# Copyright 2021 The HuggingFace Team. All rights reserved. # # Licensed under the Apache License, Version 2.0 (the "License"); # you may not use this file except in compliance with the License. # You may obtain a copy of the License at # # http://www.apache.org/licenses/LICENSE-2.0 # # Unless required by applicable law or agreed to in writing, software # distributed under the License is distributed on an "AS IS" BASIS, # WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. # See the License for the specific language governing permissions and # limitations under the License. import subprocess from typing import Union import numpy as np import requests from ..utils import add_end_docstrings, is_torch_available, is_torchaudio_available, logging from .base import Pipeline, build_pipeline_init_args if is_torch_available(): from ..models.auto.modeling_auto import MODEL_FOR_AUDIO_CLASSIFICATION_MAPPING_NAMES 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 @add_end_docstrings(build_pipeline_init_args(has_feature_extractor=True)) class AudioClassificationPipeline(Pipeline): """ Audio classification pipeline using any `AutoModelForAudioClassification`. This pipeline predicts the class of a raw waveform or an audio file. In case of an audio file, ffmpeg should be installed to support multiple audio formats. Example: ```python >>> from transformers import pipeline >>> classifier = pipeline(model="superb/wav2vec2-base-superb-ks") >>> classifier("https://huggingface.co/datasets/Narsil/asr_dummy/resolve/main/1.flac") [{'score': 0.997, 'label': '_unknown_'}, {'score': 0.002, 'label': 'left'}, {'score': 0.0, 'label': 'yes'}, {'score': 0.0, 'label': 'down'}, {'score': 0.0, 'label': 'stop'}] ``` Learn more about the basics of using a pipeline in the [pipeline tutorial](../pipeline_tutorial) This pipeline can currently be loaded from [`pipeline`] using the following task identifier: `"audio-classification"`. See the list of available models on [huggingface.co/models](https://huggingface.co/models?filter=audio-classification). """ def __init__(self, *args, **kwargs): # Default, might be overriden by the model.config. kwargs["top_k"] = 5 super().__init__(*args, **kwargs) if self.framework != "pt": raise ValueError(f"The {self.__class__} is only available in PyTorch.") self.check_model_type(MODEL_FOR_AUDIO_CLASSIFICATION_MAPPING_NAMES) def __call__( self, inputs: Union[np.ndarray, bytes, str], **kwargs, ): """ Classify the sequence(s) given as inputs. See the [`AutomaticSpeechRecognitionPipeline`] documentation for more information. Args: inputs (`np.ndarray` or `bytes` or `str` or `dict`): The inputs is either : - `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. - `bytes` it is supposed to be the content of an audio file and is interpreted by *ffmpeg* in the same way. - (`np.ndarray` of shape (n, ) of type `np.float32` or `np.float64`) Raw audio at the correct sampling rate (no further check will be done) - `dict` form can be used to pass raw audio sampled at arbitrary `sampling_rate` and let this pipeline do the resampling. The dict must be either be in the format `{"sampling_rate": int, "raw": np.array}`, or `{"sampling_rate": int, "array": np.array}`, where the key `"raw"` or `"array"` is used to denote the raw audio waveform. top_k (`int`, *optional*, defaults to None): The number of top labels that will be returned by the pipeline. If the provided number is `None` or higher than the number of labels available in the model configuration, it will default to the number of labels. Return: A list of `dict` with the following keys: - **label** (`str`) -- The label predicted. - **score** (`float`) -- The corresponding probability. """ return super().__call__(inputs, **kwargs) def _sanitize_parameters(self, top_k=None, **kwargs): # No parameters on this pipeline right now postprocess_params = {} if top_k is not None: if top_k > self.model.config.num_labels: top_k = self.model.config.num_labels postprocess_params["top_k"] = top_k return {}, {}, postprocess_params def preprocess(self, inputs): if isinstance(inputs, str): if inputs.startswith("http://") or inputs.startswith("https://"): # We need to actually check for a real protocol, otherwise it's impossible to use a local file # like http_huggingface_co.png inputs = requests.get(inputs).content else: with open(inputs, "rb") as f: inputs = f.read() if isinstance(inputs, bytes): inputs = ffmpeg_read(inputs, self.feature_extractor.sampling_rate) if isinstance(inputs, dict): # Accepting `"array"` which is the key defined in `datasets` for # better integration if not ("sampling_rate" in inputs and ("raw" in inputs or "array" in inputs)): raise ValueError( "When passing a dictionary to AudioClassificationPipeline, the dict needs to contain a " '"raw" key containing the numpy array representing the audio and a "sampling_rate" key, ' "containing the sampling_rate associated with that array" ) _inputs = inputs.pop("raw", None) if _inputs is None: # Remove path which will not be used from `datasets`. inputs.pop("path", None) _inputs = inputs.pop("array", None) in_sampling_rate = inputs.pop("sampling_rate") inputs = _inputs if in_sampling_rate != self.feature_extractor.sampling_rate: import torch if is_torchaudio_available(): from torchaudio import functional as F else: raise ImportError( "torchaudio is required to resample audio samples in AudioClassificationPipeline. " "The torchaudio package can be installed through: `pip install torchaudio`." ) inputs = F.resample( torch.from_numpy(inputs), in_sampling_rate, self.feature_extractor.sampling_rate ).numpy() if not isinstance(inputs, np.ndarray): raise ValueError("We expect a numpy ndarray as input") if len(inputs.shape) != 1: raise ValueError("We expect a single channel audio input for AudioClassificationPipeline") processed = self.feature_extractor( inputs, sampling_rate=self.feature_extractor.sampling_rate, return_tensors="pt" ) return processed def _forward(self, model_inputs): model_outputs = self.model(**model_inputs) return model_outputs def postprocess(self, model_outputs, top_k=5): probs = model_outputs.logits[0].softmax(-1) scores, ids = probs.topk(top_k) scores = scores.tolist() ids = ids.tolist() labels = [{"score": score, "label": self.model.config.id2label[_id]} for score, _id in zip(scores, ids)] return labels