From 8a08405b668e06c4670b4c13f6793e193f21a21d Mon Sep 17 00:00:00 2001
From: Yabin Li <wucong.lyb@alibaba-inc.com>
Date: 星期一, 08 五月 2023 11:43:08 +0800
Subject: [PATCH] Merge branch 'main' into dev_apis

---
 funasr/runtime/python/websocket/README.md |   58 ++++++++++++++++++++++++++++++++++++++++++++++------------
 1 files changed, 46 insertions(+), 12 deletions(-)

diff --git a/funasr/runtime/python/websocket/README.md b/funasr/runtime/python/websocket/README.md
index 73f8aeb..473c37a 100644
--- a/funasr/runtime/python/websocket/README.md
+++ b/funasr/runtime/python/websocket/README.md
@@ -1,11 +1,10 @@
-# Using funasr with websocket
-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.
+# Service with websocket-python
 
+This is a demo using funasr pipeline with websocket python-api. 
 
 ## For the Server
 
-Install the modelscope and funasr
+### Install the modelscope and funasr
 
 ```shell
 pip install -U modelscope funasr
@@ -14,18 +13,31 @@
 git clone https://github.com/alibaba/FunASR.git && cd FunASR
 ```
 
-Install the requirements for server
+### Install the requirements for server
 
 ```shell
 cd funasr/runtime/python/websocket
 pip install -r requirements_server.txt
 ```
 
-Start server
-
+### Start server
+#### ASR offline server
 ```shell
-python ASR_server.py --host "0.0.0.0" --port 10095 --asr_model "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
+python ws_server_offline.py --port 10095 --asr_model "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
 ```
+
+#### ASR streaming server
+```shell
+python ws_server_online.py --port 10095 --asr_model_online "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online"
+```
+
+#### ASR offline/online 2pass server
+
+[//]: # (```shell)
+
+[//]: # (python ws_server_online.py --host "0.0.0.0" --port 10095 --asr_model "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch")
+
+[//]: # (```)
 
 ## For the client
 
@@ -36,11 +48,33 @@
 pip install -r requirements_client.txt
 ```
 
-Start client
-
+### Start client
+#### ASR offline client
+##### Recording from mircrophone
 ```shell
-python ASR_client.py --host "127.0.0.1" --port 10095 --chunk_size 300
+# --chunk_interval, "10": 600/10=60ms, "5"=600/5=120ms, "20": 600/12=30ms
+python ws_client.py --host "0.0.0.0" --port 10095 --chunk_interval 10 --words_max_print 100
+```
+##### Loadding from wav.scp(kaldi style)
+```shell
+# --chunk_interval, "10": 600/10=60ms, "5"=600/5=120ms, "20": 600/12=30ms
+python ws_client.py --host "0.0.0.0" --port 10095 --chunk_interval 10 --words_max_print 100 --audio_in "./data/wav.scp" --send_without_sleep --output_dir "./results"
+```
+#### ASR streaming client
+##### Recording from mircrophone
+```shell
+# --chunk_size, "5,10,5"=600ms, "8,8,4"=480ms
+python ws_client.py --host "0.0.0.0" --port 10095 --chunk_size "5,10,5" --words_max_print 100
+```
+##### Loadding from wav.scp(kaldi style)
+```shell
+# --chunk_size, "5,10,5"=600ms, "8,8,4"=480ms
+python ws_client.py --host "0.0.0.0" --port 10095 --chunk_size "5,10,5" --audio_in "./data/wav.scp" --words_max_print 100 --output_dir "./results"
 ```
 
+#### ASR offline/online 2pass client
+
 ## Acknowledge
-1. We acknowledge [cgisky1980](https://github.com/cgisky1980/FunASR) for contributing the websocket service.
+1. This project is maintained by [FunASR community](https://github.com/alibaba-damo-academy/FunASR).
+2. We acknowledge [zhaoming](https://github.com/zhaomingwork/FunASR/tree/fix_bug_for_python_websocket) for contributing the websocket service.
+3. We acknowledge [cgisky1980](https://github.com/cgisky1980/FunASR) for contributing the websocket service of offline model.

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