pith. sign in

hub Canonical reference

Kimi-Audio Technical Report

Canonical reference. 73% of citing Pith papers cite this work as background.

68 Pith papers citing it
Background 73% of classified citations
abstract

We present Kimi-Audio, an open-source audio foundation model that excels in audio understanding, generation, and conversation. We detail the practices in building Kimi-Audio, including model architecture, data curation, training recipe, inference deployment, and evaluation. Specifically, we leverage a 12.5Hz audio tokenizer, design a novel LLM-based architecture with continuous features as input and discrete tokens as output, and develop a chunk-wise streaming detokenizer based on flow matching. We curate a pre-training dataset that consists of more than 13 million hours of audio data covering a wide range of modalities including speech, sound, and music, and build a pipeline to construct high-quality and diverse post-training data. Initialized from a pre-trained LLM, Kimi-Audio is continual pre-trained on both audio and text data with several carefully designed tasks, and then fine-tuned to support a diverse of audio-related tasks. Extensive evaluation shows that Kimi-Audio achieves state-of-the-art performance on a range of audio benchmarks including speech recognition, audio understanding, audio question answering, and speech conversation. We release the codes, model checkpoints, as well as the evaluation toolkits in https://github.com/MoonshotAI/Kimi-Audio.

hub tools

citation-role summary

background 11 baseline 4

citation-polarity summary

years

2026 60 2025 8

clear filters

representative citing papers

Benign Fine-Tuning Breaks Safety Alignment in Audio LLMs

cs.CR · 2026-04-17 · conditional · novelty 8.0

Benign fine-tuning on audio data breaks safety alignment in Audio LLMs by raising jailbreak success rates up to 87%, with the dominant risk axis depending on model architecture and embedding proximity to harmful content.

FlexiSLM: A Dynamic and Controllable Frame Rate Spoken Language Model

cs.SD · 2026-06-30 · unverdicted · novelty 7.0

FlexiSLM is the first spoken language model supporting dynamic and controllable frame rates on speech input and output, outperforming fixed-rate 7B models at high quality and enabling faster inference at lower rates like 6.25 Hz.

Liberating LLM Capabilities in Full-Duplex Speech Models

cs.CL · 2026-05-04 · unverdicted · novelty 7.0

LWS is a text-first paradigm for full-duplex speech LLMs that treats visible writing as a primary output channel alongside audio input and spoken response, implemented via token schema and synthetic per-second annotations.

TiCo: Time-Controllable Spoken Dialogue Model

cs.CL · 2026-03-23 · unverdicted · novelty 7.0

TiCo enables spoken dialogue models to follow explicit time constraints in generated responses using Spoken Time Markers and reinforcement learning with verifiable rewards, cutting duration error by 2.7x over its backbone.

citing papers explorer

Showing 2 of 2 citing papers after filters.