From 15b8b7bd20d3bb76e933fd2426052c19a0c6ea56 Mon Sep 17 00:00:00 2001
From: yhliang <68215459+yhliang-aslp@users.noreply.github.com>
Date: 星期二, 18 四月 2023 19:19:48 +0800
Subject: [PATCH] Merge branch 'dev_lyh' into main

---
 funasr/runtime/python/websocket/README.md |    1 -
 1 files changed, 0 insertions(+), 1 deletions(-)

diff --git a/funasr/runtime/python/websocket/README.md b/funasr/runtime/python/websocket/README.md
index 2c0dec1..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
 

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