Real-time Speech Communication Research Articles

Rate-Distortion-Perception Optimized Neural Speech Transmission System for High-Fidelity Semantic Communications.

We consider the problem of learned speech transmission. Existing methods have exploited joint source-channel coding (JSCC) to encode speech directly to transmitted symbols to improve the robustness over noisy channels. However, the fundamental limit of these methods is the failure of identification of content diversity across speech frames, leading to inefficient transmission. In this paper, we propose a novel neural speech transmission framework named NST. It can be optimized for superior rate-distortion-perception (RDP) performance toward the goal of high-fidelity semantic communication. Particularly, a learned entropy model assesses latent speech features to quantify the semantic content complexity, which facilitates the adaptive transmission rate allocation. NST enables a seamless integration of the source content with channel state information through variable-length joint source-channel coding, which maximizes the coding gain. Furthermore, we present a streaming variant of NST, which adopts causal coding based on sliding windows. Experimental results verify that NST outperforms existing speech transmission methods including separation-based and JSCC solutions in terms of RDP performance. Streaming NST achieves low-latency transmission with a slight quality degradation, which is tailored for real-time speech communication.

Speech Communication System Based on Piezoelectric Electret Mechanical Antenna

Low-frequency electromagnetic waves have strong penetration and low attenuation performance, making them widely used in long-wave communication areas such as underwater communication and earthquake prediction. Mechanical antennas are seen as a promising alternative to traditional low-frequency resonant antennas, as they offer the potential for miniaturization and low power consumption. However, existing communication systems using mechanical antennas are usually rudimentary, incompatible, and inconvenient to use. To address these problems, a novel speech communication system based on a piezoelectric electret mechanical antenna was proposed. This communication system includes a hardware system, a communication protocol for mechanical antennas, and a software system with a modular structure. This system includes hardware, a communication protocol, and a modular software system. The results of the experiments showed that this communication system provides real-time speech communication with a user-friendly design, high compatibility with different communication devices, and easy customization to meet specific requirements. This design holds great potential for applications in mineral detection, seismic research, submarine communication, industrial RF control, and other related mechanical antenna applications.

Chaos-based joint speech encryption scheme using SHA-1

In this paper, the speech signal is encrypted using multiple chaotic maps and cryptographic protocols. The input signal is fragmented and scrambled into four segments using cubic map. In order to make the signal invulnerable against attacks, scrambled signal is processed through various one-dimensional chaotic maps like cubic, logistic, skew-tent, and quadratic map. All chaotic maps are employed to provide an encryption at transmitter side; however, reversal of the same is carried out at other end. To protect the various parameters of all chaotic techniques, blowfish algorithm is employed with the private key. On top of that, additional layer of security is provided through hashing algorithm for authentication of shared data and the blowfish key of system between two ends. The computed message digest of secure hash algorithm should be same for successful communication, which authenticates and verify the parameters of chaotic maps. Furthermore, several tests have been performed on 8 KHz secured speech signal such as signal-to-noise ratio, power-spectral density, peak signal-to-noise ratio, and correlation test in order to check the efficiency of proposed system. The various statistical tests and analysis of adversary model validates the encryption model and embraces stronger security, thus making it as invulnerable to the attacks of intruder. The result promises that presented technique is extremely secure, provides end-to-end encryption and can be utilized in a real time speech communication.

Research into the use of scramblers in narrowband communication systems

One of the most important requirements for speech channel is to provide information security. Speech scrambling is one of the preferred methods for secure speech communications over the narrowband channels. That confirms the relevance of works focused on improvement and investigation of operation features of speech scrambling devices. The article shows the advantages of using scramblers in narrowband voice communication systems and presents a detailed analysis of strength to hacking of communication channels secured by frequency-domain scrambling technique. The investigation of strength to hacking is based on the developed digital signal processing algorithm which realizes the band scrambler. The developed algorithm does not require synchronization between the scrambler and the descrambler which is achieved by using of sliding window FFT technique. Presented experimental results show that the lowest level of the scrambled signal residual intelligibility can be achieved when low frequency bands of speech spectrum are permuted with high frequency ones. Also the article shows that if amount of bands is less than 32 the scrambled signal can be hacked by a simple frequency spectrum inversion in the entire frequency band of the signal. During scrambling with amount of bands greater than 32, there are variants of band permutation that ensure the level of residual intelligibility near 10‑20 % and that is secured against hacking by the spectrum inversion. For a real-time speech communication system, the maximum number of scrambler bands is limited by two factors: the delay time for the scrambling-descrambling process and inserted distortions. It was shown that the maximum amount of bands should be less or equal to 256 for scrambling time delay of 128 ms, communication channel bandwidth of 4 kHz and the level of distortions that causes reduction of intelligibility not lower than 90 %. The presented information can be used in development of frequency-domain scramblers and for selection scrambler keys, which are optimal by the criteria of strength to hacking, minimum of residual intelligibility and level of inserted distortions.

Audio dilation in real time speech communication

Algorithmically decreasing speech tempo, or “audio dilation,” can improve speech perception in attentionally demanding tasks [Gygi & Shafiro (2014), Hear. Res. 310, pp. 76–86]. On-line audio dilation [Novak, et al. (2013), Interspeech 1869–1871] is a recently developed technique that decreases the tempo of audio signals as they are generated. This pilot study investigated effects of on-line audio dilation on performance in interactive problem solving tasks. We used a Diapix task [Baker & Hazan (2011), Behav. Res. Methods 43(3), 761–770] to elicit and record spontaneous speech from pairs of participants under various dilation conditions: participants, seated in different rooms, were asked to find ten differences on two similar pictures, while their speech was either transmitted as spoken or dilated. Conditions tested include stretching one, both, or neither audio signal by 40%. Subsequent analysis shows that the technique, even using this substantial increase, did not interfere with interactive problem solving tasks, and did not lead to changes in speech production rate, measured as number of syllables per second. The lack of negative effects of on-line speech dilation provides a preliminary basis for further assessment of this method in speech perception tasks with high attentional and memory load.

Real-time, speech-in-speech hiding scheme based on least significant bit substitution and adaptive key

Real-time, embedded speech-in-speech hiding has not been widely researched. Nevertheless, it could be useful, among other cases, in secure mobile telephony. In this paper, we propose a new scheme of data hiding which takes advantage of the masking property of the Human Auditory System (HAS) to hide a secret (speech) signal into a host (speech) signal. The embedding process is carried out into the wavelet coefficients of the speech signals. The main point of the proposed scheme is that the embedding process is suitable for real-time processing, and the secret’s coefficients are relocated by an adaptive key, instead of a pseudo-noise sequence of some approaches. The latency of the embedding module makes this approach useful for real-time speech communication because the total delay added by the proposed system is low compared to the highest delay allowed for a high quality speech transmission.

Implementation and Design of Underwater Acoustic Speech Communication System Based on OFDM Technology

Due to severe signal degradation caused by multipath propagation and the limited bandwidth and the time-varying and space-variant nature of the underwater acoustic channel, underwater acoustic communication is difficult. A suitable for underwater acoustic channel communication mode must be employed. Implementation and design of the real-time speech communication system based on orthogonal frequency division multiplexing (OFDM) technology, AMBE-2000 was used to compress the speech coding. The main hardware circuit and software design of speech communication system were analyzed, speech signal was inset comfortable noise, eliminated echo, controlled volume, finally through the system debugging and channel pool test. It proved that the underwater acoustic speech communication system was feasible, speech was clear.

Research of solution for real-time speech communication over mobile Ad hoc network

Research of solution for real-time speech communication over mobile Ad hoc network

Analyzing Voice Quality in Popular VoIP Applications

Voice over IP (VoIP), which provides real-time speech communication between two users in a way that closely resembles a face-to-face conversation, has had a significant impact on the multibillion-dollar telecommunication industry. The promise of less expensive phone calls with comparable quality and better features than public switched telephone networks (PSTNs) has accelerated VoIP's adoption, both in businesses and homes. Its better integration with various forms of collaborative communications, such as instant messaging, email, and voicemail, has made it suitable for future communication solutions, but there remain several issues that are unique to VoIP systems and require further analysis.