diff --git a/AudioQnA/audio/docker/asr/asr.py b/AudioQnA/audio/docker/asr/asr.py
index e5e74edee..17b4f456c 100644
--- a/AudioQnA/audio/docker/asr/asr.py
+++ b/AudioQnA/audio/docker/asr/asr.py
@@ -8,6 +8,7 @@
 import contextlib
 import os
 import time
+import urllib.request
 
 import numpy as np
 import torch
@@ -19,7 +20,13 @@
 class AudioSpeechRecognition:
     """Convert audio to text."""
 
-    def __init__(self, model_name_or_path="openai/whisper-small", bf16=False, language=None, device="cpu"):
+    def __init__(self, model_name_or_path="openai/whisper-small", bf16=False, language="english", device="cpu"):
+        if device == "hpu":
+            # Explicitly link HPU with Torch
+            from optimum.habana.transformers.modeling_utils import adapt_transformers_to_gaudi
+
+            adapt_transformers_to_gaudi()
+
         self.device = device
         asr_model_name_or_path = os.environ.get("ASR_MODEL_PATH", model_name_or_path)
         print("Downloading model: {}".format(asr_model_name_or_path))
@@ -33,6 +40,12 @@ def __init__(self, model_name_or_path="openai/whisper-small", bf16=False, langua
             self.model = ipex.optimize(self.model, dtype=torch.bfloat16)
         self.language = language
 
+        if device == "hpu":
+            # do hpu graph warmup with a long enough input audio
+            # whisper has a receptive field of 30 seconds
+            # here we select a relatively long audio (~15 sec) to quickly warmup
+            self._warmup_whisper_hpu_graph("https://paddlespeech.bj.bcebos.com/Parakeet/docs/demos/labixiaoxin.wav")
+
     def _audiosegment_to_librosawav(self, audiosegment):
         # https://github.com/jiaaro/pydub/blob/master/API.markdown#audiosegmentget_array_of_samples
         # This way is faster than librosa.load or HuggingFace Dataset wrapper
@@ -45,16 +58,27 @@ def _audiosegment_to_librosawav(self, audiosegment):
 
         return fp_arr
 
+    def _warmup_whisper_hpu_graph(self, url):
+        print("[ASR] fetch warmup audio...")
+        urllib.request.urlretrieve(
+            url,
+            "warmup.wav",
+        )
+        print("[ASR] warmup...")
+        waveform = AudioSegment.from_file("warmup.wav").set_frame_rate(16000)
+        waveform = self._audiosegment_to_librosawav(waveform)
+        # pylint: disable=E1101
+        inputs = self.processor.feature_extractor(
+            waveform, return_tensors="pt", sampling_rate=16_000
+        ).input_features.to(self.device)
+        _ = self.model.generate(inputs, language="chinese")
+
     def audio2text(self, audio_path):
         """Convert audio to text.
 
         audio_path: the path to the input audio, e.g. ~/xxx.mp3
         """
         start = time.time()
-        if audio_path.split(".")[-1] in ["flac", "ogg", "aac", "m4a"]:
-            audio_path = self._convert_audio_type(audio_path)
-        elif audio_path.split(".")[-1] not in ["mp3", "wav"]:
-            raise Exception("[ASR ERROR] Audio format not supported!")
 
         try:
             waveform = AudioSegment.from_file(audio_path).set_frame_rate(16000)
@@ -69,20 +93,10 @@ def audio2text(self, audio_path):
             waveform, return_tensors="pt", sampling_rate=16_000
         ).input_features.to(self.device)
         with torch.cpu.amp.autocast() if self.bf16 else contextlib.nullcontext():
-            if self.language is None:
-                predicted_ids = self.model.generate(inputs)
-            elif self.language == "auto":
-                self.model.config.forced_decoder_ids = None
-                predicted_ids = self.model.generate(inputs)
-            else:
-                self.forced_decoder_ids = self.processor.get_decoder_prompt_ids(
-                    language=self.language, task="transcribe"
-                )
-                self.model.config.forced_decoder_ids = self.forced_decoder_ids
-                predicted_ids = self.model.generate(inputs)
+            predicted_ids = self.model.generate(inputs, language=self.language)
         # pylint: disable=E1101
         result = self.processor.tokenizer.batch_decode(predicted_ids, skip_special_tokens=True, normalize=True)[0]
-        if self.language == "auto" or self.language == "zh":
+        if self.language in ["chinese", "mandarin"]:
             from zhconv import convert
 
             result = convert(result, "zh-cn")
@@ -91,15 +105,20 @@ def audio2text(self, audio_path):
 
 
 if __name__ == "__main__":
-    asr = AudioSpeechRecognition(language="auto")
-    import urllib.request
+    asr = AudioSpeechRecognition(language="english")
+
+    # Test multilanguage asr
+    urllib.request.urlretrieve(
+        "https://paddlespeech.bj.bcebos.com/Parakeet/docs/demos/labixiaoxin.wav",
+        "sample.wav",
+    )
+    asr.language = "chinese"
+    text = asr.audio2text("sample.wav")
 
     urllib.request.urlretrieve(
         "https://github.com/intel/intel-extension-for-transformers/raw/main/intel_extension_for_transformers/neural_chat/assets/audio/sample.wav",
         "sample.wav",
     )
     text = asr.audio2text("sample.wav")
-    import os
 
     os.remove("sample.wav")
-    print(text)
diff --git a/AudioQnA/audio/docker/asr/asr_server.py b/AudioQnA/audio/docker/asr/asr_server.py
index a2c4ec665..4eadb1c8e 100644
--- a/AudioQnA/audio/docker/asr/asr_server.py
+++ b/AudioQnA/audio/docker/asr/asr_server.py
@@ -58,7 +58,7 @@ async def audio_to_text(file: UploadFile = File(...)):
     parser.add_argument("--port", type=int, default=8008)
     parser.add_argument("--model_name_or_path", type=str, default="openai/whisper-tiny")
     parser.add_argument("--bf16", default=False, action="store_true")
-    parser.add_argument("--language", type=str, default="auto")
+    parser.add_argument("--language", type=str, default="english")
     parser.add_argument("--device", type=str, default="cpu")
 
     args = parser.parse_args()