[Whisper] Add model for audio classification

docs/source/en/model_doc/whisper.mdx

@@ -79,6 +79,11 @@ The original code can be found [here](https://github.com/openai/whisper).
 [[autodoc]] WhisperForConditionalGeneration
     - forward
 
+## WhisperForAudioClassification
+
+[[autodoc]] WhisperForAudioClassification
+    - forward
+
 
 ## TFWhisperModel
 

docs/source/en/tasks/audio_classification.mdx

@@ -28,7 +28,7 @@ The task illustrated in this tutorial is supported by the following model archit
 
 <!--This tip is automatically generated by `make fix-copies`, do not fill manually!-->
 
-[Audio Spectrogram Transformer](../model_doc/audio-spectrogram-transformer), [Data2VecAudio](../model_doc/data2vec-audio), [Hubert](../model_doc/hubert), [SEW](../model_doc/sew), [SEW-D](../model_doc/sew-d), [UniSpeech](../model_doc/unispeech), [UniSpeechSat](../model_doc/unispeech-sat), [Wav2Vec2](../model_doc/wav2vec2), [Wav2Vec2-Conformer](../model_doc/wav2vec2-conformer), [WavLM](../model_doc/wavlm)
+[Audio Spectrogram Transformer](../model_doc/audio-spectrogram-transformer), [Data2VecAudio](../model_doc/data2vec-audio), [Hubert](../model_doc/hubert), [SEW](../model_doc/sew), [SEW-D](../model_doc/sew-d), [UniSpeech](../model_doc/unispeech), [UniSpeechSat](../model_doc/unispeech-sat), [Wav2Vec2](../model_doc/wav2vec2), [Wav2Vec2-Conformer](../model_doc/wav2vec2-conformer), [WavLM](../model_doc/wavlm), [Whisper](../model_doc/whisper)
 
 <!--End of the generated tip-->
 

src/transformers/__init__.py

@@ -2575,6 +2575,7 @@
     _import_structure["models.whisper"].extend(
         [
             "WHISPER_PRETRAINED_MODEL_ARCHIVE_LIST",
+            "WhisperForAudioClassification",
             "WhisperForConditionalGeneration",
             "WhisperModel",
             "WhisperPreTrainedModel",
@@ -5782,6 +5783,7 @@
         )
         from .models.whisper import (
             WHISPER_PRETRAINED_MODEL_ARCHIVE_LIST,
+            WhisperForAudioClassification,
             WhisperForConditionalGeneration,
             WhisperModel,
             WhisperPreTrainedModel,

src/transformers/models/auto/modeling_auto.py

@@ -877,6 +877,7 @@
         ("wav2vec2", "Wav2Vec2ForSequenceClassification"),
         ("wav2vec2-conformer", "Wav2Vec2ConformerForSequenceClassification"),
         ("wavlm", "WavLMForSequenceClassification"),
+        ("whisper", "WhisperForAudioClassification"),
     ]
 )
 

src/transformers/models/whisper/__init__.py

@@ -49,6 +49,7 @@
         "WhisperForConditionalGeneration",
         "WhisperModel",
         "WhisperPreTrainedModel",
+        "WhisperForAudioClassification",
     ]
 
 try:
@@ -99,6 +100,7 @@
     else:
         from .modeling_whisper import (
             WHISPER_PRETRAINED_MODEL_ARCHIVE_LIST,
+            WhisperForAudioClassification,
             WhisperForConditionalGeneration,
             WhisperModel,
             WhisperPreTrainedModel,

src/transformers/models/whisper/configuration_whisper.py

@@ -136,6 +136,12 @@ class WhisperConfig(PretrainedConfig):
         begin_suppress_tokens (`List[int]`, *optional*, defaults to `[220,50256]`):
             A list containing tokens that will be supressed at the beginning of the sampling process. Initialized as
             the token for `" "` (`blank_token_id`) and the `eos_token_id`
+        use_weighted_layer_sum (`bool`, *optional*, defaults to `False`):
+            Whether to use a weighted average of layer outputs with learned weights. Only relevant when using an
+            instance of [`WhisperForAudioClassification`].
+        classifier_proj_size (`int`, *optional*, defaults to 256):
+            Dimensionality of the projection before token mean-pooling for classification. Only relevant when using an
+            instance of [`WhisperForAudioClassification`].
         apply_spec_augment (`bool`, *optional*, defaults to `False`):
             Whether to apply *SpecAugment* data augmentation to the outputs of the feature encoder. For reference see
             [SpecAugment: A Simple Data Augmentation Method for Automatic Speech
@@ -214,6 +220,8 @@ def __init__(
         eos_token_id=50256,
         suppress_tokens=None,
         begin_suppress_tokens=[220, 50256],
+        use_weighted_layer_sum=False,
+        classifier_proj_size=256,
         apply_spec_augment=False,
         mask_time_prob=0.05,
         mask_time_length=10,
@@ -244,6 +252,11 @@ def __init__(
         self.scale_embedding = scale_embedding  # scale factor will be sqrt(d_model) if True
         self.max_source_positions = max_source_positions
         self.max_target_positions = max_target_positions
+
+        # Audio Classification-specific parameters. Feel free to ignore for other classes.
+        self.classifier_proj_size = classifier_proj_size
+        self.use_weighted_layer_sum = use_weighted_layer_sum
+
         # fine-tuning config parameters for SpecAugment: https://arxiv.org/abs/1904.08779
         self.apply_spec_augment = apply_spec_augment
         self.mask_time_prob = mask_time_prob

src/transformers/models/whisper/modeling_whisper.py

@@ -32,6 +32,7 @@
     BaseModelOutputWithPastAndCrossAttentions,
     Seq2SeqLMOutput,
     Seq2SeqModelOutput,
+    SequenceClassifierOutput,
 )
 from ...modeling_utils import PreTrainedModel
 from ...utils import add_start_docstrings, add_start_docstrings_to_model_forward, logging, replace_return_docstrings
@@ -701,6 +702,33 @@ def _get_feat_extract_output_lengths(self, input_lengths: torch.LongTensor):
             Whether or not to return a [`~utils.ModelOutput`] instead of a plain tuple.
 """
 
+WHISPER_ENCODER_INPUTS_DOCSTRING = r"""
+    Args:
+        input_features (`torch.FloatTensor` of shape `(batch_size, feature_size, sequence_length)`):
+            Float values mel features extracted from the raw speech waveform. Raw speech waveform can be obtained by
+            loading a `.flac` or `.wav` audio file into an array of type `List[float]` or a `numpy.ndarray`, *e.g.* via
+            the soundfile library (`pip install soundfile`). To prepare the array into `input_features`, the
+            [`AutoFeatureExtractor`] should be used for extracting the mel features, padding and conversion into a
+            tensor of type `torch.FloatTensor`. See [`~WhisperFeatureExtractor.__call__`]
+        head_mask (`torch.Tensor` of shape `(encoder_layers, encoder_attention_heads)`, *optional*):
+            Mask to nullify selected heads of the attention modules in the encoder. Mask values selected in `[0, 1]`:
+
+            - 1 indicates the head is **not masked**,
+            - 0 indicates the head is **masked**.
+        encoder_outputs (`tuple(tuple(torch.FloatTensor)`, *optional*):
+            Tuple consists of (`last_hidden_state`, *optional*: `hidden_states`, *optional*: `attentions`)
+            `last_hidden_state` of shape `(batch_size, sequence_length, hidden_size)`, *optional*) is a sequence of
+            hidden-states at the output of the last layer of the encoder.
+        output_attentions (`bool`, *optional*):
+            Whether or not to return the attentions tensors of all attention layers. See `attentions` under returned
+            tensors for more detail.
+        output_hidden_states (`bool`, *optional*):
+            Whether or not to return the hidden states of all layers. See `hidden_states` under returned tensors for
+            more detail.
+        return_dict (`bool`, *optional*):
+            Whether or not to return a [`~utils.ModelOutput`] instead of a plain tuple.
+"""
+
 
 class WhisperEncoder(WhisperPreTrainedModel):
     """
@@ -1578,3 +1606,123 @@ def _reorder_cache(past_key_values, beam_idx):
         for layer_past in past_key_values:
             reordered_past += (tuple(past_state.index_select(0, beam_idx) for past_state in layer_past),)
         return reordered_past
+
+
+@add_start_docstrings(
+    """
+    Whisper Encoder Model with a sequence classification head on top (a linear layer over the pooled output) for tasks
+    like SUPERB Keyword Spotting.
+    """,
+    WHISPER_ENCODER_INPUTS_DOCSTRING,
+)
+class WhisperForAudioClassification(WhisperPreTrainedModel):
+    def __init__(self, config):
+        super().__init__(config)
+
+        self.encoder = WhisperEncoder(config)
+        num_layers = config.num_hidden_layers + 1  # transformer layers + input embeddings
+        if config.use_weighted_layer_sum:
+            self.layer_weights = nn.Parameter(torch.ones(num_layers) / num_layers)
+        self.projector = nn.Linear(config.hidden_size, config.classifier_proj_size)
+        self.classifier = nn.Linear(config.classifier_proj_size, config.num_labels)
+
+        # Initialize weights and apply final processing
+        self.post_init()
+
+    def freeze_encoder(self):
+        """
+        Calling this function will disable the gradient computation for the Whisper encoder so that its parameters will
+        not be updated during training. Only the projection layers and classification head will be updated.
+        """
+        self.encoder._freeze_parameters()
+
+    @add_start_docstrings_to_model_forward(WHISPER_ENCODER_INPUTS_DOCSTRING)
+    @replace_return_docstrings(output_type=SequenceClassifierOutput, config_class=_CONFIG_FOR_DOC)
+    def forward(
+        self,
+        input_features: Optional[torch.LongTensor] = None,
+        head_mask: Optional[torch.Tensor] = None,
+        encoder_outputs: Optional[Tuple[Tuple[torch.FloatTensor]]] = None,
+        labels: Optional[torch.LongTensor] = None,
+        output_attentions: Optional[bool] = None,
+        output_hidden_states: Optional[bool] = None,
+        return_dict: Optional[bool] = None,
+    ) -> Union[Tuple[torch.Tensor], SequenceClassifierOutput]:
+        r"""
+        labels (`torch.LongTensor` of shape `(batch_size,)`, *optional*):
+            Labels for computing the sequence classification/regression loss. Indices should be in `[0, ...,
+            config.num_labels - 1]`. If `config.num_labels == 1` a regression loss is computed (Mean-Square loss), If
+            `config.num_labels > 1` a classification loss is computed (Cross-Entropy).
+
+        Returns:
+
+        Example:
+
+        ```python
+        >>> import torch
+        >>> from transformers import AutoFeatureExtractor, WhisperForAudioClassification
+        >>> from datasets import load_dataset
+
+        >>> feature_extractor = AutoFeatureExtractor.from_pretrained("sanchit-gandhi/whisper-medium-fleurs-lang-id")
+        >>> model = WhisperForAudioClassification.from_pretrained("sanchit-gandhi/whisper-medium-fleurs-lang-id")
+
+        >>> ds = load_dataset("google/fleurs", "all", split="validation", streaming=True)
+        >>> sample = next(iter(ds))
+
+        >>> inputs = feature_extractor(
+        ...     sample["audio"]["array"], sampling_rate=sample["audio"]["sampling_rate"], return_tensors="pt"
+        ... )
+        >>> input_features = inputs.input_features
+
+        >>> with torch.no_grad():
+        ...     logits = model(input_features).logits
+
+        >>> predicted_class_ids = torch.argmax(logits).item()
+        >>> predicted_label = model.config.id2label[predicted_class_ids]
+        >>> predicted_label
+        'af_za'
+        ```"""
+
+        output_attentions = output_attentions if output_attentions is not None else self.config.output_attentions
+        output_hidden_states = (
+            output_hidden_states if output_hidden_states is not None else self.config.output_hidden_states
+        )
+        return_dict = return_dict if return_dict is not None else self.config.use_return_dict
+
+        if encoder_outputs is None:
+            encoder_outputs = self.encoder(
+                input_features,
+                head_mask=head_mask,
+                output_attentions=output_attentions,
+                output_hidden_states=output_hidden_states,
+                return_dict=return_dict,
+            )
+
+        if self.config.use_weighted_layer_sum:
+            hidden_states = torch.stack(encoder_outputs, dim=1)
+            norm_weights = nn.functional.softmax(self.layer_weights, dim=-1)
+            hidden_states = (hidden_states * norm_weights.view(-1, 1, 1)).sum(dim=1)
+        else:
+            hidden_states = encoder_outputs[0]
+
+        hidden_states = self.projector(hidden_states)
+        pooled_output = hidden_states.mean(dim=1)
+
+        logits = self.classifier(pooled_output)
+
+        loss = None
+
+        if labels is not None:
+            loss_fct = CrossEntropyLoss()
+            loss = loss_fct(logits.view(-1, self.config.num_labels), labels.view(-1))
+
+        if not return_dict:
+            output = (logits,) + encoder_outputs[1:]
+            return ((loss,) + output) if loss is not None else output
+
+        return SequenceClassifierOutput(
+            loss=loss,
+            logits=logits,
+            hidden_states=encoder_outputs.hidden_states,
+            attentions=encoder_outputs.attentions,
+        )

src/transformers/utils/dummy_pt_objects.py

@@ -6797,6 +6797,13 @@ def __init__(self, *args, **kwargs):
 WHISPER_PRETRAINED_MODEL_ARCHIVE_LIST = None
 
 
+class WhisperForAudioClassification(metaclass=DummyObject):
+    _backends = ["torch"]
+
+    def __init__(self, *args, **kwargs):
+        requires_backends(self, ["torch"])
+
+
 class WhisperForConditionalGeneration(metaclass=DummyObject):
     _backends = ["torch"]