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# Copyright (c) 2021 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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import argparse
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import io
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import os
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import sys
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import time
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from collections import OrderedDict
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from typing import List
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from typing import Optional
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from typing import Union
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import librosa
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import numpy as np
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import paddle
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import soundfile
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from paddlenlp.transformers import AutoTokenizer
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from yacs.config import CfgNode
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from ..executor import BaseExecutor
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from ..log import logger
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from ..utils import CLI_TIMER
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from ..utils import stats_wrapper
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from ..utils import timer_register
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from paddlespeech.audio.transform.transformation import Transformation
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from paddlespeech.s2t.frontend.featurizer.text_featurizer import TextFeaturizer
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from paddlespeech.s2t.utils.utility import UpdateConfig
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__all__ = ['SSLExecutor']
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@timer_register
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class SSLExecutor(BaseExecutor):
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def __init__(self):
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super().__init__('ssl')
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self.parser = argparse.ArgumentParser(
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prog='paddlespeech.ssl', add_help=True)
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self.parser.add_argument(
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'--input', type=str, default=None, help='Audio file to recognize.')
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self.parser.add_argument(
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'--model',
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type=str,
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default='wav2vec2',
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choices=['wav2vec2', 'hubert', "wavlm"],
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help='Choose model type of asr task.')
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self.parser.add_argument(
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'--task',
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type=str,
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default='asr',
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choices=['asr', 'vector'],
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help='Choose output type for ssl task')
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self.parser.add_argument(
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'--lang',
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type=str,
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default='en',
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help='Choose model language. zh or en, zh:[wav2vec2ASR_aishell1-zh-16k], en:[wav2vec2ASR_librispeech-en-16k, hubertASR_librispeech_100-en-16k]'
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)
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self.parser.add_argument(
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"--sample_rate",
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type=int,
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default=16000,
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choices=[8000, 16000],
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help='Choose the audio sample rate of the model. 8000 or 16000')
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self.parser.add_argument(
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'--config',
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type=str,
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default=None,
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help='Config of asr task. Use deault config when it is None.')
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self.parser.add_argument(
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'--decode_method',
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type=str,
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default='ctc_greedy_search',
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choices=[
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'ctc_greedy_search',
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'ctc_prefix_beam_search',
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],
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help='only support asr task')
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self.parser.add_argument(
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'--ckpt_path',
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type=str,
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default=None,
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help='Checkpoint file of model.')
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self.parser.add_argument(
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'--yes',
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'-y',
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action="store_true",
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default=False,
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help='No additional parameters required. \
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Once set this parameter, it means accepting the request of the program by default, \
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which includes transforming the audio sample rate')
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self.parser.add_argument(
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'--rtf',
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action="store_true",
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default=False,
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help='Show Real-time Factor(RTF).')
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self.parser.add_argument(
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'--device',
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type=str,
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default=paddle.get_device(),
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help='Choose device to execute model inference.')
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self.parser.add_argument(
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'-d',
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'--job_dump_result',
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action='store_true',
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help='Save job result into file.')
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self.parser.add_argument(
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'-v',
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'--verbose',
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action='store_true',
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help='Increase logger verbosity of current task.')
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self.last_call_params = None
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def _init_from_path(self,
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model_type: str=None,
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task: str='asr',
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lang: str='en',
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sample_rate: int=16000,
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cfg_path: Optional[os.PathLike]=None,
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decode_method: str='ctc_greedy_search',
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ckpt_path: Optional[os.PathLike]=None):
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"""
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Init model and other resources from a specific path.
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"""
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logger.debug("start to init the model")
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if model_type is None:
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logger.debug(
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"Model type had not been specified, default {} was used.".
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format(model_type))
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# default max_len: unit:second
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self.max_len = 50
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if hasattr(self, 'model'):
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logger.debug('Model had been initialized.')
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return
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if cfg_path is None or ckpt_path is None:
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sample_rate_str = '16k' if sample_rate == 16000 else '8k'
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if task == 'asr':
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if model_type == 'wav2vec2':
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if lang == 'en':
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model_prefix = 'wav2vec2ASR_librispeech'
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elif lang == 'zh':
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model_prefix = 'wav2vec2ASR_aishell1'
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tag = model_prefix + '-' + lang + '-' + sample_rate_str
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elif model_type == 'hubert':
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if lang == 'en':
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model_prefix = 'hubertASR_librispeech-100h'
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elif lang == 'zh':
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logger.error("zh hubertASR is not supported yet")
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tag = model_prefix + '-' + lang + '-' + sample_rate_str
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elif model_type == 'wavlm':
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if lang == "en":
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model_prefix = "wavlmASR_librispeech"
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elif lang == "zh":
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logger.error("zh wavlmASR is not supported yet")
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tag = model_prefix + '-' + lang + '-' + sample_rate_str
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else:
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tag = model_type + '-' + lang + '-' + sample_rate_str
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self.task_resource.set_task_model(tag, version=None)
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self.res_path = self.task_resource.res_dir
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self.cfg_path = os.path.join(
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self.res_path, self.task_resource.res_dict['cfg_path'])
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self.ckpt_path = os.path.join(
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self.res_path,
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self.task_resource.res_dict['ckpt_path'] + ".pdparams")
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logger.debug(self.res_path)
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else:
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self.cfg_path = os.path.abspath(cfg_path)
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self.ckpt_path = os.path.abspath(ckpt_path + ".pdparams")
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self.res_path = os.path.dirname(
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os.path.dirname(os.path.abspath(self.cfg_path)))
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logger.debug(self.cfg_path)
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logger.debug(self.ckpt_path)
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#Init body.
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self.config = CfgNode(new_allowed=True)
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self.config.merge_from_file(self.cfg_path)
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if task == 'asr':
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with UpdateConfig(self.config):
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if lang == 'en':
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self.text_feature = TextFeaturizer(
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unit_type=self.config.unit_type,
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vocab=self.config.vocab_filepath)
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self.config.output_dim = len(self.config.vocab_filepath)
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elif lang == 'zh':
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self.text_feature = AutoTokenizer.from_pretrained(
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self.config.tokenizer)
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self.config.output_dim = self.text_feature.vocab_size
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self.config.decode.decoding_method = decode_method
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model_name = model_prefix[:model_prefix.rindex(
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'_')] # model_type: {model_name}_{dataset}
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else:
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model_name = model_type
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model_class = self.task_resource.get_model_class(model_name)
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model_conf = self.config
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model = model_class.from_config(model_conf)
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self.model = model
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self.model.eval()
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# load model
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model_dict = paddle.load(self.ckpt_path)
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if task == 'asr':
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self.model.set_state_dict(model_dict)
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else:
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getattr(self.model, model_type).set_state_dict(model_dict)
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def preprocess(self, input: Union[str, os.PathLike]):
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"""
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Input preprocess and return paddle.Tensor stored in self.input.
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Input content can be a text(tts), a file(asr, cls) or a streaming(not supported yet).
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"""
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audio_file = input
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if isinstance(audio_file, (str, os.PathLike)):
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logger.debug("Preprocess audio_file:" + audio_file)
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elif isinstance(audio_file, io.BytesIO):
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audio_file.seek(0)
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# Get the object for feature extraction
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logger.debug("get the preprocess conf")
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preprocess_conf = self.config.preprocess_config
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preprocess_args = {"train": False}
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preprocessing = Transformation(preprocess_conf)
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logger.debug("read the audio file")
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audio, audio_sample_rate = soundfile.read(
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audio_file, dtype="int16", always_2d=True)
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if self.change_format:
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if audio.shape[1] >= 2:
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audio = audio.mean(axis=1, dtype=np.int16)
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else:
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audio = audio[:, 0]
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# pcm16 -> pcm 32
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audio = self._pcm16to32(audio)
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audio = librosa.resample(
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audio, orig_sr=audio_sample_rate, target_sr=self.sample_rate)
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audio_sample_rate = self.sample_rate
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# pcm32 -> pcm 16
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audio = self._pcm32to16(audio)
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else:
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audio = audio[:, 0]
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logger.debug(f"audio shape: {audio.shape}")
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# fbank
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audio = preprocessing(audio, **preprocess_args)
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audio_len = paddle.to_tensor(audio.shape[0]).unsqueeze(axis=0)
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audio = paddle.to_tensor(audio, dtype='float32').unsqueeze(axis=0)
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self._inputs["audio"] = audio
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self._inputs["audio_len"] = audio_len
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logger.debug(f"audio feat shape: {audio.shape}")
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logger.debug("audio feat process success")
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@paddle.no_grad()
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def infer(self, model_type: str, task: str):
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"""
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Model inference and result stored in self.output.
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"""
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logger.debug("start to infer the model to get the output")
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audio = self._inputs["audio"]
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if task == 'asr':
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cfg = self.config.decode
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logger.debug(f"we will use the {model_type}ASR like model.")
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try:
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result_transcripts = self.model.decode(
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audio,
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text_feature=self.text_feature,
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decoding_method=cfg.decoding_method,
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beam_size=cfg.beam_size,
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tokenizer=getattr(self.config, 'tokenizer', None))
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self._outputs["result"] = result_transcripts[0][0]
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except Exception as e:
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logger.exception(e)
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else:
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logger.debug(
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f"we will use the {model_type} like model to extract audio feature."
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)
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try:
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out_feature = self.model(audio[:, :, 0])
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self._outputs["result"] = out_feature[0]
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except Exception as e:
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logger.exception(e)
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def postprocess(self) -> Union[str, os.PathLike]:
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"""
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Output postprocess and return human-readable results such as texts and audio files.
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"""
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return self._outputs["result"]
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def _pcm16to32(self, audio):
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assert (audio.dtype == np.int16)
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audio = audio.astype("float32")
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bits = np.iinfo(np.int16).bits
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audio = audio / (2**(bits - 1))
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return audio
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def _pcm32to16(self, audio):
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assert (audio.dtype == np.float32)
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bits = np.iinfo(np.int16).bits
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audio = audio * (2**(bits - 1))
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audio = np.round(audio).astype("int16")
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return audio
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def _check(self, audio_file: str, sample_rate: int, force_yes: bool=False):
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self.sample_rate = sample_rate
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if self.sample_rate != 16000 and self.sample_rate != 8000:
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logger.error(
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"invalid sample rate, please input --sr 8000 or --sr 16000")
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return False
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if isinstance(audio_file, (str, os.PathLike)):
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if not os.path.isfile(audio_file):
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logger.error("Please input the right audio file path")
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return False
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elif isinstance(audio_file, io.BytesIO):
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audio_file.seek(0)
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logger.debug("checking the audio file format......")
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try:
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audio, audio_sample_rate = soundfile.read(
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audio_file, dtype="int16", always_2d=True)
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audio_duration = audio.shape[0] / audio_sample_rate
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if audio_duration > self.max_len:
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logger.error(
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f"Please input audio file less then {self.max_len} seconds.\n"
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)
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return False
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except Exception as e:
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logger.exception(e)
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logger.error(
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f"can not open the audio file, please check the audio file({audio_file}) format is 'wav'. \n \
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you can try to use sox to change the file format.\n \
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For example: \n \
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sample rate: 16k \n \
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sox input_audio.xx --rate 16k --bits 16 --channels 1 output_audio.wav \n \
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sample rate: 8k \n \
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sox input_audio.xx --rate 8k --bits 16 --channels 1 output_audio.wav \n \
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")
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return False
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logger.debug("The sample rate is %d" % audio_sample_rate)
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if audio_sample_rate != self.sample_rate:
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logger.warning("The sample rate of the input file is not {}.\n \
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The program will resample the wav file to {}.\n \
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If the result does not meet your expectations，\n \
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Please input the 16k 16 bit 1 channel wav file. \
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".format(self.sample_rate, self.sample_rate))
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if force_yes is False:
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while (True):
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logger.debug(
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"Whether to change the sample rate and the channel. Y: change the sample. N: exit the prgream."
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)
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content = input("Input(Y/N):")
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if content.strip() == "Y" or content.strip(
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) == "y" or content.strip() == "yes" or content.strip(
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) == "Yes":
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logger.debug(
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"change the sampele rate, channel to 16k and 1 channel"
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)
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break
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elif content.strip() == "N" or content.strip(
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) == "n" or content.strip() == "no" or content.strip(
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) == "No":
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logger.debug("Exit the program")
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return False
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else:
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logger.warning("Not regular input, please input again")
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self.change_format = True
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else:
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logger.debug("The audio file format is right")
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self.change_format = False
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return True
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def execute(self, argv: List[str]) -> bool:
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"""
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Command line entry.
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"""
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parser_args = self.parser.parse_args(argv)
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model = parser_args.model
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task = parser_args.task
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lang = parser_args.lang
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sample_rate = parser_args.sample_rate
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config = parser_args.config
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ckpt_path = parser_args.ckpt_path
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decode_method = parser_args.decode_method
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force_yes = parser_args.yes
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rtf = parser_args.rtf
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device = parser_args.device
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if not parser_args.verbose:
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self.disable_task_loggers()
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task_source = self.get_input_source(parser_args.input)
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task_results = OrderedDict()
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has_exceptions = False
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for id_, input_ in task_source.items():
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try:
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res = self(
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audio_file=input_,
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model=model,
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task=task,
|
|
|
lang=lang,
|
|
|
sample_rate=sample_rate,
|
|
|
config=config,
|
|
|
ckpt_path=ckpt_path,
|
|
|
decode_method=decode_method,
|
|
|
force_yes=force_yes,
|
|
|
rtf=rtf,
|
|
|
device=device)
|
|
|
task_results[id_] = res
|
|
|
|
|
|
except Exception as e:
|
|
|
has_exceptions = True
|
|
|
task_results[id_] = f'{e.__class__.__name__}: {e}'
|
|
|
|
|
|
if rtf:
|
|
|
self.show_rtf(CLI_TIMER[self.__class__.__name__])
|
|
|
self.process_task_results(parser_args.input, task_results,
|
|
|
parser_args.job_dump_result)
|
|
|
if has_exceptions:
|
|
|
return False
|
|
|
else:
|
|
|
return True
|
|
|
|
|
|
@stats_wrapper
|
|
|
def __call__(self,
|
|
|
audio_file: os.PathLike,
|
|
|
model: str=None,
|
|
|
task: str='asr',
|
|
|
lang: str='en',
|
|
|
sample_rate: int=16000,
|
|
|
config: os.PathLike=None,
|
|
|
ckpt_path: os.PathLike=None,
|
|
|
decode_method: str='ctc_greedy_search',
|
|
|
force_yes: bool=False,
|
|
|
rtf: bool=False,
|
|
|
device=paddle.get_device()):
|
|
|
"""
|
|
|
Python API to call an executor.
|
|
|
"""
|
|
|
|
|
|
current_call_params = {
|
|
|
"model": model,
|
|
|
"task": task,
|
|
|
"lang": lang,
|
|
|
"sample_rate": sample_rate,
|
|
|
"config": config,
|
|
|
"ckpt_path": ckpt_path,
|
|
|
"decode_method": decode_method,
|
|
|
"force_yes": force_yes,
|
|
|
"rtf": rtf,
|
|
|
"device": device
|
|
|
}
|
|
|
if self.last_call_params is not None and self.last_call_params != current_call_params and hasattr(
|
|
|
self, 'model'):
|
|
|
del self.model
|
|
|
self.last_call_params = current_call_params
|
|
|
|
|
|
audio_file = os.path.abspath(audio_file)
|
|
|
paddle.set_device(device)
|
|
|
self._init_from_path(model, task, lang, sample_rate, config,
|
|
|
decode_method, ckpt_path)
|
|
|
if not self._check(audio_file, sample_rate, force_yes):
|
|
|
sys.exit(-1)
|
|
|
if rtf:
|
|
|
k = self.__class__.__name__
|
|
|
CLI_TIMER[k]['start'].append(time.time())
|
|
|
self.preprocess(audio_file)
|
|
|
self.infer(model, task)
|
|
|
res = self.postprocess() # Retrieve result of asr.
|
|
|
|
|
|
if rtf:
|
|
|
CLI_TIMER[k]['end'].append(time.time())
|
|
|
audio, audio_sample_rate = soundfile.read(
|
|
|
audio_file, dtype="int16", always_2d=True)
|
|
|
CLI_TIMER[k]['extra'].append(audio.shape[0] / audio_sample_rate)
|
|
|
|
|
|
return res
|
