import json import time import os from pathlib import Path import torch import torchaudio from TTS.tts.configs.xtts_config import XttsConfig from TTS.tts.models.xtts import Xtts ########################## #### Webserver Imports#### ########################## from fastapi import ( FastAPI, Form, Request, Response, Depends, ) from fastapi.responses import JSONResponse, HTMLResponse, RedirectResponse, FileResponse from fastapi.templating import Jinja2Templates from fastapi.staticfiles import StaticFiles from jinja2 import Template from contextlib import asynccontextmanager ########################### #### STARTUP VARIABLES #### ########################### # STARTUP VARIABLE - Create "this_dir" variable as the current script directory this_dir = Path(__file__).parent.resolve() # STARTUP VARIABLE - Set "device" to cuda if exists, otherwise cpu device = "cuda" if torch.cuda.is_available() else "cpu" # STARTUP VARIABLE - Import languges file for Gradio to be able to display them in the interface with open(this_dir / "languages.json", encoding="utf8") as f: languages = json.load(f) ################################################################# #### LOAD PARAMS FROM confignew.json - REQUIRED FOR BRANDING #### ################################################################# # Load config file and get settings def load_config(file_path): with open(file_path, "r") as configfile_path: configfile_data = json.load(configfile_path) return configfile_data # Define the path to the confignew.json file configfile_path = this_dir / "confignew.json" # Load confignew.json and assign it to a different variable (config_data) params = load_config(configfile_path) # check someone hasnt enabled lowvram on a system thats not cuda enabled params["low_vram"] = "false" if not torch.cuda.is_available() else params["low_vram"] # Define the path to the JSON file config_file_path = this_dir / "modeldownload.json" ############################################# #### LOAD PARAMS FROM MODELDOWNLOAD.JSON #### ############################################ # This is used only in the instance that someone has changed their model path # Define the path to the JSON file modeldownload_config_file_path = this_dir / "modeldownload.json" # Check if the JSON file exists if modeldownload_config_file_path.exists(): with open(modeldownload_config_file_path, "r") as modeldownload_config_file: modeldownload_settings = json.load(modeldownload_config_file) # Extract settings from the loaded JSON modeldownload_base_path = Path(modeldownload_settings.get("base_path", "")) modeldownload_model_path = Path(modeldownload_settings.get("model_path", "")) else: # Default settings if the JSON file doesn't exist or is empty print( f"[{params['branding']}Startup] \033[91mWarning\033[0m modeldownload.config is missing so please re-download it and save it in the alltalk_tts main folder." ) ######################## #### STARTUP CHECKS #### ######################## try: from TTS.api import TTS from TTS.utils.synthesizer import Synthesizer except ModuleNotFoundError: logger.error( f"[{params['branding']}Startup] \033[91mWarning\033[0m Could not find the TTS module. Make sure to install the requirements for the alltalk_tts extension." f"[{params['branding']}Startup] \033[91mWarning\033[0m Linux / Mac:\npip install -r extensions/alltalk_tts/requirements.txt\n" f"[{params['branding']}Startup] \033[91mWarning\033[0m Windows:\npip install -r extensions\\alltalk_tts\\requirements.txt\n" f"[{params['branding']}Startup] \033[91mWarning\033[0m If you used the one-click installer, paste the command above in the terminal window launched after running the cmd_ script. On Windows, that's cmd_windows.bat." ) raise # DEEPSPEED Import - Check for DeepSpeed and import it if it exists try: import deepspeed deepspeed_installed = True print(f"[{params['branding']}Startup] DeepSpeed \033[93mDetected\033[0m") print( f"[{params['branding']}Startup] Activate DeepSpeed in {params['branding']} settings" ) except ImportError: deepspeed_installed = False print( f"[{params['branding']}Startup] DeepSpeed \033[93mNot Detected\033[0m. See https://github.com/microsoft/DeepSpeed" ) @asynccontextmanager async def startup_shutdown(no_actual_value_it_demanded_something_be_here): await setup() yield # Shutdown logic # Create FastAPI app with lifespan app = FastAPI(lifespan=startup_shutdown) ##################################### #### MODEL LOADING AND UNLOADING #### ##################################### # MODEL LOADERS Picker For API TTS, API Local, XTTSv2 Local async def setup(): global device # Set a timer to calculate load times generate_start_time = time.time() # Record the start time of loading the model # Start loading the correct model as set by "tts_method_api_tts", "tts_method_api_local" or "tts_method_xtts_local" being True/False if params["tts_method_api_tts"]: print( f"[{params['branding']}Model] \033[94mAPI TTS Loading\033[0m {params['tts_model_name']} \033[94minto\033[93m", device, "\033[0m", ) model = await api_load_model() elif params["tts_method_api_local"]: print( f"[{params['branding']}Model] \033[94mAPI Local Loading\033[0m {modeldownload_model_path} \033[94minto\033[93m", device, "\033[0m", ) model = await api_manual_load_model() elif params["tts_method_xtts_local"]: print( f"[{params['branding']}Model] \033[94mXTTSv2 Local Loading\033[0m {modeldownload_model_path} \033[94minto\033[93m", device, "\033[0m", ) model = await xtts_manual_load_model() # Create an end timer for calculating load times generate_end_time = time.time() # Calculate start time minus end time generate_elapsed_time = generate_end_time - generate_start_time # Print out the result of the load time print( f"[{params['branding']}Model] \033[94mModel Loaded in \033[93m{generate_elapsed_time:.2f} seconds.\033[0m" ) # Set "tts_model_loaded" to true params["tts_model_loaded"] = True # Set the output path for wav files output_directory = this_dir / params["output_folder_wav_standalone"] output_directory.mkdir(parents=True, exist_ok=True) #Path(f'this_folder/outputs/').mkdir(parents=True, exist_ok=True) # MODEL LOADER For "API TTS" async def api_load_model(): global model model = TTS(params["tts_model_name"]).to(device) return model # MODEL LOADER For "API Local" async def api_manual_load_model(): global model # check to see if a custom path has been set in modeldownload.json and use that path to load the model if so if str(modeldownload_base_path) == "models": model = TTS( model_path=this_dir / "models" / modeldownload_model_path, config_path=this_dir / "models" / modeldownload_model_path / "config.json", ).to(device) else: print( f"[{params['branding']}Model] \033[94mInfo\033[0m Loading your custom model set in \033[93mmodeldownload.json\033[0m:", modeldownload_base_path / modeldownload_model_path, ) model = TTS( model_path=modeldownload_base_path / modeldownload_model_path, config_path=modeldownload_base_path / modeldownload_model_path / "config.json", ).to(device) return model # MODEL LOADER For "XTTSv2 Local" async def xtts_manual_load_model(): global model config = XttsConfig() # check to see if a custom path has been set in modeldownload.json and use that path to load the model if so if str(modeldownload_base_path) == "models": config_path = this_dir / "models" / modeldownload_model_path / "config.json" vocab_path_dir = this_dir / "models" / modeldownload_model_path / "vocab.json" checkpoint_dir = this_dir / "models" / modeldownload_model_path else: print( f"[{params['branding']}Model] \033[94mInfo\033[0m Loading your custom model set in \033[93mmodeldownload.json\033[0m:", modeldownload_base_path / modeldownload_model_path, ) config_path = modeldownload_base_path / modeldownload_model_path / "config.json" vocab_path_dir = modeldownload_base_path / modeldownload_model_path / "vocab.json" checkpoint_dir = modeldownload_base_path / modeldownload_model_path config.load_json(str(config_path)) model = Xtts.init_from_config(config) model.load_checkpoint( config, checkpoint_dir=str(checkpoint_dir), vocab_path=str(vocab_path_dir), use_deepspeed=params["deepspeed_activate"], ) model.to(device) return model # MODEL UNLOADER async def unload_model(model): print(f"[{params['branding']}Model] \033[94mUnloading model \033[0m") del model if torch.cuda.is_available(): torch.cuda.empty_cache() params["tts_model_loaded"] = False return None # MODEL - Swap model based on Gradio selection API TTS, API Local, XTTSv2 Local async def handle_tts_method_change(tts_method): global model # Update the params dictionary based on the selected radio button print( f"[{params['branding']}Model] \033[94mChanging model \033[92m(Please wait 15 seconds)\033[0m" ) # Set other parameters to False if tts_method == "API TTS": params["tts_method_api_local"] = False params["tts_method_xtts_local"] = False params["tts_method_api_tts"] = True params["deepspeed_activate"] = False elif tts_method == "API Local": params["tts_method_api_tts"] = False params["tts_method_xtts_local"] = False params["tts_method_api_local"] = True params["deepspeed_activate"] = False elif tts_method == "XTTSv2 Local": params["tts_method_api_tts"] = False params["tts_method_api_local"] = False params["tts_method_xtts_local"] = True # Unload the current model model = await unload_model(model) # Load the correct model based on the updated params await setup() # MODEL WEBSERVER- API Swap Between Models @app.route("/api/reload", methods=["POST"]) async def reload(request: Request): tts_method = request.query_params.get("tts_method") if tts_method not in ["API TTS", "API Local", "XTTSv2 Local"]: return {"status": "error", "message": "Invalid TTS method specified"} await handle_tts_method_change(tts_method) return Response( content=json.dumps({"status": "model-success"}), media_type="application/json" ) ################## #### LOW VRAM #### ################## # LOW VRAM - MODEL MOVER VRAM(cuda)<>System RAM(cpu) for Low VRAM setting async def switch_device(): global model, device # Check if CUDA is available before performing GPU-related operations if torch.cuda.is_available(): if device == "cuda": device = "cpu" model.to(device) torch.cuda.empty_cache() else: device == "cpu" device = "cuda" model.to(device) @app.post("/api/lowvramsetting") async def set_low_vram(request: Request, new_low_vram_value: bool): global device try: if new_low_vram_value is None: raise ValueError("Missing 'low_vram' parameter") if params["low_vram"] == new_low_vram_value: return Response( content=json.dumps( { "status": "success", "message": f"[{params['branding']}Model] LowVRAM is already {'enabled' if new_low_vram_value else 'disabled'}.", } ) ) params["low_vram"] = new_low_vram_value if params["low_vram"]: await unload_model(model) if torch.cuda.is_available(): device = "cpu" print( f"[{params['branding']}Model] \033[94mChanging model \033[92m(Please wait 15 seconds)\033[0m" ) print( f"[{params['branding']}Model] \033[94mLowVRAM Enabled.\033[0m Model will move between \033[93mVRAM(cuda) <> System RAM(cpu)\033[0m" ) await setup() else: # Handle the case where CUDA is not available print( f"[{params['branding']}Model] \033[91mError:\033[0m Nvidia CUDA is not available on this system. Unable to use LowVRAM mode." ) params["low_vram"] = False else: await unload_model(model) if torch.cuda.is_available(): device = "cuda" print( f"[{params['branding']}Model] \033[94mChanging model \033[92m(Please wait 15 seconds)\033[0m" ) print( f"[{params['branding']}Model] \033[94mLowVRAM Disabled.\033[0m Model will stay in \033[93mVRAM(cuda)\033[0m" ) await setup() else: # Handle the case where CUDA is not available print( f"[{params['branding']}Model] \033[91mError:\033[0m Nvidia CUDA is not available on this system. Unable to use LowVRAM mode." ) params["low_vram"] = False return Response(content=json.dumps({"status": "lowvram-success"})) except Exception as e: return Response(content=json.dumps({"status": "error", "message": str(e)})) ################### #### DeepSpeed #### ################### # DEEPSPEED - Reload the model when DeepSpeed checkbox is enabled/disabled async def handle_deepspeed_change(value): global model if value: # DeepSpeed enabled print(f"[{params['branding']}Model] \033[93mDeepSpeed Activating\033[0m") print( f"[{params['branding']}Model] \033[94mChanging model \033[92m(DeepSpeed can take 30 seconds to activate)\033[0m" ) print( f"[{params['branding']}Model] \033[91mInformation\033[0m If you have not set CUDA_HOME path, DeepSpeed may fail to load/activate" ) print( f"[{params['branding']}Model] \033[91mInformation\033[0m DeepSpeed needs to find nvcc from the CUDA Toolkit. Please check your CUDA_HOME path is" ) print( f"[{params['branding']}Model] \033[91mInformation\033[0m pointing to the correct location and use 'set CUDA_HOME=putyoutpathhere' (Windows) or" ) print( f"[{params['branding']}Model] \033[91mInformation\033[0m 'export CUDA_HOME=putyoutpathhere' (Linux) within your Python Environment" ) model = await unload_model(model) params["tts_method_api_tts"] = False params["tts_method_api_local"] = False params["tts_method_xtts_local"] = True params["deepspeed_activate"] = True await setup() else: # DeepSpeed disabled print(f"[{params['branding']}Model] \033[93mDeepSpeed De-Activating\033[0m") print( f"[{params['branding']}Model] \033[94mChanging model \033[92m(Please wait 15 seconds)\033[0m" ) params["deepspeed_activate"] = False model = await unload_model(model) await setup() return value # Return new checkbox value # DEEPSPEED WEBSERVER- API Enable/Disable DeepSpeed @app.post("/api/deepspeed") async def deepspeed(request: Request, new_deepspeed_value: bool): try: if new_deepspeed_value is None: raise ValueError("Missing 'deepspeed' parameter") if params["deepspeed_activate"] == new_deepspeed_value: return Response( content=json.dumps( { "status": "success", "message": f"DeepSpeed is already {'enabled' if new_deepspeed_value else 'disabled'}.", } ) ) params["deepspeed_activate"] = new_deepspeed_value await handle_deepspeed_change(params["deepspeed_activate"]) return Response(content=json.dumps({"status": "deepspeed-success"})) except Exception as e: return Response(content=json.dumps({"status": "error", "message": str(e)})) ######################## #### TTS GENERATION #### ######################## # TTS VOICE GENERATION METHODS (called from voice_preview and output_modifer) async def generate_audio(text, voice, language, output_file): global model if params["low_vram"] and device == "cpu": await switch_device() generate_start_time = time.time() # Record the start time of generating TTS # XTTSv2 LOCAL Method if params["tts_method_xtts_local"]: print(f"[{params['branding']}TTSGen] {text}") gpt_cond_latent, speaker_embedding = model.get_conditioning_latents( audio_path=[f"{this_dir}/voices/{voice}"], gpt_cond_len=model.config.gpt_cond_len, max_ref_length=model.config.max_ref_len, sound_norm_refs=model.config.sound_norm_refs, ) out = model.inference( text, language, gpt_cond_latent=gpt_cond_latent, speaker_embedding=speaker_embedding, temperature=float(params["local_temperature"]), length_penalty=float(model.config.length_penalty), repetition_penalty=float(params["local_repetition_penalty"]), top_k=int(model.config.top_k), # Convert to int if necessary top_p=float(model.config.top_p), enable_text_splitting=True, ) torchaudio.save(output_file, torch.tensor(out["wav"]).unsqueeze(0), 24000) # API LOCAL Methods elif params["tts_method_api_local"]: # Set the correct output path (different from the if statement) print(f"[{params['branding']}TTSGen] Using API Local") model.tts_to_file( text=text, file_path=output_file, speaker_wav=[f"{this_dir}/voices/{voice}"], language=language, temperature=float(params["local_temperature"]), length_penalty=model.config.length_penalty, repetition_penalty=float(params["local_repetition_penalty"]), top_k=model.config.top_k, top_p=model.config.top_p, ) # API TTS elif params["tts_method_api_tts"]: print(f"[{params['branding']}TTSGen] Using API TTS") model.tts_to_file( text=text, file_path=output_file, speaker_wav=[f"{this_dir}/voices/{voice}"], language=language, ) # Print Generation time and settings generate_end_time = time.time() # Record the end time to generate TTS generate_elapsed_time = generate_end_time - generate_start_time print( f"[{params['branding']}TTSGen] \033[93m{generate_elapsed_time:.2f} seconds. \033[94mLowVRAM: \033[33m{params['low_vram']} \033[94mDeepSpeed: \033[33m{params['deepspeed_activate']}\033[0m" ) # Move model back to cpu system ram if needed. if params["low_vram"] and device == "cuda": await switch_device() return # TTS VOICE GENERATION METHODS - generate TTS API @app.route("/api/generate", methods=["POST"]) async def generate(request: Request): try: # Get parameters from JSON body data = await request.json() text = data["text"] voice = data["voice"] language = data["language"] output_file = data["output_file"] # Generation logic await generate_audio(text, voice, language, output_file) return JSONResponse( content={"status": "generate-success", "data": {"audio_path": output_file}} ) except Exception as e: return JSONResponse(content={"status": "error", "message": str(e)}) ################################################### #### POPULATE FILES LIST FROM VOICES DIRECTORY #### ################################################### # List files in the "voices" directory def list_files(directory): files = [ f for f in os.listdir(directory) if os.path.isfile(os.path.join(directory, f)) and f.endswith(".wav") ] return files ############################# #### JSON CONFIG UPDATER #### ############################# # Create an instance of Jinja2Templates for rendering HTML templates templates = Jinja2Templates(directory=this_dir / "templates") # Create a dependency to get the current JSON data def get_json_data(): with open(this_dir / "confignew.json", "r") as json_file: data = json.load(json_file) return data # Define an endpoint function @app.get("/settings") async def get_settings(request: Request): wav_files = list_files(this_dir / "voices") # Render the template with the current JSON data and list of WAV files return templates.TemplateResponse( "generate_form.html", { "request": request, "data": get_json_data(), "modeldownload_model_path": modeldownload_model_path, "wav_files": wav_files, }, ) # Define an endpoint to serve static files app.mount("/static", StaticFiles(directory=str(this_dir / "templates")), name="static") @app.post("/update-settings") async def update_settings( request: Request, activate: bool = Form(...), autoplay: bool = Form(...), deepspeed_activate: bool = Form(...), delete_output_wavs: str = Form(...), ip_address: str = Form(...), language: str = Form(...), local_temperature: str = Form(...), local_repetition_penalty: str = Form(...), low_vram: bool = Form(...), tts_model_loaded: bool = Form(...), tts_model_name: str = Form(...), narrator_enabled: bool = Form(...), narrator_voice: str = Form(...), output_folder_wav: str = Form(...), port_number: str = Form(...), remove_trailing_dots: bool = Form(...), show_text: bool = Form(...), tts_method: str = Form(...), voice: str = Form(...), data: dict = Depends(get_json_data), ): # Update the settings based on the form values data["activate"] = activate data["autoplay"] = autoplay data["deepspeed_activate"] = deepspeed_activate data["delete_output_wavs"] = delete_output_wavs data["ip_address"] = ip_address data["language"] = language data["local_temperature"] = local_temperature data["local_repetition_penalty"] = local_repetition_penalty data["low_vram"] = low_vram data["tts_model_loaded"] = tts_model_loaded data["tts_model_name"] = tts_model_name data["narrator_enabled"] = narrator_enabled data["narrator_voice"] = narrator_voice data["output_folder_wav"] = output_folder_wav data["port_number"] = port_number data["remove_trailing_dots"] = remove_trailing_dots data["show_text"] = show_text data["tts_method_api_local"] = tts_method == "api_local" data["tts_method_api_tts"] = tts_method == "api_tts" data["tts_method_xtts_local"] = tts_method == "xtts_local" data["voice"] = voice # Save the updated settings back to the JSON file with open(this_dir / "confignew.json", "w") as json_file: json.dump(data, json_file) # Redirect to the settings page to display the updated settings return RedirectResponse(url="/settings", status_code=303) ################################## #### SETTINGS PAGE DEMO VOICE #### ################################## # Submit demo post here @app.post("/tts-demo-request", response_class=JSONResponse) async def tts_demo_request(request: Request, text: str = Form(...), voice: str = Form(...), language: str = Form(...), output_file: str = Form(...)): try: output_file_path = this_dir / "outputs" / output_file await generate_audio(text, voice, language, output_file_path) return JSONResponse(content={"output_file_path": str(output_file)}, status_code=200) except Exception as e: return JSONResponse(content={"error": "An error occurred"}, status_code=500) # Gives web access to the output files @app.get("/audio/{filename}") async def get_audio(filename: str): audio_path = this_dir / "outputs" / filename return FileResponse(audio_path) ######################## #### GENERATION API #### ######################## import html import re import numpy as np import soundfile as sf from typing import Union, Dict from pydantic import BaseModel, ValidationError, Field class Request(BaseModel): # Define the structure of the 'Request' class if needed pass class JSONInput(BaseModel): text_input: str = Field(..., max_length=1000, description="text_input needs to be 1000 characters or less.") text_filtering: str = Field(..., pattern="^(none|standard|html)$", description="text_filtering needs to be 'none', 'standard' or 'html'.") character_voice_gen: str = Field(..., pattern="^.*\.wav$", description="character_voice_gen needs to be the name of a wav file e.g. mysample.wav.") narrator_enabled: bool = Field(..., description="narrator_enabled needs to be true or false.") narrator_voice_gen: str = Field(..., pattern="^.*\.wav$", description="narrator_voice_gen needs to be the name of a wav file e.g. mysample.wav.") text_not_inside: str = Field(..., pattern="^(character|narrator)$", description="text_not_inside needs to be 'character' or 'narrator'.") language: str = Field(..., pattern="^(ar|zh-cn|cs|nl|en|fr|de|hu|it|ja|ko|pl|pt|ru|es|tr)$", description="language needs to be one of the following ar|zh-cn|cs|nl|en|fr|de|hu|it|ja|ko|pl|pt|ru|es|tr.") output_file_name: str = Field(..., pattern="^[a-zA-Z0-9_]+$", description="output_file_name needs to be the name without any special characters or file extension e.g. 'filename'") output_file_timestamp: bool = Field(..., description="output_file_timestamp needs to be true or false.") autoplay: bool = Field(..., description="autoplay needs to be a true or false value.") autoplay_volume: float = Field(..., ge=0.1, le=1.0, description="autoplay_volume needs to be from 0.1 to 1.0") @classmethod def validate_autoplay_volume(cls, value): if not (0.1 <= value <= 1.0): raise ValueError("Autoplay volume must be between 0.1 and 1.0") return value class TTSGenerator: @staticmethod def validate_json_input(json_data: Union[Dict, str]) -> Union[None, str]: try: if isinstance(json_data, str): json_data = json.loads(json_data) JSONInput(**json_data) return None # JSON is valid except ValidationError as e: return str(e) def standard_filtering(text_input): text_output = (text_input .replace("***", "") .replace("**", "") .replace("*", "") .replace("\n\n", "\n") .replace("'", "'") ) return text_output def narrator_filtering(narrator_text_input): processed_string = re.sub(r'\.\*\n\*', '. ', narrator_text_input) processed_string = ( processed_string .replace("\n", " ") .replace('"', '"<') .replace('"', '"<') ) processed_string = processed_string.replace('"<. *', '"< *"') processed_string = processed_string.replace('< *"', '< *') processed_string = processed_string.replace('. *', '< *') text_output = html.unescape(processed_string) return text_output def combine(output_file_timestamp, output_file_name, audio_files): audio = np.array([]) sample_rate = None try: for audio_file in audio_files: audio_data, current_sample_rate = sf.read(audio_file) if audio.size == 0: audio = audio_data sample_rate = current_sample_rate elif sample_rate == current_sample_rate: audio = np.concatenate((audio, audio_data)) else: raise ValueError("Sample rates of input files are not consistent.") except Exception as e: # Handle exceptions (e.g., file not found, invalid audio format) return None, None if output_file_timestamp: timestamp = int(time.time()) output_file_path = os.path.join(this_dir / "outputs" / f'{output_file_name}_{timestamp}_combined.wav') output_file_url = f'http://{params["ip_address"]}:{params["port_number"]}/audio/{output_file_name}_{timestamp}_combined.wav' else: output_file_path = os.path.join(this_dir / "outputs" / f'{output_file_name}_combined.wav') output_file_url = f'http://{params["ip_address"]}:{params["port_number"]}/audio/{output_file_name}.wav' try: sf.write(output_file_path, audio, samplerate=sample_rate) # Clean up unnecessary files for audio_file in audio_files: os.remove(audio_file) except Exception as e: # Handle exceptions (e.g., failed to write output file) return None, None return output_file_path, output_file_url # Generation API (separate from text-generation-webui) @app.post("/api/tts-generate", response_class=JSONResponse) async def tts_generate( text_input: str = Form(...), text_filtering: str = Form(...), character_voice_gen: str = Form(...), narrator_enabled: bool = Form(...), narrator_voice_gen: str = Form(...), text_not_inside: str = Form(...), language: str = Form(...), output_file_name: str = Form(...), output_file_timestamp: bool = Form(...), autoplay: bool = Form(...), autoplay_volume: float = Form(...), ): try: #print(f"text_filtering: {text_filtering}") #print(f"narrator_enabled: {narrator_enabled}") #print(f"text_not_inside: {text_not_inside}") #print(f"output_file_timestamp: {output_file_timestamp}") json_input_data = { "text_input": text_input, "text_filtering": text_filtering, "character_voice_gen": character_voice_gen, "narrator_enabled": narrator_enabled, "narrator_voice_gen": narrator_voice_gen, "text_not_inside": text_not_inside, "language": language, "output_file_name": output_file_name, "output_file_timestamp": output_file_timestamp, "autoplay": autoplay, "autoplay_volume": autoplay_volume, } JSONresult = TTSGenerator.validate_json_input(json_input_data) if JSONresult is None: pass else: return JSONResponse(content={"error": JSONresult}, status_code=400) if narrator_enabled: print("ORIGINAL UNTOUCHED STRING IS:", text_input,"\n") if text_filtering in ["standard", "none"]: cleaned_string = ( text_input .replace('"', '"<') .replace('*', '<*') ) print("STANDARD FILTERING IS NOW:", cleaned_string,"\n") #parts = re.split(r'\.(?
AllTalk is a web interface, based around the Coqui TTS voice cloning/speech generation system. To generate TTS, you can use the provided gradio interface or interact with the server using JSON/CURL commands.
Note: When loading up a new character in Text generation webUI it may look like nothing is happening for 20-30 seconds. Its actually processing the introduction section (greeting message) of the text and once that is completed, it will appear. You can see the activity occuring in the console window. Refreshing the page multiple times will try force the TTS engine to keep re-generating the text, so please just wait and check the console if needed.
Note: Ensure that your RP character card has asterisks around anything for the narration and double quotes around anything spoken. There is a complication ONLY with the greeting card so, ensuring it has the correct use of quotes and asterisks will help make sure that the greeting card sounds correct. I will aim to address this issue in a future update. In Text-generation-webUI parameters menu > character tab > greeting make sure that anything in there that is the narrator is in asterisks and anything spoken is in double quotes, then hit the save (💾) button.
AllTalk
Github link here
Update instructions link here
Help and issues link here
Text generation webUI
Web interface link here
Documentation link here
http://{{ params["ip_address"] }}:{{ params["port_number"] }}http://{{ params["ip_address"] }}:{{ params["port_number"] }}/readyVoice samples are stored in /alltalk_tts/voices/ and should be named using the following format name.wav
Voice outputs are stored in /alltalk_tts/outputs/
You can configure automatic maintenence deletion of old wav files by setting Del WAV's older than in the settings above.
When Disabled your output wav files will be left untouched. When set to a setting 1 Day or greater, your output wav files older than that time period will be automatically deleted on start-up of AllTalk.
This extension will download the 2.0.2 model to /alltalk_tts/models/
This TTS engine will also download the latest available model and store it wherever your OS normally stores it (Windows/Linux/Mac).
To create a new voice sample you need to make a wav file that is 22050Hz, Mono, 16 bit and between 6 to 30 seconds long, though 8 to 10 seconds is usually good enough. The model can handle up to a 30 second samples, however Ive not noticed any improvement in voice output from a much longer clips.
You want to find a nice clear selection of audio, so lets say you wanted to clone your favourite celebrity. You may go looking for an interview where they are talking. Pay close attention to the audio you are listening to and trying to sample. Are there noises in the backgroud, hiss on the soundtrack, a low humm, some quiet music playing or something? The better quality the audio the better the final TTS result. Dont forget, the AI that processes the sounds can hear everything in your sample and it will use them in the voice its trying to recreate.
Try make your clip one of nice flowing speech, like the included example files. No big pauses, gaps or other sounds. Preferably a sample that the person you are trying to copy will show a little vocal range and emotion in their voice. Also, try to avoid a clip starting or ending with breathy sounds (breathing in/out etc).
So youve downloaded your favoutie celebrity interview off Youtube, from here you need to chop it down to 6 to 30 seconds in length and resample it.
If you need to clean it up, do audio processing, volume level changes etc, do this before downsampling.
Using the latest version of Audacity select/highlight your 6 to 30 second clip and:
Tracks > Resample to 22050Hz then
Tracks > Mix > Stereo to Mono then
File > Export Audio saving it as a WAV of 22050Hz.
Save your generated wav file in the /alltalk_tts/voices/ folder.
Its worth mentioning that using AI generated audio clips may introduce unwanted sounds as its already a copy/simulation of a voice.
The reasons can be that you:
Some samples just never seem to work correctly, so maybe try a different sample. Always remember though, this is an AI model attempting to re-create a voice, so you will never get a 100% match.
This only affects the Narrator function. Most AI models should be using asterisks or double quotes to differentiate between the Narrator or the Character, however, many models sometimes switch between using asterisks and double quotes or sometimes nothing at all for the text it outputs.
This leaves a bit of a mess because sometimes un-marked text is narration and sometimes its the character talking, leaving no clear way to know where to split sentences and what voice to use. Whilst there is no 100% solution at the moment many models lean more one way or the other as to what that unmarked text will be (character or narrator).
As such, the "Text not inside" function at least gives you the choice to set how you want the TTS engine to handle situations of un-marked text.

The Low VRAM option is a crucial feature designed to enhance performance under constrained (VRAM) conditions, as the TTS models require 2GB-3GB of VRAM to run effectively. This feature strategically manages the relocation of the Text-to-Speech (TTS) model between your system's Random Access Memory (RAM) and VRAM, moving it between the two on the fly. Obviously, this is very useful for people who have smaller graphics cards and will use all their VRAM to load in their LLM.
When you dont have enough VRAM free after loading your LLM model into your VRAM (Normal Mode example below), you can see that with so little working space, your GPU will have to swap in and out bits of the TTS model, which causes horrible slowdown.
Note: An Nvidia Graphics card is required for the LowVRAM option to work, as you will just be using system RAM otherwise.
The Low VRAM mode intelligently orchestrates the relocation of the entire TTS model and stores the TTS model in your system RAM. When the TTS engine requires VRAM for processing, the entire model seamlessly moves into VRAM, causing your LLM to unload/displace some layers, ensuring optimal performance of the TTS engine.
Post-TTS processing, the model moves back to system RAM, freeing up VRAM space for your Language Model (LLM) to load back in the missing layers. This adds about 1-2 seconds to both text generation by the LLM and the TTS engine.
By transferring the entire model between RAM and VRAM, the Low VRAM option avoids fragmentation, ensuring the TTS model remains cohesive and has all the working space it needs in your GPU, without having to just work on small bits of the TTS model at a time (which causes terrible slow down).
This creates a TTS generation performance Boost for Low VRAM Users and is particularly beneficial for users with less than 2GB of free VRAM after loading their LLM, delivering a substantial 5-10x improvement in TTS generation speed.

If you have a voice that the model doesnt quite reproduce correctly, or indeed you just want to improve the reproduced voice, then finetuning is a way to train your "XTTSv2 local" model (stored in /alltalk_tts/models/xxxxx/) on a specific voice. For this you will need:
Everything has been done to make this as simple as possible. At its simplest, you can literally just download a large chunk of audio from an interview, and tell the finetuning to strip through it, find spoken parts and build your dataset. You can literally click 4 buttons, then copy a few files and you are done. At it's more complicated end you will clean up the audio a little beforehand, but its still only 4x buttons and copying a few files.
I would suggest that if its in an interview format, you cut out the interviewer speaking in audacity or your chosen audio editing package. You dont have to worry about being perfect with your cuts, the finetuning Step 1 will go and find spoken audio and cut it out for you. Is there is music over the spoken parts, for best quality you would cut out those parts, though its not 100% necessary. As always, try to avoid bad quality audio with noises in it (humming sounds, hiss etc). You can try something like Audioenhancer to try clean up noisier audio. There is no need to down-sample any of the audio, all of that is handled for you. Just give the finetuning some good quality audio to work with.
As mentioned you must have a small portion of the Nvidia CUDA Toolkit 11.8 installed. Not higher or lower versions. Specifically 11.8. You do not have to uninstall any other versions, change any graphics drivers, reinstall torch or anything like that. There are instructions within the finetuning interface on doing this or you can also find them on this link here
Ensure you have followed the instructions on setting up the Nvidia CUDA Toolkit 11.8 here or the below procedure will fail.
The below instructions are also available online here
Close all other applications that are using your GPU/VRAM and copy your audio samples into:
/alltalk_tts/finetune/put-voice-samples-in-here/
In a command prompt/terminal window you need to move into your Text generation webUI folder:
cd text-generation-webui
Start the Text generation webUI Python environment for your OS:
cmd_windows.bat, ./cmd_linux.sh, cmd_macos.sh or cmd_wsl.bat
You can double check your search path environment still works correctly with nvcc --version. It should report back 11.8:
Cuda compilation tools, release 11.8.
Move into your extensions folder:
cd extensions
Move into the alltalk_tts folder:
cd alltalk_tts
Install the finetune requirements file: pip install -r requirements_finetune.txt
Type python finetune.py and it should start up.
Follow the on-screen instructions when the web interface starts up.
When you have finished finetuning, the final tab will tell you what to do with your files and how to move your newly trained model to the correct location on disk.
DeepSpeed provides a 2x-3x speed boost for Text-to-Speech and AI tasks. It's all about making AI and TTS happen faster and more efficiently.

DeepSpeed only works with the XTTSv2 Local model and will deactivate when other models are selected, even if the checkbox still shows as being selected.
Note: DeepSpeed/AllTalk may warn if the Nvidia Cuda Toolkit and CUDA_HOME environment variable isnt set correctly. On Linux you need CUDA_HOME configured correctly; on Windows, if you use the pre-built wheel its ok without.
Note: You do not need to set Text-generation-webUi's --deepspeed setting for AllTalk to be able to use DeepSpeed.
➡️DeepSpeed requires an Nvidia Graphics card!⬅️
➡️DeepSpeed requires an Nvidia Graphics card!⬅️
DeepSpeed v11.2 will work on the current default text-generation-webui Python 3.11 environment! You have 2x options for how to setup DeepSpeed on Windows. A quick way Option 1 and a long way Option 2.
Thanks to @S95Sedan They managed to get DeepSpeed 11.2 working on Windows via making some edits to the original Microsoft DeepSpeed v11.2 installation. The original post is here.
OPTION 1 - Pre-Compiled Wheel Deepspeed v11.2 (Python 3.11 and 3.10)
It's worth noting that all models and methods can and do sound different from one another. Many people complained about the quality of audio produced by the 2.0.3 model, so this extension will download the 2.0.2 model to your models folder and give you the choice to use 2.0.2 (API Local and XTTSv2 Local) or use the most current model 2.0.3 (API TTS). As/When a new model is released by Coqui it will be downloaded by the TTS service on startup and stored wherever the TTS service keeps new models for your operating system.
It is recommended not to modify these settings unless you fully comprehend their effects. A general overview is provided below for reference.
Changes to these settings won't take effect until you restart AllTalk/Text generation webUI.
These settings only affect API Local and XTTSv2 Local methods.
In the context of text-to-speech (TTS), the Repetition Penalty influences how the model handles the repetition of sounds, phonemes, or intonation patterns. Here's how it works:
The factory setting for repetition penalty is 10.0
Temperature influences the randomness of the generated speech. Here's how it affects the output:
The factory setting for temperature is 0.70
Factory settings should be fine for most people, however if you choose to experiment, setting extremely high or low values, especially without a good understanding of their effects, may lead to flat-sounding output or very strange-sounding output. It's advisable to experiment with adjustments incrementally and observe the impact on the generated speech to find a balance that suits your desired outcome.
AllTalk performs a variety of checks on startup and will warn out messages at the console should you need to do something such as update your TTS version.
A basic environment check to ensure everything should work e.g. is the model already downloaded, are the configuration files set correctly etc.
AllTalk will download the Xtts model (version 2.0.2) into your models folder. Many people didnt like the quality of the 2.0.3 model, however the latest model will be accessable on the API TTS setting (2.0.3 at the time of writing) so you have the best of both worlds.
Its possible to set a custom model for the API Local and XTTSv2 Local methods, or indeed point it at the same model that API TTS uses (wherever it is stored on your OS of choice).
Many people did not like the sound quality of the Coqui 2.0.3 model, and as such the AllTalk downloads the 2.0.2 model seperately from the 2.0.3 model that TTS service downloads and manages.
Typically the 2.0.2 model is stored in your /alltalk_tts/models folder and it is downloaded on first start-up of the AllTalk_tts extension. However, you may either want to use a custom model version of your choosing, or re-point AllTalk to a different path on your system, or even point it so that API Local and XTTSv2 Local both use the same model that API TTS is using.
If you do choose to change the location there are a couple of things to note.
To change the model path, there are at minimum 2x settings you need to alter in the modeldownload.json file, base_path and model_path.
You would edit the settings in the modeldownload.json file as follows (make a backup of your current file in case):
Windows path example: c:\\mystuff\\mydownloads\\myTTSmodel\\{files in here}
base_path would be "c:\\mystuff\\mydownloads"
model_path would be: "myTTSmodel"
Note: On Windows systems, you have to specify a double backslash \\ for each folder level in the path (as above)
Linux path example: /home/myaccount/myTTSmodel/{files in here}
base_path would be "/home/myaccount"
model_path would be: "TTSmodel"
Once you restart Alltalk, it will check this path for the files and output any details at the console.
When you are happy it's' working correctly, you are welcome to go delete the models folder stored at /alltalk_tts/models.
If you wish to change the files that the modeldownloader is pulling at startup, you can futher edit the modeldownload.json and change the https addresses within this files files_to_download section e.g.
"files_to_download": {
"LICENSE.txt": "https://huggingface.co/coqui/XTTS-v2/resolve/v2.0.2/LICENSE.txt?download=true",
"README.md": "https://huggingface.co/coqui/XTTS-v2/resolve/v2.0.2/README.md?download=true",
"config.json": "https://huggingface.co/coqui/XTTS-v2/resolve/v2.0.2/config.json?download=true",
"model.pth": "https://huggingface.co/coqui/XTTS-v2/resolve/v2.0.2/model.pth?download=true",
"vocab.json": "https://huggingface.co/coqui/XTTS-v2/resolve/v2.0.2/vocab.json?download=true"
}
confignew.json file:
| Key | Default Value | Explanation |
|---|---|---|
| "activate" | true | Sets activation state within Text-generation-webUI. |
| "autoplay" | true | Sets autoplay within Text generation webUI. |
| "branding" | "AllTalk" | Used to change the default name. Requires a space e.g. "Mybrand ". |
| "deepspeed_activate" | false | Sets DeepSpeed activation on startup. |
| "delete_output_wavs" | "Disabled" | Sets duration of outputs to delete. |
| "ip_address" | "127.0.0.1" | Sets default IP address. |
| "language" | "English" | Sets default language for Text-generation-webUI TTS. |
| "low_vram" | false | Sets default setting for LowVRAM mode. |
| "local_temperature" | "0.70" | Sets default model temp for API Local and XTTSv2 Local. |
| "local_repetition_penalty" | "10.0" | Sets default model repetition for API Local and XTTSv2 Local. |
| "tts_model_loaded" | true | AllTalk internal use only. Do not change. |
| "tts_model_name" | "tts_models/multilingual/multi-dataset/xtts_v2" | Sets default model that API TTS is looking for through the TTS service (separate to API Local and XTTSv2 Local). |
| "narrator_enabled" | true | Sets default narrator on/off in Text-generation-webUI TTS. |
| "narrator_voice" | "female_02.wav" | Sets default wav to use for narrator in Text-generation-webUI TTS. |
| "port_number" | "7851" | Sets default port number for AllTalk. |
| "output_folder_wav" | "extensions/alltalk_tts/outputs/" | Sets default output path Text-generation-webUI should use for finding outputs. |
| "output_folder_wav_standalone" | "outputs/" | Sets default output path in standalone mode. |
| "remove_trailing_dots" | false | Sets trailing dot removal pre-generating TTS. |
| "show_text" | true | Sets if text should be displayed below audio in Text-generation-webUI. |
| "tts_method_api_local" | false | Sets API Local as the default model/method for TTS. |
| "tts_method_api_tts" | false | Sets API TTS as the default model/method for TTS. |
| "tts_method_xtts_local" | true | Sets XTTSv2 Local as the default model/method for TTS. |
| "voice" | "female_01.wav" | Sets default voice for TTS. |
modeldownload.json file:
| Key | Value | Description | ||||||||||||
|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|
| "base_path" | "models" | Sets local model base path for API Local and XTTSv2 Local. | ||||||||||||
| "model_path" | "xttsv2_2.0.2" | Sets local model folder for API Local and XTTSv2 Local below the base path. | ||||||||||||
| "files_to_download" |
|
Sets the model files required to be downloaded into \base_path\model_path\ and where to download them from. |
The Text-to-Speech (TTS) Generation API allows you to generate speech from text input using various configuration options. This API supports both character and narrator voices, providing flexibility for creating dynamic and engaging audio content.
Standard TTS speech Example (standard text) generating a time-stamped file
curl -X POST "http://127.0.0.1:7851/api/tts-generate" -d "text_input=All of this is text spoken by the character. This is text not inside quotes, though that doesnt matter in the slightest" -d "text_filtering=standard" -d "character_voice_gen=female_01.wav" -d "narrator_enabled=false" -d "narrator_voice_gen=male_01.wav" -d "text_not_inside=character" -d "language=en" -d "output_file_name=myoutputfile" -d "output_file_timestamp=true" -d "autoplay=true" -d "autoplay_volume=0.8"
Narrator Example (standard text) generating a time-stamped file
curl -X POST "http://127.0.0.1:7851/api/tts-generate" -d "text_input=*This is text spoken by the narrator* \"This is text spoken by the character\". This is text not inside quotes." -d "text_filtering=standard" -d "character_voice_gen=female_01.wav" -d "narrator_enabled=true" -d "narrator_voice_gen=male_01.wav" -d "text_not_inside=character" -d "language=en" -d "output_file_name=myoutputfile" -d "output_file_timestamp=true" -d "autoplay=true" -d "autoplay_volume=0.8"
Note that if your text that needs to be generated contains double quotes you will need to escape them with \" (Please see the narrator example).
text_input: The text you want the TTS engine to produce. Use escaped double quotes for character speech and asterisks for narrator speech if using the narrator function. Example:
-d "text_input=*This is text spoken by the narrator* \"This is text spoken by the character\". This is text not inside quotes."
text_filtering: Filter for text. Options:
-d "text_filtering=none"
-d "text_filtering=standard"
-d "text_filtering=html"
Example:
character_voice_gen: The WAV file name for the character's voice.
-d "character_voice_gen=female_01.wav"
narrator_enabled: Enable or disable the narrator function. If true, minimum text filtering is set to standard. Anything between double quotes is considered the character's speech, and anything between asterisks is considered the narrator's speech.
-d "narrator_enabled=true"
-d "narrator_enabled=false"
narrator_voice_gen: The WAV file name for the narrator's voice.
-d "narrator_voice_gen=male_01.wav"
text_not_inside: Specify the handling of lines not inside double quotes or asterisks, for the narrator feature. Options:
-d "text_not_inside=character"
-d "text_not_inside=narrator"
language: Choose the language for TTS. Options:
-d "language=en"
output_file_name: The name of the output file (excluding the .wav extension).
-d "output_file_name=myoutputfile"
output_file_timestamp: Add a timestamp to the output file name. If true, each file will have a unique timestamp; otherwise, the same file name will be overwritten each time you generate TTS.
-d "output_file_timestamp=true"
-d "output_file_timestamp=false"
autoplay: Feature not yet available. Enable or disable autoplay. Still needs to be specified in the JSON request.
-d "autoplay=true"
-d "autoplay=false"
autoplay_volume: Feature not yet available. Set the autoplay volume. Should be between 0.1 and 1.0. Still needs to be specified in the JSON request.
-d "autoplay_volume=0.8"
The API returns a JSON object with the following properties:
Example JSON TTS Generation Response:
{"status": "generate-success", "output_file_path": "C:\\text-generation-webui\\extensions\\alltalk_tts\\outputs\\myoutputfile_1703149973.wav", "output_file_url": "http://127.0.0.1:7851/audio/myoutputfile_1703149973.wav"}
curl -X POST "http://127.0.0.1:7851/api/reload?tts_method=API%20Local"
curl -X POST "http://127.0.0.1:7851/api/reload?tts_method=API%20TTS"
curl -X POST "http://127.0.0.1:7851/api/reload?tts_method=XTTSv2%20Local"
Switch between the 3 models respectively.
JSON return {"status": "model-success"}
curl -X POST "http://127.0.0.1:7851/api/deepspeed?new_deepspeed_value=True"
Replace True with False to disable DeepSpeed mode.
JSON return {"status": "deepspeed-success"}
curl -X POST "http://127.0.0.1:7851/api/lowvramsetting?new_low_vram_value=True"
Replace True with False to disable Low VRAM mode.
JSON return {"status": "lowvram-success"}
Command line outputs are more verbose to assist in understanding backend processes and debugging.
Its possible during startup you can get a warning message such as:
[AllTalk Startup] Warning TTS Subprocess has NOT started up yet, Will keep trying for 80 seconds maximum
This is normal behavior if the subprocess is taking a while to start, however, if there is an issue starting the subprocess, you may see multiples of this message and an it will time out after 80 seconds, resulting in the TTS engine not starting. Its likely that you are not in the correct python environment or one that has a TTS engine inside, if this happens, though the system will output a warning about that ahead of this message
Typically the command line console will output any warning or error messages. If you need to reset your default configuation, the settings are all listed above in the configuration details.