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Arabic Text-To-Speech (TTS) Data Preparation

Summary

This paper offers practical guidance on building recorded data for Arabic text-to-speech. The authors argue that Arabic lags behind other languages in TTS and that the quality of the human-recorded ground truth largely decides how natural and intelligible the resulting voice will be. They combine a linguistic view with a studio-production view. They first outline a TTS pipeline aimed at banking and finance: text normalization, diacritization, spell checking, utterance segmentation and audio generation. They then describe the data work.

  • Selecting the corpus: open-domain or closed-domain text, phonetically balanced, with controlled utterance length.
  • Diacritizing the text: partial or full, to remove lexical ambiguity.
  • Directing the recording sessions: voice-actor selection, tempo, session length, accent, intonation, pauses and noise.

Finally, they propose ways for listeners to judge speech quality: naturalness, degraded MOS, intelligibility and comprehensibility tests, plus a checklist of perceptual factors. The paper reports no experiments or released dataset. Its value lies in the specifications it gives to voice coaches, actors and annotators working on a low-resource, diacritic-dependent language.

Key points

  • The proposed pipeline: normalization turns numbers, dates, abbreviations and foreign words into Arabic text, then diacritization, an Arabic spell checker, utterance segmentation, and finally audio generation. The target domain is finance and banking.
  • The data should be phonetically balanced and balanced in length. Phrases are capped at about 12 units, roughly what a speaker can say without pausing.
  • The paper tabulates eight Arabic diacritics with IPA and Buckwalter equivalents. Partial diacritization is presented as a cheaper way to resolve ambiguity, for example the many readings of an undiacritized word meaning "study/teach/lesson".
  • Recording rules include:
    • at most about 2 hours of recording per day;
    • a 10-minute break every hour;
    • returning to reference sentences after every 50 utterances;
    • re-recording instead of technical cuts;
    • pausing only at punctuation, with short vowels dropped at pauses;
    • keeping the delivery natural rather than acted.
  • Evaluation uses four tests: naturalness, degraded MOS with randomized natural and synthetic clips, intelligibility (MOS and WER from listener transcriptions), and comprehensibility (questions scored 0/1).

Interesting ideas

  • I1. Morphological analyzers can use whatever diacritics are present as hard filters, discarding any analysis that contradicts them, so sparse annotation still prunes ambiguity cheaply. — For Mova (tokens): Partial cues such as a known lemma, a capitalization hint or a silver POS can act as constraints that eliminate candidate analyses in the tagger, rather than as soft features.
  • I2. Speakers periodically go back to a fixed set of reference sentences so that tempo and style do not drift over a long session. — For Mova (learn): A small frozen anchor set can be re-scored at regular intervals during training and self-play to catch drift before it contaminates silver data.
  • I3. Defects are fixed by redoing the whole recording, never by splicing fragments, because patches break consistency. — For Mova (back): When generated text from the text → graph → text compressor has a local error, regenerating the whole sentence from the graph should yield more coherent output than patching words in place.
  • I4. Intelligibility (writing down what was heard, regardless of meaning) is tested separately from comprehensibility (answering questions about the content). — For Mova (back): The compressor can be scored twice, once on surface reconstruction and once on whether Mova's readers answer the same questions on the regenerated text as on the original.
  • I5. Evaluators rate a shuffled mix of real human recordings and synthetic ones, so the human items act as a built-in reference and bias check. — For Mova (learn): Gold human trees and frames can be hidden among model outputs when LLM teachers or witness parsers judge quality, serving as negative and positive controls for the judges themselves.

Our summary of the paper, in our own words — not its text. The work belongs to its authors; read it at the links above.

Fields: Linguistics & languages

Cite

The paper itself, not this page. BibTeX from the library's record of the work. DOI 10.48550/arxiv.2204.03255 · arXiv 2204.03255

@misc{masri2022arabic,
  title = {{Arabic Text-To-Speech (TTS) Data Preparation}},
  author = {Masri, Hala Al and Za'ter, Muhy Eddin},
  year = {2022},
  doi = {10.48550/arxiv.2204.03255},
  eprint = {2204.03255},
  archivePrefix = {arXiv},
  url = {https://doi.org/10.48550/arxiv.2204.03255}
}

References

14 references, 1 of them with a card of ours. ↗ leads to the work, «find» searches for one we have no id for yet.

  1. Paul Taylor. Text-to-speech synthesis. Cambridge university press, 2009. find
  2. Andrew J Hunt and Alan W Black. Unit selection in a concatenative speech synthesis system using a large speech database. In 1996 IEEE International Conference on Acoustics, Speech, and Signal Processing Conference Proceedings, volume 1, pages 373–376. IEEE, 1996. find
  3. Alan W Black and Paul A Taylor. Automatically clustering similar units for unit selection in speech synthesis. 1997. find
  4. Keiichi Tokuda, Takayoshi Yoshimura, Takashi Masuko, Takao Kobayashi, and Tadashi Kitamura. Speech parameter generation algorithms for hmm-based speech synthesis. In 2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No. 00CH37100), volume 3, pages 1315–1318. IEEE, 2000. find
  5. Keiichi Tokuda, Yoshihiko Nankaku, Tomoki Toda, Heiga Zen, Junichi Yamagishi, and Keiichiro Oura. Speech synthesis based on hidden markov models. Proceedings of the IEEE, 101(5):1234–1252, 2013. find
  6. Heiga Zen, Keiichi Tokuda, and Alan W Black. Statistical parametric speech synthesis. speech communication, 51(11):1039–1064, 2009. find
  7. Heiga Ze, Andrew Senior, and Mike Schuster. Statistical parametric speech synthesis using deep neural networks. In 2013 ieee international conference on acoustics, speech and signal processing, pages 7962–7966. IEEE, 2013. find
  8. Aaron van den Oord, Sander Dieleman, Heiga Zen, Karen Simonyan, Oriol Vinyals, Alex Graves, Nal Kalchbrenner, Andrew Senior, and Koray Kavukcuoglu. Wavenet: A generative model for raw audio. arXiv preprint arXiv:1609.03499, 2016. ↗ our card
  9. Yuxuan Wang, RJ Skerry-Ryan, Daisy Stanton, Yonghui Wu, Ron J Weiss, Navdeep Jaitly, Zongheng Yang, Ying Xiao, Zhifeng Chen, Samy Bengio, et al. Tacotron: Towards end-to-end speech synthesis. arXiv preprint arXiv:1703.10135, 2017. ↗
  10. Akira Tamamori, Tomoki Hayashi, Kazuhiro Kobayashi, Kazuya Takeda, and Tomoki Toda. Speaker-dependent wavenet vocoder. In Interspeech, volume 2017, pages 1118–1122, 2017. find
  11. Jonathan Shen, Ruoming Pang, Ron J Weiss, Mike Schuster, Navdeep Jaitly, Zongheng Yang, Zhifeng Chen, Yu Zhang, Yuxuan Wang, Rj Skerrv-Ryan, et al. Natural tts synthesis by conditioning wavenet on mel spectrogram predictions. In 2018 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), pages 4779–4783. IEEE, 2018. find
  12. Taha Zerrouki, Mohammed M Abu Shquier, Amar Balla, Nabila Bousbia, Imededdine Sakraoui, and Fateh Boudardara. Adapting espeak to arabic language: Converting arabic text to speech language using espeak. International Journal of Reasoning-Based Intelligent Systems, 11(1):76–89, 2019. find
  13. Fady K Fahmy, Mahmoud I Khalil, and Hazem M Abbas. A transfer learning end-to-end arabic text-to-speech (tts) deep architecture. In IAPR Workshop on Artificial Neural Networks in Pattern Recognition, pages 266–277. Springer, 2020. find
  14. MOHAMED BEN MERIEM. Phonetic transcription of arabic. International Journal of Natural Language Processing, 1(1), 2020. find

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