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1. SpoTNet: A spoofing-aware Transformer Network for Effective Synthetic Speech Detection

   https://doi.org/10.1145/3592572.3592841  

   Khan, Awais; Malik, Khalid Mahmood (June 2023, MAD '23: Proceedings of the 2nd ACM International Workshop on Multimedia AI against Disinformation)

   The prevalence of voice spoofing attacks in today’s digital world has become a critical security concern. Attackers employ various techniques, such as voice conversion (VC) and text-to-speech (TTS), to generate synthetic speech that imitates the victim’s voice and gain access to sensitive information. The recent advances in synthetic speech generation pose a significant threat to modern security systems, while traditional voice authentication methods are incapable of detecting them effectively. To address this issue, a novel solution for logical access (LA)-based synthetic speech detection is proposed in this paper. SpoTNet is an attention-based spoofing transformer network that includes crafted front-end spoofing features and deep attentive features retrieved using the developed logical spoofing transformer encoder (LSTE). The derived attentive features were then processed by the proposed multi-layer spoofing classifier to classify speech samples as bona fide or synthetic. In synthetic speeches produced by the TTS algorithm, the spectral characteristics of the synthetic speech are altered to match the target speaker’s formant frequencies, while in VC attacks, the temporal alignment of the speech segments is manipulated to preserve the target speaker’s prosodic features. By highlighting these observations, this paper targets the prosodic and phonetic-based crafted features, i.e., the Mel-spectrogram, spectral contrast, and spectral envelope, presenting an effective preprocessing pipeline proven to be effective in synthetic speech detection. The proposed solution achieved state-of-the-art performance against eight recent feature fusion methods with lower EER of 0.95% on the ASVspoof-LA dataset, demonstrating its potential to advance the field of speaker identification and improve speaker recognition systems. 

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2. Exploiting Frequency Response for the Identification of Microphone using Artificial Neural Networks

   Hafeez, A; Khalid, Malik; Hafiz, Malik (June 2019, AES Int. Conf. Audio Forensics 2019)

   Microphone identification addresses the challenge of identifying the microphone signature from the recorded signal. An audio recording system (consisting of microphone, A/D converter, codec, etc.) leaves its unique traces in the recorded signal. Microphone system can be modeled as a linear time invariant system. The impulse response of this system is convoluted with the audio signal which is recorded using “the” microphone. This paper makes an attempt to identify "the" microphone from the frequency response of the microphone. To estimate the frequency response of a microphone, we employ sine sweep method which is independent of speech characteristics. Sinusoidal signals of increasing frequencies are generated, and subsequently we record the audio of each frequency. Detailed evaluation of sine sweep method shows that the frequency response of each microphone is stable. A neural network based classifier is trained to identify the microphone from recorded signal. Results show that the proposed method achieves microphone identification having 100% accuracy. 

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3. Cooperative Speech Separation With a Microphone Array and Asynchronous Wearable Devices

   https://doi.org/10.21437/Interspeech.2022-11025  

   Corey, Ryan; Mittal, Manan; Sarkar, Kanad; Singer, Andrew C. (September 2022, Proc. Interspeech 2022)

   We consider the problem of separating speech from several talkers in background noise using a fixed microphone array and a set of wearable devices. Wearable devices can provide reliable information about speech from their wearers, but they typically cannot be used directly for multichannel source separation due to network delay, sample rate offsets, and relative motion. Instead, the wearable microphone signals are used to compute the speech presence probability for each talker at each time-frequency index. Those parameters, which are robust against small sample rate offsets and relative motion, are used to track the second-order statistics of the speech sources and background noise. The fixed array then separates the speech signals using an adaptive linear time-varying multichannel Wiener filter. The proposed method is demonstrated using real-room recordings from three human talkers with binaural earbud microphones and an eight-microphone tabletop array. 

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4. I Spy You: Eavesdropping Continuous Speech on Smartphones via Motion Sensors.

   https://doi.org/10.1145/3569486  

   Zhang, Shijia; Liu, Yilin; Gowda, Mahanth (January 2023, Proceedings of the ACM on Interactive, Mobile, Wearable and Ubiquitous Technologies)

   This paper presents iSpyU, a system that shows the feasibility of recognition of natural speech content played on a phone during conference calls (Skype, Zoom, etc) using a fusion of motion sensors such as accelerometer and gyroscope. While microphones require permissions from the user to be accessible by an app developer, the motion sensors are zero-permission sensors, thus accessible by a developer without alerting the user. This allows a malicious app to potentially eavesdrop on sensitive speech content played by the user's phone. In designing the attack, iSpyU tackles a number of technical challenges including: (i) Low sampling rate of motion sensors (500 Hz in comparison to 44 kHz for a microphone). (ii) Lack of availability of large-scale training datasets to train models for Automatic Speech Recognition (ASR) with motion sensors. iSpyU systematically addresses these challenges by a combination of techniques in synthetic training data generation, ASR modeling, and domain adaptation. Extensive measurement studies on modern smartphones show a word level accuracy of 53.3 - 59.9% over a dictionary of 2000-10000 words, and a character level accuracy of 70.0 - 74.8%. We believe such levels of accuracy poses a significant threat when viewed from a privacy perspective. 

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5. Audio Capture Using Piezoelectric Sensors on Vibrating Panel Surfaces

   DiPassio, Tre; Heilemann, Michael C.; Thompson, Benjamin; Bocko, Mark F. (May 2023, 154th Convention of the Audio Engineering Society)

   The microphone systems employed by smart devices such as cellphones and tablets require case penetrations that leave them vulnerable to environmental damage. A structural sensor mounted on the back of the display screen can be employed to record audio by capturing the bending vibration signals induced in the display panel by an incident acoustic wave - enabling a functional microphone on a fully sealed device. Distributed piezoelectric sensing elements and low-noise accelerometers were bonded to the surfaces of several different panels and used to record acoustic speech signals. The quality of the recorded signals was assessed using the speech transmission index, and the recordings were transcribed to text using an automatic speech recognition system. Although the quality of the speech signals recorded by the piezoelectric sensors was reduced compared to the quality of speech recorded by the accelerometers, the word-error-rate of each transcription increased only by approximately 2% on average, suggesting that distributed piezoelectric sensors can be used as a low-cost surface microphone for smart devices that employ automatic speech recognition. A method of crosstalk cancellation was also implemented to enable the simultaneous recording and playback of audio signals by an array of piezoelectric elements and evaluated by the measured improvement in the recording’s signal-to-interference ratio. 

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