Dialogue models falter in noisy, multi-speaker environments, often producing irrelevant responses and awkward turn-taking. We present AV-Dialog, the first multimodal dialog framework that uses both audio and visual cues to track the target speaker, predict turn-taking, and generate coherent responses. By combining acoustic tokenization with multi-task, multi-stage training on monadic, synthetic, and real audio-visual dialogue datasets, AV-Dialog achieves robust streaming transcription, semantically grounded turn-boundary detection and accurate responses, resulting in a natural conversational flow. Experiments show that AV-Dialog outperforms audio-only models under interference, reducing transcription errors, improving turn-taking prediction, and enhancing human-rated dialogue quality. These results highlight the power of seeing as well as hearing for speaker-aware interaction, paving the way for {spoken} dialogue agents that perform {robustly} in real-world, noisy environments.

In crowded settings, the human brain can focus on speech from a target speaker, given prior knowledge of how they sound. We introduce a novel intelligent hearable system that achieves this capability, enabling target speech hearing to ignore all interfering speech and noise, but the target speaker. A naïve approach is to require a clean speech example to enroll the target speaker. This is however not well aligned with the hearable application domain since obtaining a clean example is challenging in real world scenarios, creating a unique user interface problem. We present the first enrollment interface where the wearer looks at the target speaker for a few seconds to capture a single, short, highly noisy, binaural example of the target speaker. This noisy example is used for enrollment and subsequent speech extraction in the presence of interfering speakers and noise. Our system achieves a signal quality improvement of 7.01 dB using less than 5 seconds of noisy enrollment audio and can process 8 ms of audio chunks in 6.24 ms on an embedded CPU. Our user studies demonstrate generalization to real-world static and mobile speakers in previously unseen indoor and outdoor multipath environments. Finally, our enrollment interface for noisy examples does not cause performance degradation compared to clean examples, while being convenient and user-friendly. Taking a step back, this paper takes an important step towards enhancing the human auditory perception with artificial intelligence.

Despite broad interest in modeling spoken dialogue agents, most approaches are inherently "half-duplex" -restricted to turn-based interaction with responses requiring explicit prompting by the user or implicit tracking of interruption or silence events.Human dialogue, by contrast, is "full-duplex" allowing for rich synchronicity in the form of quick and dynamic turn-taking, overlapping speech, and backchanneling.Technically, the challenge of achieving full-duplex dialogue with LLMs lies in modeling synchrony as pre-trained LLMs do not have a sense of "time".To bridge this gap, we propose Synchronous LLMs for fullduplex spoken dialogue modeling.We design a novel mechanism to integrate time information into Llama3-8b so that they run synchronously with the real-world clock.We also introduce a training recipe that uses 212k hours of synthetic spoken dialogue data generated from text dialogue data to create a model that generates meaningful and natural spoken dialogue, with just 2k hours of real-world spoken dialogue data.Synchronous LLMs outperform state-of-the-art in dialogue meaningfulness while maintaining naturalness.Finally, we demonstrate the model's ability to participate in full-duplex dialogue by simulating interaction between two agents trained on different datasets, while considering Internet-scale latencies of up to 240ms.

Existing tiled manycore architectures propose to convert abundant silicon resources into general-purpose parallel processors with unmatched computational density and programmability. However, as we approach 100 K cores in one chip, conventional manycore architectures struggle to navigate three key axes: scalability, programmability, and density. Many manycores sacrifice programmability for density; or scalability for programmability. In this paper, we explore HammerBlade, which simultaneously achieves scalability, programmability and density. HammerBlade is a fully open-source RISC-V manycore architecture, which has been silicon-validated with a 2048-core ASIC implementation using a 14/16nm process. We evaluate the system using a suite of parallel benchmarks that captures a broad spectrum of computation and communication patterns.

Integrating cameras into wireless smart rings has been challenging due to size and power constraints. We introduce IRIS, the first wireless vision-enabled smart ring system for smart home interactions. Equipped with a camera, Bluetooth radio, inertial measurement unit (IMU), and an onboard battery, IRIS meets the small size, weight, and power (SWaP) requirements for ring devices. IRIS is context-aware, adapting its gesture set to the detected device, and can last for 16-24 hours on a single charge. IRIS leverages the scene semantics to achieve instance-level device recognition. In a study involving 23 participants, IRIS consistently outpaced voice commands, with a higher proportion of participants expressing a preference for IRIS over voice commands regarding toggling a device’s state, granular control, and social acceptability. Our work pushes the boundary of what is possible with ring form-factor devices, addressing system challenges and opening up novel interaction capabilities.

We present the first neural network model to achieve real-time and streaming target sound extraction. To accomplish this, we propose Waveformer, an encoder-decoder architecture with a stack of dilated causal convolution layers as the encoder, and a transformer decoder layer as the decoder. This hybrid architecture uses dilated causal convolutions for processing large receptive fields in a computationally efficient manner, while also leveraging the generalization performance of transformer-based architectures. Our evaluations show as much as 2.2–3.3 dB improvement in SI-SNRi compared to the prior models for this task while having a 1.2–4x smaller model size and a 1.5–2x lower runtime. We provide code, dataset, and audio samples: https://waveformer.cs.washington.edu/.

We present NeuriCam, a novel deep learning-based system to achieve video capture from low-power dual-mode IoT camera systems. Our idea is to design a dual-mode camera system where the first mode is low power (1.1 mW) but only outputs grey-scale, low resolution and noisy video and the second mode consumes much higher power (100 mW) but outputs color and higher resolution images. To reduce total energy consumption, we heavily duty cycle the high power mode to output an image only once every second. The data for this camera system is then wirelessly sent to a nearby plugged-in gateway, where we run our real-time neural network decoder to reconstruct a higher-resolution color video. To achieve this, we introduce an attention feature filter mechanism that assigns different weights to different features, based on the correlation between the feature map and the contents of the input frame at each spatial location. We design a wireless hardware prototype using off-the-shelf cameras and address practical issues including packet loss and perspective mismatch. Our evaluations show that our dual-camera approach reduces energy consumption by 7x compared to existing systems. Further, our model achieves an average greyscale PSNR gain of 3.7 dB over prior single and dual-camera video super-resolution methods and 5.6 dB RGB gain over prior color propagation methods.

Imagine being able to listen to the birds chirping in a park without hearing the chatter from other hikers, or being able to block out traffic noise on a busy street while still being able to hear emergency sirens and car honks. We introduce semantic hearing, a novel capability for hearable devices that enables them to, in real-time, focus on, or ignore, specific sounds from real-world environments, while also preserving the spatial cues. To achieve this, we make two technical contributions: 1) we present the first neural network that can achieve binaural target sound extraction in the presence of interfering sounds and background noise, and 2) we design a training methodology that allows our system to generalize to real-world use. Results show that our system can operate with 20 sound classes and that our transformer-based network has a runtime of 6.56 ms on a connected smartphone. In-the-wild evaluation with participants in previously unseen indoor and outdoor scenarios shows that our proof-of-concept system can extract the target sounds and generalize to preserve the spatial cues in its binaural output. Project page with code: https://semantichearing.cs.washington.edu

Future CPU-manycore heterogeneous systems can provide high peak throughput by integrating thousands of simple, independent, energy-efficient cores in a single die. However, there are two key challenges to translating this high peak throughput into improved end-to-end workload performance: 1) manycore co-processors rely on simple hardware putting significant demands on the software programmer and 2) manycore co-processors use in-order cores that struggle to tolerate long memory latencies. To address the manycore programmability challenge, this article presents a dense and sparse tensor processing framework based on PyTorch that enables domain experts to easily accelerate off-the-shelf workloads on CPU-manycore heterogeneous systems. To address the manycore memory latency challenge, we use our extended PyTorch framework to explore the potential for decoupled access/execute (DAE) software and hardware mechanisms. More specifically, we propose two software-only techniques, naïve-software DAE and systolic-software DAE, along with a lightweight hardware access accelerator to further improve area-normalized throughput. We evaluate our techniques using a combination of PyTorch operator microbenchmarking and real-world PyTorch workloads running on a detailed register-transfer-level model of a 128-core manycore architecture. Our evaluation on three real-world dense and sparse tensor workloads suggests these workloads can achieve approximately 2–$6\times $performance improvement when scaled to a future 2000-core CPU-manycore heterogeneous system compared to an 18-core out-of-order CPU baseline, while potentially achieving higher area-normalized throughput and improved energy efficiency compared to general-purpose graphics processing units.