O.E. Scharenborg
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60 records found
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First, a data audit of four public datasets shows that simple classifiers can identify which dataset a recording came from using several kinds of audio measurements. This establishes that source-related clues are available, but not that the deepfake detectors use them. To test detector reliance directly, the thesis introduces the mixing ratio protocol. The protocol changes which of two datasets supplies the genuine and deepfake recordings during training, while the evaluation recordings remain fixed. If a detector uses dataset source as a shortcut, reversing this relationship should cause its decisions to change.
This is what occurs. Across all three detectors, reversing the training composition moves the equal error rate (EER) on the same test recordings from 0.003–0.028 to 0.78–0.87. An EER of 0 is perfect, 0.5 is chance, and a value above 0.5 means that the predictions are systematically inverted. A similar response appears with a second dataset pair. By contrast, the results are much flatter when the same procedure is applied to two groups of speakers from one dataset. This control shows that the mixing procedure does not create the main response by itself.
Further experiments test whether one audio property explains the dependence. Reducing the frequency range, matching the average signal level, and removing low-frequency energy do not remove it. Dataset source can also be predicted from several different kinds of measurements. These findings are consistent with source information being represented through several overlapping or untested properties, but they do not identify which properties each detector used. Overall, dataset identity is available as an exploitable shortcut in speech deepfake detection. Future benchmarks should therefore include diverse recording and generation pipelines in both genuine and deepfake classes. ...
First, a data audit of four public datasets shows that simple classifiers can identify which dataset a recording came from using several kinds of audio measurements. This establishes that source-related clues are available, but not that the deepfake detectors use them. To test detector reliance directly, the thesis introduces the mixing ratio protocol. The protocol changes which of two datasets supplies the genuine and deepfake recordings during training, while the evaluation recordings remain fixed. If a detector uses dataset source as a shortcut, reversing this relationship should cause its decisions to change.
This is what occurs. Across all three detectors, reversing the training composition moves the equal error rate (EER) on the same test recordings from 0.003–0.028 to 0.78–0.87. An EER of 0 is perfect, 0.5 is chance, and a value above 0.5 means that the predictions are systematically inverted. A similar response appears with a second dataset pair. By contrast, the results are much flatter when the same procedure is applied to two groups of speakers from one dataset. This control shows that the mixing procedure does not create the main response by itself.
Further experiments test whether one audio property explains the dependence. Reducing the frequency range, matching the average signal level, and removing low-frequency energy do not remove it. Dataset source can also be predicted from several different kinds of measurements. These findings are consistent with source information being represented through several overlapping or untested properties, but they do not identify which properties each detector used. Overall, dataset identity is available as an exploitable shortcut in speech deepfake detection. Future benchmarks should therefore include diverse recording and generation pipelines in both genuine and deepfake classes.
ASR Systems for Deaf/Hard-of Hearing People
Targeted Data Augmentation and Personalized Fine-Tuning for Dutch d/Dhh Speech Recognition
This thesis investigates whether MoE delivers that promise on dysarthric ASR, using the FastConformer-CTC Large encoder fine-tuned per fold on the TORGO and UASpeech corpora across 41 leave-one-speaker-out folds. Three research questions are addressed: how standard sparsely-gated MoE compares to a fine-tuned dense baseline (RQ1); what specialization the trained experts develop (RQ2); and whether either training-time supervision or inference-time router steering can push the model toward useful specialization (RQ3).
The answer to all three questions is negative. Five standard MoE configurations match but do not beat the fine-tuned dense baseline, and a parameter-equal follow-up shows that at equal feed-forward capacity, the MoE structure is actively worse than dense by up to 2.89 pp. The trained experts develop genuine functional specialization, but the axis of specialization is encoder depth, not speaker group or phonetic content: different layers prefer different expert pairs, and within any layer, routing is roughly content-independent and group-independent. Training-time supervision toward group-aware specialization collapses at inference, as the depth-based organization overrides the supervised partition. Inference-time router steering produces real per-speaker improvements under post-hoc selection of the bias strength, but neither of two deployment-realistic mechanisms for choosing that bias recovers the gain.
The MoE structure on this task does meaningful work, but not the work the task needs: it organizes the encoder’s computation along an axis the dense baseline already covers through its per-layer feed-forward weights. Routing-input variants, cross-architecture replication, and continual-learning extensions are flagged as the most direct follow-ups. ...
This thesis investigates whether MoE delivers that promise on dysarthric ASR, using the FastConformer-CTC Large encoder fine-tuned per fold on the TORGO and UASpeech corpora across 41 leave-one-speaker-out folds. Three research questions are addressed: how standard sparsely-gated MoE compares to a fine-tuned dense baseline (RQ1); what specialization the trained experts develop (RQ2); and whether either training-time supervision or inference-time router steering can push the model toward useful specialization (RQ3).
The answer to all three questions is negative. Five standard MoE configurations match but do not beat the fine-tuned dense baseline, and a parameter-equal follow-up shows that at equal feed-forward capacity, the MoE structure is actively worse than dense by up to 2.89 pp. The trained experts develop genuine functional specialization, but the axis of specialization is encoder depth, not speaker group or phonetic content: different layers prefer different expert pairs, and within any layer, routing is roughly content-independent and group-independent. Training-time supervision toward group-aware specialization collapses at inference, as the depth-based organization overrides the supervised partition. Inference-time router steering produces real per-speaker improvements under post-hoc selection of the bias strength, but neither of two deployment-realistic mechanisms for choosing that bias recovers the gain.
The MoE structure on this task does meaningful work, but not the work the task needs: it organizes the encoder’s computation along an axis the dense baseline already covers through its per-layer feed-forward weights. Routing-input variants, cross-architecture replication, and continual-learning extensions are flagged as the most direct follow-ups.
Improving Automatic Speech Recognition for Dutch Children with Developmental Language Disorder using Synthetic Data
Synthetic Speech Augmentation Using Text-to-Speech
Our results demonstrate that applying a moderate probability of linguistically motivated DLD transformations significantly outperforms standard, error-free childlike synthetic speech. Despite this relative linguistic success, the synthetic fine-tuning did not surpass the absolute performance of the unadapted zero-shot baseline. Furthermore, experiments indicate that scaling synthetic speaker variability introduces disruptive generation artifacts rather than improving model robustness. These findings demonstrate the validity of targeted linguistic simulation for atypical speech while exposing the severe acoustic limitations of current zero-shot TTS models in clinical, data-scarce domains. ...
Our results demonstrate that applying a moderate probability of linguistically motivated DLD transformations significantly outperforms standard, error-free childlike synthetic speech. Despite this relative linguistic success, the synthetic fine-tuning did not surpass the absolute performance of the unadapted zero-shot baseline. Furthermore, experiments indicate that scaling synthetic speaker variability introduces disruptive generation artifacts rather than improving model robustness. These findings demonstrate the validity of targeted linguistic simulation for atypical speech while exposing the severe acoustic limitations of current zero-shot TTS models in clinical, data-scarce domains.
speakers. For Dutch, previous evaluations found higher word error rates for native children than for native teenagers, but they did not show which types of words contribute to the difference. This paper studies whether that difference is concentrated in particular word categories. A word category is the part of speech assigned to a word in the reference transcript, for example noun, verb, pronoun, article, or preposition. The analysis uses native Dutch child and teenager speech from the JASMIN-CGN corpus in two speaking styles: read speech and speech from human machine interaction tasks. The analysis compares Google Chirp 2 and Whisper-large-v3 outputs with the reference transcripts and groups substitutions and deletions by word category. Insertions are reported separately because an inserted system word has no matching reference word category. Children have higher word error rates than teenagers in every condition. The difference is about 9 percentage points in read speech and about 4 to 5 percentage points in human machine interaction speech. In read speech, most of the substitution and deletion difference comes from nouns, verbs, and pronouns. In human machine interaction speech, errors are spread across categories, so no single category explains much of the difference. A control that compares utterances with the same counts of word categories reduces the verb and pronoun differences, while the noun difference remains largest. A check of the specific words behind these categories shows why this matters: many noun and verb errors come from words found only in the child read reference transcripts, whereas pronoun errors mostly involve words used by both groups. The results show that word categories are useful for locating where the error rate difference appears, but they are not a complete explanation without checking the specific words and speaking material behind
each category. ...
speakers. For Dutch, previous evaluations found higher word error rates for native children than for native teenagers, but they did not show which types of words contribute to the difference. This paper studies whether that difference is concentrated in particular word categories. A word category is the part of speech assigned to a word in the reference transcript, for example noun, verb, pronoun, article, or preposition. The analysis uses native Dutch child and teenager speech from the JASMIN-CGN corpus in two speaking styles: read speech and speech from human machine interaction tasks. The analysis compares Google Chirp 2 and Whisper-large-v3 outputs with the reference transcripts and groups substitutions and deletions by word category. Insertions are reported separately because an inserted system word has no matching reference word category. Children have higher word error rates than teenagers in every condition. The difference is about 9 percentage points in read speech and about 4 to 5 percentage points in human machine interaction speech. In read speech, most of the substitution and deletion difference comes from nouns, verbs, and pronouns. In human machine interaction speech, errors are spread across categories, so no single category explains much of the difference. A control that compares utterances with the same counts of word categories reduces the verb and pronoun differences, while the noun difference remains largest. A check of the specific words behind these categories shows why this matters: many noun and verb errors come from words found only in the child read reference transcripts, whereas pronoun errors mostly involve words used by both groups. The results show that word categories are useful for locating where the error rate difference appears, but they are not a complete explanation without checking the specific words and speaking material behind
each category.
A Whisper-based encoder-DNN-CTC model is extended with a triplet-loss objective to improve phoneme-level discrimination. Four negative sampling strategies are compared: randomly selected, phonologically motivated, and two empirically derived from the model's own prediction errors, one estimated on the training set and one via cross-validation. Each is evaluated under two training regimes: contrastive fine-tuning of a pretrained model and training from scratch.
All contrastive approaches significantly outperform a CTC-only baseline. The strongest results are obtained with phonologically motivated and cross-validation-based empirical negatives when training from scratch, yielding up to a 10.7% relative reduction in phoneme error rate. Under fine-tuning, differences between sampling strategies are negligible. In contrast, when trained from scratch, the phonological and cross-validation-based empirical strategies significantly outperform randomly selected and training-set-based empirical negatives.
These findings suggest that, for this speaker, contrastive learning for dysarthric speech benefits from phonologically informed or empirically derived negative pairs rather than random selection. A practical trade-off emerges between the two strongest strategies: phonologically motivated sampling requires no speaker-specific preprocessing and is immediately applicable to new speakers, but generates a large number of triplets and is computationally expensive at training time. Cross-validation-based empirical sampling requires building a speaker-specific confusion matrix upfront, but produces fewer, more targeted triplets and trains more efficiently. Given comparable performance, the choice between them reduces to whether preprocessing overhead or training-time resources are the limiting constraint. ...
A Whisper-based encoder-DNN-CTC model is extended with a triplet-loss objective to improve phoneme-level discrimination. Four negative sampling strategies are compared: randomly selected, phonologically motivated, and two empirically derived from the model's own prediction errors, one estimated on the training set and one via cross-validation. Each is evaluated under two training regimes: contrastive fine-tuning of a pretrained model and training from scratch.
All contrastive approaches significantly outperform a CTC-only baseline. The strongest results are obtained with phonologically motivated and cross-validation-based empirical negatives when training from scratch, yielding up to a 10.7% relative reduction in phoneme error rate. Under fine-tuning, differences between sampling strategies are negligible. In contrast, when trained from scratch, the phonological and cross-validation-based empirical strategies significantly outperform randomly selected and training-set-based empirical negatives.
These findings suggest that, for this speaker, contrastive learning for dysarthric speech benefits from phonologically informed or empirically derived negative pairs rather than random selection. A practical trade-off emerges between the two strongest strategies: phonologically motivated sampling requires no speaker-specific preprocessing and is immediately applicable to new speakers, but generates a large number of triplets and is computationally expensive at training time. Cross-validation-based empirical sampling requires building a speaker-specific confusion matrix upfront, but produces fewer, more targeted triplets and trains more efficiently. Given comparable performance, the choice between them reduces to whether preprocessing overhead or training-time resources are the limiting constraint.
Comparing Human Listeners and Dutch ASR on Transcribing Child Speech
The Effect of Familiarity with Child Speech on Transcription Performance
A balanced set of 40 utterances were taken from the JASMIN database (speakers aged 7-11), and were transcribed by 20 humans (10 familiar with child speech (parent/caretaker) and 10 unfamiliar). Transcripts were also gathered from two state-of-the-art ASR systems (Google Telephony and a Conformer model). These transcripts were evaluated against reference transcripts using Word Error Rate (WER). Statistical significance was tested.
Results show that overall ASR transcription performance was comparable to human performance, and in some cases slightly, but not significantly, better. Familiar listeners did not outperform unfamiliar listeners. In fact, there was no significant performance difference between the two groups of humans. Within the 7-11 age range, no clear relationship between speaker age and WER was shown, but results were sensitive to sentence difficulty outliers and "speaker effects". ...
A balanced set of 40 utterances were taken from the JASMIN database (speakers aged 7-11), and were transcribed by 20 humans (10 familiar with child speech (parent/caretaker) and 10 unfamiliar). Transcripts were also gathered from two state-of-the-art ASR systems (Google Telephony and a Conformer model). These transcripts were evaluated against reference transcripts using Word Error Rate (WER). Statistical significance was tested.
Results show that overall ASR transcription performance was comparable to human performance, and in some cases slightly, but not significantly, better. Familiar listeners did not outperform unfamiliar listeners. In fact, there was no significant performance difference between the two groups of humans. Within the 7-11 age range, no clear relationship between speaker age and WER was shown, but results were sensitive to sentence difficulty outliers and "speaker effects".
Human vs AI: Recognising Teenage Speech
How good are humans at recognizing teenage speech samples compared to state-of-the-art AI-based automatic speech recognisers?
State-of-the-art Automatic Speech Recognition Systems on Dutch Regional Dialects
Exploring Bias in Dutch-trained Automatic Speech Recognition Systems