speech-test
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Browse files- README.md +96 -0
- config.json +108 -0
- preprocessor_config.json +9 -0
- pytorch_model.bin +3 -0
README.md
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---
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language: en
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datasets:
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- superb
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tags:
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- speech
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- audio
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- wav2vec2
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- audio-classification
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license: apache-2.0
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---
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# Wav2Vec2-Large for Keyword Spotting
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## Model description
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This is a ported version of
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[S3PRL's Wav2Vec2 for the SUPERB Keyword Spotting task](https://github.com/s3prl/s3prl/tree/master/s3prl/downstream/speech_commands).
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The base model is [wav2vec2-large-lv60](https://huggingface.co/facebook/wav2vec2-large-lv60), which is pretrained on 16kHz
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sampled speech audio. When using the model make sure that your speech input is also sampled at 16Khz.
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For more information refer to [SUPERB: Speech processing Universal PERformance Benchmark](https://arxiv.org/abs/2105.01051)
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## Task and dataset description
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Keyword Spotting (KS) detects preregistered keywords by classifying utterances into a predefined set of
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words. The task is usually performed on-device for the fast response time. Thus, accuracy, model size, and
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inference time are all crucial. SUPERB uses the widely used
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[Speech Commands dataset v1.0](https://www.tensorflow.org/datasets/catalog/speech_commands) for the task.
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The dataset consists of ten classes of keywords, a class for silence, and an unknown class to include the
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false positive.
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For the original model's training and evaluation instructions refer to the
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[S3PRL downstream task README](https://github.com/s3prl/s3prl/tree/master/s3prl/downstream#ks-keyword-spotting).
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## Usage examples
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You can use the model via the Audio Classification pipeline:
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```python
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from datasets import load_dataset
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from transformers import pipeline
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dataset = load_dataset("anton-l/superb_demo", "ks", split="test")
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classifier = pipeline("audio-classification", model="superb/wav2vec2-large-superb-ks")
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labels = classifier(dataset[0]["file"], top_k=5)
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```
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Or use the model directly:
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```python
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import torch
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from datasets import load_dataset
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from transformers import Wav2Vec2ForSequenceClassification, Wav2Vec2FeatureExtractor
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from torchaudio.sox_effects import apply_effects_file
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effects = [["channels", "1"], ["rate", "16000"], ["gain", "-3.0"]]
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def map_to_array(example):
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speech, _ = apply_effects_file(example["file"], effects)
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example["speech"] = speech.squeeze(0).numpy()
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return example
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# load a demo dataset and read audio files
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dataset = load_dataset("anton-l/superb_demo", "ks", split="test")
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dataset = dataset.map(map_to_array)
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model = Wav2Vec2ForSequenceClassification.from_pretrained("superb/wav2vec2-large-superb-ks")
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feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained("superb/wav2vec2-large-superb-ks")
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# compute attention masks and normalize the waveform if needed
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inputs = feature_extractor(dataset[:4]["speech"], sampling_rate=16000, padding=True, return_tensors="pt")
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logits = model(**inputs).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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labels = [model.config.id2label[_id] for _id in predicted_ids.tolist()]
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```
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## Eval results
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The evaluation metric is accuracy.
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| | **s3prl** | **transformers** |
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|--------|-----------|------------------|
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|**test**| `0.9666` | `N/A` |
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### BibTeX entry and citation info
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```bibtex
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@article{yang2021superb,
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title={SUPERB: Speech processing Universal PERformance Benchmark},
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author={Yang, Shu-wen and Chi, Po-Han and Chuang, Yung-Sung and Lai, Cheng-I Jeff and Lakhotia, Kushal and Lin, Yist Y and Liu, Andy T and Shi, Jiatong and Chang, Xuankai and Lin, Guan-Ting and others},
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journal={arXiv preprint arXiv:2105.01051},
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year={2021}
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}
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```
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config.json
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{
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"_name_or_path": "facebook/wav2vec2-large-lv60",
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"activation_dropout": 0.1,
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"apply_spec_augment": true,
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"architectures": [
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"Wav2Vec2ForSequenceClassification"
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],
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"attention_dropout": 0.1,
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"bos_token_id": 1,
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"classifier_proj_size": 256,
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"codevector_dim": 768,
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"contrastive_logits_temperature": 0.1,
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"conv_bias": true,
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"conv_dim": [
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512,
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512,
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512,
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512,
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512,
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512,
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512
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],
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"conv_kernel": [
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10,
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3,
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3,
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3,
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2,
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2
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],
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"conv_stride": [
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5,
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2,
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],
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"ctc_loss_reduction": "sum",
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"ctc_zero_infinity": false,
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"diversity_loss_weight": 0.1,
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"do_stable_layer_norm": true,
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"eos_token_id": 2,
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"feat_extract_activation": "gelu",
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"feat_extract_dropout": 0.0,
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"feat_extract_norm": "layer",
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"feat_proj_dropout": 0.1,
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"feat_quantizer_dropout": 0.0,
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"final_dropout": 0.1,
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"gradient_checkpointing": false,
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"hidden_act": "gelu",
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"hidden_dropout": 0.1,
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"hidden_dropout_prob": 0.1,
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"hidden_size": 1024,
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"id2label": {
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"0": "yes",
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"1": "no",
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"2": "up",
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"3": "down",
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"4": "left",
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"5": "right",
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"6": "on",
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"7": "off",
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"8": "stop",
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"9": "go",
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"10": "_unknown_",
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"11": "_silence_"
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},
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"initializer_range": 0.02,
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"intermediate_size": 4096,
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"label2id": {
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"_silence_": 11,
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"_unknown_": 10,
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"down": 3,
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"go": 9,
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"left": 4,
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"no": 1,
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"off": 7,
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"on": 6,
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"right": 5,
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"stop": 8,
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"up": 2,
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"yes": 0
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},
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"layer_norm_eps": 1e-05,
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"layerdrop": 0.1,
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"mask_feature_length": 10,
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"mask_feature_prob": 0.0,
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"mask_time_length": 10,
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"mask_time_prob": 0.05,
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"model_type": "wav2vec2",
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"num_attention_heads": 16,
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"num_codevector_groups": 2,
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"num_codevectors_per_group": 320,
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"num_conv_pos_embedding_groups": 16,
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"num_conv_pos_embeddings": 128,
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"num_feat_extract_layers": 7,
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"num_hidden_layers": 24,
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"num_negatives": 100,
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"pad_token_id": 0,
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"proj_codevector_dim": 768,
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"torch_dtype": "float32",
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"transformers_version": "4.11.0.dev0",
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"use_weighted_layer_sum": true,
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"vocab_size": 32
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}
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preprocessor_config.json
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{
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"do_normalize": false,
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"feature_extractor_type": "Wav2Vec2FeatureExtractor",
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"feature_size": 1,
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"padding_side": "right",
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"padding_value": 0.0,
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"return_attention_mask": true,
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"sampling_rate": 16000
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}
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pytorch_model.bin
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version https://git-lfs.github.com/spec/v1
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oid sha256:be36f743cf72df29a247523f697f35fab379511ea935547fda6fa79ca9c9e874
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size 1262980715
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