update the audio handler note, test=doc

paddlespeech/server/utils/audio_handler.py

-import numpy as np
-import logging
-import argparse
-import asyncio
-import codecs
+# 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.
 import json
 import logging
-import os
 
 import numpy as np
 import soundfile
 import websockets
+
 from paddlespeech.cli.log import logger
+
+
 class ASRAudioHandler:
     def __init__(self, url="127.0.0.1", port=8090):
+        """PaddleSpeech Online ASR Server Client  audio handler
+           Online asr server use the websocket protocal
+        Args:
+            url (str, optional): the server ip. Defaults to "127.0.0.1".
+            port (int, optional): the server port. Defaults to 8090.
+        """
         self.url = url
         self.port = port
         self.url = "ws://" + self.url + ":" + str(self.port) + "/ws/asr"
 
     def read_wave(self, wavfile_path: str):
+        """read the audio file from specific wavfile path
+
+        Args:
+            wavfile_path (str): the audio wavfile, 
+                                 we assume that audio sample rate matches the model
+
+        Yields:
+            numpy.array: the samall package audio pcm data
+        """
         samples, sample_rate = soundfile.read(wavfile_path, dtype='int16')
         x_len = len(samples)
 
         chunk_size = 85 * 16  #80ms, sample_rate = 16kHz
-        if x_len % chunk_size!= 0:
+        if x_len % chunk_size != 0:
             padding_len_x = chunk_size - x_len % chunk_size
         else:
             padding_len_x = 0
@@ -40,11 +65,19 @@ class ASRAudioHandler:
             yield x_chunk
 
     async def run(self, wavfile_path: str):
+        """Send a audio file to online server
+
+        Args:
+            wavfile_path (str): audio path
+
+        Returns:
+            str: the final asr result
+        """
         logging.info("send a message to the server")
-        # self.read_wave()
-        # send websocket handshake protocal
+
+        # 1. send websocket handshake protocal
         async with websockets.connect(self.url) as ws:
-            # server has already received handshake protocal
+            # 2. server has already received handshake protocal
             # client start to send the command
             audio_info = json.dumps(
                 {
@@ -59,14 +92,14 @@ class ASRAudioHandler:
             msg = await ws.recv()
             logger.info("receive msg={}".format(msg))
 
-            # send chunk audio data to engine
+            # 3. send chunk audio data to engine
             for chunk_data in self.read_wave(wavfile_path):
                 await ws.send(chunk_data.tobytes())
                 msg = await ws.recv()
                 msg = json.loads(msg)
                 logger.info("receive msg={}".format(msg))
 
-            # finished 
+            # 4. we must send finished signal to the server
             audio_info = json.dumps(
                 {
                     "name": "test.wav",
@@ -78,9 +111,9 @@ class ASRAudioHandler:
                 separators=(',', ': '))
             await ws.send(audio_info)
             msg = await ws.recv()
-            
-            # decode the bytes to str
+
+            # 5. decode the bytes to str
             msg = json.loads(msg)
             logger.info("final receive msg={}".format(msg))
             result = msg
-            return result
+            return result
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