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PaddleSpeech/deepspeech/exps/deepspeech2/bin/deploy/server.py

135 lines
4.8 KiB

Support paddle 2.x (#538) * 2.x model * model test pass * fix data * fix soundfile with flac support * one thread dataloader test pass * export feasture size add trainer and utils add setup model and dataloader update travis using Bionic dist * add venv; test under venv * fix unittest; train and valid * add train and config * add config and train script * fix ctc cuda memcopy error * fix imports * fix train valid log * fix dataset batch shuffle shift start from 1 fix rank_zero_only decreator error close tensorboard when train over add decoding config and code * test process can run * test with decoding * test and infer with decoding * fix infer * fix ctc loss lr schedule sortagrad logger * aishell egs * refactor train add aishell egs * fix dataset batch shuffle and add batch sampler log print model parameter * fix model and ctc * sequence_mask make all inputs zeros, which cause grad be zero, this is a bug of LessThanOp add grad clip by global norm add model train test notebook * ctc loss remove run prefix using ord value as text id * using unk when training compute_loss need text ids ord id using in test mode, which compute wer/cer * fix tester * add lr_deacy refactor code * fix tools * fix ci add tune fix gru model bugs add dataset and model test * fix decoding * refactor repo fix decoding * fix musan and rir dataset * refactor io, loss, conv, rnn, gradclip, model, utils * fix ci and import * refactor model add export jit model * add deploy bin and test it * rm uselss egs * add layer tools * refactor socket server new model from pretrain * remve useless * fix instability loss and grad nan or inf for librispeech training * fix sampler * fix libri train.sh * fix doc * add license on cpp * fix doc * fix libri script * fix install * clip 5 wer 7.39, clip 400 wer 7.54, 1.8 clip 400 baseline 7.49
4 years ago
# Copyright (c) 2021 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.
"""Server-end for the ASR demo."""
import os
import time
import argparse
import functools
import paddle
import numpy as np
from deepspeech.utils.socket_server import warm_up_test
from deepspeech.utils.socket_server import AsrTCPServer
from deepspeech.utils.socket_server import AsrRequestHandler
from deepspeech.training.cli import default_argument_parser
from deepspeech.exps.deepspeech2.config import get_cfg_defaults
from deepspeech.frontend.utility import read_manifest
from deepspeech.utils.utility import add_arguments, print_arguments
from deepspeech.models.deepspeech2 import DeepSpeech2Model
from deepspeech.io.dataset import ManifestDataset
def start_server(config, args):
"""Start the ASR server"""
dataset = ManifestDataset(
config.data.test_manifest,
config.data.vocab_filepath,
config.data.mean_std_filepath,
augmentation_config="{}",
max_duration=config.data.max_duration,
min_duration=config.data.min_duration,
stride_ms=config.data.stride_ms,
window_ms=config.data.window_ms,
n_fft=config.data.n_fft,
max_freq=config.data.max_freq,
target_sample_rate=config.data.target_sample_rate,
specgram_type=config.data.specgram_type,
use_dB_normalization=config.data.use_dB_normalization,
target_dB=config.data.target_dB,
random_seed=config.data.random_seed,
keep_transcription_text=True)
model = DeepSpeech2Model.from_pretrained(dataset, config,
args.checkpoint_path)
model.eval()
# prepare ASR inference handler
def file_to_transcript(filename):
feature = dataset.process_utterance(filename, "")
audio = np.array([feature[0]]).astype('float32') #[1, D, T]
audio_len = feature[0].shape[1]
audio_len = np.array([audio_len]).astype('int64') # [1]
result_transcript = model.decode(
paddle.to_tensor(audio),
paddle.to_tensor(audio_len),
vocab_list=dataset.vocab_list,
decoding_method=config.decoding.decoding_method,
lang_model_path=config.decoding.lang_model_path,
beam_alpha=config.decoding.alpha,
beam_beta=config.decoding.beta,
beam_size=config.decoding.beam_size,
cutoff_prob=config.decoding.cutoff_prob,
cutoff_top_n=config.decoding.cutoff_top_n,
num_processes=config.decoding.num_proc_bsearch)
return result_transcript[0]
# warming up with utterrances sampled from Librispeech
print('-----------------------------------------------------------')
print('Warming up ...')
warm_up_test(
audio_process_handler=file_to_transcript,
manifest_path=args.warmup_manifest,
num_test_cases=3)
print('-----------------------------------------------------------')
# start the server
server = AsrTCPServer(
server_address=(args.host_ip, args.host_port),
RequestHandlerClass=AsrRequestHandler,
speech_save_dir=args.speech_save_dir,
audio_process_handler=file_to_transcript)
print("ASR Server Started.")
server.serve_forever()
def main(config, args):
start_server(config, args)
if __name__ == "__main__":
parser = default_argument_parser()
add_arg = functools.partial(add_arguments, argparser=parser)
# yapf: disable
add_arg('host_ip', str,
'localhost',
"Server's IP address.")
add_arg('host_port', int, 8086, "Server's IP port.")
add_arg('speech_save_dir', str,
'demo_cache',
"Directory to save demo audios.")
add_arg('warmup_manifest', str, None, "Filepath of manifest to warm up.")
args = parser.parse_args()
print_arguments(args)
# https://yaml.org/type/float.html
config = get_cfg_defaults()
if args.config:
config.merge_from_file(args.config)
if args.opts:
config.merge_from_list(args.opts)
config.freeze()
print(config)
args.warmup_manifest = config.data.test_manifest
print_arguments(args)
if args.dump_config:
with open(args.dump_config, 'w') as f:
print(config, file=f)
main(config, args)