activity
20122024
most citedGemini 1.5: Unlocking multimodal understanding across millions of tokens of context

297 citations · 1.2k across the 31 of their papers we have counts for

collaborators

36 papers

cs.CL2024★ 297 cited

Gemini 1.5: Unlocking multimodal understanding across millions of tokens of context

Gemini Team, Petko Georgiev, Ving Ian Lei +1132

In this report, we introduce the Gemini 1.5 family of models, representing the next generation of highly compute-efficient multimodal models capable of recalling and reasoning over…

cs.SD2023

TokenSplit: Using Discrete Speech Representations for Direct, Refined, and Transcript-Conditioned Speech Separation and Recognition

Hakan Erdogan, Scott Wisdom, Xuankai Chang +4

We present TokenSplit, a speech separation model that acts on discrete token sequences. The model is trained on multiple tasks simultaneously: separate and transcribe each speech s…

cs.CL2023★ 43 cited

AudioPaLM: A Large Language Model That Can Speak and Listen

Paul K. Rubenstein, Chulayuth Asawaroengchai, Duc Dung Nguyen +27

We introduce AudioPaLM, a large language model for speech understanding and generation. AudioPaLM fuses text-based and speech-based language models, PaLM-2 [Anil et al., 2023] and…

cs.SD2023★ 18 cited

SoundStorm: Efficient Parallel Audio Generation

Zalán Borsos, Matt Sharifi, Damien Vincent +3

We present SoundStorm, a model for efficient, non-autoregressive audio generation. SoundStorm receives as input the semantic tokens of AudioLM, and relies on bidirectional attentio…

cs.SD2023★ 1 cited

LMCodec: A Low Bitrate Speech Codec With Causal Transformer Models

Teerapat Jenrungrot, Michael Chinen, W. Bastiaan Kleijn +4

We introduce LMCodec, a causal neural speech codec that provides high quality audio at very low bitrates. The backbone of the system is a causal convolutional codec that encodes au…

cs.SD2023★ 3 cited

Speak, Read and Prompt: High-Fidelity Text-to-Speech with Minimal Supervision

Eugene Kharitonov, Damien Vincent, Zalán Borsos +6

We introduce SPEAR-TTS, a multi-speaker text-to-speech (TTS) system that can be trained with minimal supervision. By combining two types of discrete speech representations, we cast…

cs.SD2023★ 189 cited

MusicLM: Generating Music From Text

Andrea Agostinelli, Timo I. Denk, Zalán Borsos +10

We introduce MusicLM, a model generating high-fidelity music from text descriptions such as "a calming violin melody backed by a distorted guitar riff". MusicLM casts the process o…

cs.SD2022★ 24 cited

AudioLM: a Language Modeling Approach to Audio Generation

Zalán Borsos, Raphaël Marinier, Damien Vincent +8

We introduce AudioLM, a framework for high-quality audio generation with long-term consistency. AudioLM maps the input audio to a sequence of discrete tokens and casts audio genera…

eess.AS2022

Text-Driven Separation of Arbitrary Sounds

Kevin Kilgour, Beat Gfeller, Qingqing Huang +3

We propose a method of separating a desired sound source from a single-channel mixture, based on either a textual description or a short audio sample of the target source. This is…

cs.SD2022★ 2 cited

SpeechPainter: Text-conditioned Speech Inpainting

Zalán Borsos, Matt Sharifi, Marco Tagliasacchi

We propose SpeechPainter, a model for filling in gaps of up to one second in speech samples by leveraging an auxiliary textual input. We demonstrate that the model performs speech…

eess.AS2022

CycleGAN-Based Unpaired Speech Dereverberation

Hannah Muckenhirn, Aleksandr Safin, Hakan Erdogan +4

Typically, neural network-based speech dereverberation models are trained on paired data, composed of a dry utterance and its corresponding reverberant utterance. The main limitati…

cs.SD2022★ 1 cited

Disentangling speech from surroundings with neural embeddings

Ahmed Omran, Neil Zeghidour, Zalán Borsos +3

We present a method to separate speech signals from noisy environments in the embedding space of a neural audio codec. We introduce a new training procedure that allows our model t…

eess.AS2022

Real time spectrogram inversion on mobile phone

Oleg Rybakov, Marco Tagliasacchi, Yunpeng Li +3

We present two methods of real time magnitude spectrogram inversion: streaming Griffin Lim(GL) and streaming MelGAN. We demonstrate the impact of looking ahead on perceptual qualit…

cs.LG2021★ 1 cited

Data Summarization via Bilevel Optimization

Zalán Borsos, Mojmír Mutný, Marco Tagliasacchi +1

The increasing availability of massive data sets poses a series of challenges for machine learning. Prominent among these is the need to learn models under hardware or human resour…

cs.SD2021

SoundStream: An End-to-End Neural Audio Codec

Neil Zeghidour, Alejandro Luebs, Ahmed Omran +2

We present SoundStream, a novel neural audio codec that can efficiently compress speech, music and general audio at bitrates normally targeted by speech-tailored codecs. SoundStrea…

cs.SD2021

Self-Supervised Learning from Automatically Separated Sound Scenes

Eduardo Fonseca, Aren Jansen, Daniel P. W. Ellis +7

Real-world sound scenes consist of time-varying collections of sound sources, each generating characteristic sound events that are mixed together in audio recordings. The associati…

cs.SD2021★ 30 cited

LEAF: A Learnable Frontend for Audio Classification

Neil Zeghidour, Olivier Teboul, Félix de Chaumont Quitry +1

Mel-filterbanks are fixed, engineered audio features which emulate human perception and have been used through the history of audio understanding up to today. However, their undeni…

eess.AS2020

One-shot conditional audio filtering of arbitrary sounds

Beat Gfeller, Dominik Roblek, Marco Tagliasacchi

We consider the problem of separating a particular sound source from a single-channel mixture, based on only a short sample of the target source. Using SoundFilter, a wave-to-wave…

cs.SD2020

MicAugment: One-shot Microphone Style Transfer

Zalán Borsos, Yunpeng Li, Beat Gfeller +1

A crucial aspect for the successful deployment of audio-based models "in-the-wild" is the robustness to the transformations introduced by heterogeneous acquisition conditions. In t…

cs.LG2020

Semi-supervised Batch Active Learning via Bilevel Optimization

Zalán Borsos, Marco Tagliasacchi, Andreas Krause

Active learning is an effective technique for reducing the labeling cost by improving data efficiency. In this work, we propose a novel batch acquisition strategy for active learni…

eess.AS2020★ 1 cited

Real-time Speech Frequency Bandwidth Extension

Yunpeng Li, Marco Tagliasacchi, Oleg Rybakov +2

In this paper we propose a lightweight model for frequency bandwidth extension of speech signals, increasing the sampling frequency from 8kHz to 16kHz while restoring the high freq…

eess.AS2020

SEANet: A Multi-modal Speech Enhancement Network

Marco Tagliasacchi, Yunpeng Li, Karolis Misiunas +1

We explore the possibility of leveraging accelerometer data to perform speech enhancement in very noisy conditions. Although it is possible to only partially reconstruct user's spe…

eess.AS2020★ 14 cited

Learning to Denoise Historical Music

Yunpeng Li, Beat Gfeller, Marco Tagliasacchi +1

We propose an audio-to-audio neural network model that learns to denoise old music recordings. Our model internally converts its input into a time-frequency representation by means…

eess.AS2020★ 126 cited

Towards Learning a Universal Non-Semantic Representation of Speech

Joel Shor, Aren Jansen, Ronnie Maor +7

The ultimate goal of transfer learning is to reduce labeled data requirements by exploiting a pre-existing embedding model trained for different datasets or tasks. The visual and l…