transformers/examples/pytorch/audio-classification/README.md
Anton Lozhkov 4213728067
[Examples] Add an official audio classification example (#13722)
* Restore broken merge

* Additional args, DDP, remove CommonLanguage

* Update examples for V100, add training results

* Style

* Apply suggestions from code review

Co-authored-by: Sylvain Gugger <35901082+sgugger@users.noreply.github.com>

* Remove custom datasets for simplicity, apply suggestions from code review

* Add the attention_mask flag, reorganize README

Co-authored-by: Sylvain Gugger <35901082+sgugger@users.noreply.github.com>
2021-10-01 18:52:45 +02:00

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# Audio classification examples
The following examples showcase how to fine-tune `Wav2Vec2` for audio classification using PyTorch.
Speech recognition models that have been pretrained in unsupervised fashion on audio data alone,
*e.g.* [Wav2Vec2](https://huggingface.co/transformers/master/model_doc/wav2vec2.html),
[HuBERT](https://huggingface.co/transformers/master/model_doc/hubert.html),
[XLSR-Wav2Vec2](https://huggingface.co/transformers/master/model_doc/xlsr_wav2vec2.html), have shown to require only
very little annotated data to yield good performance on speech classification datasets.
## Single-GPU
The following command shows how to fine-tune [wav2vec2-base](https://huggingface.co/facebook/wav2vec2-base) on the 🗣️ [Keyword Spotting subset](https://huggingface.co/datasets/superb#ks) of the SUPERB dataset.
```bash
python run_audio_classification.py \
--model_name_or_path facebook/wav2vec2-base \
--dataset_name superb \
--dataset_config_name ks \
--output_dir wav2vec2-base-keyword-spotting \
--overwrite_output_dir \
--remove_unused_columns False \
--do_train \
--do_eval \
--fp16 \
--learning_rate 3e-5 \
--max_length_seconds 1 \
--warmup_ratio 0.1 \
--num_train_epochs 5 \
--per_device_train_batch_size 32 \
--gradient_accumulation_steps 4 \
--per_device_eval_batch_size 32 \
--dataloader_num_workers 4 \
--logging_strategy steps \
--logging_steps 10 \
--evaluation_strategy epoch \
--save_strategy epoch \
--load_best_model_at_end True \
--save_total_limit 3 \
--seed 0 \
--push_to_hub
```
On a single V100 GPU (16GB), this script should run in ~10 minutes and yield accuracy of **98.4%**.
👀 See the results here: [anton-l/wav2vec2-base-keyword-spotting](https://huggingface.co/anton-l/wav2vec2-base-keyword-spotting)
## Multi-GPU
The following command shows how to fine-tune [wav2vec2-base](https://huggingface.co/facebook/wav2vec2-base) for 🌎 **Language Identification** on the [CommonLanguage dataset](https://huggingface.co/datasets/anton-l/common_language).
```bash
python run_audio_classification.py \
--model_name_or_path facebook/wav2vec2-base \
--dataset_name anton-l/common_language \
--audio_column_name path \
--label_column_name language \
--output_dir wav2vec2-base-lang-id \
--overwrite_output_dir \
--remove_unused_columns False \
--do_train \
--do_eval \
--fp16 \
--learning_rate 3e-5 \
--max_length_seconds 16 \
--attention_mask False \
--warmup_ratio 0.1 \
--num_train_epochs 10 \
--per_device_train_batch_size 8 \
--gradient_accumulation_steps 4 \
--per_device_eval_batch_size 1 \
--dataloader_num_workers 8 \
--logging_strategy steps \
--logging_steps 10 \
--evaluation_strategy epoch \
--save_strategy epoch \
--load_best_model_at_end True \
--save_total_limit 3 \
--seed 0 \
--push_to_hub
```
On 4 V100 GPUs (16GB), this script should run in ~1 hour and yield accuracy of **79.45%**.
👀 See the results here: [anton-l/wav2vec2-base-lang-id](https://huggingface.co/anton-l/wav2vec2-base-lang-id)
## Sharing your model on 🤗 Hub
0. If you haven't already, [sign up](https://huggingface.co/join) for a 🤗 account
1. Make sure you have `git-lfs` installed and git set up.
```bash
$ apt install git-lfs
```
2. Log in with your HuggingFace account credentials using `huggingface-cli`
```bash
$ huggingface-cli login
# ...follow the prompts
```
3. When running the script, pass the following arguments:
```bash
python run_audio_classification.py \
--push_to_hub \
--hub_model_id <username/model_id> \
...
```