arXiv Computation and Language

O\'ido: Open-Vocabulary Speech Recognition on a $5 Microcontroller without a Neural Accelerator

O'ido is an inference engine that performs open‑vocabulary English speech recognition entirely on an ESP32‑S3 microcontroller, running a 13‑million‑parameter Conformer‑CTC model with int8 quantization and a 1.3‑million‑parameter language model. On LibriSpeech it achieves 3.70%/8.23% WER on test‑clean/test‑other, only 0.12 points below full precision, and outperforms the vendor’s recognizer and several laptop‑based tiny models. The firmware processes audio in about twice the duration of the input on a single core, with a streaming variant that delivers partial transcripts while speaking, though it is not yet real‑time. whyItMatters":"This work demonstrates that sophisticated, open‑vocabulary speech recognition can run on inexpensive, low‑power microcontrollers without dedicated neural accelerators, expanding the reach of voice interfaces to devices that previously could only support keyword spotting or fixed command sets."

arXiv AI
Oct 7

Offline AI Modules: Voice-First Offline Architecture, Hardware Reference Stack, Quantization and Benchmarking

The paper introduces the Offline AI Modules workstream, which provides a voice‑first offline architecture, a low‑cost hardware reference bill of materials, and a reproducible quantization and benchmarking pipeline for instruction‑tuned language models in the 2‑5B parameter range. It evaluates three models across four quantization formats on two hardware tiers—NVIDIA Jetson Orin NX and Raspberry Pi5—measuring deployment metrics and multilingual quality. The key result is that Q4_K_M quantization offers the best size‑to‑quality trade‑off, enabling high decode throughput and strong topic classification accuracy while staying within memory limits on both tiers.

By Sunday Afariogun, Odunolaoluwa Jenrola, Zeinab Nezami
arXiv Computation and Language
Oct 22

How Far Can the Gemma 4 Speech Encoder Go Without Changing Its Weights? Saudi and Gulf Arabic Speech Recognition with a Frozen and an Adapter-Augmented Encoder

The paper evaluates a 305 M‑parameter audio encoder shared by Gemma 4 and EmbeddingGemma 2 for Saudi and Gulf Arabic speech recognition without altering its weights. Two approaches are compared: a frozen system that trains only a 22.6 M‑parameter CTC reader, and an adapted system that adds 3.1 M low‑rank adapters, a 41.5 M‑parameter hybrid CTC/attention reader, and joint decoding with a sub‑word language model. The adapted system matches or surpasses a 0.6 B‑parameter fine‑tuned recogniser on several datasets, while the frozen system remains competitive in character error rate and runs three‑quarters faster on CPU with greedy decoding.

By Amr Abdelkhalik
arXiv AI
Oct 5

Refinement Buys Intelligibility, Search Buys Identity: What Test-Time Compute Buys in Masked-Diffusion TTS

The study investigates how two computational dimensions—model depth and refinement steps—affect intelligibility and speaker identity in masked-diffusion text‑to‑speech systems. Experiments with 15 models (19–133 M parameters) and up to 16 refinement steps show that refinement improves intelligibility more than identity, with a 1.86× asymmetry that persists even after retraining. Best‑of‑K search can recover identity when refinement fails, and analysis indicates that depth and steps target distinct bottlenecks, requiring separate optimization.

By Nityanand Mathur, Hamees Sayed, Ayush Pratap Singh
arXiv Computation and Language
Oct 23

WaxalNet: Benchmarking and Adapting Speech Recognition for Spontaneous Speech in 19 African Languages

The paper introduces WaxalNet, a benchmark and adaptation study for spontaneous speech recognition across 19 African languages using the WAXAL corpus. It evaluates zero‑shot and fine‑tuned performance of several multilingual foundation models (Whisper, MMS, Omnilingual ASR, SeamlessM4T‑v2), showing that fine‑tuning can substantially reduce word error rates and that LLM decoders generally outperform CTC decoders when the encoder is shared. The authors also demonstrate that a single multilingual model can serve all languages with only a modest WER penalty and provide a native‑speaker audit of failure modes.

By Victor Tolulope Olufemi, Oreoluwa Babatunde, Ramsey Njema, Bolarinwa Gbotemi, Wanchi Lucia Yen, John Uzodinma, Sunday Ajayi, Oluwademilade Williams, Kausar Moshood, Innocent Elendu Anyaele, Akebert Arefaine, Candace Hunzwi, Wongel Dawit Daniel, Emmilly Namuganga, Cleophas Kadima, Athanase Bahizire, Onitsiky Ranaivoson, Emmanuel Aaron, Nicholaus Ladislaus, Idris Muhammed, Jonathan Enoch Simenya, Martin Koome, Matewos Tegete Endaylalu, Peter Ifeoluwa Adeyemo, Hondi Prisca Birindwa, Ukachi Agnes Eze-Mbey, Yacoba Oduro-Yeboah, Pericles Adjovi, Mikel K. Ngueajio, Toluwani Aremu, Prasenjit Mitra
arXiv Computation and Language
Sep 24

Text Scores Can Miss Waveform Use: A Qwen2-Audio Quantization Case Study

The paper presents a new evaluation protocol for post‑training quantization of speech language models that separates lexical output, transcript‑insufficient endpoints, and packed implementations. In a Qwen2‑Audio case study, a 6‑bit allocation selected for translation improves chrF scores but degrades emotion recognition, while uniform and front‑layer controls perform better on emotion tasks. Similar patterns hold at 7 bits, and a 4‑bit study shows consistent emotion deficits across all low‑bit allocations, with no advantage for the selected scheme. The study highlights a precision‑dependent mismatch between lexical output, waveform‑dependent behavior, and nominal precision, without claiming a general failure of low‑bit models or a deployment benefit for the selected allocation.

By Mengzhe Geng, Jinxi Jin, Junhao Xu
arXiv Computation and Language
2d ago

InteractionBench: Measuring Timing, Task Accuracy, Whisper and Volume, and Grounding in Real-Time Voice Systems

InteractionBench is a benchmark that evaluates real‑time voice systems on four dimensions—conversational timing, spoken‑task accuracy, whispering and volume, and groundedness—using the same recorded user audio across thirteen scenarios. It tests seven systems (GPT‑Live, GPT‑Realtime, Gemini Live, Moshi, PersonaPlex, Freeze‑Omni, and a turn‑based cascade) over 1,519 runs, revealing correlations between timing and task accuracy and highlighting differences in whispering, grammar correction, and groundedness performance. The study also notes protocol variations and the impact of network delay on timing scores.

By Richard Yucheng He, Chen Xu, Yihang Liu, Tairan Chen, Baodong Cao