transformers/examples/research_projects/wav2vec2/README.md
Mohamed El-Geish af8afdc88d
wav2vec2: support datasets other than LibriSpeech (#10581)
* wav2vec2: support datasets other than LibriSpeech

* Formatting run_asr.py to pass code quality test

* bundled orthography options and added verbose logs

* fixing a typo in timit fine-tuning script

* update comment for clarity

* resize_lm_head and load custom vocab from file

* adding a max_duration_in_seconds filter

* do not assign `duration_filter` lambda, use a def

* log untransliterated text as well

* fix base model for arabic

* fix duration filter when target_sr is not set

* drop duration_in_seconds when unneeded

* script for wav2vec2-large-lv60-timit-asr

* fix for "tha" in arabic corpus (huggingface#10581)

* adding more options to work with common_voice

* PR feedback (huggingface#10581)

* small README change
2021-03-18 10:20:26 +03:00

130 lines
5.6 KiB
Markdown

## Fine-tuning Wav2Vec2
The `run_asr.py` script allows one to fine-tune pretrained Wav2Vec2 models that can be found [here](https://huggingface.co/models?search=facebook/wav2vec2).
This finetuning script can also be run as a google colab [TODO: here]( ).
The script is actively maintained by [Patrick von Platen](https://github.com/patrickvonplaten).
Feel free to ask a question on the [Forum](https://discuss.huggingface.co/) or post an issue on [GitHub](https://github.com/huggingface/transformers/issues/new/choose) and adding `@patrickvonplaten` as a tag.
### Fine-Tuning with TIMIT
Let's take a look at the [script](./finetune_base_timit_asr.sh) used to fine-tune [wav2vec2-base](https://huggingface.co/facebook/wav2vec2-base)
with the [TIMIT dataset](https://huggingface.co/datasets/timit_asr):
```bash
#!/usr/bin/env bash
python run_asr.py \
--output_dir="./wav2vec2-base-timit-asr" \
--num_train_epochs="30" \
--per_device_train_batch_size="20" \
--per_device_eval_batch_size="20" \
--evaluation_strategy="steps" \
--save_steps="500" \
--eval_steps="100" \
--logging_steps="50" \
--learning_rate="5e-4" \
--warmup_steps="3000" \
--model_name_or_path="facebook/wav2vec2-base" \
--fp16 \
--dataset_name="timit_asr" \
--train_split_name="train" \
--validation_split_name="test" \
--orthography="timit" \
--preprocessing_num_workers="$(nproc)" \
--group_by_length \
--freeze_feature_extractor \
--verbose_logging \
```
The resulting model and inference examples can be found [here](https://huggingface.co/elgeish/wav2vec2-base-timit-asr).
Some of the arguments above may look unfamiliar, let's break down what's going on:
`--orthography="timit"` applies certain text preprocessing rules, for tokenization and normalization, to clean up the dataset.
In this case, we use the following instance of `Orthography`:
```python
Orthography(
do_lower_case=True,
# break compounds like "quarter-century-old" and replace pauses "--"
translation_table=str.maketrans({"-": " "}),
)
```
The instance above is used as follows:
* creates a tokenizer with `do_lower_case=True` (ignores casing for input and lowercases output when decoding)
* replaces `"-"` with `" "` to break compounds like `"quarter-century-old"` and to clean up suspended hyphens
* cleans up consecutive whitespaces (replaces them with a single space: `" "`)
* removes characters not in vocabulary (lacking respective sound units)
`--verbose_logging` logs text preprocessing updates and when evaluating, using the validation split every `eval_steps`,
logs references and predictions.
### Fine-Tuning with Arabic Speech Corpus
Other datasets, like the [Arabic Speech Corpus dataset](https://huggingface.co/datasets/arabic_speech_corpus),
require more work! Let's take a look at the [script](./finetune_large_xlsr_53_arabic_speech_corpus.sh)
used to fine-tune [wav2vec2-large-xlsr-53](https://huggingface.co/elgeish/wav2vec2-large-xlsr-53-arabic):
```bash
#!/usr/bin/env bash
python run_asr.py \
--output_dir="./wav2vec2-large-xlsr-53-arabic-speech-corpus" \
--num_train_epochs="50" \
--per_device_train_batch_size="1" \
--per_device_eval_batch_size="1" \
--gradient_accumulation_steps="8" \
--evaluation_strategy="steps" \
--save_steps="500" \
--eval_steps="100" \
--logging_steps="50" \
--learning_rate="5e-4" \
--warmup_steps="3000" \
--model_name_or_path="elgeish/wav2vec2-large-xlsr-53-arabic" \
--fp16 \
--dataset_name="arabic_speech_corpus" \
--train_split_name="train" \
--validation_split_name="test" \
--max_duration_in_seconds="15" \
--orthography="buckwalter" \
--preprocessing_num_workers="$(nproc)" \
--group_by_length \
--freeze_feature_extractor \
--target_feature_extractor_sampling_rate \
--verbose_logging \
```
First, let's understand how this dataset represents Arabic text; it uses a format called
[Buckwalter transliteration](https://en.wikipedia.org/wiki/Buckwalter_transliteration).
We use the [lang-trans](https://github.com/kariminf/lang-trans) package to convert back to Arabic when logging.
The Buckwalter format only includes ASCII characters, some of which are non-alpha (e.g., `">"` maps to `"أ"`).
`--orthography="buckwalter"` applies certain text preprocessing rules, for tokenization and normalization, to clean up the dataset. In this case, we use the following instance of `Orthography`:
```python
Orthography(
vocab_file=pathlib.Path(__file__).parent.joinpath("vocab/buckwalter.json"),
word_delimiter_token="/", # "|" is Arabic letter alef with madda above
words_to_remove={"sil"}, # fixing "sil" in arabic_speech_corpus dataset
untransliterator=arabic.buckwalter.untransliterate,
translation_table=str.maketrans(translation_table = {
"-": " ", # sometimes used to represent pauses
"^": "v", # fixing "tha" in arabic_speech_corpus dataset
}),
)
```
The instance above is used as follows:
* creates a tokenizer with Buckwalter vocabulary and `word_delimiter_token="/"`
* replaces `"-"` with `" "` to clean up hyphens and fixes the orthography for `"ث"`
* removes words used as indicators (in this case, `"sil"` is used for silence)
* cleans up consecutive whitespaces (replaces them with a single space: `" "`)
* removes characters not in vocabulary (lacking respective sound units)
`--verbose_logging` logs text preprocessing updates and when evaluating, using the validation split every `eval_steps`,
logs references and predictions. Using the Buckwalter format, text is also logged in Arabic abjad.
`--target_feature_extractor_sampling_rate` resamples audio to target feature extractor's sampling rate (16kHz).
`--max_duration_in_seconds="15"` filters out examples whose audio is longer than the specified limit,
which helps with capping GPU memory usage.