jonatasgrosman commited on
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README.md ADDED
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+ ---
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+ language:
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+ - de
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+ license: apache-2.0
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+ tags:
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+ - automatic-speech-recognition
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+ - mozilla-foundation/common_voice_8_0
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+ - de
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+ - robust-speech-event
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+ datasets:
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+ - mozilla-foundation/common_voice_8_0
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+ model-index:
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+ - name: XLS-R-1B - German
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+ results:
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+ - task:
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+ name: Automatic Speech Recognition
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+ type: automatic-speech-recognition
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+ dataset:
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+ name: Common Voice 8
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+ type: mozilla-foundation/common_voice_8_0
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+ args: de
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+ metrics:
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+ - name: Test WER
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+ type: wer
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+ value: 10.95
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+ - name: Test CER
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+ type: cer
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+ value: 2.72
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+ - name: Test WER (+LM)
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+ type: wer
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+ value: 9.32
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+ - name: Test CER (+LM)
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+ type: cer
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+ value: 2.39
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+ ---
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+
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+ # XLS-R-1B-GERMAN
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+
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+ This model is a fine-tuned version of [facebook/wav2vec2-xls-r-1b](https://huggingface.co/facebook/wav2vec2-xls-r-1b) on the MOZILLA-FOUNDATION/COMMON_VOICE_8_0 - DE dataset.
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+
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+
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+ ## Evaluation Commands
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+
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+ 1. To evaluate on `mozilla-foundation/common_voice_8_0` with split `test`
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+
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+ ```bash
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+ python eval.py --model_id jonatasgrosman/wav2vec2-xls-r-1b-german --dataset mozilla-foundation/common_voice_8_0 --config de --split test
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+ ```
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+
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+ 2. To evaluate on `speech-recognition-community-v2/dev_data`
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+
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+ ```bash
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+ python eval.py --model_id jonatasgrosman/wav2vec2-xls-r-1b-german --dataset speech-recognition-community-v2/dev_data --config de --split validation --chunk_length_s 5.0 --stride_length_s 1.0
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+ ```
alphabet.json ADDED
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+ {"labels": ["", "<s>", "</s>", "\u2047", " ", "'", "-", "a", "b", "c", "d", "e", "f", "g", "h", "i", "j", "k", "l", "m", "n", "o", "p", "q", "r", "s", "t", "u", "v", "w", "x", "y", "z", "\u00df", "\u00e4", "\u00f6", "\u00fc"], "is_bpe": false}
config.json ADDED
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+ {
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+ "_name_or_path": "facebook/wav2vec2-xls-r-1b",
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+ "activation_dropout": 0.05,
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+ "adapter_kernel_size": 3,
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+ "adapter_stride": 2,
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+ "add_adapter": false,
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+ "apply_spec_augment": true,
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+ "architectures": [
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+ "Wav2Vec2ForCTC"
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+ ],
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+ "attention_dropout": 0.05,
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+ "bos_token_id": 1,
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+ "classifier_proj_size": 256,
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+ "codevector_dim": 1024,
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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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+ 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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+ 2,
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+ 2,
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+ 2,
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+ 2,
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+ 2
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+ ],
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+ "ctc_loss_reduction": "mean",
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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": 0.05,
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+ "feat_quantizer_dropout": 0.0,
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+ "final_dropout": 0.05,
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+ "hidden_act": "gelu",
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+ "hidden_dropout": 0.05,
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+ "hidden_size": 1280,
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+ "initializer_range": 0.02,
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+ "intermediate_size": 5120,
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+ "layer_norm_eps": 1e-05,
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+ "layerdrop": 0.05,
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+ "mask_feature_length": 10,
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+ "mask_feature_min_masks": 0,
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+ "mask_feature_prob": 0.0,
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+ "mask_time_length": 10,
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+ "mask_time_min_masks": 2,
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+ "mask_time_prob": 0.05,
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+ "model_type": "wav2vec2",
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+ "num_adapter_layers": 3,
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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": 48,
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+ "num_negatives": 100,
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+ "output_hidden_size": 1280,
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+ "pad_token_id": 0,
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+ "proj_codevector_dim": 1024,
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+ "tdnn_dilation": [
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+ 1,
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+ 2,
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+ 3,
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+ 1,
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+ 1
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+ ],
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+ "tdnn_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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+ 1500
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+ ],
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+ "tdnn_kernel": [
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+ 5,
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+ 3,
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+ 3,
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+ 1,
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+ 1
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+ ],
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+ "torch_dtype": "float32",
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+ "transformers_version": "4.16.0.dev0",
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+ "use_weighted_layer_sum": false,
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+ "vocab_size": 37,
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+ "xvector_output_dim": 512
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+ }
eval.py ADDED
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+ #!/usr/bin/env python3
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+ from datasets import load_dataset, load_metric, Audio, Dataset
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+ from transformers import pipeline, AutoFeatureExtractor, AutoTokenizer
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+ import re
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+ import torch
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+ import argparse
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+ from typing import Dict
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+
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+
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+ def log_results(result: Dataset, args: Dict[str, str]):
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+ """ DO NOT CHANGE. This function computes and logs the result metrics. """
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+
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+ log_outputs = args.log_outputs
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+ dataset_id = "_".join(args.dataset.split("/") + [args.config, args.split])
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+
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+ # load metric
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+ wer = load_metric("wer")
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+ cer = load_metric("cer")
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+
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+ # compute metrics
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+ wer_result = wer.compute(references=result["target"], predictions=result["prediction"])
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+ cer_result = cer.compute(references=result["target"], predictions=result["prediction"])
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+
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+ # print & log results
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+ result_str = (
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+ f"WER: {wer_result}\n"
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+ f"CER: {cer_result}"
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+ )
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+ print(result_str)
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+
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+ with open(f"{dataset_id}_eval_results.txt", "w") as f:
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+ f.write(result_str)
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+
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+ # log all results in text file. Possibly interesting for analysis
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+ if log_outputs is not None:
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+ pred_file = f"log_{dataset_id}_predictions.txt"
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+ target_file = f"log_{dataset_id}_targets.txt"
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+
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+ with open(pred_file, "w") as p, open(target_file, "w") as t:
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+
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+ # mapping function to write output
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+ def write_to_file(batch, i):
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+ p.write(f"{i}" + "\n")
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+ p.write(batch["prediction"] + "\n")
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+ t.write(f"{i}" + "\n")
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+ t.write(batch["target"] + "\n")
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+
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+ result.map(write_to_file, with_indices=True)
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+
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+
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+ def normalize_text(text: str, invalid_chars_regex: str, to_lower: bool) -> str:
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+ """ DO ADAPT FOR YOUR USE CASE. this function normalizes the target text. """
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+
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+ text = text.lower() if to_lower else text.upper()
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+
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+ text = re.sub(invalid_chars_regex, " ", text)
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+
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+ text = re.sub("\s+", " ", text).strip()
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+
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+ return text
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+
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+
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+ def main(args):
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+ # load dataset
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+ dataset = load_dataset(args.dataset, args.config, split=args.split, use_auth_token=True)
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+
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+ # for testing: only process the first two examples as a test
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+ # dataset = dataset.select(range(10))
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+
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+ # load processor
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+ feature_extractor = AutoFeatureExtractor.from_pretrained(args.model_id)
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+ sampling_rate = feature_extractor.sampling_rate
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+
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+ # resample audio
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+ dataset = dataset.cast_column("audio", Audio(sampling_rate=sampling_rate))
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+
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+ # load eval pipeline
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+ if args.device is None:
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+ args.device = 0 if torch.cuda.is_available() else -1
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+ asr = pipeline("automatic-speech-recognition", model=args.model_id, device=args.device)
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+
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+ # build normalizer config
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+ tokenizer = AutoTokenizer.from_pretrained(args.model_id)
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+ tokens = [x for x in tokenizer.convert_ids_to_tokens(range(0, tokenizer.vocab_size))]
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+ special_tokens = [
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+ tokenizer.pad_token, tokenizer.word_delimiter_token,
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+ tokenizer.unk_token, tokenizer.bos_token,
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+ tokenizer.eos_token,
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+ ]
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+ non_special_tokens = [x for x in tokens if x not in special_tokens]
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+ invalid_chars_regex = f"[^\s{re.escape(''.join(set(non_special_tokens)))}]"
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+ normalize_to_lower = False
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+ for token in non_special_tokens:
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+ if token.isalpha() and token.islower():
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+ normalize_to_lower = True
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+ break
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+
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+ # map function to decode audio
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+ def map_to_pred(batch, args=args, asr=asr, invalid_chars_regex=invalid_chars_regex, normalize_to_lower=normalize_to_lower):
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+ prediction = asr(batch["audio"]["array"], chunk_length_s=args.chunk_length_s, stride_length_s=args.stride_length_s)
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+
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+ batch["prediction"] = prediction["text"]
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+ batch["target"] = normalize_text(batch["sentence"], invalid_chars_regex, normalize_to_lower)
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+ return batch
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+
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+ # run inference on all examples
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+ result = dataset.map(map_to_pred, remove_columns=dataset.column_names)
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+
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+ # compute and log_results
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+ # do not change function below
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+ log_results(result, args)
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+
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+
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+ if __name__ == "__main__":
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+ parser = argparse.ArgumentParser()
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+
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+ parser.add_argument(
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+ "--model_id", type=str, required=True, help="Model identifier. Should be loadable with 🤗 Transformers"
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+ )
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+ parser.add_argument(
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+ "--dataset", type=str, required=True, help="Dataset name to evaluate the `model_id`. Should be loadable with 🤗 Datasets"
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+ )
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+ parser.add_argument(
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+ "--config", type=str, required=True, help="Config of the dataset. *E.g.* `'en'` for Common Voice"
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+ )
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+ parser.add_argument(
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+ "--split", type=str, required=True, help="Split of the dataset. *E.g.* `'test'`"
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+ )
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+ parser.add_argument(
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+ "--chunk_length_s", type=float, default=None, help="Chunk length in seconds. Defaults to None. For long audio files a good value would be 5.0 seconds."
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+ )
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+ parser.add_argument(
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+ "--stride_length_s", type=float, default=None, help="Stride of the audio chunks. Defaults to None. For long audio files a good value would be 1.0 seconds."
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+ )
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+ parser.add_argument(
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+ "--log_outputs", action='store_true', help="If defined, write outputs to log file for analysis."
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+ )
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+ parser.add_argument(
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+ "--device",
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+ type=int,
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+ default=None,
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+ help="The device to run the pipeline on. -1 for CPU (default), 0 for the first GPU and so on.",
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+ )
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+ args = parser.parse_args()
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+
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+ main(args)
language_model/attrs.json ADDED
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+ {"alpha": 0.5, "beta": 1.5, "unk_score_offset": -10.0, "score_boundary": true}
language_model/lm.binary ADDED
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+ size 302214597
language_model/unigrams.txt ADDED
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log_mozilla-foundation_common_voice_8_0_de_test_predictions.txt ADDED
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log_mozilla-foundation_common_voice_8_0_de_test_predictions_greedy.txt ADDED
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log_mozilla-foundation_common_voice_8_0_de_test_targets.txt ADDED
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log_mozilla-foundation_common_voice_8_0_de_test_targets_greedy.txt ADDED
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mozilla-foundation_common_voice_8_0_de_test_eval_results.txt ADDED
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+ WER: 0.09321675555494932
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+ CER: 0.023974465348382053
mozilla-foundation_common_voice_8_0_de_test_eval_results_greedy.txt ADDED
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+ WER: 0.10953718985718768
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+ CER: 0.02727745510990778
preprocessor_config.json ADDED
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+ {
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+ "do_normalize": true,
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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,
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+ "processor_class": "Wav2Vec2ProcessorWithLM",
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+ "return_attention_mask": true,
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+ "sampling_rate": 16000
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+ }
pytorch_model.bin ADDED
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special_tokens_map.json ADDED
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+ {"bos_token": "<s>", "eos_token": "</s>", "unk_token": "<unk>", "pad_token": "<pad>"}
vocab.json ADDED
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+ {"<pad>": 0, "<s>": 1, "</s>": 2, "<unk>": 3, "|": 4, "'": 5, "-": 6, "a": 7, "b": 8, "c": 9, "d": 10, "e": 11, "f": 12, "g": 13, "h": 14, "i": 15, "j": 16, "k": 17, "l": 18, "m": 19, "n": 20, "o": 21, "p": 22, "q": 23, "r": 24, "s": 25, "t": 26, "u": 27, "v": 28, "w": 29, "x": 30, "y": 31, "z": 32, "ß": 33, "ä": 34, "ö": 35, "ü": 36}