
f"{request.app.state.config.TTS_OPENAI_API_BASE_URL}/audio/models" | |
) | |
response.raise_for_status() | |
data = response.json() | |
available_models = data.get("models", []) | |
except Exception as e: | |
log.error(f"Error fetching models from custom endpoint: {str(e)}") | |
available_models = [{"id": "tts-1"}, {"id": "tts-1-hd"}] | |
else: | |
available_models = [{"id": "tts-1"}, {"id": "tts-1-hd"}] | |
elif request.app.state.config.TTS_ENGINE == "elevenlabs": | |
try: | |
response = requests.get( | |
"https://api.elevenlabs.io/v1/models", | |
headers={ | |
"xi-api-key": request.app.state.config.TTS_API_KEY, | |
"Content-Type": "application/json", | |
}, | |
timeout=5, | |
) | |
response.raise_for_status() | |
models = response.json() | |
available_models = [ | |
{"name": model["name"], "id": model["model_id"]} for model in models | |
] | |
except requests.RequestException as e: | |
log.error(f"Error fetching voices: {str(e)}") | |
return available_models | |
async def get_models(request: Request, user=Depends(get_verified_user)): | |
return {"models": get_available_models(request)} | |
def get_available_voices(request) -> dict: | |
"""Returns {voice_id: voice_name} dict""" | |
available_voices = {} | |
if request.app.state.config.TTS_ENGINE == "openai": | |
# Use custom endpoint if not using the official OpenAI API URL | |
if not request.app.state.config.TTS_OPENAI_API_BASE_URL.startswith( | |
"https://api.openai.com" | |
): | |
try: | |
response = requests.get( | |
f"{request.app.state.config.TTS_OPENAI_API_BASE_URL}/audio/voices" | |
) | |
response.raise_for_status() | |
data = response.json() | |
voices_list = data.get("voices", []) | |
available_voices = {voice["id"]: voice["name"] for voice in voices_list} | |
except Exception as e: | |
log.error(f"Error fetching voices from custom endpoint: {str(e)}") | |
available_voices = { | |
"alloy": "alloy", | |
"echo": "echo", | |
"fable": "fable", | |
"onyx": "onyx", | |
"nova": "nova", | |
"shimmer": "shimmer", | |
} | |
else: | |
available_voices = { | |
"alloy": "alloy", | |
"echo": "echo", | |
"fable": "fable", | |
"onyx": "onyx", | |
"nova": "nova", | |
"shimmer": "shimmer", | |
} | |
elif request.app.state.config.TTS_ENGINE == "elevenlabs": | |
try: | |
available_voices = get_elevenlabs_voices( | |
api_key=request.app.state.config.TTS_API_KEY | |
) | |
except Exception: | |
# Avoided @lru_cache with exception | |
pass | |
elif request.app.state.config.TTS_ENGINE == "azure": | |
try: | |
region = request.app.state.config.TTS_AZURE_SPEECH_REGION | |
base_url = request.app.state.config.TTS_AZURE_SPEECH_BASE_URL | |
url = ( | |
base_url or f"https://{region}.tts.speech.microsoft.com" | |
) + "/cognitiveservices/voices/list" | |
headers = { | |
"Ocp-Apim-Subscription-Key": request.app.state.config.TTS_API_KEY | |
} | |
response = requests.get(url, headers=headers) | |
response.raise_for_status() | |
voices = response.json() | |
for voice in voices: | |
available_voices[voice["ShortName"]] = ( | |
f"{voice['DisplayName']} ({voice['ShortName']})" | |
) | |
except requests.RequestException as e: | |
log.error(f"Error fetching voices: {str(e)}") | |
return available_voices | |
def get_elevenlabs_voices(api_key: str) -> dict: | |
""" | |
Note, set the following in your .env file to use Elevenlabs: | |
AUDIO_TTS_ENGINE=elevenlabs | |
AUDIO_TTS_API_KEY=sk_... # Your Elevenlabs API key | |
AUDIO_TTS_VOICE=EXAVITQu4vr4xnSDxMaL # From https://api.elevenlabs.io/v1/voices | |
AUDIO_TTS_MODEL=eleven_multilingual_v2 | |
""" | |
try: | |
# TODO: Add retries | |
response = requests.get( | |
"https://api.elevenlabs.io/v1/voices", | |
headers={ | |
"xi-api-key": api_key, | |
"Content-Type": "application/json", | |
}, | |
) | |
response.raise_for_status() | |
voices_data = response.json() | |
voices = {} | |
for voice in voices_data.get("voices", []): | |
voices[voice["voice_id"]] = voice["name"] | |
except requests.RequestException as e: | |
# Avoid @lru_cache with exception | |
log.error(f"Error fetching voices: {str(e)}") | |
raise RuntimeError(f"Error fetching voices: {str(e)}") | |
return voices | |
async def get_voices(request: Request, user=Depends(get_verified_user)): | |
return { | |
"voices": [ | |
{"id": k, "name": v} for k, v in get_available_voices(request).items() | |
] | |
} | |