add asr websocket server note, test=doc

demos/speech_recognition/run.sh

@@ -0,0 +1,10 @@
+#!/bin/bash
+
+wget -c https://paddlespeech.bj.bcebos.com/PaddleAudio/zh.wav https://paddlespeech.bj.bcebos.com/PaddleAudio/en.wav
+
+# asr
+paddlespeech asr --input ./zh.wav
+
+
+# asr + punc
+paddlespeech asr --input ./zh.wav | paddlespeech text --task punc

paddlespeech/server/tests/asr/online/web/app.py

@@ -1,12 +1,11 @@
 #!/usr/bin/env python3
 # -*- coding: utf-8 -*-
-
 # Copyright 2021 Mobvoi Inc. All Rights Reserved.
 # Author: zhendong.peng@mobvoi.com (Zhendong Peng)
-
 import argparse
 
-from flask import Flask, render_template
+from flask import Flask
+from flask import render_template
 
 parser = argparse.ArgumentParser(description='training your network')
 parser.add_argument('--port', default=19999, type=int, help='port id')
@@ -14,9 +13,11 @@ args = parser.parse_args()
 
 app = Flask(__name__)
 
+
 @app.route('/')
 def index():
     return render_template('index.html')
 
+
 if __name__ == '__main__':
     app.run(host='0.0.0.0', port=args.port, debug=True)

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

@@ -15,10 +15,11 @@
 # -*- coding: UTF-8 -*-
 import argparse
 import asyncio
+import codecs
 import json
 import logging
 import os
-import codecs
+
 import numpy as np
 import soundfile
 import websockets
@@ -44,8 +45,8 @@ class ASRAudioHandler:
         padding = np.zeros((padding_len_x), dtype=samples.dtype)
         padded_x = np.concatenate([samples, padding], axis=0)
 
-        assert ( x_len + padding_len_x ) % chunk_size == 0
+        assert (x_len + padding_len_x) % chunk_size == 0
-        num_chunk = (x_len + padding_len_x ) / chunk_size
+        num_chunk = (x_len + padding_len_x) / chunk_size
         num_chunk = int(num_chunk)
 
         for i in range(0, num_chunk):
@@ -56,12 +57,7 @@ class ASRAudioHandler:
 
     async def run(self, wavfile_path: str):
         logging.info("send a message to the server")
-        # 读取音频
-        # self.read_wave()
-        # 发送 websocket 的 handshake 协议头
         async with websockets.connect(self.url) as ws:
-            # server 端已经接收到 handshake 协议头
-            # 发送开始指令
             audio_info = json.dumps(
                 {
                     "name": "test.wav",
@@ -98,7 +94,6 @@ class ASRAudioHandler:
             msg = json.loads(msg)
             logging.info("receive msg={}".format(msg))
 
-
         return result
 
 

paddlespeech/server/utils/buffer.py

@@ -24,12 +24,22 @@ class Frame(object):
 
 class ChunkBuffer(object):
     def __init__(self,
-                 window_n=7,    # frame
+                 window_n=7,
-                 shift_n=4,     # frame
+                 shift_n=4,
-                 window_ms=20,  # ms
+                 window_ms=20,
-                 shift_ms=10,   # ms
+                 shift_ms=10,
                  sample_rate=16000,
                  sample_width=2):
+        """audio sample data point buffer
+
+        Args:
+            window_n (int, optional): decode window frame length. Defaults to 7 frame.
+            shift_n (int, optional): decode shift frame length. Defaults to 4 frame.
+            window_ms (int, optional): frame length, ms. Defaults to 20 ms.
+            shift_ms (int, optional): shift length, ms. Defaults to 10 ms.
+            sample_rate (int, optional): audio sample rate. Defaults to 16000.
+            sample_width (int, optional): sample point bytes. Defaults to 2 bytes.
+        """
         self.window_n = window_n
         self.shift_n = shift_n
         self.window_ms = window_ms
@@ -38,11 +48,14 @@ class ChunkBuffer(object):
         self.sample_width = sample_width  # int16 = 2; float32 = 4
         self.remained_audio = b''
 
-        self.window_sec = float((self.window_n - 1) * self.shift_ms + self.window_ms) / 1000.0 
+        self.window_sec = float((self.window_n - 1) * self.shift_ms +
+                                self.window_ms) / 1000.0
         self.shift_sec = float(self.shift_n * self.shift_ms / 1000.0)
 
-        self.window_bytes = int(self.window_sec * self.sample_rate * self.sample_width)
+        self.window_bytes = int(self.window_sec * self.sample_rate *
-        self.shift_bytes = int(self.shift_sec * self.sample_rate * self.sample_width)
+                                self.sample_width)
+        self.shift_bytes = int(self.shift_sec * self.sample_rate *
+                               self.sample_width)
 
     def frame_generator(self, audio):
         """Generates audio frames from PCM audio data.
@@ -57,7 +70,8 @@ class ChunkBuffer(object):
         timestamp = 0.0
 
         while offset + self.window_bytes <= len(audio):
-            yield Frame(audio[offset:offset + self.window_bytes], timestamp, self.window_sec)
+            yield Frame(audio[offset:offset + self.window_bytes], timestamp,
+                        self.window_sec)
             timestamp += self.shift_sec
             offset += self.shift_bytes
 

paddlespeech/server/ws/asr_socket.py

@@ -79,11 +79,6 @@ async def websocket_endpoint(websocket: WebSocket):
             elif "bytes" in message:
                 message = message["bytes"]
 
-                # vad for input bytes audio
-                # vad.add_audio(message)
-                # message = b''.join(f for f in vad.vad_collector()
-                #                    if f is not None)
-
                 engine_pool = get_engine_pool()
                 asr_engine = engine_pool['asr']
                 asr_results = ""
@@ -95,6 +90,7 @@ async def websocket_endpoint(websocket: WebSocket):
                                                                   sample_rate)
                     asr_engine.run(x_chunk, x_chunk_lens)
                     asr_results = asr_engine.postprocess()
+
                 asr_results = asr_engine.postprocess()
                 resp = {'asr_results': asr_results}