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Arun Narayanan

15 accepted papers

2025

Bone Conducted Signal Guided Speech Enhancement For Voice Assistant on Earbuds

ICASSP 2025accepted

In this work we present a multi-modal, streaming enhancement network to improve speech recognition for voice assistants on earbuds. The proposed model is guided by a bone conducted signal (BCS) to separate the interfering sources from the target speaker signal. We train the model on a simulated spee…

Cited by 0SourceScholar
2024

Extreme Encoder Output Frame Rate Reduction: Improving Computational Latencies of Large End-to-End Models

ICASSP 2024accepted

The accuracy of end-to-end (E2E) automatic speech recognition (ASR) models continues to improve as they are scaled to larger sizes, with some now reaching billions of parameters. Widespread deployment and adoption of these models, however, requires computationally efficient strategies for decoding.…

Cited by 11SourceScholar
2024

Improving Acoustic Echo Cancellation for Voice Assistants Using Neural Echo Suppression and Multi-Microphone Noise Reduction

ICASSP 2024accepted

Keyword spotting (KS) and automatic speech recognition (ASR) on smart speakers in a home environment with interfering signals from loudspeakers are challenging tasks to this day, despite improvements in acoustic echo cancellation (AEC) systems. In this work we propose to combine a single microphone…

Cited by 0SourceScholar
2023

Cleanformer: A Multichannel Array Configuration-Invariant Neural Enhancement Frontend for ASR in Smart Speakers

ICASSP 2023accepted

This work introduces Cleanformer —a streaming multichannel neural enhancement frontend for automatic speech recognition (ASR). This model has a Conformer-based architecture which takes as inputs a single channel each of raw and enhanced signals, and uses self-attention to derive a time-frequency mas…

Cited by 0SourceScholar
2023

Conditional Conformer: Improving Speaker Modulation For Single And Multi-User Speech Enhancement

ICASSP 2023accepted

Recently, Feature-wise Linear Modulation (FiLM) has been shown to outperform other approaches to incorporate speaker embedding into speech separation and VoiceFilter models. We propose an improved method of incorporating such embeddings into a Voice- Filter frontend for automatic speech recognition…

Cited by 5SourceScholar
2022

Improving The Latency And Quality Of Cascaded Encoders

ICASSP 2022accepted

In this paper, we explore reducing computational latency of the 2-pass cascaded encoder model [1]. Specifically, we experiment with reducing the size of the causal 1st-pass and adding capacity to the non-causal 2nd-pass, such that the overall latency can be reduced without loss of quality. In additi…

Cited by 0SourceScholar
2022

Transducer-Based Streaming Deliberation for Cascaded Encoders

ICASSP 2022accepted

Previous research on applying deliberation networks to automatic speech recognition has achieved excellent results. The attention decoder based deliberation model often works as a rescorer to improve first-pass recognition results, and requires the full first-pass hypothesis for second-pass delibera…

Cited by 0SourceScholar
2021

A Better and Faster end-to-end Model for Streaming ASR

ICASSP 2021accepted

End-to-end (E2E) models have shown to outperform state-of-the-art conventional models for streaming speech recognition [1] across many dimensions, including quality (as measured by word error rate (WER)) and endpointer latency [2]. However, the model still tends to delay the predictions towards the…

Cited by 0SourceScholar
2021

Cascaded Encoders for Unifying Streaming and Non-Streaming ASR

ICASSP 2021accepted

End-to-end (E2E) automatic speech recognition (ASR) models, by now, have shown competitive performance on several benchmarks. These models are structured to either operate in streaming or non-streaming mode. This work presents cascaded encoders for building a single E2E ASR model that can operate in…

Cited by 0SourceScholar
2021

FastEmit: Low-Latency Streaming ASR with Sequence-Level Emission Regularization

ICASSP 2021accepted

Streaming automatic speech recognition (ASR) aims to emit each hypothesized word as quickly and accurately as possible. However, emitting fast without degrading quality, as measured by word error rate (WER), is highly challenging. Existing approaches including Early and Late Penalties [1] and Constr…

Cited by 0SourceScholar
2021

Improving Streaming Automatic Speech Recognition with Non-Streaming Model Distillation on Unsupervised Data

ICASSP 2021accepted

Streaming end-to-end automatic speech recognition (ASR) models are widely used on smart speakers and on-device applications. Since these models are expected to transcribe speech with minimal latency, they are constrained to be causal with no future context, compared to their non-streaming counterpar…

Cited by 0SourceScholar
2021

Less is More: Improved RNN-T Decoding Using Limited Label Context and Path Merging

ICASSP 2021accepted

End-to-end models that condition the output sequence on all previously predicted labels have emerged as popular alternatives to conventional systems for automatic speech recognition (ASR). Since distinct label histories correspond to distinct models states, such models are decoded using an approxima…

Cited by 37SourceScholar
2020

A Streaming On-Device End-To-End Model Surpassing Server-Side Conventional Model Quality and Latency

ICASSP 2020accepted

Thus far, end-to-end (E2E) models have not been shown to outperform state-of-the-art conventional models with respect to both quality, i.e., word error rate (WER), and latency, i.e., the time the hypothesis is finalized after the user stops speaking. In this paper, we develop a first-pass Recurrent…

Cited by 0SourceScholar
2018

Spectral Distortion Model for Training Phase-Sensitive Deep-Neural Networks for Far-Field Speech Recognition

ICASSP 2018accepted

In this paper, we present an algorithm which introduces phase-perturbation to the training database when training phase-sensitive deep neural-network models. Traditional features such as log-mel or cepstral features do not have have any phase-relevant information. However features such as raw-wavefo…

Cited by 3SourceScholar
2016

Factored spatial and spectral multichannel raw waveform CLDNNs

ICASSP 2016accepted

Multichannel ASR systems commonly separate speech enhancement, including localization, beamforming and postfiltering, from acoustic modeling. Recently, we explored doing multichannel enhancement jointly with acoustic modeling, where beamforming and frequency decomposition was folded into one layer o…

Cited by 0SourceScholar