Moonshine Streaming Small — Spanish
Spanish streaming speech recognition, 112.9M parameters. Same architecture as moonshine-ai/moonshine-streaming-small, trained for Spanish with a 12,288-entry Spanish tokenizer. The [Tiny model](https://huggingface.co/moonshine-ai/moonshine-streaming-tiny- es) is a quarter of the size.
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.
Checkpoint identity
This repository is a conversion of one specific training checkpoint, recorded here because the weights behind a language move as later stages win:
| Checkpoint | es12k_small_stageC_soup012_macro4.99.safetensors |
| Stage | C (read-speech mix), weight-averaged over three checkpoints |
| Architecture | spindlier_prime_adapted |
| Tokenizer | tokenizer_es12k.json, vocab 12,288 |
| Snapshot taken | 2026-08-24 |
| Parameters | 112.9M |
If you need reproducibility, pin the revision of this repository rather than
tracking main.
Usage
pip install --upgrade transformers datasets[audio]
from transformers import MoonshineStreamingForConditionalGeneration, AutoProcessor
import torch
model = MoonshineStreamingForConditionalGeneration.from_pretrained(
"moonshine-ai/moonshine-streaming-small-es"
).eval()
processor = AutoProcessor.from_pretrained("moonshine-ai/moonshine-streaming-small-es")
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.
Architecture
| Encoder | 10 layers, width 620, 8 heads, sliding windows (16, 4) on the first two and last two layers and (16, 0) between |
| Decoder | 10 layers, width 512, 8 heads, RoPE over 32 of each head's 64 dimensions |
| Frontend | 50 Hz features, CMVN, asinh compression, two causal stride-2 convolutions |
| Adapter | learned absolute positional embeddings, then a projection from 620 to 512 |
The lookahead layers give roughly 80 ms of lookahead; the intermediate layers have none.
Training data
Trained on a large-scale automatically labeled Spanish corpus, plus a much smaller human-labeled read-speech set:
- Crawled corpus, roughly 160,000 hours, pseudo-labeled and unaudited.
- Track A read speech, roughly 1,700 hours, human-transcribed (Multilingual LibriSpeech, Common Voice, LibriVox, VoxPopuli, FLEURS and OpenSLR Argentinian/Chilean/Colombian sets).
The crawled 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. No human-verified transcript was used for the bulk of training.
Evaluation
Spanish is scored on word error rate (WER), after the usual case and punctuation normalization. Mandarin and Japanese in this model family are instead scored on no-space CER, because they are written without spaces; every other language, this one included, uses WER.
suite_es is FLEURS Spanish and Multilingual LibriSpeech Spanish. Both are
read speech. The Multilingual LibriSpeech panel is European Spanish audiobooks;
the model is not measured here on Latin American spontaneous speech.
Seeded 400-utterance sample, batch 1
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.
| Panel | WER |
|---|---|
fleurs_es | 4.91 |
mls_es | 4.95 |
| macro | 4.933 |
This repository against the training checkpoint
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. A conversion that loads and emits
plausible text can still have a permuted weight mapping, which only a score
catches.
fleurs_es | mls_es | macro | |
|---|---|---|---|
| Training checkpoint | 4.91 | 4.95 | 4.933 |
| This repository | 4.89 | 4.57 | 4.732 |
399/400 and 398/400 transcripts are byte-identical.
The quantized build we ship
The .ort package served to the Moonshine deployment library is quantized to
int8 from these same weights, and scores 4.938 against 4.933 for the float
checkpoint on the same sample under the same stopping rule -- a difference of
+0.004, which is inside the noise of a 400-clip sample and should not be read
as the quantized build being better or worse. That build is a different
artifact from this repository, which is float32.
Limitations
- Machine-labeled training data. See above; the model reproduces its teacher's mistakes as well as its strengths.
- Repetition loops on short clips. Like other seq2seq ASR models this one can fall into a repetition loop, and short or noisy clips are where it happens. Cap the output length, as the usage snippet does.
- Evaluated on 2 panels only. No evaluation of telephony, children's speech, heavy dialect, or noisy far-field conditions.
Out-of-scope use
Not intended for non-consensual surveillance, speaker identification, or high-stakes decisions.
License
MIT.