From d1ac7608d409ba86f34c93cf6bf8bd85b9b106b6 Mon Sep 17 00:00:00 2001 From: huangyuxin Date: Thu, 19 May 2022 09:12:16 +0000 Subject: [PATCH] add script for calc RTF --- demos/streaming_asr_server/websocket_client.py | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/demos/streaming_asr_server/websocket_client.py b/demos/streaming_asr_server/websocket_client.py index 3451b8d0..8a4fe330 100644 --- a/demos/streaming_asr_server/websocket_client.py +++ b/demos/streaming_asr_server/websocket_client.py @@ -13,6 +13,9 @@ # limitations under the License. #!/usr/bin/python # -*- coding: UTF-8 -*- + +# script for calc RTF: grep -rn RTF log.txt | awk '{print $NF}' | awk -F "=" '{sum += $NF} END {print "all time",sum, "audio num", NR, "RTF", sum/NR}' + import argparse import asyncio import codecs @@ -40,7 +43,7 @@ def main(args): result = result["result"] logger.info(f"asr websocket client finished : {result}") - # support to process batch audios from wav.scp + # support to process batch audios from wav.scp if args.wavscp and os.path.exists(args.wavscp): logging.info(f"start to process the wavscp: {args.wavscp}") with codecs.open(args.wavscp, 'r', encoding='utf-8') as f,\ -- GitLab