lspr98/conformer-stt-s3 — explained in plain English
Analysis updated 2026-05-18
Build a voice controlled device that transcribes speech entirely on the microcontroller, with no internet needed.
Add offline speech recognition to a battery powered gadget for privacy or areas without network coverage.
Benchmark on device speech recognition accuracy and speed across different ESP32S3 boards.
| lspr98/conformer-stt-s3 | gsalvadoi/guinxu-engine | burakcan/meshcore-mishmesh | |
|---|---|---|---|
| Stars | 48 | 48 | 47 |
| Language | C++ | C++ | C++ |
| Setup difficulty | hard | moderate | moderate |
| Complexity | 5/5 | 3/5 | 3/5 |
| Audience | developer | developer | general |
Figures from each repo's GitHub metadata at analysis time.
Requires a specific ESP32S3 board with enough SRAM, PSRAM, and flash, plus a wired I2S microphone.
conformer-stt-s3 brings automatic speech recognition, meaning software that turns spoken audio into written text, onto a small and inexpensive microcontroller called the ESP32S3, rather than requiring a powerful server or cloud service. It works by shrinking a 13 million parameter speech recognition model originally built by Nvidia down to a smaller, compressed version that can fit and run directly on this tiny chip, transcribing English speech into lower case text. This makes it useful for voice controlled devices such as smart gadgets or appliances. Running the model on such small hardware is naturally slower than running it on a full sized server, but the tradeoff brings real benefits. The device uses much less power, which matters for gadgets that sit idle listening for speech most of the time. Since the audio is processed entirely on the device itself and never sent anywhere else, it offers strong privacy, and it also allows battery powered devices to work in places with no internet connection at all. The README reports the model's accuracy across several standard speech testing datasets, with error rates ranging from about 5.6 percent on the cleanest dataset up to around 21.4 percent on a harder, noisier one. It also reports timing measurements on two specific development boards, showing that transcribing several seconds of audio can take anywhere from about one second up to around thirteen seconds depending on how much audio is being processed at once. Because the model cannot listen and transcribe in true real time, the project includes a simple voice activity detector that decides which chunks of incoming audio are worth sending to the model at all, based on how loud each chunk is compared to a configurable threshold. To build and run this project you need a compatible ESP32S3 development board with specific amounts of memory, a wired I2S microphone such as the INMP441, and Espressif's own ESP-IDF development toolchain, which the author recommends running inside a container for consistency.
Runs a compressed speech recognition model directly on a small ESP32S3 microcontroller, transcribing English speech without the cloud.
Mainly C++. The stack also includes C++, ESP-IDF, ESP32-S3.
Setup difficulty is rated hard, with roughly 1h+ to a first successful run.
Mainly developer.
This repo across BitVibe Labs
Verify against the repo before relying on details.