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Training in progress, step 500

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Files changed (6) hide show
  1. .gitignore +1 -0
  2. config.json +107 -0
  3. eval.py +128 -0
  4. preprocessor_config.json +9 -0
  5. pytorch_model.bin +3 -0
  6. training_args.bin +3 -0
.gitignore ADDED
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+ checkpoint-*/
config.json ADDED
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+ {
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+ "_name_or_path": "facebook/wav2vec2-xls-r-300m",
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+ "activation_dropout": 0.1,
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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": 0.0,
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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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+ 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": 0.0,
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+ "feat_quantizer_dropout": 0.0,
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+ "final_dropout": 0.0,
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+ "hidden_act": "gelu",
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+ "hidden_dropout": 0.0,
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+ "hidden_size": 1024,
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+ "initializer_range": 0.02,
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+ "intermediate_size": 4096,
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+ "layer_norm_eps": 1e-05,
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+ "layerdrop": 0.0,
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+ "mask_feature_length": 64,
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+ "mask_feature_min_masks": 0,
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+ "mask_feature_prob": 0.25,
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+ "mask_time_length": 10,
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+ "mask_time_min_masks": 2,
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+ "mask_time_prob": 0.75,
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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": 24,
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+ "num_negatives": 100,
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+ "output_hidden_size": 1024,
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+ "pad_token_id": 43,
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+ "proj_codevector_dim": 768,
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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": 46,
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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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+ import argparse
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+ import re
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+ from typing import Dict
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+
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+ from datasets import Audio, Dataset, load_dataset, load_metric
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+
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+ from transformers import AutoFeatureExtractor, pipeline
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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 = f"WER: {wer_result}\n" f"CER: {cer_result}"
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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) -> 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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+ chars_to_ignore_regex = '[,?.!\-\;\:"β€œ%β€˜β€οΏ½β€”β€™β€¦β€“]' # noqa: W605 IMPORTANT: this should correspond to the chars that were ignored during training
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+
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+ text = re.sub(chars_to_ignore_regex, "", text.lower())
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+
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+ # In addition, we can normalize the target text, e.g. removing new lines characters etc...
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+ # note that order is important here!
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+ token_sequences_to_ignore = ["\n\n", "\n", " ", " "]
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+
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+ for t in token_sequences_to_ignore:
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+ text = " ".join(text.split(t))
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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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+ asr = pipeline("automatic-speech-recognition", model=args.model_id)
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+
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+ # map function to decode audio
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+ def map_to_pred(batch):
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+ prediction = asr(
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+ 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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+
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+ batch["prediction"] = prediction["text"]
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+ batch["target"] = normalize_text(batch["sentence"])
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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 πŸ€— Transformers"
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+ )
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+ parser.add_argument(
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+ "--dataset",
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+ type=str,
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+ required=True,
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+ help="Dataset name to evaluate the `model_id`. Should be loadable with πŸ€— 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("--split", type=str, required=True, help="Split of the dataset. *E.g.* `'test'`")
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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 5 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 1 second."
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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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+ args = parser.parse_args()
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+
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+ main(args)
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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+ "return_attention_mask": true,
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+ "sampling_rate": 16000
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+ }
pytorch_model.bin ADDED
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training_args.bin ADDED
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