Searcharxiv⌕ Search
Home/Archive/arXiv

arXiv · 0905.1235

The Modular Audio Recognition Framework (MARF) and its Applications: Scientific and Software Engineering Notes

Serguei A. Mokhov·Stephen Sinclair·Ian Clément·Dimitrios Nicolacopoulos
View original source ↗Read source document ↗Export metadata ↓

Abstract

MARF is an open-source research platform and a collection of voice/sound/speech/text and natural language processing (NLP) algorithms written in Java and arranged into a modular and extensible framework facilitating addition of new algorithms. MARF can run distributively over the network and may act as a library in applications or be used as a source for learning and extension. A few example applications are provided to show how to use the framework. There is an API reference in the Javadoc format as well as this set of accompanying notes with the detailed description of the architectural design, algorithms, and applications. MARF and its applications are released under a BSD-style license and is hosted at SourceForge.net. This document provides the details and the insight on the internals of MARF and some of the mentioned applications.

Explore related subjects

cs.SDcs.CLcs.CVcs.MMcs.NE

Keep this discovery

BibTeXRIS

Serguei A. Mokhov, Stephen Sinclair, Ian Clément, Dimitrios Nicolacopoulos. 2009-05-08. The Modular Audio Recognition Framework (MARF) and its Applications: Scientific and Software Engineering Notes. https://doi.org/10.1007/978-1-4020-8741-7_84 10.1007/978-3-540-68825-9_21 10.1145/1370256.1370262

Cite the original work for its findings. Save a collection to share your selection of sources.

KEEP EXPLORING

Related papers

arXiv2026

Xiaomi-CocktailASR-1 Technical Report

Recently, large language model (LLM) based ASR models have achieved significant progress, yet they generally lack support for multi-speaker scenarios, where the cocktail party problem remains a critical bottleneck for further advancing ASR. Existing TS-ASR methods, including end-to-end architectures with speaker embeddings and latest LLM-based explorations suffer from degraded single-speaker performance and the inability to reject when the target speaker is absent. In this paper, we propose Xiaomi-CocktailASR-1, an LLM-based end-to-end TS-ASR architecture. By utilizing reference speech as voiceprint prompts, it directly transcribes the target speaker's speech without requiring speech separation. Xiaomi-CocktailASR-1 maintains competitive performance in single-speaker scenarios, comparable to mainstream ASR models. It also features a negative sample rejection capability, outputting empty text when the target speaker is absent from the mixed speech. Additionally, Xiaomi-CocktailASR-1 supports a Chain-of-Thought (CoT) reasoning mode to provide explicit reasoning steps. Extensive experiments on various synthetic and real-world multispeaker benchmarks demonstrate that Xiaomi-CocktailASR-1 achieves state-of-the-art performance, effectively addressing the cocktail party problem through a unified architecture that balances multispeaker and single-speaker recognition accuracy, along with rejection capability.

cs.SD↗
arXiv2026

EConv-TasNet: Efficient Conv-TasNet for Effective Speech Separation

Conv-TasNet has served as a strong baseline for time-domain speech separation, and many studies have extended it with advanced architectures such as dual-path networks, U-Nets, and attention mechanisms. However, these methods often introduce high computational cost and complexity, limiting their deployment in resource-constrained scenarios. To address this issue, we propose eConv-TasNet, an efficient variant of Conv-TasNet that improves both effectiveness and efficiency without relying on resource-intensive modules. The proposed model consists of a group-wise early-splitting (GES) module and a multi-group feature aggregation (MGFA) module. GES generates discriminative speaker embeddings at intermediate stages, while MGFA progressively aggregates these group-level representations for refined mask estimation. Experimental results show that eConv-TasNet reduces model size by 22.4%, accelerates inference by 18.9%, and improves SI-SNRi by 14.0%-28.0% across three public benchmarks. Moreover, it achieves competitive performance compared with state-of-the-art methods while requiring significantly fewer parameters and lower inference cost. These results demonstrate a favorable efficiency-effectiveness trade-off for edge deployment.

cs.SD↗