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PaddleSpeech/demos/streaming_asr_server/local/websocket_client_srt.py

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6.0 KiB

#!/usr/bin/python
# 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.
# calc avg RTF(NOT Accurate): grep -rn RTF log.txt | awk '{print $NF}' | awk -F "=" '{sum += $NF} END {print "all time",sum, "audio num", NR, "RTF", sum/NR}'
# python3 websocket_client.py --server_ip 127.0.0.1 --port 8290 --punc.server_ip 127.0.0.1 --punc.port 8190 --wavfile ./zh.wav
# python3 websocket_client.py --server_ip 127.0.0.1 --port 8290 --wavfile ./zh.wav
import argparse
import asyncio
import codecs
import os
from pydub import AudioSegment
import re
from paddlespeech.cli.log import logger
from paddlespeech.server.utils.audio_handler import ASRWsAudioHandler
def convert_to_wav(input_file):
# Load audio file
audio = AudioSegment.from_file(input_file)
# Set parameters for audio file
audio = audio.set_channels(1)
audio = audio.set_frame_rate(16000)
# Create output filename
output_file = os.path.splitext(input_file)[0] + ".wav"
# Export audio file as WAV
audio.export(output_file, format="wav")
logger.info(f"{input_file} converted to {output_file}")
def format_time(sec):
# Convert seconds to SRT format (HH:MM:SS,ms)
hours = int(sec/3600)
minutes = int((sec%3600)/60)
seconds = int(sec%60)
milliseconds = int((sec%1)*1000)
return f'{hours:02d}:{minutes:02d}:{seconds:02d},{milliseconds:03d}'
def results2srt(results, srt_file):
"""convert results from paddlespeech to srt format for subtitle
Args:
results (dict): results from paddlespeech
"""
# times contains start and end time of each word
times = results['times']
# result contains the whole sentence including punctuation
result = results['result']
# split result into several sencences by '' and '。'
sentences = re.split('|。', result)[:-1]
# print("sentences: ", sentences)
# generate relative time for each sentence in sentences
relative_times = []
word_i = 0
for sentence in sentences:
relative_times.append([])
for word in sentence:
if relative_times[-1] == []:
relative_times[-1].append(times[word_i]['bg'])
if len(relative_times[-1]) == 1:
relative_times[-1].append(times[word_i]['ed'])
else:
relative_times[-1][1] = times[word_i]['ed']
word_i += 1
# print("relative_times: ", relative_times)
# generate srt file acoording to relative_times and sentences
with open(srt_file, 'w') as f:
for i in range(len(sentences)):
# Write index number
f.write(str(i+1)+'\n')
# Write start and end times
start = format_time(relative_times[i][0])
end = format_time(relative_times[i][1])
f.write(start + ' --> ' + end + '\n')
# Write text
f.write(sentences[i]+'\n\n')
logger.info(f"results saved to {srt_file}")
def main(args):
logger.info("asr websocket client start")
handler = ASRWsAudioHandler(
args.server_ip,
args.port,
endpoint=args.endpoint,
punc_server_ip=args.punc_server_ip,
punc_server_port=args.punc_server_port)
loop = asyncio.get_event_loop()
# check if the wav file is mp3 format
# if so, convert it to wav format using convert_to_wav function
if args.wavfile and os.path.exists(args.wavfile):
if args.wavfile.endswith(".mp3"):
convert_to_wav(args.wavfile)
args.wavfile = args.wavfile.replace(".mp3", ".wav")
# support to process single audio file
if args.wavfile and os.path.exists(args.wavfile):
logger.info(f"start to process the wavscp: {args.wavfile}")
result = loop.run_until_complete(handler.run(args.wavfile))
# result = result["result"]
# logger.info(f"asr websocket client finished : {result}")
results2srt(result, args.wavfile.replace(".wav", ".srt"))
# support to process batch audios from wav.scp
if args.wavscp and os.path.exists(args.wavscp):
logger.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["result"]
w.write(f"{utt_name} {result}\n")
if __name__ == "__main__":
logger.info("Start to do streaming asr client")
parser = argparse.ArgumentParser()
parser.add_argument(
'--server_ip', type=str, default='127.0.0.1', help='server ip')
parser.add_argument('--port', type=int, default=8090, help='server port')
parser.add_argument(
'--punc.server_ip',
type=str,
default=None,
dest="punc_server_ip",
help='Punctuation server ip')
parser.add_argument(
'--punc.port',
type=int,
default=8091,
dest="punc_server_port",
help='Punctuation server port')
parser.add_argument(
"--endpoint",
type=str,
default="/paddlespeech/asr/streaming",
help="ASR websocket endpoint")
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)