diff --git a/src/transformers/models/whisper/feature_extraction_whisper.py b/src/transformers/models/whisper/feature_extraction_whisper.py
index f2349120280b..418cb23b68b0 100644
--- a/src/transformers/models/whisper/feature_extraction_whisper.py
+++ b/src/transformers/models/whisper/feature_extraction_whisper.py
@@ -215,6 +215,29 @@ def _np_extract_fbank_features(self, waveform: np.array) -> np.ndarray:
 
         return log_spec
 
+    @staticmethod
+    # Copied from transformers.models.wav2vec2.feature_extraction_wav2vec2.Wav2Vec2FeatureExtractor.zero_mean_unit_var_norm
+    def zero_mean_unit_var_norm(
+        input_values: List[np.ndarray], attention_mask: List[np.ndarray], padding_value: float = 0.0
+    ) -> List[np.ndarray]:
+        """
+        Every array in the list is normalized to have zero mean and unit variance
+        """
+        if attention_mask is not None:
+            attention_mask = np.array(attention_mask, np.int32)
+            normed_input_values = []
+
+            for vector, length in zip(input_values, attention_mask.sum(-1)):
+                normed_slice = (vector - vector[:length].mean()) / np.sqrt(vector[:length].var() + 1e-7)
+                if length < normed_slice.shape[0]:
+                    normed_slice[length:] = padding_value
+
+                normed_input_values.append(normed_slice)
+        else:
+            normed_input_values = [(x - x.mean()) / np.sqrt(x.var() + 1e-7) for x in input_values]
+
+        return normed_input_values
+
     def __call__(
         self,
         raw_speech: Union[np.ndarray, List[float], List[np.ndarray], List[List[float]]],
@@ -225,6 +248,7 @@ def __call__(
         padding: Optional[str] = "max_length",
         max_length: Optional[int] = None,
         sampling_rate: Optional[int] = None,
+        do_normalize: Optional[bool] = None,
         **kwargs,
     ) -> BatchFeature:
         """
@@ -266,6 +290,9 @@ def __call__(
                 pipeline.
             padding_value (`float`, defaults to 0.0):
                 The value that is used to fill the padding values / vectors.
+            do_normalize (`bool`, *optional*, defaults to `False`):
+                Whether or not to zero-mean unit-variance normalize the input. Normalizing can help to significantly
+                improve the performance of the model.
         """
 
         if sampling_rate is not None:
@@ -312,6 +339,18 @@ def __call__(
         # make sure list is in array format
         input_features = padded_inputs.get("input_features").transpose(2, 0, 1)
 
+        if return_attention_mask:
+            # rescale from sample (48000) to feature (3000)
+            padded_inputs["attention_mask"] = padded_inputs["attention_mask"][:, :: self.hop_length]
+
+        # zero-mean and unit-variance normalization
+        if do_normalize:
+            padded_inputs["input_features"] = self.zero_mean_unit_var_norm(
+                padded_inputs["input_features"],
+                attention_mask=padded_inputs["attention_mask"],
+                padding_value=self.padding_value,
+            )
+
         input_features = [self._np_extract_fbank_features(waveform) for waveform in input_features[0]]
 
         if isinstance(input_features[0], List):
diff --git a/tests/models/whisper/test_feature_extraction_whisper.py b/tests/models/whisper/test_feature_extraction_whisper.py
index f1ef36b1f28f..57c12b86ddbc 100644
--- a/tests/models/whisper/test_feature_extraction_whisper.py
+++ b/tests/models/whisper/test_feature_extraction_whisper.py
@@ -21,6 +21,7 @@
 import unittest
 
 import numpy as np
+from datasets import load_dataset
 
 from transformers import is_speech_available
 from transformers.testing_utils import check_json_file_has_correct_format, require_torch, require_torchaudio
@@ -198,8 +199,6 @@ def test_double_precision_pad(self):
             self.assertTrue(pt_processed.input_features.dtype == torch.float32)
 
     def _load_datasamples(self, num_samples):
-        from datasets import load_dataset
-
         ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
         # automatic decoding with librispeech
         speech_samples = ds.sort("id").select(range(num_samples))[:num_samples]["audio"]
@@ -222,3 +221,12 @@ def test_integration(self):
         feaure_extractor = WhisperFeatureExtractor()
         input_features = feaure_extractor(input_speech, return_tensors="pt").input_features
         self.assertTrue(torch.allclose(input_features[0, 0, :30], EXPECTED_INPUT_FEATURES, atol=1e-4))
+
+    def test_zero_mean_unit_variance_normalization_trunc_np_longest(self):
+        feat_extract = self.feature_extraction_class(**self.feat_extract_tester.prepare_feat_extract_dict())
+        audio = self._load_datasamples(1)[0]
+        audio = ((audio - audio.min()) / (audio.max() - audio.min())) * 65535  # Rescale to [0, 65535] to show issue
+        audio = feat_extract.zero_mean_unit_var_norm([audio], attention_mask=None)[0]
+
+        self.assertTrue(np.all(np.mean(audio) < 1e-3))
+        self.assertTrue(np.all(np.abs(np.var(audio) - 1) < 1e-3))