Public benchmarks are important measures of Automatic Speech Recognition (ASR) model capabilities. However, by nature of being public, there is risk of models being optimized for these benchmarks in ways that do not generalize well to real-world data. We present a methodology for quantifying benchmark optimization, focusing on cases where the audio underdetermines the reference transcript. We identify three families of behavioral probes that reveal models' capabilities of reproducing benchmark reference spans despite underdetermined audio: reference disagreement, masked-number recovery, and orthographic switching. We find that the highest-scoring open source models output verbatim reference transcript spans even when the relevant audio is contradictory, masked, or ambiguous. Using a variety of mechanistic probes, we show that models respond to narrow acoustic cues to override the faithful representation of the audio in favor of a benchmark-optimized policy. We show the benchmark-optimized behavior can be causally manipulated via low-rank linear steering or simply appending audio to the end of a segment in some cases. Overall, our results indicate that high-performing models exhibit benchmark-conditioned behaviors that can inflate benchmark performance without reflecting improved general-purpose transcription ability.
Figures & tables
Probe case
Reference rendering r
Comparison rendering a
Reference insertion
we should return to the Geneva format
we should return to Geneva format
Reference deletion
Mr. President
Thank you Mr. President
Reference substitution
in a position
in a role
Masked word
we also had 50 international US experts
we also had <masked> international US experts
Orthographic variant
Mr President
Mister President
Figure 1: Example renderings used by the behavioral probes. Red marks the reference span being tested; green marks the competing audio-true or acoustically equivalent rendering. Reference insertion, deletion, and substitution are the three edit types used by the reference-disagreement probe. Masking induces a deletion-style comparison by silencing a word (where <masked> indicates a silent audio interval, not a literal model output), while orthographic switching compares two acoustically identical spellings.
Figure 2: Cross-model audit on VoxPopuli. WER (%) is the VoxPopuli-test score from the June 2026 Open ASR Leaderboard [ 38 ] . Kimi Audio is not on the leaderboard, and its score is computed using the leaderboard’s scoring. Consensus-panel members are scored against edits flagged unanimously by the remaining three members (§ 3.3 ).
Figure 3: ( 3(a) ) Masked-number accept-ref per corpus. ( 3(b) ) Orthographic switch rate on archaic spacing.
Figure 4: Difference in masked-number recovery between LibriSpeech test-set narrator clones and held-out libri-fresh narrator clones reading identical sentences (test-set minus held-out; positive values indicate greater recovery for test-set narrator clones). Sentence-clustered bootstrap 95% confidence intervals.
Figure 5: The trigger battery, VoxPopuli. Top row: voice conditions on identical transcripts—( 5(a) ) reference-disagreement and ( 5(b) ) masked-number accept-ref . Bottom row: the same probes with the trigger removed instead of the voice varied—truncation to the edit, an appended 8 s conversational donor, or the learned register direction projected out of one encoder layer ( 5(c) , 5(d) ). Wilson 95% CIs.
Figure 6: Switching the benchmark-optimized policy on and off. ( 6(a) ) Input level: On real clips (top left) a conversational donor collapses accept-ref while a VoxPopuli donor leaves it intact; on ep-fresh clones of the same sentences (top right) a VoxPopuli donor re-ignites it while the conversational donor does not. ( 6(b) ) Activation level: projecting out the learned direction on real benchmark edits (bottom left) and adding it on generic-voice clones (bottom right). The remaining consensus-panel members show no effect, like Voxtral-Mini-3B.
Appendix figures & tables12 assets
Supplementary material from the paper’s appendix.
Appendix
consensus
human-annotated edits
model
accept-ref
accept-ref
95% CI
n
Cohere-Transcribe
0.30
0.52
[0.47, 0.58]
253/483
Granite-Speech-4.1-2B
0.21
0.42
[0.36, 0.47]
211/508
Canary-Qwen-2.5B
0.23
0.41
[0.36, 0.47]
210/507
Higgs-Audio-v3-8B
0.21
0.39
[0.33, 0.44]
200/517
Phi-4-Multimodal
0.19
0.38
[0.33, 0.44]
195/510
Appendix
Table 1: accept-ref on VoxPopuli-AA human-annotated edits, beside the consensus rates of Figure 2 . Ordering is nearly identical aside from Granite and Canary swapping places.
Figure 7: Reference-disagreement accept-ref on the real VoxPopuli recordings under content-preserving perturbations (additive noise 10 dB; measured room reverberation, RT60 0.60 ). Wilson 95% CIs.
Figure 8: Honorific switch rate ( Mr / Mister ), all 11 models. A rate above 0.5 (dashed) means the model tracks each corpus’s convention at rates above chance.
Figure 9: Base-anchored Mister rate for the same sentence rendered in a cloned VoxPopuli-speaker voice vs. a cloned LibriSpeech-narrator voice; ticks mark the real-recording rate. Left: VoxPopuli Mr sentences ( n=47 ; voice transfer). Right: LibriSpeech Mister sentences ( n=108 ; voice erosion). Granite retains conventions on real audio but loses them under any TTS clone, so its clone cells are relatively uninformative; Whisper, Phi-4, and Voxtral never emit Mister .
Figure 10: Audio lift λ(r) (Eq. 1 ), in nats per character; bootstrap 95% CIs. Parakeet-TDT is excluded (no teacher-forced readout). ( 10(a) ) The reference disagreement audio lift. ( 10(b) ) The masked entity audio lift for different voice conditions for 115 paired sentences.
model
full
truncated
donor ablated
activation ablated
Cohere-Transcribe
0.30
0.13
0.06
0.04
Canary-Qwen-2.5B
0.23
0.12
0.05
0.02
Granite-Speech-4.1-2B
0.21
0.12
0.20
0.14
Higgs-Audio-v3-8B
0.21
0.12
0.03
–
Phi-4-Multimodal
0.19
0.09
0.05
0.20
Parakeet-TDT-0.6B-v2
0.18
0.08
0.06
0.01
Appendix
Table 2: Reference-disagreement accept-ref on consensus edits as audio context is removed or the trigger is ablated. truncated cuts the audio to a tight window around the edit span ( ±1 aligned word ±0.25 s); donor ablated appends an 8 s conversational donor to the full clip; activation ablated projects the learned register direction out of a single encoder layer.
consensus accept-ref
masked accept-ref
model
VoxPopuli
ep-fresh
VoxPopuli
ep-fresh
Cohere-Transcribe
0.304
0.122
0.185
0.074
Canary-Qwen-2.5B
0.233
0.150
0.051
0.062
Granite-Speech-4.1-2B
0.208
0.117
0.051
0.062
Higgs-Audio-v3-8B
0.211
0.168
0.070
0.040
Phi-4-Multimodal
0.188
0.193
0.076
0.044
Appendix
Table 3: Full-probe accept-ref on real audio with each corpus’s own content: VoxPopuli-test vs ep-fresh . Masked columns score the corpus-paired subsets, so the VoxPopuli masked rates differ from the full masked set of § 4 .
Figure 11: Context gating on all consensus edits, conditioned on reproduction: for each model with a high accept-ref , the edits it reproduces on the full clip, re-decoded on the isolated window of Table 2 . Reference-extra words mostly disappear and suppressed audible words resurface once the surrounding benchmark context is cut away. Wilson 95% CIs.
Figure 12: Edit-locus dissociation (Cohere, Granite, Canary-Qwen). Isolating the span restores faithful transcription for both edit types (context gating; the all-model version is Figure 11 ). Patching in a context-free encoding of the span removes reference insertions (encoder-side) but never restores reference omissions (decoder-side).
Figure 13: Masked accept-ref , real VoxPopuli minus a register-matched held-out clone (fresh post-cutoff parliament), same transcript, paired 95% CI. Positive means the model reads a silenced number from the benchmark voice but not a held-out voice; significant for Cohere and Canary (red).
real
vox-clone
ep-fresh
generic
real − ep-fresh
real − generic
Cohere-Transcribe
+1.52
+1.26
+0.92
+0.54
+0.60[+0.30,+0.94]
+0.98[+0.66,+1.33]
Canary-Qwen-2.5B
+1.22
+1.05
+0.65
+0.77
+0.57[+0.27,+0.90]
+0.45[+0.17,+0.71]
Granite-Speech-4.1-2B
+0.12
+0.15
+0.16
+0.01
−0.04[−0.24,+0.17]
+0.11[−0.11,+0.32]
Phi-4-Multimodal
+0.46
+0.51
+0.24
+0.27
+0.23[+0.01,+0.50]
+0.19[+0.04,+0.35]
Higgs-Audio-v3-8B
+0.50
+0.57
+0.37
+0.11
+0.13[−0.00,+0.26]
+0.39[+0.23,+0.56]
Whisper-Large-v3
+0.70
+0.89
+0.85
+1.01
−0.16[−0.37,+0.06]
−0.31[−0.47,−0.15]
Appendix
Table 4: Audio lift λ(r) (Eq. 1 , nats/char) of the silenced number span by voice condition ( 115 paired sentences passing the intelligibility gate in every clone condition). Diff columns: paired differences, bootstrap 95% CIs; bold marks CIs excluding zero. Whisper’s lift rises on clean TTS, so a drop in lift is likely not a synthesis artifact; Qwen3’s lift is negative in every condition, so its diffs do not indicate recovery. Parakeet-TDT has no teacher-forced readout.
model
truncated
attn-isolated
translate
full
Voxtral-Mini-3B
1.00
1.00
1.00
1.00
Whisper-Large-v3
1.00
1.00
1.00
1.00
Moonshine-Streaming
1.00
1.00
–
1.00
Qwen3-ASR-0.6B
0.94
0.95
1.00
1.00
Kimi-Audio-7B
1.00
0.89
–
0.89
Cohere-Transcribe
0.94
0.26
–
0.00
Appendix
Table 5: The opening-courtesy case study: rate at which the audible courtesy is present in the output. truncated : the audio is cut to the opener; attn-isolated keeps the full-clip audio encoding but restricts the decoder’s attention over it to the opener’s frames. translate : the same audio decoded under an English → Spanish translation instruction. full : the entire clip. – marks models without translation or attention-isolation capabilities.
ASR systems have been often reported to underperform on atypical speech. An often conflated compounding factor is the existence of two valid transcription references: verbatim (actual produced speech, including repetitions/prolongations) and intended (the canonical form of the text with disfluencies removed) in atypical speech recognition depending on context and use-case. Most ASR evaluations conflate this duality into a single ground truth and reward systems that delete disfluencies, ignoring verbatim faithfulness. We benchmark 11 ASR models from encoder-decoder, CTC and transducer families using both verbatim and intended references on atypical stuttered speech as a case study. Our quantitative assessment underlines the disparity in model performance and rankings using the two transcript styles. Through this analysis, we highlight the importance of selecting a suitable transcription reference for valid model selection depending on the use-case, particularly for atypical ASR.
Large Audio-Language Models show consistent performance gains across speech and audio benchmarks, yet high scores may not reflect true auditory perception. If a model can answer questions without processing the acoustic signal, the benchmark fails as a measure of auditory understanding. We present a diagnostic framework using two axes: text prior, which measures answerability from text and general knowledge alone, and audio reliance, which assesses actual dependency on the acoustic signal. Evaluating eight LALMs across three benchmarks, we find that models retain 60-72% of their full audio scores even without any audio input. Moreover, among items that require audio, only 3.0-4.2% need the complete audio clip; the majority can be resolved using localized fragments. These findings challenge the assumption that benchmark performance equals robust audio understanding, and we conclude with practical guidelines for improving evaluation reliability and benchmark design.
Leonardo Haw-Yang Foo, Chih-Kai Yang, Chen-An Li +2
National Taiwan University · NTU Artificial Intelligence Center of Research Excellence (NTU AI-CoRE)
Advances in large language models (LLMs) have enabled significant capabilities in audio processing, resulting in state-of-the-art models now known as Large Audio Language Models (LALMs). However, minimal work has been done to measure audio understanding beyond automatic speech recognition (ASR). This paper closes that gap by proposing a benchmark suite, SCENEBench (Spatial, Cross-lingual, Environmental, Non-speech Evaluation), that targets a broad form of audio comprehension across four real-world categories: background sound understanding, noise localization, cross-linguistic speech understanding, and vocal characterizer recognition. These four categories are selected based on understudied needs from accessibility technology and industrial noise monitoring. In addition to performance, we also measure model latency. The purpose of this benchmark suite is to assess audio beyond just what words are said - rather, how they are said and the non-speech components of the audio. Because our audio samples are synthetically constructed (e.g., by overlaying two natural audio samples), we further validate our benchmark against 20 natural audio items per task, sub-sampled from existing datasets to match our task criteria, to assess ecological validity. We assess five state-of-the-art LALMs and find critical gaps: performance varies across tasks, with some tasks performing below random chance and others achieving high accuracy. These results provide direction for targeted improvements in model capabilities.