diff --git a/docs/python_package.md b/docs/python_package.md
index 9c72ba0f..98cccaa2 100644
--- a/docs/python_package.md
+++ b/docs/python_package.md
@@ -21,6 +21,8 @@ $ wespeaker --task embedding --audio_file audio.wav --output_file embedding.txt
 $ wespeaker --task embedding_kaldi --wav_scp wav.scp --output_file /path/to/embedding
 $ wespeaker --task similarity --audio_file audio.wav --audio_file2 audio2.wav
 $ wespeaker --task diarization --audio_file audio.wav
+$ wespeaker --task diarization --audio_file audio.wav --device cuda:0 # use CUDA on Windows/Linux
+$ wespeaker --task diarization --audio_file audio.wav --device mps    # use Metal Performance Shaders on MacOS
 ```
 
 You can specify the following parameters. (use `-h` for details)
@@ -33,7 +35,7 @@ You can specify the following parameters. (use `-h` for details)
     - diarization_list: apply speaker diarization for a kaldi-style wav.scp
 * `-l` or `--language`: use Chinese/English speaker models
 * `-p` or `--pretrain`: the path of pretrained model, `avg_model.pt` and `config.yaml` should be contained
-* `-g` or `--gpu`: use GPU for inference, number $< 0$ means using CPU
+* `--device`: set pytorch device, `cpu`, `cuda`, `cuda:0` or `mps`
 * `--campplus`:
   use [`campplus_cn_common_200k` of damo](https://www.modelscope.cn/models/iic/speech_campplus_sv_zh-cn_16k-common/summary)
 * `--eres2net`:
@@ -69,14 +71,14 @@ which can either be the ones we provided and trained by yourself.
 import wespeaker
 
 model = wespeaker.load_model('chinese')
-# set_gpu to enable the cuda inference, number < 0 means using CPU
-model.set_gpu(0)
+# set the device on which tensors are or will be allocated.
+model.set_device('cuda:0')
 
 # embedding/embedding_kaldi/similarity/diarization
 embedding = model.extract_embedding('audio.wav')
 utt_names, embeddings = model.extract_embedding_list('wav.scp')
 similarity = model.compute_similarity('audio1.wav', 'audio2.wav')
-diar_result = model.diarize('audio.wav')
+diar_result = model.diarize('audio.wav', 'give_this_utt_a_name')
 
 # register and recognize
 model.register('spk1', 'spk1_audio1.wav')
diff --git a/wespeaker/cli/speaker.py b/wespeaker/cli/speaker.py
index be064ac1..67030556 100644
--- a/wespeaker/cli/speaker.py
+++ b/wespeaker/cli/speaker.py
@@ -72,11 +72,7 @@ def set_resample_rate(self, resample_rate: int):
     def set_vad(self, apply_vad: bool):
         self.apply_vad = apply_vad
 
-    def set_gpu(self, device_id: int):
-        if device_id >= 0:
-            device = 'cuda:{}'.format(device_id)
-        else:
-            device = 'cpu'
+    def set_device(self, device: str):
         self.device = torch.device(device)
         self.model = self.model.to(self.device)
 
@@ -304,7 +300,7 @@ def main():
         model = load_model_local(args.pretrain)
     model.set_resample_rate(args.resample_rate)
     model.set_vad(args.vad)
-    model.set_gpu(args.gpu)
+    model.set_device(args.device)
     model.set_diarization_params(min_duration=args.diar_min_duration,
                                  window_secs=args.diar_window_secs,
                                  period_secs=args.diar_period_secs,
diff --git a/wespeaker/cli/utils.py b/wespeaker/cli/utils.py
index 3d5125f2..34eb26b0 100644
--- a/wespeaker/cli/utils.py
+++ b/wespeaker/cli/utils.py
@@ -52,11 +52,12 @@ def get_args():
                         type=str,
                         default="",
                         help='model directory')
-    parser.add_argument('-g',
-                        '--gpu',
-                        type=int,
-                        default=-1,
-                        help='which gpu to use (number <0 means using cpu)')
+    parser.add_argument('--device',
+                        type=str,
+                        default='cpu',
+                        help="device type (most commonly cpu or cuda,"
+                             "but also potentially mps, xpu, xla or meta)"
+                             "and optional device ordinal for the device type.")
     parser.add_argument('--audio_file', help='audio file')
     parser.add_argument('--audio_file2',
                         help='audio file2, specifically for similarity task')
diff --git a/wespeaker/diar/extract_emb.py b/wespeaker/diar/extract_emb.py
index da16fc85..6e0f4cfa 100644
--- a/wespeaker/diar/extract_emb.py
+++ b/wespeaker/diar/extract_emb.py
@@ -37,7 +37,7 @@ def init_session(source, device):
     opts = ort.SessionOptions()
     opts.inter_op_num_threads = 1
     opts.intra_op_num_threads = 1
-    opts.log_severity_level = 0
+    opts.log_severity_level = 1
     session = ort.InferenceSession(source,
                                    sess_options=opts,
                                    providers=providers)
diff --git a/wespeaker/frontend/s3prl.py b/wespeaker/frontend/s3prl.py
index d96168d5..37bf8c8e 100644
--- a/wespeaker/frontend/s3prl.py
+++ b/wespeaker/frontend/s3prl.py
@@ -57,7 +57,7 @@ def __init__(self,
 
         if layer != -1:
             layer_selections = [layer]
-            assert not multilayer_feature,\
+            assert not multilayer_feature, \
                 "multilayer_feature must be False if layer is specified"
         else:
             layer_selections = None