Papers for

call center software teams

Papers whose findings have a practical use for this group, as judged from the abstract. Open a paper to read what it means in practice.

Steerable full duplex speech models improve conversational control and timing

SteerDuplex: Steerable Duplex Speech Dialogue Models

Abstract: Full-duplex spoken dialogue models support low-latency turn taking, interruption handling, and backchanneling, yet a key capability remains underexplored: steerability, the ability to reliably shift conversational behavior along attributes such as tone, persona, speaking rate, and voice style in response to user instructions. We introduce a taxonomy of text- and audio-based steerability that identifies substantial gaps in current full-duplex models. To address this gap, we introduce SteerDuplex, a Moshi-based full-duplex speech model fine-tuned on natural conversations and synthetic dialogues targeting instruction following, vocal delivery, reasoning, and duplex interaction. We further apply two-stage reinforcement learning (RL) with hybrid rewards, combining verifiable interaction checks and judge-based semantic feedback to improve timing and response continuity. To evaluate full-duplex spoken steerability, we introduce SteerBench, a benchmark with 390 spoken prompts and 1,067 human-authored binary audio and text rubrics spanning tone, persona, style/accent, and speed/length. On SteerBench, supervised training improves audio-steering average pass rate by 44.5 percentage points over the strongest evaluated open baseline. On Audio MultiChallenge, task average pass rate improves by 7 points over its strongest evaluated open baseline. RL further raises source-clean interruption response from 72.5% to 82.5% and reduces synthetic pause barge-in from 26.5% to 9%. Steering and aggregate task scores remain comparable or higher, while reward probes reveal reward hacking through incomplete responses. Our model and benchmark support systematic research on spoken steerability, with reward analysis showing why timing gains must be evaluated alongside response completeness.

Fri 11 SeptArtificial IntelligenceComputation and Language
The gist
People want voice assistants and dialogue systems to talk more naturally and follow instructions about how to speak, like changing tone or speed. The authors found that current full-duplex speech models, which can listen and talk at the same time, were missing this ability to change their speaking style reliably. They created SteerDuplex, a model trained to adjust conversation style and timing based on user instructions, and tested it on a new benchmark called SteerBench. Their model showed big improvements in controlling voice style and handling turn-taking smoothly, though some issues remain with incomplete responses.
Open 2609.12623v1

X2-NativeCursor improves real-time text progress tracking in streaming speech

X2-NativeCursor: Native-Token Text Progress Tracking for Incremental-Text Streaming Codec TTS

Abstract: Incremental-text streaming text-to-speech (TTS) needs online text progress tracking for synchronized highlighting, interruption handling, and dialogue-history updates. Input text arrives before it is spoken, so text arrival alone cannot indicate speech progress. Existing waveform-based alignment requires complete audio or adds acoustic processing during streaming. We propose X2-NativeCursor, a lightweight observer that tracks progress from native speech tokens before waveform decoding without changing the TTS generator. Its normalization plan links spoken labels to their original-text spans. Text and native-token encoders feed a local matcher that estimates the current label position. A separate output rule converts revisable position estimates into a cursor that never moves backward. Mean absolute error against an automatic reference is 0.151 Chinese characters with 80-ms lookahead, versus 1.253 characters with 320-ms lookahead for an online waveform baseline. Alignment real-time factor also decreases from 0.3598 to 0.0180 relative to this baseline. Lower tracking error is retained under a second automatic alignment reference. We evaluate X2-NativeCursor on Qwen3-TTS and validate its adaptation to CosyVoice2 by training a separate observer for each backbone. Code is publicly available at https://github.com/X-Square-Robot/X2Streaming-TTS.

Wed 9 SeptComputation and Language
The gist
It’s hard for speech software to keep track of which word is being spoken in real time, especially when the text arrives before the speech is produced. The authors created X2-NativeCursor, a lightweight tool that follows the speech progress by looking at internal speech tokens instead of waiting for audio output. This method tracks spoken words more accurately and faster without changing the main speech generator. It works well for Chinese and adapts to different speech models.
Open 2609.09677v1

Audio visual system improves dialogue clarity in noisy speech

Noise Adaptive Streaming Audio-Visual Speech Token Enhancement for Robust Full-Duplex Spoken Dialogue Models

Abstract: Full-duplex spoken dialogue systems enable simultaneous listening and speaking, but their audio-only perception often fails under background noise and overlapping speech, leading to incoherent responses. Recent audio-visual dialogue approaches show that incorporating visual cues such as lip movements improve robustness under audio corruption. However, existing approaches often adapt the large speech dialogue model itself to process visual input, requiring costly multimodal training. We propose AV-STE, a modular streaming audio-visual front-end that restores corrupted semantic speech tokens from noisy audio and lip video before they reach the speech LLM. The downstream dialogue model remains entirely frozen, preserving its pretrained conversational capabilities. When integrated with frozen Moshi, AV-STE improves average GPT-4o-judged response coherence from 1.42 to 1.91 under same-dataset speaker interference while largely preserving turn-taking behavior. Gains also transfer to out-of-domain Seamless Interaction.

Tue 8 SeptSoundArtificial IntelligenceHuman-Computer Interaction
The gist
Spoken dialogue systems that listen and talk at the same time often get confused when there is background noise or multiple people talking. To fix this, the authors created a special tool called AV-STE that uses both sound and lip reading from video to clean up what is being said before sending it to the conversation brain. This way, the main dialogue system stays the same and still understands and talks well. Their method made the conversation clearer and easier to follow when people were talking over each other or in noisy places.
Open 2609.08390v1