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

143 lines
4.8 KiB

# 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.
#!/usr/bin/python
# -*- coding: UTF-8 -*-
import argparse
import asyncio
import codecs
import json
import logging
import os
import numpy as np
import soundfile
import websockets
class ASRAudioHandler:
def __init__(self, url="127.0.0.1", port=8090):
self.url = url
self.port = port
self.url = "ws://" + self.url + ":" + str(self.port) + "/ws/asr"
def read_wave(self, wavfile_path: str):
samples, sample_rate = soundfile.read(wavfile_path, dtype='int16')
x_len = len(samples)
# chunk_stride = 40 * 16 #40ms, sample_rate = 16kHz
chunk_size = 80 * 16 #80ms, sample_rate = 16kHz
if x_len % chunk_size != 0:
padding_len_x = chunk_size - x_len % chunk_size
else:
padding_len_x = 0
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
num_chunk = (x_len + padding_len_x) / chunk_size
num_chunk = int(num_chunk)
for i in range(0, num_chunk):
start = i * chunk_size
end = start + chunk_size
x_chunk = padded_x[start:end]
yield x_chunk
async def run(self, wavfile_path: str):
logging.info("send a message to the server")
# self.read_wave()
# send websocket handshake protocal
async with websockets.connect(self.url) as ws:
# server has already received handshake protocal
# client start to send the command
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 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)
logging.info("receive msg={}".format(msg))
result = msg
# 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()
# decode the bytes to str
msg = json.loads(msg)
logging.info("receive msg={}".format(msg))
return result
def main(args):
logging.basicConfig(level=logging.INFO)
logging.info("asr websocket client start")
handler = ASRAudioHandler("127.0.0.1", 8090)
loop = asyncio.get_event_loop()
# support to process single audio file
if args.wavfile and os.path.exists(args.wavfile):
logging.info(f"start to process the wavscp: {args.wavfile}")
result = loop.run_until_complete(handler.run(args.wavfile))
result = result["asr_results"]
logging.info(f"asr websocket client finished : {result}")
# 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,\
codecs.open("result.txt", 'w', encoding='utf-8') as w:
for line in f:
utt_name, utt_path = line.strip().split()
result = loop.run_until_complete(handler.run(utt_path))
result = result["asr_results"]
w.write(f"{utt_name} {result}\n")
if __name__ == "__main__":
parser = argparse.ArgumentParser()
parser.add_argument(
"--wavfile",
action="store",
help="wav file path ",
default="./16_audio.wav")
parser.add_argument(
"--wavscp", type=str, default=None, help="The batch audios dict text")
args = parser.parse_args()
main(args)