From f2e7ea83c9d8b580d09eb31acf6c2fa60e683e3b Mon Sep 17 00:00:00 2001
From: zhaomingwork <zhaomingwork@qq.com>
Date: 星期六, 13 五月 2023 08:02:33 +0800
Subject: [PATCH] fix bug for cpp msg not return right name
---
funasr/bin/asr_inference_paraformer.py | 575 ++++++++++++++++++++++++++++++++------------------------
1 files changed, 328 insertions(+), 247 deletions(-)
diff --git a/funasr/bin/asr_inference_paraformer.py b/funasr/bin/asr_inference_paraformer.py
index 487f750..ecdb62a 100644
--- a/funasr/bin/asr_inference_paraformer.py
+++ b/funasr/bin/asr_inference_paraformer.py
@@ -41,15 +41,22 @@
from funasr.utils import asr_utils, wav_utils, postprocess_utils
from funasr.models.frontend.wav_frontend import WavFrontend
from funasr.models.e2e_asr_paraformer import BiCifParaformer, ContextualParaformer
+from funasr.models.e2e_asr_contextual_paraformer import NeatContextualParaformer
from funasr.export.models.e2e_asr_paraformer import Paraformer as Paraformer_export
-
+from funasr.utils.timestamp_tools import ts_prediction_lfr6_standard
+from funasr.bin.tp_inference import SpeechText2Timestamp
+from funasr.bin.vad_inference import Speech2VadSegment
+from funasr.bin.punctuation_infer import Text2Punc
+from funasr.utils.vad_utils import slice_padding_fbank
+from funasr.tasks.vad import VADTask
+from funasr.utils.timestamp_tools import time_stamp_sentence, ts_prediction_lfr6_standard
class Speech2Text:
"""Speech2Text class
Examples:
>>> import soundfile
- >>> speech2text = Speech2Text("asr_config.yml", "asr.pth")
+ >>> speech2text = Speech2Text("asr_config.yml", "asr.pb")
>>> audio, rate = soundfile.read("speech.wav")
>>> speech2text(audio)
[(text, token, token_int, hypothesis object), ...]
@@ -190,7 +197,8 @@
@torch.no_grad()
def __call__(
- self, speech: Union[torch.Tensor, np.ndarray], speech_lengths: Union[torch.Tensor, np.ndarray] = None
+ self, speech: Union[torch.Tensor, np.ndarray], speech_lengths: Union[torch.Tensor, np.ndarray] = None,
+ begin_time: int = 0, end_time: int = None,
):
"""Inference
@@ -233,7 +241,7 @@
pre_token_length = pre_token_length.round().long()
if torch.max(pre_token_length) < 1:
return []
- if not isinstance(self.asr_model, ContextualParaformer):
+ if not isinstance(self.asr_model, ContextualParaformer) and not isinstance(self.asr_model, NeatContextualParaformer):
if self.hotword_list:
logging.warning("Hotword is given but asr model is not a ContextualParaformer.")
decoder_outs = self.asr_model.cal_decoder_with_predictor(enc, enc_len, pre_acoustic_embeds, pre_token_length)
@@ -241,6 +249,10 @@
else:
decoder_outs = self.asr_model.cal_decoder_with_predictor(enc, enc_len, pre_acoustic_embeds, pre_token_length, hw_list=self.hotword_list)
decoder_out, ys_pad_lens = decoder_outs[0], decoder_outs[1]
+
+ if isinstance(self.asr_model, BiCifParaformer):
+ _, _, us_alphas, us_peaks = self.asr_model.calc_predictor_timestamp(enc, enc_len,
+ pre_token_length) # test no bias cif2
results = []
b, n, d = decoder_out.size()
@@ -283,8 +295,14 @@
text = self.tokenizer.tokens2text(token)
else:
text = None
+ timestamp = []
+ if isinstance(self.asr_model, BiCifParaformer):
+ _, timestamp = ts_prediction_lfr6_standard(us_alphas[i][:enc_len[i]*3],
+ us_peaks[i][:enc_len[i]*3],
+ copy.copy(token),
+ vad_offset=begin_time)
+ results.append((text, token, token_int, hyp, timestamp, enc_len_batch_total, lfr_factor))
- results.append((text, token, token_int, hyp, enc_len_batch_total, lfr_factor))
# assert check_return_type(results)
return results
@@ -343,225 +361,6 @@
hotword_list = None
return hotword_list
-class Speech2TextExport:
- """Speech2TextExport class
-
- """
-
- def __init__(
- self,
- asr_train_config: Union[Path, str] = None,
- asr_model_file: Union[Path, str] = None,
- cmvn_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,
- hotword_list_or_file: str = None,
- **kwargs,
- ):
-
- # 1. Build ASR model
- asr_model, asr_train_args = ASRTask.build_model_from_file(
- asr_train_config, asr_model_file, cmvn_file, device
- )
- frontend = None
- if asr_train_args.frontend is not None and asr_train_args.frontend_conf is not None:
- frontend = WavFrontend(cmvn_file=cmvn_file, **asr_train_args.frontend_conf)
-
- logging.info("asr_model: {}".format(asr_model))
- logging.info("asr_train_args: {}".format(asr_train_args))
- asr_model.to(dtype=getattr(torch, dtype)).eval()
-
- token_list = asr_model.token_list
-
-
-
- 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
-
- self.device = device
- self.dtype = dtype
- self.nbest = nbest
- self.frontend = frontend
-
- model = Paraformer_export(asr_model, onnx=False)
- self.asr_model = model
-
- @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)
-
- 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.asr_model.frontend = None
- else:
- feats = speech
- feats_len = speech_lengths
-
- enc_len_batch_total = feats_len.sum()
- 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)
-
- decoder_outs = self.asr_model(**batch)
- decoder_out, ys_pad_lens = decoder_outs[0], decoder_outs[1]
-
- results = []
- b, n, d = decoder_out.size()
- for i in range(b):
- am_scores = decoder_out[i, :ys_pad_lens[i], :]
-
- 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(
- yseq.tolist(), 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 and x != 2, 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))
-
- 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],
- cmvn_file: Optional[str] = None,
- raw_inputs: Union[np.ndarray, torch.Tensor] = 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,
-
- **kwargs,
-):
- inference_pipeline = inference_modelscope(
- maxlenratio=maxlenratio,
- minlenratio=minlenratio,
- batch_size=batch_size,
- beam_size=beam_size,
- ngpu=ngpu,
- ctc_weight=ctc_weight,
- lm_weight=lm_weight,
- penalty=penalty,
- log_level=log_level,
- asr_train_config=asr_train_config,
- asr_model_file=asr_model_file,
- cmvn_file=cmvn_file,
- raw_inputs=raw_inputs,
- lm_train_config=lm_train_config,
- lm_file=lm_file,
- token_type=token_type,
- key_file=key_file,
- word_lm_train_config=word_lm_train_config,
- bpemodel=bpemodel,
- allow_variable_data_keys=allow_variable_data_keys,
- streaming=streaming,
- output_dir=output_dir,
- dtype=dtype,
- seed=seed,
- ngram_weight=ngram_weight,
- nbest=nbest,
- num_workers=num_workers,
-
- **kwargs,
- )
- return inference_pipeline(data_path_and_name_and_type, raw_inputs)
def inference_modelscope(
@@ -591,11 +390,15 @@
nbest: int = 1,
num_workers: int = 1,
output_dir: Optional[str] = None,
+ timestamp_infer_config: Union[Path, str] = None,
+ timestamp_model_file: Union[Path, str] = None,
param_dict: dict = None,
**kwargs,
):
assert check_argument_types()
-
+ ncpu = kwargs.get("ncpu", 1)
+ torch.set_num_threads(ncpu)
+
if word_lm_train_config is not None:
raise NotImplementedError("Word LM is not implemented")
if ngpu > 1:
@@ -612,7 +415,9 @@
export_mode = param_dict.get("export_mode", False)
else:
hotword_list_or_file = None
-
+
+ if kwargs.get("device", None) == "cpu":
+ ngpu = 0
if ngpu >= 1 and torch.cuda.is_available():
device = "cuda"
else:
@@ -643,10 +448,17 @@
nbest=nbest,
hotword_list_or_file=hotword_list_or_file,
)
- if export_mode:
- speech2text = Speech2TextExport(**speech2text_kwargs)
+
+ speech2text = Speech2Text(**speech2text_kwargs)
+
+ if timestamp_model_file is not None:
+ speechtext2timestamp = SpeechText2Timestamp(
+ timestamp_cmvn_file=cmvn_file,
+ timestamp_model_file=timestamp_model_file,
+ timestamp_infer_config=timestamp_infer_config,
+ )
else:
- speech2text = Speech2Text(**speech2text_kwargs)
+ speechtext2timestamp = None
def _forward(
data_path_and_name_and_type,
@@ -660,11 +472,9 @@
hotword_list_or_file = None
if param_dict is not None:
hotword_list_or_file = param_dict.get('hotword')
-
- if 'hotword' in kwargs:
+ if 'hotword' in kwargs and kwargs['hotword'] is not None:
hotword_list_or_file = kwargs['hotword']
-
- if speech2text.hotword_list is None:
+ if hotword_list_or_file is not None or 'hotword' in kwargs:
speech2text.hotword_list = speech2text.generate_hotwords_list(hotword_list_or_file)
# 3. Build data-iterator
@@ -684,6 +494,11 @@
allow_variable_data_keys=allow_variable_data_keys,
inference=True,
)
+
+ if param_dict is not None:
+ use_timestamp = param_dict.get('use_timestamp', True)
+ else:
+ use_timestamp = True
forward_time_total = 0.0
length_total = 0.0
@@ -726,7 +541,19 @@
result = [results[batch_id][:-2]]
key = keys[batch_id]
- for n, (text, token, token_int, hyp) in zip(range(1, nbest + 1), result):
+ for n, result in zip(range(1, nbest + 1), result):
+ text, token, token_int, hyp = result[0], result[1], result[2], result[3]
+ timestamp = result[4] if len(result[4]) > 0 else None
+ # conduct timestamp prediction here
+ # timestamp inference requires token length
+ # thus following inference cannot be conducted in batch
+ if timestamp is None and speechtext2timestamp:
+ ts_batch = {}
+ ts_batch['speech'] = batch['speech'][batch_id].unsqueeze(0)
+ ts_batch['speech_lengths'] = torch.tensor([batch['speech_lengths'][batch_id]])
+ ts_batch['text_lengths'] = torch.tensor([len(token)])
+ us_alphas, us_peaks = speechtext2timestamp(**ts_batch)
+ ts_str, timestamp = ts_prediction_lfr6_standard(us_alphas[0], us_peaks[0], token, force_time_shift=-3.0)
# Create a directory: outdir/{n}best_recog
if writer is not None:
ibest_writer = writer[f"{n}best_recog"]
@@ -738,13 +565,25 @@
ibest_writer["rtf"][key] = rtf_cur
if text is not None:
- text_postprocessed, _ = postprocess_utils.sentence_postprocess(token)
+ if use_timestamp and timestamp is not None:
+ postprocessed_result = postprocess_utils.sentence_postprocess(token, timestamp)
+ else:
+ postprocessed_result = postprocess_utils.sentence_postprocess(token)
+ timestamp_postprocessed = ""
+ if len(postprocessed_result) == 3:
+ text_postprocessed, timestamp_postprocessed, word_lists = postprocessed_result[0], \
+ postprocessed_result[1], \
+ postprocessed_result[2]
+ else:
+ text_postprocessed, word_lists = postprocessed_result[0], postprocessed_result[1]
item = {'key': key, 'value': text_postprocessed}
+ if timestamp_postprocessed != "":
+ item['timestamp'] = timestamp_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_postprocessed
+ ibest_writer["text"][key] = " ".join(word_lists)
logging.info("decoding, utt: {}, predictions: {}".format(key, text))
rtf_avg = "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))
@@ -753,6 +592,257 @@
ibest_writer["rtf"]["rtf_avf"] = rtf_avg
return asr_result_list
+ return _forward
+
+
+def inference_modelscope_vad_punc(
+ 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],
+ cmvn_file: Optional[str] = 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,
+ output_dir: Optional[str] = None,
+ dtype: str = "float32",
+ seed: int = 0,
+ ngram_weight: float = 0.9,
+ nbest: int = 1,
+ num_workers: int = 1,
+ vad_infer_config: Optional[str] = None,
+ vad_model_file: Optional[str] = None,
+ vad_cmvn_file: Optional[str] = None,
+ time_stamp_writer: bool = True,
+ punc_infer_config: Optional[str] = None,
+ punc_model_file: Optional[str] = None,
+ outputs_dict: Optional[bool] = True,
+ param_dict: dict = None,
+ **kwargs,
+):
+ assert check_argument_types()
+ ncpu = kwargs.get("ncpu", 1)
+ torch.set_num_threads(ncpu)
+
+ 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 param_dict is not None:
+ hotword_list_or_file = param_dict.get('hotword')
+ else:
+ hotword_list_or_file = None
+
+ 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
+ speech2vadsegment_kwargs = dict(
+ vad_infer_config=vad_infer_config,
+ vad_model_file=vad_model_file,
+ vad_cmvn_file=vad_cmvn_file,
+ device=device,
+ dtype=dtype,
+ )
+ # logging.info("speech2vadsegment_kwargs: {}".format(speech2vadsegment_kwargs))
+ speech2vadsegment = Speech2VadSegment(**speech2vadsegment_kwargs)
+
+ # 3. Build speech2text
+ speech2text_kwargs = dict(
+ asr_train_config=asr_train_config,
+ asr_model_file=asr_model_file,
+ cmvn_file=cmvn_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,
+ hotword_list_or_file=hotword_list_or_file,
+ )
+ speech2text = Speech2Text(**speech2text_kwargs)
+ text2punc = None
+ if punc_model_file is not None:
+ text2punc = Text2Punc(punc_infer_config, punc_model_file, device=device, dtype=dtype)
+
+ if output_dir is not None:
+ writer = DatadirWriter(output_dir)
+ ibest_writer = writer[f"1best_recog"]
+ ibest_writer["token_list"][""] = " ".join(speech2text.asr_train_args.token_list)
+
+ 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,
+ ):
+
+ hotword_list_or_file = None
+ if param_dict is not None:
+ hotword_list_or_file = param_dict.get('hotword')
+
+ if 'hotword' in kwargs:
+ hotword_list_or_file = kwargs['hotword']
+
+ if speech2text.hotword_list is None:
+ speech2text.hotword_list = speech2text.generate_hotwords_list(hotword_list_or_file)
+
+ # 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,
+ fs=fs,
+ batch_size=1,
+ key_file=key_file,
+ num_workers=num_workers,
+ preprocess_fn=VADTask.build_preprocess_fn(speech2vadsegment.vad_infer_args, False),
+ collate_fn=VADTask.build_collate_fn(speech2vadsegment.vad_infer_args, False),
+ allow_variable_data_keys=allow_variable_data_keys,
+ inference=True,
+ )
+
+ if param_dict is not None:
+ use_timestamp = param_dict.get('use_timestamp', True)
+ else:
+ use_timestamp = True
+
+ finish_count = 0
+ file_count = 1
+ lfr_factor = 6
+ # 7 .Start for-loop
+ asr_result_list = []
+ 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)
+ ibest_writer = writer[f"1best_recog"]
+
+ 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}"
+
+ vad_results = speech2vadsegment(**batch)
+ _, vadsegments = vad_results[0], vad_results[1][0]
+
+ speech, speech_lengths = batch["speech"], batch["speech_lengths"]
+
+ n = len(vadsegments)
+ data_with_index = [(vadsegments[i], i) for i in range(n)]
+ sorted_data = sorted(data_with_index, key=lambda x: x[0][1] - x[0][0])
+ results_sorted = []
+ for j, beg_idx in enumerate(range(0, n, batch_size)):
+ end_idx = min(n, beg_idx + batch_size)
+ speech_j, speech_lengths_j = slice_padding_fbank(speech, speech_lengths, sorted_data[beg_idx:end_idx])
+
+ batch = {"speech": speech_j, "speech_lengths": speech_lengths_j}
+ batch = to_device(batch, device=device)
+ results = speech2text(**batch)
+
+ if len(results) < 1:
+ results = [["", [], [], [], [], [], []]]
+ results_sorted.extend(results)
+ restored_data = [0] * n
+ for j in range(n):
+ index = sorted_data[j][1]
+ restored_data[index] = results_sorted[j]
+ result = ["", [], [], [], [], [], []]
+ for j in range(n):
+ result[0] += restored_data[j][0]
+ result[1] += restored_data[j][1]
+ result[2] += restored_data[j][2]
+ if len(restored_data[j][4]) > 0:
+ for t in restored_data[j][4]:
+ t[0] += vadsegments[j][0]
+ t[1] += vadsegments[j][0]
+ result[4] += restored_data[j][4]
+ # result = [result[k]+restored_data[j][k] for k in range(len(result[:-2]))]
+
+ key = keys[0]
+ # result = result_segments[0]
+ text, token, token_int = result[0], result[1], result[2]
+ time_stamp = result[4] if len(result[4]) > 0 else None
+
+ if use_timestamp and time_stamp is not None:
+ postprocessed_result = postprocess_utils.sentence_postprocess(token, time_stamp)
+ else:
+ postprocessed_result = postprocess_utils.sentence_postprocess(token)
+ text_postprocessed = ""
+ time_stamp_postprocessed = ""
+ text_postprocessed_punc = postprocessed_result
+ if len(postprocessed_result) == 3:
+ text_postprocessed, time_stamp_postprocessed, word_lists = postprocessed_result[0], \
+ postprocessed_result[1], \
+ postprocessed_result[2]
+ else:
+ text_postprocessed, word_lists = postprocessed_result[0], postprocessed_result[1]
+
+ text_postprocessed_punc = text_postprocessed
+ punc_id_list = []
+ if len(word_lists) > 0 and text2punc is not None:
+ text_postprocessed_punc, punc_id_list = text2punc(word_lists, 20)
+
+ item = {'key': key, 'value': text_postprocessed_punc}
+ if text_postprocessed != "":
+ item['text_postprocessed'] = text_postprocessed
+ if time_stamp_postprocessed != "":
+ item['time_stamp'] = time_stamp_postprocessed
+
+ item['sentences'] = time_stamp_sentence(punc_id_list, time_stamp_postprocessed, text_postprocessed)
+
+ asr_result_list.append(item)
+ finish_count += 1
+ # asr_utils.print_progress(finish_count / file_count)
+ if writer is not None:
+ # Write the result to each file
+ ibest_writer["token"][key] = " ".join(token)
+ ibest_writer["token_int"][key] = " ".join(map(str, token_int))
+ ibest_writer["vad"][key] = "{}".format(vadsegments)
+ ibest_writer["text"][key] = " ".join(word_lists)
+ ibest_writer["text_with_punc"][key] = text_postprocessed_punc
+ if time_stamp_postprocessed is not None:
+ ibest_writer["time_stamp"][key] = "{}".format(time_stamp_postprocessed)
+
+ logging.info("decoding, utt: {}, predictions: {}".format(key, text_postprocessed_punc))
+ return asr_result_list
+
return _forward
@@ -929,18 +1019,9 @@
kwargs = vars(args)
kwargs.pop("config", None)
kwargs['param_dict'] = param_dict
- inference(**kwargs)
+ inference_pipeline = inference_modelscope(**kwargs)
+ return inference_pipeline(kwargs["data_path_and_name_and_type"], param_dict=param_dict)
if __name__ == "__main__":
main()
-
- # from modelscope.pipelines import pipeline
- # from modelscope.utils.constant import Tasks
- #
- # inference_16k_pipline = pipeline(
- # task=Tasks.auto_speech_recognition,
- # model='damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch')
- #
- # rec_result = inference_16k_pipline(audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav')
- # print(rec_result)
--
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