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Upload 12 files
Browse files- app.py +78 -0
- assets/whistress_model.svg +0 -0
- requirements.txt +7 -0
- whistress/__init__.py +1 -0
- whistress/inference_client/__init__.py +1 -0
- whistress/inference_client/utils.py +163 -0
- whistress/inference_client/whistress_client.py +26 -0
- whistress/model/__init__.py +1 -0
- whistress/model/model.py +318 -0
- whistress/weights/additional_decoder_block.pt +3 -0
- whistress/weights/classifier.pt +3 -0
- whistress/weights/metadata.json +3 -0
app.py
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import torch
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import gradio as gr
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from pathlib import Path
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from whistress import WhiStressInferenceClient
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CURRENT_DIR = Path(__file__).parent
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# Load the model
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device = "cuda" if torch.cuda.is_available() else "cpu"
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model = WhiStressInferenceClient(device=device)
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def get_whistress_predictions(audio):
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"""
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Get the transcription and emphasis scores for the given audio input.
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Args:
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audio (sr, numpy.ndarray): The audio input as a NumPy array.
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Returns:
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List[Tuple[str, int]]: A list of tuples containing words and their emphasis scores.
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"""
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audio = {
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"sampling_rate": audio[0],
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"array": audio[1],
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}
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return model.predict(audio=audio, transcription=None, return_pairs=True)
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# App UI
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with gr.Blocks() as demo:
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with gr.Row():
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with gr.Column(scale=2):
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gr.Markdown(
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"""
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# WhiStress: Enriching Transcriptions with Sentence Stress Detection
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The WhiStress model allows you to detect important words in your speech.
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Check out our paper: 📚 [WhiStress](https://arxiv.org/),
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## Architecture
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The model is built on [Whisper](https://arxiv.org/abs/2212.04356) model,
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using `whisper-small.en` [model](https://huggingface.co/openai/whisper-small.en)
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as the backbone.
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WhiStress includes an additional decoder based classifier that predicts the stress label of each transcription token.
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## Training Data
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The model was trained using [TinyStress-15K](https://huggingface.co/datasets/loud-whisper-project/tinyStories-audio-emphasized),
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that is derived from [tinyStories](https://huggingface.co/datasets/roneneldan/TinyStories) dataset.
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## Inference Demo
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Upload an audio file or record your own voice to transcribe the speech and emphasize the important words.
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For maximal performance, please speak clearly.
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"""
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)
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with gr.Column(scale=1):
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# Define Gradio interface for displaying image with HTML component
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gr.Image(
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f"{CURRENT_DIR}/assets/whistress_model.svg",
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label="Architecture",
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)
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gr.Interface(
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get_whistress_predictions,
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gr.Audio(
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sources=["microphone", "upload"],
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label="Upload speech or record your own",
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type="numpy",
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),
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gr.HighlightedText(),
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allow_flagging="never",
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)
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def launch():
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demo.launch()
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if __name__ == "__main__":
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launch()
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assets/whistress_model.svg
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requirements.txt
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torch==2.1.0
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torchaudio==2.1.0
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torchlibrosa==0.1.0
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librosa==0.10.2.post1
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transformers==4.44.0
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numpy==1.26.4
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gradio==5.31.0
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whistress/__init__.py
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from .inference_client import WhiStressInferenceClient
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whistress/inference_client/__init__.py
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from .whistress_client import WhiStressInferenceClient
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whistress/inference_client/utils.py
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import torch
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from transformers import WhisperConfig
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import librosa
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import numpy as np
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import pathlib
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from torch.nn import functional as F
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from ..model import WhiStress
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PATH_TO_WEIGHTS = pathlib.Path(__file__).parent.parent / "weights"
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def get_loaded_model(device="cuda"):
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whisper_model_name = f"openai/whisper-small.en"
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whisper_config = WhisperConfig()
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whistress_model = WhiStress(
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whisper_config, layer_for_head=9, whisper_backbone_name=whisper_model_name
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).to(device)
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whistress_model.processor.tokenizer.model_input_names = [
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"input_ids",
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"attention_mask",
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"labels_head",
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]
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whistress_model.load_model(PATH_TO_WEIGHTS)
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whistress_model.to(device)
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whistress_model.eval()
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return whistress_model
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def get_word_emphasis_pairs(
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transcription_preds, emphasis_preds, processor, filter_special_tokens=True
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):
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emphasis_preds_list = emphasis_preds.tolist()
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transcription_preds_words = [
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processor.tokenizer.decode([i], skip_special_tokens=False)
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for i in transcription_preds
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]
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if filter_special_tokens:
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special_tokens_indices = [
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i
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for i, x in enumerate(transcription_preds)
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if x in processor.tokenizer.all_special_ids
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]
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emphasis_preds_list = [
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x
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for i, x in enumerate(emphasis_preds_list)
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if i not in special_tokens_indices
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]
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transcription_preds_words = [
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x
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for i, x in enumerate(transcription_preds_words)
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if i not in special_tokens_indices
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]
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return list(zip(transcription_preds_words, emphasis_preds_list))
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def inference_from_audio(audio: np.ndarray, model: WhiStress, device: str):
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input_features = model.processor.feature_extractor(
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audio, sampling_rate=16000, return_tensors="pt"
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)["input_features"]
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out_model = model.generate_dual(input_features=input_features.to(device))
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emphasis_probs = F.softmax(out_model.logits, dim=-1)
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emphasis_preds = torch.argmax(emphasis_probs, dim=-1)
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emphasis_preds_right_shifted = torch.cat((emphasis_preds[:, -1:], emphasis_preds[:, :-1]), dim=1)
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word_emphasis_pairs = get_word_emphasis_pairs(
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out_model.preds[0],
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emphasis_preds_right_shifted[0],
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model.processor,
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filter_special_tokens=True,
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)
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return word_emphasis_pairs
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def prepare_audio(audio, target_sr=16000):
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# resample to 16kHz
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sr = audio["sampling_rate"]
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y = audio["array"]
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y = np.array(y, dtype=float)
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y_resampled = librosa.resample(y, orig_sr=sr, target_sr=target_sr)
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# Normalize the audio (scale to [-1, 1])
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y_resampled /= max(abs(y_resampled))
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return y_resampled
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def merge_stressed_tokens(tokens_with_stress):
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"""
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tokens_with_stress is a list of tuples: (token_string, stress_value)
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e.g.:
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[(" I", 0), (" didn", 1), ("'t", 0), (" say", 0), (" he", 0), (" stole", 0),
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(" the", 0), (" money", 0), (".", 0)]
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Returns a list of merged tuples, combining subwords into full words.
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"""
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merged = []
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current_word = ""
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current_stress = 0 # 0 means not stressed, 1 means stressed
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for token, stress in tokens_with_stress:
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# If token starts with a space (or is the very first), we treat it as a new word
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# or if current_word is empty (first iteration).
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if token.startswith(" ") or current_word == "":
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# If we already have something in current_word, push it into merged
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# before starting a new one
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if current_word:
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merged.append((current_word, current_stress))
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# Start a new word
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current_word = token
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current_stress = stress
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else:
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# Otherwise, it's a subword that should be appended to the previous word
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current_word += token
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# If any sub-token is stressed, the whole merged word is stressed
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current_stress = max(current_stress, stress)
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# Don't forget to append the final word
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if current_word:
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merged.append((current_word, current_stress))
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return merged
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def inference_from_audio_and_transcription(
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audio: np.ndarray, transcription, model: WhiStress, device: str
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):
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input_features = model.processor.feature_extractor(
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audio, sampling_rate=16000, return_tensors="pt"
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)["input_features"]
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# convert transcription to input_ids
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input_ids = model.processor.tokenizer(
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transcription,
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return_tensors="pt",
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padding="max_length",
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truncation=True,
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max_length=30,
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)["input_ids"]
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out_model = model(
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input_features=input_features.to(device),
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decoder_input_ids=input_ids.to(device),
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)
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emphasis_probs = F.softmax(out_model.logits, dim=-1)
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emphasis_preds = torch.argmax(emphasis_probs, dim=-1)
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emphasis_preds_right_shifted = torch.cat((emphasis_preds[:, -1:], emphasis_preds[:, :-1]), dim=1)
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word_emphasis_pairs = get_word_emphasis_pairs(
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input_ids[0],
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emphasis_preds_right_shifted[0],
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model.processor,
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filter_special_tokens=True,
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)
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return word_emphasis_pairs
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def scored_transcription(audio, model, strip_words=True, transcription: str = None, device="cuda"):
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audio_arr = prepare_audio(audio)
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token_stress_pairs = None
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if transcription: # if we want to use the ground truth transcription
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token_stress_pairs = inference_from_audio_and_transcription(audio_arr, transcription, model, device)
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else:
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token_stress_pairs = inference_from_audio(audio_arr, model, device)
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# token_stress_pairs = inference_from_audio(audio_arr, model)
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word_level_stress = merge_stressed_tokens(token_stress_pairs)
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if strip_words:
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word_level_stress = [(word.strip(), stress) for word, stress in word_level_stress]
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return word_level_stress
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whistress/inference_client/whistress_client.py
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import numpy as np
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from .utils import get_loaded_model, scored_transcription
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from typing import Union, Dict
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class WhiStressInferenceClient:
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def __init__(self, device="cuda"):
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self.device = device
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self.whistress = get_loaded_model(self.device)
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def predict(
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self, audio: Dict[str, Union[np.ndarray, int]], transcription=None, return_pairs=True
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):
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word_emphasis_pairs = scored_transcription(
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audio=audio,
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model=self.whistress,
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device=self.device,
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strip_words=True,
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transcription=transcription
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)
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if return_pairs:
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return word_emphasis_pairs
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# returs transcription str and list of emphasized words
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return " ".join([x[0] for x in word_emphasis_pairs]), [
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x[0] for x in word_emphasis_pairs if x[1] == 1
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]
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whistress/model/__init__.py
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from .model import WhiStress
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whistress/model/model.py
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|
1 |
+
from transformers import (
|
2 |
+
WhisperForConditionalGeneration,
|
3 |
+
WhisperProcessor,
|
4 |
+
PreTrainedModel,
|
5 |
+
WhisperConfig,
|
6 |
+
)
|
7 |
+
from transformers.models.whisper.modeling_whisper import WhisperDecoderLayer
|
8 |
+
from transformers.modeling_outputs import BaseModelOutput
|
9 |
+
import torch.nn.functional as F
|
10 |
+
import torch.nn as nn
|
11 |
+
import torch
|
12 |
+
import os
|
13 |
+
from dataclasses import dataclass
|
14 |
+
from typing import Optional
|
15 |
+
import json
|
16 |
+
|
17 |
+
|
18 |
+
@dataclass
|
19 |
+
class CustomModelOutput(BaseModelOutput):
|
20 |
+
loss: Optional[torch.FloatTensor] = None
|
21 |
+
logits: torch.FloatTensor = None
|
22 |
+
head_preds: torch.FloatTensor = None
|
23 |
+
labels_head: Optional[torch.FloatTensor] = None
|
24 |
+
whisper_logits: torch.FloatTensor = None
|
25 |
+
preds: Optional[torch.Tensor] = None
|
26 |
+
|
27 |
+
|
28 |
+
# Define a new head (e.g., a classification layer)
|
29 |
+
class LinearHead(nn.Module):
|
30 |
+
def __init__(self, input_dim, output_dim):
|
31 |
+
super(LinearHead, self).__init__()
|
32 |
+
self.linear = nn.Linear(input_dim, output_dim)
|
33 |
+
|
34 |
+
def forward(self, x):
|
35 |
+
return self.linear(x)
|
36 |
+
|
37 |
+
|
38 |
+
class FCNN(nn.Module):
|
39 |
+
def __init__(self, input_dim, output_dim):
|
40 |
+
super(FCNN, self).__init__()
|
41 |
+
hidden_dim = 2 * input_dim
|
42 |
+
self.fc1 = nn.Linear(input_dim, hidden_dim)
|
43 |
+
self.fc2 = nn.Linear(hidden_dim, output_dim)
|
44 |
+
|
45 |
+
def forward(self, x):
|
46 |
+
x = F.relu(self.fc1(x))
|
47 |
+
x = self.fc2(x)
|
48 |
+
return x
|
49 |
+
|
50 |
+
|
51 |
+
class WhiStress(PreTrainedModel):
|
52 |
+
|
53 |
+
config_class = WhisperConfig
|
54 |
+
model_input_names = ["input_features", "labels_head", "whisper_labels"]
|
55 |
+
|
56 |
+
def __init__(
|
57 |
+
self,
|
58 |
+
config: WhisperConfig,
|
59 |
+
layer_for_head: Optional[int] = None,
|
60 |
+
whisper_backbone_name="openai/whisper-small.en",
|
61 |
+
):
|
62 |
+
super().__init__(config)
|
63 |
+
self.whisper_backbone_name = whisper_backbone_name
|
64 |
+
self.whisper_model = WhisperForConditionalGeneration.from_pretrained(
|
65 |
+
self.whisper_backbone_name,
|
66 |
+
).eval()
|
67 |
+
self.processor = WhisperProcessor.from_pretrained(self.whisper_backbone_name)
|
68 |
+
|
69 |
+
input_dim = self.whisper_model.config.d_model # Model's hidden size
|
70 |
+
output_dim = 2 # Number of classes or output features for the new head
|
71 |
+
|
72 |
+
config = self.whisper_model.config
|
73 |
+
# add additional decoder block using the existing Whisper config
|
74 |
+
self.additional_decoder_block = WhisperDecoderLayer(config)
|
75 |
+
self.classifier = FCNN(input_dim, output_dim)
|
76 |
+
# add weighted loss for CE
|
77 |
+
neg_weight = 1.0
|
78 |
+
pos_weight = 0.7 / 0.3
|
79 |
+
class_weights = torch.tensor([neg_weight, pos_weight])
|
80 |
+
self.loss_fct = nn.CrossEntropyLoss(ignore_index=-100, weight=class_weights)
|
81 |
+
self.layer_for_head = -1 if layer_for_head is None else layer_for_head
|
82 |
+
|
83 |
+
def to(self, device: str = ("cuda" if torch.cuda.is_available() else "cpu")):
|
84 |
+
self.whisper_model.to(device)
|
85 |
+
self.additional_decoder_block.to(device)
|
86 |
+
self.classifier.to(device)
|
87 |
+
super().to(device)
|
88 |
+
return self
|
89 |
+
|
90 |
+
def load_model(self, save_dir=None):
|
91 |
+
# load only the classifier and extra decoder layer (saved locally)
|
92 |
+
if save_dir is not None:
|
93 |
+
print('loading model from:', save_dir)
|
94 |
+
device = "cuda" if torch.cuda.is_available() else "cpu"
|
95 |
+
self.classifier.load_state_dict(
|
96 |
+
torch.load(
|
97 |
+
os.path.join(save_dir, "classifier.pt"),
|
98 |
+
weights_only=False,
|
99 |
+
map_location=torch.device(device),
|
100 |
+
)
|
101 |
+
)
|
102 |
+
self.additional_decoder_block.load_state_dict(
|
103 |
+
torch.load(
|
104 |
+
os.path.join(save_dir, "additional_decoder_block.pt"),
|
105 |
+
weights_only=False,
|
106 |
+
map_location=torch.device(device),
|
107 |
+
)
|
108 |
+
)
|
109 |
+
# read and load the layer_for_head.json
|
110 |
+
# the json format is {"layer_for_head": 9}
|
111 |
+
with open(os.path.join(save_dir, "metadata.json"), "r") as f:
|
112 |
+
metadata = json.load(f)
|
113 |
+
self.layer_for_head = metadata["layer_for_head"]
|
114 |
+
return
|
115 |
+
|
116 |
+
def train(self, mode: Optional[bool] = True):
|
117 |
+
# freeze whisper and train classifier
|
118 |
+
self.whisper_model.eval()
|
119 |
+
# mark whisper model requires grad false
|
120 |
+
for param in self.whisper_model.parameters():
|
121 |
+
param.requires_grad = False
|
122 |
+
for param in self.additional_decoder_block.parameters():
|
123 |
+
param.requires_grad = True
|
124 |
+
for param in self.classifier.parameters():
|
125 |
+
param.requires_grad = True
|
126 |
+
self.additional_decoder_block.train()
|
127 |
+
self.classifier.train()
|
128 |
+
|
129 |
+
def eval(self):
|
130 |
+
self.whisper_model.eval()
|
131 |
+
self.additional_decoder_block.eval()
|
132 |
+
self.classifier.eval()
|
133 |
+
|
134 |
+
def forward(
|
135 |
+
self,
|
136 |
+
input_features,
|
137 |
+
attention_mask=None,
|
138 |
+
decoder_input_ids=None,
|
139 |
+
labels_head=None,
|
140 |
+
whisper_labels=None,
|
141 |
+
):
|
142 |
+
device = "cuda" if torch.cuda.is_available() else "cpu"
|
143 |
+
self.whisper_model.eval()
|
144 |
+
|
145 |
+
# pass the inputs through the model
|
146 |
+
backbone_outputs = self.whisper_model(
|
147 |
+
input_features=input_features,
|
148 |
+
attention_mask=attention_mask,
|
149 |
+
decoder_input_ids=decoder_input_ids,
|
150 |
+
output_hidden_states=True,
|
151 |
+
labels=whisper_labels,
|
152 |
+
)
|
153 |
+
|
154 |
+
# Extract the hidden states of the last layer of the decoder
|
155 |
+
decoder_last_layer_hidden_states = backbone_outputs.decoder_hidden_states[
|
156 |
+
self.layer_for_head
|
157 |
+
].to(device)
|
158 |
+
|
159 |
+
# Extract the hidden states of the layer of the encoder who encapsulates best the prosodic features
|
160 |
+
layer_for_head_hidden_states = backbone_outputs.encoder_hidden_states[
|
161 |
+
self.layer_for_head
|
162 |
+
].to(device)
|
163 |
+
# Pass the decoder last hidden layers through the new head (decoder_block + lin cls)
|
164 |
+
|
165 |
+
additional_decoder_block_outputs = self.additional_decoder_block(
|
166 |
+
hidden_states=decoder_last_layer_hidden_states,
|
167 |
+
encoder_hidden_states=layer_for_head_hidden_states,
|
168 |
+
)
|
169 |
+
head_logits = self.classifier(additional_decoder_block_outputs[0].to(device))
|
170 |
+
|
171 |
+
# calculate softmax
|
172 |
+
head_probs = F.softmax(head_logits, dim=-1)
|
173 |
+
preds = head_probs.argmax(dim=-1).to(device)
|
174 |
+
if labels_head is not None:
|
175 |
+
preds = torch.where(
|
176 |
+
torch.isin(
|
177 |
+
labels_head, torch.tensor(list([-100])).to(device) # 50257, 50362,
|
178 |
+
),
|
179 |
+
torch.tensor(-100),
|
180 |
+
preds,
|
181 |
+
)
|
182 |
+
# Calculate custom loss if labels are provided
|
183 |
+
loss = None
|
184 |
+
if labels_head is not None:
|
185 |
+
# CrossEntropyLoss for the custom head
|
186 |
+
loss = self.loss_fct(
|
187 |
+
head_logits.reshape(-1, head_logits.size(-1)), labels_head.reshape(-1)
|
188 |
+
)
|
189 |
+
return CustomModelOutput(
|
190 |
+
logits=head_logits,
|
191 |
+
labels_head=labels_head,
|
192 |
+
whisper_logits=backbone_outputs.logits,
|
193 |
+
loss=loss,
|
194 |
+
preds=preds,
|
195 |
+
)
|
196 |
+
|
197 |
+
def generate(
|
198 |
+
self,
|
199 |
+
input_features,
|
200 |
+
max_length=128,
|
201 |
+
labels_head=None,
|
202 |
+
whisper_labels=None,
|
203 |
+
**generate_kwargs,
|
204 |
+
):
|
205 |
+
"""
|
206 |
+
Generate both the Whisper output and custom head output sequences in alignment.
|
207 |
+
"""
|
208 |
+
device = "cuda" if torch.cuda.is_available() else "cpu"
|
209 |
+
# Generate the Whisper output sequence
|
210 |
+
whisper_outputs = self.whisper_model.generate(
|
211 |
+
input_features=input_features,
|
212 |
+
max_length=max_length,
|
213 |
+
labels=whisper_labels,
|
214 |
+
do_sample=False,
|
215 |
+
**generate_kwargs,
|
216 |
+
)
|
217 |
+
|
218 |
+
# pass the inputs through the model
|
219 |
+
backbone_outputs = self.whisper_model(
|
220 |
+
input_features=input_features,
|
221 |
+
decoder_input_ids=whisper_outputs,
|
222 |
+
output_hidden_states=True,
|
223 |
+
)
|
224 |
+
|
225 |
+
# Extract the hidden states of the last layer of the decoder
|
226 |
+
decoder_last_layer_hidden_states = backbone_outputs.decoder_hidden_states[
|
227 |
+
self.layer_for_head
|
228 |
+
].to(device)
|
229 |
+
|
230 |
+
# Extract the hidden states of the last layer of the encoder
|
231 |
+
layer_for_head_hidden_states = backbone_outputs.encoder_hidden_states[
|
232 |
+
self.layer_for_head
|
233 |
+
].to(device)
|
234 |
+
# Pass the decoder last hidden layers through the new head (decoder_block + lin cls)
|
235 |
+
|
236 |
+
additional_decoder_block_outputs = self.additional_decoder_block(
|
237 |
+
hidden_states=decoder_last_layer_hidden_states,
|
238 |
+
encoder_hidden_states=layer_for_head_hidden_states,
|
239 |
+
)
|
240 |
+
head_logits = self.classifier(additional_decoder_block_outputs[0].to(device))
|
241 |
+
# calculate softmax
|
242 |
+
head_probs = F.softmax(head_logits, dim=-1)
|
243 |
+
preds = head_probs.argmax(dim=-1).to(device)
|
244 |
+
preds = torch.where(
|
245 |
+
torch.isin(
|
246 |
+
whisper_outputs, torch.tensor(list([50256])).to(device) # 50257, 50362,
|
247 |
+
),
|
248 |
+
torch.tensor(-100),
|
249 |
+
preds,
|
250 |
+
)
|
251 |
+
# preds_shifted = torch.cat((preds[:, 1:], preds[:, :1]), dim=1)
|
252 |
+
return preds
|
253 |
+
|
254 |
+
def generate_dual(
|
255 |
+
self,
|
256 |
+
input_features,
|
257 |
+
attention_mask=None,
|
258 |
+
max_length=200,
|
259 |
+
labels_head=None,
|
260 |
+
whisper_labels=None,
|
261 |
+
**generate_kwargs,
|
262 |
+
):
|
263 |
+
"""
|
264 |
+
Generate both the Whisper output and custom head output sequences in alignment.
|
265 |
+
"""
|
266 |
+
device = "cuda" if torch.cuda.is_available() else "cpu"
|
267 |
+
# Generate the Whisper output sequence
|
268 |
+
whisper_outputs = self.whisper_model.generate(
|
269 |
+
input_features=input_features,
|
270 |
+
attention_mask=attention_mask,
|
271 |
+
max_length=max_length,
|
272 |
+
labels=whisper_labels,
|
273 |
+
return_dict_in_generate=True,
|
274 |
+
**generate_kwargs,
|
275 |
+
)
|
276 |
+
|
277 |
+
# pass the inputs through the model
|
278 |
+
backbone_outputs = self.whisper_model(
|
279 |
+
input_features=input_features,
|
280 |
+
attention_mask=attention_mask,
|
281 |
+
decoder_input_ids=whisper_outputs.sequences,
|
282 |
+
output_hidden_states=True,
|
283 |
+
)
|
284 |
+
|
285 |
+
# Extract the hidden states of the last layer of the decoder
|
286 |
+
decoder_last_layer_hidden_states = backbone_outputs.decoder_hidden_states[
|
287 |
+
self.layer_for_head
|
288 |
+
].to(device)
|
289 |
+
|
290 |
+
# Extract the hidden states of the last layer of the encoder
|
291 |
+
layer_for_head_hidden_states = backbone_outputs.encoder_hidden_states[
|
292 |
+
self.layer_for_head
|
293 |
+
].to(device)
|
294 |
+
# Pass the decoder last hidden layers through the new head (decoder_block + lin cls)
|
295 |
+
|
296 |
+
additional_decoder_block_outputs = self.additional_decoder_block(
|
297 |
+
hidden_states=decoder_last_layer_hidden_states,
|
298 |
+
encoder_hidden_states=layer_for_head_hidden_states,
|
299 |
+
)
|
300 |
+
head_logits = self.classifier(additional_decoder_block_outputs[0].to(device))
|
301 |
+
head_probs = F.softmax(head_logits, dim=-1)
|
302 |
+
preds = head_probs.argmax(dim=-1).to(device)
|
303 |
+
preds = torch.where(
|
304 |
+
torch.isin(
|
305 |
+
whisper_outputs.sequences, torch.tensor(list([50256])).to(device) # 50257, 50362,
|
306 |
+
),
|
307 |
+
torch.tensor(-100),
|
308 |
+
preds,
|
309 |
+
)
|
310 |
+
return CustomModelOutput(
|
311 |
+
logits=head_logits,
|
312 |
+
head_preds=preds,
|
313 |
+
whisper_logits=whisper_outputs.logits,
|
314 |
+
preds=whisper_outputs.sequences
|
315 |
+
)
|
316 |
+
|
317 |
+
def __str__(self):
|
318 |
+
return "WhiStress"
|
whistress/weights/additional_decoder_block.pt
ADDED
@@ -0,0 +1,3 @@
|
|
|
|
|
|
|
|
|
1 |
+
version https://git-lfs.github.com/spec/v1
|
2 |
+
oid sha256:b7d440821c831364c5046e859843926120550a38143f89e1bace82a2ed03cc77
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3 |
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size 37809834
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whistress/weights/classifier.pt
ADDED
@@ -0,0 +1,3 @@
|
|
|
|
|
|
|
|
|
1 |
+
version https://git-lfs.github.com/spec/v1
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2 |
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oid sha256:599257b647cbca9fc21aac4ede87651cd43d03c3338e705bd59d919ee19ebc6f
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3 |
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size 4739176
|
whistress/weights/metadata.json
ADDED
@@ -0,0 +1,3 @@
|
|
|
|
|
|
|
|
|
1 |
+
{
|
2 |
+
"layer_for_head": 9
|
3 |
+
}
|