From 24f73665e2d8ea8e4de2fe4f900bc539d7f7b989 Mon Sep 17 00:00:00 2001
From: hnluo <haoneng.lhn@alibaba-inc.com>
Date: 星期一, 17 四月 2023 15:49:45 +0800
Subject: [PATCH] Merge pull request #367 from alibaba-damo-academy/dev_lhn2

---
 funasr/runtime/python/websocket/README.md |    5 +++--
 1 files changed, 3 insertions(+), 2 deletions(-)

diff --git a/funasr/runtime/python/websocket/README.md b/funasr/runtime/python/websocket/README.md
index ce44728..353cfa6 100644
--- a/funasr/runtime/python/websocket/README.md
+++ b/funasr/runtime/python/websocket/README.md
@@ -2,7 +2,6 @@
 We can send streaming audio data to server in real-time with grpc client every 300 ms e.g., and get transcribed text when stop speaking.
 The audio data is in streaming, the asr inference process is in offline.
 
-# Steps
 
 ## For the Server
 
@@ -31,13 +30,15 @@
 
 Install the requirements for client
 ```shell
+git clone https://github.com/alibaba/FunASR.git && cd FunASR
+cd funasr/runtime/python/websocket
 pip install -r requirements_client.txt
 ```
 
 Start client
 
 ```shell
-python ASR_client.py --host "localhost" --port 10095 --chunk_size 300
+python ASR_client.py --host "127.0.0.1" --port 10095 --chunk_size 300
 ```
 
 ## Acknowledge

--
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