DESIGN, IMPLEMENTATION, AND ANALYSIS OF ADAPTIVE M-ARY FREQUENCY SHIFT KEYING ON GSM VOICE CHANNEL

A mobile voice channel is a high priority service in cellular communication systems, has a wide coverage, and is almost always available. With these advantages, mobile voice channels can be used to transmit digital data remotely in rural areas that are not covered with 3G/4G networks. In a mobile voice channel, the Adaptive Multi Rate (AMR) whose rate can vary from 4.75 to 12.2 kbps is used today as a vocoder. In data communication through AMR channel with non-adaptive modulation, the modulation configuration must work on the lowest AMR rate, so the configuration is not optimal for higher channel rates. To optimize the data transmission rate within an AMR channel, we proposed an adaptive M-ary Frequency Shift Keying (MFSK) modulation method with a zero crossing demodulator. The Adaptive MFSK modulation is designed to adjust its modulation configurations based on a quality of the channel that shown by vocoder rate. In this project, using signal quality indicator provided by GSM modem, the modulator adjust its symbol time length to fit the maximum bit error rate (BER). Demodulator detects the symbol without need the symbol length information and counts the bit error rate. The adaptive MFSK modulation proposed in this research has higher data rate than fixed lower rate modulation in the similar BER range. The adaptive modulation has lower BER than fixed higher rate modulation in the similar data rate range

The voice activity detection (VAD) recorder and VAD network recorder : a thesis presented in partial fulfilment of the requirements for the degree of Master of Science in Computer Science at Massey University

The project is to provide a feasibility study for the AudioGraph tool, focusing on two application areas: the VAD (voice activity detector) recorder and the VAD network recorder. The first one achieves a low bit-rate speech recording on the fly, using a GSM compression coder with a simple VAD algorithm; and the second one provides two-way speech over IP, fulfilling echo cancellation with a simplex channel. The latter is required for implementing a synchronous AudioGraph. In the first chapter we introduce the background of this project, specifically, the VoIP technology, the AudioGraph tool, and the VAD algorithms. We also discuss the problems set for this project. The second chapter presents all the relevant techniques in detail, including sound representation, speech-coding schemes, sound file formats, PowerPlant and Macintosh programming issues, and the simple VAD algorithm we have developed. The third chapter discusses the implementation issues, including the systems' objective, architecture, the problems encountered and solutions used. The fourth chapter illustrates the results of the two applications. The user documentations for the applications are given, and after that, we analyse the parameters based on the results. We also present the default settings of the parameters, which could be used in the AudioGraph system. The last chapter provides conclusions and future work

Error Correction For Automotive Telematics Systems

One benefit of data communication over the voice channel of the cellular network is to reliably transmit real-time high priority data in case of life critical situations. An important implementation of this use-case is the pan-European eCall automotive standard, which has already been deployed since 2018. This is the first international standard for mobile emergency call that was adopted by multiple regions in Europe and the world. Other countries in the world are currently working on deploying a similar emergency communication system, such as in Russia and China. Moreover, many experiments and road tests are conducted yearly to validate and improve the requirements of the system. The results have proven that the requirements are unachievable thus far, with a success rate of emergency data delivery of only 70%. The eCall in-band modem transmits emergency information from the in-vehicle system (IVS) over the voice channel of the circuit switch real time communication system to the public safety answering point (PSAP) in case of a collision. The voice channel is characterized by the non-linear vocoder which is designed to compress speech waveforms. In addition, multipath fading, caused by the surrounding buildings and hills, results in severe signal distortion and causes delays in the transmission of the emergency information. Therefore, to reliably transmit data over the voice channels, the in-band modem modulates the data into speech-like (SL) waveforms, and employs a powerful forward error correcting (FEC) code to secure the real-time transmission. In this dissertation, the Turbo coded performance of the eCall in-band modem is first evaluated through the adaptive white Gaussian noise (AWGN) channel and the adaptive multi-rate (AMR) voice channel. The modulation used is biorthogonal pulse position modulation (BPPM). Simulations are conducted for both the fast and robust eCall modem. The results show that the distortion added by the vocoder is significantly large and degrades the system performance. In addition, the robust modem performs better than the fast modem. For instance, to achieve a bit error rate (BER) of 10^{-6} using the AMR compression rate of 7.4 kbps, the signal-to-noise ratio (SNR) required is 5.5 dB for the robust modem while a SNR of 7.5 dB is required for the fast modem. On the other hand, the fading effect is studied in the eCall channel. It was shown that the fading distribution does not follow a Rayleigh distribution. The performance of the in-band modem is evaluated through the AWGN, AMR and fading channel. The results are compared with a Rayleigh fading channel. The analysis shows that strong fading still exists in the voice channel after power control. The results explain the large delays and failure of the emergency data transmission to the PSAP. Thus, the eCall standard needs to re-evaluate their requirements in order to consider the impact of fading on the transmission of the modulated signals. The results can be directly applied to design real-time emergency communication systems, including modulation and coding

Error Correction For Automotive Telematics Systems

One benefit of data communication over the voice channel of the cellular network is to reliably transmit real-time high priority data in case of life critical situations. An important implementation of this use-case is the pan-European eCall automotive standard, which has already been deployed since 2018. This is the first international standard for mobile emergency call that was adopted by multiple regions in Europe and the world. Other countries in the world are currently working on deploying a similar emergency communication system, such as in Russia and China. Moreover, many experiments and road tests are conducted yearly to validate and improve the requirements of the system. The results have proven that the requirements are unachievable thus far, with a success rate of emergency data delivery of only 70%. The eCall in-band modem transmits emergency information from the in-vehicle system (IVS) over the voice channel of the circuit switch real time communication system to the public safety answering point (PSAP) in case of a collision. The voice channel is characterized by the non-linear vocoder which is designed to compress speech waveforms. In addition, multipath fading, caused by the surrounding buildings and hills, results in severe signal distortion and causes delays in the transmission of the emergency information. Therefore, to reliably transmit data over the voice channels, the in-band modem modulates the data into speech-like (SL) waveforms, and employs a powerful forward error correcting (FEC) code to secure the real-time transmission. In this dissertation, the Turbo coded performance of the eCall in-band modem is first evaluated through the adaptive white Gaussian noise (AWGN) channel and the adaptive multi-rate (AMR) voice channel. The modulation used is biorthogonal pulse position modulation (BPPM). Simulations are conducted for both the fast and robust eCall modem. The results show that the distortion added by the vocoder is significantly large and degrades the system performance. In addition, the robust modem performs better than the fast modem. For instance, to achieve a bit error rate (BER) of 10^{-6} using the AMR compression rate of 7.4 kbps, the signal-to-noise ratio (SNR) required is 5.5 dB for the robust modem while a SNR of 7.5 dB is required for the fast modem. On the other hand, the fading effect is studied in the eCall channel. It was shown that the fading distribution does not follow a Rayleigh distribution. The performance of the in-band modem is evaluated through the AWGN, AMR and fading channel. The results are compared with a Rayleigh fading channel. The analysis shows that strong fading still exists in the voice channel after power control. The results explain the large delays and failure of the emergency data transmission to the PSAP. Thus, the eCall standard needs to re-evaluate their requirements in order to consider the impact of fading on the transmission of the modulated signals. The results can be directly applied to design real-time emergency communication systems, including modulation and coding

Channel Fading Statistics For Real-Time Data Transmission In Emergency Call Systems And Unmanned Aerial Systems

The Third Generation Partnership Project (3GPP) selected an in-band modem to transmit emergency data over cellular voice channel for the European Union emergency call (eCall) system. However, the road test results presented by the Harmonized eCall European Pilot project showed that the success rate of data delivery was only 71%, indicating that there is significant potential to improve its performance. In this dissertation, a testbed is designed for the eCall system that satisfies the 3GPP TS 26.267/268/269 standards. A method is proposed to measure the power of the received signal that passes through the in-band channel. Experiments are performed with the in-vehicle system testbed in a laboratory or a car travelling in city, suburb, country- side, or freeway. Fading statistics of the received signal after power control are found and discussed, together with cumulative distribution function (CDF), level crossing rate (LCR), and average fade duration (AFD). It is found that with probability less than or equal to 0.1%, fading and attenuation can vary from -19 dB for the continuous wave (CW) signal at 500 Hz to -9.5 dB for the CW signal at 2000 Hz. This dissertation recommends moving the CW signals at 500 Hz and 800 Hz for detection and synchronization in the 3GPP standard to 1500 Hz and 2000 Hz, respectively. This will give 9.5 dB improvement in detection and synchronization. The fading results are used to calculate the bit error rate (BER) performance for the eCall in-band modem. Synchronization detection probability are obtained by transmitting the synchronization preamble through various adaptive multi-rate vocoders and an additive white Gaussian noise channel. The testbed and proposed method are also used to measure the power of signals received by an unmanned aerial systems (UAS) and by the receiver in the operation center, respectively. Field experiments are carried out by flying the UAS above different locations. Statistics, including CDF, LCR, and AFD, are calculated for the six test-sites. The results of the fading statistics, synchronization detection probability, and BER can be directly applied to design real-time communication systems, including detection, delay estimation, modulation and coding

Proceedings of the Second International Mobile Satellite Conference (IMSC 1990)

Presented here are the proceedings of the Second International Mobile Satellite Conference (IMSC), held June 17-20, 1990 in Ottawa, Canada. Topics covered include future mobile satellite communications concepts, aeronautical applications, modulation and coding, propagation and experimental systems, mobile terminal equipment, network architecture and control, regulatory and policy considerations, vehicle antennas, and speech compression