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Ste gano graphic systems are used for the transmission of hidden data in the original signal. The article describes the algorithm of the hidden data transmission using the speech signal as a carrier. The echo method is used for data embedding. In order to improve the decoding efficiency of embedded data, the procedure of voicing correction and mechanism of informed coding were developed and implemented...
Research on multilingual speech recognition remains attractive yet challenging. Recent studies focus on learning shared structures under the multi-task paradigm, in particular a feature sharing structure. This approach has been found effective to improve performance on each individual language. However, this approach is only useful when the deployed system supports just one language. In a true multilingual...
The increasing number of standardized speech codecs implies interoperability issues between networks deploying different and sometime incompatible standards. To assure interoperability, transcoding from one codec format to another is necessary at gateways between networks. This transcoding reduces speech quality, involves computation load and additive delay. Recently, a fair amount of work has been...
Speech coders operating in time domain can be extended with a frequency domain mode to improve encoding of music, even though this is challenging at low delay. In such a scenario, the short analysis window limits the benefit of the transform coder, while a delayless switch between the two coders constrains the system further. The paper presents an LPC and MDCT-based audio coder part of the new 3GPP...
A highly error resilient mode of the newly standardized 3GPP EVS speech codec is described. Compared to the AMR-WB codec and other conversational codecs, the EVS channel aware mode offers significantly improved error resilience in voice communication over packet-switched networks such as Voice-over-IP (VoIP) and Voice-over-LTE (VoLTE). The error resilience is achieved using a form of in-band forward...
The objective of this research is to study the performance of a high quality speech compression in real-time on a single-chip system. Based on voice over Internet protocol (VoIP) requirements, we have decided to implement a high quality speech coding (with signal-to-noise ratio, SNR of more than 10 dB), at a low bit rate of 8 kbit/s or less. The coder must have delay not more than 100 ms. The development...
The transmission of the Voice over Internet protocol requires high quality of speech to update this service. VoIP signal quality can be degraded by network layer problems which includes delay, packet loss and jitter. But the frame loss while transmitting voice over internet services may generate audible distortion in the perceived speech quality. The simulation model of VoIP network under impairments...
In digital mobile communication systems, speech coding is very important to increase the bandwidth efficiency. Usually, speech coding algorithms determine speech parameters, which are highly sensitive to transmission errors. In this paper, we use the residual redundancy remaining after using the Enhanced Variable Rate Codec (EVRC) algorithm for error concealment. Average residual redundancies of the...
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