add training scripts

ds_config.json

@@ -0,0 +1,50 @@
+{
+    "fp16": {
+        "enabled": "auto",
+        "loss_scale": 0,
+        "loss_scale_window": 1000,
+        "initial_scale_power": 16,
+        "hysteresis": 2,
+        "min_loss_scale": 1
+    },
+
+    "optimizer": {
+        "type": "AdamW",
+        "params": {
+            "lr": "auto",
+            "betas": "auto",
+            "eps": "auto",
+            "weight_decay": "auto"
+        }
+    },
+
+   "scheduler": {
+         "type": "WarmupDecayLR",
+         "params": {
+             "last_batch_iteration": -1,
+             "total_num_steps": "auto",
+             "warmup_min_lr": "auto",
+             "warmup_max_lr": "auto",
+             "warmup_num_steps": "auto"
+         }
+     },
+
+    "zero_optimization": {
+        "stage": 2,
+        "offload_optimizer": {
+            "device": "cpu",
+            "pin_memory": true
+        },
+        "allgather_partitions": true,
+        "allgather_bucket_size": 2e8,
+        "overlap_comm": true,
+        "reduce_scatter": true,
+        "reduce_bucket_size": 2e8,
+        "contiguous_gradients": true
+    },
+
+    "gradient_accumulation_steps": "auto",
+    "gradient_clipping": "auto",
+    "train_batch_size": "auto",
+    "train_micro_batch_size_per_gpu": "auto"
+}

run.sh

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+deepspeed run_audio_classification.py \
+    --deepspeed ds_config.json \
+    --model_name_or_path openai/whisper-medium \
+    --dataset_name google/xtreme_s \
+    --dataset_config_name fleurs.all \
+    --output_dir ./ \
+    --overwrite_output_dir \
+    --remove_unused_columns False \
+    --do_train \
+    --do_eval \
+    --fp16 \
+    --learning_rate 3e-5 \
+    --max_length_seconds 30 \
+    --label_column_name lang_id \
+    --attention_mask False \
+    --warmup_ratio 0.1 \
+    --num_train_epochs 3 \
+    --per_device_train_batch_size 16 \
+    --gradient_accumulation_steps 2 \
+    --gradient_checkpointing True \
+    --per_device_eval_batch_size 32 \
+    --dataloader_num_workers 8 \
+    --logging_strategy steps \
+    --logging_steps 25 \
+    --evaluation_strategy epoch \
+    --save_strategy epoch \
+    --load_best_model_at_end True \
+    --metric_for_best_model accuracy \
+    --seed 0 \
+    --freeze_feature_encoder False \
+    --push_to_hub

run_audio_classification.py

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+#!/usr/bin/env python
+# coding=utf-8
+# Copyright 2021 The HuggingFace Inc. team. All rights reserved.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+import logging
+import os
+import sys
+import warnings
+from dataclasses import dataclass, field
+from random import randint
+from typing import Optional
+
+import datasets
+import evaluate
+import numpy as np
+from datasets import DatasetDict, load_dataset
+
+import transformers
+from transformers import (
+    AutoConfig,
+    AutoFeatureExtractor,
+    AutoModelForAudioClassification,
+    HfArgumentParser,
+    Trainer,
+    TrainingArguments,
+    set_seed,
+)
+from transformers.trainer_utils import get_last_checkpoint
+from transformers.utils import check_min_version, send_example_telemetry
+from transformers.utils.versions import require_version
+
+
+logger = logging.getLogger(__name__)
+
+# Will error if the minimal version of Transformers is not installed. Remove at your own risks.
+check_min_version("4.27.0.dev0")
+
+require_version("datasets>=1.14.0", "To fix: pip install -r examples/pytorch/audio-classification/requirements.txt")
+
+
+def random_subsample(wav: np.ndarray, max_length: float, sample_rate: int = 16000):
+    """Randomly sample chunks of `max_length` seconds from the input audio"""
+    sample_length = int(round(sample_rate * max_length))
+    if len(wav) <= sample_length:
+        return wav
+    random_offset = randint(0, len(wav) - sample_length - 1)
+    return wav[random_offset : random_offset + sample_length]
+
+
+@dataclass
+class DataTrainingArguments:
+    """
+    Arguments pertaining to what data we are going to input our model for training and eval.
+    Using `HfArgumentParser` we can turn this class
+    into argparse arguments to be able to specify them on
+    the command line.
+    """
+
+    dataset_name: Optional[str] = field(default=None, metadata={"help": "Name of a dataset from the datasets package"})
+    dataset_config_name: Optional[str] = field(
+        default=None, metadata={"help": "The configuration name of the dataset to use (via the datasets library)."}
+    )
+    train_file: Optional[str] = field(
+        default=None, metadata={"help": "A file containing the training audio paths and labels."}
+    )
+    eval_file: Optional[str] = field(
+        default=None, metadata={"help": "A file containing the validation audio paths and labels."}
+    )
+    train_split_name: str = field(
+        default="train",
+        metadata={
+            "help": "The name of the training data set split to use (via the datasets library). Defaults to 'train'"
+        },
+    )
+    eval_split_name: str = field(
+        default="validation",
+        metadata={
+            "help": (
+                "The name of the training data set split to use (via the datasets library). Defaults to 'validation'"
+            )
+        },
+    )
+    audio_column_name: str = field(
+        default="audio",
+        metadata={"help": "The name of the dataset column containing the audio data. Defaults to 'audio'"},
+    )
+    label_column_name: str = field(
+        default="label", metadata={"help": "The name of the dataset column containing the labels. Defaults to 'label'"}
+    )
+    max_train_samples: Optional[int] = field(
+        default=None,
+        metadata={
+            "help": (
+                "For debugging purposes or quicker training, truncate the number of training examples to this "
+                "value if set."
+            )
+        },
+    )
+    max_eval_samples: Optional[int] = field(
+        default=None,
+        metadata={
+            "help": (
+                "For debugging purposes or quicker training, truncate the number of evaluation examples to this "
+                "value if set."
+            )
+        },
+    )
+    max_length_seconds: float = field(
+        default=20,
+        metadata={"help": "Audio clips will be randomly cut to this length during training if the value is set."},
+    )
+
+
+@dataclass
+class ModelArguments:
+    """
+    Arguments pertaining to which model/config/tokenizer we are going to fine-tune from.
+    """
+
+    model_name_or_path: str = field(
+        default="facebook/wav2vec2-base",
+        metadata={"help": "Path to pretrained model or model identifier from huggingface.co/models"},
+    )
+    config_name: Optional[str] = field(
+        default=None, metadata={"help": "Pretrained config name or path if not the same as model_name"}
+    )
+    cache_dir: Optional[str] = field(
+        default=None, metadata={"help": "Where do you want to store the pretrained models downloaded from the Hub"}
+    )
+    model_revision: str = field(
+        default="main",
+        metadata={"help": "The specific model version to use (can be a branch name, tag name or commit id)."},
+    )
+    feature_extractor_name: Optional[str] = field(
+        default=None, metadata={"help": "Name or path of preprocessor config."}
+    )
+    freeze_feature_encoder: bool = field(
+        default=True, metadata={"help": "Whether to freeze the feature encoder layers of the model."}
+    )
+    attention_mask: bool = field(
+        default=True, metadata={"help": "Whether to generate an attention mask in the feature extractor."}
+    )
+    use_auth_token: bool = field(
+        default=False,
+        metadata={
+            "help": (
+                "Will use the token generated when running `huggingface-cli login` (necessary to use this script "
+                "with private models)."
+            )
+        },
+    )
+    freeze_feature_extractor: Optional[bool] = field(
+        default=None, metadata={"help": "Whether to freeze the feature extractor layers of the model."}
+    )
+    ignore_mismatched_sizes: bool = field(
+        default=False,
+        metadata={"help": "Will enable to load a pretrained model whose head dimensions are different."},
+    )
+
+    def __post_init__(self):
+        if not self.freeze_feature_extractor and self.freeze_feature_encoder:
+            warnings.warn(
+                "The argument `--freeze_feature_extractor` is deprecated and "
+                "will be removed in a future version. Use `--freeze_feature_encoder`"
+                "instead. Setting `freeze_feature_encoder==True`.",
+                FutureWarning,
+            )
+        if self.freeze_feature_extractor and not self.freeze_feature_encoder:
+            raise ValueError(
+                "The argument `--freeze_feature_extractor` is deprecated and "
+                "should not be used in combination with `--freeze_feature_encoder`."
+                "Only make use of `--freeze_feature_encoder`."
+            )
+
+
+def main():
+    # See all possible arguments in src/transformers/training_args.py
+    # or by passing the --help flag to this script.
+    # We now keep distinct sets of args, for a cleaner separation of concerns.
+
+    parser = HfArgumentParser((ModelArguments, DataTrainingArguments, TrainingArguments))
+    if len(sys.argv) == 2 and sys.argv[1].endswith(".json"):
+        # If we pass only one argument to the script and it's the path to a json file,
+        # let's parse it to get our arguments.
+        model_args, data_args, training_args = parser.parse_json_file(json_file=os.path.abspath(sys.argv[1]))
+    else:
+        model_args, data_args, training_args = parser.parse_args_into_dataclasses()
+
+    # Sending telemetry. Tracking the example usage helps us better allocate resources to maintain them. The
+    # information sent is the one passed as arguments along with your Python/PyTorch versions.
+    send_example_telemetry("run_audio_classification", model_args, data_args)
+
+    # Setup logging
+    logging.basicConfig(
+        format="%(asctime)s - %(levelname)s - %(name)s - %(message)s",
+        datefmt="%m/%d/%Y %H:%M:%S",
+        handlers=[logging.StreamHandler(sys.stdout)],
+    )
+
+    if training_args.should_log:
+        # The default of training_args.log_level is passive, so we set log level at info here to have that default.
+        transformers.utils.logging.set_verbosity_info()
+
+    log_level = training_args.get_process_log_level()
+    logger.setLevel(log_level)
+    transformers.utils.logging.set_verbosity(log_level)
+    transformers.utils.logging.enable_default_handler()
+    transformers.utils.logging.enable_explicit_format()
+
+    # Log on each process the small summary:
+    logger.warning(
+        f"Process rank: {training_args.local_rank}, device: {training_args.device}, n_gpu: {training_args.n_gpu} "
+        + f"distributed training: {bool(training_args.local_rank != -1)}, 16-bits training: {training_args.fp16}"
+    )
+    logger.info(f"Training/evaluation parameters {training_args}")
+
+    # Set seed before initializing model.
+    set_seed(training_args.seed)
+
+    # Detecting last checkpoint.
+    last_checkpoint = None
+    if os.path.isdir(training_args.output_dir) and training_args.do_train and not training_args.overwrite_output_dir:
+        last_checkpoint = get_last_checkpoint(training_args.output_dir)
+        if last_checkpoint is None and len(os.listdir(training_args.output_dir)) > 0:
+            raise ValueError(
+                f"Output directory ({training_args.output_dir}) already exists and is not empty. "
+                "Use --overwrite_output_dir to train from scratch."
+            )
+        elif last_checkpoint is not None and training_args.resume_from_checkpoint is None:
+            logger.info(
+                f"Checkpoint detected, resuming training at {last_checkpoint}. To avoid this behavior, change "
+                "the `--output_dir` or add `--overwrite_output_dir` to train from scratch."
+            )
+
+    # Initialize our dataset and prepare it for the audio classification task.
+    raw_datasets = DatasetDict()
+    raw_datasets["train"] = load_dataset(
+        data_args.dataset_name,
+        data_args.dataset_config_name,
+        split=data_args.train_split_name,
+        use_auth_token=True if model_args.use_auth_token else None,
+    )
+    raw_datasets["eval"] = load_dataset(
+        data_args.dataset_name,
+        data_args.dataset_config_name,
+        split=data_args.eval_split_name,
+        use_auth_token=True if model_args.use_auth_token else None,
+    )
+
+    if data_args.audio_column_name not in raw_datasets["train"].column_names:
+        raise ValueError(
+            f"--audio_column_name {data_args.audio_column_name} not found in dataset '{data_args.dataset_name}'. "
+            "Make sure to set `--audio_column_name` to the correct audio column - one of "
+            f"{', '.join(raw_datasets['train'].column_names)}."
+        )
+
+    if data_args.label_column_name not in raw_datasets["train"].column_names:
+        raise ValueError(
+            f"--label_column_name {data_args.label_column_name} not found in dataset '{data_args.dataset_name}'. "
+            "Make sure to set `--label_column_name` to the correct text column - one of "
+            f"{', '.join(raw_datasets['train'].column_names)}."
+        )
+
+    # Setting `return_attention_mask=True` is the way to get a correctly masked mean-pooling over
+    # transformer outputs in the classifier, but it doesn't always lead to better accuracy
+    feature_extractor = AutoFeatureExtractor.from_pretrained(
+        model_args.feature_extractor_name or model_args.model_name_or_path,
+        return_attention_mask=model_args.attention_mask,
+        cache_dir=model_args.cache_dir,
+        revision=model_args.model_revision,
+        use_auth_token=True if model_args.use_auth_token else None,
+    )
+
+    # `datasets` takes care of automatically loading and resampling the audio,
+    # so we just need to set the correct target sampling rate.
+    raw_datasets = raw_datasets.cast_column(
+        data_args.audio_column_name, datasets.features.Audio(sampling_rate=feature_extractor.sampling_rate)
+    )
+
+    model_input_name = feature_extractor.model_input_names[0]
+
+    def train_transforms(batch):
+        """Apply train_transforms across a batch."""
+        subsampled_wavs = []
+        for audio in batch[data_args.audio_column_name]:
+            wav = random_subsample(
+                audio["array"], max_length=data_args.max_length_seconds, sample_rate=feature_extractor.sampling_rate
+            )
+            subsampled_wavs.append(wav)
+        inputs = feature_extractor(subsampled_wavs, sampling_rate=feature_extractor.sampling_rate)
+        output_batch = {model_input_name: inputs.get(model_input_name)}
+        output_batch["labels"] = list(batch[data_args.label_column_name])
+
+        return output_batch
+
+    def val_transforms(batch):
+        """Apply val_transforms across a batch."""
+        wavs = [audio["array"] for audio in batch[data_args.audio_column_name]]
+        inputs = feature_extractor(wavs, sampling_rate=feature_extractor.sampling_rate)
+        output_batch = {model_input_name: inputs.get(model_input_name)}
+        output_batch["labels"] = list(batch[data_args.label_column_name])
+
+        return output_batch
+
+    # Prepare label mappings.
+    # We'll include these in the model's config to get human readable labels in the Inference API.
+    labels = raw_datasets["train"].features[data_args.label_column_name].names
+    label2id, id2label = {}, {}
+    for i, label in enumerate(labels):
+        label2id[label] = str(i)
+        id2label[str(i)] = label
+
+    # Load the accuracy metric from the datasets package
+    metric = evaluate.load("accuracy")
+
+    # Define our compute_metrics function. It takes an `EvalPrediction` object (a namedtuple with
+    # `predictions` and `label_ids` fields) and has to return a dictionary string to float.
+    def compute_metrics(eval_pred):
+        """Computes accuracy on a batch of predictions"""
+        predictions = np.argmax(eval_pred.predictions, axis=1)
+        return metric.compute(predictions=predictions, references=eval_pred.label_ids)
+
+    config = AutoConfig.from_pretrained(
+        model_args.config_name or model_args.model_name_or_path,
+        num_labels=len(labels),
+        label2id=label2id,
+        id2label=id2label,
+        finetuning_task="audio-classification",
+        cache_dir=model_args.cache_dir,
+        revision=model_args.model_revision,
+        use_auth_token=True if model_args.use_auth_token else None,
+    )
+    model = AutoModelForAudioClassification.from_pretrained(
+        model_args.model_name_or_path,
+        from_tf=bool(".ckpt" in model_args.model_name_or_path),
+        config=config,
+        cache_dir=model_args.cache_dir,
+        revision=model_args.model_revision,
+        use_auth_token=True if model_args.use_auth_token else None,
+        ignore_mismatched_sizes=model_args.ignore_mismatched_sizes,
+    )
+
+    # freeze the convolutional waveform encoder
+    if model_args.freeze_feature_encoder:
+        model.freeze_feature_encoder()
+
+    if training_args.do_train:
+        if data_args.max_train_samples is not None:
+            raw_datasets["train"] = (
+                raw_datasets["train"].shuffle(seed=training_args.seed).select(range(data_args.max_train_samples))
+            )
+        # Set the training transforms
+        raw_datasets["train"].set_transform(train_transforms, output_all_columns=False)
+
+    if training_args.do_eval:
+        if data_args.max_eval_samples is not None:
+            raw_datasets["eval"] = (
+                raw_datasets["eval"].shuffle(seed=training_args.seed).select(range(data_args.max_eval_samples))
+            )
+        # Set the validation transforms
+        raw_datasets["eval"].set_transform(val_transforms, output_all_columns=False)
+
+    # Initialize our trainer
+    trainer = Trainer(
+        model=model,
+        args=training_args,
+        train_dataset=raw_datasets["train"] if training_args.do_train else None,
+        eval_dataset=raw_datasets["eval"] if training_args.do_eval else None,
+        compute_metrics=compute_metrics,
+        tokenizer=feature_extractor,
+    )
+
+    # Training
+    if training_args.do_train:
+        checkpoint = None
+        if training_args.resume_from_checkpoint is not None:
+            checkpoint = training_args.resume_from_checkpoint
+        elif last_checkpoint is not None:
+            checkpoint = last_checkpoint
+        train_result = trainer.train(resume_from_checkpoint=checkpoint)
+        trainer.save_model()
+        trainer.log_metrics("train", train_result.metrics)
+        trainer.save_metrics("train", train_result.metrics)
+        trainer.save_state()
+
+    # Evaluation
+    if training_args.do_eval:
+        metrics = trainer.evaluate()
+        trainer.log_metrics("eval", metrics)
+        trainer.save_metrics("eval", metrics)
+
+    # Write model card and (optionally) push to hub
+    kwargs = {
+        "finetuned_from": model_args.model_name_or_path,
+        "tasks": "audio-classification",
+        "dataset": data_args.dataset_name,
+        "tags": ["audio-classification"],
+    }
+    if training_args.push_to_hub:
+        trainer.push_to_hub(**kwargs)
+    else:
+        trainer.create_model_card(**kwargs)
+
+
+if __name__ == "__main__":
+    main()