File size: 4,243 Bytes
171bc27 |
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 |
import torch
from speechbrain.inference.interfaces import Pretrained
class ASR(Pretrained):
"""A ready-to-use class for utterance-level classification (e.g, speaker-id,
language-id, emotion recognition, keyword spotting, etc).
The class assumes that an self-supervised encoder like wav2vec2/hubert and a classifier model
are defined in the yaml file. If you want to
convert the predicted index into a corresponding text label, please
provide the path of the label_encoder in a variable called 'lab_encoder_file'
within the yaml.
The class can be used either to run only the encoder (encode_batch()) to
extract embeddings or to run a classification step (classify_batch()).
```
Example
-------
>>> import torchaudio
>>> from speechbrain.pretrained import EncoderClassifier
>>> # Model is downloaded from the speechbrain HuggingFace repo
>>> tmpdir = getfixture("tmpdir")
>>> classifier = EncoderClassifier.from_hparams(
... source="speechbrain/spkrec-ecapa-voxceleb",
... savedir=tmpdir,
... )
>>> # Compute embeddings
>>> signal, fs = torchaudio.load("samples/audio_samples/example1.wav")
>>> embeddings = classifier.encode_batch(signal)
>>> # Classification
>>> prediction = classifier .classify_batch(signal)
"""
def __init__(self, *args, **kwargs):
super().__init__(*args, **kwargs)
def encode_batch(self, wavs, wav_lens=None, normalize=False):
"""Encodes the input audio into a single vector embedding.
The waveforms should already be in the model's desired format.
You can call:
``normalized = <this>.normalizer(signal, sample_rate)``
to get a correctly converted signal in most cases.
Arguments
---------
wavs : torch.tensor
Batch of waveforms [batch, time, channels] or [batch, time]
depending on the model. Make sure the sample rate is fs=16000 Hz.
wav_lens : torch.tensor
Lengths of the waveforms relative to the longest one in the
batch, tensor of shape [batch]. The longest one should have
relative length 1.0 and others len(waveform) / max_length.
Used for ignoring padding.
normalize : bool
If True, it normalizes the embeddings with the statistics
contained in mean_var_norm_emb.
Returns
-------
torch.tensor
The encoded batch
"""
batch = batch.to(self.device)
sig, self.sig_lens = batch.sig
tokens_bos, _ = batch.tokens_bos
sig, self.sig_lens = sig.to(self.device), self.sig_lens.to(self.device)
# Forward pass
encoded_outputs = self.modules.encoder_w2v2(sig.detach())
embedded_tokens = self.modules.embedding(tokens_bos)
decoder_outputs, _ = self.modules.decoder(embedded_tokens, encoded_outputs, self.sig_lens)
# Output layer for seq2seq log-probabilities
logits = self.modules.seq_lin(decoder_outputs)
predictions = {"seq_logprobs": self.hparams.log_softmax(logits)}
predictions["tokens"], _, _, _ = self.hparams.test_search(encoded_outputs, self.sig_lens)
return predictions
def classify_file(self, path):
"""Classifies the given audiofile into the given set of labels.
Arguments
---------
path : str
Path to audio file to classify.
Returns
-------
out_prob
The log posterior probabilities of each class ([batch, N_class])
score:
It is the value of the log-posterior for the best class ([batch,])
index
The indexes of the best class ([batch,])
text_lab:
List with the text labels corresponding to the indexes.
(label encoder should be provided).
"""
waveform = self.load_audio(path)
# Fake a batch:
batch = waveform.unsqueeze(0)
rel_length = torch.tensor([1.0])
outputs = self.encode_batch(batch, rel_length)["tokens"]
return outputs
def forward(self, wavs, wav_lens=None):
return self.encode_batch(wavs=wavs, wav_lens=wav_lens) |