""" Ray Serve deployment for Coqui TTS. Runs on: elminster (RTX 2070 8GB, CUDA) """ import base64 import io import os import time from typing import Any from ray import serve from ray_serve.mlflow_logger import InferenceLogger @serve.deployment(name="TTSDeployment", num_replicas=1) class TTSDeployment: def __init__(self): import torch from TTS.api import TTS self.model_name = os.environ.get("MODEL_NAME", "tts_models/en/ljspeech/tacotron2-DDC") # Detect device self.use_gpu = torch.cuda.is_available() print(f"Loading TTS model: {self.model_name}") print(f"Using GPU: {self.use_gpu}") self.tts = TTS(model_name=self.model_name, progress_bar=False) if self.use_gpu: self.tts = self.tts.to("cuda") print("TTS model loaded successfully") # MLflow metrics self._mlflow = InferenceLogger( experiment_name="ray-serve-tts", run_name=f"tts-{self.model_name.split('/')[-1]}", tags={"model.name": self.model_name, "model.framework": "coqui-tts", "gpu": str(self.use_gpu)}, flush_every=5, ) self._mlflow.initialize(params={"model_name": self.model_name, "use_gpu": str(self.use_gpu)}) async def __call__(self, request: dict[str, Any]) -> dict[str, Any]: """ Handle text-to-speech requests. Expected request format: { "text": "Text to synthesize", "speaker": "speaker_name", "language": "en", "speed": 1.0, "output_format": "wav", "return_base64": true } """ import numpy as np from scipy.io import wavfile _start = time.time() text = request.get("text", "") speaker = request.get("speaker") language = request.get("language") speed = request.get("speed", 1.0) output_format = request.get("output_format", "wav") return_base64 = request.get("return_base64", True) if not text: return {"error": "No text provided"} # Generate speech try: # TTS.tts returns a numpy array of audio samples wav = self.tts.tts( text=text, speaker=speaker, language=language, speed=speed, ) # Convert to numpy array if needed if not isinstance(wav, np.ndarray): wav = np.array(wav) # Normalize to int16 wav_int16 = (wav * 32767).astype(np.int16) # Get sample rate from model config sample_rate = ( self.tts.synthesizer.output_sample_rate if hasattr(self.tts, "synthesizer") else 22050 ) # Write to buffer buffer = io.BytesIO() wavfile.write(buffer, sample_rate, wav_int16) audio_bytes = buffer.getvalue() duration = len(wav) / sample_rate # Log to MLflow self._mlflow.log_request( latency_s=time.time() - _start, audio_duration_s=duration, text_chars=len(text), realtime_factor=(time.time() - _start) / duration if duration > 0 else 0, ) response = { "model": self.model_name, "sample_rate": sample_rate, "duration": duration, "format": output_format, } if return_base64: response["audio"] = base64.b64encode(audio_bytes).decode("utf-8") else: response["audio_bytes"] = audio_bytes return response except Exception as e: return { "error": str(e), "model": self.model_name, } def list_speakers(self) -> dict[str, Any]: """List available speakers for multi-speaker models.""" speakers = [] if hasattr(self.tts, "speakers") and self.tts.speakers: speakers = self.tts.speakers return { "model": self.model_name, "speakers": speakers, "is_multi_speaker": len(speakers) > 0, } app = TTSDeployment.bind()