Huashu Design
alchaincyf · Skill
Turns a one-line request in your AI agent into a finished design: a launch animation, a clickable app prototype, an editable slide deck, an infographic or a se…
Turns Japanese speech into text with a 615.8M speech recognition model built for speed and accuracy. For developers and teams who transcribe Japanese audio on a Linux machine with an NVIDIA GPU.
Haas-ja-v1.0 is a kit in the Productivity category, listed by Mindket. It is free. This is version 1, last updated on . Download it on this page, a zip of its files. Nothing in it runs by itself.
pip install -U huggingface_hub hf download third-intelligence/Haas-ja-v1.0
Haas-ja-v1.0 turns spoken Japanese into text, with speed and accuracy as its two goals. It builds on NVIDIA's Parakeet-TDT design and swaps in Kimi Delta Attention in its prediction network. The card reports a strict CER of 6.12% on a benchmark made from five public Japanese datasets.
It suits developers and teams who need Japanese audio turned into text at speed.
| Industry | Who uses it | How they use it | Example |
|---|---|---|---|
| Media | Video teams | Transcribe Japanese interviews and clips | A script from a recorded interview |
| Business | Operations teams | Turn Japanese meetings into text | Notes from a recorded call |
| Research | Speech researchers | Compare Japanese recognizers | A test on the card's five-dataset benchmark |
People who work in other languages, or on a Mac or a machine without an NVIDIA GPU, can skip it.
| Use case | Who | What they get back |
|---|---|---|
| Short clips | Developers | The Japanese text of a recording |
| Long recordings | Operations teams | Text from audio over 30 seconds, split at quiet points |
| Names and terms | Editors | A custom vocabulary so rare names are written as you want |
You need these first:
pip install -U huggingface_hub hf download third-intelligence/Haas-ja-v1.0
Load the model, load an audio file, and call transcribe. For recordings longer than 30 seconds use the long-form call, which splits the audio at quiet points and works through the pieces in batches.
import haas_asr
model = haas_asr.load_model("third-intelligence/Haas-ja-v1.0")
audio = haas_asr.load_audio("example.wav")
output = model.transcribe([audio])
print(output[0].text)audio = haas_asr.load_audio("example.flac")
text = haas_asr.transcribe_longform(model, audio, batch_size=64)| Tool | What it does |
|---|---|
| haas_asr.load_model | Returns the model as a standard NeMo object, set to bfloat16 on the GPU |
| haas_asr.load_audio | Reads WAV, FLAC, MP3 or OGG, makes it mono and resamples it to 16 kHz |
| model.transcribe | Transcribes a list of audio clips |
| haas_asr.transcribe_longform | Splits long audio at quiet points and transcribes it in batches |
You load a recording of a meeting in Japanese and call transcribe on it. The model returns the spoken Japanese as text, and printing the first result shows it. A recording over 30 seconds goes through the long-form call instead.
Speech recognition turns spoken words into written text. A first small thing to try is one short Japanese WAV file and the four lines of the first snippet.
For experts, the stack pairs a FastConformer encoder with a TDT decoder, as in Parakeet-TDT. The decoder also predicts how long each token lasts, so it can skip several encoder frames at a time. Kimi Delta Attention replaces the LSTM in the prediction network, which the card says raises accuracy at almost the same speed. The cost is a hard dependency on Triton and CUDA.
Turns Japanese speech into text with a 615.8M speech recognition model built for speed and accuracy. For developers and teams who transcribe Japanese audio on a Linux machine with an NVIDIA GPU.
Made by third-intelligence on Hugging Face. These notes are Mindket's, written from its model page and model card; the model's own files and license are on Hugging Face.
What it is: Haas-ja-v1.0 turns spoken Japanese into text, with speed and accuracy as its two goals. It builds on NVIDIA's Parakeet-TDT design and swaps in Kimi Delta Attention in its prediction network. The card reports a strict CER of 6.12% on a benchmark made from five public Japanese datasets.
Who it suits: It suits developers and teams who need Japanese audio turned into text at speed.
Industry | Who uses it | How they use it | Example
Media | Video teams | Transcribe Japanese interviews and clips | A script from a recorded interview
Business | Operations teams | Turn Japanese meetings into text | Notes from a recorded call
Research | Speech researchers | Compare Japanese recognizers | A test on the card's five-dataset benchmark
People who work in other languages, or on a Mac or a machine without an NVIDIA GPU, can skip it.
Where to use it:
Use case | Who | What they get back
Short clips | Developers | The Japanese text of a recording
Long recordings | Operations teams | Text from audio over 30 seconds, split at quiet points
Names and terms | Editors | A custom vocabulary so rare names are written as you want
How to set it up: You need these first:
pip install -U huggingface_hub
hf download third-intelligence/Haas-ja-v1.0
How to use it: Load the model, load an audio file, and call transcribe. For recordings longer than 30 seconds use the long-form call, which splits the audio at quiet points and works through the pieces in batches.
import haas_asr
model = haas_asr.load_model("third-intelligence/Haas-ja-v1.0")
audio = haas_asr.load_audio("example.wav")
output = model.transcribe([audio])
print(output[0].text)
audio = haas_asr.load_audio("example.flac")
text = haas_asr.transcribe_longform(model, audio, batch_size=64)
Tool | What it does
haas_asr.load_model | Returns the model as a standard NeMo object, set to bfloat16 on the GPU
haas_asr.load_audio | Reads WAV, FLAC, MP3 or OGG, makes it mono and resamples it to 16 kHz
model.transcribe | Transcribes a list of audio clips
haas_asr.transcribe_longform | Splits long audio at quiet points and transcribes it in batches
Example: You load a recording of a meeting in Japanese and call transcribe on it. The model returns the spoken Japanese as text, and printing the first result shows it. A recording over 30 seconds goes through the long-form call instead.
For newcomers and experts: Speech recognition turns spoken words into written text. A first small thing to try is one short Japanese WAV file and the four lines of the first snippet.
For experts, the stack pairs a FastConformer encoder with a TDT decoder, as in Parakeet-TDT. The decoder also predicts how long each token lasts, so it can skip several encoder frames at a time. Kimi Delta Attention replaces the LSTM in the prediction network, which the card says raises accuracy at almost the same speed. The cost is a hard dependency on Triton and CUDA.
Load a Japanese audio file
It is made mono and resampled to 16 kHz
The encoder reads the audio frames
The decoder predicts tokens and durations
You get the Japanese text
Good to know:
Title: Parameters by part, in millions
Encoder: 608.9 M
Prediction network: 3.9 M
Joint network: 3.0 M
Download its files with Hugging Face's command line:
pip install -U huggingface_hub
hf download third-intelligence/Haas-ja-v1.0
Then follow the usage in its model card. Nothing in this kit runs by itself.
README.md 5 KB
Listed Hugging Face models in the same category that share the most tags with this one.
| Product | Downloads, last 30 days | Likes | Parameters | License | Last change |
|---|---|---|---|---|---|
| Haas-ja-v1.0 This page | 3,665 | 77 | — | apache-2.0 | |
| OrukeetShares fastconformer · nemo · automatic-speech-recognition | 35,262 | 110 | 627M | cc-by-sa-4.0 | |
| parakeet-ultra-ggufShares conformer · automatic-speech-recognition | 1,375,499 | 8 | 627M | cc-by-4.0 | |
| Ming-3.0-audio-tinyShares speech · audio · automatic-speech-recognition | 5,202 | 62 | 8.9B | mit | |
| orukqwen-asr-1.7bShares speech · audio · automatic-speech-recognition | 222 | 6 | 2B | apache-2.0 | |
| MOSS-Transcribe-DiarizeShares speech · audio | 121,031 | 520 | 909M | apache-2.0 |
Posts and comments that name this project on Hacker News, Bluesky, X, Reddit, Facebook and YouTube, found by a weekly search and on Trendshift. Mindket doesn’t write or check them. Open one to read it where it was posted.
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