diff --git a/cosyvoice/cli/frontend.py b/cosyvoice/cli/frontend.py
index 6d397cc99..f8d75d92b 100644
--- a/cosyvoice/cli/frontend.py
+++ b/cosyvoice/cli/frontend.py
@@ -17,13 +17,13 @@
 import onnxruntime
 import torch
 import numpy as np
-import whisper
 from typing import Callable
 import torchaudio.compliance.kaldi as kaldi
 import os
 import re
 import inflect
 from cosyvoice.utils.file_utils import logging, load_wav
+from cosyvoice.utils.audio_utils import log_mel_spectrogram
 from cosyvoice.utils.frontend_utils import contains_chinese, replace_blank, replace_corner_mark, remove_bracket, spell_out_number, split_paragraph, is_only_punctuation
 
 
@@ -95,7 +95,7 @@ def _extract_text_token_generator(self, text_generator):
     def _extract_speech_token(self, prompt_wav):
         speech = load_wav(prompt_wav, 16000)
         assert speech.shape[1] / 16000 <= 30, 'do not support extract speech token for audio longer than 30s'
-        feat = whisper.log_mel_spectrogram(speech, n_mels=128)
+        feat = log_mel_spectrogram(speech, n_mels=128)
         speech_token = self.speech_tokenizer_session.run(None,
                                                          {self.speech_tokenizer_session.get_inputs()[0].name:
                                                           feat.detach().cpu().numpy(),
diff --git a/cosyvoice/dataset/processor.py b/cosyvoice/dataset/processor.py
index deba209ba..ce1491f79 100644
--- a/cosyvoice/dataset/processor.py
+++ b/cosyvoice/dataset/processor.py
@@ -17,12 +17,12 @@
 import pyarrow.parquet as pq
 from io import BytesIO
 import numpy as np
-import whisper
 import torch
 import torchaudio
 from torch.nn.utils.rnn import pad_sequence
 import torch.nn.functional as F
 import pyworld as pw
+from cosyvoice.utils.audio_utils import log_mel_spectrogram
 from cosyvoice.utils.onnx import embedding_extractor, online_feature
 
 AUDIO_FORMAT_SETS = {'flac', 'mp3', 'm4a', 'ogg', 'opus', 'wav', 'wma'}
@@ -193,7 +193,7 @@ def compute_whisper_fbank(data, num_frames=-1, mode='train'):
         if num_frames != -1:
             assert sample['speech'].shape[1] % num_frames == 0, 'speech length is not aligned with speech_token'
         sample['speech_16k'] = torchaudio.transforms.Resample(orig_freq=sample['sample_rate'], new_freq=16000)(sample['speech'])
-        sample['whisper_feat'] = whisper.log_mel_spectrogram(sample['speech_16k'], n_mels=128).squeeze(dim=0).transpose(0, 1)
+        sample['whisper_feat'] = log_mel_spectrogram(sample['speech_16k'], n_mels=128).squeeze(dim=0).transpose(0, 1)
         yield sample
 
 
diff --git a/cosyvoice/tokenizer/tokenizer.py b/cosyvoice/tokenizer/tokenizer.py
index 6ecf4ae84..fab635231 100644
--- a/cosyvoice/tokenizer/tokenizer.py
+++ b/cosyvoice/tokenizer/tokenizer.py
@@ -4,8 +4,6 @@
 from typing import Optional
 import torch
 from transformers import AutoTokenizer
-from whisper.tokenizer import Tokenizer
-
 import tiktoken
 
 LANGUAGES = {
@@ -213,7 +211,7 @@ def get_tokenizer(
     num_languages: int = 99,
     language: Optional[str] = None,
     task: Optional[str] = None,  # Literal["transcribe", "translate", None]
-) -> Tokenizer:
+):
     if language is not None:
         language = language.lower()
         if language not in LANGUAGES:
@@ -233,6 +231,7 @@ def get_tokenizer(
 
     encoding = get_encoding(name=encoding_name, num_languages=num_languages)
 
+    from whisper.tokenizer import Tokenizer
     return Tokenizer(
         encoding=encoding, num_languages=num_languages, language=language, task=task
     )
diff --git a/cosyvoice/utils/audio_utils.py b/cosyvoice/utils/audio_utils.py
new file mode 100644
index 000000000..b7330cf28
--- /dev/null
+++ b/cosyvoice/utils/audio_utils.py
@@ -0,0 +1,56 @@
+# Copyright (c) 2024 Alibaba Inc (authors: Xiang Lyu)
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#   http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+import torch
+import torchaudio
+
+
+def log_mel_spectrogram(audio, n_mels=128, n_fft=400, hop_length=160, sample_rate=16000):
+    """Compute a log-mel spectrogram from a waveform tensor.
+
+    This is a drop-in replacement for ``whisper.log_mel_spectrogram`` that uses
+    only ``torch`` and ``torchaudio``, avoiding the heavy ``openai-whisper``
+    dependency.  The output is numerically equivalent for the default Whisper
+    parameters (n_fft=400, hop_length=160, sample_rate=16000).
+
+    Args:
+        audio: 1-D or 2-D float tensor of raw audio at *sample_rate* Hz.
+        n_mels: Number of mel-frequency bins.
+        n_fft: FFT window size.
+        hop_length: Hop length for STFT.
+        sample_rate: Expected sample rate of *audio*.
+
+    Returns:
+        Tensor of shape ``(n_mels, n_frames)`` (if 1-D input) or
+        ``(batch, n_mels, n_frames)`` (if 2-D input).
+    """
+    window = torch.hann_window(n_fft).to(audio.device)
+    stft = torch.stft(audio, n_fft, hop_length, window=window, return_complex=True)
+    magnitudes = stft[..., :-1].abs() ** 2
+
+    mel_filters = torchaudio.functional.melscale_fbanks(
+        n_freqs=n_fft // 2 + 1,
+        f_min=0.0,
+        f_max=sample_rate / 2.0,
+        n_mels=n_mels,
+        sample_rate=sample_rate,
+        norm="slaney",
+        mel_scale="slaney",
+    ).to(audio.device)
+
+    mel_spec = mel_filters.T @ magnitudes
+    log_spec = torch.clamp(mel_spec, min=1e-10).log10()
+    log_spec = torch.maximum(log_spec, log_spec.amax() - 8.0)
+    log_spec = (log_spec + 4.0) / 4.0
+    return log_spec
diff --git a/requirements.txt b/requirements.txt
index 989bccf61..366e9054e 100644
--- a/requirements.txt
+++ b/requirements.txt
@@ -22,7 +22,6 @@ omegaconf==2.3.0
 onnx==1.16.0
 onnxruntime-gpu==1.18.0; sys_platform == 'linux'
 onnxruntime==1.18.0; sys_platform == 'darwin' or sys_platform == 'win32'
-openai-whisper==20231117
 protobuf==4.25
 pyarrow==18.1.0
 pydantic==2.7.0
diff --git a/tools/extract_speech_token.py b/tools/extract_speech_token.py
index 976a23b48..e1404174c 100755
--- a/tools/extract_speech_token.py
+++ b/tools/extract_speech_token.py
@@ -20,7 +20,7 @@
 import onnxruntime
 import numpy as np
 import torchaudio
-import whisper
+from cosyvoice.utils.audio_utils import log_mel_spectrogram
 
 
 def single_job(utt):
@@ -34,7 +34,7 @@ def single_job(utt):
         logging.warning('do not support extract speech token for audio longer than 30s')
         speech_token = []
     else:
-        feat = whisper.log_mel_spectrogram(audio, n_mels=128)
+        feat = log_mel_spectrogram(audio, n_mels=128)
         speech_token = ort_session.run(None, {ort_session.get_inputs()[0].name: feat.detach().cpu().numpy(),
                                               ort_session.get_inputs()[1].name: np.array([feat.shape[2]], dtype=np.int32)})[0].flatten().tolist()
     return utt, speech_token