import argparse import gc import logging import numpy as np import queue import socket import struct import threading import traceback import wave from importlib.resources import files import torch import torchaudio from huggingface_hub import hf_hub_download from hydra.utils import get_class from omegaconf import OmegaConf from f5_tts.infer.utils_infer import ( chunk_text, preprocess_ref_audio_text, load_vocoder, load_model, infer_batch_process, ) logging.basicConfig(level=logging.INFO) logger = logging.getLogger(__name__) class AudioFileWriterThread(threading.Thread): """Threaded file writer to avoid blocking the TTS streaming process.""" def __init__(self, output_file, sampling_rate): super().__init__() self.output_file = output_file self.sampling_rate = sampling_rate self.queue = queue.Queue() self.stop_event = threading.Event() self.audio_data = [] def run(self): """Process queued audio data and write it to a file.""" logger.info("AudioFileWriterThread started.") with wave.open(self.output_file, "wb") as wf: wf.setnchannels(1) wf.setsampwidth(2) wf.setframerate(self.sampling_rate) while not self.stop_event.is_set() or not self.queue.empty(): try: chunk = self.queue.get(timeout=0.1) if chunk is not None: chunk = np.int16(chunk * 32767) self.audio_data.append(chunk) wf.writeframes(chunk.tobytes()) except queue.Empty: continue def add_chunk(self, chunk): """Add a new chunk to the queue.""" self.queue.put(chunk) def stop(self): """Stop writing and ensure all queued data is written.""" self.stop_event.set() self.join() logger.info("Audio writing completed.") class TTSStreamingProcessor: def __init__(self, model, ckpt_file, vocab_file, ref_audio, ref_text, device=None, dtype=torch.float32): self.device = device or ( "cuda" if torch.cuda.is_available() else "xpu" if torch.xpu.is_available() else "mps" if torch.backends.mps.is_available() else "cpu" ) model_cfg = OmegaConf.load(str(files("f5_tts").joinpath(f"configs/{model}.yaml"))) self.model_cls = get_class(f"f5_tts.model.{model_cfg.model.backbone}") self.model_arc = model_cfg.model.arch self.mel_spec_type = model_cfg.model.mel_spec.mel_spec_type self.sampling_rate = model_cfg.model.mel_spec.target_sample_rate self.model = self.load_ema_model(ckpt_file, vocab_file, dtype) self.vocoder = self.load_vocoder_model() self.update_reference(ref_audio, ref_text) self._warm_up() self.file_writer_thread = None self.first_package = True def load_ema_model(self, ckpt_file, vocab_file, dtype): return load_model( self.model_cls, self.model_arc, ckpt_path=ckpt_file, mel_spec_type=self.mel_spec_type, vocab_file=vocab_file, ode_method="euler", use_ema=True, device=self.device, ).to(self.device, dtype=dtype) def load_vocoder_model(self): return load_vocoder(vocoder_name=self.mel_spec_type, is_local=False, local_path=None, device=self.device) def update_reference(self, ref_audio, ref_text): self.ref_audio, self.ref_text = preprocess_ref_audio_text(ref_audio, ref_text) self.audio, self.sr = torchaudio.load(self.ref_audio) ref_audio_duration = self.audio.shape[-1] / self.sr ref_text_byte_len = len(self.ref_text.encode("utf-8")) self.max_chars = int(ref_text_byte_len / (ref_audio_duration) * (25 - ref_audio_duration)) self.few_chars = int(ref_text_byte_len / (ref_audio_duration) * (25 - ref_audio_duration) / 2) self.min_chars = int(ref_text_byte_len / (ref_audio_duration) * (25 - ref_audio_duration) / 4) def _warm_up(self): logger.info("Warming up the model...") gen_text = "Warm-up text for the model." for _ in infer_batch_process( (self.audio, self.sr), self.ref_text, [gen_text], self.model, self.vocoder, progress=None, device=self.device, streaming=True, ): pass logger.info("Warm-up completed.") def generate_stream(self, text, conn): text_batches = chunk_text(text, max_chars=self.max_chars) if self.first_package: text_batches = chunk_text(text_batches[0], max_chars=self.few_chars) + text_batches[1:] text_batches = chunk_text(text_batches[0], max_chars=self.min_chars) + text_batches[1:] self.first_package = False audio_stream = infer_batch_process( (self.audio, self.sr), self.ref_text, text_batches, self.model, self.vocoder, progress=None, device=self.device, streaming=True, chunk_size=2048, ) # Reset the file writer thread if self.file_writer_thread is not None: self.file_writer_thread.stop() self.file_writer_thread = AudioFileWriterThread("output.wav", self.sampling_rate) self.file_writer_thread.start() for audio_chunk, _ in audio_stream: if len(audio_chunk) > 0: logger.info(f"Generated audio chunk of size: {len(audio_chunk)}") # Send audio chunk via socket conn.sendall(struct.pack(f"{len(audio_chunk)}f", *audio_chunk)) # Write to file asynchronously self.file_writer_thread.add_chunk(audio_chunk) logger.info("Finished sending audio stream.") conn.sendall(b"END") # Send end signal # Ensure all audio data is written before exiting self.file_writer_thread.stop() def handle_client(conn, processor): try: with conn: conn.setsockopt(socket.IPPROTO_TCP, socket.TCP_NODELAY, 1) while True: data = conn.recv(1024) if not data: processor.first_package = True break data_str = data.decode("utf-8").strip() logger.info(f"Received text: {data_str}") try: processor.generate_stream(data_str, conn) except Exception as inner_e: logger.error(f"Error during processing: {inner_e}") traceback.print_exc() break except Exception as e: logger.error(f"Error handling client: {e}") traceback.print_exc() def start_server(host, port, processor): with socket.socket(socket.AF_INET, socket.SOCK_STREAM) as s: s.bind((host, port)) s.listen() logger.info(f"Server started on {host}:{port}") while True: conn, addr = s.accept() logger.info(f"Connected by {addr}") handle_client(conn, processor) if __name__ == "__main__": parser = argparse.ArgumentParser() parser.add_argument("--host", default="0.0.0.0") parser.add_argument("--port", default=9998) parser.add_argument( "--model", default="F5TTS_v1_Base", help="The model name, e.g. F5TTS_v1_Base", ) parser.add_argument( "--ckpt_file", default=str(hf_hub_download(repo_id="SWivid/F5-TTS", filename="F5TTS_v1_Base/model_1250000.safetensors")), help="Path to the model checkpoint file", ) parser.add_argument( "--vocab_file", default="", help="Path to the vocab file if customized", ) parser.add_argument( "--ref_audio", default=str(files("f5_tts").joinpath("infer/examples/basic/basic_ref_en.wav")), help="Reference audio to provide model with speaker characteristics", ) parser.add_argument( "--ref_text", default="", help="Reference audio subtitle, leave empty to auto-transcribe", ) parser.add_argument("--device", default=None, help="Device to run the model on") parser.add_argument("--dtype", default=torch.float32, help="Data type to use for model inference") args = parser.parse_args() try: # Initialize the processor with the model and vocoder processor = TTSStreamingProcessor( model=args.model, ckpt_file=args.ckpt_file, vocab_file=args.vocab_file, ref_audio=args.ref_audio, ref_text=args.ref_text, device=args.device, dtype=args.dtype, ) # Start the server start_server(args.host, args.port, processor) except KeyboardInterrupt: gc.collect()