Audio deepfake detection · Real communication channels
RealCommBenchmarking and Adapting Audio Deepfake Detection
over Real Communication Channels
Abstract
Acoustic capture, device processing, and transmission coding in real calls jointly alter speech signals and may affect the reliability of audio deepfake detection. To systematically evaluate these effects, we introduce RealComm, a dataset pairing digital speech with recordings of the same utterances transmitted through real mobile calls. The acquisition design covers acoustic and wired injection, handset and speakerphone operation, and caller-side denoising, with bona fide and synthetic speech recorded under matched conditions. The dataset includes RealCommBench for evaluation and RealCommTrain for adaptation. We further develop RealComm-Aug, a staged waveform augmentation pipeline for simulating acquisition and transmission effects, and investigate its combination with real-call training. Experiments with existing detectors reveal substantial degradation after call transmission, even for models with strong digital-domain performance; acoustic injection is particularly challenging. Paired quality and spectral analyses show that quality degradation is associated with poorer detection across recording conditions for the evaluated detectors, while high-frequency attenuation alone does not explain detection difficulty. Simulated augmentation and real-call training provide complementary benefits, with their combination achieving the lowest call-domain error for each of three evaluated detectors. For Teffic-Audio, the equal error rate decreases from 37.94% to 22.37%.
Dataset and acquisition
Every source utterance has a digital reference and multiple real-call versions. Bona fide and synthetic speech traverse the same recording conditions; all versions of a source remain in one split.
| Split | Digital sources | Recorded sources | Conditions / source | Call recordings |
|---|---|---|---|---|
| Train | 1,792 | 224 | 44 | 9,856 |
| Development | 448 | 56 | 44 | 2,464 |
| Test | 560 | 560 | 42 | 23,520 |
The source pool is balanced by authenticity and language (English / Mandarin), with synthetic speech from seven generators. Train and test share the generator set. Repeated call versions expand condition coverage rather than the number of independent source utterances.
Paired audio examples
Explore 16 source examples in English and Mandarin: bona fide speech and all seven TTS systems, spanning eight directed device routes. Within each example, the digital reference and three call configurations share the same source utterance; the call route is fixed.
Choose a language and voice source above. For bona fide and CosyVoice2 examples, switch denoising off/on to compare matched recordings. Other examples show their recorded denoising state below each player; “no user-controlled switch” is distinct from off. OtA uses acoustic injection; LtM uses a wired microphone input. WAV files are unchanged and are not loudness-normalized; selection does not use detector scores.
Zero-shot detection
Real-call transmission substantially degrades the evaluated detectors. EER rises by more than 30 percentage points for every detector below, despite digital-domain EERs of 0–15.71%. Acoustic injection is consistently harder than wired injection, and degradation occurs in both languages and across all seven generation systems.
Complete RealComm evaluation results will be published and maintained on SpoofRadar, our unified leaderboard for speech deepfake detection. This page presents the manuscript’s fixed results; the platform will provide full evaluation tables and subsequent updates, identifying model/checkpoint versions and evaluation settings.
| Detector | Digital | Call | ΔEER (pp) |
|---|---|---|---|
| WavLM-L | 4.64 | 40.00 | +35.36 |
| Whisper | 15.71 | 47.88 | +32.17 |
| AASIST | 7.50 | 40.19 | +32.69 |
| Teffic-Audio | 0.00 | 37.95 | +37.95 |
| DF Arena | 7.86 | 38.77 | +30.91 |
The main comparisons follow the current manuscript. The initial XLSR-SLS evaluation is retained in its protocol discussion (38.57% digital / 50.31% call EER); its weak digital baseline limits interpretation of the additional channel effect.
Speech quality and spectral changes
Poorer speech quality is associated with greater detection degradation across recording conditions. Real calls increase recognition errors and reduce speaker similarity and predicted overall quality. These quality changes track EER increases for the evaluated detectors, while high-frequency attenuation alone does not explain which conditions are harder.
Correlations are descriptive, not causal. Quality metrics use full utterances; detection uses the model-specific input prefix protocol described in the manuscript.
RealComm-Aug
RealComm-Aug samples one signal-path scenario and its operations for each waveform, with the same distribution for bona fide and synthetic speech. Acoustic environment, device processing, file coding, and call transmission form distinct stages and exits. The receiver adds no further replay.
FExisting model; no RealComm updates.
AOriginal digital training mixture with RealComm-Aug.
TOriginal digital mixture plus RealCommTrain, retaining the original digital augmentation.
A+TThe same data as T, with RealComm-Aug on digital examples.
Every adapted model starts from its corresponding F checkpoint. Real-call examples bypass the added data-side augmentation; existing model-internal neural-codec augmentation retains its training configuration. No augmentation is applied at evaluation.
Adaptation results
Real-call training gives larger gains than simulation alone. Their combination produces the lowest pooled call EER for all three models, reaching 22.37% for Teffic-Audio. Digital-domain results reveal a tradeoff for AASIST.
| Detector | Domain | F | A | T | A+T |
|---|---|---|---|---|---|
| WavLM-L | Digital | 4.64 | 4.29 | 3.21 | 3.21 |
| WavLM-L | Call | 40.00 | 38.71 | 35.48 | 34.16 |
| AASIST | Digital | 7.50 | 8.93 | 6.07 | 8.21 |
| AASIST | Call | 40.19 | 38.49 | 37.96 | 36.90 |
| Teffic-Audio | Digital | 0.00 | 0.36 | 0.36 | 0.00 |
| Teffic-Audio | Call | 37.94 | 34.92 | 23.84 | 22.37 |
Bold indicates the best setting within each model and domain. Teffic-Audio F uses a matched FP32 re-evaluation here (37.94%); the frozen zero-shot submission is 37.95%. Results are from single training runs. T and A+T use RealCommTrain, which is not included in the benchmark release; these adaptation results are distinct from zero-shot benchmark entries.
Performance on public test sets
Call-domain gains do not guarantee improvement on every external test set. Augmentation alone improves ASVspoof 2021 LA results for all three adapted detectors. Combined adaptation improves real-call detection, but increases In-the-Wild EER by 0.38, 0.78, and 0.85 percentage points for WavLM-L, AASIST, and Teffic-Audio, respectively.
Resources
The current study covers a fixed set of devices, two languages and seven generators. The results do not establish robustness to arbitrary hardware, communication services or unseen synthesis systems.