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Alejandro Luebs

6 accepted papers

2023

High-Fidelity Audio Compression with Improved RVQGAN

NeurIPS 2023spotlight

Language models have been successfully used to model natural signals, such as images, speech, and music. A key component of these models is a high quality neural compression model that can compress high-dimensional natural signals into lower dimensional discrete tokens. To that end, we introduce a h…

2021

Generative Speech Coding with Predictive Variance Regularization

ICASSP 2021accepted

The recent emergence of machine-learning based generative models for speech suggests a significant reduction in bit rate for speech codecs is possible. However, the performance of generative models deteriorates significantly with the distortions present in real-world input signals. We argue that thi…

Cited by 0SourceScholar
2019

Low Bit-rate Speech Coding with VQ-VAE and a WaveNet Decoder

ICASSP 2019accepted

In order to efficiently transmit and store speech signals, speech codecs create a minimally redundant representation of the input signal which is then decoded at the receiver with the best possible perceptual quality. In this work we demonstrate that a neural network architecture based on VQ-VAE wit…

Cited by 0SourceScholar
2018

Wavenet Based Low Rate Speech Coding

ICASSP 2018accepted

Traditional parametric coding of speech facilitates low rate but provides poor reconstruction quality because of the inadequacy of the model used. We describe how a WaveNet generative speech model can be used to generate high quality speech from the bit stream of a standard parametric coder operatin…

Cited by 155SourceScholar
2017

Practically efficient nonlinear acoustic echo cancellers using cascaded block RLS and FLMS adaptive filters

ICASSP 2017accepted

This paper presents a practically efficient implementation for non-linear acoustic echo cancellation (NAEC). The echo path is modeled by a novel hybrid Taylor-Volterra pre-processor followed by a linear FIR filter. A cascaded block RLS and unconstrained FLMS adaptive algorithm is developed to jointl…

Cited by 21SourceScholar
2016

Globally optimized least-squares post-filtering for microphone array speech enhancement

ICASSP 2016accepted

Existing post-filtering techniques for microphone array speech enhancement have two common deficiencies. First, they assume that the noise is either white or diffuse and cannot deal with point inter-ferers. Second, they estimate the post-filter coefficients using only two microphones at a time and t…

Cited by 21SourceScholar