diff --git a/speechx/examples/ds2_ol/README.md b/speechx/examples/ds2_ol/README.md
index 3fabd3e7..64d0afaa 100644
--- a/speechx/examples/ds2_ol/README.md
+++ b/speechx/examples/ds2_ol/README.md
@@ -4,8 +4,11 @@ Please go to `aishell` to test it.
 
 * aishell
 Deepspeech2 Streaming Decoding under aishell dataset.
+* websocket
+Streaming ASR with websocket.
 
 The below is for developing and offline testing:
 * nnet
 * feat
 * decoder
+
diff --git a/speechx/examples/ds2_ol/aishell/path.sh b/speechx/examples/ds2_ol/aishell/path.sh
old mode 100644
new mode 100755
diff --git a/speechx/examples/ds2_ol/aishell/websocket_client.sh b/speechx/examples/ds2_ol/aishell/websocket_client.sh
old mode 100644
new mode 100755
diff --git a/speechx/examples/ds2_ol/aishell/websocket_server.sh b/speechx/examples/ds2_ol/aishell/websocket_server.sh
old mode 100644
new mode 100755
diff --git a/speechx/examples/ds2_ol/decoder/recognizer_test_main.cc b/speechx/examples/ds2_ol/decoder/recognizer_test_main.cc
index 198a8ec2..e6fed0ed 100644
--- a/speechx/examples/ds2_ol/decoder/recognizer_test_main.cc
+++ b/speechx/examples/ds2_ol/decoder/recognizer_test_main.cc
@@ -60,6 +60,7 @@ int main(int argc, char* argv[]) {
             for (int i = 0; i < cur_chunk_size; ++i) {
                 wav_chunk(i) = waveform(sample_offset + i);
             }
+            // wav_chunk = waveform.Range(sample_offset + i, cur_chunk_size);
 
             recognizer.Accept(wav_chunk);
             if (cur_chunk_size < chunk_sample_size) {
@@ -67,8 +68,10 @@ int main(int argc, char* argv[]) {
             }
             recognizer.Decode();
 
+            // no overlap
             sample_offset += cur_chunk_size;
         }
+
         std::string result;
         result = recognizer.GetFinalResult();
         recognizer.Reset();
diff --git a/speechx/examples/ds2_ol/websocket/CMakeLists.txt b/speechx/examples/ds2_ol/websocket/CMakeLists.txt
index 754b528e..ed542aad 100644
--- a/speechx/examples/ds2_ol/websocket/CMakeLists.txt
+++ b/speechx/examples/ds2_ol/websocket/CMakeLists.txt
@@ -6,5 +6,4 @@ target_link_libraries(websocket_server_main PUBLIC frontend kaldi-feat-common nn
 
 add_executable(websocket_client_main ${CMAKE_CURRENT_SOURCE_DIR}/websocket_client_main.cc)
 target_include_directories(websocket_client_main PRIVATE ${SPEECHX_ROOT} ${SPEECHX_ROOT}/kaldi)
-target_link_libraries(websocket_client_main PUBLIC frontend kaldi-feat-common nnet decoder fst utils gflags glog kaldi-base kaldi-matrix kaldi-util kaldi-decoder websocket ${DEPS})
-
+target_link_libraries(websocket_client_main PUBLIC frontend kaldi-feat-common nnet decoder fst utils gflags glog kaldi-base kaldi-matrix kaldi-util kaldi-decoder websocket ${DEPS})
\ No newline at end of file
diff --git a/speechx/speechx/decoder/param.h b/speechx/speechx/decoder/param.h
index cd50ef53..a2cbbe50 100644
--- a/speechx/speechx/decoder/param.h
+++ b/speechx/speechx/decoder/param.h
@@ -37,13 +37,16 @@ DEFINE_int32(receptive_field_length,
 DEFINE_int32(downsampling_rate,
              4,
              "two CNN(kernel=5) module downsampling rate.");
+DEFINE_string(
+    model_input_names,
+    "audio_chunk,audio_chunk_lens,chunk_state_h_box,chunk_state_c_box",
+    "model input names");
 DEFINE_string(model_output_names,
-              "save_infer_model/scale_0.tmp_1,save_infer_model/"
-              "scale_1.tmp_1,save_infer_model/scale_2.tmp_1,save_infer_model/"
-              "scale_3.tmp_1",
+              "softmax_0.tmp_0,tmp_5,concat_0.tmp_0,concat_1.tmp_0",
               "model output names");
 DEFINE_string(model_cache_names, "5-1-1024,5-1-1024", "model cache names");
 
+
 namespace ppspeech {
 // todo refactor later
 FeaturePipelineOptions InitFeaturePipelineOptions() {