update en readme.md, test=doc

paddlespeech/server/bin/paddlespeech_client.py

@@ -35,7 +35,7 @@ from paddlespeech.server.utils.audio_process import wav2pcm
 from paddlespeech.server.utils.util import wav2base64
 
 __all__ = [
-    'TTSClientExecutor', 'ASRClientExecutor', 'ASRClientExecutor',
+    'TTSClientExecutor', 'ASRClientExecutor', 'ASROnlineClientExecutor',
     'CLSClientExecutor'
 ]
 

paddlespeech/server/tests/asr/online/README.md

+([简体中文](./README_cn.md)|English)
+
+# Speech Service
+
+## Introduction
+
+This document introduces a client for streaming asr service: microphone
+
+
+## Usage
+### 1. Install
+Refer [Install](https://github.com/PaddlePaddle/PaddleSpeech/blob/develop/docs/source/install.md).
+
+ **paddlepaddle 2.2.1** 或以上版本。
+It is recommended to use **paddlepaddle 2.2.1** or above.
+You can choose one way from meduim and hard to install paddlespeech.
+
+
+### 2. Prepare config File
+
+
+The input of  ASR client demo should be a WAV file(`.wav`), and the sample rate must be the same as the model.
+
+Here are sample files for thisASR client demo that can be downloaded:
+```bash
+wget -c https://paddlespeech.bj.bcebos.com/PaddleAudio/zh.wav
+```
+
+### 2. Streaming ASR Client Usage
+
+- microphone
+   ```
+   python microphone_client.py
+
+   ```

paddlespeech/server/tests/asr/online/README_cn.md

+([English](./README.md)|中文)
+
+# 语音服务
+
+## 介绍
+本文档介绍如何使用流式ASR的一种不同客户端:麦克风。 
+
+
+## 使用方法
+### 1. 安装
+请看 [安装文档](https://github.com/PaddlePaddle/PaddleSpeech/blob/develop/docs/source/install.md).
+
+推荐使用 **paddlepaddle 2.2.1** 或以上版本。
+你可以从 medium，hard 三中方式中选择一种方式安装 PaddleSpeech。
+
+
+### 2. 准备测试文件
+
+这个 ASR client 的输入应该是一个 WAV 文件（`.wav`），并且采样率必须与模型的采样率相同。
+
+可以下载此 ASR client的示例音频：
+```bash
+wget -c https://paddlespeech.bj.bcebos.com/PaddleAudio/zh.wav
+```
+
+### 2. 流式 ASR 客户端使用方法
+
+- Python模拟流式服务命令行
+   ```
+
+   # 流式ASR
+   paddlespeech_client asr_online --server_ip 127.0.0.1 --port 8091 --input ./zh.wav
+
+   ```
+
+
+- 麦克风
+   ```
+   # 直接调用麦克风设备
+   python microphone_client.py
+
+   ```

paddlespeech/server/tests/asr/online/microphone_client.py

+# Copyright (c) 2022 PaddlePaddle Authors. All Rights Reserved.
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#     http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+"""
+record wave from the mic
+"""
+import asyncio
+import json
+import logging
+import threading
+import wave
+from signal import SIGINT
+from signal import SIGTERM
+
+import pyaudio
+import websockets
+
+
+class ASRAudioHandler(threading.Thread):
+    def __init__(self, url="127.0.0.1", port=8091):
+        threading.Thread.__init__(self)
+        self.url = url
+        self.port = port
+        self.url = "ws://" + self.url + ":" + str(self.port) + "/ws/asr"
+        self.fileName = "./output.wav"
+        self.chunk = 5120
+        self.format = pyaudio.paInt16
+        self.channels = 1
+        self.rate = 16000
+        self._running = True
+        self._frames = []
+        self.data_backup = []
+
+    def startrecord(self):
+        """
+        start a new thread to record wave
+        """
+        threading._start_new_thread(self.recording, ())
+
+    def recording(self):
+        """
+        recording wave
+        """
+        self._running = True
+        self._frames = []
+        p = pyaudio.PyAudio()
+        stream = p.open(
+            format=self.format,
+            channels=self.channels,
+            rate=self.rate,
+            input=True,
+            frames_per_buffer=self.chunk)
+        while (self._running):
+            data = stream.read(self.chunk)
+            self._frames.append(data)
+            self.data_backup.append(data)
+
+        stream.stop_stream()
+        stream.close()
+        p.terminate()
+
+    def save(self):
+        """
+        save wave data
+        """
+        p = pyaudio.PyAudio()
+        wf = wave.open(self.fileName, 'wb')
+        wf.setnchannels(self.channels)
+        wf.setsampwidth(p.get_sample_size(self.format))
+        wf.setframerate(self.rate)
+        wf.writeframes(b''.join(self.data_backup))
+        wf.close()
+        p.terminate()
+
+    def stoprecord(self):
+        """
+        stop recording
+        """
+        self._running = False
+
+    async def run(self):
+        aa = input("是否开始录音？   (y/n)")
+        if aa.strip() == "y":
+            self.startrecord()
+            logging.info("*" * 10 + "开始录音，请输入语音")
+
+            async with websockets.connect(self.url) as ws:
+                # 发送开始指令
+                audio_info = json.dumps(
+                    {
+                        "name": "test.wav",
+                        "signal": "start",
+                        "nbest": 5
+                    },
+                    sort_keys=True,
+                    indent=4,
+                    separators=(',', ': '))
+                await ws.send(audio_info)
+                msg = await ws.recv()
+                logging.info("receive msg={}".format(msg))
+
+                # send bytes data
+                logging.info("结束录音请: Ctrl + c。继续请按回车。")
+                try:
+                    while True:
+                        while len(self._frames) > 0:
+                            await ws.send(self._frames.pop(0))
+                            msg = await ws.recv()
+                            logging.info("receive msg={}".format(msg))
+                except asyncio.CancelledError:
+                    # quit
+                    # send finished 
+                    audio_info = json.dumps(
+                        {
+                            "name": "test.wav",
+                            "signal": "end",
+                            "nbest": 5
+                        },
+                        sort_keys=True,
+                        indent=4,
+                        separators=(',', ': '))
+                    await ws.send(audio_info)
+                    msg = await ws.recv()
+                    logging.info("receive msg={}".format(msg))
+
+                    self.stoprecord()
+                    logging.info("*" * 10 + "录音结束")
+                    self.save()
+        elif aa.strip() == "n":
+            exit()
+        else:
+            print("无效输入!")
+            exit()
+
+
+if __name__ == "__main__":
+
+    logging.basicConfig(level=logging.INFO)
+    logging.info("asr websocket client start")
+
+    handler = ASRAudioHandler("127.0.0.1", 8091)
+    loop = asyncio.get_event_loop()
+    main_task = asyncio.ensure_future(handler.run())
+    for signal in [SIGINT, SIGTERM]:
+        loop.add_signal_handler(signal, main_task.cancel)
+    try:
+        loop.run_until_complete(main_task)
+    finally:
+        loop.close()
+
+    logging.info("asr websocket client finished")