Study of the Improvement of Fixed Telephone Based on the CAS

2012 ◽  
Vol 546-547 ◽  
pp. 1256-1260
Author(s):  
Hai Dong Lei
Keyword(s):  
Signal Detection ◽  
Message Processing ◽  
Short Message ◽  
Voice Communication ◽  
Processing Module ◽  
The Voice

In this paper, based on the traditional fixed telephony features to increase the short message processing module. To meet the voice communication between users, the fixed telephone can realize mutual development between the SMS function .The main telephone lines in the completion of the CAS signal detection, and traditional communications in FSK and DTMF signal reception and transmission.

scholarly journals Voice and Position Simultaneous Communication System Based on Beidou Navigation Constellation

2020 ◽  
Vol 38 (5) ◽  
pp. 1010-1017
Author(s):  
Zhen Ao ◽  
Feng Li ◽  
Qiang Ma ◽  
Guiqing He
Keyword(s):  
Communication System ◽  
Linear Prediction ◽  
Voice Conversion ◽  
Prediction Algorithm ◽  
Short Message ◽  
Navigation Systems ◽  
Analysis Program ◽  
Data Format ◽  
Voice Communication ◽  
The Voice

Considering China Beidou has unique two-way communication capability for short messages that are not available in other navigation systems such as GPS, a 600bps vocoder adapted to the short message channel of Beidou is developed. The sinusoidal excitation linear prediction algorithm is adopted by the vocoder to achieve voice communication with clear communication quality. Furthermore, a coordinate compression algorithm for processing positioning information is designed to provide more transmission space for speech encoded data. Based on the above-mentioned results, a communication system that only the Beidou navigation system is used to complete two-way secure voice and positioning simultaneous interpretation is realized. The system firstly uses a voice conversion program to convert the voice coded data obtained by the vocoder codec module into the Beidou short message data format; and then the voice code analysis program and the latitude and longitude analysis program is used to parse the voice code and location information; Finally, the experimental results of voice communication and positioning transmission are verified on the Beidou short message transceiver, and the subjective MOS test score indicates that the way is paved for the practical use of Beidou short message voice communication.

The Grammar of Mobile Phone Written Language

2010 ◽  
pp. 110-198
Author(s):  
Adams B. Bodomo
Keyword(s):  
Mobile Phone ◽  
English Language ◽  
Service Providers ◽  
Written Language ◽  
Communication Process ◽  
Short Message ◽  
Voice Communication ◽  
Communicative Practice ◽  
Message Service

In the last chapter, I concentrated mainly on mobile phone voice communication. In this chapter, I will focus on mobile phone text communication. Mobile phone texting or communication through short message service (SMS) started slowly, as we saw in chapter 6, but has quickly emerged as a frequent daily linguistic, literacy or general communicative practice in which two or more people exchange messages by coding and decoding texts received and sent from their cell phones. Mobile phone texting is almost now as pervasive and as ubiquitous as mobile phone voice communication, if not more among some segments of users like young people. This communication process can be witnessed in buses, at homes, in offices, in restaurants, out in the woods, on the high seas, and even in the air! Hong Kong’s main English language newspaper, the South China Morning Post (SCMP) edition of April 11, 2004 indicates that as huge a volume of 200 million SMS messages are exchanged monthly. SMS has become a multi-million dollar business for service providers.

On the Application of DSP Technology in Real-Time Voice Communication System

2010 ◽  
Vol 159 ◽  
pp. 727-732
Author(s):  
Wen Ge Feng
Keyword(s):  
Signal Processing ◽  
Real Time ◽  
Software Design ◽  
Interface Design ◽  
Design Principle ◽  
Signal Acquisition ◽  
Voice Communication ◽  
Voice Signal ◽  
Signal Acquisition And Processing ◽  
The Voice

This paper describes the advantages of DSP TMS320VC5402 in the voice coding communication and focuses on interface design of real-time voice signal processing as well as the hardware and software design of the system from the aspect of voice signal acquisition and processing. Besides, it also introduces the corresponding design principle of hardware and software.

A Speech Enhancement Algorithm Based on the “Music Noise” Analysis

2014 ◽  
Vol 543-547 ◽  
pp. 2784-2787
Author(s):  
Ying Ma ◽  
Xiao Hua Zhang ◽  
Bing Lei Xing
Keyword(s):  
Speech Enhancement ◽  
Noise Analysis ◽  
Noise Pollution ◽  
Spectral Subtraction ◽  
Voice Communication ◽  
Spectrum Correction ◽  
Spectral Subtraction Method ◽  
The Voice ◽  
Enhancement Algorithm ◽  
Communication Equipment

Interference is inevitable process of voice communication will be from the surrounding environment and transmission medium noise, communication equipment, electronic noise, and other speakers. These interference makes the voice receiver received for noisy speech signal with noise pollution. According to the traditional spectral subtraction residual musical noise is too strong, the weighted processing is reduced and the power spectrum correction, spectral subtraction method was adopted to improve the traditional. According to the analysis of real speech data collection simulation, improved spectral subtraction can effectively reduce the musical noise, can satisfy the requirement of speech enhancement.

Adaptive Playout Buffering Schemes for IP Voice Communication

2011 ◽  
pp. 30-36
Author(s):  
Stefano Ferretti ◽  
Marco Roccetti ◽  
Claudio E. Palazzi
Keyword(s):  
Communication Systems ◽  
Multimedia Systems ◽  
The Internet ◽  
Voice Communication ◽  
Switched Networks ◽  
Research Areas ◽  
Internet Users ◽  
Audio Communication ◽  
Voice Data ◽  
The Voice

Audio communication over IP-based networks represents one of the most interesting research areas in the field of distributed multimedia systems. Today, routing the voice over Internet enables cheaper communication services than those deployed over traditional circuit-switched networks. BoAT (Roccetti, Ghini, Pau, Salomoni, & Bonfigli, 2001a), Ekiga, FreePhone (Bolot & Vega Garcia, 1996), iCall, Kiax, NeVot (Schulzrinne, 1992), rat (Hardman, Sasse, & Kouvelas, 1998), Skype, Tapioca, vat (Jacobson & McCanne, n.d.), WengoPhone, and YATE, are just few examples of free VoIP software available to Internet users. Without any doubts, new (wired and wireless) highspeed, broadband networks facilitate the transmission of the voice over the Internet and have determined the success of these applications. However, the best effort service offered by the Internet architecture does not provide any guarantee on the delivery of (voice) data packets. Thus, to maintain a correct time consistency of the transmitted audio stream, these voice communication systems must be equipped with schemes able to deal with the unpredictability of network latency, delay jitter, and possible packet loss.

A Discontinuous Transmission Method for LPC Speech Codec

2014 ◽  
Vol 644-650 ◽  
pp. 4346-4350
Author(s):  
Hong En Xie ◽  
Qiang Li ◽  
Qin Jun Shu
Keyword(s):  
Background Noise ◽  
Auditory Perception ◽  
Activity Detection ◽  
Characteristic Parameters ◽  
Transmission Method ◽  
Voice Communication ◽  
Transmission Frequency ◽  
Speech Codec ◽  
The Voice ◽  
Voice Activity

In order to improve the utilization of transmission bandwidth in voice communication, this paper proposes a discontinuous transmission method for LPC speech codec. Firstly, by using the algorithm of voice activity detection (VAD), the received signal is divided into voice frame and mute frame. Transitional frame is introduced when the voice frame is converted to mute frame. Then voice frames and transitional frames are encoded at a normal rate, but mute frames are encoded into silence description (SID) frame at a lower rate, which is sent by a method of discontinuous transmission mode. The transmission frequency of SID frame is adjusted automatically according to the fluctuation of characteristic parameters of background noise in mute frames. Finally, the method is applied to the simulation in the MELP vocoder, and the results show that this method has better adaptability in the transmission of mute signal and the synthesized background noise presents good comfort and continuity in the auditory perception.

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