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Update app.py
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app.py
CHANGED
@@ -1,618 +1,66 @@
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import os
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import queue
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from huggingface_hub import snapshot_download
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import hydra
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import numpy as np
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import wave
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import io
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import pyrootutils
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import gc
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# Download if not exists
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os.makedirs("checkpoints", exist_ok=True)
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#snapshot_download(repo_id="fishaudio/fish-speech-1.4", local_dir="./checkpoints/fish-speech-1.4")
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print("All checkpoints downloaded")
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import html
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import os
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import threading
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from argparse import ArgumentParser
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from pathlib import Path
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from functools import partial
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import gradio as gr
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import librosa
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import torch
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import
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# torch.cuda.is_available = lambda: False
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# torchaudio.set_audio_backend("soundfile")
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from loguru import logger
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from transformers import AutoTokenizer
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from tools.llama.generate import launch_thread_safe_queue
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from tools.vqgan.inference import load_model as load_vqgan_model
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from fish_speech.text.chn_text_norm.text import Text as ChnNormedText
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from tools.api import decode_vq_tokens, encode_reference
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from tools.auto_rerank import batch_asr, calculate_wer, is_chinese, load_model
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from tools.llama.generate import (
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GenerateRequest,
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GenerateResponse,
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WrappedGenerateResponse,
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launch_thread_safe_queue,
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)
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from tools.vqgan.inference import load_model as load_decoder_model
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# Make einx happy
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os.environ["EINX_FILTER_TRACEBACK"] = "false"
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HEADER_MD = """# Fish Speech
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## The demo in this space is version 1.4, Please check [Fish Audio](https://fish.audio) for the best model.
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## 该 Demo 为 Fish Speech 1.4 版本, 请在 [Fish Audio](https://fish.audio) 体验最新 DEMO.
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A text-to-speech model based on VQ-GAN and Llama developed by [Fish Audio](https://fish.audio).
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由 [Fish Audio](https://fish.audio) 研发的基于 VQ-GAN 和 Llama 的多语种语音合成.
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You can find the source code [here](https://github.com/fishaudio/fish-speech) and models [here](https://huggingface.co/fishaudio/fish-speech-1.4).
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你可以在 [这里](https://github.com/fishaudio/fish-speech) 找到源代码和 [这里](https://huggingface.co/fishaudio/fish-speech-1.4) 找到模型.
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Related code and weights are released under CC BY-NC-SA 4.0 License.
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相关代码,权重使用 CC BY-NC-SA 4.0 许可证发布.
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We are not responsible for any misuse of the model, please consider your local laws and regulations before using it.
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我们不对模型的任何滥用负责,请在使用之前考虑您当地的法律法规.
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The model running in this WebUI is Fish Speech V1.4 Medium.
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在此 WebUI 中运行的模型是 Fish Speech V1.4 Medium.
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"""
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TEXTBOX_PLACEHOLDER = """Put your text here. 在此处输入文本."""
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try:
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import spaces
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GPU_DECORATOR = spaces.GPU
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except ImportError:
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def GPU_DECORATOR(func):
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def wrapper(*args, **kwargs):
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return func(*args, **kwargs)
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return wrapper
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def build_html_error_message(error):
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return f"""
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<div style="color: red;
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font-weight: bold;">
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{html.escape(error)}
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</div>
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"""
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@GPU_DECORATOR
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@torch.inference_mode()
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def inference(
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text,
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enable_reference_audio,
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reference_audio,
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reference_text,
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max_new_tokens,
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chunk_length,
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top_p,
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repetition_penalty,
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temperature,
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streaming=False
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):
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if args.max_gradio_length > 0 and len(text) > args.max_gradio_length:
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return (
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None,
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None,
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"Text is too long, please keep it under {} characters.".format(
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args.max_gradio_length
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),
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)
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# Parse reference audio aka prompt
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prompt_tokens = encode_reference(
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decoder_model=decoder_model,
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reference_audio=reference_audio,
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enable_reference_audio=enable_reference_audio,
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)
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# LLAMA Inference
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request = dict(
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device=decoder_model.device,
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max_new_tokens=max_new_tokens,
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text=text,
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top_p=top_p,
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repetition_penalty=repetition_penalty,
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temperature=temperature,
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compile=args.compile,
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iterative_prompt=chunk_length > 0,
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chunk_length=chunk_length,
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max_length=2048,
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prompt_tokens=prompt_tokens if enable_reference_audio else None,
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prompt_text=reference_text if enable_reference_audio else None,
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)
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response_queue = queue.Queue()
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llama_queue.put(
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GenerateRequest(
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request=request,
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response_queue=response_queue,
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)
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)
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segments = []
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while True:
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result: WrappedGenerateResponse = response_queue.get()
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if result.status == "error":
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return None, None, build_html_error_message(result.response)
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result: GenerateResponse = result.response
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if result.action == "next":
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break
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with torch.autocast(
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device_type=(
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"cpu"
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if decoder_model.device.type == "mps"
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else decoder_model.device.type
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),
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dtype=args.precision,
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):
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fake_audios = decode_vq_tokens(
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decoder_model=decoder_model,
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codes=result.codes,
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)
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fake_audios = fake_audios.float().cpu().numpy()
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segments.append(fake_audios)
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if len(segments) == 0:
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return (
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None,
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None,
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build_html_error_message(
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"No audio generated, please check the input text."
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),
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)
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# Return the final audio
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audio = np.concatenate(segments, axis=0)
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return None, (decoder_model.spec_transform.sample_rate, audio), None
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if torch.cpu.is_available():
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torch.cpu.empty_cache()
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gc.collect()
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def inference_with_auto_rerank(
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text,
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enable_reference_audio,
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reference_audio,
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reference_text,
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max_new_tokens,
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chunk_length,
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top_p,
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repetition_penalty,
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temperature,
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use_auto_rerank,
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streaming=False,
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):
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max_attempts = 2 if use_auto_rerank else 1
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best_wer = float("inf")
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best_audio = None
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best_sample_rate = None
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for attempt in range(max_attempts):
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_, (sample_rate, audio), message = inference(
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text,
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enable_reference_audio,
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reference_audio,
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reference_text,
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max_new_tokens,
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chunk_length,
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top_p,
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repetition_penalty,
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temperature,
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streaming=False,
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)
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if audio is None:
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return None, None, message
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if not use_auto_rerank:
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return None, (sample_rate, audio), None
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asr_result = batch_asr(asr_model, [audio], sample_rate)[0]
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wer = calculate_wer(text, asr_result["text"])
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if wer <= 0.3 and not asr_result["huge_gap"]:
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return None, (sample_rate, audio), None
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if wer < best_wer:
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best_wer = wer
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best_audio = audio
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best_sample_rate = sample_rate
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if attempt == max_attempts - 1:
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break
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return None, (best_sample_rate, best_audio), None
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n_audios = 4
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global_audio_list = []
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global_error_list = []
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def inference_wrapper(
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text,
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enable_reference_audio,
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reference_audio,
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reference_text,
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max_new_tokens,
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chunk_length,
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top_p,
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repetition_penalty,
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temperature,
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batch_infer_num,
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if_load_asr_model,
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):
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audios = []
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errors = []
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for _ in range(batch_infer_num):
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result = inference_with_auto_rerank(
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text,
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enable_reference_audio,
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reference_audio,
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reference_text,
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max_new_tokens,
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chunk_length,
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top_p,
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repetition_penalty,
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temperature,
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if_load_asr_model,
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)
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_, audio_data, error_message = result
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audios.append(
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gr.Audio(value=audio_data if audio_data else None, visible=True),
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)
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errors.append(
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gr.HTML(value=error_message if error_message else None, visible=True),
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)
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for _ in range(batch_infer_num, n_audios):
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audios.append(
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gr.Audio(value=None, visible=False),
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)
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errors.append(
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gr.HTML(value=None, visible=False),
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)
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return None, *audios, *errors
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def wav_chunk_header(sample_rate=44100, bit_depth=16, channels=1):
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buffer = io.BytesIO()
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with wave.open(buffer, "wb") as wav_file:
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wav_file.setnchannels(channels)
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wav_file.setsampwidth(bit_depth // 8)
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wav_file.setframerate(sample_rate)
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wav_header_bytes = buffer.getvalue()
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buffer.close()
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return wav_header_bytes
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def normalize_text(user_input, use_normalization):
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if use_normalization:
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return ChnNormedText(raw_text=user_input).normalize()
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else:
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return user_input
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asr_model = None
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def change_if_load_asr_model(if_load):
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global asr_model
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if if_load:
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gr.Warning("Loading faster whisper model...")
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if asr_model is None:
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asr_model = load_model()
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return gr.Checkbox(label="Unload faster whisper model", value=if_load)
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if if_load is False:
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gr.Warning("Unloading faster whisper model...")
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del asr_model
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asr_model = None
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if torch.cpu.is_available():
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torch.cpu.empty_cache()
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gc.collect()
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return gr.Checkbox(label="Load faster whisper model", value=if_load)
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def change_if_auto_label(if_load, if_auto_label, enable_ref, ref_audio, ref_text):
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if if_load and asr_model is not None:
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if (
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if_auto_label
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and enable_ref
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and ref_audio is not None
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and ref_text.strip() == ""
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):
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data, sample_rate = librosa.load(ref_audio)
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res = batch_asr(asr_model, [data], sample_rate)[0]
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ref_text = res["text"]
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else:
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gr.Warning("Whisper model not loaded!")
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return gr.Textbox(value=ref_text)
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def build_app():
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with gr.Blocks(theme=gr.themes.Base()) as app:
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gr.Markdown(HEADER_MD)
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# Use light theme by default
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app.load(
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None,
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None,
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js="() => {const params = new URLSearchParams(window.location.search);if (!params.has('__theme')) {params.set('__theme', '%s');window.location.search = params.toString();}}"
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% args.theme,
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)
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# Inference
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with gr.Row():
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with gr.Column(scale=3):
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text = gr.Textbox(
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label="Input Text", placeholder=TEXTBOX_PLACEHOLDER, lines=10
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)
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refined_text = gr.Textbox(
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label="Realtime Transform Text",
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placeholder=
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"Normalization Result Preview (Currently Only Chinese)",
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lines=5,
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interactive=False,
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)
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with gr.Row():
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if_refine_text = gr.Checkbox(
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label="Text Normalization (ZH)",
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value=False,
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scale=1,
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)
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if_load_asr_model = gr.Checkbox(
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label="Load / Unload ASR model for auto-reranking",
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value=False,
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scale=3,
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)
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with gr.Row():
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with gr.Tab(label="Advanced Config"):
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chunk_length = gr.Slider(
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label="Iterative Prompt Length, 0 means off",
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minimum=0,
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maximum=500,
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value=200,
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step=8,
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)
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max_new_tokens = gr.Slider(
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label="Maximum tokens per batch, 0 means no limit",
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minimum=0,
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maximum=2048,
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value=1024, # 0 means no limit
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step=8,
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)
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top_p = gr.Slider(
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label="Top-P",
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minimum=0.6,
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maximum=0.9,
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value=0.7,
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step=0.01,
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)
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repetition_penalty = gr.Slider(
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label="Repetition Penalty",
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minimum=1,
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maximum=1.5,
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value=1.2,
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step=0.01,
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)
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temperature = gr.Slider(
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label="Temperature",
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minimum=0.6,
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maximum=0.9,
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value=0.7,
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step=0.01,
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)
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with gr.Tab(label="Reference Audio"):
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gr.Markdown(
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447 |
-
"5 to 10 seconds of reference audio, useful for specifying speaker."
|
448 |
-
)
|
449 |
-
|
450 |
-
enable_reference_audio = gr.Checkbox(
|
451 |
-
label="Enable Reference Audio",
|
452 |
-
)
|
453 |
-
|
454 |
-
# Add dropdown for selecting example audio files
|
455 |
-
example_audio_files = [f for f in os.listdir("examples") if f.endswith(".wav")]
|
456 |
-
example_audio_dropdown = gr.Dropdown(
|
457 |
-
label="Select Example Audio",
|
458 |
-
choices=[""] + example_audio_files,
|
459 |
-
value=""
|
460 |
-
)
|
461 |
-
|
462 |
-
reference_audio = gr.Audio(
|
463 |
-
label="Reference Audio",
|
464 |
-
type="filepath",
|
465 |
-
)
|
466 |
-
with gr.Row():
|
467 |
-
if_auto_label = gr.Checkbox(
|
468 |
-
label="Auto Labeling",
|
469 |
-
min_width=100,
|
470 |
-
scale=0,
|
471 |
-
value=False,
|
472 |
-
)
|
473 |
-
reference_text = gr.Textbox(
|
474 |
-
label="Reference Text",
|
475 |
-
lines=1,
|
476 |
-
placeholder="在一无所知中,梦里的一天结束了,一个新的「轮回」便会开始。",
|
477 |
-
value="",
|
478 |
-
)
|
479 |
-
with gr.Tab(label="Batch Inference"):
|
480 |
-
batch_infer_num = gr.Slider(
|
481 |
-
label="Batch infer nums",
|
482 |
-
minimum=1,
|
483 |
-
maximum=n_audios,
|
484 |
-
step=1,
|
485 |
-
value=1,
|
486 |
-
)
|
487 |
-
|
488 |
-
with gr.Column(scale=3):
|
489 |
-
for _ in range(n_audios):
|
490 |
-
with gr.Row():
|
491 |
-
error = gr.HTML(
|
492 |
-
label="Error Message",
|
493 |
-
visible=True if _ == 0 else False,
|
494 |
-
)
|
495 |
-
global_error_list.append(error)
|
496 |
-
with gr.Row():
|
497 |
-
audio = gr.Audio(
|
498 |
-
label="Generated Audio",
|
499 |
-
type="numpy",
|
500 |
-
interactive=False,
|
501 |
-
visible=True if _ == 0 else False,
|
502 |
-
)
|
503 |
-
global_audio_list.append(audio)
|
504 |
-
|
505 |
-
with gr.Row():
|
506 |
-
stream_audio = gr.Audio(
|
507 |
-
label="Streaming Audio",
|
508 |
-
streaming=True,
|
509 |
-
autoplay=True,
|
510 |
-
interactive=False,
|
511 |
-
show_download_button=True,
|
512 |
-
)
|
513 |
-
with gr.Row():
|
514 |
-
with gr.Column(scale=3):
|
515 |
-
generate = gr.Button(
|
516 |
-
value="\U0001F3A7 " + "Generate", variant="primary"
|
517 |
-
)
|
518 |
-
generate_stream = gr.Button(
|
519 |
-
value="\U0001F3A7 " + "Streaming Generate",
|
520 |
-
variant="primary",
|
521 |
-
)
|
522 |
-
|
523 |
-
text.input(
|
524 |
-
fn=normalize_text, inputs=[text, if_refine_text], outputs=[refined_text]
|
525 |
-
)
|
526 |
-
|
527 |
-
if_load_asr_model.change(
|
528 |
-
fn=change_if_load_asr_model,
|
529 |
-
inputs=[if_load_asr_model],
|
530 |
-
outputs=[if_load_asr_model],
|
531 |
-
)
|
532 |
-
|
533 |
-
if_auto_label.change(
|
534 |
-
fn=lambda: gr.Textbox(value=""),
|
535 |
-
inputs=[],
|
536 |
-
outputs=[reference_text],
|
537 |
-
).then(
|
538 |
-
fn=change_if_auto_label,
|
539 |
-
inputs=[
|
540 |
-
if_load_asr_model,
|
541 |
-
if_auto_label,
|
542 |
-
enable_reference_audio,
|
543 |
-
reference_audio,
|
544 |
-
reference_text,
|
545 |
-
],
|
546 |
-
outputs=[reference_text],
|
547 |
-
)
|
548 |
-
|
549 |
-
def select_example_audio(audio_file):
|
550 |
-
if audio_file:
|
551 |
-
audio_path = os.path.join("examples", audio_file)
|
552 |
-
lab_file = os.path.splitext(audio_file)[0] + ".lab"
|
553 |
-
lab_path = os.path.join("examples", lab_file)
|
554 |
-
|
555 |
-
if os.path.exists(lab_path):
|
556 |
-
with open(lab_path, "r", encoding="utf-8") as f:
|
557 |
-
lab_content = f.read().strip()
|
558 |
-
else:
|
559 |
-
lab_content = ""
|
560 |
-
|
561 |
-
return audio_path, lab_content, True
|
562 |
-
return None, "", False
|
563 |
-
|
564 |
-
# Connect the dropdown to update reference audio and text
|
565 |
-
example_audio_dropdown.change(
|
566 |
-
fn=select_example_audio,
|
567 |
-
inputs=[example_audio_dropdown],
|
568 |
-
outputs=[reference_audio, reference_text, enable_reference_audio]
|
569 |
-
)
|
570 |
-
# # Submit
|
571 |
-
generate.click(
|
572 |
-
inference_wrapper,
|
573 |
-
[
|
574 |
-
refined_text,
|
575 |
-
enable_reference_audio,
|
576 |
-
reference_audio,
|
577 |
-
reference_text,
|
578 |
-
max_new_tokens,
|
579 |
-
chunk_length,
|
580 |
-
top_p,
|
581 |
-
repetition_penalty,
|
582 |
-
temperature,
|
583 |
-
batch_infer_num,
|
584 |
-
if_load_asr_model,
|
585 |
-
],
|
586 |
-
[stream_audio, *global_audio_list, *global_error_list],
|
587 |
-
concurrency_limit=1,
|
588 |
-
)
|
589 |
-
return app
|
590 |
-
|
591 |
|
592 |
def parse_args():
|
593 |
parser = ArgumentParser()
|
594 |
parser.add_argument(
|
595 |
"--llama-checkpoint-path",
|
596 |
-
type=
|
597 |
-
default="checkpoints/fish-speech-1.4",
|
|
|
598 |
)
|
599 |
parser.add_argument(
|
600 |
"--decoder-checkpoint-path",
|
601 |
-
type=
|
602 |
default="checkpoints/fish-speech-1.4/firefly-gan-vq-fsq-8x1024-21hz-generator.pth",
|
|
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|
603 |
)
|
604 |
-
parser.add_argument("--decoder-config-name", type=str, default="firefly_gan_vq")
|
605 |
-
parser.add_argument("--device", type=str, default="cuda")
|
606 |
-
parser.add_argument("--half", action="store_true")
|
607 |
-
parser.add_argument("--compile", action="store_true",default=True)
|
608 |
-
parser.add_argument("--max-gradio-length", type=int, default=0)
|
609 |
-
parser.add_argument("--theme", type=str, default="light")
|
610 |
-
|
611 |
return parser.parse_args()
|
612 |
|
613 |
|
614 |
-
|
615 |
args = parse_args()
|
|
|
616 |
args.precision = torch.half if args.half else torch.bfloat16
|
617 |
|
618 |
logger.info("Loading Llama model...")
|
@@ -632,22 +80,25 @@ if __name__ == "__main__":
|
|
632 |
|
633 |
logger.info("Decoder model loaded, warming up...")
|
634 |
|
635 |
-
#
|
636 |
-
|
637 |
-
|
638 |
-
|
639 |
-
|
640 |
-
|
641 |
-
|
642 |
-
|
643 |
-
|
644 |
-
|
645 |
-
|
646 |
-
temperature=0.7,
|
647 |
-
)
|
648 |
)
|
649 |
|
650 |
logger.info("Warming up done, launching the web UI...")
|
651 |
|
|
|
652 |
app = build_app()
|
653 |
app.launch(show_api=True)
|
|
|
|
|
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|
|
1 |
import os
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|
2 |
import torch
|
3 |
+
from argparse import ArgumentParser
|
|
|
|
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|
4 |
from loguru import logger
|
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|
|
|
5 |
from tools.llama.generate import launch_thread_safe_queue
|
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|
6 |
from tools.vqgan.inference import load_model as load_decoder_model
|
7 |
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|
8 |
|
9 |
def parse_args():
|
10 |
parser = ArgumentParser()
|
11 |
parser.add_argument(
|
12 |
"--llama-checkpoint-path",
|
13 |
+
type=str,
|
14 |
+
default="checkpoints/fish-speech-1.4-sft-yth-lora",
|
15 |
+
help="Path to the Llama checkpoint"
|
16 |
)
|
17 |
parser.add_argument(
|
18 |
"--decoder-checkpoint-path",
|
19 |
+
type=str,
|
20 |
default="checkpoints/fish-speech-1.4/firefly-gan-vq-fsq-8x1024-21hz-generator.pth",
|
21 |
+
help="Path to the VQ-GAN checkpoint"
|
22 |
+
)
|
23 |
+
parser.add_argument(
|
24 |
+
"--decoder-config-name",
|
25 |
+
type=str,
|
26 |
+
default="firefly_gan_vq",
|
27 |
+
help="VQ-GAN config name"
|
28 |
+
)
|
29 |
+
parser.add_argument(
|
30 |
+
"--device",
|
31 |
+
type=str,
|
32 |
+
default="cpu",
|
33 |
+
help="Device to run on (cpu or cuda)"
|
34 |
+
)
|
35 |
+
parser.add_argument(
|
36 |
+
"--half",
|
37 |
+
action="store_true",
|
38 |
+
help="Use half precision"
|
39 |
+
)
|
40 |
+
parser.add_argument(
|
41 |
+
"--compile",
|
42 |
+
action="store_true",
|
43 |
+
default=True,
|
44 |
+
help="Compile the model for optimized inference"
|
45 |
+
)
|
46 |
+
parser.add_argument(
|
47 |
+
"--max-gradio-length",
|
48 |
+
type=int,
|
49 |
+
default=0,
|
50 |
+
help="Maximum length for Gradio input"
|
51 |
+
)
|
52 |
+
parser.add_argument(
|
53 |
+
"--theme",
|
54 |
+
type=str,
|
55 |
+
default="light",
|
56 |
+
help="Theme for the Gradio app"
|
57 |
)
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
58 |
return parser.parse_args()
|
59 |
|
60 |
|
61 |
+
def main():
|
62 |
args = parse_args()
|
63 |
+
|
64 |
args.precision = torch.half if args.half else torch.bfloat16
|
65 |
|
66 |
logger.info("Loading Llama model...")
|
|
|
80 |
|
81 |
logger.info("Decoder model loaded, warming up...")
|
82 |
|
83 |
+
# Perform a dry run to warm up the model
|
84 |
+
inference(
|
85 |
+
text="Hello, world!",
|
86 |
+
enable_reference_audio=False,
|
87 |
+
reference_audio=None,
|
88 |
+
reference_text="",
|
89 |
+
max_new_tokens=0,
|
90 |
+
chunk_length=100,
|
91 |
+
top_p=0.7,
|
92 |
+
repetition_penalty=1.2,
|
93 |
+
temperature=0.7,
|
|
|
|
|
94 |
)
|
95 |
|
96 |
logger.info("Warming up done, launching the web UI...")
|
97 |
|
98 |
+
# Launch the Gradio app
|
99 |
app = build_app()
|
100 |
app.launch(show_api=True)
|
101 |
+
|
102 |
+
|
103 |
+
if __name__ == "__main__":
|
104 |
+
main()
|