Model reference · open weights

tedlium2_streaming_transformer

Audio espnet Speech→text 1 build Open weights 21 dl/mo

tedlium2_streaming_transformer is an open-weight audio or speech model from espnet. tedlium2_streaming_transformer (BF16) weighs 140 MB; the smallest configuration that runs it is RTX 3060 12 GB.

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  • tedlium2_streaming_transformer is an automatic speech recognition model developed by espnet that performs streaming decoding in 40-millisecond chunks.
  • It was trained on the TED-LIUM 2 dataset using a BPE-500 language model and supports English.
  • The model achieves a word error rate of 10.8 on the test set and is distributed under the cc-by-4.0 license.

Summary of the espnet/tedlium2_streaming_transformer model card, 2026-10-01

What it is

Released byespnet
Released2026-09-20
VRAM140 MB for the weights

What it runs on

Memory and cards for tedlium2_streaming_transformer (BF16)

140 MBweights, file size
1.6 GBruntime overhead
CardOne streamMemory
RTX 3060 12 GBfits11.6 GB
RTX 4060 Ti 16 GBfits15.4 GB
RTX 3090 24 GBfits23.4 GB
RTX 4090 24 GBfits23.4 GB
RTX 5090 32 GBfits31.0 GB
L40S 48 GBfits44.0 GB
A100 80 GBfits78.2 GB
H100 80 GBfits78.1 GB
RTX PRO 6000 Blackwell 96 GBfits93.8 GB
DGX Spark (GB10) 128 GB unifiedfits107 GB
H200 141 GBfits138 GB
B200 180 GBfits176 GB

From the model card

What espnet says about tedlium2_streaming_transformer

Read the model card

ESPnet2 streaming ASR model

espnet/tedlium2_streaming_transformer

A streaming Transformer: it decodes while the audio is still arriving, a block at a time, rather than waiting for the end of the utterance. Trained on TED-LIUM 2 with the egs2/tedlium2/asr1 recipe and conf/train_asr_streaming_transformer.yaml, with a BPE-500 language model beside it.

Trained and originally published by Keqi Deng, as D-Keqi/espnet_asr_train_asr_streaming_transformer_raw_en_bpe500_sp_valid.acc.ave. This copy is here so that the demo notebooks and the documentation point at a model the organization keeps alive; the weights are the same ones.

Usage

import soundfile as sf
from espnet2.bin.asr_inference_streaming import Speech2TextStreaming

s2t = Speech2TextStreaming.from_pretrained(
    "espnet/tedlium2_streaming_transformer",
    device="cpu",
    beam_size=20,
    ctc_weight=0.5,
    penalty=0.0,
    nbest=1,
    disable_repetition_detection=True,
)

speech, rate = sf.read("audio.wav", dtype="float32")  # 16 kHz, one channel

# 640 samples is 40 ms: the hypothesis comes back after each slice, and grows
chunk = 640
slices = len(speech) // chunk
for i in range(slices):
    results = s2t(speech=speech[i * chunk : (i + 1) * chunk], is_final=False)
    if results:
        print(results[0][0])

results = s2t(speech=speech[slices * chunk :], is_final=True)
print(results[0][0])

There is a notebook that runs this end to end: Demos/asr_streaming_demo.ipynb.

For non-streaming recognition, OWSM-CTC is far more accurate and covers 150+ languages.

Results

TED-LIUM 2, decoded with the language model in this repository.

datasetWERCER
dev11.45.4
test10.85.3

The full tables are in exp/asr_train_asr_streaming_transformer_raw_en_bpe500_sp/RESULTS.md.

Trained with espnet 0.9.8 and pytorch 1.5.1, in November 2021, and loaded by current ESPnet through the meta.yaml in this repository.

Citing ESPnet

@inproceedings{watanabe2018espnet,
  author={Shinji Watanabe and Takaaki Hori and Shigeki Karita and Tomoki Hayashi and Jiro Nishitoba and Yuya Unno and Nelson {Enrique Yalta Soplin} and Jahn Heymann and Matthew Wiesner and Nanxin Chen and Adithya Renduchintala and Tsubasa Ochiai},
  title={{ESPnet}: End-to-End Speech Processing Toolkit},
  year={2018},
  booktitle={Proceedings of Interspeech},
  pages={2207--2211},
  doi={10.21437/Interspeech.2018-1456},
  url={http://dx.doi.org/10.21437/Interspeech.2018-1456}
}

Quoted from the model card on Hugging Face. The full card is behind the Hugging Face link above.

How it works

How audio & music work

Audio or textinputAudio modelrecognise / synthesiseText or audiooutputSpeech-to-text turns audio into text; text-to-speech and music models turn text into audio.
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