jonatasgrosman
commited on
Commit
•
14359da
1
Parent(s):
bca42c0
add model
Browse files- README.md +54 -0
- alphabet.json +1 -0
- config.json +107 -0
- eval.py +146 -0
- language_model/attrs.json +1 -0
- language_model/lm.binary +3 -0
- language_model/unigrams.txt +0 -0
- preprocessor_config.json +10 -0
- pytorch_model.bin +3 -0
- special_tokens_map.json +1 -0
- vocab.json +1 -0
README.md
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---
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language:
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- en
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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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- en
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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 - English
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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: en
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metrics:
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- name: Test WER
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type: wer
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value:
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- name: Test CER
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type: cer
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value:
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- name: Test WER (+LM)
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type: wer
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value:
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- name: Test CER (+LM)
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type: cer
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value:
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---
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# XLS-R-1B-English
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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 - EN dataset.
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## Evaluation Commands
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1. To evaluate on `mozilla-foundation/common_voice_8_0` with split `test`
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```bash
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python eval.py --model_id jonatasgrosman/wav2vec2-xls-r-1b-english --dataset mozilla-foundation/common_voice_8_0 --config en --split test
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```
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2. To evaluate on `speech-recognition-community-v2/dev_data`
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```bash
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python eval.py --model_id jonatasgrosman/wav2vec2-xls-r-1b-english --dataset speech-recognition-community-v2/dev_data --config en --split validation --chunk_length_s 5.0 --stride_length_s 1.0
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```
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alphabet.json
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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"], "is_bpe": false}
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config.json
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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": 33,
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"xvector_output_dim": 512
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}
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eval.py
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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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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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log_outputs = args.log_outputs
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dataset_id = "_".join(args.dataset.split("/") + [args.config, args.split])
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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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# 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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# 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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with open(f"{dataset_id}_eval_results.txt", "w") as f:
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f.write(result_str)
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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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with open(pred_file, "w") as p, open(target_file, "w") as t:
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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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result.map(write_to_file, with_indices=True)
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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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text = text.lower() if to_lower else text.upper()
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text = re.sub(invalid_chars_regex, " ", text)
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text = re.sub("\s+", " ", text).strip()
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return text
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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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# for testing: only process the first two examples as a test
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# dataset = dataset.select(range(10))
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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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# resample audio
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dataset = dataset.cast_column("audio", Audio(sampling_rate=sampling_rate))
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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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# 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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# 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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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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# run inference on all examples
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result = dataset.map(map_to_pred, remove_columns=dataset.column_names)
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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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if __name__ == "__main__":
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parser = argparse.ArgumentParser()
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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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)
|
135 |
+
parser.add_argument(
|
136 |
+
"--log_outputs", action='store_true', help="If defined, write outputs to log file for analysis."
|
137 |
+
)
|
138 |
+
parser.add_argument(
|
139 |
+
"--device",
|
140 |
+
type=int,
|
141 |
+
default=None,
|
142 |
+
help="The device to run the pipeline on. -1 for CPU (default), 0 for the first GPU and so on.",
|
143 |
+
)
|
144 |
+
args = parser.parse_args()
|
145 |
+
|
146 |
+
main(args)
|
language_model/attrs.json
ADDED
@@ -0,0 +1 @@
|
|
|
|
|
1 |
+
{"alpha": 0.5, "beta": 1.5, "unk_score_offset": -10.0, "score_boundary": true}
|
language_model/lm.binary
ADDED
@@ -0,0 +1,3 @@
|
|
|
|
|
|
|
|
|
1 |
+
version https://git-lfs.github.com/spec/v1
|
2 |
+
oid sha256:47e16abf6384ebd1b3395144330b60710dc43f3d16c4b2b4794071cd117230e5
|
3 |
+
size 862913451
|
language_model/unigrams.txt
ADDED
The diff for this file is too large to render.
See raw diff
|
|
preprocessor_config.json
ADDED
@@ -0,0 +1,10 @@
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
1 |
+
{
|
2 |
+
"do_normalize": true,
|
3 |
+
"feature_extractor_type": "Wav2Vec2FeatureExtractor",
|
4 |
+
"feature_size": 1,
|
5 |
+
"padding_side": "right",
|
6 |
+
"padding_value": 0,
|
7 |
+
"processor_class": "Wav2Vec2ProcessorWithLM",
|
8 |
+
"return_attention_mask": true,
|
9 |
+
"sampling_rate": 16000
|
10 |
+
}
|
pytorch_model.bin
ADDED
@@ -0,0 +1,3 @@
|
|
|
|
|
|
|
|
|
1 |
+
version https://git-lfs.github.com/spec/v1
|
2 |
+
oid sha256:a89f9eaf110691dbb7d717a7dc1ce8a23825ff31404da81abebcda49c8a8ddd3
|
3 |
+
size 3850481841
|
special_tokens_map.json
ADDED
@@ -0,0 +1 @@
|
|
|
|
|
1 |
+
{"bos_token": "<s>", "eos_token": "</s>", "unk_token": "<unk>", "pad_token": "<pad>"}
|
vocab.json
ADDED
@@ -0,0 +1 @@
|
|
|
|
|
1 |
+
{"<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}
|