datasets
Training and evaluation data, with the modality, task and licence stated up front. Listed live from the Hugging Face Hub.
LEMAS-Dataset-train
Overview
This dataset is part of LEMAS-Project (lemas-project.github.io/LEMAS-Project).
It contains a large-scale training set (150k+ hours) and a curated evaluation set
(500 utterances per language) covering 10 languages, all with word-level alignment.
Fields
key: unique utterance identifier; the first two characters indicate the language ID
audio: relative path to the MP3 audio file (in the eval set, this key is renamed to "file_name" for compatibility with the viewer)… See the full description on the dataset page: https://huggingface.co/datasets/LEMAS-Project/LEMAS-Dataset-train.Earnings22-Cleaned-AA
Earnings22-Cleaned-AA
Quick links: AA Speech-to-Text Leaderboard | AA-WER v2.0 article
Earnings22-Cleaned-AA is a cleaned subset of the English Earnings-22 test data from esb/datasets, a corpus of corporate earnings calls from global companies with speakers of many different nationalities and accents. This cleaned subset is the Earnings-22 portion included in AA-WER v2. We manually reviewed and corrected errors in the original ground-truth transcriptions to ensure fairer evaluation… See the full description on the dataset page: https://huggingface.co/datasets/ArtificialAnalysis/Earnings22-Cleaned-AA.Granary
Granary: Speech Recognition and Translation Dataset in 25 European Languages
Granary is a large-scale, open-source multilingual speech dataset covering 25 European languages for Automatic Speech Recognition (ASR) and Automatic Speech Translation (AST) tasks.
Overview
Granary addresses the scarcity of high-quality speech data for low-resource languages by consolidating multiple datasets under a unified framework:
🗣️ ~1M hours of… See the full description on the dataset page: https://huggingface.co/datasets/nvidia/Granary.Audio2Tool
Audio2Tool: Speak, Call, Act — A Dataset for Benchmarking Speech Tool Use
Authors: Ramit Pahwa1,∗,∗∗, Apoorva Beedu1,∗, Parivesh Priye1, Rutu Gandhi†1, Saloni Takawale†1, Aruna Baijal1, Zengli Yang1
1 Rivian & Volkswagen Technologies · ∗ equal contribution · ∗∗ corresponding author · † equal contribution
📄 Project page / demo: https://audio2tool.github.io/
📦 Dataset: https://huggingface.co/datasets/RVtech/Audio2Tool
✉️ Contact (corresponding… See the full description on the dataset page: https://huggingface.co/datasets/RVtech/Audio2Tool.StreamAudio-2M
StreamAudio-2M
Large-scale streaming-audio dataset for audio-LLM / audio-agent training. Each row is a
stream: a sequence of audio turns sharing one unified schema. ~2.28M unique audio clips
are organised into six task subsets.
Subsets
Subset
Rows
Description
Stream_Audio_Understanding
90,738
Montages of audio-understanding clips (AudioSet / FMA): captions, choice & open QA
Real_time_ASR
28,109
Streams of ASR clips (CommonVoice / GigaSpeech /… See the full description on the dataset page: https://huggingface.co/datasets/zhifeixie/StreamAudio-2M.apptek_callcenter_dialogues
AppTek Call-Center Dialogues: A Multi-Accent Long-Form Benchmark for English ASR
AppTek Call-Center Dialogues is a long-form conversational speech dataset for automatic speech recognition (ASR), featuring diverse English accents
across multiple service-oriented domains and designed to evaluate models on realistic call-center interactions.
128.6 hours of speech
14 English accent groups
16 service domains
5–15 minute conversations (long-form)
Split-channel audio (one… See the full description on the dataset page: https://huggingface.co/datasets/apptek-com/apptek_callcenter_dialogues.UltraVoice
UltraVoice: Scaling Fine-Grained Style-Controlled Speech Conversations for Spoken Dialogue Models
📝 Abstract
Spoken dialogue models currently lack the ability for fine-grained speech style control, a critical capability for human-like interaction that is often overlooked in favor of purely functional capabilities like reasoning and question answering. To address this limitation, we introduce UltraVoice, the first large-scale speech dialogue dataset… See the full description on the dataset page: https://huggingface.co/datasets/tutu0604/UltraVoice.ContextASR-Bench
ContextASR-Bench: A Massive Contextual Speech Recognition Benchmark
Automatic Speech Recognition (ASR) has been extensively investigated, yet prior benchmarks have largely focused on assessing the acoustic robustness of ASR models, leaving evaluations of their linguistic capabilities relatively underexplored. This largely stems from the limited parameter sizes and training corpora of conventional ASR models, leaving them with insufficient world knowledge, which is crucial for… See the full description on the dataset page: https://huggingface.co/datasets/MrSupW/ContextASR-Bench.VoxPopuli-Cleaned-AA
VoxPopuli-Cleaned-AA
Quick links: AA Speech to Text Leaderboard | AA-WER v2.0 article
VoxPopuli-Cleaned-AA is a cleaned subset of the English VoxPopuli test data from esb/datasets, a speech dataset derived from European Parliament recordings. This cleaned subset is the VoxPopuli portion included in AA-WER v2. We manually reviewed and corrected errors in the original ground-truth transcriptions to ensure fairer evaluation of Speech to Text (STT) models.
This dataset is part of AA-WER… See the full description on the dataset page: https://huggingface.co/datasets/ArtificialAnalysis/VoxPopuli-Cleaned-AA.thai-aligner-bench
Thai Aligner Bench
🚧 Development in progress.
How accurately can a forced aligner place Thai token and word boundaries in
speech? This is a self-contained benchmark: one Python file
(aligner_bench.py) plus 1,572 clips of Thai speech with frame-exact timing
ground truth. No Thai NLP stack or other code is needed — just
numpy soundfile torch torchaudio transformers.
The ground truth is what makes the dataset useful: the audio was rendered by a
TTS model whose duration predictor… See the full description on the dataset page: https://huggingface.co/datasets/wayu-ai/thai-aligner-bench.speech-translation-and-summarization
English-Centric Multilingual Audio Dataset
This dataset contains generated article and summary audio for English-centric multilingual directions.
Each direction folder contains metadata JSONL files and corresponding audio files for few_shot and test splits.
Included directions
amharic_english / english_amharic
arabic_english / english_arabic
bengali_english / english_bengali
chinese_simplified_english / english_chinese_simplified
english_english
french_english /… See the full description on the dataset page: https://huggingface.co/datasets/McGill-NLP/speech-translation-and-summarization.spanish-slang-stt-data
Spanish Regional Speech-to-Text Dataset
A multilingual Spanish speech recognition dataset covering 4 regional dialects for fine-tuning Whisper and other ASR models.
Dataset Description
This dataset contains ~39,000 audio samples with transcriptions across 4 Spanish-speaking regions:
Region
Samples
Description
Mexico
17,725
Mexican Spanish including CIEMPIESS corpus
Spain
11,360
Castilian Spanish from TEDx and Common Voice
Argentina
5,839
Rioplatense Spanish… See the full description on the dataset page: https://huggingface.co/datasets/shraavb/spanish-slang-stt-data.voice-code-bench
VoiceCodeBench
VoiceCodeBench is a test-only benchmark for evaluating whether automatic
speech recognition (ASR) systems preserve exact structured values in English
workplace speech.
Paper: VoiceCodeBench: Evaluating Exact Structured-Token Recovery in Automatic Speech Recognition
The benchmark targets cases where a transcript is software input: callback
numbers, email addresses, command-line flags, file paths, URLs, account
identifiers, dates, measurements, and similar values… See the full description on the dataset page: https://huggingface.co/datasets/besimple-ai/voice-code-bench.simsamu
Simsamu dataset
This repository contains recordings of simulated medical dispatch dialogs in the
french language, annotated for diarization and transcription. It is published
under the MIT license.
These dialogs were recorded as part of the training of emergency medicine
interns, which consisted in simulating a medical dispatch call where the interns
took turns playing the caller and the regulating doctor.
Each situation was decided randomly in advance, blind to who was playing the… See the full description on the dataset page: https://huggingface.co/datasets/medkit/simsamu.quran-tajweed-phonetics
The complete phonetic layer of the Quran in the riwaya of Hafs 'an
'Asim via tariq al-Shatibiyyah: 6,236 ayat, 522,475 phones, every
phone carrying its tajweed attribution: madd class with its transmitted
length range, ghunna grade, qalqalah class, tafkheem with its rank, sakt,
the seventeen sifat, and the rule that produced it.
Built and maintained by Quran Lab, a waqf building open technology in
the service of the Quran.
How it was built and verified
Indexed from the… See the full description on the dataset page: https://huggingface.co/datasets/Quran-Lab/quran-tajweed-phonetics.Earnings22-Cleaned-AA-chunked
Earnings22-Cleaned-AA-chunked
Quick links: AA Streaming Speech to Text Leaderboard | Speech to Text methodology
Earnings22-Cleaned-AA-chunked is a chunked version of Earnings22-Cleaned-AA, the cleaned Earnings-22 subset used by Artificial Analysis for streaming Speech to Text evaluation.
The original Earnings-22 data comes from esb/datasets, a corpus of corporate earnings calls. Artificial Analysis manually reviewed and corrected the reference transcripts in the cleaned subset… See the full description on the dataset page: https://huggingface.co/datasets/ArtificialAnalysis/Earnings22-Cleaned-AA-chunked.contact-attendant-zhtw
Contact-Attendant zh-TW/en — speech → tool-call dialogs
The training & evaluation data behind Luigi/Qwen3-ASR-0.6B-Agent
— a 0.6B speech agent that hears a spoken request and emits a search_contacts tool call for a
bilingual (Traditional Chinese / English) office phone directory.
This dataset is fully self-contained: the audio clips, the multi-turn dialog transcripts, the
closed contact directory, and the scripts that generated them. With it you can reproduce the
fine-tuned… See the full description on the dataset page: https://huggingface.co/datasets/Luigi/contact-attendant-zhtw.HearInContextEnglish | 中文
HearInContext
A Benchmark for Implicit Context in Speech Recognition
Illustrative example: the same spoken request is disambiguated as flour or flower by different assistant histories. The dialogue and waveform are illustrative.
Same audio. Different contexts. Different meanings.
HearInContext is a Mandarin–English contextual speech recognition benchmark. It pairs the same audio with dialogue histories supporting different meanings to evaluate… See the full description on the dataset page: https://huggingface.co/datasets/OPPOer/HearInContext.darija-asr-corpus
Darija ASR Corpus (dataset-core)
Arabizi (Latin-script) transcriptions of Moroccan Darija speech, produced for a
Whisper fine-tuning pipeline (paper not yet published -- citation forthcoming).
This repo contains four source subsets: DODa, DVoice, Wiki, and
YouTube. Each subset carries its own upstream license/terms -- see below --
because they are drawn from four different original projects.
Subsets
Config
Rows
Audio bundled?
Upstream license
Upstream source… See the full description on the dataset page: https://huggingface.co/datasets/abnajlae/darija-asr-corpus.librivox-tracksA dataset of all audio files & corresponding sources uploaded to LibriVox before 1st November 2025.
Donate to Internet Archive, who hosts most of the audio data
Donate to Project Gutenberg, who hosts most of the source books
Be a good netizen and use proper caching & sensible rate limiting when downloading files.
November 2025 Update
book.id, section.id, and section.num are now integers instead of strings.
reader has been replaced with a readers array. No entries currently… See the full description on the dataset page: https://huggingface.co/datasets/pykeio/librivox-tracks.chinese-lips-speech-slide-probe
Chinese-LiPS Speech + Slide Probe
A self-contained probe set for testing whether visual slide context helps
simultaneous speech translation — with the input as audio, not transcripts.
Why audio matters: feeding a transcript to a text LLM deletes the acoustic
ambiguity (homophones, polysemy) that slide context is meant to resolve; the
transcript already commits to one reading. Any honest test of "does vision help
streaming ST" must consume speech.
Contents… See the full description on the dataset page: https://huggingface.co/datasets/gavinlaw/chinese-lips-speech-slide-probe.bengali-talkshow-audio
Bengali Talkshow Audio Dataset
A large-scale collection of 1,180 Bengali talk show audio recordings totaling 789+ hours of multi-speaker speech, sourced from Bangladeshi television talk shows and political debate programs.
Dataset Description
This dataset contains audio from Bengali-language TV talk shows, political debates, and news discussion programs from major Bangladeshi television channels. Each recording features multiple speakers engaged in discussion, making it… See the full description on the dataset page: https://huggingface.co/datasets/nymtheescobar/bengali-talkshow-audio.pavo-bench
PAVO-Bench: 50K-Turn Benchmark for ASR-LLM-TTS Pipeline Routing
Code: github.com/vnmoorthy/pavo-bench · Paper: TMLR 2026 (accepted) · Authors: NarasingaMoorthy VeiluKanthaPerumal (UPenn), Mohammed Imthathullah (Google)
pip install git+https://github.com/vnmoorthy/pavo-bench.git
Headline results (vs fixed-cloud baseline, 50,000 voice turns)
Metric
Result
Significance
P95 end-to-end latency (H100, LibriSpeech)
−10.3% (−167 ms)
—
Median latency
−34%… See the full description on the dataset page: https://huggingface.co/datasets/vnmoorthy/pavo-bench.massive-yt-edu-queue
Massive YouTube Educational Video Queue
Full metadata and content classification for 4,489,228 YouTube educational videos totaling 3,975,157 hours.
Description
This dataset contains metadata, content categorization, and license risk assessment for ~4.5M YouTube videos identified as potentially educational. It serves as the discovery and processing queue for the massive-yt-edu-transcriptions project, which aims to create the world's largest open educational transcript… See the full description on the dataset page: https://huggingface.co/datasets/thepowerfuldeez/massive-yt-edu-queue.ark-asr-3b-open-asr-leaderboard-results
ARK-ASR-3B Open ASR Leaderboard Results
Raw JSONL manifests for AutoArk-AI/ARK-ASR-3B on the public English
short-form hf-audio/open-asr-leaderboard splits.
These manifests were generated on a local 8x RTX 4090 machine and scored with
the shared Open ASR Leaderboard scorer:
PYTHONPATH=. python - <<'PY'
from normalizer.eval_utils import score_results
score_results(
'ark_asr/results.AutoArk-AI-ARK-ASR-3B_20260622_official',
'AutoArk-AI/ARK-ASR-3B',
)
PY
Important:… See the full description on the dataset page: https://huggingface.co/datasets/Edge0/ark-asr-3b-open-asr-leaderboard-results.EGYSpeak
EGYSpeak
A curated dataset of 147,979 single-speaker Egyptian Arabic (pure dialect) audio clips with transcriptions, sourced from the fadisarwat/egyptian-arabic-lines Kaggle dataset and processed through a rigorous ASR pipeline.
Quick Start
1. Download the dataset:
from huggingface_hub import snapshot_download
snapshot_download(
repo_id="MohamedGomaa30/EGYSpeak",
repo_type="dataset",
local_dir="EGYSpeak",
)
2. Extract the dataset:
from… See the full description on the dataset page: https://huggingface.co/datasets/MohamedGomaa30/EGYSpeak.Tech-Sentences-For-ASR-Training
TechVoice Dataset
Work in Progress – This dataset is actively being expanded with new recordings.
Dataset Statistics
Metric
Current
Target
Progress
Duration
38m 43s
5h 0m 0s
██░░░░░░░░░░░░░░░░░░ 12.9%
Words
10,412
50,000
████░░░░░░░░░░░░░░░░ 20.8%
Total Recordings: 205 samples
Total Characters: 74,312
A specialized speech dataset for fine-tuning Automatic Speech Recognition (ASR) models on technical and developer vocabulary. Contains human-recorded… See the full description on the dataset page: https://huggingface.co/datasets/danielrosehill/Tech-Sentences-For-ASR-Training.teochew_wild
Teochew-Wild：首个正字标注的野外潮州话数据集
本数据集（Teochew-Wild）是从网络上发音清晰、噪声较少的音视频内容中获取的，原始音视频的数据来源为：民生新闻、潮汕讲古、地方电视节目、故事书、抖音自媒体口播等，我借鉴了Emilla提出的数据集自动处理流水线，对原始数据进行归一化、降噪和剪切（部分自动剪切效果差的使用手工修正）；
Teochew-Wild总共包括20个发音标准、念错率低的潮汕母语说话人、共12500条音频片段，包含潮州市区、汕头市区、澄海、榕江音、潮安南部等多个区域的口音，语料内容覆盖书面用语与口头用语，并同时提供正字和拼音标注，是首个公开可用、标注准确率高的潮州话数据集，主要面向语音识别和语音合成任务。
文件说明 （File Structure Explanation）
├── label_for_qwen_asr/ # 预处理标签文件夹，完全适配Qwen-ASR模型读取格式
├── README.md # 项目说明文档（本文档）… See the full description on the dataset page: https://huggingface.co/datasets/panlr/teochew_wild.multichannel-meetings-10h
GroundTruth Multi-Channel Meeting Audio Dataset (10h)
Summary
This dataset contains approximately 10 hours of co-located, multi-speaker meeting recordings, each captured simultaneously via a room (built-in) microphone and individual close-talk lapel microphones worn by each participant.
Each meeting includes:
One full meeting recording (room microphone)
Individual close-talk recordings for each participant (one file per speaker)
Structured metadata describing speakers… See the full description on the dataset page: https://huggingface.co/datasets/ground-truth/multichannel-meetings-10h.MM-ContextASR-Bench
MM-ContextASR Bench
Metadata and evaluation splits for Multimodal Conversational Context for
LLM-Based ASR: Data Construction, Training, and Benchmark.
Dataset summary
Config
Examples
Audio
Context
Primary metric
mm_contextasr
1,250 (250 current utterances × 5 histories)
1,439 WAV files included
Controlled user-assistant dialogue
entity Recall
kespeech
19,212
Source ID only
Same-speaker speech and transcript
CER, SER, entity Recall
cv_yue
3,525… See the full description on the dataset page: https://huggingface.co/datasets/lilonghao/MM-ContextASR-Bench.