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PersonaPlex: Voice and role control for full duplex conversational speech models
➡️ Code: nvidia/personaplex
➡️ Demo: PersonaPlex Project Page
➡️ Paper: Coming Soon
Description:
Personaplex is a real-time speech-to-speech conversational model that jointly performs streaming speech understanding and speech generation. The model operates on continuous audio encoded with a neural codec and predicts both text tokens and audio tokens autoregressively to produce its spoken responses. Incoming user audio is incrementally encoded and fed to the model while Personaplex simultaneously generates its own outgoing speech, enabling natural conversational dynamics such as interruptions, barge-ins, overlaps, and rapid turn-taking. Personaplex runs in a dual-stream configuration in which listening and speaking occur concurrently. This design allows the model to update its internal state based on the user’s ongoing speech while still producing fluent output audio, supporting highly interactive conversations. Before the conversation begins, Personaplex is conditioned on two prompts: a voice prompt and a text prompt. The voice prompt consists of a sequence of audio tokens that establish the target vocal characteristics and speaking style. The text prompt specifies persona attributes such as role, background, and scenario context. Together, these prompts define the model's conversational identity and guide its linguistic and acoustic behavior throughout the interaction.
This model is ready for commercial use.
License/Terms of Use:
GOVERNING TERMS: Use of this model is governed by the NVIDIA Open Model License Agreement. ADDITIONAL INFORMATION: CC-BY-4.0.
Use Case:
Wherever NVIDIA’s speech-to-speech conversational models are used, PersonaPlex can generate English speech response for English speech input.
Deployment Geography:
Global
Release Date:
Hugging Face [01/15/2026] via [https://huggingface.co/nvidia/personaplex-7b-v1]
Github [01/15/2026] via [https://github.com/NVIDIA/personaplex]
Model Architecture:
Architecture Type: Transformer
Network Architecture: Moshi
Moshi uses:
- Mimi Speech Encoder (ConvNet, Transformer)
- Moshi Temporal Transformer + Depth Transformer
- Mimi Speech Decoder (Transformer, ConvNet)
** This model was developed based on Moshi (Moshiko weights)
** Number of model parameters: 7B
Input(s):
Input Type(s): Text (prompt), Audio (user speech)
Input Format: String, WAV/WebAudio
Input Parameters: One-Dimensional (1D)
Other Properties Related to Input: 24kHz sample rate for audio.
Output(s)
Output Type(s): Text (agent text), Audio (agent speech)
Output Format: String, WAV/WebAudio
Output Parameters: One-Dimensional (1D)
Other Properties Related to Output: 24kHz sample rate for audio.
Our AI models are designed and/or optimized to run on NVIDIA GPU-accelerated systems. By leveraging NVIDIA’s hardware (e.g. GPU cores) and software frameworks (e.g., CUDA libraries), the model achieves faster training and inference times compared to CPU-only solutions.
Software Integration:
Runtime Engine: PyTorch
Supported Hardware Microarchitecture Compatibility:
- NVIDIA Ampere (A100)
- NVIDIA Hopper (H100)
Preferred/Supported Operating System(s):
- Linux
The integration of foundation and fine-tuned models into AI systems requires additional testing using use-case-specific data to ensure safe and effective deployment. Following the V-model methodology, iterative testing and validation at both unit and system levels are essential to mitigate risks, meet technical and functional requirements, and ensure compliance with safety and ethical standards before deployment.
Model Version(s):
- v1.0
Training, Testing, and Evaluation Datasets:
Training Dataset:
Link: Fisher English: Part1, Part2
Data Modality: Audio (speech)
Audio Training Data Size: Less than 10,000 Hours
Data Collection Method by dataset: Human
Labeling Method by dataset: Automated
Properties: 7303 conversations (upto 10 minutes each).
Testing/Evaluation Dataset:
Link: FullDuplexBench
Data Collection Method by dataset: Hybrid: Human, Synthetic, Automated.
Labeling Method by dataset: Automated.
Properties: The FullDuplexBench public benchmark aggregates various synthetic and real datasets.
Additionally speaker similarity (SSIM) between voice prompts and model outputs on the User Interruption portion of the FullDuplexBench benchmark were measured using WavLM-TDNN embedding cosine similarity.
FullDuplexBench Benchmark Scores:
| Metric | Value |
|---|---|
| Pause Handling(Synthetic): TOR↓ | 0.358 |
| Pause Handling(Candor): TOR↓ | 0.431 |
| Backchannel: TOR↓ | 0.273 |
| Backchannel: Freq↑ | 0.042 |
| Backchannel: JSD↓ | 0.662 |
| Smooth Turn Taking: TOR↑ | 0.908 |
| Smooth Turn Taking: Latency↓ | 0.170 |
| User Interruption: TOR↑ | 0.950 |
| User Interruption: GPT-4o↑ | 4.290 |
| User Interruption: Latency↓ | 0.240 |
| User Interruption: SSIM(WavLM)↑ | 0.650 |
Comparison With Other Conversational AI Systems: PersonaPlex outperforms other open-source and commercial systems on conversational dynamics, response and interruption latency, and task adherence in both question-answering assistant and customer service roles.
Inference:
Acceleration Engine: PyTorch
Test Hardware: NVIDIA A100 80 GB
Ethical Considerations:
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kyutai/moshiko-pytorch-bf16