stable-codec-speech-16k-base
is a Transformer-based codec model designed for high-quality, low-bitrate audio coding. It processes audio waveforms by encoding them into discrete tokens, which can later be decoded back into the original audio waveform.
IMPORTANT: This is the base version of the model without CTC finetuning, provided for reproducibility of results in the
arXiv paper
. For 99% of usecases,
stable-codec-speech-16k
is a better choice.
Please note: For individuals or organizations generating annual revenue of US $1,000,000 (or local currency equivalent) or more, regardless of the source of that revenue, you must obtain an enterprise commercial license directly from Stability AI before commercially using Stable Codec, any derivative work of Stable Codec (such as a “fine tune” model), or their outputs. You may submit a request for an Enterprise License at
https://stability.ai/enterprise
. Please refer to Stability AI's Community License, available at
https://stability.ai/license
, for more information.
Model details
: This released model is a speech codec designed to compress real-world speech data into a suitable format for generative modeling. It provides a foundational tool for developing downstream applications in speech understanding and generation, such as text-to-speech systems and conversational AI models.
Please check our
arXiv page
and
Github repo
for details.
License
Community License:
Free for research, non-commercial, and commercial use by organizations and individuals generating annual revenue of US $1,000,000 (or local currency equivalent) or more, regardless of the source of that revenue. If your annual revenue exceeds US $1M, any commercial use of this model or derivative works thereof requires obtaining an Enterprise License directly from Stability AI. You may submit a request for an Enterprise License at
https://stability.ai/enterprise
. Please refer to Stability AI's Community License, available at
https://stability.ai/license
, for more information.
Intended uses include the following:
Efficient compression of speech signals for storage or streaming purposes.
Enhancing speech-based applications, such as telecommunication systems and real-time communication platforms.
Research and development in audio coding and speech synthesis, including understanding and improving codec performance.
Development of downstream applications including speech recognition and generation.
This model is purely trained on non-overlapping clean English speech, and exhibits optimal performance in these situations. It is not suitable for applications requiring high-fidelity music or environmental sound coding.
Contact
Please report any issues with the model or contact us:
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