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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 | |
| 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_TIMEOUT, | |
| ENV, | |
| SRC_LOG_LEVELS, | |
| DEVICE_TYPE, | |
| ENABLE_FORWARD_USER_INFO_HEADERS, | |
| ) | |
| router = APIRouter() | |
| # Constants | |
| MAX_FILE_SIZE_MB = 25 | |
| 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 get_audio_convert_format(file_path): | |
| """Check if the given file needs to be converted to a different format.""" | |
| if not os.path.isfile(file_path): | |
| log.error(f"File not found: {file_path}") | |
| return False | |
| try: | |
| info = mediainfo(file_path) | |
| if ( | |
| info.get("codec_name") == "aac" | |
| and info.get("codec_type") == "audio" | |
| and info.get("codec_tag_string") == "mp4a" | |
| ): | |
| return "mp4" | |
| elif info.get("format_name") == "ogg": | |
| return "ogg" | |
| except Exception as e: | |
| log.error(f"Error getting audio format: {e}") | |
| return False | |
| return None | |
| def convert_audio_to_wav(file_path, output_path, conversion_type): | |
| """Convert MP4/OGG audio file to WAV format.""" | |
| audio = AudioSegment.from_file(file_path, format=conversion_type) | |
| audio.export(output_path, format="wav") | |
| log.info(f"Converted {file_path} to {output_path}") | |
| 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 {} | |
| ), | |
| }, | |
| ) 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, | |
| }, | |
| ) 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, | |
| ) 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 transcribe(request: Request, file_path): | |
| log.info(f"transcribe: {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": | |
| convert_format = get_audio_convert_format(file_path) | |
| if convert_format: | |
| ext = convert_format.split(".")[-1] | |
| os.rename(file_path, file_path.replace(".{ext}", f".{convert_format}")) | |
| # Convert unsupported audio file to WAV format | |
| convert_audio_to_wav( | |
| file_path.replace(".{ext}", f".{convert_format}"), | |
| file_path, | |
| convert_format, | |
| ) | |
| 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 compress_audio(file_path): | |
| if os.path.getsize(file_path) > MAX_FILE_SIZE: | |
| 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 = f"{file_dir}/{id}_compressed.opus" | |
| audio.export(compressed_path, format="opus", bitrate="32k") | |
| log.debug(f"Compressed audio to {compressed_path}") | |
| if ( | |
| os.path.getsize(compressed_path) > MAX_FILE_SIZE | |
| ): # Still larger than MAX_FILE_SIZE after compression | |
| raise Exception(ERROR_MESSAGES.FILE_TOO_LARGE(size=f"{MAX_FILE_SIZE_MB}MB")) | |
| return compressed_path | |
| else: | |
| return file_path | |
| def transcription( | |
| request: Request, | |
| file: UploadFile = File(...), | |
| user=Depends(get_verified_user), | |
| ): | |
| log.info(f"file.content_type: {file.content_type}") | |
| supported_filetypes = ( | |
| "audio/mpeg", | |
| "audio/wav", | |
| "audio/ogg", | |
| "audio/x-m4a", | |
| "audio/webm", | |
| ) | |
| if not file.content_type.startswith(supported_filetypes): | |
| 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: | |
| try: | |
| file_path = compress_audio(file_path) | |
| except Exception as e: | |
| log.exception(e) | |
| raise HTTPException( | |
| status_code=status.HTTP_400_BAD_REQUEST, | |
| detail=ERROR_MESSAGES.DEFAULT(e), | |
| ) | |
| data = transcribe(request, file_path) | |
| file_path = file_path.split("/")[-1] | |
| return {**data, "filename": 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() | |
| ] | |
| } | |