Source code for transformers.pipelines.automatic_speech_recognition

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# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
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#
#     http://www.apache.org/licenses/LICENSE-2.0
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# Unless required by applicable law or agreed to in writing, software
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import subprocess
from typing import TYPE_CHECKING, Union

import numpy as np

from ..utils import logging
from .base import Pipeline


if TYPE_CHECKING:
    from ...feature_extraction_sequence_utils import SequenceFeatureExtractor

logger = logging.get_logger(__name__)


def ffmpeg_read(bpayload: bytes, sampling_rate: int) -> np.array:
    """
    Helper function to read an audio file through ffmpeg.
    """
    ar = f"{sampling_rate}"
    ac = "1"
    format_for_conversion = "f32le"
    ffmpeg_command = [
        "ffmpeg",
        "-i",
        "pipe:0",
        "-ac",
        ac,
        "-ar",
        ar,
        "-f",
        format_for_conversion,
        "-hide_banner",
        "-loglevel",
        "quiet",
        "pipe:1",
    ]

    try:
        ffmpeg_process = subprocess.Popen(ffmpeg_command, stdin=subprocess.PIPE, stdout=subprocess.PIPE)
    except FileNotFoundError:
        raise ValueError("ffmpeg was not found but is required to load audio files from filename")
    output_stream = ffmpeg_process.communicate(bpayload)
    out_bytes = output_stream[0]

    audio = np.frombuffer(out_bytes, np.float32)
    if audio.shape[0] == 0:
        raise ValueError("Malformed soundfile")
    return audio


[docs]class AutomaticSpeechRecognitionPipeline(Pipeline): """ Pipeline that aims at extracting spoken text contained within some audio. The input can be either a raw waveform or a audio file. In case of the audio file, ffmpeg should be installed for to support multiple audio formats """ def __init__(self, feature_extractor: "SequenceFeatureExtractor", *args, **kwargs): """ Arguments: feature_extractor (:obj:`~transformers.SequenceFeatureExtractor`): The feature extractor that will be used by the pipeline to encode waveform for the model. model (:obj:`~transformers.PreTrainedModel` or :obj:`~transformers.TFPreTrainedModel`): The model that will be used by the pipeline to make predictions. This needs to be a model inheriting from :class:`~transformers.PreTrainedModel` for PyTorch and :class:`~transformers.TFPreTrainedModel` for TensorFlow. tokenizer (:obj:`~transformers.PreTrainedTokenizer`): The tokenizer that will be used by the pipeline to encode data for the model. This object inherits from :class:`~transformers.PreTrainedTokenizer`. modelcard (:obj:`str` or :class:`~transformers.ModelCard`, `optional`): Model card attributed to the model for this pipeline. framework (:obj:`str`, `optional`): The framework to use, either :obj:`"pt"` for PyTorch or :obj:`"tf"` for TensorFlow. The specified framework must be installed. If no framework is specified, will default to the one currently installed. If no framework is specified and both frameworks are installed, will default to the framework of the :obj:`model`, or to PyTorch if no model is provided. device (:obj:`int`, `optional`, defaults to -1): Device ordinal for CPU/GPU supports. Setting this to -1 will leverage CPU, a positive will run the model on the associated CUDA device id. """ super().__init__(*args, **kwargs) self.feature_extractor = feature_extractor if self.framework == "tf": raise ValueError("The AutomaticSpeechRecognitionPipeline is only available in PyTorch.")
[docs] def __call__( self, inputs: Union[np.ndarray, bytes, str], **kwargs, ): """ Classify the sequence(s) given as inputs. See the :obj:`~transformers.AutomaticSpeechRecognitionPipeline` documentation for more information. Args: inputs (:obj:`np.ndarray` or :obj:`bytes` or :obj:`str`): The inputs is either a raw waveform (:obj:`np.ndarray` of shape (n, ) of type :obj:`np.float32` or :obj:`np.float64`) at the correct sampling rate (no further check will be done) or a :obj:`str` that is the filename of the audio file, the file will be read at the correct sampling rate to get the waveform using `ffmpeg`. This requires `ffmpeg` to be installed on the system. If `inputs` is :obj:`bytes` it is supposed to be the content of an audio file and is interpreted by `ffmpeg` in the same way. Return: A :obj:`dict` with the following keys: - **text** (:obj:`str`) -- The recognized text. """ if isinstance(inputs, str): with open(inputs, "rb") as f: inputs = f.read() if isinstance(inputs, bytes): inputs = ffmpeg_read(inputs, self.feature_extractor.sampling_rate) assert isinstance(inputs, np.ndarray), "We expect a numpy ndarray as input" assert len(inputs.shape) == 1, "We expect a single channel audio input for AutomaticSpeechRecognitionPipeline" processed = self.feature_extractor( inputs, sampling_rate=self.feature_extractor.sampling_rate, return_tensors="pt" ) processed = self.ensure_tensor_on_device(**processed) name = self.model.__class__.__name__ if name.endswith("ForConditionalGeneration"): input_ids = processed["input_features"] tokens = self.model.generate(input_ids=input_ids) tokens = tokens.squeeze(0) elif name.endswith("ForCTC"): outputs = self.model(**processed) tokens = outputs.logits.squeeze(0).argmax(dim=-1) skip_special_tokens = False if "CTC" in self.tokenizer.__class__.__name__ else True recognized_string = self.tokenizer.decode(tokens, skip_special_tokens=skip_special_tokens) return {"text": recognized_string}