mirror of
https://github.com/SWivid/F5-TTS.git
synced 2026-01-15 22:42:43 -08:00
Merge pull request #476 from lpscr/main
add in gradio finetune 8bit value fix some stuff and add new transcripe into easy
This commit is contained in:
@@ -15,6 +15,7 @@ from f5_tts.infer.utils_infer import (
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preprocess_ref_audio_text,
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remove_silence_for_generated_wav,
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save_spectrogram,
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transcribe,
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target_sample_rate,
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)
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from f5_tts.model import DiT, UNetT
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@@ -82,6 +83,9 @@ class F5TTS:
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model_cls, model_cfg, ckpt_file, mel_spec_type, vocab_file, ode_method, use_ema, self.device
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)
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def transcribe(self, ref_audio, language=None):
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return transcribe(ref_audio, language)
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def export_wav(self, wav, file_wave, remove_silence=False):
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sf.write(file_wave, wav, self.target_sample_rate)
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@@ -150,6 +150,22 @@ def initialize_asr_pipeline(device=device, dtype=None):
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)
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# transcribe
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def transcribe(ref_audio, language=None):
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global asr_pipe
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if asr_pipe is None:
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initialize_asr_pipeline(device=device)
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return asr_pipe(
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ref_audio,
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chunk_length_s=30,
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batch_size=128,
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generate_kwargs={"task": "transcribe", "language": language} if language else {"task": "transcribe"},
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return_timestamps=False,
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)["text"].strip()
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# load model checkpoint for inference
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@@ -306,17 +322,8 @@ def preprocess_ref_audio_text(ref_audio_orig, ref_text, clip_short=True, show_in
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show_info("Using cached reference text...")
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ref_text = _ref_audio_cache[audio_hash]
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else:
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global asr_pipe
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if asr_pipe is None:
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initialize_asr_pipeline(device=device)
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show_info("No reference text provided, transcribing reference audio...")
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ref_text = asr_pipe(
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ref_audio,
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chunk_length_s=30,
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batch_size=128,
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generate_kwargs={"task": "transcribe"},
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return_timestamps=False,
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)["text"].strip()
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ref_text = transcribe(ref_audio)
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# Cache the transcribed text (not caching custom ref_text, enabling users to do manual tweak)
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_ref_audio_cache[audio_hash] = ref_text
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else:
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@@ -13,6 +13,9 @@ from importlib.resources import files
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target_sample_rate = 24000
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n_mel_channels = 100
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hop_length = 256
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win_length = 1024
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n_fft = 1024
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mel_spec_type = "vocos" # 'vocos' or 'bigvgan'
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# -------------------------- Argument Parsing --------------------------- #
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@@ -40,7 +43,7 @@ def parse_args():
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parser.add_argument("--max_samples", type=int, default=64, help="Max sequences per batch")
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parser.add_argument("--grad_accumulation_steps", type=int, default=1, help="Gradient accumulation steps")
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parser.add_argument("--max_grad_norm", type=float, default=1.0, help="Max gradient norm for clipping")
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parser.add_argument("--epochs", type=int, default=10, help="Number of training epochs")
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parser.add_argument("--epochs", type=int, default=100, help="Number of training epochs")
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parser.add_argument("--num_warmup_updates", type=int, default=300, help="Warmup steps")
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parser.add_argument("--save_per_updates", type=int, default=10000, help="Save checkpoint every X steps")
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parser.add_argument("--last_per_steps", type=int, default=50000, help="Save last checkpoint every X steps")
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@@ -121,11 +124,15 @@ def main():
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vocab_char_map, vocab_size = get_tokenizer(tokenizer_path, tokenizer)
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print("\nvocab : ", vocab_size)
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print("\nvocoder : ", mel_spec_type)
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mel_spec_kwargs = dict(
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target_sample_rate=target_sample_rate,
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n_mel_channels=n_mel_channels,
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n_fft=n_fft,
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hop_length=hop_length,
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win_length=win_length,
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n_mel_channels=n_mel_channels,
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target_sample_rate=target_sample_rate,
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mel_spec_type=mel_spec_type,
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)
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model = CFM(
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@@ -26,12 +26,13 @@ from datasets import Dataset as Dataset_
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from datasets.arrow_writer import ArrowWriter
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from safetensors.torch import save_file
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from scipy.io import wavfile
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from transformers import pipeline
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from cached_path import cached_path
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from f5_tts.api import F5TTS
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from f5_tts.model.utils import convert_char_to_pinyin
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from f5_tts.infer.utils_infer import transcribe
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from importlib.resources import files
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training_process = None
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system = platform.system()
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python_executable = sys.executable or "python"
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@@ -47,8 +48,6 @@ file_train = "src/f5_tts/train/finetune_cli.py"
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device = "cuda" if torch.cuda.is_available() else "mps" if torch.backends.mps.is_available() else "cpu"
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pipe = None
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# Save settings from a JSON file
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def save_settings(
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@@ -70,6 +69,7 @@ def save_settings(
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tokenizer_file,
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mixed_precision,
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logger,
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ch_8bit_adam,
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):
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path_project = os.path.join(path_project_ckpts, project_name)
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os.makedirs(path_project, exist_ok=True)
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@@ -93,6 +93,7 @@ def save_settings(
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"tokenizer_file": tokenizer_file,
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"mixed_precision": mixed_precision,
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"logger": logger,
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"bnb_optimizer": ch_8bit_adam,
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}
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with open(file_setting, "w") as f:
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json.dump(settings, f, indent=4)
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@@ -124,6 +125,7 @@ def load_settings(project_name):
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"tokenizer_file": "",
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"mixed_precision": "none",
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"logger": "wandb",
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"bnb_optimizer": False,
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}
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return (
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settings["exp_name"],
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@@ -143,12 +145,15 @@ def load_settings(project_name):
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settings["tokenizer_file"],
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settings["mixed_precision"],
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settings["logger"],
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settings["bnb_optimizer"],
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)
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with open(file_setting, "r") as f:
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settings = json.load(f)
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if "logger" not in settings:
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settings["logger"] = "wandb"
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if "bnb_optimizer" not in settings:
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settings["bnb_optimizer"] = False
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return (
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settings["exp_name"],
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settings["learning_rate"],
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@@ -167,6 +172,7 @@ def load_settings(project_name):
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settings["tokenizer_file"],
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settings["mixed_precision"],
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settings["logger"],
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settings["bnb_optimizer"],
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)
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@@ -381,18 +387,17 @@ def start_training(
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mixed_precision="fp16",
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stream=False,
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logger="wandb",
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ch_8bit_adam=False,
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):
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global training_process, tts_api, stop_signal, pipe
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global training_process, tts_api, stop_signal
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if tts_api is not None or pipe is not None:
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if tts_api is not None:
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if tts_api is not None:
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del tts_api
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if pipe is not None:
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del pipe
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gc.collect()
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torch.cuda.empty_cache()
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tts_api = None
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pipe = None
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path_project = os.path.join(path_data, dataset_name)
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@@ -447,11 +452,10 @@ def start_training(
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f"--dataset_name {dataset_name}"
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)
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if finetune:
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cmd += f" --finetune {finetune}"
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cmd += f" --finetune {finetune}"
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if file_checkpoint_train != "":
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cmd += f" --file_checkpoint_train {file_checkpoint_train}"
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cmd += f" --pretrain {file_checkpoint_train}"
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if tokenizer_file != "":
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cmd += f" --tokenizer_path {tokenizer_file}"
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@@ -460,7 +464,10 @@ def start_training(
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cmd += f" --log_samples True --logger {logger} "
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print(cmd)
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if ch_8bit_adam:
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cmd += " --bnb_optimizer True "
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print("run command : \n" + cmd + "\n")
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save_settings(
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dataset_name,
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@@ -481,6 +488,7 @@ def start_training(
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tokenizer_file,
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mixed_precision,
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logger,
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ch_8bit_adam,
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)
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try:
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@@ -641,27 +649,6 @@ def create_data_project(name, tokenizer_type):
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return gr.update(choices=project_list, value=name)
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def transcribe(file_audio, language="english"):
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global pipe
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if pipe is None:
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pipe = pipeline(
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"automatic-speech-recognition",
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model="openai/whisper-large-v3-turbo",
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torch_dtype=torch.float16,
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device=device,
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)
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text_transcribe = pipe(
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file_audio,
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chunk_length_s=30,
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batch_size=128,
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generate_kwargs={"task": "transcribe", "language": language},
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return_timestamps=False,
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)["text"].strip()
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return text_transcribe
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def transcribe_all(name_project, audio_files, language, user=False, progress=gr.Progress()):
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path_project = os.path.join(path_data, name_project)
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path_dataset = os.path.join(path_project, "dataset")
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@@ -758,11 +745,9 @@ def get_correct_audio_path(
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# Case 2: If it has a supported extension but is not a full path
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elif has_supported_extension(audio_input) and not os.path.isabs(audio_input):
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file_audio = os.path.join(base_path, audio_input)
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print("2")
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# Case 3: If only the name is given (no extension and not a full path)
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elif not has_supported_extension(audio_input) and not os.path.isabs(audio_input):
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print("3")
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for ext in supported_formats:
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potential_file = os.path.join(base_path, f"{audio_input}.{ext}")
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if os.path.exists(potential_file):
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@@ -816,9 +801,12 @@ def create_metadata(name_project, ch_tokenizer, progress=gr.Progress()):
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continue
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if duration < 1 or duration > 25:
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error_files.append([file_audio, "duration < 1 or > 25 "])
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if duration > 25:
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error_files.append([file_audio, "duration > 25 sec"])
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if duration < 1:
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error_files.append([file_audio, "duration < 1 sec "])
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continue
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if len(text) < 4:
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if len(text) < 3:
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error_files.append([file_audio, "very small text len 3"])
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continue
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@@ -1208,7 +1196,9 @@ def get_random_sample_infer(project_name):
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)
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def infer(project, file_checkpoint, exp_name, ref_text, ref_audio, gen_text, nfe_step, use_ema):
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def infer(
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project, file_checkpoint, exp_name, ref_text, ref_audio, gen_text, nfe_step, use_ema, speed, seed, remove_silence
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):
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global last_checkpoint, last_device, tts_api, last_ema
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if not os.path.isfile(file_checkpoint):
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@@ -1238,8 +1228,17 @@ def infer(project, file_checkpoint, exp_name, ref_text, ref_audio, gen_text, nfe
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print("update >> ", device_test, file_checkpoint, use_ema)
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with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as f:
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tts_api.infer(gen_text=gen_text, ref_text=ref_text, ref_file=ref_audio, nfe_step=nfe_step, file_wave=f.name)
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return f.name, tts_api.device
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tts_api.infer(
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gen_text=gen_text.lower().strip(),
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ref_text=ref_text.lower().strip(),
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ref_file=ref_audio,
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nfe_step=nfe_step,
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file_wave=f.name,
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speed=speed,
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seed=seed,
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remove_silence=remove_silence,
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)
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return f.name, tts_api.device, str(tts_api.seed)
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def check_finetune(finetune):
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@@ -1506,6 +1505,7 @@ Skip this step if you have your dataset, raw.arrow, duration.json, and vocab.txt
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```"""
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)
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ch_tokenizern = gr.Checkbox(label="Create Vocabulary", value=False, visible=False)
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bt_prepare = bt_create = gr.Button("Prepare")
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txt_info_prepare = gr.Text(label="Info", value="")
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txt_vocab_prepare = gr.Text(label="Vocab", value="")
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@@ -1560,6 +1560,7 @@ If you encounter a memory error, try reducing the batch size per GPU to a smalle
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last_per_steps = gr.Number(label="Last per Steps", value=100)
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with gr.Row():
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ch_8bit_adam = gr.Checkbox(label="Use 8-bit Adam optimizer")
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mixed_precision = gr.Radio(label="mixed_precision", choices=["none", "fp16", "bf16"], value="none")
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cd_logger = gr.Radio(label="logger", choices=["wandb", "tensorboard"], value="wandb")
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start_button = gr.Button("Start Training")
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@@ -1584,6 +1585,7 @@ If you encounter a memory error, try reducing the batch size per GPU to a smalle
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tokenizer_filev,
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mixed_precisionv,
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cd_loggerv,
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ch_8bit_adamv,
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) = load_settings(projects_selelect)
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exp_name.value = exp_namev
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learning_rate.value = learning_ratev
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@@ -1602,6 +1604,7 @@ If you encounter a memory error, try reducing the batch size per GPU to a smalle
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tokenizer_file.value = tokenizer_filev
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mixed_precision.value = mixed_precisionv
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cd_logger.value = cd_loggerv
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ch_8bit_adam.value = ch_8bit_adamv
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ch_stream = gr.Checkbox(label="Stream Output Experiment", value=True)
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txt_info_train = gr.Text(label="Info", value="")
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@@ -1660,6 +1663,7 @@ If you encounter a memory error, try reducing the batch size per GPU to a smalle
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mixed_precision,
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ch_stream,
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cd_logger,
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ch_8bit_adam,
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],
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outputs=[txt_info_train, start_button, stop_button],
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)
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@@ -1732,12 +1736,17 @@ If you encounter a memory error, try reducing the batch size per GPU to a smalle
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with gr.TabItem("Test Model"):
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gr.Markdown("""```plaintext
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SOS: Check the use_ema setting (True or False) for your model to see what works best for you.
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SOS: Check the use_ema setting (True or False) for your model to see what works best for you. use seed -1 from random
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```""")
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exp_name = gr.Radio(label="Model", choices=["F5-TTS", "E2-TTS"], value="F5-TTS")
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list_checkpoints, checkpoint_select = get_checkpoints_project(projects_selelect, False)
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nfe_step = gr.Number(label="NFE Step", value=32)
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with gr.Row():
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nfe_step = gr.Number(label="NFE Step", value=32)
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speed = gr.Slider(label="Speed", value=1.0, minimum=0.3, maximum=2.0, step=0.1)
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seed = gr.Number(label="Seed", value=-1, minimum=-1)
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remove_silence = gr.Checkbox(label="Remove Silence")
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ch_use_ema = gr.Checkbox(label="Use EMA", value=True)
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with gr.Row():
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cm_checkpoint = gr.Dropdown(
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@@ -1757,14 +1766,27 @@ SOS: Check the use_ema setting (True or False) for your model to see what works
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with gr.Row():
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txt_info_gpu = gr.Textbox("", label="Device")
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seed_info = gr.Text(label="Seed :")
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check_button_infer = gr.Button("Infer")
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gen_audio = gr.Audio(label="Audio Gen", type="filepath")
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check_button_infer.click(
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fn=infer,
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inputs=[cm_project, cm_checkpoint, exp_name, ref_text, ref_audio, gen_text, nfe_step, ch_use_ema],
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outputs=[gen_audio, txt_info_gpu],
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inputs=[
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cm_project,
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cm_checkpoint,
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exp_name,
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ref_text,
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ref_audio,
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gen_text,
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nfe_step,
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ch_use_ema,
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speed,
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seed,
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remove_silence,
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],
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outputs=[gen_audio, txt_info_gpu, seed_info],
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)
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bt_checkpoint_refresh.click(fn=get_checkpoints_project, inputs=[cm_project], outputs=[cm_checkpoint])
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