diff --git a/backend/python/whisperx/backend.py b/backend/python/whisperx/backend.py
index 7318e10b7726..dc62022862b2 100644
--- a/backend/python/whisperx/backend.py
+++ b/backend/python/whisperx/backend.py
@@ -16,6 +16,7 @@
 sys.path.insert(0, os.path.join(os.path.dirname(__file__), '..', 'common'))
 sys.path.insert(0, os.path.join(os.path.dirname(__file__), 'common'))
 from grpc_auth import get_auth_interceptors
+from transcript_utils import require_diarization_token, seconds_to_nanoseconds
 
 
 
@@ -81,6 +82,11 @@ def AudioTranscription(self, request, context):
         import whisperx
         from whisperx.diarize import DiarizationPipeline
 
+        try:
+            require_diarization_token(request.diarize, self.hf_token)
+        except ValueError as err:
+            context.abort(grpc.StatusCode.FAILED_PRECONDITION, str(err))
+
         resultSegments = []
         text = ""
         try:
@@ -117,8 +123,8 @@ def AudioTranscription(self, request, context):
             # Build result segments
             for idx, seg in enumerate(transcript["segments"]):
                 seg_text = seg.get("text", "")
-                start = int(seg.get("start", 0))
-                end = int(seg.get("end", 0))
+                start = seconds_to_nanoseconds(seg.get("start", 0))
+                end = seconds_to_nanoseconds(seg.get("end", 0))
                 speaker = seg.get("speaker", "")
 
                 resultSegments.append(backend_pb2.TranscriptSegment(
diff --git a/backend/python/whisperx/test_transcript_utils.py b/backend/python/whisperx/test_transcript_utils.py
new file mode 100644
index 000000000000..debe2ea6ecf7
--- /dev/null
+++ b/backend/python/whisperx/test_transcript_utils.py
@@ -0,0 +1,25 @@
+import unittest
+
+import transcript_utils
+
+
+class TestTranscriptUtils(unittest.TestCase):
+    def test_diarization_requires_hugging_face_token(self):
+        with self.assertRaisesRegex(
+            ValueError,
+            "HF_TOKEN is required for WhisperX diarization",
+        ):
+            transcript_utils.require_diarization_token(True, None)
+
+    def test_diarization_does_not_require_token_when_disabled(self):
+        transcript_utils.require_diarization_token(False, None)
+
+    def test_seconds_are_serialized_as_nanoseconds(self):
+        self.assertEqual(
+            transcript_utils.seconds_to_nanoseconds(3.25),
+            3_250_000_000,
+        )
+
+
+if __name__ == "__main__":
+    unittest.main()
diff --git a/backend/python/whisperx/transcript_utils.py b/backend/python/whisperx/transcript_utils.py
new file mode 100644
index 000000000000..a8ac5751057e
--- /dev/null
+++ b/backend/python/whisperx/transcript_utils.py
@@ -0,0 +1,12 @@
+"""Helpers for WhisperX transcript responses."""
+
+
+def require_diarization_token(diarize, token):
+    """Reject diarization when WhisperX cannot load its gated pipeline."""
+    if diarize and not token:
+        raise ValueError("HF_TOKEN is required for WhisperX diarization")
+
+
+def seconds_to_nanoseconds(seconds):
+    """Convert WhisperX timestamps to the duration unit used by LocalAI."""
+    return int(seconds * 1_000_000_000)
diff --git a/docs/content/features/audio-to-text.md b/docs/content/features/audio-to-text.md
index 0312d392a98e..5a5e833cf3a3 100644
--- a/docs/content/features/audio-to-text.md
+++ b/docs/content/features/audio-to-text.md
@@ -11,6 +11,7 @@ The transcription endpoint allows to convert audio files to text. The endpoint s
 - **[whisper.cpp](https://github.com/ggerganov/whisper.cpp)**: A C++ library for audio transcription (default)
 - **moonshine**: Ultra-fast transcription engine optimized for low-end devices
 - **faster-whisper**: Fast Whisper implementation with CTranslate2
+- **WhisperX**: Whisper transcription with word alignment and optional speaker diarization. Set `HF_TOKEN` and pass `diarize=true` to load WhisperX's gated pyannote diarization pipeline.
 - **[parakeet-cpp](https://github.com/mudler/parakeet.cpp)**: A C++/ggml port of NVIDIA NeMo Parakeet (FastConformer TDT/CTC/RNNT/hybrid). Runs quantized GGUFs on CPU or GPU, emits word-level timestamps, and supports cache-aware streaming (the `realtime_eou` model surfaces end-of-utterance events).
 - **llama-cpp**: Route transcription to any multimodal-audio GGUF model served by the `llama-cpp` backend (e.g. [Qwen3-ASR](https://huggingface.co/ggml-org/Qwen3-ASR-0.6B-GGUF), Voxtral, Qwen2-Audio). Under the hood the request is converted into a chat completion with the audio attached via the model's audio encoder - the same path the upstream llama.cpp server uses. Set `backend: llama-cpp` in the model YAML and point `mmproj` at the matching audio encoder.
 - **voxtral**: Voxtral-family models served by a dedicated backend
@@ -109,7 +110,7 @@ In addition to `file` and `model`, the endpoint accepts the following multipart
 | `timestamp_granularities[]` | Multi-value form field: `word` and/or `segment`. Honored when the backend produces the requested granularity. |
 | `response_format` | One of `json` (default for backwards-compat), `verbose_json`, `text`, `srt`, `vtt`, `lrc`. |
 | `stream` | When `true`, the endpoint emits an SSE stream of `transcript.text.delta` events followed by a final `transcript.text.done` event. |
-| `diarize` | LocalAI extension - speaker diarization (whisper.cpp only). |
+| `diarize` | LocalAI extension - speaker diarization. WhisperX requires `HF_TOKEN`; requests fail with `FailedPrecondition` when it is missing. |
 
 The response body for `verbose_json` includes `text`, `language`, `duration`, and `segments[]` (with `speaker` populated when diarization is enabled).