guapaQAQ
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Update Gametime (basic) Parquet data
Browse files- README.md +122 -0
- basic/test.parquet +3 -0
- basic/train.parquet +3 -0
README.md
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---
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# Data-files config (no script, Parquet-only)
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configs:
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- config_name: basic
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data_files:
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- split: train
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path:
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- "basic/train-*.parquet"
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- "basic/train.parquet"
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- split: test
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path:
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- "basic/test-*.parquet"
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- "basic/test.parquet"
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default: true
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pretty_name: "Gametime (Basic)"
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tags:
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- audio
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- speech
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- tts
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- asr
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- benchmark
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task_categories:
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- automatic-speech-recognition
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- text-to-speech
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language:
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- en
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license: cc-by-4.0
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size_categories:
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- n<100K
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---
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# Gametime — Basic (Parquet)
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**Gametime** is a lightweight speech benchmark for quick TTS ↔︎ ASR sanity checks on short, instruction-like utterances.
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This repo uses **Parquet-only** data files (no `Audio` feature, no WAVs checked in). Each example contains:
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- `id, category, dataset, split, template_idx, item_idx, text`
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- `audio_bytes` (raw file bytes), `audio_format` (e.g., `"wav"`), `sampling_rate` (e.g., `16000`)
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Why? This keeps streaming reliable and dependency-light.
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---
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## TL;DR — Stream and decode
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```python
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from datasets import load_dataset
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import io, soundfile as sf
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ds = load_dataset("gametime-benchmark/gametime", "basic", split="train", streaming=True)
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ex = next(iter(ds))
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buf = io.BytesIO(ex["audio_bytes"]) # raw bytes from Parquet
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wav, sr = sf.read(buf, dtype="float32") # decodes in-memory
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print(ex["id"], sr, len(wav), ex["text"])
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````
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* Works with **streaming=True** (no full download).
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* No `torchcodec` required. You only need `soundfile` (libsndfile).
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---
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## Schema
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Columns in each Parquet row:
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* `id`: unique identifier (e.g., `1-a-Sequence-Number/train/1-a-Sequence-Number-01-01.wav`)
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* `category`: `"basic"`
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* `dataset`: group name (e.g., `1-a-Sequence-Number`)
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* `split`: `train` or `test`
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* `template_idx`, `item_idx`: indices if applicable (empty string otherwise)
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* `text`: reference text to speak/recognize
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* `audio_bytes`: raw WAV bytes
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* `audio_format`: `"wav"`
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* `sampling_rate`: e.g., `16000`
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---
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## Example: quick ASR sanity check (Whisper)
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```python
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# pip install transformers datasets soundfile evaluate
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import io, soundfile as sf, evaluate
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from datasets import load_dataset
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from transformers import pipeline
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asr = pipeline("automatic-speech-recognition", model="openai/whisper-small")
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wer = evaluate.load("wer")
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ds = load_dataset("gametime-benchmark/gametime", "basic", split="test", streaming=True)
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refs, hyps = [], []
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for i, ex in enumerate(ds):
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wav, sr = sf.read(io.BytesIO(ex["audio_bytes"]), dtype="float32")
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hyp = asr({"array": wav, "sampling_rate": sr})["text"]
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refs.append(ex["text"]); hyps.append(hyp)
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if i >= 199: # quick 200-sample check
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break
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print("WER:", wer.compute(references=refs, predictions=hyps))
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```
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---
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## Notes
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* **Advanced** category (with `<sil="Xms"/>` timing tokens) is planned separately.
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* License: **CC BY 4.0** for text, metadata, and audio.
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---
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## Citation
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```
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@dataset{gametime_basic_2025,
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title = {Gametime: Basic Speech Benchmark},
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year = {2025},
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url = {https://huggingface.co/datasets/gametime-benchmark/gametime},
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note = {Parquet-only split "basic" for TTS/ASR sanity checks}
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}
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```
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basic/test.parquet
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version https://git-lfs.github.com/spec/v1
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oid sha256:8e255c2fd8b2b33be1714b888849a985f75bd8e195a4efd8bb645d68bebdffe1
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size 240085790
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basic/train.parquet
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version https://git-lfs.github.com/spec/v1
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oid sha256:283c7ce2dc1833ecb0b82fe2c45f943ab9727a27ee23a325ebc52f80d7acb247
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size 473425907
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