From 62aab4a6da1b17dd36fd9332f3cd536d93c58b3a Mon Sep 17 00:00:00 2001
From: 九耳 <mengzhe.cmz@alibaba-inc.com>
Date: 星期四, 30 三月 2023 14:15:59 +0800
Subject: [PATCH] Merge branch 'dev_cmz2' of github.com:alibaba-damo-academy/FunASR into dev_cmz2

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

diff --git a/funasr/runtime/python/websocket/README.md b/funasr/runtime/python/websocket/README.md
index e8cdae5..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
 
@@ -31,6 +30,8 @@
 
 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
 ```
 

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