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Digital Audio Broadcasting (DAB+) is becoming a reality in the Czech Republic. The first nationwide DAB+ network, based on the regular broadcasting, is being completed. This paper presents the principles that were used to achieve a quick and efficient penetration of the DAB+ signal in the Czech population and the highways. Attention is focused on practical experience with the use of High-Efficiency Advanced Audio Coding (HE-AAC) emphasizing maximum efficiency of the multiplex. This is done with respect to the subjective perception of sound quality by the audience. Final audio processing and appropriate signal pre-processing are considered. The paper also focuses on how to use Forward Error Correction (FEC) coding to increase the reach of transmitters and the reasons for employing the specific transmitter network configuration, including indoor reception. The results of this complex method are demonstrated on the network rollout in particular periods, while the key assumptions were verified. The entire development process can be monitored on the maps of coverage.

Digital Audio Broadcast (DAB) transmitters can be successfully used as illumination sources in Passive Coherent Location (PCL). However, extending the integration time in such a configuration leads to the occurrence of periodical artifacts in the bistatic range/Doppler plots, resulting from the time structure of the DAB signal. In this paper, we propose some methods of signal preprocessing (based on symbol removal, substitution by noise, and duplication) that operate on the DAB transmission frame level and improve the received signal’s correlation properties. We also demonstrate that two of these methods allow us to avoid the mentioned artifacts and thus to improve the quality of the range/Doppler plots with detection results. We evaluate the performance of the proposed methods using real DAB signals acquired in an experimental PCL platform. We also provide the analysis of the Signal to Noise Ratio (SNR) during the detection of a moving target which shows that the proposed solution, based on symbol duplication, can offer around 3 dB of gain in SNR. Finally, we carry out the computational complexity analysis showing that the proposed method can be implemented with a minimal cost after some optimizations.

Anti-aliasing filters play an important role in analog front-end circuits, and their suitable design can improve the performance of digital audio broadcasting systems. In this paper, a Gm-C anti-aliasing filter based on carbon nanotube field effect transistors (CNTFETs) is proposed. Using CNTFETs provides low power consumption and appropriate frequency response. Simulation results show that these characteristics are widely dependent on the physical parameters of CNTFETs. The effects of chirality, the number of CNTs in each CNTFET, and distance of CNTs from each other on power consumption and frequency response are investigated. Among these parameters, the chirality and number of CNTs respectively have nonlinear and linear relationships with both power consumption and cutoff frequency. On the other hand, changing the distance between CNTs has the least effect on the output characteristics. Also, the output features of the filter for different gate oxides and channel lengths are studied. The overall result of the simulations demonstrates that the filters based on CNTFETs are promising candidates to be used in future nano-scale integrated circuits.

This paper describes the complex experiences and the results of the Digital Audio Broadcasting (DAB+) implementation in the conditions of the Czech Republic. It analyses the background of the EUREKA 147 program, its targets, and to what extent they were reached. The main focus is concentrated on the evaluation of the real steps which had to be done during the implementation process before the first regular broadcasting could be started and the difficulties encountered. The key point became experimental broadcasting “DAB Prague”. It demonstrated the advantages of this new platform and gave the opportunity to test its functions and physical behavior in real conditions, under the different parameter settings. The important issue is the allocation of the band III by National Regulatory Authority and the capacity of DAB+ multiplexes. The paper also demonstrates the experience from the setup of the DAB+ headend. It describes the content solution in context of using non-entropic audio coding, setting the codec algorithms and the way to use accompanying multimedia services. The results of the tests and measurements presented in this paper were used in the real project of the regular nationwide DAB+ network.

With more and more new mobile devices (such as mobile phones, tablets, and wearable devices) entering people’s daily life, along with the application and development of relevant technologies based on users’ location information, location based service is becoming a basic service demand of people’s life. This paper puts forward a research on location technology based on frequency modulation band digital audio broadcasting (FM China Digital Radio, FM-CDR). A new method of adding timestamp information to the FM-CDR frame structure is proposed, which verified that the change to the system does not affect the normal transmission and reception of broadcast content under the original standards and can accurately extract the recognition signal and timing information of BS. In the complex environment, the estimation algorithm of signal parameters such as received signal strength (RSS), time of arrival (TOA), and time difference of arrival (TDOA) of terrestrial radio broadcast signals is studied. In this paper, a new method based on multisource data fusion is proposed, which can meet the need of localization in various environments and overcome the deficiency of single localization method.

One of the most crucial task of terrestrial audio broadcasting is the transition to digital system and infrastructure. In this paper, the china digital radio (CDR) system structure is presented including framing structure, transfer mode, as well as spectrum mode. In addition, this paper gives a comprehensive introduction to the deployment of CDR technology in China.

The single frequency network (SFN) is a popular solution in modern digital audio and television system networks for extending effective coverage, compared to its traditional single-transmitter counterpart. As benefits of this configuration appear to be obvious, this paper focuses on the exact analysis of so-called SFN gain&mdash, a quantitative effect of advantage in terms of the received signal strength. The investigations cover a statistical analysis of SFN gain values, obtained by means of computer simulations, with respect to the factors influencing the coverage, i.e., the protection level, the reception mode (fixed, portable, mobile), and the receiver location (outdoor, indoor). The analyses conclude with an observation that the most noteworthy contribution of the SFN gain is observed on the far edges of the networks, and the least one close to the transmitters. It is also observed that the highest values of the SFN gain can be expected in the fixed mode, while the protection level has the lowest impact.

Digital Audio Broadcasting (DAB) and all similar systems for digital radio and television broadcasting are inevitably associated with lossy psychoacoustic audio compression. The coding algorithms are continuously being improved. To achieve high sound quality a lower bitrate, required by the broadcasters, is now sufficient. This paper compares the relevant digital audio codecs: MPEG 2 and AAC in three profiles (AAC LC, HE-AAC v1 and HE-AAC v2). The well-known MP3 format is also added for a better comparison. A brief description of the basic functional principles of the codecs is followed by a comparison of their efficiency keeping in mind the subjectively comparable sound quality. The main contribution of this paper is the verification of the relationship of the bitrate level and sound quality in broadcasting environment and the finding out the influence of other, often more significant factors, such as the primary quality of the input recordings and the concatenation of non-entropic coding, on the subjective perception in the digital broadcasting chain. These findings are supported by the results of a unique research analysis providing an insight into which specific audio encoding configurations are used for DAB+ radio broadcasting in practice, in Europe as a whole and in individual European countries.

Broadcasters need to decide on bitrates of the services in the multiplex transmitted via Digital Audio Broadcasting Plus system. The bitrate should be set as low as possible for maximal number of services, but with high quality, not lower than in conventional analog systems. In this paper, the objective method Perceptual Evaluation of Audio Quality is used to analyze the perceived audio quality for appropriate codecs --- MP2 and AAC offering three profiles. The main aim is to determine dependencies on the type of signal --- music and speech, the number of channels --- stereo and mono, and the bitrate. Results indicate that only MP2 codec and AAC Low Complexity profile reach imperceptible quality loss. The MP2 codec needs higher bitrate than AAC Low Complexity profile for the same quality. For the both versions of AAC High-Efficiency profiles, the limit bitrates are determined above which less complex profiles outperform the more complex ones and higher bitrates above these limits are not worth using. It is shown that stereo music has worse quality than stereo speech generally, whereas for mono, the dependencies vary upon the codec/profile. Furthermore, numbers of services satisfying various quality criteria are presented.

This paper proposes an adaptive impedance matching subsystem, called self-structuring impedance matcher (SSIM), for in-vehicle digital audio broadcasting (DAB) applications. The goal of the SSIM is to make antennas originally aiming at other vehicular applications, such as a digital television (DTV, 470–894 MHz) and an interphone car radio (136–174 MHz), operate in the targeted DAB frequency band (170–240 MHz), thereby lowering the startup cost of DAB devices. The SSIM consists of a template of a matching circuit, a microprocessor, and a receiving-type sensor. The template uses a nonuniform microstrip line interconnected with relay switches. To enhance the dynamic range and reduce the insertion loss for the template simultaneously, this work employs a new design technique based on solving the mathematical programming problem of matching requirements. The resultant matching domain covers a large area of the Smith chart, although only five switches are used. Experimental results show that the SSIM successfully adapts an interphone/DTV antenna into a DAB receiving mode. The operational bandwidth meets the DAB specification, and the transducer gain is larger than −1.83 dB.

Orthogonal Frequency Division Multiplexing (OFDM) is a type of Multicarrier Modulation (MCM) technique in which entire bandwidth is divided into large number of small sub-carriers and each subcarrier is transmitted parallel to achieve higher data rates. It has various applications like Digital Audio Broadcasting (DAB), Digital Video Broadcasting (DVB) and wireless LAN.OFDM technique is widely used in wireless communication system because of its very high data rate. The performance of FFT based OFDM system using Linear and cyclic channel coding and Constant Modulus Algorithm (CMA) equalizer is simulated using simulink model. The BER saving using the optimized proposed model with both linear and cyclic channel coding along with CMA equalizer is evaluated. The proposed work using cyclic channel coding with QPSK/QAM modulation and CMA as channel equalization under AWGN channel results in 52.6% and 96.3% BER reduction as compared to conventional OFDM model without channel coding, channel equalization and channel fading. So, CMA equalizer is used to enhance the performance of OFDM system.

Orthogonal frequency division multiplexing is a type of multicarrier modulation technique which is used in wireless communication. OFDM can be used in various wireless and wired application such Digital audio broadcasting, digital video broadcasting & wireless LAN. The Performance of OFDM system depends on various parameters such as using different transform, different modulation and by putting various channel coding. In this paper OFDM system is modeled by using different transform (DWT/FFT), different modulation (BPSK, QPSK, QAM) and different channel coding (Linear / Cyclic block codes) for AWGN channel. The performance parameter is calculated as Bit error rate (BER) for various transform/modulation/channel coding based OFDM systems. MATLAB Simulink tool is used to calculate the bit error rate (BER).

The Digital Radio Mondiale (DRM) digital radio standard is the only one recommended by the ITU to be used in Short Wave bands. These bands present unique characteristics for the broadcasting, very big coverage areas and long-range transmission. But they suffer of complicated propagation channel with big time variations that make difficult the regular broadcasting operation. In order to evaluate the DRM performance, field trials were made in Brazil by the Brazilian Government. The objective was to evaluate DRM technology for long-range transmission in Short Wave frequencies. In this paper, the results of these field trials are explained. Two frequencies were used during a period of three weeks at different hours of the day. The transmission was made from French Guyana and the reception was tested in Brasilia.

The digital audio broadcasting (DAB) system is mainly designed for large-area coverage and is suited for a single-frequency network (SFN) operating in Very High Frequency (VHF) Band III. To operate successfully, the SFN requires a different network-planning approach for multiple-frequency networks and should avoid self-interference. This paper focuses on nationwide SFN planning and DAB channel-allocation planning to replace all frequency-modulation (FM) services with terrestrial DAB services in VHF Band III. In South Korea, VHF Band III has been used for terrestrial digital multimedia broadcasting (T-DMB) services; thus, available frequencies should be investigated for T-DAB services and the nationwide SFN. An approach for broadcasting channel-interference analysis is proposed herein. For the interference analysis, the geographical features and the information for stations of legacy FM radio broadcasting and T-DMB, such as transmission parameters, are considered. The ratio of the interference area to the coverage is calculated via the channel-interference analysis. Areas affected by the self-interference in the nationwide SFN are investigated. The results of this paper provide insights regarding channel-allocation planning for T-DAB services and an approach for nationwide SFN planning.

This paper investigates the impact of different audio codecs typically deployed in current digital audio broadcasting (DAB) systems and web-casting applications, which represent a main source of quality impairment in these systems and applications, on the quality perceived by the end user. Both subjective and objective assessments are used. Two different audio quality prediction models, namely Perceptual Evaluation of Audio Quality (PEAQ) and Perceptual Objective Listening Quality Assessment (POLQA) Music, are evaluated by comparing the predictions with subjectively obtained grades. The results show that the degradations introduced by the typical lossy audio codecs deployed in current DAB systems and web-casting applications operating at the lowest bit rate typically used in these distribution systems and applications seriously impact the subjective audio quality perceived by the end user. Furthermore, it is shown that a retrained POLQA Music provides the best overall correlations between predicted objective measurements and subjective scores allowing to predict the final perceived quality with good accuracy when scores are averaged over a small set of musical fragments ( $\mathbf {R = 0.95}$ ).

Digital audio broadcasting services transmit substantial amounts of data that is encoded to minimize bandwidth whilst maximizing user quality of experience. Many large service providers continually alter codecs to improve the encoding process. Performing subjective tests to validate each codec alteration would be impractical, necessitating the use of objective perceptual audio quality models. This paper evaluates the quality scores from ViSQOLAudio, an objective perceptual audio quality model, against the quality scores of PEAQ, POLQA, and PEMO-Q on three datasets containing fullband audio encoded with a variety of codecs and bitrates. The results show that ViSQOLAudio was more accurate than all other models on two of the datasets and performed well on the third, demonstrating the utility of ViSQOLAudio for predicting the perceptual audio quality for encoded music.

Existing emergency alert techniques of terrestrial-digital multimedia broadcasting (T-DMB) and digital audio broadcasting (DAB) can be executed only when the receivers are activated. To overcome the limitation of existing technique, a novel emergency wake-up alert system within a conventional T-DMB service network is proposed. The emergency wake-up alert signal is periodically generated by emergency transmission data unit (ETDU) and then transmitted through the conventional T-DMB/DAB station. After receiving the emergency wake-up alert signal, T-DMB/DAB receiver can be wake-up, activating and result in showing emergency broadcasting contents to people. To obtain frequency efficiency, the emergency wake-up alert signal is overlaid onto the conventional T-DMB frequency band using a direct sequence spread spectrum technique. To assess the performance of the developed ETDU, simple RF transmitter, test receiver and measurement equipment are used. The performance is checked with different values of emergency wake-up alert signal to T-DMB power ratio. The laboratory test results shows that the wake-up alert signal can be transmitted without degrading the existing T-DMB signal quality and can be detected at very low power emergency wakeup alert signal, e.g., as low emergency wake-up alert signal to T-DMB power ratio as -22.2dB.

This paper describes a simple delay diversity technique for terrestrial digital multimedia broadcasting (T-DMB) and digital audio broadcasting in a single-frequency network (SFN). For the diversity technique, a delay diversity scheme is adopted. In the delay diversity scheme, a non-delayed signal is transmitted in the first antenna, and delayed versions of the signal are transmitted in each additional antenna. For an SFN environment with multiple transmitters, delay diversity can be executed by controlling the emission times of the transmitters. This SFN delay diversity scheme does not require any hardware changes in either the transmitter or receiver, and perfect backward compatibility can be acquired. To evaluate the performance improvement, laboratory tests are executed with various types of commercial T-DMB receivers as well as a measurement receiver. The improvement in the bit error rate performance is evaluated using a measurement receiver, and an improvement of the threshold of visibility value is evaluated for commercial receivers. Test results show that the T-DMB system can obtain diversity gain using the described technique.

Norway has announced the end of analogue radio, and plans to be the first country to turn off one of five FM transmitters on 11 January 2017; this will save £35m as FM costs eight times more to run than DAB. Although a number of countries are in the analogue-to-digital transition period, Norway will be the first to fully scrap traditional terrestrial radio and turn to DAB, providing 22 national channels, and potentially a further 20 more. Despite a successful switchover for terrestrial television in 2012, it is highly unlikely that radio stations will follow suit as the UK government has proposed a new deadline όf 2020 or abandoning the idea altogether. Ed Vaizey, the UK minister of state for culture and the digital economy, has acknowledged that the public's acceptance of DAB radios has been limited, with a lack of investment in DAB technologies in the UK.

This paper depicts the field trials and results for in-band digital radio technologies in Korea. For the in-band methods, HD Radio and DRM+ are considered. In HD Radio, MP3 hybrid mode and MP5 all digital modes are selected, and 4-QAM and 16-QAM are selected for DRM+. To evaluate the practical performance of each technology, system parameters and received electric field strength are evaluated in various test routes and environments. In the field trial, listening tests are also executed by test staffs, and the results of listening tests are analyzed and compared with results of system parameters. Since the trials consider various practical reception environments, the test results can be used as references for decision of digital radio standard.

The DRM+ (Digital Radio Mondiale) system is the extension up to 300 MHz of the DRM digital radio broadcasting system. During 2010 three different trials were conducted in Brazil with DRM+ system in VHF band II (popularly called FM band). These trials were defined to have a big, medium and small coverage area for big, local and community broadcasters. For this purpose, different transmission powers were used. The transmission was done with only DRM signal in low power trial. In the high and medium power trials the transmission was done in “combined mode”, that is, the FM and DRM+ signals were transmitted in adjacent channels from the same transmission site. In this paper the results of these trials are explained and compared with theoretical values.

A surface-acoustic-wave-less first folded-conversion and second down-conversion receiver for the AM/FM/digital radio mondiale (DRM)/digital audio broadcasting (DAB) radio standards is presented. This architecture is more suitable for broadcasting radio with ultra-narrow signal bandwidth since it relaxes the requirement on the local area (LO) generator by decreasing its tuning range. To reduce the blockers and out-of-band (OB) interferers, current amplifying and polyphase filters are introduced simultaneously. An inverter-based switched capacitor array technique is proposed in voltage-controlled oscillator (VCO) design to isolate the noise from the substrate. With a single VCO, the receiver can cover the frequency range from 100 kHz to 1.5 GHz. Without the RF pre-filters, the in-band ${\rm IIP}_{3}$ is 1.5 dBm, and the OB ${\rm IIP}_{3}$ is up to 17.4 dBm. The LO phase noise in the worst case is $-$ 119 dBc/Hz@1 MHz. The chip is fabricated in a 0.18- $\mu{\hbox{m}}$ CMOS technology, and mounted on a QFN package. It consumes only 86.4 mW under 1.8-V supply. To our best knowledge, this is first receiver reported thus far that supports all four AM/FM/DRM/DAB standards.

Many countries are converting their analog communication and broadcasting services into digital services. However, digital radio broadcasting service is still an issue to be addressed. In order to provide digital radio broadcasting service, it is very important to choose the most appropriate digital radio method for their frequency environment and listeners. This paper depicts laboratory test results of in-band digital radio technologies, HD Radio and DRM+. To contribute to decision of digital radio standard and design of digital radio transmission network, various results of many test items are expressed.

This paper depicts the digital radio test bed design and performance evaluation for digital audio broadcasting, DAB+, and terrestrial digital multimedia broadcasting audio systems that are representative digital radio broadcasting standards operating in band III (out-of-band). We consider a laboratory trial to measure the performance of each digital radio system. The existing broadcasting measurement systems have some limitations, such as the small number of available measurement items and broadcasting systems. However, the introduced test bed can cover above representative three digital radio systems and measure various performance parameters. This paper minutely describes not only the test bed designs but also various test results according to the measurement items. Therefore, this paper can help select an appropriate digital radio standard.

Frequency is an important parameter for the operation and control of power systems. One novel application of frequency measurements involves using this information to authenticate a digital audio/video recording presented as forensic evidence in legal proceedings. To apply this technique, called the electrical network frequency (ENF) criterion, both a reference frequency database and an accurate frequency estimation method are required. This paper briefly introduces a wide-area frequency monitoring network (FNET) as the reference frequency database and analyzes statistical features of frequency of the four North American interconnections in terms of different time scales. Combined with digital filtering and a three-sample interpolation, the short-time Fourier transform (STFT) is adjusted to estimate the ENF embedded in digital audio recordings. A procedure of using the ENF criterion, ranging from signal preprocessing to ENF estimation and reference frequency database matching, is then proposed. Further, oscillator error of the digital recording device is considered and an iterative error correction method is given to assist with the frequency database matching. Factors which influence the accuracy of frequency estimation, such as parameter selections of the STFT and signal-to-noise ratio, are also discussed. Test results show that the procedure is capable of performing digital audio authentication.

As an effective technique for combating multipath fading and for high-bit-rate transmission over wireless channels, orthogonal frequency division multiplexing (OFDM) is extensively used in modern digital television terrestrial broadcasting (DTTB) systems to support high performance bandwidth-efficient multimedia services. Chinese digital television terrestrial broadcasting standard recommends three kinds of Frame Head mode with PN420/PN595/PN945 as training sequences for the time domain synchronous OFDM (TDS-OFDM) based Digital Terrestrial Multimedia Broadcasting (DTMB) systems. At the receiver side, Frame Head mode detection and symbol detection should be done before subsequent synchronization and channel estimation. In this paper, a Frame Head mode detection and symbol detection scheme robust to carrier frequency offset is proposed for the TDS-OFDM based DTMB systems. Performance studies under different channel situations are used to verify the efficiency of the proposed scheme.

A novel low-loss image-reject mixer (IRM) using the source follower isolation method for digital radio mondiale/digital audio broadcasting applications is presented. It is based on double-quadrature mixers (DQMs) and passive polyphase filters (PPFs). The DQM is designed with four passive CMOS mixers. A source follower is inserted between the RF PPF and DQM to decrease the conversion loss of the IRM. Compared with the traditional isolation method, the source follower isolation method shows much higher on-resistance while looking into the RF port of the mixer; meanwhile, the chip area will be greatly decreased. This chip is fabricated in a 0.18-μm CMOS technology. The total conversion loss is 10.5 dB, and the conversion loss per stage of PPFs is 2.1 dB on average. The image-reject ratio is larger than 40 dB in the bandwidth of 1.536 MHz without trimming and calibration. The PldB gain compression point is 7.6 dBm. The chip core area is 1.1 × 1.3 mm2.

Digital Audio Broadcasting (DAB) receiver requires fast Automatic Gain Control (AGC) to compensate the strength loss due to the multipath propagation in mobile receiving condition. But fast AGC could also incorrectly amplify DAB's null symbol and the following phase reference symbol to cause the synchronization problem. In this paper, we mathematically analyze the AGC circuit under a Rayleigh fading channel and give the basic rule to generate the AGC feedback voltage. Then a fast and high accuracy AGC algorithm with least affection on the DAB null symbol is proposed. The algorithm is simplified to have only add, shift and compare operations, making it easy for hardware and software implementation. Finally the algorithm is simulated and verified on Field Program Gate Array (FPGA) device. The test results show that the proposed AGC algorithm can be settled within 0.5 ms, and keep the received signal stable at a moving speed from stationary to over 120 km/h, both in city and suburban area. The algorithm and the analysis method can also be used for AGC design in other wireless systems.

The 26 MHz band is currently allocated for international broadcasting by means of the ionospheric propagation. However, sky waves are strongly dependent on the ionosphere conditions and signal reception is not always possible. As a consequence, this HF frequency range is scarcely utilized and an optimization of its use is required. The DRM (Digital Radio Mondiale) standard provides enough robustness to enable reliable audio reception taking advantage of the ground wave propagation at these frequencies. According to this, local coverage is viable in this part of the spectrum, as proved the field trials and studies carried out in the last years. Nevertheless, it is still necessary to establish the general guidelines for the set up and consolidation of the corresponding digital networks. Thus, the planning factors and procedures for digital local broadcasting services in the 26 MHz band are defined in this paper, along with reference parameter values and recommended system configurations for this new application.

This paper investigates the network deployment of single frequency networks (SFN) based on OFDM schemes that are standardized for terrestrial broadcasting systems, for digital audio broadcasting (DAB) and for digital video broadcasting (DVB) systems. The concept of green network planning is presented. The term `green' refers to low carbon, energy efficiency and low exposure to radiation, parameters important for the sustainable growth. For the purpose of our investigation a mountainous area of Northern Greece is examined, that is described by a digital terrain elevation model (DTEM) and field computations are based on multi shape slope uniform theory of diffraction (UTD) technique. A genetic algorithm (GA) optimization method is developed for the network planning purposes. A comparison between various planning strategies is presented. It is shown that great CO2 reductions, cost savings and low exposure to radiation can be achieved when the network planning considers a `green' strategy.

In this paper, we propose an improved sampling frequency offset (SFO) estimation scheme for orthogonal frequency division multiplexing (OFDM) based digital radio mondiale (DRM) broadcasting systems. To demonstrate the performance of the proposed frequency estimator, analytical expression of the mean square error (MSE) is derived and performances are compared with conventional pilot-assisted estimators. Based on the simulation and theoretical analysis, our proposed estimator shows an improved estimation performance compared to the conventional estimators.

Orthogonal frequency division multiplexing (OFDM), the multicarrier modulation technique with high bandwidth efficiency and robustness against multipath fading, is used in several high-speed broadband communication systems including digital video broadcasting (DVB), digital audio broadcasting (DAB), asymmetric digital subscriber line (ADSL), wireless local area network (WLAN IEEE 802.11a,g), high performance radio local area network (HIPERLAN 2), and worldwide interoperability for microwave access (WiMAX IEEE 802.16). However, the transmit signal in OFDM system has a high peak-to-average power ratio (PAPR), one of the major drawbacks of multicarrier transmission, and therefore high dynamic range is required in both the digital to analog converter (DAC) and the power amplifier (PA) for proper operation. Otherwise, the nonlinearities of PA cause out-of-band distortion and increase in bit error rate (BER). Therefore, the reduction in PAPR in OFDM system is desirable in order to obtain power efficiency and increase BER performance. In this paper, a new parametric PAPR reduction technique, L"2-by-3 transform from sliding norm transform (SNT) family is proposed. Based on its parameter, a significant PAPR reduction is obtained. This paper presents the analysis of power saving and computational complexity of the proposed method L"2-by-3 and comparison to two other methods; selected mapping and partial transmit sequence.

This paper describes the bit error rate (BER) performance of the Enhanced Packet Mode (EPM) of the Eureka 147 Digital Audio Broadcast (DAB) system in a mobile environment. The BER performance of DAB-EPM is characterized by computer simulations in frequency selective (typical case) and flat faded (worst case) mobile channels. For each of these channels, simulations are performed at vehicle speeds ranging from very slow (walking speed) to very fast (high speed trains). Simulation results show that, in typical frequency selective mobile environments, the target BER of 10 8 required to deliver adequate packet data service via the EPM is achieved at Carrier to Noise Ratio (C/N) values smaller than the target value of 14 dB used in coverage and frequency planning at most of the vehicle speeds tested. Moreover, the coverage area of packet data services is equivalent to the coverage of audio services delivered with the MPEG Layer 2 audio codec and the Unequal Error Protection Level 3 of DAB.

This letter proposes a low complexity integer frequency offset (IFO) estimation scheme in an orthogonal frequency division multiplexing (OFDM) based digital radio mondiale plus (DRM+) system with nonuniform phased pilot symbols. To reduce the computational complexity, the pilot symbols used for frequency estimation in the DRM+ system are partitioned into a number of pilot subsets, and pilots in each subset are phase-rotated to have a unique phase so that the estimator is made computationally efficient.

In this paper, a Terrestrial Digital Multimedia Broadcasting (T-DMB) system for a multichannel audio service is proposed. The multichannel audio information on the T-DMB system proposed in this paper consists of a stereo audio signal coded by BSAC and additional multichannel side information coded by MPEG surround. These two types of information are conveyed through two elementary streams (ES), respectively. To control two elementary streams, two object descriptor (OD) streams are exploited, each of which transmits OD commands, including ObjectDescriptorUpdate and ES_DescriptorUpdate, related to a stereo audio signal and additional multichannel side information, respectively. With this method, the proposed system can convey multichannel side information while guaranteeing backward compatibility for the legacy T-DMB terminals, and moreover makes it possible to provide various advanced service scenarios with outstanding extensibility.

In this paper, we designed an integrated mobile terrestrial Digital Multimedia Broadcasting (DMB) player. The newly designed DMB player provides audio and video displays in accordance with the T-DMB, CMMB, and ISDB-T standards. In order to support displaying audio and video stream data of these three different standards selectively, we developed a unified Transport Stream (TS) parser, implemented new DMB player applications that provide the necessary functionalities and menus. In the near future users of this new terrestrial DMB player will be able to receive mobile terrestrial DMB data as they travel in Korea, China and Japan without the need of a corrective DMB receiver device for each country.

This paper presents a CORDIC (Coordinate Rotation Digital Computer)-based split-radix fast Fourier transform (FFT) core for OFDM systems, for example, Ultra Wide Band (UWB), Asymmetric Digital Subscriber Line (ADSL), Digital Audio Broadcasting (DAB), Digital Video Broadcasting - Terrestrial (DVB-T), Very High Bitrate DSL (VHDSL), and Worldwide Interoperability for Microwave Access (WiMAX). The high-speed 128/256/512/1024/2048/4096/8192-point FFT processor has been implemented by 0.18 @mm (1p6m) at 1.8 V, in which all the control signals are generated internally. This programmable FFT processor outperforms the conventional ones in terms of both power consumption and core area.

In a shared telecommunications medium such as the Eureka 147 Digital Audio Broadcasting (DAB) system, we expect that available bandwidth will be optimally allocated to different users. This paper proposes optimal bandwidth allocation by use of statistical multiplexing that both service providers and multiplex operators can benefit from. Moreover, an auditory-based objective measure (i.e., ITU-R PEAQ) has been employed to make sure that the encoded audio materials maintain ?broadcast quality?. In our simulations of statistical multiplexing of five different ?broadcast quality? radio programs, we show that a variable bit rate (VBR) stream with an average bit rate of 192 kbps could be saved as extra bandwidth in DAB to create new services. We also show that this VBR stream provides at least 112 kbps of extra capacity 98.5% of the time. The extra bandwidth can be used to transmit new audio programs, data or any side information to augment the radio listening experience.

A number of mechanisms can result in the generation of spurious tones in Orthogonal Frequency Division Multiplexing (OFDM) receivers. In this work we determine the Signal-to-Noise-Ratio (SNR) degradation at the output of the Fast-Fourier-Transform (FFT) block and evaluate the performance of a Digital Audio Broadcasting (DAB) receiver in the presence of such tonal interference. We show that the degradation varies with the central frequency of the interference and demonstrate that relatively weak spurious tones can compromise the performance of a DAB receiver. We provide specific evaluations of the severity of the above effect for all possible DAB systems parameterizations and evaluate the permissible levels of spurious tones in such systems. It is shown that for DAB EEP Mode A protection level 4, tonal interference of 14 dB lower than the signal can compromise the receiver performance.

This paper presents a new audio quality monitoring algorithm: ldquoAQUArdquo for assessing the performance of Digital Radio Mondiale (DRM) receivers. AQUA targets performance evaluation of consumer grade receivers in fading channels objectively. Consumer grade DRM receivers do not generally possess a Receiver Status and Control Interface (RSCI), which prohibits the possibility of analysing digitally decoded audio bit stream after reception. AQUA relies on the analysis of the received analog audio output signal instead. The paper describes the development of the AQUA algorithm and the measurement system used for receiver audio quality testing. The performance of AQUA is compared to conventional RSCI derived metrics like Bit Error Rate (BER) and Frame Error Rate (FER) of the decoded audio data. A series of tests are conducted in a variety of propagation conditions using a DRM dedicated testing system with a fading channel emulator. It is shown that FER estimates using AQUA are in good agreement with conventional FER figures derived from RSCI in different fading channel scenarios. This allows AQUA to be used as a reliable alternative to conventional FER performance analysis for automated objective testing of DRM receivers.

Terrestrial or Handheld Digital Video Broadcasting (DVB-T/H) and Terrestrial Digital Multimedia Broadcasting (T-DMB) are two popular broadcasting standards that enable digital television transmissions to handheld receivers. This paper presents a comprehensive performance comparison between the physical layers of DVB-T/H and T-DMB when employed for mobile communications. By exploiting a recently proposed fast simulation model, we assess the BER of the two coded OFDM systems in several realistic scenarios, taking into account Rayleigh and Rice channels, different mobile speeds, inner and outer channel coding, channel estimation, and one or two receive antennas. Our comparison shows that the DVB-T/H physical layer performance highly depends on the delay spread of the channel, whereas T-DMB is less sensitive to the frequency selectivity of the channel. As a result, DVB-T/H yields better performance than T-DMB in typical Rayleigh channels with significant delay spread. On the contrary, at high SNR, T-DMB outperforms DVB-T/H in Rice channels with low delay spread. As a side result, we show the performance improvement of DVB-H produced by MPE-FEC at the data link layer.

In this study, we implemented an enhanced A/V synchronization method including BIFS processing for new portable T-DMB players. The T-DMB supports simultaneous audio, video, and BIFS transmission. However, delayed correction or non-correction of the audio and video synchronization can be annoying to the user. Here, a novel synchronization strategy is applied for video, audio, and BIFS using a control buffer. The proposed method preferentially synchronizes audio data with an internal clock system and performs resynchronization by comparing audio CTS signal with video CTS signal. The designed system could complete the synchronization of audio and video including BIFS data, thus allowing stable T-DMB services in unreliable transmission environments such as shadow regions.

After coarse synchronization in orthogonal frequency division multiplexing (OFDM) systems, there might still be a residual frequency offset (RFO) and a sampling clock frequency offset (SFO), which seriously degrade the performance of the systems. This paper proposes a simple way of jointly estimating the RFO and SFO by using a continual pilot (CP) defined in an OFDM-based digital video broadcasting (DVB) system. For deriving an unbiased joint RFO and SFO estimator in the current DVB system, a CP subset is selected to simultaneously remove the effect of RFO and SFO. We can observe by simulation that the proposed frequency estimator endowed with a properly selected CP subset is unbiased and performs robustly.

As the US migrates to digital radio, broadcasters are making important decisions about the transmission rates they use on primary and secondary audio channels. One way to evaluate coders at different bit rates is to elicit consumers' opinions of the audio quality. However, it is evident from several lines of research that consumers' preferences do not always match their ability to process information efficiency. Therefore when evaluating coders at low bit-rates it is important to measure consumer's processing efficiency as well as examining their preferences. With regard to audio, past research has shown that degraded speech, including speech masked with noise and synthetic speech, impairs individuals' ability to recall content. The current research explores whether speech coded at very low bit rates (i.e., 9 kbps and 24 kbps) also impairs consumers' memory for text passages. Concurrently, we examine people's subjective ratings to see whether these ratings correlated with our objective recall measure. Results suggest that participants' recall of information dropped significantly at 9 kbps, a finding which has important implications for broadcasters. Participants' audio quality ratings did not correlate with their ability to remember story details, supporting the notion that merely asking consumers to rate the quality of a signal may not provide a complete picture of how impaired audio will affect their behavior. However, interestingly, their professed level of interest, comprehension, and enjoyment with the story did correlate with their memory scores, making these questions better candidates for subjective tests of very low bit rate coders.

STiMi is a mobile multimedia broadcasting trade standard that was released by the Chinese State Administration of Radio Film and Television in 2006. This paper details the algorithm research and development work on the acquisition for STiMi. First, the STiMi physical layer is briefly reviewed. Secondly, individual algorithms, including time slot detection, fractional frequency offset estimation, integer frequency offset estimation and fine timing synchronization are discussed with simulation results presented. An implemented STiMi receiver prototype is also introduced with a focus on the acquisition algorithms implementation. Finally, the whole acquisition procedure is summarized. Acquisition algorithms and structures presented in this paper are adopted for prototyping and perform well.

Commercial terrestrial digital multimedia broadcasting (T-DMB) service was launched in the Republic of Korea. Reflecting recent demands of high quality video, Advanced T-DMB(AT-DMB) adopting a hierarchical modulation is being developed as the way to increase the payload data. Based on the AT-DMB system, this paper proposes a method to provide increased video quality with high spatial resolution. By using the emerging H.264/MPEG-4 AVC amendment 3 scalable extension (SVC), the proposed method is able to support high spatial resolution as well as the currently serviced spatial resolution while retaining the backward compatibility to T-DMB. In the experimental results, the benefits in terms of coding efficiency and spatial scalability are proven in comparison with the conventional T-DMB and alternative methods to provide a high spatial resolution video.

Implementing an IP-friendly transport mechanism in broadcasting systems would enable more convenient multimedia broadcasting applications. The overhead of current multiplexing methods for IP packets that are based on MPEG-2 systems is large, and they are not well suited for efficient transmission of IP packets. We developed an efficient scheme for transporting IP packets over broadcasting channels. The scheme includes compression of IP header information, transmission control signals for listing IP multicast group addresses, and a simple encapsulation. This paper describes the multiplexing scheme and its theoretical overhead. We conducted an experiment on a prototype transmission system that shows our scheme is efficiency.

This paper presents a blind synchronizer for orthogonal frequency-division multiplexing (OFDM) systems based on signal-to-interference-and-noise ratio (SINR) maximization. Due to the incurred losses from intersymbol interference (ISI) and intercarrier interference (ICI) introduced by synchronization errors, the SINR of the received data drastically drops. By taking advantage of this characteristic, both the symbol time and carrier frequency offsets are intuitively estimated by maximizing the SINR metric. For the SINR metric, two blind SINR estimations are investigated. The estimations do not need prior knowledge of the channel profiles and transmitted data. As such, the proposed maximum SINR (MSINR) synchronization algorithm is nondata aided so that the transmission efficiency can be improved. Moreover, to reduce the computational complexity, the early-late gate technique is proposed for the implementation of the synchronizer. Simulation results exhibit better performance for the MSINR algorithm than conventional techniques in multipath fading channels.

Symbol timing offset (STO) can result in intersymbol interference (ISI) and a rotated phase which value is proportional to the subcarrier index at the FFT output in an OFDM receiver. In order to avoid ISI, the FFT window start position has to be put in advance of the estimated point obtained by coarse STO estimation algorithms. But a large number of forward-shift samples will deteriorate the estimation of data subcarrier channels requiring interpolation from pilot subcarrier channels due to the phase rotation caused by a residual STO. In this paper we analyze the influence of STO on channel interpolation and propose a new compensation method for channel correction with STO. From the performance analysis of simulation results in the DVB-T application, the new algorithm not only has a better performance but also is attractive in using a simple residual STO estimation method such that few pilots are required for fine STO estimation compared to conventional approaches.

In orthogonal frequency division multiplexing (OFDM) system, phase noise (PHN) from oscillator can severely reduce the performance by causing: common phase error (CPE) and inter-carrier interference (ICI). The impact of PHN on received signal can be characterized by the circular convolution of the transmitted signal and the discrete Fourier transform (DFT) of PHN signal in frequency domain. Digital television/terrestrial multimedia broadcasting (DTMB), announced as the Chinese digital television terrestrial broadcasting (DTTB) standard, adopted the time-domain synchronous OFDM system (TDS-OFDM) system which has 36 continuous transmission parameter signaling (TPS) symbols in frequency domain which be used for PHN suppression. Based on a linearized parametric model for PHN, least squared (LS) estimation and fast Fourier transform (FFT) approaching method for PHN correction are proposed in this paper. The effects of the PHN on channel estimation in both TDS-OFDM system and conventional cyclic prefix OFDM (CP-OFDM) are also investigated. Simulations show that the proposed algorithm can effectively reduce the PHN and improve the symbol error rata (SER) of TDS-OFDM systems over both additive white Gaussian noise (AWGN) and broadcasting multipath channel. In practice, the FFT-approaching method can be applied to the DTMB receiver-chip design directly with low implementation cost.