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#!/usr/bin/python
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# Copyright (c) 2022 PaddlePaddle Authors. All Rights Reserved.
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#
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# Licensed under the Apache License, Version 2.0 (the "License");
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# you may not use this file except in compliance with the License.
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# You may obtain a copy of the License at
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#
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# http://www.apache.org/licenses/LICENSE-2.0
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#
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# Unless required by applicable law or agreed to in writing, software
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# distributed under the License is distributed on an "AS IS" BASIS,
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# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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# See the License for the specific language governing permissions and
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# limitations under the License.
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# 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}'
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# 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
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# python3 websocket_client.py --server_ip 127.0.0.1 --port 8290 --wavfile ./zh.wav
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import argparse
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import asyncio
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import codecs
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import os
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from pydub import AudioSegment
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import re
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from paddlespeech.cli.log import logger
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from paddlespeech.server.utils.audio_handler import ASRWsAudioHandler
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def convert_to_wav(input_file):
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# Load audio file
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audio = AudioSegment.from_file(input_file)
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# Set parameters for audio file
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audio = audio.set_channels(1)
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audio = audio.set_frame_rate(16000)
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# Create output filename
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output_file = os.path.splitext(input_file)[0] + ".wav"
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# Export audio file as WAV
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audio.export(output_file, format="wav")
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logger.info(f"{input_file} converted to {output_file}")
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def format_time(sec):
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# Convert seconds to SRT format (HH:MM:SS,ms)
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hours = int(sec/3600)
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minutes = int((sec%3600)/60)
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seconds = int(sec%60)
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milliseconds = int((sec%1)*1000)
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return f'{hours:02d}:{minutes:02d}:{seconds:02d},{milliseconds:03d}'
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def results2srt(results, srt_file):
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"""convert results from paddlespeech to srt format for subtitle
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Args:
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results (dict): results from paddlespeech
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"""
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# times contains start and end time of each word
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times = results['times']
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# result contains the whole sentence including punctuation
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result = results['result']
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# split result into several sencences by '，' and '。'
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sentences = re.split('，|。', result)[:-1]
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# print("sentences: ", sentences)
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# generate relative time for each sentence in sentences
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relative_times = []
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word_i = 0
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for sentence in sentences:
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relative_times.append([])
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for word in sentence:
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if relative_times[-1] == []:
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relative_times[-1].append(times[word_i]['bg'])
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if len(relative_times[-1]) == 1:
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relative_times[-1].append(times[word_i]['ed'])
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else:
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relative_times[-1][1] = times[word_i]['ed']
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word_i += 1
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# print("relative_times: ", relative_times)
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# generate srt file acoording to relative_times and sentences
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with open(srt_file, 'w') as f:
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for i in range(len(sentences)):
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# Write index number
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f.write(str(i+1)+'\n')
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# Write start and end times
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start = format_time(relative_times[i][0])
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end = format_time(relative_times[i][1])
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f.write(start + ' --> ' + end + '\n')
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# Write text
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f.write(sentences[i]+'\n\n')
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logger.info(f"results saved to {srt_file}")
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def main(args):
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logger.info("asr websocket client start")
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handler = ASRWsAudioHandler(
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args.server_ip,
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args.port,
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endpoint=args.endpoint,
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punc_server_ip=args.punc_server_ip,
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punc_server_port=args.punc_server_port)
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loop = asyncio.get_event_loop()
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# check if the wav file is mp3 format
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# if so, convert it to wav format using convert_to_wav function
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if args.wavfile and os.path.exists(args.wavfile):
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if args.wavfile.endswith(".mp3"):
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convert_to_wav(args.wavfile)
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args.wavfile = args.wavfile.replace(".mp3", ".wav")
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# support to process single audio file
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if args.wavfile and os.path.exists(args.wavfile):
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logger.info(f"start to process the wavscp: {args.wavfile}")
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result = loop.run_until_complete(handler.run(args.wavfile))
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# result = result["result"]
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# logger.info(f"asr websocket client finished : {result}")
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results2srt(result, args.wavfile.replace(".wav", ".srt"))
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# support to process batch audios from wav.scp
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if args.wavscp and os.path.exists(args.wavscp):
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logger.info(f"start to process the wavscp: {args.wavscp}")
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with codecs.open(args.wavscp, 'r', encoding='utf-8') as f,\
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codecs.open("result.txt", 'w', encoding='utf-8') as w:
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for line in f:
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utt_name, utt_path = line.strip().split()
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result = loop.run_until_complete(handler.run(utt_path))
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result = result["result"]
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w.write(f"{utt_name} {result}\n")
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if __name__ == "__main__":
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logger.info("Start to do streaming asr client")
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parser = argparse.ArgumentParser()
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parser.add_argument(
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'--server_ip', type=str, default='127.0.0.1', help='server ip')
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parser.add_argument('--port', type=int, default=8090, help='server port')
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parser.add_argument(
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'--punc.server_ip',
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type=str,
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default=None,
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dest="punc_server_ip",
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help='Punctuation server ip')
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parser.add_argument(
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'--punc.port',
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type=int,
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default=8091,
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dest="punc_server_port",
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help='Punctuation server port')
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parser.add_argument(
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"--endpoint",
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type=str,
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default="/paddlespeech/asr/streaming",
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help="ASR websocket endpoint")
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parser.add_argument(
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"--wavfile",
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action="store",
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help="wav file path ",
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default="./16_audio.wav")
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parser.add_argument(
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"--wavscp", type=str, default=None, help="The batch audios dict text")
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args = parser.parse_args()
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main(args)
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twoDogy
2 years ago
committed by
GitHub