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# 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_INIT_ARGS, Pipeline
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(PIPELINE_INIT_ARGS)
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
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