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