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PaddleSpeech/demos/streaming_asr_server/local/websocket_client_srt.py

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6.0 KiB

Cherry-pick to r1.4 branch (#3798) * [TTS]add Diffsinger with opencpop dataset (#3005) * Update requirements.txt * fix vits reduce_sum's input/output dtype, test=tts (#3028) * [TTS] add opencpop PWGAN example (#3031) * add opencpop voc, test=tts * soft link * Update textnorm_test_cases.txt * [TTS] add opencpop HIFIGAN example (#3038) * add opencpop voc, test=tts * soft link * add opencpop hifigan, test=tts * update * fix dtype diff of last expand_v2 op of VITS (#3041) * [ASR]add squeezeformer model (#2755) * add squeezeformer model * change CodeStyle, test=asr * change CodeStyle, test=asr * fix subsample rate error, test=asr * merge classes as required, test=asr * change CodeStyle, test=asr * fix missing code, test=asr * split code to new file, test=asr * remove rel_shift, test=asr * Update README.md * Update README_cn.md * Update README.md * Update README_cn.md * Update README.md * fix input dtype of elementwise_mul op from bool to int64 (#3054) * [TTS] add svs frontend (#3062) * [TTS]clean starganv2 vc model code and add docstring (#2987) * clean code * add docstring * [Doc] change define asr server config to chunk asr config, test=doc (#3067) * Update README.md * Update README_cn.md * get music score, test=doc (#3070) * [TTS]fix elementwise_floordiv's fill_constant (#3075) * fix elementwise_floordiv's fill_constant * add float converter for min_value in attention * fix paddle2onnx's install version, install the newest paddle2onnx in run.sh (#3084) * [TTS] update svs_music_score.md (#3085) * rm unused dep, test=tts (#3097) * Update bug-report-tts.md (#3120) * [TTS]Fix VITS lite infer (#3098) * [TTS]add starganv2 vc trainer (#3143) * add starganv2 vc trainer * fix StarGANv2VCUpdater and losses * fix StarGANv2VCEvaluator * add some typehint * [TTS]【Hackathon + No.190】 + 模型复现:iSTFTNet (#3006) * iSTFTNet implementation based on hifigan, not affect the function and execution of HIFIGAN * modify the comment in iSTFT.yaml * add the comments in hifigan * iSTFTNet implementation based on hifigan, not affect the function and execution of HIFIGAN * modify the comment in iSTFT.yaml * add the comments in hifigan * add iSTFTNet.md * modify the format of iSTFTNet.md * modify iSTFT.yaml and hifigan.py * Format code using pre-commit * modify hifigan.py,delete the unused self.istft_layer_id , move the self.output_conv behind else, change conv_post to output_conv * update iSTFTNet_csmsc_ckpt.zip download link * modify iSTFTNet.md * modify hifigan.py and iSTFT.yaml * modify iSTFTNet.md * add function for generating srt file (#3123) * add function for generating srt file 在原来websocket_client.py的基础上,增加了由wav或mp3格式的音频文件生成对应srt格式字幕文件的功能 * add function for generating srt file 在原来websocket_client.py的基础上,增加了由wav或mp3格式的音频文件生成对应srt格式字幕文件的功能 * keep origin websocket_client.py 恢复原本的websocket_client.py文件 * add generating subtitle function into README * add generate subtitle funciton into README * add subtitle generation function * add subtitle generation function * fix example/aishell local/train.sh if condition bug, test=asr (#3146) * fix some preprocess bugs (#3155) * add amp for U2 conformer. * fix scaler save * fix scaler save and load. * mv scaler.unscale_ blow grad_clip. * [TTS]add StarGANv2VC preprocess (#3163) * [TTS] [黑客松]Add JETS (#3109) * Update quick_start.md (#3175) * [BUG] Fix progress bar unit. (#3177) * Update quick_start_cn.md (#3176) * [TTS]StarGANv2 VC fix some trainer bugs, add add reset_parameters (#3182) * VITS learning rate revised, test=tts * VITS learning rate revised, test=tts * [s2t] mv dataset into paddlespeech.dataset (#3183) * mv dataset into paddlespeech.dataset * add aidatatang * fix import * Fix some typos. (#3178) * [s2t] move s2t data preprocess into paddlespeech.dataset (#3189) * move s2t data preprocess into paddlespeech.dataset * avg model, compute wer, format rsl into paddlespeech.dataset * fix format rsl * fix avg ckpts * Update pretrained model in README (#3193) * [TTS]Fix losses of StarGAN v2 VC (#3184) * VITS learning rate revised, test=tts * VITS learning rate revised, test=tts * add new aishell model for better CER. * add readme * [s2t] fix cli args to config (#3194) * fix cli args to config * fix train cli * Update README.md * [ASR] Support Hubert, fintuned on the librispeech dataset (#3088) * librispeech hubert, test=asr * librispeech hubert, test=asr * hubert decode * review * copyright, notes, example related * hubert cli * pre-commit format * fix conflicts * fix conflicts * doc related * doc and train config * librispeech.py * support hubert cli * [ASR] fix asr 0-d tensor. (#3214) * Update README.md * Update README.md * fix: 🐛 修复服务端 python ASREngine 无法使用conformer_talcs模型 (#3230) * fix: 🐛 fix python ASREngine not pass codeswitch * docs: 📝 Update Docs * 修改模型判断方式 * Adding WavLM implementation * fix model m5s * Code clean up according to comments in https://github.com/PaddlePaddle/PaddleSpeech/pull/3242 * fix error in tts/st * Changed the path for the uploaded weight * Update phonecode.py # 固话的正则 错误修改 参考https://github.com/speechio/chinese_text_normalization/blob/master/python/cn_tn.py 固化的正则为: pattern = re.compile(r"\D((0(10|2[1-3]|[3-9]\d{2})-?)?[1-9]\d{6,7})\D") * Adapted wavlmASR model to pretrained weights and CLI * Changed the MD5 of the pretrained tar file due to bug fixes * Deleted examples/librispeech/asr5/format_rsl.py * Update released_model.md * Code clean up for CIs * Fixed the transpose usages ignored before * Update setup.py * refactor mfa scripts * Final cleaning; Modified SSL/infer.py and README for wavlm inclusion in model options * updating readme and readme_cn * remove tsinghua pypi * Update setup.py (#3294) * Update setup.py * refactor rhy * fix ckpt * add dtype param for arange API. (#3302) * add scripts for tts code switch * add t2s assets * more comment on tts frontend * fix librosa==0.8.1 numpy==1.23.5 for paddleaudio align with this version * move ssl into t2s.frontend; fix spk_id for 0-D tensor; * add ssml unit test * add en_frontend file * add mix frontend test * fix long text oom using ssml; filter comma; update polyphonic * remove print * hotfix english G2P * en frontend unit text * fix profiler (#3323) * old grad clip has 0d tensor problem, fix it (#3334) * update to py3.8 * remove fluid. * add roformer * fix bugs * add roformer result * support position interpolation for langer attention context windown length. * RoPE with position interpolation * rope for streaming decoding * update result * fix rotary embeding * Update README.md * fix weight decay * fix develop view confict with model's * Add XPU support for SpeedySpeech (#3502) * Add XPU support for SpeedySpeech * fix typos * update description of nxpu * Add XPU support for FastSpeech2 (#3514) * Add XPU support for FastSpeech2 * optimize * Update ge2e_clone.py (#3517) 修复在windows上的多空格错误 * Fix Readme. (#3527) * Update README.md * Update README_cn.md * Update README_cn.md * Update README.md * FIX: Added missing imports * FIX: Fixed the implementation of a special method * 【benchmark】add max_mem_reserved for benchmark (#3604) * fix profiler * add max_mem_reserved for benchmark * fix develop bug function:view to reshape (#3633) * 【benchmark】fix gpu_mem unit (#3634) * fix profiler * add max_mem_reserved for benchmark * fix benchmark * 增加文件编码读取 (#3606) Fixed #3605 * bugfix: audio_len should be 1D, no 0D, which will raise list index out (#3490) of range error in the following decode process Co-authored-by: Luzhenhui <luzhenhui@mqsz.com> * Update README.md (#3532) Fixed a typo * fixed version for paddlepaddle. (#3701) * fixed version for paddlepaddle. * fix code style * 【Fix Speech Issue No.5】issue 3444 transformation import error (#3779) * fix paddlespeech.s2t.transform.transformation import error * fix paddlespeech.s2t.transform import error * 【Fix Speech Issue No.8】issue 3652 merge_yi function has a bug (#3786) * 【Fix Speech Issue No.8】issue 3652 merge_yi function has a bug * 【Fix Speech Issue No.8】issue 3652 merge_yi function has a bug * 【test】add cli test readme (#3784) * add cli test readme * fix code style * 【test】fix test cli bug (#3793) * add cli test readme * fix code style * fix bug * Update setup.py (#3795) * adapt view behavior change, fix KeyError. (#3794) * adapt view behavior change, fix KeyError. * fix readme demo run error. * fixed opencc version --------- Co-authored-by: liangym <34430015+lym0302@users.noreply.github.com> Co-authored-by: TianYuan <white-sky@qq.com> Co-authored-by: 夜雨飘零 <yeyupiaoling@foxmail.com> Co-authored-by: zxcd <228587199@qq.com> Co-authored-by: longRookie <68834517+longRookie@users.noreply.github.com> Co-authored-by: twoDogy <128727742+twoDogy@users.noreply.github.com> Co-authored-by: lemondy <lemondy9@gmail.com> Co-authored-by: ljhzxc <33015549+ljhzxc@users.noreply.github.com> Co-authored-by: PiaoYang <495384481@qq.com> Co-authored-by: WongLaw <mailoflawrence@gmail.com> Co-authored-by: Hui Zhang <zhtclz@foxmail.com> Co-authored-by: Shuangchi He <34329208+Yulv-git@users.noreply.github.com> Co-authored-by: TianHao Zhang <32243340+Zth9730@users.noreply.github.com> Co-authored-by: guanyc <guanyc@gmail.com> Co-authored-by: jiamingkong <kinetical@live.com> Co-authored-by: zoooo0820 <zoooo0820@qq.com> Co-authored-by: shuishu <990941859@qq.com> Co-authored-by: LixinGuo <18510030324@126.com> Co-authored-by: gmm <38800877+mmglove@users.noreply.github.com> Co-authored-by: Wang Huan <wanghuan29@baidu.com> Co-authored-by: Kai Song <50285351+USTCKAY@users.noreply.github.com> Co-authored-by: skyboooox <zcj924@gmail.com> Co-authored-by: fazledyn-or <ataf@openrefactory.com> Co-authored-by: luyao-cv <1367355728@qq.com> Co-authored-by: Color_yr <402067010@qq.com> Co-authored-by: JeffLu <luzhenhui@gmail.com> Co-authored-by: Luzhenhui <luzhenhui@mqsz.com> Co-authored-by: satani99 <42287151+satani99@users.noreply.github.com> Co-authored-by: mjxs <52824616+kk-2000@users.noreply.github.com> Co-authored-by: Mattheliu <leonliuzx@outlook.com>
5 months ago
#!/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)