| import io |
| import os |
| import threading |
| import time |
| from typing import Optional |
|
|
| import numpy as np |
| import soundfile as sf |
| import torch |
| import torchaudio |
| from huggingface_hub import login as hf_login |
| from loguru import logger |
| from snac import SNAC |
| from transformers import AutoModelForCausalLM, AutoTokenizer |
|
|
| from app.speaker_map import recommended_speed |
|
|
|
|
| TOKENISER_LENGTH = 128256 |
| END_OF_SPEECH_ID = TOKENISER_LENGTH + 2 |
| PAD_TOKEN_ID = TOKENISER_LENGTH + 7 |
| AUDIO_START_ID = TOKENISER_LENGTH + 10 |
| SAMPLE_RATE = 24000 |
|
|
| DEFAULTS = { |
| "temperature": float(os.getenv("TTS_TEMPERATURE", "0.4")), |
| "top_p": float(os.getenv("TTS_TOP_P", "0.9")), |
| "repetition_penalty": float(os.getenv("TTS_REPETITION_PENALTY", "1.05")), |
| "max_seq_length": int(os.getenv("TTS_MAX_SEQ_LENGTH", "2048")), |
| "max_words": int(os.getenv("TTS_MAX_WORDS", "50")), |
| "denoise": os.getenv("TTS_DENOISE", "false").lower() == "true", |
| } |
|
|
|
|
| class TTSRuntime: |
| def __init__(self) -> None: |
| self._model = None |
| self._tokenizer = None |
| self._snac = None |
| self._df_model = None |
| self._df_state = None |
| self._device = "cuda" if torch.cuda.is_available() else "cpu" |
| self._loaded = False |
| self._lock = threading.Lock() |
|
|
| @property |
| def is_ready(self) -> bool: |
| return self._loaded |
|
|
| @staticmethod |
| def _resolve_hf_token(cli_token: Optional[str]) -> Optional[str]: |
| if cli_token: |
| return cli_token |
| return ( |
| os.getenv("HF_TOKEN") |
| or os.getenv("HUGGINGFACE_TOKEN") |
| or os.getenv("HUGGING_FACE_HUB_TOKEN") |
| ) |
|
|
| def load(self, model_name: Optional[str] = None, hf_token: Optional[str] = None) -> None: |
| with self._lock: |
| if self._loaded: |
| return |
|
|
| |
| |
| os.environ["HF_HUB_ENABLE_HF_TRANSFER"] = "0" |
|
|
| model_name = model_name or os.getenv("MODEL_NAME", "Mevearth2/Quantized-Merged-TTS") |
| token = self._resolve_hf_token(hf_token) |
|
|
| if token: |
| try: |
| hf_login(token=token, add_to_git_credential=False) |
| logger.info("HF auth success") |
| except Exception as exc: |
| logger.warning(f"HF login warning: {exc}") |
|
|
| logger.info(f"Loading tokenizer/model from {model_name}") |
| self._tokenizer = AutoTokenizer.from_pretrained(model_name, token=token) |
| self._model = AutoModelForCausalLM.from_pretrained( |
| model_name, |
| token=token, |
| torch_dtype=torch.float16 if self._device == "cuda" else torch.float32, |
| ) |
| self._model.to(self._device) |
| self._model.eval() |
|
|
| pad_token = self._tokenizer.decode([PAD_TOKEN_ID]) |
| self._tokenizer.pad_token = pad_token |
| self._tokenizer.padding_side = "left" |
|
|
| try: |
| from unsloth import FastLanguageModel |
|
|
| FastLanguageModel.for_inference(self._model) |
| logger.info("Unsloth inference enabled") |
| except Exception: |
| logger.info("Unsloth not enabled") |
|
|
| logger.info("Loading SNAC decoder") |
| self._snac = SNAC.from_pretrained("hubertsiuzdak/snac_24khz") |
|
|
| if DEFAULTS["denoise"]: |
| try: |
| from df.enhance import init_df |
|
|
| self._df_model, self._df_state, _ = init_df() |
| logger.info("DeepFilter initialized") |
| except Exception as exc: |
| logger.warning(f"DeepFilter unavailable: {exc}") |
|
|
| self._loaded = True |
|
|
| @staticmethod |
| def _build_prompt(utterance: str, language: str, user_id: str) -> str: |
| return ( |
| "<custom_token_3><|begin_of_text|>" |
| f"{language}{user_id}: {utterance}" |
| "<|eot_id|><custom_token_4><custom_token_5><custom_token_1>" |
| ) |
|
|
| @staticmethod |
| def _extract_audio_ids(output_ids: torch.Tensor) -> list[int]: |
| raw_audio_ids = [tok.item() for tok in output_ids if tok.item() >= AUDIO_START_ID] |
| clean = [] |
| full_groups = len(raw_audio_ids) // 7 |
| for i in range(full_groups): |
| base = i * 7 |
| for j in range(7): |
| clean.append(raw_audio_ids[base + j] - AUDIO_START_ID) |
| return clean |
|
|
| @staticmethod |
| def _snac_tokens_to_codebooks(clean_audio_ids: list[int]): |
| codes = [[], [], []] |
| full_groups = len(clean_audio_ids) // 7 |
|
|
| for i in range(full_groups): |
| b = i * 7 |
| codes[0].append(clean_audio_ids[b + 0]) |
| codes[1].append(clean_audio_ids[b + 1] - 4096) |
| codes[2].append(clean_audio_ids[b + 2] - (2 * 4096)) |
| codes[2].append(clean_audio_ids[b + 3] - (3 * 4096)) |
| codes[1].append(clean_audio_ids[b + 4] - (4 * 4096)) |
| codes[2].append(clean_audio_ids[b + 5] - (5 * 4096)) |
| codes[2].append(clean_audio_ids[b + 6] - (6 * 4096)) |
|
|
| if len(codes[0]) == 0 or len(codes[1]) == 0 or len(codes[2]) == 0: |
| return None |
|
|
| return [ |
| torch.tensor(codes[0]).unsqueeze(0), |
| torch.tensor(codes[1]).unsqueeze(0), |
| torch.tensor(codes[2]).unsqueeze(0), |
| ] |
|
|
| @staticmethod |
| def _apply_speed(audio: np.ndarray, speed: float) -> np.ndarray: |
| if abs(speed - 1.0) <= 1e-4: |
| return audio |
| |
| if hasattr(torchaudio, "sox_effects") and hasattr(torchaudio.sox_effects, "apply_effects_tensor"): |
| audio_t = torch.from_numpy(audio).unsqueeze(0) |
| out_t, _ = torchaudio.sox_effects.apply_effects_tensor( |
| audio_t, |
| SAMPLE_RATE, |
| effects=[["tempo", f"{speed}"]], |
| ) |
| return out_t.squeeze(0).cpu().numpy() |
|
|
| |
| |
| in_len = int(audio.shape[0]) |
| out_len = max(1, int(round(in_len / speed))) |
| x_old = np.linspace(0.0, 1.0, num=in_len, dtype=np.float64) |
| x_new = np.linspace(0.0, 1.0, num=out_len, dtype=np.float64) |
| stretched = np.interp(x_new, x_old, audio.astype(np.float64)) |
| return stretched.astype(np.float32) |
|
|
| def _apply_denoise(self, audio: np.ndarray) -> np.ndarray: |
| if self._df_model is None or self._df_state is None: |
| return audio |
| try: |
| import librosa |
| from df.enhance import enhance |
|
|
| audio_48k = librosa.resample(audio, orig_sr=SAMPLE_RATE, target_sr=48000) |
| audio_48k_t = torch.from_numpy(audio_48k).unsqueeze(0) |
| denoised = enhance(self._df_model, self._df_state, audio_48k_t) |
| denoised_np = denoised.squeeze(0).cpu().numpy() |
| return librosa.resample(denoised_np, orig_sr=48000, target_sr=SAMPLE_RATE) |
| except Exception as exc: |
| logger.warning(f"Denoise failed: {exc}") |
| return audio |
|
|
| def synthesize_wav_bytes(self, utterance: str, language: str, user_id: str) -> tuple[bytes, int]: |
| if not self._loaded: |
| raise RuntimeError("Runtime is not loaded") |
|
|
| start = time.perf_counter() |
| safe_utterance = " ".join(utterance.split()[: DEFAULTS["max_words"]]) |
| prompt = self._build_prompt(safe_utterance, language, user_id) |
| inputs = self._tokenizer(prompt, add_special_tokens=False, return_tensors="pt") |
|
|
| input_ids = inputs.input_ids.to(self._device) |
| attention_mask = inputs.attention_mask.to(self._device) |
| max_new_tokens = max(32, DEFAULTS["max_seq_length"] - input_ids.shape[1]) |
|
|
| with torch.inference_mode(): |
| output = self._model.generate( |
| input_ids=input_ids, |
| attention_mask=attention_mask, |
| max_new_tokens=max_new_tokens, |
| do_sample=True, |
| temperature=DEFAULTS["temperature"], |
| top_p=DEFAULTS["top_p"], |
| repetition_penalty=DEFAULTS["repetition_penalty"], |
| eos_token_id=END_OF_SPEECH_ID, |
| ) |
|
|
| clean_audio_ids = self._extract_audio_ids(output[0]) |
| if not clean_audio_ids: |
| raise RuntimeError("No audio token IDs generated") |
|
|
| codes = self._snac_tokens_to_codebooks(clean_audio_ids) |
| if codes is None: |
| raise RuntimeError("Insufficient audio token IDs for SNAC decode") |
|
|
| with torch.inference_mode(): |
| audio = self._snac.decode(codes) |
|
|
| audio_np = audio.detach().squeeze().cpu().numpy().astype(np.float32) |
| audio_np = self._apply_speed(audio_np, recommended_speed(language, str(user_id))) |
| audio_np = self._apply_denoise(audio_np) |
|
|
| wav_buf = io.BytesIO() |
| sf.write(wav_buf, audio_np, SAMPLE_RATE, format="WAV") |
| wav_bytes = wav_buf.getvalue() |
| duration_ms = int((time.perf_counter() - start) * 1000) |
| return wav_bytes, duration_ms |
|
|