Natural language processing (NLP) and speech technologies have made significant progress in recent years; however, they remain largely focused on standardized language varieties. Dialects, despite their cultural significance and widespread use, are underrepresented in linguistic resources and computational models, resulting in performance disparities. To address this gap, we introduce Saar-Voice, a six-hour speech corpus for the Saarbrücken dialect of German. The dataset was created by first collecting text through digitized books and locally sourced materials. A subset of this text was recorded by nine speakers, and we conducted analyses on both the textual and speech components to assess the dataset's characteristics and quality. We discuss methodological challenges related to orthographic and speaker variation, and explore grapheme-to-phoneme (G2P) conversion. The resulting corpus provides aligned textual and audio representations. This serves as a foundation for future research on dialect-aware text-to-speech (TTS), particularly in low-resource scenarios, including zero-shot and few-shot model adaptation.
Figures & tables
Figure 1: The isogloss dat-das-line distinguishing areas in which Rhine Franconian dialects (“das”) and Moselle Franconian dialects (“dat”) are spoken ( Drenda, 2008 ) in Saarland (smaller area in the south-west marked with thin border lines) and Rhineland-Palatinate.
Source
Domain
# Sent
# Words
# U. Words
Printed Books
An da Saar gefonn
Poetry
698
6,228
1,978
Geschaffd - Gelääbd
Poetry; Prose
394
3,302
1,214
Saa, was de willschd
Prose; Poetry
2,216
19,711
3,147
Was wääs dann isch
Poetry; Prose
2,525
18,445
3,490
Locally Sourced
Table 1: Textual corpus statistics of the collected texts for the Saarbrücken dialect.
Speakers
Total speakers
9
Gender ratio
4F / 5M
Speech
Recorded sentences
4,871
Total duration (hh:mm:ss)
05:54:48
Avg. duration per sent. (sec)
4.37
Table 2: Saar-Voice Corpus Statistics.
Speaker
Gender
Total audio (hh:mm:ss)
# sentences
Mean F0
Median F0
F0 range (5–95%)
W/s
P01
F
00:23:17
301
203
200
141-278
1.8249
P02
F
01:02:28
1075
251
249
203-312
2.4166
P03
M
01:15:37
899
115
107
80-176
1.7352
P04
M
00:27:56
400
143
140
100-196
2.2056
P05
F
00:27:18
399
213
212
175-256
2.0203
P06
M
00:18:43
300
119
115
84-160
2.3334
Table 3: Speaker-level acoustic statistics. F0 in Hz. W/s = words per second.
Figure 2: SNR distribution on Saar-Voice.
Word
English
Epitran
Ours
berechne
calculate (Imp.)
\textipa b \textschwa \textscr \textepsilon xne \textlengthmark
\textipa b \textschwa \textscr \textepsilon xn \textschwa
bidde
please
\textipa b \textsci tde \textlengthmark
\textipa b \textsci d \textschwa
Midde
middle
\textipa m \textsci tde \textlengthmark
\textipa m \textsci d \textschwa
odder
or
\textipa \textopeno td \textschwa \textscr
\textipa \textopeno d \textturna
buuch
book (Imp.)
\textipa bu \textlengthmark u \textlengthmark x
\textipa bu \textlengthmark x
ääner
one
\textipa æ \textlengthmark æ \textlengthmark n \textschwa \textscr
\textipa æ \textlengthmark n \textturna
Table 4: Examples of G2P Errors. Corrections of transcriptions are estimates by a native speaker.
Arabic automatic speech recognition (ASR) faces unique challenges due to diglossia, extensive regional dialect variation, and limited speech resources. Existing speech datasets often focus on single dialects or large-scale broadcast/web data, leading to trade-offs between linguistic diversity and annotation quality. We present BULBUL, a multi-dialect Arabic ASR dataset collected from 275 speakers in 11 Arab countries. BULBUL includes structured dialect and sub-dialect coverage, as well as recordings of classical Arabic and modern standard Arabic spoken by participants in their native dialectal accents to support accent-aware modeling. The quality of the recordings was ensured through a two-level human verification process. We further benchmark a range of recent ASR systems, establishing strong baselines for modern dialectal and accented Arabic ASR.
While modern Automatic Speech Recognition (ASR) systems achieve high accuracy on benchmark corpora, their performance often degrades when there is real-world variability. This work focuses on variability arising due to accented, spontaneous, and domain-specific speech. In particular, we introduce PAper REading DAtaset (PAREDA), a first-of-its-kind multi-accent speech dataset consisting of discussions on academic Natural Language Processing (NLP) papers between speakers with Australian, Indian-English, and Chinese English accents. Each session elicits a spontaneous monologue (a summary of a paper's abstract) and a non-monologue (a question-and-answer session between participants), resulting in a corpus rich with technical jargon and conversational phenomena. We evaluate the performance of SOTA ASR models on PAREDA, analysing the impact of accent mixing and increased speech rate. Our results show that, in the zero-shot setting, models perform worse, confirming the dataset's challenging nature. However, fine-tuning on PAREDA significantly reduces the Word Error Rate (WER), demonstrating that our dataset captures linguistic characteristics often missing from existing corpora. PAREDA serves as a valuable new resource for building and evaluating more robust and inclusive ASR systems for specialised, real-world applications.
Current end-to-end speech dialogue models are primarily optimized for mainstream languages and remain limited in low-resource dialect scenarios due to the scarcity of dialect speech data. Moreover, during dialect adaptation, the semantic representation space of speech dialogue models continuously evolves, while conventional speech supervision remains unchanged, leading to semantic inconsistency between hidden representations and speech targets and degrading speech stability and naturalness. To address these issues, we propose DialectS2S, an end-to-end speech dialogue model for Chinese dialects. We first develop a scalable dialect speech dialogue synthesis pipeline for efficient data construction. We further introduce a two-stage post-training strategy with self-aligned speech supervision, which aligns the semantic content of speech supervision with the evolved semantic representations of the model to improve dialect speech generation quality. Experimental results show that DialectS2S consistently outperforms existing baselines across multiple Chinese dialects in speech dialogue, achieving substantial improvements in dialect consistency, response quality, and speech intelligibility. Our work provides an efficient and scalable solution for end-to-end speech dialogue modeling in low-resource dialect scenarios. To facilitate future research and practical applications, we fully open-source the DialectS2S framework, including model checkpoints, training datasets, and fine-tuning code.
Yi Shu, Tianyu Peng, Yingzhuo Deng +5
School of Artificial Intelligence, University of Chinese Academy of Sciences · Institute of Automation, Chinese Academy of Sciences · Wuhan AI Research +1