FunASR/funasr/bin/asr_inference_paraformer.py
2022-12-28 11:51:17 +08:00

650 lines
22 KiB
Python
Executable File

#!/usr/bin/env python3
import argparse
import logging
import sys
import time
from pathlib import Path
from typing import Optional
from typing import Sequence
from typing import Tuple
from typing import Union
from typing import Dict
from typing import Any
from typing import List
import numpy as np
import torch
from typeguard import check_argument_types
from funasr.fileio.datadir_writer import DatadirWriter
from funasr.modules.beam_search.beam_search import BeamSearchPara as BeamSearch
from funasr.modules.beam_search.beam_search import Hypothesis
from funasr.modules.scorers.ctc import CTCPrefixScorer
from funasr.modules.scorers.length_bonus import LengthBonus
from funasr.modules.subsampling import TooShortUttError
from funasr.tasks.asr import ASRTaskParaformer as ASRTask
from funasr.tasks.lm import LMTask
from funasr.text.build_tokenizer import build_tokenizer
from funasr.text.token_id_converter import TokenIDConverter
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.utils import asr_utils, wav_utils, postprocess_utils
from funasr.models.frontend.wav_frontend import WavFrontend
from modelscope.utils.logger import get_logger
logger = get_logger()
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 Speech2Text:
"""Speech2Text class
Examples:
>>> import soundfile
>>> speech2text = Speech2Text("asr_config.yml", "asr.pth")
>>> audio, rate = soundfile.read("speech.wav")
>>> speech2text(audio)
[(text, token, token_int, hypothesis object), ...]
"""
def __init__(
self,
asr_train_config: Union[Path, str] = None,
asr_model_file: Union[Path, str] = None,
lm_train_config: Union[Path, str] = None,
lm_file: Union[Path, str] = None,
token_type: str = None,
bpemodel: str = None,
device: str = "cpu",
maxlenratio: float = 0.0,
minlenratio: float = 0.0,
dtype: str = "float32",
beam_size: int = 20,
ctc_weight: float = 0.5,
lm_weight: float = 1.0,
ngram_weight: float = 0.9,
penalty: float = 0.0,
nbest: int = 1,
frontend_conf: dict = None,
**kwargs,
):
assert check_argument_types()
# 1. Build ASR model
scorers = {}
asr_model, asr_train_args = ASRTask.build_model_from_file(
asr_train_config, asr_model_file, device
)
if asr_model.frontend is None and frontend_conf is not None:
frontend = WavFrontend(**frontend_conf)
asr_model.frontend = frontend
# logging.info("asr_model: {}".format(asr_model))
# logging.info("asr_train_args: {}".format(asr_train_args))
asr_model.to(dtype=getattr(torch, dtype)).eval()
ctc = CTCPrefixScorer(ctc=asr_model.ctc, eos=asr_model.eos)
token_list = asr_model.token_list
scorers.update(
ctc=ctc,
length_bonus=LengthBonus(len(token_list)),
)
# 2. Build Language model
if lm_train_config is not None:
lm, lm_train_args = LMTask.build_model_from_file(
lm_train_config, lm_file, device
)
scorers["lm"] = lm.lm
# 3. Build ngram model
# ngram is not supported now
ngram = None
scorers["ngram"] = ngram
# 4. Build BeamSearch object
# transducer is not supported now
beam_search_transducer = None
weights = dict(
decoder=1.0 - ctc_weight,
ctc=ctc_weight,
lm=lm_weight,
ngram=ngram_weight,
length_bonus=penalty,
)
beam_search = BeamSearch(
beam_size=beam_size,
weights=weights,
scorers=scorers,
sos=asr_model.sos,
eos=asr_model.eos,
vocab_size=len(token_list),
token_list=token_list,
pre_beam_score_key=None if ctc_weight == 1.0 else "full",
)
beam_search.to(device=device, dtype=getattr(torch, dtype)).eval()
for scorer in scorers.values():
if isinstance(scorer, torch.nn.Module):
scorer.to(device=device, dtype=getattr(torch, dtype)).eval()
# logging.info(f"Beam_search: {beam_search}")
# logging.info(f"Decoding device={device}, dtype={dtype}")
# 5. [Optional] Build Text converter: e.g. bpe-sym -> Text
if token_type is None:
token_type = asr_train_args.token_type
if bpemodel is None:
bpemodel = asr_train_args.bpemodel
if token_type is None:
tokenizer = None
elif token_type == "bpe":
if bpemodel is not None:
tokenizer = build_tokenizer(token_type=token_type, bpemodel=bpemodel)
else:
tokenizer = None
else:
tokenizer = build_tokenizer(token_type=token_type)
converter = TokenIDConverter(token_list=token_list)
# logging.info(f"Text tokenizer: {tokenizer}")
self.asr_model = asr_model
self.asr_train_args = asr_train_args
self.converter = converter
self.tokenizer = tokenizer
has_lm = lm_weight == 0.0 or lm_file is None
if ctc_weight == 0.0 and has_lm:
beam_search = None
self.beam_search = beam_search
self.beam_search_transducer = beam_search_transducer
self.maxlenratio = maxlenratio
self.minlenratio = minlenratio
self.device = device
self.dtype = dtype
self.nbest = nbest
@torch.no_grad()
def __call__(
self, speech: Union[torch.Tensor, np.ndarray], speech_lengths: Union[torch.Tensor, np.ndarray] = None
):
"""Inference
Args:
speech: Input speech data
Returns:
text, token, token_int, hyp
"""
assert check_argument_types()
# Input as audio signal
if isinstance(speech, np.ndarray):
speech = torch.tensor(speech)
# data: (Nsamples,) -> (1, Nsamples)
# lengths: (1,)
# if len(speech.size()) < 3:
# speech = speech.unsqueeze(0).to(getattr(torch, self.dtype))
# speech_lengths = speech.new_full([1], dtype=torch.long, fill_value=speech.size(1))
lfr_factor = max(1, (speech.size()[-1]//80)-1)
batch = {"speech": speech, "speech_lengths": speech_lengths}
# a. To device
batch = to_device(batch, device=self.device)
# b. Forward Encoder
enc, enc_len = self.asr_model.encode(**batch)
if isinstance(enc, tuple):
enc = enc[0]
# assert len(enc) == 1, len(enc)
enc_len_batch_total = torch.sum(enc_len).item()
predictor_outs = self.asr_model.calc_predictor(enc, enc_len)
pre_acoustic_embeds, pre_token_length = predictor_outs[0], predictor_outs[1]
pre_token_length = pre_token_length.round().long()
decoder_outs = self.asr_model.cal_decoder_with_predictor(enc, enc_len, pre_acoustic_embeds, pre_token_length)
decoder_out, ys_pad_lens = decoder_outs[0], decoder_outs[1]
results = []
b, n, d = decoder_out.size()
for i in range(b):
x = enc[i, :enc_len[i], :]
am_scores = decoder_out[i, :pre_token_length[i], :]
if self.beam_search is not None:
nbest_hyps = self.beam_search(
x=x, am_scores=am_scores, maxlenratio=self.maxlenratio, minlenratio=self.minlenratio
)
nbest_hyps = nbest_hyps[: self.nbest]
else:
yseq = am_scores.argmax(dim=-1)
score = am_scores.max(dim=-1)[0]
score = torch.sum(score, dim=-1)
# pad with mask tokens to ensure compatibility with sos/eos tokens
yseq = torch.tensor(
[self.asr_model.sos] + yseq.tolist() + [self.asr_model.eos], device=yseq.device
)
nbest_hyps = [Hypothesis(yseq=yseq, score=score)]
for hyp in nbest_hyps:
assert isinstance(hyp, (Hypothesis)), type(hyp)
# remove sos/eos and get results
last_pos = -1
if isinstance(hyp.yseq, list):
token_int = hyp.yseq[1:last_pos]
else:
token_int = hyp.yseq[1:last_pos].tolist()
# remove blank symbol id, which is assumed to be 0
token_int = list(filter(lambda x: x != 0, token_int))
# Change integer-ids to tokens
token = self.converter.ids2tokens(token_int)
if self.tokenizer is not None:
text = self.tokenizer.tokens2text(token)
else:
text = None
results.append((text, token, token_int, hyp, enc_len_batch_total, lfr_factor))
# assert check_return_type(results)
return results
def inference(
maxlenratio: float,
minlenratio: float,
batch_size: int,
beam_size: int,
ngpu: int,
ctc_weight: float,
lm_weight: float,
penalty: float,
log_level: Union[int, str],
data_path_and_name_and_type,
asr_train_config: Optional[str],
asr_model_file: Optional[str],
audio_lists: Union[List[Any], bytes] = None,
lm_train_config: Optional[str] = None,
lm_file: Optional[str] = None,
token_type: Optional[str] = None,
key_file: Optional[str] = None,
word_lm_train_config: Optional[str] = None,
bpemodel: Optional[str] = None,
allow_variable_data_keys: bool = False,
streaming: bool = False,
output_dir: Optional[str] = None,
dtype: str = "float32",
seed: int = 0,
ngram_weight: float = 0.9,
nbest: int = 1,
num_workers: int = 1,
frontend_conf: dict = None,
fs: Union[dict, int] = 16000,
lang: Optional[str] = None,
**kwargs,
):
assert check_argument_types()
if word_lm_train_config is not None:
raise NotImplementedError("Word LM 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:
device = "cuda"
else:
device = "cpu"
hop_length: int = 160
sr: int = 16000
if isinstance(fs, int):
sr = fs
else:
if 'model_fs' in fs and fs['model_fs'] is not None:
sr = fs['model_fs']
# data_path_and_name_and_type for modelscope: (data from audio_lists)
# ['speech', 'sound', 'am.mvn']
# data_path_and_name_and_type for funasr:
# [('/mnt/data/jiangyu.xzy/exp/maas/mvn.1.scp', 'speech', 'kaldi_ark')]
if isinstance(data_path_and_name_and_type[0], Tuple):
features_type: str = data_path_and_name_and_type[0][1]
elif isinstance(data_path_and_name_and_type[0], str):
features_type: str = data_path_and_name_and_type[1]
else:
raise NotImplementedError("unknown features type:{0}".format(data_path_and_name_and_type))
if features_type != 'sound':
frontend_conf = None
flag_modelscope = False
else:
flag_modelscope = True
if frontend_conf is not None:
if 'hop_length' in frontend_conf:
hop_length = frontend_conf['hop_length']
finish_count = 0
file_count = 1
if flag_modelscope and not isinstance(data_path_and_name_and_type[0], Tuple):
data_path_and_name_and_type_new = [
audio_lists, data_path_and_name_and_type[0], data_path_and_name_and_type[1]
]
if isinstance(audio_lists, bytes):
file_count = 1
else:
file_count = len(audio_lists)
if len(data_path_and_name_and_type) >= 3 and frontend_conf is not None:
mvn_file = data_path_and_name_and_type[2]
mvn_data = wav_utils.extract_CMVN_featrures(mvn_file)
frontend_conf['mvn_data'] = mvn_data
# 1. Set random-seed
set_all_random_seed(seed)
# 2. Build speech2text
speech2text_kwargs = dict(
asr_train_config=asr_train_config,
asr_model_file=asr_model_file,
lm_train_config=lm_train_config,
lm_file=lm_file,
token_type=token_type,
bpemodel=bpemodel,
device=device,
maxlenratio=maxlenratio,
minlenratio=minlenratio,
dtype=dtype,
beam_size=beam_size,
ctc_weight=ctc_weight,
lm_weight=lm_weight,
ngram_weight=ngram_weight,
penalty=penalty,
nbest=nbest,
frontend_conf=frontend_conf,
)
speech2text = Speech2Text(**speech2text_kwargs)
# 3. Build data-iterator
if flag_modelscope:
loader = ASRTask.build_streaming_iterator_modelscope(
data_path_and_name_and_type_new,
dtype=dtype,
batch_size=batch_size,
key_file=key_file,
num_workers=num_workers,
preprocess_fn=ASRTask.build_preprocess_fn(speech2text.asr_train_args, False),
collate_fn=ASRTask.build_collate_fn(speech2text.asr_train_args, False),
allow_variable_data_keys=allow_variable_data_keys,
inference=True,
sample_rate=fs
)
else:
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=ASRTask.build_preprocess_fn(speech2text.asr_train_args, False),
collate_fn=ASRTask.build_collate_fn(speech2text.asr_train_args, False),
allow_variable_data_keys=allow_variable_data_keys,
inference=True,
)
forward_time_total = 0.0
length_total = 0.0
# 7 .Start for-loop
# FIXME(kamo): The output format should be discussed about
asr_result_list = []
if output_dir is not None:
writer = DatadirWriter(output_dir)
else:
writer = None
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}"
# batch = {k: v for k, v in batch.items() if not k.endswith("_lengths")}
# logging.info("decoding, utt_id: {}".format(keys))
# N-best list of (text, token, token_int, hyp_object)
time_beg = time.time()
results = speech2text(**batch)
time_end = time.time()
forward_time = time_end - time_beg
lfr_factor = results[0][-1]
length = results[0][-2]
forward_time_total += forward_time
length_total += length
logging.info(
"decoding, feature length: {}, forward_time: {:.4f}, rtf: {:.4f}".
format(length, forward_time, 100 * forward_time / (length*lfr_factor)))
for batch_id in range(_bs):
result = [results[batch_id][:-2]]
key = keys[batch_id]
for n, (text, token, token_int, hyp) in zip(range(1, nbest + 1), result):
# Create a directory: outdir/{n}best_recog
if writer is not None:
ibest_writer = writer[f"{n}best_recog"]
# Write the result to each file
ibest_writer["token"][key] = " ".join(token)
ibest_writer["token_int"][key] = " ".join(map(str, token_int))
ibest_writer["score"][key] = str(hyp.score)
if text is not None:
text_postprocessed = postprocess_utils.sentence_postprocess(token)
item = {'key': key, 'value': text_postprocessed}
asr_result_list.append(item)
finish_count += 1
asr_utils.print_progress(finish_count / file_count)
if writer is not None:
ibest_writer["text"][key] = text
logging.info("decoding, utt: {}, predictions: {}".format(key, text))
logging.info("decoding, feature length total: {}, forward_time total: {:.4f}, rtf avg: {:.4f}".
format(length_total, forward_time_total, 100 * forward_time_total / (length_total*lfr_factor)))
return asr_result_list
def set_parameters(language: str = None,
sample_rate: Union[int, Dict[Any, int]] = None):
if language is not None:
global global_asr_language
global_asr_language = language
if sample_rate is not None:
global global_sample_rate
global_sample_rate = sample_rate
def get_parser():
parser = config_argparse.ArgumentParser(
description="ASR Decoding",
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=True)
parser.add_argument(
"--ngpu",
type=int,
default=0,
help="The number of gpus. 0 indicates CPU mode",
)
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=1,
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=True,
action="append",
)
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(
"--asr_train_config",
type=str,
help="ASR training configuration",
)
group.add_argument(
"--asr_model_file",
type=str,
help="ASR model parameter file",
)
group.add_argument(
"--lm_train_config",
type=str,
help="LM training configuration",
)
group.add_argument(
"--lm_file",
type=str,
help="LM parameter file",
)
group.add_argument(
"--word_lm_train_config",
type=str,
help="Word LM training configuration",
)
group.add_argument(
"--word_lm_file",
type=str,
help="Word LM parameter file",
)
group.add_argument(
"--ngram_file",
type=str,
help="N-gram parameter file",
)
group.add_argument(
"--model_tag",
type=str,
help="Pretrained model tag. If specify this option, *_train_config and "
"*_file will be overwritten",
)
group = parser.add_argument_group("Beam-search related")
group.add_argument(
"--batch_size",
type=int,
default=1,
help="The batch size for inference",
)
group.add_argument("--nbest", type=int, default=1, help="Output N-best hypotheses")
group.add_argument("--beam_size", type=int, default=20, help="Beam size")
group.add_argument("--penalty", type=float, default=0.0, help="Insertion penalty")
group.add_argument(
"--maxlenratio",
type=float,
default=0.0,
help="Input length ratio to obtain max output length. "
"If maxlenratio=0.0 (default), it uses a end-detect "
"function "
"to automatically find maximum hypothesis lengths."
"If maxlenratio<0.0, its absolute value is interpreted"
"as a constant max output length",
)
group.add_argument(
"--minlenratio",
type=float,
default=0.0,
help="Input length ratio to obtain min output length",
)
group.add_argument(
"--ctc_weight",
type=float,
default=0.5,
help="CTC weight in joint decoding",
)
group.add_argument("--lm_weight", type=float, default=1.0, help="RNNLM weight")
group.add_argument("--ngram_weight", type=float, default=0.9, help="ngram weight")
group.add_argument("--streaming", type=str2bool, default=False)
group.add_argument(
"--frontend_conf",
default=None,
help="",
)
group.add_argument("--audio_lists", 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 = parser.add_argument_group("Text converter related")
group.add_argument(
"--token_type",
type=str_or_none,
default=None,
choices=["char", "bpe", None],
help="The token type for ASR model. "
"If not given, refers from the training args",
)
group.add_argument(
"--bpemodel",
type=str_or_none,
default=None,
help="The model path of sentencepiece. "
"If not given, refers from the training args",
)
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()