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- import argparse
- import logging
- from optparse import Option
- import sys
- import json
- from pathlib import Path
- from typing import Any
- from typing import List
- from typing import Optional
- from typing import Sequence
- from typing import Tuple
- from typing import Union
- from typing import Dict
- import numpy as np
- import torch
- from typeguard import check_argument_types
- from funasr.fileio.datadir_writer import DatadirWriter
- from funasr.datasets.preprocessor import LMPreprocessor
- from funasr.tasks.asr import ASRTaskAligner as ASRTask
- from funasr.torch_utils.device_funcs import to_device
- from funasr.torch_utils.set_all_random_seed import set_all_random_seed
- from funasr.utils import config_argparse
- from funasr.utils.cli_utils import get_commandline_args
- from funasr.utils.types import str2bool
- from funasr.utils.types import str2triple_str
- from funasr.utils.types import str_or_none
- from funasr.models.frontend.wav_frontend import WavFrontend
- from funasr.text.token_id_converter import TokenIDConverter
- from funasr.utils.timestamp_tools import ts_prediction_lfr6_standard
- header_colors = '\033[95m'
- end_colors = '\033[0m'
- global_asr_language: str = 'zh-cn'
- global_sample_rate: Union[int, Dict[Any, int]] = {
- 'audio_fs': 16000,
- 'model_fs': 16000
- }
- class SpeechText2Timestamp:
- def __init__(
- self,
- timestamp_infer_config: Union[Path, str] = None,
- timestamp_model_file: Union[Path, str] = None,
- timestamp_cmvn_file: Union[Path, str] = None,
- device: str = "cpu",
- dtype: str = "float32",
- **kwargs,
- ):
- assert check_argument_types()
- # 1. Build ASR model
- tp_model, tp_train_args = ASRTask.build_model_from_file(
- timestamp_infer_config, timestamp_model_file, device=device
- )
- if 'cuda' in device:
- tp_model = tp_model.cuda() # force model to cuda
- frontend = None
- if tp_train_args.frontend is not None:
- frontend = WavFrontend(cmvn_file=timestamp_cmvn_file, **tp_train_args.frontend_conf)
-
- logging.info("tp_model: {}".format(tp_model))
- logging.info("tp_train_args: {}".format(tp_train_args))
- tp_model.to(dtype=getattr(torch, dtype)).eval()
- logging.info(f"Decoding device={device}, dtype={dtype}")
- self.tp_model = tp_model
- self.tp_train_args = tp_train_args
- token_list = self.tp_model.token_list
- self.converter = TokenIDConverter(token_list=token_list)
- self.device = device
- self.dtype = dtype
- self.frontend = frontend
- self.encoder_downsampling_factor = 1
- if tp_train_args.encoder_conf["input_layer"] == "conv2d":
- self.encoder_downsampling_factor = 4
-
- @torch.no_grad()
- def __call__(
- self,
- speech: Union[torch.Tensor, np.ndarray],
- speech_lengths: Union[torch.Tensor, np.ndarray] = None,
- text_lengths: Union[torch.Tensor, np.ndarray] = None
- ):
- assert check_argument_types()
- # Input as audio signal
- if isinstance(speech, np.ndarray):
- speech = torch.tensor(speech)
- if self.frontend is not None:
- feats, feats_len = self.frontend.forward(speech, speech_lengths)
- feats = to_device(feats, device=self.device)
- feats_len = feats_len.int()
- self.tp_model.frontend = None
- else:
- feats = speech
- feats_len = speech_lengths
- # lfr_factor = max(1, (feats.size()[-1]//80)-1)
- batch = {"speech": feats, "speech_lengths": feats_len}
- # a. To device
- batch = to_device(batch, device=self.device)
- # b. Forward Encoder
- enc, enc_len = self.tp_model.encode(**batch)
- if isinstance(enc, tuple):
- enc = enc[0]
- # c. Forward Predictor
- _, _, us_alphas, us_peaks = self.tp_model.calc_predictor_timestamp(enc, enc_len, text_lengths.to(self.device)+1)
- return us_alphas, us_peaks
- def inference(
- batch_size: int,
- ngpu: int,
- log_level: Union[int, str],
- data_path_and_name_and_type,
- timestamp_infer_config: Optional[str],
- timestamp_model_file: Optional[str],
- timestamp_cmvn_file: Optional[str] = None,
- raw_inputs: Union[np.ndarray, torch.Tensor] = None,
- key_file: Optional[str] = None,
- allow_variable_data_keys: bool = False,
- output_dir: Optional[str] = None,
- dtype: str = "float32",
- seed: int = 0,
- num_workers: int = 1,
- split_with_space: bool = True,
- seg_dict_file: Optional[str] = None,
- **kwargs,
- ):
- inference_pipeline = inference_modelscope(
- batch_size=batch_size,
- ngpu=ngpu,
- log_level=log_level,
- timestamp_infer_config=timestamp_infer_config,
- timestamp_model_file=timestamp_model_file,
- timestamp_cmvn_file=timestamp_cmvn_file,
- key_file=key_file,
- allow_variable_data_keys=allow_variable_data_keys,
- output_dir=output_dir,
- dtype=dtype,
- seed=seed,
- num_workers=num_workers,
- split_with_space=split_with_space,
- seg_dict_file=seg_dict_file,
- **kwargs,
- )
- return inference_pipeline(data_path_and_name_and_type, raw_inputs)
- def inference_modelscope(
- batch_size: int,
- ngpu: int,
- log_level: Union[int, str],
- # data_path_and_name_and_type,
- timestamp_infer_config: Optional[str],
- timestamp_model_file: Optional[str],
- timestamp_cmvn_file: Optional[str] = None,
- # raw_inputs: Union[np.ndarray, torch.Tensor] = None,
- key_file: Optional[str] = None,
- allow_variable_data_keys: bool = False,
- output_dir: Optional[str] = None,
- dtype: str = "float32",
- seed: int = 0,
- num_workers: int = 1,
- split_with_space: bool = True,
- seg_dict_file: Optional[str] = None,
- **kwargs,
- ):
- assert check_argument_types()
- ncpu = kwargs.get("ncpu", 1)
- torch.set_num_threads(ncpu)
-
- if batch_size > 1:
- raise NotImplementedError("batch decoding is not implemented")
- if ngpu > 1:
- raise NotImplementedError("only single GPU decoding is supported")
- logging.basicConfig(
- level=log_level,
- format="%(asctime)s (%(module)s:%(lineno)d) %(levelname)s: %(message)s",
- )
- if ngpu >= 1 and torch.cuda.is_available():
- device = "cuda"
- else:
- device = "cpu"
- # 1. Set random-seed
- set_all_random_seed(seed)
- # 2. Build speech2vadsegment
- speechtext2timestamp_kwargs = dict(
- timestamp_infer_config=timestamp_infer_config,
- timestamp_model_file=timestamp_model_file,
- timestamp_cmvn_file=timestamp_cmvn_file,
- device=device,
- dtype=dtype,
- )
- logging.info("speechtext2timestamp_kwargs: {}".format(speechtext2timestamp_kwargs))
- speechtext2timestamp = SpeechText2Timestamp(**speechtext2timestamp_kwargs)
- preprocessor = LMPreprocessor(
- train=False,
- token_type=speechtext2timestamp.tp_train_args.token_type,
- token_list=speechtext2timestamp.tp_train_args.token_list,
- bpemodel=None,
- text_cleaner=None,
- g2p_type=None,
- text_name="text",
- non_linguistic_symbols=speechtext2timestamp.tp_train_args.non_linguistic_symbols,
- split_with_space=split_with_space,
- seg_dict_file=seg_dict_file,
- )
- if output_dir is not None:
- writer = DatadirWriter(output_dir)
- tp_writer = writer[f"timestamp_prediction"]
- # ibest_writer["token_list"][""] = " ".join(speech2text.asr_train_args.token_list)
- else:
- tp_writer = None
-
- def _forward(
- data_path_and_name_and_type,
- raw_inputs: Union[np.ndarray, torch.Tensor] = None,
- output_dir_v2: Optional[str] = None,
- fs: dict = None,
- param_dict: dict = None,
- **kwargs
- ):
- output_path = output_dir_v2 if output_dir_v2 is not None else output_dir
- writer = None
- if output_path is not None:
- writer = DatadirWriter(output_path)
- tp_writer = writer[f"timestamp_prediction"]
- else:
- tp_writer = None
- # 3. Build data-iterator
- if data_path_and_name_and_type is None and raw_inputs is not None:
- if isinstance(raw_inputs, torch.Tensor):
- raw_inputs = raw_inputs.numpy()
- data_path_and_name_and_type = [raw_inputs, "speech", "waveform"]
-
- loader = ASRTask.build_streaming_iterator(
- data_path_and_name_and_type,
- dtype=dtype,
- batch_size=batch_size,
- key_file=key_file,
- num_workers=num_workers,
- preprocess_fn=preprocessor,
- collate_fn=ASRTask.build_collate_fn(speechtext2timestamp.tp_train_args, False),
- allow_variable_data_keys=allow_variable_data_keys,
- inference=True,
- )
- tp_result_list = []
- for keys, batch in loader:
- assert isinstance(batch, dict), type(batch)
- assert all(isinstance(s, str) for s in keys), keys
- _bs = len(next(iter(batch.values())))
- assert len(keys) == _bs, f"{len(keys)} != {_bs}"
- logging.info("timestamp predicting, utt_id: {}".format(keys))
- _batch = {'speech':batch['speech'],
- 'speech_lengths':batch['speech_lengths'],
- 'text_lengths':batch['text_lengths']}
- us_alphas, us_cif_peak = speechtext2timestamp(**_batch)
- for batch_id in range(_bs):
- key = keys[batch_id]
- token = speechtext2timestamp.converter.ids2tokens(batch['text'][batch_id])
- ts_str, ts_list = ts_prediction_lfr6_standard(us_alphas[batch_id], us_cif_peak[batch_id], token, force_time_shift=-3.0)
- logging.warning(ts_str)
- item = {'key': key, 'value': ts_str, 'timestamp':ts_list}
- if tp_writer is not None:
- tp_writer["tp_sync"][key+'#'] = ts_str
- tp_writer["tp_time"][key+'#'] = str(ts_list)
- tp_result_list.append(item)
- return tp_result_list
- return _forward
- def get_parser():
- parser = config_argparse.ArgumentParser(
- description="Timestamp Prediction Inference",
- formatter_class=argparse.ArgumentDefaultsHelpFormatter,
- )
- # Note(kamo): Use '_' instead of '-' as separator.
- # '-' is confusing if written in yaml.
- parser.add_argument(
- "--log_level",
- type=lambda x: x.upper(),
- default="INFO",
- choices=("CRITICAL", "ERROR", "WARNING", "INFO", "DEBUG", "NOTSET"),
- help="The verbose level of logging",
- )
- parser.add_argument("--output_dir", type=str, required=False)
- parser.add_argument(
- "--ngpu",
- type=int,
- default=0,
- help="The number of gpus. 0 indicates CPU mode",
- )
- parser.add_argument(
- "--gpuid_list",
- type=str,
- default="",
- help="The visible gpus",
- )
- parser.add_argument("--seed", type=int, default=0, help="Random seed")
- parser.add_argument(
- "--dtype",
- default="float32",
- choices=["float16", "float32", "float64"],
- help="Data type",
- )
- parser.add_argument(
- "--num_workers",
- type=int,
- default=0,
- help="The number of workers used for DataLoader",
- )
- group = parser.add_argument_group("Input data related")
- group.add_argument(
- "--data_path_and_name_and_type",
- type=str2triple_str,
- required=False,
- action="append",
- )
- group.add_argument("--raw_inputs", type=list, default=None)
- # example=[{'key':'EdevDEWdIYQ_0021','file':'/mnt/data/jiangyu.xzy/test_data/speech_io/SPEECHIO_ASR_ZH00007_zhibodaihuo/wav/EdevDEWdIYQ_0021.wav'}])
- group.add_argument("--key_file", type=str_or_none)
- group.add_argument("--allow_variable_data_keys", type=str2bool, default=False)
- group = parser.add_argument_group("The model configuration related")
- group.add_argument(
- "--timestamp_infer_config",
- type=str,
- help="VAD infer configuration",
- )
- group.add_argument(
- "--timestamp_model_file",
- type=str,
- help="VAD model parameter file",
- )
- group.add_argument(
- "--timestamp_cmvn_file",
- type=str,
- help="Global cmvn file",
- )
- group = parser.add_argument_group("infer related")
- group.add_argument(
- "--batch_size",
- type=int,
- default=1,
- help="The batch size for inference",
- )
- group.add_argument(
- "--seg_dict_file",
- type=str,
- default=None,
- help="The batch size for inference",
- )
- group.add_argument(
- "--split_with_space",
- type=bool,
- default=False,
- help="The batch size for inference",
- )
- return parser
- def main(cmd=None):
- print(get_commandline_args(), file=sys.stderr)
- parser = get_parser()
- args = parser.parse_args(cmd)
- kwargs = vars(args)
- kwargs.pop("config", None)
- inference(**kwargs)
- if __name__ == "__main__":
- main()
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