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OpenSTBench

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arXiv PyPI Python License: MIT GitHub X-LANCE

OpenSTBench is a multidimensional evaluation toolkit for speech translation. It is designed for heterogeneous systems, including speech-to-text translation (S2TT), speech-to-speech translation (S2ST), offline systems, and streaming systems.

The toolkit organizes evaluation into three dimensions:

  • Translation Quality: whether the translated text preserves the source meaning.
  • Speech Quality: whether generated speech is natural, text-consistent, speaker-preserving, emotion-preserving, and faithful to non-verbal or paralinguistic events.
  • Temporal Quality: whether generated speech preserves duration structure and, for streaming systems, whether output is responsive.

Installation

pip install OpenSTBench

For local development:

git clone https://github.com/sjtuayj/OpenSTBench.git
cd OpenSTBench
conda create -n openstbench python=3.10 -y
conda activate openstbench
pip install -e .

Optional extras:

pip install "OpenSTBench[comet]"
pip install "OpenSTBench[whisper]"
pip install "OpenSTBench[speech_quality]"
pip install "OpenSTBench[emotion]"
pip install "OpenSTBench[paralinguistics]"
pip install "OpenSTBench[metricx]"
pip install "OpenSTBench[all]"

MetricX follows the official google-research/metricx runtime requirements. Installing OpenSTBench[metricx] or OpenSTBench[all] pins the MetricX-compatible stack, including transformers[torch]==4.30.2, sentencepiece==0.1.99, datasets==2.13.1, protobuf==3.20.3, and accelerate>=0.26.0.

BLEURT is installed separately:

pip install git+https://github.com/lucadiliello/bleurt-pytorch.git

Package Names

  • PyPI package: OpenSTBench
  • Python import: openstbench

Evaluation Dimensions

Dimension Evaluator System type Main outputs
Translation Quality TranslationEvaluator S2TT, S2ST text or generated speech sacreBLEU, chrF++, COMET, BLEURT, MetricX, MetricX_QE, and their ASR_ variants
Speech Quality SpeechQualityEvaluator S2ST UTMOS, WER_Consistency, CER_Consistency
Speech Quality SpeakerSimilarityEvaluator S2ST average_wavlm_large_similarity, average_resemblyzer_similarity
Speech Quality EmotionEvaluator S2ST Emotion2Vec_Cosine_Similarity, Audio_Emotion_Accuracy
Speech Quality ParalinguisticEvaluator S2ST Acoustic_Event_Count_F1, Acoustic_Event_Localization_F1, Acoustic_Event_Onset_Error
Temporal Quality TemporalConsistencyEvaluator S2ST Duration_Consistency_SLC_0.2, Duration_Consistency_SLC_0.4
Temporal Quality LatencyEvaluator Streaming S2TT/S2ST First_Audio_Delay_(StartOffset_ms), Overall_Translation_Delay_(ATD_ms), End_Action_Delay_(CustomATD_ms), Real_Time_Factor_(RTF)

Offline and streaming are supported system settings, not separate metric dimensions. Use the evaluators that match the available outputs: text, generated speech, source/target audio pairs, event annotations, or streaming traces.

Experimental Overview

The radar plot below illustrates the multidimensional view produced by OpenSTBench for representative streaming and offline speech translation systems. It summarizes how systems can differ across translation quality, speech quality, and temporal quality: a system with strong translation quality may still show different behavior in speech realization, speaker or emotion preservation, paralinguistic fidelity, temporal consistency, and latency or efficiency.

OpenSTBench experimental radar overview

Datasets

The paper uses the following datasets. Please follow the license and access terms of each original dataset.

Dataset Used for Link
MSLT dev Translation quality, speech quality, temporal consistency, latency Microsoft Speech Language Translation Corpus
LibriTTS-based paired speaker set Speaker preservation The constructed OpenSTBench paired set is available on Hugging Face Datasets; the source corpus is LibriTTS
RAVDESS Emotion preservation Audio_Speech_Actors_01-24.zip from the RAVDESS Zenodo record
MCAE-SPPS Emotion preservation MCAE-SPPS on OSF
NonverbalTTS test Paralinguistic fidelity deepvk/NonverbalTTS
SynParaSpeech Paralinguistic fidelity shawnpi/SynParaSpeech

Quick Start

from openstbench import TranslationEvaluator

evaluator = TranslationEvaluator(
    use_bleu=True,
    use_chrf=True,
    use_comet=False,
    use_bleurt=False,
    use_metricx=True,
    device="cuda",
)

scores = evaluator.evaluate_all(
    reference=["我喜欢看电影。", "今天天气很好。"],
    target_text=["我喜欢看电影。", "今天天气很好。"],
    source=["I like watching movies.", "The weather is nice today."],
    target_lang="zh",
    asr_text=["我喜欢看电影。", "今天天气很好。"],
)

print(scores)

ASR translation-quality evaluation is enabled by default and runs only when target_audio or precomputed asr_text is supplied. To transcribe generated speech directly, pass a file, a list of files, or a directory as target_audio; Whisper defaults to medium and can be changed with TranslationEvaluator(whisper_model="large-v3"). Set use_asr=False to disable every ASR_ metric.

Examples

Complete parameter templates are kept in examples/. The README intentionally stays compact; use these files for configurable parameters, input formats, and output fields.

  • examples/python/translation_eval.py
  • examples/python/speech_quality_eval.py
  • examples/python/speaker_similarity_eval.py
  • examples/python/emotion_eval.py
  • examples/python/paralinguistic_eval.py
  • examples/python/paralinguistic_identity_baseline.py
  • examples/python/temporal_consistency_eval.py
  • examples/python/latency_eval.py
  • examples/bash/install_extras.sh
  • examples/bash/run_latency_cli.sh

Latency can also be run from the module CLI:

python -m openstbench.latency.cli --help

Conventions

  • Text inputs generally accept list[str], one-sample-per-line .txt files, and .json files where supported by the evaluator.
  • Audio inputs generally accept folders, list[str], .txt path lists, and .json path lists where supported by the evaluator.
  • For zh, ja, and ko, speech consistency reports CER_Consistency; other languages report WER_Consistency.
  • Evaluators that accept pretrained model sources use a local-first rule. If the supplied local path exists, OpenSTBench uses it; otherwise it falls back to the configured remote model id.
  • Optional dependencies are loaded only when the corresponding evaluator needs them.
  • MetricX is enabled by default in TranslationEvaluator. It follows the official google-research/metricx README, uses text only, reports error scores in [0, 25] where lower is better, and can be disabled with use_metricx=False.
  • TranslationEvaluator also enables ASR variants by default. With target_audio, Whisper transcribes the generated speech before the same enabled translation metrics are applied; with asr_text, the supplied transcript is used directly. Calls without either input preserve the text-only behavior.

Acknowledgements

Citation

If you find our work useful, please cite as:

@misc{an2026openstbenchsemanticevaluationspeech,
      title={OpenSTBench: Beyond Semantic Evaluation for Speech Translation}, 
      author={Yanjie An and Yuxiang Zhao and Yichi Zhang and Qixi Zheng and Yujie Tu and Keqi Deng and Kai Yu and Xie Chen},
      year={2026},
      eprint={2605.30792},
      archivePrefix={arXiv},
      primaryClass={eess.AS},
      url={https://arxiv.org/abs/2605.30792}, 
}

License

OpenSTBench's original code is released under the MIT License. See LICENSE.

Some latency evaluation components include code adapted from SimulEval, which is licensed under the Creative Commons Attribution-ShareAlike 4.0 International License (CC BY-SA 4.0). Those adapted portions are distributed under CC BY-SA 4.0. See THIRD_PARTY_NOTICES.md for details.

The datasets referenced by OpenSTBench, including the datasets used in the paper, are not covered by the OpenSTBench code license. They are provided by their original authors or distributors under their own licenses and terms of use. Some datasets are restricted to research or non-commercial use.

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