--- library_name: llima license: apache-2.0 pipeline_tag: automatic-speech-recognition base_model: openai/whisper-medium base_model_relation: quantized language: - en - zh - de - es - ru - ko - fr - ja - pt - tr - pl - ca - nl - ar - sv - it - id - hi - fi - vi - he - uk - el - ms - cs - ro - da - hu - ta - "no" - th - ur - hr - bg - lt - la - mi - ml - cy - sk - te - fa - lv - bn - sr - az - sl - kn - et - mk - br - eu - is - hy - ne - mn - bs - kk - sq - sw - gl - mr - pa - si - km - sn - yo - so - af - oc - ka - be - tg - sd - gu - am - yi - lo - uz - fo - ht - ps - tk - nn - mt - sa - lb - my - bo - tl - mg - as - tt - haw - ln - ha - ba - jw - su tags: - whisper - audio - automatic-speech-recognition - speech-translation - embedded - sima - modalix - a16w8 --- # Whisper Medium A16W8: Optimized for SiMa.ai Modalix ## Overview This repository contains a precompiled and quantized version of [OpenAI Whisper Medium](https://huggingface.co/openai/whisper-medium), optimized for execution on the **SiMa.ai Modalix** platform. | Property | Value | |---|---| | Model architecture | Whisper encoder-decoder Transformer | | Parameters | 769 million | | Quantization | A16W8: 16-bit activations, 8-bit weights | | Target | SiMa.ai Modalix | | Tasks | Multilingual transcription and speech translation to English | | Languages | 99 | | Maximum audio window | 30 seconds | | Maximum decoder positions | 448 | | Compatible Neat release | 0.4.0 | | Source model | [openai/whisper-medium](https://huggingface.co/openai/whisper-medium) | > This repository is not a standard Transformers checkpoint. The `elf_files/` > and `devkit/` directories contain compiled artifacts consumed by the SiMa.ai > Neat runtime on Modalix. ## Prerequisites To use this model, you need: 1. A **SiMa.ai Modalix device**. 2. The **SiMa.ai Neat Runtime** installed or updated on Modalix. 3. The **Hugging Face CLI**, optionally, for downloading the model on a host before copying it to Modalix. Follow the [SiMa.ai Neat getting started guide](https://developer.sima.ai/software/getting-started/) to install or update the Neat Runtime. ## Installation and Deployment ### Download the Model On Modalix, download the precompiled model with the LLiMa model manager: ```bash llima pull whisper-medium-a16w8 ``` The model is installed under: ```text /media/nvme/llima/models/whisper-medium-a16w8 ``` Alternatively, download it on a host and copy it to Modalix: ```bash hf download simaai/whisper-medium-a16w8 \ --local-dir whisper-medium-a16w8 scp -r whisper-medium-a16w8 \ sima@:/media/nvme/llima/models/ ``` Replace `` with the IP address or hostname of the Modalix device. ## Usage This model is intended for applications built with the SiMa.ai Neat GenAI APIs. Use `ASRModel` for direct, in-process transcription or `GenAIServer` when clients access the model over HTTP. ### Direct Application Integration The Neat `ASRModel` API provides the lowest-overhead path for speech transcription inside an application: ```cpp #include "neat/genai.h" #include int main() { simaai::neat::genai::ASRModel model( "/media/nvme/llima/models/whisper-medium-a16w8"); simaai::neat::genai::GenerationRequest request; request.audio_file = "speech.wav"; request.language = "en"; auto result = model.run(request); std::cout << result.text << "\n"; } ``` For the complete C++ and Python workflows, including audio tensors and streaming, see [GenAI Model](https://developer.sima.ai/software/develop-apps/development-workflow/genai-model). ### Serve with the Neat GenAI Server Use `GenAIServer` when a browser, service, or remote client needs to access the model through an HTTP API. The packaged Neat tutorial can serve this ASR model by itself: ```bash ./lib/sima-neat/tutorials/tutorial_021_serve_genai_models \ --asr /media/nvme/llima/models/whisper-medium-a16w8 ``` Verify that the model was registered: ```bash curl http://:9998/v1/models ``` Send an audio transcription request with the tutorial client: ```bash python3 share/sima-neat/tutorials/021_serve_genai_models/request_audio_transcription.py \ --server-ip \ --model asr \ speech.wav ``` See [Serve GenAI Models](https://developer.sima.ai/software/tutorials/serve-genai-models) for the complete C++, Python, and HTTP workflow. ## Artifact Layout ```text devkit/ Runtime configuration, tokenizer, and embeddings elf_files/ Compiled Modalix accelerator programs compile.sh Compilation entry point gen_models--openai--whisper.py Whisper compilation utility ``` The compilation scripts are included for provenance and reproducibility. They are not required when using the precompiled artifacts. ## Evaluation No accuracy or performance results specific to this A16W8 compiled artifact are currently reported. Results on the [source model card](https://huggingface.co/openai/whisper-medium) describe the unquantized upstream checkpoint and should not be treated as measurements of this build. ## Limitations - These artifacts run on SiMa.ai Modalix and cannot be loaded with `transformers.AutoModel`. - The runtime processes an audio window of up to 30 seconds. Segment longer recordings before inference. - This build produces text without word- or segment-level timestamps. - Speaker diarization is not provided. - Quantization can cause minor differences from the full-precision source model. - Whisper can hallucinate text during silence or noisy audio and performs unevenly across languages, accents, and domains. Evaluate the model on representative data before production deployment. ## Resources - [SiMa.ai Neat](https://developer.sima.ai/software/getting-started/) - [GenAI Model API](https://developer.sima.ai/software/develop-apps/development-workflow/genai-model) - [Serve GenAI Models](https://developer.sima.ai/software/tutorials/serve-genai-models) - [GenAI with LLiMa](https://developer.sima.ai/software/genai-llima/) - [OpenAI Whisper Medium](https://huggingface.co/openai/whisper-medium) - [Whisper paper](https://arxiv.org/abs/2212.04356) ## Citation ```bibtex @misc{radford2022whisper, doi = {10.48550/ARXIV.2212.04356}, url = {https://arxiv.org/abs/2212.04356}, author = {Radford, Alec and Kim, Jong Wook and Xu, Tao and Brockman, Greg and McLeavey, Christine and Sutskever, Ilya}, title = {Robust Speech Recognition via Large-Scale Weak Supervision}, publisher = {arXiv}, year = {2022} } ```