diff --git a/docs/contributing/model/transcription.md b/docs/contributing/model/transcription.md
index a23de100da39..db868686e2a7 100644
--- a/docs/contributing/model/transcription.md
+++ b/docs/contributing/model/transcription.md
@@ -193,7 +193,7 @@ Provide a fast duration→token estimate to improve streaming usage statistics:
 
 The API server takes care of basic audio I/O and optional chunking before building prompts:
 
-- Resampling: Input audio is resampled to `SpeechToTextConfig.sample_rate` using `librosa`.
+- Resampling: Input audio is resampled to `SpeechToTextConfig.sample_rate` using `AudioResampler`.
 - Chunking: If `SpeechToTextConfig.allow_audio_chunking` is True and the duration exceeds `max_audio_clip_s`, the server splits the audio into overlapping chunks and generates a prompt per chunk. Overlap is controlled by `overlap_chunk_second`.
 - Energy-aware splitting: When `min_energy_split_window_size` is set, the server finds low-energy regions to minimize cutting within words.
 
@@ -206,8 +206,8 @@ Relevant server logic:
     async def _preprocess_speech_to_text(...):
         language = self.model_cls.validate_language(request.language)
         ...
-        y, sr = librosa.load(bytes_, sr=self.asr_config.sample_rate)
-        duration = librosa.get_duration(y=y, sr=sr)
+        y, sr = load_audio(bytes_, sr=self.asr_config.sample_rate)
+        duration = get_audio_duration(y=y, sr=sr)
         do_split_audio = (self.asr_config.allow_audio_chunking
                         and duration > self.asr_config.max_audio_clip_s)
         chunks = [y] if not do_split_audio else self._split_audio(y, int(sr))
diff --git a/docs/features/multimodal_inputs.md b/docs/features/multimodal_inputs.md
index ee82c34fa0eb..d9b49f7cb7f9 100644
--- a/docs/features/multimodal_inputs.md
+++ b/docs/features/multimodal_inputs.md
@@ -300,12 +300,12 @@ Full example: [examples/offline_inference/audio_language.py](../../examples/offl
 Speech-to-text models like Whisper have a maximum audio length they can process (typically 30 seconds). For longer audio files, vLLM provides a utility to intelligently split audio into chunks at quiet points to minimize cutting through speech.
 
 ```python
-import librosa
 from vllm import LLM, SamplingParams
 from vllm.multimodal.audio import split_audio
+from vllm.multimodal.media.audio import load_audio
 
 # Load long audio file
-audio, sr = librosa.load("long_audio.wav", sr=16000)
+audio, sr = load_audio("long_audio.wav", sr=16000)
 
 # Split into chunks at low-energy (quiet) regions
 chunks = split_audio(
@@ -832,7 +832,7 @@ Then, you can use the OpenAI client as follows:
         base_url=openai_api_base,
     )
 
-    # Any format supported by librosa is supported
+    # Any format supported by soundfile/PyAV is supported
     audio_url = AudioAsset("winning_call").url
     audio_base64 = encode_base64_content_from_url(audio_url)
 
diff --git a/examples/online_serving/openai_chat_completion_client_for_multimodal.py b/examples/online_serving/openai_chat_completion_client_for_multimodal.py
index c4407923ed2d..ba3adf55c90e 100644
--- a/examples/online_serving/openai_chat_completion_client_for_multimodal.py
+++ b/examples/online_serving/openai_chat_completion_client_for_multimodal.py
@@ -267,7 +267,7 @@ def run_audio(model: str, max_completion_tokens: int) -> None:
                     {
                         "type": "input_audio",
                         "input_audio": {
-                            # Any format supported by librosa is supported
+                            # Any format supported by soundfile/PyAV is supported
                             "data": audio_base64,
                             "format": "wav",
                         },
@@ -292,7 +292,7 @@ def run_audio(model: str, max_completion_tokens: int) -> None:
                     {
                         "type": "audio_url",
                         "audio_url": {
-                            # Any format supported by librosa is supported
+                            # Any format supported by soundfile/PyAV is supported
                             "url": audio_url
                         },
                     },
@@ -316,7 +316,7 @@ def run_audio(model: str, max_completion_tokens: int) -> None:
                     {
                         "type": "audio_url",
                         "audio_url": {
-                            # Any format supported by librosa is supported
+                            # Any format supported by soundfile/PyAV is supported
                             "url": f"data:audio/ogg;base64,{audio_base64}"
                         },
                     },
diff --git a/examples/online_serving/openai_realtime_client.py b/examples/online_serving/openai_realtime_client.py
index 2bd3c7e60d55..fda3d7cb4564 100644
--- a/examples/online_serving/openai_realtime_client.py
+++ b/examples/online_serving/openai_realtime_client.py
@@ -12,7 +12,6 @@
 Requirements:
 - vllm with audio support
 - websockets
-- librosa
 - numpy
 
 The script:
@@ -26,12 +25,12 @@
 import asyncio
 import json
 
-import librosa
 import numpy as np
 import pybase64 as base64
 import websockets
 
 from vllm.assets.audio import AudioAsset
+from vllm.multimodal.media.audio import load_audio
 
 
 def audio_to_pcm16_base64(audio_path: str) -> str:
@@ -39,7 +38,7 @@ def audio_to_pcm16_base64(audio_path: str) -> str:
     Load an audio file and convert it to base64-encoded PCM16 @ 16kHz.
     """
     # Load audio and resample to 16kHz mono
-    audio, _ = librosa.load(audio_path, sr=16000, mono=True)
+    audio, _ = load_audio(audio_path, sr=16000, mono=True)
     # Convert to PCM16
     pcm16 = (audio * 32767).astype(np.int16)
     # Encode as base64
diff --git a/tests/entrypoints/openai/correctness/test_transcription_api_correctness.py b/tests/entrypoints/openai/correctness/test_transcription_api_correctness.py
index 194c52eae35e..f847b7b0d88d 100644
--- a/tests/entrypoints/openai/correctness/test_transcription_api_correctness.py
+++ b/tests/entrypoints/openai/correctness/test_transcription_api_correctness.py
@@ -13,7 +13,6 @@
 import time
 from statistics import mean, median
 
-import librosa
 import pytest
 import soundfile
 import torch
@@ -21,6 +20,7 @@
 from evaluate import load
 from transformers.models.whisper.english_normalizer import EnglishTextNormalizer
 
+from vllm.multimodal.audio import get_audio_duration
 from vllm.tokenizers import get_tokenizer
 
 from ....models.registry import HF_EXAMPLE_MODELS
@@ -84,7 +84,7 @@ async def process_dataset(model, client, data, concurrent_request):
         trust_remote_code=model_info.trust_remote_code,
     )
 
-    # Warmup call as the first `librosa.load` server-side is quite slow.
+    # Warmup call as the first `load_audio` server-side is quite slow.
     audio, sr = data[0]["audio"]["array"], data[0]["audio"]["sampling_rate"]
     _ = await bound_transcribe(sem, client, tokenizer, (audio, sr), "")
 
@@ -118,7 +118,7 @@ def print_performance_metrics(results, total_time):
 
 def add_duration(sample):
     y, sr = sample["audio"]["array"], sample["audio"]["sampling_rate"]
-    sample["duration_ms"] = librosa.get_duration(y=y, sr=sr) * 1000
+    sample["duration_ms"] = get_audio_duration(y=y, sr=sr) * 1000
     return sample
 
 
diff --git a/tests/entrypoints/openai/realtime/test_realtime_validation.py b/tests/entrypoints/openai/realtime/test_realtime_validation.py
index bb6b02f5c99e..e317090fa543 100644
--- a/tests/entrypoints/openai/realtime/test_realtime_validation.py
+++ b/tests/entrypoints/openai/realtime/test_realtime_validation.py
@@ -5,7 +5,6 @@
 import json
 import warnings
 
-import librosa
 import numpy as np
 import pybase64 as base64
 import pytest
@@ -14,6 +13,7 @@
 from tests.entrypoints.openai.conftest import add_attention_backend
 from tests.utils import ROCM_ENV_OVERRIDES, ROCM_EXTRA_ARGS, RemoteOpenAIServer
 from vllm.assets.audio import AudioAsset
+from vllm.multimodal.media.audio import load_audio
 
 # Increase engine iteration timeout for ROCm where first-use JIT compilation
 # can exceed the default 60s, causing a silent deadlock in feed_tokens.
@@ -56,7 +56,7 @@ async def send_event(ws, event: dict) -> None:
 def mary_had_lamb_audio_chunks() -> list[str]:
     """Audio split into ~1 second chunks for streaming."""
     path = AudioAsset("mary_had_lamb").get_local_path()
-    audio, _ = librosa.load(str(path), sr=16000, mono=True)
+    audio, _ = load_audio(str(path), sr=16000, mono=True)
 
     # Split into ~0.1 second chunks (1600 samples at 16kHz)
     chunk_size = 1600
diff --git a/tests/entrypoints/openai/speech_to_text/test_transcription_validation_whisper.py b/tests/entrypoints/openai/speech_to_text/test_transcription_validation_whisper.py
index 8dba1b59742b..511179f7fcb1 100644
--- a/tests/entrypoints/openai/speech_to_text/test_transcription_validation_whisper.py
+++ b/tests/entrypoints/openai/speech_to_text/test_transcription_validation_whisper.py
@@ -6,7 +6,6 @@
 import io
 import json
 
-import librosa
 import numpy as np
 import openai
 import pytest
@@ -14,6 +13,7 @@
 import soundfile as sf
 
 from tests.utils import RemoteOpenAIServer
+from vllm.multimodal.media.audio import load_audio
 from vllm.platforms import current_platform
 
 MODEL_NAME = "openai/whisper-large-v3-turbo"
@@ -134,7 +134,7 @@ async def test_bad_requests(mary_had_lamb, whisper_client):
 @pytest.mark.asyncio
 async def test_long_audio_request(mary_had_lamb, whisper_client):
     mary_had_lamb.seek(0)
-    audio, sr = librosa.load(mary_had_lamb)
+    audio, sr = load_audio(mary_had_lamb)
     # Add small silence after each audio for repeatability in the split process
     audio = np.pad(audio, (0, 1600))
     repeated_audio = np.tile(audio, 10)
diff --git a/tests/entrypoints/openai/speech_to_text/test_translation_validation.py b/tests/entrypoints/openai/speech_to_text/test_translation_validation.py
index 16b9614d957e..a8b17bf34324 100644
--- a/tests/entrypoints/openai/speech_to_text/test_translation_validation.py
+++ b/tests/entrypoints/openai/speech_to_text/test_translation_validation.py
@@ -7,7 +7,6 @@
 import json
 
 import httpx
-import librosa
 import numpy as np
 import openai
 import pytest
@@ -17,6 +16,7 @@
 from tests.entrypoints.openai.conftest import add_attention_backend
 from tests.utils import RemoteOpenAIServer
 from vllm.logger import init_logger
+from vllm.multimodal.media.audio import load_audio
 
 logger = init_logger(__name__)
 
@@ -264,7 +264,7 @@ async def test_long_audio_request(foscolo, client_and_model):
     if model_name == "google/gemma-3n-E2B-it":
         pytest.skip("Gemma3n does not support long audio requests")
     foscolo.seek(0)
-    audio, sr = librosa.load(foscolo)
+    audio, sr = load_audio(foscolo)
     repeated_audio = np.tile(audio, 2)
     # Repeated audio to buffer
     buffer = io.BytesIO()
diff --git a/tests/models/multimodal/generation/test_phi4mm.py b/tests/models/multimodal/generation/test_phi4mm.py
index 7f1a12f04474..1a4fb35a28aa 100644
--- a/tests/models/multimodal/generation/test_phi4mm.py
+++ b/tests/models/multimodal/generation/test_phi4mm.py
@@ -4,7 +4,6 @@
 import os
 from collections.abc import Sequence
 
-import librosa
 import pytest
 import regex as re
 from huggingface_hub import snapshot_download
@@ -14,6 +13,7 @@
 from vllm.logprobs import SampleLogprobs
 from vllm.lora.request import LoRARequest
 from vllm.multimodal.image import convert_image_mode, rescale_image_size
+from vllm.multimodal.media.audio import load_audio
 
 from ....conftest import (
     IMAGE_ASSETS,
@@ -290,7 +290,7 @@ def test_vision_speech_models(
     num_logprobs: int,
 ) -> None:
     # use the example speech question so that the model outputs are reasonable
-    audio = librosa.load(speech_question, sr=None)
+    audio = load_audio(speech_question, sr=None)
     image = convert_image_mode(ImageAsset("cherry_blossom").pil_image, "RGB")
 
     inputs_vision_speech = [
diff --git a/tests/models/multimodal/generation/test_whisper.py b/tests/models/multimodal/generation/test_whisper.py
index babf7e7a4978..186e7e054ce1 100644
--- a/tests/models/multimodal/generation/test_whisper.py
+++ b/tests/models/multimodal/generation/test_whisper.py
@@ -4,11 +4,11 @@
 from collections.abc import Sequence
 from typing import Any
 
-import librosa
 import pytest
 from transformers import AutoModelForSpeechSeq2Seq
 
 from vllm.assets.audio import AudioAsset
+from vllm.multimodal.audio import AudioResampler
 from vllm.platforms import current_platform
 
 from ....conftest import HfRunner, PromptAudioInput, VllmRunner
@@ -93,13 +93,12 @@ def run_test(
 def resampled_assets() -> list[tuple[Any, int]]:
     audio_assets = [AudioAsset("mary_had_lamb"), AudioAsset("winning_call")]
     sampled_assets = []
+    resampler = AudioResampler(target_sr=WHISPER_SAMPLE_RATE)
     for asset in audio_assets:
         audio, orig_sr = asset.audio_and_sample_rate
         # Resample to Whisper's expected sample rate (16kHz)
         if orig_sr != WHISPER_SAMPLE_RATE:
-            audio = librosa.resample(
-                audio, orig_sr=orig_sr, target_sr=WHISPER_SAMPLE_RATE
-            )
+            audio = resampler.resample(audio, orig_sr=orig_sr)
         sampled_assets.append(
             (audio, WHISPER_SAMPLE_RATE),
         )
diff --git a/tests/multimodal/media/test_audio.py b/tests/multimodal/media/test_audio.py
index 4361066ab885..3729e71f24e7 100644
--- a/tests/multimodal/media/test_audio.py
+++ b/tests/multimodal/media/test_audio.py
@@ -3,12 +3,12 @@
 from pathlib import Path
 from unittest.mock import patch
 
-import librosa
 import numpy as np
 import pybase64 as base64
 import pytest
 
 from vllm.multimodal.media import AudioMediaIO
+from vllm.multimodal.media.audio import load_audio
 
 from ...conftest import AudioTestAssets
 
@@ -73,6 +73,6 @@ def test_audio_media_io_from_video(video_assets):
     video_path = video_assets[0].video_path
     with open(video_path, "rb") as f:
         audio, sr = audio_io.load_bytes(f.read())
-    audio_ref, sr_ref = librosa.load(video_path, sr=None)
+    audio_ref, sr_ref = load_audio(video_path, sr=None)
     assert sr == sr_ref
     np.testing.assert_allclose(audio_ref, audio, atol=1e-4)
diff --git a/vllm/multimodal/media/audio.py b/vllm/multimodal/media/audio.py
index 26d58d13a731..37b8662a76b6 100644
--- a/vllm/multimodal/media/audio.py
+++ b/vllm/multimodal/media/audio.py
@@ -29,9 +29,9 @@
     soundfile = PlaceholderModule("soundfile")  # type: ignore[assignment]
 
 
-# Public libsndfile error codes exposed via `soundfile.LibsndfileError.code`, soundfile
-# being librosa's main backend. Used to validate if an audio loading error is due to a
-# server error vs a client error (invalid audio file).
+# Public libsndfile error codes exposed via `soundfile.LibsndfileError.code`,
+# soundfile being the main audio loading backend. Used to validate if an audio
+# loading error is due to a server error vs a client error (invalid audio file).
 # 0 = sf_error(NULL) race condition: when multiple threads fail sf_open_virtual
 #     concurrently, one thread may clear the global error before another reads it,
 #     producing code=0 ("Garbled error message from libsndfile" in soundfile).
diff --git a/vllm/transformers_utils/processors/cohere_asr.py b/vllm/transformers_utils/processors/cohere_asr.py
index f742074a4e3d..e1257de4e735 100644
--- a/vllm/transformers_utils/processors/cohere_asr.py
+++ b/vllm/transformers_utils/processors/cohere_asr.py
@@ -4,11 +4,11 @@
 import math
 import random
 
-import librosa
 import numpy as np
 import torch
 import torch.nn.functional as F
 from torch import nn
+from torchaudio.functional import melscale_fbanks
 from transformers import AutoFeatureExtractor, AutoProcessor, BatchFeature
 from transformers.feature_extraction_sequence_utils import (
     SequenceFeatureExtractor,
@@ -129,17 +129,15 @@ def __init__(
         self.pad_min_duration = 0.0
         self.pad_direction = "both"
 
-        filterbanks = torch.tensor(
-            librosa.filters.mel(
-                sr=sample_rate,
-                n_fft=self.n_fft,
-                n_mels=nfilt,
-                fmin=lowfreq,
-                fmax=highfreq,
-                norm=mel_norm,
-            ),
-            dtype=torch.float,
-        ).unsqueeze(0)
+        filterbanks = melscale_fbanks(
+            n_freqs=self.n_fft // 2 + 1,
+            f_min=lowfreq,
+            f_max=highfreq,
+            n_mels=nfilt,
+            sample_rate=sample_rate,
+            norm=mel_norm,
+            mel_scale="slaney",
+        ).T.unsqueeze(0)
         self.register_buffer("fb", filterbanks)
 
         # Calculate maximum sequence length