Audio-Text-to-Text
Transformers
Safetensors
English
Chinese
qwen2
text-generation
speech-language-model
streaming
audio
multimodal
qwen2.5-omni
text-generation-inference
Instructions to use zhifeixie/AudioInteraction with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use zhifeixie/AudioInteraction with Transformers:
# Load model directly from transformers import AutoTokenizer, AutoModelForCausalLM tokenizer = AutoTokenizer.from_pretrained("zhifeixie/AudioInteraction") model = AutoModelForCausalLM.from_pretrained("zhifeixie/AudioInteraction") - Notebooks
- Google Colab
- Kaggle
Create README.md
Browse files
README.md
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---
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license: apache-2.0
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---
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language:
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- en
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- zh
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license: apache-2.0
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pipeline_tag: audio-text-to-text
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tags:
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- speech-language-model
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- streaming
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- audio
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- multimodal
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- qwen2.5-omni
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datasets:
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- zhifeixie/StreamAudio-2M
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base_model:
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- Qwen/Qwen2.5-Omni-3B
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---
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# Mini-Omni3: Streaming Audio-In, Text-Out Conversational Model
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[**Code**](https://github.com/xzf-thu/Mini-Omni3) <!-- TODO: confirm repo URL -->
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Mini-Omni3 is a streaming speech-language model that listens to audio in real time and decides, at each audio chunk, whether to keep listening or to start replying with text. The model alternates between a **LISTENING** state, where it consumes one encoder-output chunk per step and emits either `KEEP_SILENCE` or `TEXT_BEGIN`, and a **SPEAKING** state, where it autoregressively generates a text turn until `TEXT_END` and then returns to listening for the next chunk.
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This design lets the model handle both spoken questions ("answer it") and ambient sounds ("decide based on the sound whether help is needed") within a single streaming session, without an external VAD or turn-taking heuristic.
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The release contains the Mini-Omni3 language-model weights (sharded safetensors), a chunk-wise audio encoder adapted from Qwen2.5-Omni, and the matching tokenizer and model config.
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## Model Details
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- **Model name:** Mini-Omni3
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- **Task:** Streaming audio-conditioned text generation (audio in, text out)
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- **Audio encoder:** Qwen2.5-Omni audio tower (chunk-wise)
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- **Audio framing:** 16 kHz, padded to 0.4-second (6400-sample) boundaries; 10 encoder-output frames per chunk
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- **Decoding states:** LISTENING (emits `KEEP_SILENCE` / `TEXT_BEGIN`) and SPEAKING (emits text until `TEXT_END`)
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- **Default sampling:** temperature 0.3, top-k 3
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- **Default max new tokens:** 4096 per session
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- **License:** Apache-2.0
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## Repository Contents
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```text
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Mini-Omni3/
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βββ MiniOmni3_LM_sharded/ # Sharded safetensors of the LM weights
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β βββ model.safetensors.index.json
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β βββ model-0000N-of-0000N.safetensors
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βββ MiniOmni3_ChunkwisedEncoder.pth # Audio encoder weights (Qwen2.5-Omni audio tower)
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βββ qwen_2_5_omni_config/ # Audio-encoder config (nested: thinker_config.audio_config)
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βββ model_config.yaml # GPT config consumed by Config.from_file
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βββ tokenizer.json # Tokenizer
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βββ README.md # This card
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```
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## Intended Use
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Mini-Omni3 is intended for streaming conversational agents that need to react to audio as it arrives β for example, voice assistants that may interject mid-utterance, alarms that respond to ambient sound, or low-latency dialogue systems where waiting for a full utterance before replying is too slow. The model is not a transcription system; it produces a conversational reply (or silence) rather than a verbatim transcript.
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## Quick Start
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### Installation
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```bash
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git clone https://github.com/xzf-thu/Mini-Omni3.git # TODO: confirm repo URL
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cd Mini-Omni3
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conda create -n mini-omni3 python=3.10 -y
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conda activate mini-omni3
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pip install -r requirements.txt
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```
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### Download the checkpoint
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```python
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from huggingface_hub import snapshot_download
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local_dir = snapshot_download(
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repo_id="zhifeixie/Mini-Omni3",
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repo_type="model",
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)
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print(local_dir)
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```
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### Python Usage
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```python
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from src.miniomni3.generate.run import run_inference
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run_inference(
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checkpoint_dir=local_dir, # the path snapshot_download returned
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audio_paths=["/path/to/audio.wav"], # offline mode: one round per path
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device="cuda:0", # or "mps" / "cpu"
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)
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```
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For interactive use, omit `audio_paths` and `run_inference` will prompt for an audio path each round:
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```python
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run_inference(checkpoint_dir=local_dir, rounds=5, device="cuda:0")
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```
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## Streaming Protocol
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A single session looks like:
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```text
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[system prompt tokens]
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ββββ LISTENING ββββ
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β AUDIO_BEGIN PAD*10 ASSISTANT β KEEP_SILENCE (keep listening)
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β AUDIO_BEGIN PAD*10 ASSISTANT β TEXT_BEGIN EMOTION (start replying)
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βββββββββββββββββββ
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ββββ SPEAKING βββββ
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β β¦ text tokens β¦ TEXT_END (reply finished)
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βββββββββββββββββββ
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ββββ LISTENING ββββ (next audio chunk)
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β¦
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```
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The model is trained to emit at most one `TEXT_BEGIN` per audio chunk. Each assistant turn begins with `TEXT_BEGIN`, followed by an emotion token, the reply tokens, and `TEXT_END`. Turns starting with `KEEP_SILENCE` indicate the model chose not to respond to that chunk.
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## Training Summary
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<!-- TODO: fill in once details are public.
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Suggested fields:
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- Pretraining base
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- SFT / instruction-tuning data
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- Streaming-objective data construction (how KEEP_SILENCE / TEXT_BEGIN supervision was generated)
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- Total tokens / hours of audio
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- Hardware and duration
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-->
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## Evaluation
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<!-- TODO: fill in once benchmarks are decided.
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Candidate metrics:
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- Spoken-QA accuracy on held-out audio prompts
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- False-trigger rate on ambient / non-speech audio (lower is better)
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- Response-onset latency in encoder chunks from end of question
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- Text quality of replies (e.g. GPT-judge or human preference)
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-->
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## Limitations
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- The model produces text, not speech. Pair it with a TTS system for end-to-end voice interaction.
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- Audio must be 16 kHz mono; non-conforming inputs are resampled by `whisper.load_audio` and padded to 0.4-second boundaries before encoding.
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- Decisions are made at 0.4-second granularity (one encoder chunk), which sets a floor on response-onset latency.
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- Trailing partial audio chunks shorter than 10 encoder frames are dropped before generation.
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## Citation
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<!-- TODO: replace with the real arxiv id and year once published. -->
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```bibtex
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@misc{xie_miniomni3,
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title = {Mini-Omni3: Streaming Audio-In, Text-Out Conversational Modeling},
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author = {Zhifei Xie and collaborators},
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year = {2026},
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note = {Preprint in preparation}
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}
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```
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## Acknowledgements
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Mini-Omni3 builds on the Qwen2.5-Omni audio encoder. We thank the Qwen team and the maintainers of OpenAI Whisper for the audio-loading utilities used in this project.
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