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nineninesix/
kani-tts-400m-0.3-pt

Name: kani-tts-400m-0.3-pt
Availability: InStock

NightPrince/MasriSpeech-Full

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KaniTTS Pretrain v0.3

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A high-speed, high-fidelity Text-to-Speech model optimized for real-time conversational AI applications.

Overview

KaniTTS uses a two-stage pipeline combining a large language model with an efficient audio codec for exceptional speed and audio quality. The architecture generates compressed token representations through a backbone LLM, then rapidly synthesizes waveforms via neural audio codec, achieving extremely low latency.

Key Specifications:

Model Size: 400M parameters
Sample Rate: 22kHz
Language: English, Chinese, Korean, Spanish, German, Japanese, German, Kyrgyz
License: Apache 2.0

Quickstart: Install from PyPI & Run Inference

It’s a lightweight so you can install, load a model, and speak in minutes. Designed for quick starts and simple workflows—no heavy setup, just pip install and run. More detailes...

Install

bash

pip install kani-tts
pip install -U "transformers==4.57.1" # for LFM2 !!!

Quick Start

python

from kani_tts import KaniTTS

model = KaniTTS('nineninesix/kani-tts-400m-0.3-pt')

# Generate audio from text
audio, text = model("Your text here")

# Save to file (requires soundfile)
model.save_audio(audio, "output.wav")

Working with Multi-Speaker Models

This model NOT support multiple speakers. But You can check if Your model supports speakers and select a specific voice:

python

from kani_tts import KaniTTS

model = KaniTTS('your_mode_repo/name')

# Check if model supports multiple speakers
print(f"Model type: {model.status}")  # 'singlspeaker' or 'multispeaker'

# Display available speakers (pretty formatted)
model.show_speakers()

# Or access the speaker list directly
print(model.speaker_list)

# Generate audio with a specific speaker
audio, text = model("Your text here", speaker_id="speaker_n")

Custom Configuration

python

from kani_tts import KaniTTS

model = KaniTTS(
    'nineninesix/kani-tts-400m-0.3-pt',
    temperature=0.7,           # Control randomness (default: 1.0)
    top_p=0.9,                 # Nucleus sampling (default: 0.95)
    max_new_tokens=2000,       # Max audio length (default: 1200)
    repetition_penalty=1.2,    # Prevent repetition (default: 1.1)
    suppress_logs=True,        # Suppress library logs (default: True)
    show_info=True,            # Show model info on init (default: True)
)

audio, text = model("Your text here")

Playing Audio in Jupyter Notebooks

You can listen to generated audio directly in Jupyter notebooks or IPython:

python

from kani_tts import KaniTTS
from IPython.display import Audio as aplay

model = KaniTTS('nineninesix/kani-tts-400m-0.3-pt')
audio, text = model("Your text here")

# Play audio in notebook
aplay(audio, rate=model.sample_rate)

Performance

On NovitaAI RTX 5090 using vLLM:

RTF: ~0.2 (5 times faster than realtime)
Memory: 16GB GPU VRAM used
Source Code: https://github.com/nineninesix-ai/kanitts-vllm

GPU Benchmark Results

GPU Model	VRAM	Cost ($/hr)	RTF
RTX 5090	32GB	$0.423	0.190
RTX 4080	16GB	$0.220	0.200
RTX 5060 Ti	16GB	$0.138	0.529
RTX 4060 Ti	16GB	$0.122	0.537
RTX 3060	12GB	$0.093	0.600

Datasets

Use Cases

Conversational AI: Real-time speech for chatbots and virtual assistants
Edge/Server Deployment: Resource-efficient inference on affordable hardware
Accessibility: Screen readers and language learning applications
Research: Fine-tuning for specific voices, accents, or emotions

Limitations

Performance degrades with inputs exceeding 15 seconds (need to use sliding window chunking)
Limited expressivity without fine-tuning for specific emotions
May inherit biases from training data in prosody or pronunciation
Optimized primarily for English; other languages may require additional training

Optimization Tips

Multilingual Performance: Continually pretrain on target language datasets and fine-tune NanoCodec
Batch Processing: Use batches of 8-16 for high-throughput scenarios
Hardware: Optimized for NVIDIA Blackwell architecture GPUs

Resources

Models:

Pretrained Model: https://huggingface.co/nineninesix/kani-tts-500m-0.3-pt
English: https://huggingface.co/nineninesix/kani-tts-400m-en
Chinese: https://huggingface.co/nineninesix/kani-tts-400m-zh
Korean: https://huggingface.co/nineninesix/kani-tts-400m-ko
German: https://huggingface.co/nineninesix/kani-tts-400m-de
Spanish: https://huggingface.co/nineninesix/kani-tts-400m-es
Arabic: https://huggingface.co/nineninesix/kani-tts-400m-ar
Japanese: https://huggingface.co/nineninesix/kani-tts-370m-expo2025-osaka-ja

Examples:

Space: https://huggingface.co/spaces/nineninesix/KaniTTS
OpenAI compatible API: https://github.com/nineninesix-ai/kanitts-vllm
Finetuning code pipeline: https://github.com/nineninesix-ai/KaniTTS-Finetune-pipeline
Dataset preparation pipeline: https://github.com/nineninesix-ai/nano-codec-dataset-pipeline
Example Dataset: https://huggingface.co/datasets/nineninesix/expresso-conversational-en-nano-codec-dataset
ComfyUI node: https://github.com/wildminder/ComfyUI-KaniTTS by WildAi
NextJS basic app: https://github.com/nineninesix-ai/open-audio. It uses the OpenAI npm package to connect to the OpenAI-compatible server API provided by kanitts-vllm.
GitHub Repository: https://github.com/nineninesix-ai/kani-tts

Links:

Website: https://www.nineninesix.ai
Contact Form: https://airtable.com/appX2G2TpoRk4M5Bf/pagO2xbIOjiwulPcP/form

Acknowledgments

Built on top of LiquidAI LFM2 350M as the backbone and Nvidia NanoCodec for audio processing.

Responsible Use

Prohibited activities include:

Illegal content or harmful, threatening, defamatory, or obscene material
Hate speech, harassment, or incitement of violence
Generating false or misleading information
Impersonating individuals without consent
Malicious activities such as spamming, phishing, or fraud

By using this model, you agree to comply with these restrictions and all applicable laws.

Contact

Have a question, feedback, or need support? Please fill out our contact form and we'll get back to you as soon as possible.

Citation

@inproceedings{emilialarge,
  author={He, Haorui and Shang, Zengqiang and Wang, Chaoren and Li, Xuyuan and Gu, Yicheng and Hua, Hua and Liu, Liwei and Yang, Chen and Li, Jiaqi and Shi, Peiyang and Wang, Yuancheng and Chen, Kai and Zhang, Pengyuan and Wu, Zhizheng},
  title={Emilia: A Large-Scale, Extensive, Multilingual, and Diverse Dataset for Speech Generation},
  booktitle={arXiv:2501.15907},
  year={2025}
}

@article{emonet_voice_2025,
  author={Schuhmann, Christoph and Kaczmarczyk, Robert and Rabby, Gollam and Friedrich, Felix and Kraus, Maurice and Nadi, Kourosh and Nguyen, Huu and Kersting, Kristian and Auer, Sören},
  title={EmoNet-Voice: A Fine-Grained, Expert-Verified Benchmark for Speech Emotion Detection},
  journal={arXiv preprint arXiv:2506.09827},
  year={2025}
}

@dataset{masrispeech_full,
  author       = {Yahya Muhammad Alnwsany},
  title        = {MasriSpeech-Full: Large-Scale Egyptian Arabic Speech Corpus},
  year         = {2025},
  publisher    = {Hugging Face},
  url          = {https://huggingface.co/collections/NightPrince/masrispeech-dataset-68594e59e46fd12c723f1544}
}