From 3cd3473bf7a3b41484baa86d9092248d78e7af39 Mon Sep 17 00:00:00 2001
From: 游雁 <zhifu.gzf@alibaba-inc.com>
Date: 星期五, 21 四月 2023 17:17:37 +0800
Subject: [PATCH] docs

---
 funasr/runtime/python/websocket/README.md |   10 ++++++----
 1 files changed, 6 insertions(+), 4 deletions(-)

diff --git a/funasr/runtime/python/websocket/README.md b/funasr/runtime/python/websocket/README.md
index ce44728..73f8aeb 100644
--- a/funasr/runtime/python/websocket/README.md
+++ b/funasr/runtime/python/websocket/README.md
@@ -2,16 +2,16 @@
 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
 
 Install the modelscope and funasr
 
 ```shell
-pip install "modelscope[audio_asr]" -f https://modelscope.oss-cn-beijing.aliyuncs.com/releases/repo.html
+pip install -U modelscope funasr
+# For the users in China, you could install with the command:
+# pip install -U modelscope funasr -i https://mirror.sjtu.edu.cn/pypi/web/simple
 git clone https://github.com/alibaba/FunASR.git && cd FunASR
-pip install --editable ./
 ```
 
 Install the requirements for server
@@ -31,13 +31,15 @@
 
 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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