Downloads · 30 days
119
37% of all-time downloads
moonshine-ai/moonshine-streaming-tiny-ja
moonshine-streaming-tiny-ja is a automatic speech recognition model from moonshine-ai. Use it when you need speech turned into text. It is set up for transformers. The card lists the license as mit.
Japanese streaming speech recognition, 27.0M parameters. Same architecture as moonshine-ai/moonshine-streaming-tiny, trained for Japanese with a 12,288-entry Japanese tokenizer.
Downloads · 30 days
119
37% of all-time downloads
All-time downloads
321
Public
Parameters
27M
108 MB on disk
Likes
2
Public
Click a slice to open those files.
.safetensors108 MB · 100%
From the Hugging Face model README
Japanese streaming speech recognition, 27.0M parameters. Same architecture as moonshine-ai/moonshine-streaming-tiny, trained for Japanese with a 12,288-entry Japanese tokenizer.
Moonshine Streaming pairs a 50 Hz time-domain audio frontend with a sliding-window Transformer encoder, so it transcribes incrementally rather than waiting for an utterance to finish. It is intended for on-device use on edge-class hardware.
This repository is a conversion of a specific training checkpoint, recorded here because the training run that produced it was still in progress when this snapshot was taken and a better one may replace it:
| Checkpoint | ja12k_tiny_stageC_best.safetensors |
| Stage | C (read-speech mix) |
| Architecture | slinkier_prime_adapted |
| Tokenizer | tokenizer_ja12k.json, vocab 12,288 |
| Snapshot taken | 2026-08-23 |
| Tensors / parameters | 163 / 27.0M |
If you need reproducibility, pin the revision of this repository rather than
tracking main.
pip install --upgrade transformers datasets[audio]
from transformers import MoonshineStreamingForConditionalGeneration, AutoProcessor
import torch
model = MoonshineStreamingForConditionalGeneration.from_pretrained(
"moonshine-ai/moonshine-streaming-tiny-ja"
).eval()
processor = AutoProcessor.from_pretrained("moonshine-ai/moonshine-streaming-tiny-ja")
inputs = processor(audio, return_tensors="pt", sampling_rate=16000)
# Cap the output length. Like other seq2seq ASR models this one can fall into a
# repetition loop, and short or noisy clips are where it happens.
seq_lens = inputs.attention_mask.sum(dim=-1)
max_new_tokens = int((seq_lens * 6.5 / 16000).max().item()) + 2
generated = model.generate(**inputs, max_new_tokens=max_new_tokens)
print(processor.batch_decode(generated, skip_special_tokens=True)[0])
Pass the attention_mask. The encoder applies its per-layer sliding windows
only when it is given one; called without a mask it attends over the whole
utterance instead, which is a different model from the one that was trained. The
processor returns the mask, so the snippet above is the safe form. The processor
also pads audio to a whole number of 80-sample frames, which the frontend
requires.
| Encoder | 6 layers, width 320, 8 heads, sliding windows (16, 4) on the first two and last two layers and (16, 0) between |
| Decoder | 6 layers, width 320, 8 heads, RoPE over 32 of each head's 40 dimensions |
| Frontend | 50 Hz features, CMVN, asinh compression, two causal stride-2 convolutions |
| Adapter | learned absolute positional embeddings before the decoder |
The lookahead layers give roughly 80 ms of lookahead; the intermediate layers have none.
Trained on a large-scale automatically labeled Japanese corpus, plus a read-speech mix in the final stage:
The podcast and YouTube transcripts are pseudo-labels: they were produced by running a Whisper-family teacher model over crawled audio, not by human transcription. The model therefore inherits the teacher's error modes, including its handling of proper nouns, numerals and code-switching, and its transcription conventions for a language written without spaces. No human-verified transcript was used for the bulk of training.
Japanese is scored on character error rate with spaces removed
(cer_nospace), never WER. Japanese is written without spaces, so tokenization
differences alone can read as several hundred percent WER while the characters
are correct.
suite_ja is FLEURS Japanese (650 utterances) and ReazonSpeech Japanese (5,263).
| Panel | CER |
|---|---|
fleurs_ja | 11.50 |
reazonspeech_ja | 26.73 |
| macro | 19.115 |
Batch 1 is the honest number for deployment. Batched evaluation zero-pads short clips up to the longest in the batch, and that trailing silence flatters the model by more than a point on spontaneous speech.
| Panel | CER |
|---|---|
fleurs_ja | 11.62 |
reazonspeech_ja | 27.77 |
| macro | 19.70 |
These weights were converted from the neo training checkpoint, and the
conversion was checked by measurement rather than inspection: same seeded sample,
same batch size, same normalizer.
fleurs_ja | reazonspeech_ja | macro | |
|---|---|---|---|
| Training checkpoint | 11.62 | 27.77 | 19.699 |
| This repository | 11.35 | 28.08 | 19.712 |
397/400 and 385/400 transcripts are byte-identical. The residual comes from the frontend's 80-sample frame alignment, which this path pads and the training path does not.
Not intended for non-consensual surveillance, speaker identification, or high-stakes decisions.
MIT.