#!/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)