wavencoder 0.1.1

WavEncoder - PyTorch backed audio encoder

WavEncoder

WavEncoder is a Python library for encoding audio signal, transforms for audio augmention and training audio classification models with PyTorch backend.

Package Contents

Layers	Models	Transforms	Trainer and utils
Attention Dot Soft Additive Multiplicative SincNet layer Time Delay Neural Network(TDNN)	PreTrained wav2vec SincNet RawNet Baseline 1DCNN LSTM Classifier LSTM Attention Classifier	Noise(Environmet/Gaussian White Noise) Speed Change PadCrop Clip Reverberation TimeShift TimeMask FrequencyMask	Classification Trainer Classification Testing Download Noise Dataset Download Impulse Response Dataset

Wav Models to be added

• wav2vec [1]
• wav2vec2 [2]
• SincNet [3]
• PASE [4]
• MockingJay [5]
• RawNet [6]
• GaborNet [7]
• LEAF [8]
• CNN-1D
• CNN-LSTM
• CNN-LSTM-Attn
• CNN-Transformer

Check the Demo Colab Notebook.

Installation

Use the package manager pip to install wavencoder.

pip install wavencoder

Usage

Import pretrained encoder, baseline models and classifiers

import torch
import wavencoder

x = torch.randn(1, 16000) # [1, 16000]
encoder = wavencoder.models.Wav2Vec(pretrained=True)
z = encoder(x) # [1, 512, 98]

classifier = wavencoder.models.LSTM_Attn_Classifier(512, 64, 2,                          
                                                    return_attn_weights=True, 
                                                    attn_type='soft')
y_hat, attn_weights = classifier(z) # [1, 2], [1, 98]

Use wavencoder with PyTorch Sequential or class modules

import torch
import torch.nn as nn
import wavencoder

model = nn.Sequential(
        wavencoder.models.Wav2Vec(),
        wavencoder.models.LSTM_Attn_Classifier(512, 64, 2,                          
                                               return_attn_weights=True, 
                                               attn_type='soft')
)

x = torch.randn(1, 16000) # [1, 16000]
y_hat, attn_weights = model(x) # [1, 2], [1, 98]

import torch
import torch.nn as nn
import wavencoder

class AudioClassifier(nn.Module):
    def __init__(self):
        super(AudioClassifier, self).__init__()
        self.encoder = wavencoder.models.Wav2Vec(pretrained=True)
        self.classifier = nn.Linear(512, 2)

    def forward(self, x):
        z = self.encoder(x)
        z = torch.mean(z, dim=2)
        out = self.classifier(z)
        return out

model = AudioClassifier()
x = torch.randn(1, 16000) # [1, 16000]
y_hat = model(x) # [1, 2]

Train the encoder-classifier models

from wavencoder.models import Wav2Vec, LSTM_Attn_Classifier
from wavencoder.trainer import train, test_evaluate_classifier, test_predict_classifier

model = nn.Sequential(
    Wav2Vec(pretrained=False),
    LSTM_Attn_Classifier(512, 64, 2)
)

trainloader = ...
valloader = ...
testloader = ...

trained_model, train_dict = train(model, trainloader, valloader, n_epochs=20)
test_prediction_dict = test_predict_classifier(trained_model, testloader)

Add Transforms to your DataLoader for Augmentation/Processing the wav signal

from wavencoder.transforms import Compose, AdditiveNoise, SpeedChange, Clipping, PadCrop, Reverberation

audio, _ = torchaudio.load('test.wav')

transforms = Compose([
                    AdditiveNoise('path-to-noise-folder', p=0.5, snr_levels=[5, 10, 15], p=0.5), 
                    SpeedChange(factor_range=(-0.5, 0.0), p=0.5), 
                    Clipping(p=0.5),
                    PadCrop(48000, crop_position='random', pad_position='random') 
                    ])

transformed_audio = transforms(audio)

Contributing

Pull requests are welcome. For major changes, please open an issue first to discuss what you would like to change.

Please make sure to update tests as appropriate.

License

MIT

Reference

	Paper	Code
[1]	Wav2Vec: Unsupervised Pre-training for Speech Recognition	GitHub
[2]	Wav2vec 2.0: Learning the structure of speech from raw audio	GitHub
[3]	Speaker Recognition from Raw Waveform with SincNet	GitHub
[4]	Learning Problem-agnostic Speech Representations from Multiple Self-supervised Tasks	GitHub
[5]	Mockingjay: Unsupervised Speech Representation Learning with Deep Bidirectional Transformer Encoders	GitHub
[6]	Improved RawNet with Feature Map Scaling for Text-independent Speaker Verification using Raw Waveforms	GitHub