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@@ -19,7 +19,6 @@ from typing import List
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import numpy as np
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import torch
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-import torchaudio
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from typeguard import check_argument_types
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from funasr.fileio.datadir_writer import DatadirWriter
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@@ -40,11 +39,12 @@ from funasr.utils.types import str2bool
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from funasr.utils.types import str2triple_str
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from funasr.utils.types import str_or_none
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from funasr.utils import asr_utils, wav_utils, postprocess_utils
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-from funasr.models.frontend.wav_frontend import WavFrontend
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-from funasr.models.e2e_asr_paraformer import BiCifParaformer, ContextualParaformer
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+from funasr.models.frontend.wav_frontend import WavFrontend, WavFrontendOnline
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from funasr.export.models.e2e_asr_paraformer import Paraformer as Paraformer_export
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+
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np.set_printoptions(threshold=np.inf)
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+
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class Speech2Text:
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"""Speech2Text class
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@@ -89,7 +89,7 @@ class Speech2Text:
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)
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frontend = None
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if asr_train_args.frontend is not None and asr_train_args.frontend_conf is not None:
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- frontend = WavFrontend(cmvn_file=cmvn_file, **asr_train_args.frontend_conf)
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+ frontend = WavFrontendOnline(cmvn_file=cmvn_file, **asr_train_args.frontend_conf)
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logging.info("asr_model: {}".format(asr_model))
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logging.info("asr_train_args: {}".format(asr_train_args))
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@@ -189,8 +189,7 @@ class Speech2Text:
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@torch.no_grad()
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def __call__(
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- self, cache: dict, speech: Union[torch.Tensor, np.ndarray], speech_lengths: Union[torch.Tensor, np.ndarray] = None,
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- begin_time: int = 0, end_time: int = None,
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+ self, cache: dict, speech: Union[torch.Tensor], speech_lengths: Union[torch.Tensor] = None
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):
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"""Inference
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@@ -201,38 +200,57 @@ class Speech2Text:
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"""
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assert check_argument_types()
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-
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- # Input as audio signal
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- if isinstance(speech, np.ndarray):
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- speech = torch.tensor(speech)
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- if self.frontend is not None:
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- feats, feats_len = self.frontend.forward(speech, speech_lengths)
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- feats = to_device(feats, device=self.device)
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- feats_len = feats_len.int()
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- self.asr_model.frontend = None
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+ results = []
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+ cache_en = cache["encoder"]
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+ if speech.shape[1] < 16 * 60 and cache["is_final"]:
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+ cache["last_chunk"] = True
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+ feats = cache["feats"]
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+ feats_len = torch.tensor([feats.shape[1]])
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else:
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- feats = speech
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- feats_len = speech_lengths
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- lfr_factor = max(1, (feats.size()[-1] // 80) - 1)
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- feats_len = cache["encoder"]["stride"] + cache["encoder"]["pad_left"] + cache["encoder"]["pad_right"]
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- feats = feats[:,cache["encoder"]["start_idx"]:cache["encoder"]["start_idx"]+feats_len,:]
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- feats_len = torch.tensor([feats_len])
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- batch = {"speech": feats, "speech_lengths": feats_len, "cache": cache}
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-
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- # a. To device
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- batch = to_device(batch, device=self.device)
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+ if self.frontend is not None:
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+ feats, feats_len = self.frontend.forward(speech, speech_lengths, cache_en["is_final"])
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+ feats = to_device(feats, device=self.device)
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+ feats_len = feats_len.int()
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+ self.asr_model.frontend = None
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+ else:
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+ feats = speech
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+ feats_len = speech_lengths
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+
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+ if feats.shape[1] != 0:
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+ if cache_en["is_final"]:
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+ if feats.shape[1] + cache_en["chunk_size"][2] < cache_en["chunk_size"][1]:
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+ cache_en["last_chunk"] = True
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+ else:
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+ # first chunk
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+ feats_chunk1 = feats[:, :cache_en["chunk_size"][1], :]
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+ feats_len = torch.tensor([feats_chunk1.shape[1]])
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+ results_chunk1 = self.infer(feats_chunk1, feats_len, cache)
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+
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+ # last chunk
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+ cache_en["last_chunk"] = True
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+ feats_chunk2 = feats[:, -(feats.shape[1] + cache_en["chunk_size"][2] - cache_en["chunk_size"][1]):, :]
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+ feats_len = torch.tensor([feats_chunk2.shape[1]])
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+ results_chunk2 = self.infer(feats_chunk2, feats_len, cache)
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+
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+ return results_chunk1 + results_chunk2
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+
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+ results = self.infer(feats, feats_len, cache)
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+ return results
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+
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+ @torch.no_grad()
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+ def infer(self, feats: Union[torch.Tensor], feats_len: Union[torch.Tensor], cache: List = None):
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+ batch = {"speech": feats, "speech_lengths": feats_len}
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+ batch = to_device(batch, device=self.device)
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# b. Forward Encoder
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- enc, enc_len = self.asr_model.encode_chunk(feats, feats_len, cache)
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+ enc, enc_len = self.asr_model.encode_chunk(feats, feats_len, cache=cache)
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if isinstance(enc, tuple):
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enc = enc[0]
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# assert len(enc) == 1, len(enc)
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enc_len_batch_total = torch.sum(enc_len).item() * self.encoder_downsampling_factor
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predictor_outs = self.asr_model.calc_predictor_chunk(enc, cache)
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- pre_acoustic_embeds, pre_token_length, alphas, pre_peak_index = predictor_outs[0], predictor_outs[1], \
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- predictor_outs[2], predictor_outs[3]
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- pre_token_length = pre_token_length.floor().long()
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+ pre_acoustic_embeds, pre_token_length= predictor_outs[0], predictor_outs[1]
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if torch.max(pre_token_length) < 1:
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return []
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decoder_outs = self.asr_model.cal_decoder_with_predictor_chunk(enc, pre_acoustic_embeds, cache)
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@@ -279,166 +297,12 @@ class Speech2Text:
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text = self.tokenizer.tokens2text(token)
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else:
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text = None
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-
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- results.append((text, token, token_int, hyp, enc_len_batch_total, lfr_factor))
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+ results.append(text)
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# assert check_return_type(results)
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return results
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-class Speech2TextExport:
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- """Speech2TextExport class
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-
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- """
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-
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- def __init__(
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- self,
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- asr_train_config: Union[Path, str] = None,
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- asr_model_file: Union[Path, str] = None,
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- cmvn_file: Union[Path, str] = None,
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- lm_train_config: Union[Path, str] = None,
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- lm_file: Union[Path, str] = None,
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- token_type: str = None,
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- bpemodel: str = None,
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- device: str = "cpu",
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- maxlenratio: float = 0.0,
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- minlenratio: float = 0.0,
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- dtype: str = "float32",
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- beam_size: int = 20,
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- ctc_weight: float = 0.5,
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- lm_weight: float = 1.0,
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- ngram_weight: float = 0.9,
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- penalty: float = 0.0,
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- nbest: int = 1,
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- frontend_conf: dict = None,
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- hotword_list_or_file: str = None,
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- **kwargs,
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- ):
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-
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- # 1. Build ASR model
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- asr_model, asr_train_args = ASRTask.build_model_from_file(
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- asr_train_config, asr_model_file, cmvn_file, device
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- )
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- frontend = None
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- if asr_train_args.frontend is not None and asr_train_args.frontend_conf is not None:
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- frontend = WavFrontend(cmvn_file=cmvn_file, **asr_train_args.frontend_conf)
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-
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- logging.info("asr_model: {}".format(asr_model))
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- logging.info("asr_train_args: {}".format(asr_train_args))
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- asr_model.to(dtype=getattr(torch, dtype)).eval()
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-
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- token_list = asr_model.token_list
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-
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- logging.info(f"Decoding device={device}, dtype={dtype}")
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-
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- # 5. [Optional] Build Text converter: e.g. bpe-sym -> Text
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- if token_type is None:
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- token_type = asr_train_args.token_type
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- if bpemodel is None:
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- bpemodel = asr_train_args.bpemodel
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-
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- if token_type is None:
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- tokenizer = None
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- elif token_type == "bpe":
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- if bpemodel is not None:
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- tokenizer = build_tokenizer(token_type=token_type, bpemodel=bpemodel)
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- else:
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- tokenizer = None
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- else:
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- tokenizer = build_tokenizer(token_type=token_type)
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- converter = TokenIDConverter(token_list=token_list)
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- logging.info(f"Text tokenizer: {tokenizer}")
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-
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- # self.asr_model = asr_model
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- self.asr_train_args = asr_train_args
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- self.converter = converter
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- self.tokenizer = tokenizer
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-
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- self.device = device
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- self.dtype = dtype
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- self.nbest = nbest
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- self.frontend = frontend
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-
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- model = Paraformer_export(asr_model, onnx=False)
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- self.asr_model = model
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-
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- @torch.no_grad()
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- def __call__(
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- self, speech: Union[torch.Tensor, np.ndarray], speech_lengths: Union[torch.Tensor, np.ndarray] = None
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- ):
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- """Inference
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-
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- Args:
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- speech: Input speech data
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- Returns:
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- text, token, token_int, hyp
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-
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- """
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- assert check_argument_types()
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-
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- # Input as audio signal
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- if isinstance(speech, np.ndarray):
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- speech = torch.tensor(speech)
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-
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- if self.frontend is not None:
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- feats, feats_len = self.frontend.forward(speech, speech_lengths)
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- feats = to_device(feats, device=self.device)
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- feats_len = feats_len.int()
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- self.asr_model.frontend = None
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- else:
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- feats = speech
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- feats_len = speech_lengths
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-
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- enc_len_batch_total = feats_len.sum()
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- lfr_factor = max(1, (feats.size()[-1] // 80) - 1)
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- batch = {"speech": feats, "speech_lengths": feats_len}
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-
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- # a. To device
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- batch = to_device(batch, device=self.device)
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-
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- decoder_outs = self.asr_model(**batch)
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- decoder_out, ys_pad_lens = decoder_outs[0], decoder_outs[1]
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-
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- results = []
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- b, n, d = decoder_out.size()
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- for i in range(b):
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- am_scores = decoder_out[i, :ys_pad_lens[i], :]
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-
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- yseq = am_scores.argmax(dim=-1)
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- score = am_scores.max(dim=-1)[0]
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- score = torch.sum(score, dim=-1)
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- # pad with mask tokens to ensure compatibility with sos/eos tokens
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- yseq = torch.tensor(
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- yseq.tolist(), device=yseq.device
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- )
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- nbest_hyps = [Hypothesis(yseq=yseq, score=score)]
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-
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- for hyp in nbest_hyps:
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- assert isinstance(hyp, (Hypothesis)), type(hyp)
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-
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- # remove sos/eos and get results
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- last_pos = -1
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- if isinstance(hyp.yseq, list):
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- token_int = hyp.yseq[1:last_pos]
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- else:
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- token_int = hyp.yseq[1:last_pos].tolist()
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-
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- # remove blank symbol id, which is assumed to be 0
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- token_int = list(filter(lambda x: x != 0 and x != 2, token_int))
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-
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- # Change integer-ids to tokens
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- token = self.converter.ids2tokens(token_int)
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-
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- if self.tokenizer is not None:
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- text = self.tokenizer.tokens2text(token)
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- else:
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- text = None
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-
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- results.append((text, token, token_int, hyp, enc_len_batch_total, lfr_factor))
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-
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- return results
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-
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-
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def inference(
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maxlenratio: float,
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minlenratio: float,
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@@ -536,8 +400,6 @@ def inference_modelscope(
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**kwargs,
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):
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assert check_argument_types()
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- ncpu = kwargs.get("ncpu", 1)
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- torch.set_num_threads(ncpu)
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if word_lm_train_config is not None:
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raise NotImplementedError("Word LM is not implemented")
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@@ -580,11 +442,9 @@ def inference_modelscope(
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penalty=penalty,
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nbest=nbest,
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)
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- if export_mode:
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- speech2text = Speech2TextExport(**speech2text_kwargs)
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- else:
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- speech2text = Speech2Text(**speech2text_kwargs)
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-
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+
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+ speech2text = Speech2Text(**speech2text_kwargs)
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+
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def _load_bytes(input):
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middle_data = np.frombuffer(input, dtype=np.int16)
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middle_data = np.asarray(middle_data)
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@@ -599,7 +459,33 @@ def inference_modelscope(
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offset = i.min + abs_max
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array = np.frombuffer((middle_data.astype(dtype) - offset) / abs_max, dtype=np.float32)
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return array
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-
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+
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+ def _prepare_cache(cache: dict = {}, chunk_size=[5,10,5], batch_size=1):
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+ if len(cache) > 0:
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+ return cache
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+
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+ cache_en = {"start_idx": 0, "cif_hidden": torch.zeros((batch_size, 1, 320)),
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+ "cif_alphas": torch.zeros((batch_size, 1)), "chunk_size": chunk_size, "last_chunk": False,
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+ "feats": torch.zeros((batch_size, chunk_size[0] + chunk_size[2], 560))}
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+ cache["encoder"] = cache_en
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+
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+ cache_de = {"decode_fsmn": None}
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+ cache["decoder"] = cache_de
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+
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+ return cache
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+
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+ def _cache_reset(cache: dict = {}, chunk_size=[5,10,5], batch_size=1):
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+ if len(cache) > 0:
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+ cache_en = {"start_idx": 0, "cif_hidden": torch.zeros((batch_size, 1, 320)),
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+ "cif_alphas": torch.zeros((batch_size, 1)), "chunk_size": chunk_size, "last_chunk": False,
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+ "feats": torch.zeros((batch_size, chunk_size[0] + chunk_size[2], 560))}
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+ cache["encoder"] = cache_en
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+
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+ cache_de = {"decode_fsmn": None}
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+ cache["decoder"] = cache_de
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+
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+ return cache
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+
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def _forward(
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data_path_and_name_and_type,
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raw_inputs: Union[np.ndarray, torch.Tensor] = None,
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@@ -610,123 +496,35 @@ def inference_modelscope(
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):
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# 3. Build data-iterator
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+ if data_path_and_name_and_type is not None and data_path_and_name_and_type[2] == "bytes":
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+ raw_inputs = _load_bytes(data_path_and_name_and_type[0])
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+ raw_inputs = torch.tensor(raw_inputs)
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+ if data_path_and_name_and_type is None and raw_inputs is not None:
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+ if isinstance(raw_inputs, np.ndarray):
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+ raw_inputs = torch.tensor(raw_inputs)
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is_final = False
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cache = {}
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+ chunk_size = [5, 10, 5]
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if param_dict is not None and "cache" in param_dict:
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cache = param_dict["cache"]
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if param_dict is not None and "is_final" in param_dict:
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is_final = param_dict["is_final"]
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+ if param_dict is not None and "chunk_size" in param_dict:
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+ chunk_size = param_dict["chunk_size"]
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- if data_path_and_name_and_type is not None and data_path_and_name_and_type[2] == "bytes":
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- raw_inputs = _load_bytes(data_path_and_name_and_type[0])
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- raw_inputs = torch.tensor(raw_inputs)
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- if data_path_and_name_and_type is not None and data_path_and_name_and_type[2] == "sound":
|
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- raw_inputs = torchaudio.load(data_path_and_name_and_type[0])[0][0]
|
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- is_final = True
|
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- if data_path_and_name_and_type is None and raw_inputs is not None:
|
|
|
- if isinstance(raw_inputs, np.ndarray):
|
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|
- raw_inputs = torch.tensor(raw_inputs)
|
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# 7 .Start for-loop
|
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|
# FIXME(kamo): The output format should be discussed about
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+ raw_inputs = torch.unsqueeze(raw_inputs, axis=0)
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+ input_lens = torch.tensor([raw_inputs.shape[1]])
|
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asr_result_list = []
|
|
|
- results = []
|
|
|
- asr_result = ""
|
|
|
- wait = True
|
|
|
- if len(cache) == 0:
|
|
|
- cache["encoder"] = {"start_idx": 0, "pad_left": 0, "stride": 10, "pad_right": 5, "cif_hidden": None, "cif_alphas": None, "is_final": is_final, "left": 0, "right": 0}
|
|
|
- cache_de = {"decode_fsmn": None}
|
|
|
- cache["decoder"] = cache_de
|
|
|
- cache["first_chunk"] = True
|
|
|
- cache["speech"] = []
|
|
|
- cache["accum_speech"] = 0
|
|
|
|
|
|
- if raw_inputs is not None:
|
|
|
- if len(cache["speech"]) == 0:
|
|
|
- cache["speech"] = raw_inputs
|
|
|
- else:
|
|
|
- cache["speech"] = torch.cat([cache["speech"], raw_inputs], dim=0)
|
|
|
- cache["accum_speech"] += len(raw_inputs)
|
|
|
- while cache["accum_speech"] >= 960:
|
|
|
- if cache["first_chunk"]:
|
|
|
- if cache["accum_speech"] >= 14400:
|
|
|
- speech = torch.unsqueeze(cache["speech"], axis=0)
|
|
|
- speech_length = torch.tensor([len(cache["speech"])])
|
|
|
- cache["encoder"]["pad_left"] = 5
|
|
|
- cache["encoder"]["pad_right"] = 5
|
|
|
- cache["encoder"]["stride"] = 10
|
|
|
- cache["encoder"]["left"] = 5
|
|
|
- cache["encoder"]["right"] = 0
|
|
|
- results = speech2text(cache, speech, speech_length)
|
|
|
- cache["accum_speech"] -= 4800
|
|
|
- cache["first_chunk"] = False
|
|
|
- cache["encoder"]["start_idx"] = -5
|
|
|
- cache["encoder"]["is_final"] = False
|
|
|
- wait = False
|
|
|
- else:
|
|
|
- if is_final:
|
|
|
- cache["encoder"]["stride"] = len(cache["speech"]) // 960
|
|
|
- cache["encoder"]["pad_left"] = 0
|
|
|
- cache["encoder"]["pad_right"] = 0
|
|
|
- speech = torch.unsqueeze(cache["speech"], axis=0)
|
|
|
- speech_length = torch.tensor([len(cache["speech"])])
|
|
|
- results = speech2text(cache, speech, speech_length)
|
|
|
- cache["accum_speech"] = 0
|
|
|
- wait = False
|
|
|
- else:
|
|
|
- break
|
|
|
- else:
|
|
|
- if cache["accum_speech"] >= 19200:
|
|
|
- cache["encoder"]["start_idx"] += 10
|
|
|
- cache["encoder"]["stride"] = 10
|
|
|
- cache["encoder"]["pad_left"] = 5
|
|
|
- cache["encoder"]["pad_right"] = 5
|
|
|
- cache["encoder"]["left"] = 0
|
|
|
- cache["encoder"]["right"] = 0
|
|
|
- speech = torch.unsqueeze(cache["speech"], axis=0)
|
|
|
- speech_length = torch.tensor([len(cache["speech"])])
|
|
|
- results = speech2text(cache, speech, speech_length)
|
|
|
- cache["accum_speech"] -= 9600
|
|
|
- wait = False
|
|
|
- else:
|
|
|
- if is_final:
|
|
|
- cache["encoder"]["is_final"] = True
|
|
|
- if cache["accum_speech"] >= 14400:
|
|
|
- cache["encoder"]["start_idx"] += 10
|
|
|
- cache["encoder"]["stride"] = 10
|
|
|
- cache["encoder"]["pad_left"] = 5
|
|
|
- cache["encoder"]["pad_right"] = 5
|
|
|
- cache["encoder"]["left"] = 0
|
|
|
- cache["encoder"]["right"] = cache["accum_speech"] // 960 - 15
|
|
|
- speech = torch.unsqueeze(cache["speech"], axis=0)
|
|
|
- speech_length = torch.tensor([len(cache["speech"])])
|
|
|
- results = speech2text(cache, speech, speech_length)
|
|
|
- cache["accum_speech"] -= 9600
|
|
|
- wait = False
|
|
|
- else:
|
|
|
- cache["encoder"]["start_idx"] += 10
|
|
|
- cache["encoder"]["stride"] = cache["accum_speech"] // 960 - 5
|
|
|
- cache["encoder"]["pad_left"] = 5
|
|
|
- cache["encoder"]["pad_right"] = 0
|
|
|
- cache["encoder"]["left"] = 0
|
|
|
- cache["encoder"]["right"] = 0
|
|
|
- speech = torch.unsqueeze(cache["speech"], axis=0)
|
|
|
- speech_length = torch.tensor([len(cache["speech"])])
|
|
|
- results = speech2text(cache, speech, speech_length)
|
|
|
- cache["accum_speech"] = 0
|
|
|
- wait = False
|
|
|
- else:
|
|
|
- break
|
|
|
-
|
|
|
- if len(results) >= 1:
|
|
|
- asr_result += results[0][0]
|
|
|
- if asr_result == "":
|
|
|
- asr_result = "sil"
|
|
|
- if wait:
|
|
|
- asr_result = "waiting_for_more_voice"
|
|
|
- item = {'key': "utt", 'value': asr_result}
|
|
|
- asr_result_list.append(item)
|
|
|
- else:
|
|
|
- return []
|
|
|
+ cache = _prepare_cache(cache, chunk_size=chunk_size, batch_size=1)
|
|
|
+ cache["encoder"]["is_final"] = is_final
|
|
|
+ asr_result = speech2text(cache, raw_inputs, input_lens)
|
|
|
+ item = {'key': "utt", 'value': asr_result}
|
|
|
+ asr_result_list.append(item)
|
|
|
+ if is_final:
|
|
|
+ cache = _cache_reset(cache, chunk_size=chunk_size, batch_size=1)
|
|
|
return asr_result_list
|
|
|
|
|
|
return _forward
|
|
|
@@ -921,4 +719,3 @@ if __name__ == "__main__":
|
|
|
# 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)
|
|
|
|
|
|
-
|