游雁
2023-04-21 3cd3473bf7a3b41484baa86d9092248d78e7af39
funasr/bin/asr_inference_mfcca.py
@@ -41,8 +41,6 @@
from funasr.utils import asr_utils, wav_utils, postprocess_utils
import pdb
header_colors = '\033[95m'
end_colors = '\033[0m'
global_asr_language: str = 'zh-cn'
global_sample_rate: Union[int, Dict[Any, int]] = {
@@ -55,7 +53,7 @@
    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), ...]
@@ -194,8 +192,8 @@
        # Input as audio signal
        if isinstance(speech, np.ndarray):
            speech = torch.tensor(speech)
        if(speech.dim()==3):
            speech = torch.squeeze(speech, 2)
        #speech = speech.unsqueeze(0).to(getattr(torch, self.dtype))
        speech = speech.to(getattr(torch, self.dtype))
        # lenghts: (1,)
@@ -474,6 +472,8 @@
    **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 word_lm_train_config is not None:
@@ -534,6 +534,8 @@
            data_path_and_name_and_type,
            dtype=dtype,
            batch_size=batch_size,
            fs=fs,
            mc=True,
            key_file=key_file,
            num_workers=num_workers,
            preprocess_fn=ASRTask.build_preprocess_fn(speech2text.asr_train_args, False),