Add inference section to task guides

docs/source/en/tasks/audio_classification.mdx

@@ -14,27 +14,44 @@ specific language governing permissions and limitations under the License.
 
 <Youtube id="KWwzcmG98Ds"/>
 
-Audio classification assigns a label or class to audio data. It is similar to text classification, except an audio input is continuous and must be discretized, whereas text can be split into tokens. Some practical applications of audio classification include identifying intent, speakers, and even animal species by their sounds.
+Audio classification - just like with text - assigns a class label output from the input data. The only difference is instead of text inputs, you have raw audio waveforms. Some practical applications of audio classification include identifying speaker intent, language classification, and even animal species by their sounds.
 
-This guide will show you how to fine-tune [Wav2Vec2](https://huggingface.co/facebook/wav2vec2-base) on the [MInDS-14](https://huggingface.co/datasets/PolyAI/minds14) to classify intent.
+This guide will show you how to:
+
+1. Finetune [Wav2Vec2](https://huggingface.co/facebook/wav2vec2-base) on the [MInDS-14](https://huggingface.co/datasets/PolyAI/minds14) dataset to classify speaker intent.
+2. Use your finetuned model for inference.
 
 <Tip>
 
 See the audio classification [task page](https://huggingface.co/tasks/audio-classification) for more information about its associated models, datasets, and metrics.
 
 </Tip>
 
+Before you begin, make sure you have all the necessary libraries installed:
+
+```bash
+pip install transformers datasets evaluate
+```
+
+We encourage you to login to your Hugging Face account so you can upload and share your model with the community. When prompted, enter your token to login:
+
+```py
+>>> from huggingface_hub import notebook_login
+
+>>> notebook_login()
+```
+
 ## Load MInDS-14 dataset
 
-Load the [MInDS-14](https://huggingface.co/datasets/PolyAI/minds14) from the 🤗 Datasets library:
+Start by loading the MInDS-14 dataset from the 🤗 Datasets library:
 
 ```py
 >>> from datasets import load_dataset, Audio
 
 >>> minds = load_dataset("PolyAI/minds14", name="en-US", split="train")
 ```
 
-Split this dataset into a train and test set:
+Split the dataset's `train` split into a smaller train and test set with the [`~datasets.Dataset.train_test_split`] method. This'll give you a chance to experiment and make sure everything works before spending more time on the full dataset.
 
 ```py
 >>> minds = minds.train_test_split(test_size=0.2)
@@ -56,7 +73,7 @@ DatasetDict({
 })
 ```
 
-While the dataset contains a lot of other useful information, like `lang_id` and `english_transcription`, you will focus on the `audio` and `intent_class` in this guide. Remove the other columns:
+While the dataset contains a lot of useful information, like `lang_id` and `english_transcription`, you'll focus on the `audio` and `intent_class` in this guide. Remove the other columns with the [`~datasets.Dataset.remove_columns`] method:
 
 ```py
 >>> minds = minds.remove_columns(["path", "transcription", "english_transcription", "lang_id"])
@@ -73,7 +90,12 @@ Take a look at an example now:
  'intent_class': 2}
 ```
 
-The `audio` column contains a 1-dimensional `array` of the speech signal that must be called to load and resample the audio file. The `intent_class` column is an integer that represents the class id of intent. Create a dictionary that maps a label name to an integer and vice versa. The mapping will help the model recover the label name from the label number:
+There are two fields:
+
+- `audio`: a 1-dimensional `array` of the speech signal that must be called to load and resample the audio file. 
+- `intent_class`: represents the class id of the speaker's intent. 
+
+To make it easier for the model to get the label name from the label id, create a dictionary that maps the label name to an integer and vice versa:
 
 ```py
 >>> labels = minds["train"].features["intent_class"].names
@@ -83,26 +105,24 @@ The `audio` column contains a 1-dimensional `array` of the speech signal that mu
 ...     id2label[str(i)] = label
 ```
 
-Now you can convert the label number to a label name for more information:
+Now you can convert the label id to a label name:
 
 ```py
 >>> id2label[str(2)]
 'app_error'
 ```
 
-Each keyword - or label - corresponds to a number; `2` indicates `app_error` in the example above.
-
 ## Preprocess
 
-Load the Wav2Vec2 feature extractor to process the audio signal:
+The next step is to load a Wav2Vec2 feature extractor to process the audio signal:
 
 ```py
 >>> from transformers import AutoFeatureExtractor
 
 >>> feature_extractor = AutoFeatureExtractor.from_pretrained("facebook/wav2vec2-base")
 ```
 
-The [MInDS-14](https://huggingface.co/datasets/PolyAI/minds14) dataset has a sampling rate of 8000khz. You will need to resample the dataset to use the pretrained Wav2Vec2 model:
+The MInDS-14 dataset has a sampling rate of 8000khz (you can find this information in it's [dataset card](https://huggingface.co/datasets/PolyAI/minds14)), which means you'll need to resample the dataset to 16000kHz to use the pretrained Wav2Vec2 model:
 
 ```py
 >>> minds = minds.cast_column("audio", Audio(sampling_rate=16_000))
@@ -114,11 +134,11 @@ The [MInDS-14](https://huggingface.co/datasets/PolyAI/minds14) dataset has a sam
  'intent_class': 2}
 ```
 
-The preprocessing function needs to:
+Now create a preprocessing function that:
 
-1. Call the `audio` column to load and if necessary resample the audio file.
-2. Check the sampling rate of the audio file matches the sampling rate of the audio data a model was pretrained with. You can find this information on the Wav2Vec2 [model card](https://huggingface.co/facebook/wav2vec2-base).
-3. Set a maximum input length so longer inputs are batched without being truncated.
+1. Calls the `audio` column to load, and if necessary, resample the audio file.
+2. Checks if the sampling rate of the audio file matches the sampling rate of the audio data a model was pretrained with. You can find this information in the Wav2Vec2 [model card](https://huggingface.co/facebook/wav2vec2-base).
+3. Set a maximum input length to batch longer inputs without truncating them.
 
 ```py
 >>> def preprocess_function(examples):
@@ -129,18 +149,46 @@ The preprocessing function needs to:
 ...     return inputs
 ```
 
-Use 🤗 Datasets [`~datasets.Dataset.map`] function to apply the preprocessing function over the entire dataset. You can speed up the `map` function by setting `batched=True` to process multiple elements of the dataset at once. Remove the columns you don't need, and rename `intent_class` to `label` because that is what the model expects:
+To apply the preprocessing function over the entire dataset, use 🤗 Datasets [`~datasets.Dataset.map`] function. You can speed up `map` by setting `batched=True` to process multiple elements of the dataset at once. Remove the columns you don't need, and rename `intent_class` to `label` because that's the name the model expects:
 
 ```py
 >>> encoded_minds = minds.map(preprocess_function, remove_columns="audio", batched=True)
 >>> encoded_minds = encoded_minds.rename_column("intent_class", "label")
 ```
 
+## Evaluate
+
+Including a metric during training is often helpful for evaluating your model's performance. You can quickly load a evaluation method with the 🤗 [Evaluate](https://huggingface.co/docs/evaluate/index) library. For this task, load the [accuracy](https://huggingface.co/spaces/evaluate-metric/accuracy) metric (see the 🤗 Evaluate [quick tour](https://huggingface.co/docs/evaluate/a_quick_tour) to learn more about how to load and compute a metric):
+
+```py
+>>> import evaluate
+
+>>> accuracy = evaluate.load("accuracy")
+```
+
+Then create a function that passes your predictions and labels to [`~evaluate.EvaluationModule.compute`] to calculate the accuracy:
+
+```py
+>>> import numpy as np
+
+
+>>> def compute_metrics(eval_pred):
+...     predictions = np.argmax(eval_pred.predictions, axis=1)
+...     return accuracy.compute(predictions=predictions, references=eval_pred.label_ids)
+```
+
+Your `compute_metrics` function is ready to go now, and you'll return to it when you setup your training.
+
 ## Train
 
 <frameworkcontent>
 <pt>
-Load Wav2Vec2 with [`AutoModelForAudioClassification`]. Specify the number of labels, and pass the model the mapping between label number and label class:
+<Tip>
+
+If you aren't familiar with finetuning a model with the [`Trainer`], take a look at the basic tutorial [here](../training#train-with-pytorch-trainer)!
+
+</Tip>
+You're ready to start training your model now! Load Wav2Vec2 with [`AutoModelForAudioClassification`] along with the number of expected labels, and the label mappings:
 
 ```py
 >>> from transformers import AutoModelForAudioClassification, TrainingArguments, Trainer
@@ -151,25 +199,28 @@ Load Wav2Vec2 with [`AutoModelForAudioClassification`]. Specify the number of la
 ... )
 ```
 
-<Tip>
-
-If you aren't familiar with fine-tuning a model with the [`Trainer`], take a look at the basic tutorial [here](../training#finetune-with-trainer)!
-
-</Tip>
-
 At this point, only three steps remain:
 
-1. Define your training hyperparameters in [`TrainingArguments`].
-2. Pass the training arguments to [`Trainer`] along with the model, datasets, and feature extractor.
-3. Call [`~Trainer.train`] to fine-tune your model.
+1. Define your training hyperparameters in [`TrainingArguments`]. The only required parameter is `output_dir` which specifies where to save your model. You'll push this model to the Hub by setting `push_to_hub=True` (you need to be signed in to Hugging Face to upload your model). At the end of each epoch, the [`Trainer`] will evaluate the accuracy and save the training checkpoint.
+2. Pass the training arguments to [`Trainer`] along with the model, dataset, tokenizer, data collator, and `compute_metrics` function.
+3. Call [`~Trainer.train`] to finetune your model.
+
 
 ```py
 >>> training_args = TrainingArguments(
-...     output_dir="./results",
+...     output_dir="my_awesome_mind_model",
 ...     evaluation_strategy="epoch",
 ...     save_strategy="epoch",
 ...     learning_rate=3e-5,
-...     num_train_epochs=5,
+...     per_device_train_batch_size=32,
+...     gradient_accumulation_steps=4,
+...     per_device_eval_batch_size=32,
+...     num_train_epochs=10,
+...     warmup_ratio=0.1,
+...     logging_steps=10,
+...     load_best_model_at_end=True,
+...     metric_for_best_model="accuracy",
+...     push_to_hub=True,
 ... )
 
 >>> trainer = Trainer(
@@ -178,15 +229,89 @@ At this point, only three steps remain:
 ...     train_dataset=encoded_minds["train"],
 ...     eval_dataset=encoded_minds["test"],
 ...     tokenizer=feature_extractor,
+...     compute_metrics=compute_metrics,
 ... )
 
 >>> trainer.train()
 ```
+
+Once training is completed, share your model to the Hub with the [`~transformers.Trainer.push_to_hub`] method so everyone can use your model:
+
+```py
+>>> trainer.push_to_hub()
+```
 </pt>
 </frameworkcontent>
 
 <Tip>
 
-For a more in-depth example of how to fine-tune a model for audio classification, take a look at the corresponding [PyTorch notebook](https://colab.research.google.com/github/huggingface/notebooks/blob/main/examples/audio_classification.ipynb).
+For a more in-depth example of how to finetune a model for audio classification, take a look at the corresponding [PyTorch notebook](https://colab.research.google.com/github/huggingface/notebooks/blob/main/examples/audio_classification.ipynb).
 
 </Tip>
+
+## Inference
+
+Great, now that you've finetuned a model, you can use it for inference!
+
+Load an audio file you'd like to run inference on. Remember to resample the sampling rate of the audio file to match the sampling rate of the model if you need to!
+
+```py
+>>> from datasets import load_dataset, Audio
+
+>>> dataset = load_dataset("PolyAI/minds14", name="en-US", split="train")
+>>> dataset = dataset.cast_column("audio", Audio(sampling_rate=16000))
+>>> sampling_rate = dataset.features["audio"].sampling_rate
+>>> audio_file = dataset[0]["audio"]["path"]
+```
+
+The simplest way to try out your finetuned model for inference is to use it in a [`pipeline`]. Instantiate a `pipeline` for audio classification with your model, and pass your audio file to it:
+
+```py
+>>> from transformers import pipeline
+
+>>> classifier = pipeline("audio-classification", model="stevhliu/my_awesome_minds_model")
+>>> classifier(audio_file)
+[
+    {'score': 0.09766869246959686, 'label': 'cash_deposit'},
+    {'score': 0.07998877018690109, 'label': 'app_error'},
+    {'score': 0.0781070664525032, 'label': 'joint_account'},
+    {'score': 0.07667109370231628, 'label': 'pay_bill'},
+    {'score': 0.0755252093076706, 'label': 'balance'}
+]
+```
+
+You can also manually replicate the results of the `pipeline` if you'd like:
+
+<frameworkcontent>
+<pt>
+Load a feature extractor to preprocess the audio file and return the `input` as PyTorch tensors:
+
+```py
+>>> from transformers import AutoFeatureExtractor
+
+>>> feature_extractor = AutoFeatureExtractor.from_pretrained("stevhliu/my_awesome_minds_model")
+>>> inputs = feature_extractor(dataset[0]["audio"]["array"], sampling_rate=sampling_rate, return_tensors="pt")
+```
+
+Pass your inputs to the model and return the logits:
+
+```py
+>>> from transformers import AutoModelForAudioClassification
+
+>>> model = AutoModelForAudioClassification.from_pretrained("stevhliu/my_awesome_minds_model")
+>>> with torch.no_grad():
+...     logits = model(**inputs).logits
+```
+
+Get the class with the highest probability, and use the model's `id2label` mapping to convert it to a label:
+
+```py
+>>> import torch
+
+>>> predicted_class_ids = torch.argmax(logits).item()
+>>> predicted_label = model.config.id2label[predicted_class_ids]
+>>> predicted_label
+'cash_deposit'
+```
+</pt>
+</frameworkcontent>