@rohanpaul_ai: An open multilingual ASR model just posted 3.54 average WER across 5 languages Hojo-ASR-Multi-V1 combines an acoustic e…

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Summary

Hojo-ASR-Multi-V1 开源多语言语音识别模型以 3.54 的平均 WER 横跨 5 种语言取得出色成绩,其采用声学编码器搭配 Qwen3-4B 大语言模型解码器的架构,已在 Hugging Face 上以 Apache 2.0 许可发布。

An open multilingual ASR model just posted 3.54 average WER across 5 languages Hojo-ASR-Multi-V1 combines an acoustic encoder with a Qwen3-4B decoder, giving speech recognition an unusually strong language model at the final decoding stage. - Built with a Qwen3-4B LLM decoder - Designed for multilingual conversational speech recognition - Publicly available on Hugging Face, under the Apache 2.0 license - Installable through the hojo-asr Python package - Supports local inference - Accepts WAV file paths, audio byte streams and SCP inputs There is a bigger technical story behind the 3.54 WER number. Traditional ASR systems rely heavily on acoustic and language modeling components designed specifically for speech, while Hojo hands the decoding stage to a Qwen3-4B LLM after the audio has been encoded and adapted. So recognition gets access to a much richer model of how words naturally fit together while it is deciding what was actually said. What makes Hojo even more interesting is that it is reaching that level across multiple languages rather than being built around only 1 language. And it does this with an openly available model that developers can actually run and inspect themselves. The architecture is probably the most interesting part for me. Hojo uses an Encoder, Adapter and LLM setup with Qwen3-4B as the decoder, so the final transcript is produced using both acoustic information from the speech and the language knowledge inside the LLM. That gives the decoder much more context for resolving words that may sound ambiguous from audio alone.
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An open multilingual ASR model just posted 3.54 average WER across 5 languages

Hojo-ASR-Multi-V1 combines an acoustic encoder with a Qwen3-4B decoder, giving speech recognition an unusually strong language model at the final decoding stage.

  • Built with a Qwen3-4B LLM decoder
  • Designed for multilingual conversational speech recognition
  • Publicly available on Hugging Face, under the Apache 2.0 license
  • Installable through the hojo-asr Python package
  • Supports local inference
  • Accepts WAV file paths, audio byte streams and SCP inputs

There is a bigger technical story behind the 3.54 WER number.

Traditional ASR systems rely heavily on acoustic and language modeling components designed specifically for speech, while Hojo hands the decoding stage to a Qwen3-4B LLM after the audio has been encoded and adapted. So recognition gets access to a much richer model of how words naturally fit together while it is deciding what was actually said.

What makes Hojo even more interesting is that it is reaching that level across multiple languages rather than being built around only 1 language. And it does this with an openly available model that developers can actually run and inspect themselves.

The architecture is probably the most interesting part for me.

Hojo uses an Encoder, Adapter and LLM setup with Qwen3-4B as the decoder, so the final transcript is produced using both acoustic information from the speech and the language knowledge inside the LLM.

That gives the decoder much more context for resolving words that may sound ambiguous from audio alone.

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Qwen-Audio-3.1-ASR 技术报告介绍了一系列音频语言模型,在语音识别、说话人分离转录和通用音频理解方面较 Qwen-Audio-3.0-ASR 有显著提升,支持 30 种语言和 16 种汉语方言,并提供流式与非流式等多种部署变体。

@MaxForAI: If you are working on voice agents, you should try this project. A team from NTU, NUS, and Shanghai AI Lab released: Mega-ASR. This fully open-source ASR is built on Qwen3-ASR, aiming to break the long-standing bottleneck of ASR performance in noisy, reverberant, or other impaired real-world environments...

X AI KOLs Timeline

NTU, NUS, and Shanghai AI Lab jointly released Mega-ASR, a fully open-source ASR model built on Qwen3-ASR. Using the Voices-in-the-Wild-2M dataset and progressive acoustic-to-semantic optimization, it achieves up to 30% relative Word Error Rate (WER) reduction in real-world noisy environments. With only 1.7B parameters, it enables efficient inference on consumer-grade hardware.