Ant Group and vLLM community release Humming—a kernel library for quantized inference supporting Zhenwu 810E
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Ant Group and vLLM community release Humming—a kernel library for quantized inference supporting Zhenwu 810E

Ant Group, in collaboration with the vLLM community, has released Humming—a GEMM kernel library designed for the inference of quantized Large Language Models (LLMs) using Just-In-Time (JIT) compilation.

This project is released under the Apache 2.0 license via Ant Group's inclusionAI organization and is mirrored in the vLLM project. Humming addresses a common challenge in quantized serving: the need to create a specialized kernel for every combination of weight bit width, activation format, scaling scheme, dense or Mixture-of-Experts (MoE) layout structure, and GPU generation.

The library translates supported export formats into a unified quantization description, then performs JIT compilation of kernels from reusable CUDA components for a specified target shape, device, and computational options.

According to the project description, one architecture supports integer weights from INT1 to INT8 and floating-point weights from FP4 to FP8, combined with activations of FP16, BF16, FP8, FP4, INT8, or INT4, provided the formats are compatible. It supports both dense GEMM and MoE GEMM, and is targeted at NVIDIA GPUs starting from the Turing architecture.

The library dependencies are limited to PyTorch and NVCC, and the pip package size is approximately 100 KB. A significant aspect is the extension of hardware support beyond NVIDIA: Humming fully supports the T-Head Zhenwu 810E AI accelerator, which is CUDA-compatible, allowing the use of the same kernels and configuration paths on this device. T-Head is a subsidiary of Alibaba, and Zhenwu 810E was previously used in Chinese AI deployments.

Performance results presented in the joint announcement are based on proprietary microbenchmarks and team serving tests. At small batch sizes, dense W4A16 showed an improvement of approximately 3–4 times compared to cuBLAS across several devices. On B300, the W4A8 ratio to baseline DeepGEMM is estimated at about 1.6 times in some cases with small and medium batch sizes.

During end-to-end testing of DeepSeek-V4.1-Flash on eight H20 GPUs, transitioning from MoE Marlin W4A16 to Humming W4A8 increased output throughput by approximately 14%–43% and reduced the average time to first token by 25%–32%, with the largest gain observed at higher parallelism. These results have not yet undergone independent verification.

Starting with vLLM version 0.29.0, vLLM installations with CUDA include Humming as a dependency. Operators can activate it using the flags --linear-backend humming and --moe-backend humming. It should be noted that Humming is a joint project of Ant Group and the vLLM community, not exclusively an Ant Group product.

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Ant Group releases Ming-3.0-tts: small speech synthesis models with voice control
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Ant Group releases Ming-3.0-tts: small speech synthesis models with voice control

The inclusionAI team from Ant Group has released Ming-3.0-tts, a family of speech generation models, to the public. The goal is to provide detailed control over the voice in models that are compact enough for local deployment. The code, task guide, and evaluation tables are available on GitHub, while the weights are hosted on Hugging Face and ModelScope; all materials are distributed under the MIT license.

The release includes two models. Ming-3.0-tts-0.5b is a dense model with approximately half a billion parameters. Ming-3.0-tts-tiny is a mixture-of-experts model built on Ant's Ling V3 architecture, which has a total of 7.9 billion parameters, with 1.3 billion active per token. Both models utilize a unified generative structure: a language model, governed by task-specific control tokens and text, influences the DiT patch-head and audio tokenizer.

The capabilities of these models are extensive. Users can clone a reference voice, create a new voice based on a textual description, or choose from over 100 built-in Chinese voices and five English ones. Speech delivery can be regulated using natural language instructions, and parameters such as emotion, speed, pitch, and volume can be adjusted directly. Fourteen built-in tags allow for the insertion of laughter, sighs, and other vocal sounds in both Chinese and English. The models support Chinese, English, German, Spanish, French, Korean, and Japanese, as well as Sichuanese and Cantonese.

The tiny model additionally supports multi-speaker dialogues generated from scripts and voice samples, which is geared towards creating audio in the style of long podcasts. The team notes that these capabilities function as separate task modes rather than arbitrary combinations.

According to the repository, Ming-3.0-tts ranked first among open-weight models in the Artificial Analysis Controlled Voice leaderboard and achieved first place overall in the Pronunciation Robustness Score, reaching 93.2% correctly pronounced highlighted segments as of October 2026. However, the team warns that these rankings change as new models are added. On open benchmarks, the model demonstrates state-of-the-art results among listed open-source models across all six NVBench paralinguistic metrics with one tag, error frequency on the base Cantonese dataset for the 0.5b model, and word error frequency in German and Spanish for the tiny model.

The README file also honestly lists weaknesses. Voice similarity in recently added languages generally does not make the top three leaders, pitch control is the least accurate attribute, the error rate of the tiny Korean model lags significantly behind the smaller model, and the audio quality for podcasts still falls short of the earlier Ming-omni-tts release. The roadmap indicates accelerated vLLM-Omni inference and a technical report as next steps. The project urges users not to clone real people's voices without consent and to label synthesized speech where it might be mistaken for a real recording.

4Paradigm and RiseUnion launch ModelSphere—a platform for serving heterogeneous LLMs
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4Paradigm and RiseUnion launch ModelSphere—a platform for serving heterogeneous LLMs

Companies 4Paradigm and RiseUnion have released ModelSphere, a production platform for deploying artificial intelligence, to the public. This platform integrates compute resource management, model serving, and runtime environment configuration under the Apache 2.0 license. The project code is available on GitHub at modelsphere/modelsphere, where the Apache-2.0 license is also specified, and it was republished on the day of the launch.

Representatives from the companies note that ModelSphere operates on top of engines such as vLLM and SGLang. While these stacks aim for maximum utilization of a single server, ModelSphere handles cluster routing, a shared KV cache pool, automatic scaling based on quality of service, crash recovery, and online auto-tuning.

The joint announcement details built-in features, including a graphical deployment dashboard, elastic scaling, serving with prefix and decoding separation, three-level KV caching, high-availability load balancing, and automatic recovery after service failure.

Special attention is given to heterogeneous chips. Thanks to the EngineX inference level, ModelSphere allows scheduling resources from Huawei Ascend, Iluvatar, Moore Threads, and Hygon DCU within a single pool, eliminating the need for teams to rewrite low-level code for each accelerator. The project already includes adaptations for DeepSeek-V4 and V4-Flash, Qwen, Kimi, MiniMax, and GLM, with plans to release production-ready configurations shortly after new open models appear.

Regarding performance, the announcement claims that thanks to AutoTune and hardware-aware balancing features, peak throughput can increase by approximately 285% compared to the internal baseline, described as a fourfold jump. It should be noted that this figure is a statement by the companies themselves and has not undergone independent audit, so it should be viewed as a marketing estimate until reproduced by external parties.

According to the stack diagram presented by the partners, node engines remain responsible for cores and batching, while ModelSphere manages request routing, considering cache locality, a unified KV pool across replicas, SLO-based scaling decisions, and operational monitoring. The core idea is that enterprises already using mixed accelerators can deploy stable LLM APIs without having to build a private control plane from scratch.

The release is free for enterprise use under the Apache 2.0 license and is positioned as infrastructure software, not a story about funding or revenue.

Xiaomi releases open audio model MiDashengLM-Spatial capable of detecting sound sources
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Xiaomi releases open audio model MiDashengLM-Spatial capable of detecting sound sources

Xiaomi has introduced MiDashengLM-Spatial, a new component of its Dasheng series of audio models. This model is designed not only to understand the content of sound but also its spatial origin.

The full model weights and inference code have been published by the company under the Apache 2.0 license, which permits both research and commercial use. The model is built upon MiDashengLM-7B, a large sound understanding model that Xiaomi previously made open source.

To enable spatial perception, a spatial audio encoder named Spatial-Dasheng was added to the model. According to a technical report posted on arXiv, it utilizes a hierarchical semantic-to-spatial conditioning module that separately captures semantic and spatial information before merging them. This results in a unified end-to-end model capable of analyzing speech, background noise, and music, as well as inferring the direction, distance, and movement of the sound source.

On the Hugging Face page, the model is presented with approximately 8 billion parameters. Spatial cues depend on multi-channel input; the model description states that it was trained on binaural two-channel audio, although mono input is also supported, it does not carry directional information.

Xiaomi asserts that the general audio capabilities of the model are preserved, and the research team reports that performance in basic audio processing tasks remains comparable to large audio-language models of around 8 billion parameters.

Xiaomi positions MiDashengLM-Spatial as the first open, end-to-end model for spatial audio understanding in the industry. It should be noted that this claim is contained within the article and qualified with the phrase 'to our knowledge,' and it has not yet been independently verified.

Over the past two years, Xiaomi has made its fundamental sound models, Xiaomi Dasheng and MiDashengLM-7B, open source. The company notes that technologies from the Dasheng series are already used in over 30 applications across its smart home and automotive cockpit products, covering speech recognition, ambient sounds, and music. This figure refers to the existing series, not the new MiDashengLM-Spatial model.

Regarding MiDashengLM-Spatial itself, Xiaomi plans to gradually integrate it into smart home and automotive cockpit scenarios, where knowing the origin of a voice or alarm signal will allow assistants to react more accurately. The company has not yet provided timelines or names of specific products.

The release includes a project demo page, a technical report on arXiv, evaluation code on GitHub, and model weights on Hugging Face.

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