---
title: "Unsupervised Data Selection for TTS — Baali 2023"
description: "The paper asks whether usable Arabic text-to-speech can be built without a studio corpus by reusing broadcast news collected for ASR."
authors: ["Massa Baali", "Tomoki Hayashi", "Hamdy Mubarak", "Soumi Maiti", "Shinji Watanabe", "Wassim El-Hajj", "Ahmed Ali"]
year: 2023
venue: "arXiv:2301.09099"
source: "https://arxiv.org/abs/2301.09099"
doi: "10.48550/arxiv.2301.09099"
arxiv: "2301.09099"
licence: "arXiv — non-exclusive license to distribute"
licence_url: "http://arxiv.org/licenses/nonexclusive-distrib/1.0/"
summary_licence: "Elamur Cosmic License"
canonical: "https://research.elamur.ai/free/en/papers/baali-2023-unsupervised-data-selection-tts-using/"
---
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# Unsupervised Data Selection for TTS: Using Arabic Broadcast News as a Case Study

**Authors:** [Massa Baali](https://md.research.elamur.ai/free/en/authors/massa-baali/), Tomoki Hayashi, [Hamdy Mubarak](https://md.research.elamur.ai/free/en/authors/hamdy-mubarak/), Soumi Maiti, [Shinji Watanabe](https://md.research.elamur.ai/free/en/authors/shinji-watanabe/), [Wassim El-Hajj](https://md.research.elamur.ai/free/en/authors/wassim-el-hajj/), [Ahmed Ali](https://md.research.elamur.ai/free/en/authors/ahmed-ali/) · **Year:** 2023 · **Venue:** arXiv:2301.09099 · **Licence:** [arXiv — non-exclusive license to distribute](http://arxiv.org/licenses/nonexclusive-distrib/1.0/)

**Links:** [Publisher's page](https://arxiv.org/abs/2301.09099) · [PDF](https://arxiv.org/pdf/2301.09099) · [DOI 10.48550/arxiv.2301.09099](https://doi.org/10.48550/arxiv.2301.09099) · [arXiv 2301.09099](https://arxiv.org/abs/2301.09099) · [OpenAlex](https://openalex.org/W4317940367)

**Citations:** 4 citations (openalex-snapshot · 2026-10-14) · [5](#cited-by) in our library, of the 140884 papers whose bibliographies we read

## Summary
The paper asks whether usable Arabic text-to-speech can be built without a studio corpus by reusing broadcast news collected for ASR. The authors start from Aljazeera recordings in MGB-2/QASR and keep only two anchor speakers, one male and one female. They then filter for clean segments in two ways: a linguist sorted segments into six classes, and three automatic MOS predictors (MOSNet, wv-MOS, DNSMOS) scored each segment, with a threshold of 4. An Arabic ASR system repairs faulty transcripts, and automatic diacritization adds the missing vowels. For modelling, Tacotron2 and Transformer-TTS are pre-trained on English LJSpeech and fine-tuned on about one hour of Arabic. These autoregressive models then provide durations to a FastSpeech2-based Conformer. The final system reaches 3.9% CER, against 1.3% for real recordings, and scores MOS 4.4 for intelligibility and 4.2 for naturalness. Listeners often recognized the actual broadcaster. The study shows that a small, carefully filtered subset of found data can be enough to build TTS for a language with few resources.

## Key points
- **Source data:** MGB-2 Aljazeera broadcasts from 2004 to 2015, covering more than 4,000 episodes across 19 programs, with audio at 16 kHz. The speaker metadata was messy; Barack Obama alone appeared under 9 name variants. It was cleaned through repeated automatic parsing followed by manual checks.
- **Manual labelling:** Among the anchors' segments, the two biggest problems were bad recording quality (40%) and overlapping speech (33%). About one hour of clean audio per speaker survived.
- **Automatic selection:** DNSMOS and wv-MOS agreed closely with the human "good" labels. A TTS trained on the DNSMOS selection reached 4.0% CER, compared with 3.9% for the hand-picked selection.
- **Diacritization:** Adding vowels to the transcripts cut CER from 32.2% to 3.9%.
- **Training and release:** The AR models start from weights pre-trained on LJSpeech (24 h). The token embedding layer is not transferred because the symbol sets differ, and guided attention loss is added. Audio is generated with a Parallel WaveGAN vocoder. The recipe is published in ESPnet as `qasr_tts`.

## Interesting ideas
- **Name the failure types first, then validate the filter:** The authors defined explicit classes of bad data on a sample labelled by an expert, and only then checked an automatic filter against that sample. The filter's quality becomes measurable instead of assumed. — For Mova (learn): Classify silver-tree failures (e.g. wrong attachment, tokenization error, garbage input) on a hand-checked subset. Then measure how well witness-agreement filtering recovers the trees that humans judged good.
- **A small clean set can beat a large noisy one:** A cheap quality score with a fixed threshold produced a training subset that matched expert selection. This suggests aggressive filtering often pays off more than collecting more volume. — For Mova (learn): Train parsers only on silver sentences above a strict agreement threshold, and compare against the full silver pool to show how much noise costs.
- **Repair good samples instead of discarding them:** Segments with clean audio but faulty transcripts were corrected with a second model (ASR) rather than thrown away. This keeps scarce data that would otherwise be lost. — For Mova (learn): When witness parsers disagree on only one arc, apply self-induced repair rules to fix that arc. This keeps the rest of the sentence's tree rather than dropping it.
- **Restore missing input information before modelling:** Adding diacritics, which the text simply lacked, made a larger difference than any architecture choice. — For Mova (tokens): Normalize information that is often missing from input before tagging, such as dropped apostrophes, casing, and contractions. This keeps downstream errors from being blamed on the parser.
- **Use a slow, robust model to train a fast one:** The autoregressive model, which learns alignment reliably, provided durations to the faster non-autoregressive model, which could not learn alignment well from one hour of data. — For Mova (tree): The GPU transformer parser gparse could provide arc or label targets for training the faster perceptron parser on unlabelled text.

*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](https://md.research.elamur.ai/free/en/shelves/linguistics/)

## Cited by

Cited by

5 papers of our library cite it (of the 140884 whose bibliographies we read); 2 of them have a page here, 3 not yet.

- [FOOCTTS: Generating Arabic Speech with Acoustic Environment for Football Commentator](https://md.research.elamur.ai/free/en/papers/baali-2023-fooctts-generating-arabic-speech-acoustic/)
- [ClArTTS: An Open-Source Classical Arabic Text-to-Speech Corpus](https://md.research.elamur.ai/free/en/papers/kulkarni-2023-clartts-open-source-classical-arabic/)

## Cite

The paper itself, not this page. BibTeX from the library's record of the work.

```bibtex
@misc{baali2023unsupervised,
  title = {{Unsupervised Data Selection for TTS: Using Arabic Broadcast News as a Case Study}},
  author = {Baali, Massa and Hayashi, Tomoki and Mubarak, Hamdy and Maiti, Soumi and Watanabe, Shinji and El-Hajj, Wassim and Ali, Ahmed},
  year = {2023},
  doi = {10.48550/arxiv.2301.09099},
  eprint = {2301.09099},
  archivePrefix = {arXiv},
  url = {https://doi.org/10.48550/arxiv.2301.09099}
}
```

- [← previous Unravelling Indirect Answers to Wh-Questions: Corpus Construction, Analysis, and Generation](https://md.research.elamur.ai/free/en/papers/yusupujiang-2023-unravelling-indirect-answers-wh-questions/)
- [next → Using LSTM and GRU With a New Dataset for Named Entity Recognition in the Arabic Language](https://md.research.elamur.ai/free/en/papers/shaker-2023-using-lstm-gru-new-dataset/)
- [↑ back to papers of 2023](https://md.research.elamur.ai/free/en/papers/2023/page/8/)

## References

31 references, 5 of them with a card of ours. ↗ leads to the work, «find» searches for one we have no id for yet.

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28. P. Guo, F. Boyer, X. Chang, T. Hayashi, Y. Higuchi, H. Inaguma, N. Kamo, C. Li, D. Garcia-Romero, J. Shi, et al., “Recent developments on ESPnet toolkit boosted by conformer,” ICASSP, 2021. [find](https://scholar.google.com/scholar?q=P%2E+Guo%2C+F%2E+Boyer%2C+X%2E+Chang%2C+T%2E+Hayashi%2C+Y%2E+Higuchi%2C+H%2E+Inaguma%2C+N%2E+Kamo%2C+C%2E+Li%2C+D%2E+Garcia%2DRomero%2C+J%2E+Shi%2C+et+al%2E%2C+%E2%80%9CRecent+developments+on+ESPnet+toolkit+boosted+by+conformer%2C%E2%80%9D+ICASSP%2C+2021%2E)
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30. N. Perraudin, P. Balazs, and P. L. Søndergaard, “A fast griffin-lim algorithm,” in 2013 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics. [find](https://scholar.google.com/scholar?q=N%2E+Perraudin%2C+P%2E+Balazs%2C+and+P%2E+L%2E+S%C3%B8ndergaard%2C+%E2%80%9CA+fast+griffin%2Dlim+algorithm%2C%E2%80%9D+in+2013+IEEE+Workshop+on+Applications+of+Signal+Processing+to+Audio+and+Acoustics%2E)
31. R. Yamamoto, E. Song, and J. Kim, “Parallel wavegan: A fast waveform generation model based on generative adversarial networks with multi-resolution spectrogram,” in ICASSP, 2020. [find](https://scholar.google.com/scholar?q=R%2E+Yamamoto%2C+E%2E+Song%2C+and+J%2E+Kim%2C+%E2%80%9CParallel+wavegan%3A+A+fast+waveform+generation+model+based+on+generative+adversarial+networks+with+multi%2Dresolution+spectrogram%2C%E2%80%9D+in+ICASSP%2C+2020%2E)
