Abstract: Radio broadcasting technology has evolved rapidly over the last few years due to ever increasing demands for as high quality sound services with ancillary data transmission in mobile environment. DAB+ is the upgraded version of digital audio broadcasting (DAB). DAB and DAB+ coexist in many countries, so receivers are required to be compatible with both standards. In order to store and transfer audio information efficiently in modern electronic systems, audio data needs to be compressed. In this paper, maximum likelihood DAB audio coding is designed to achieve errorless audio decoding for DAB. A 1/2 bit rate encoder and decoder to achieve higher throughput is designed. Along with this technique clock gating technique is also proposed where clock gating is a predominant technique used for power saving. It is observed that the commonly used synthesis based gating still leaves a large amount of redundant clock pulses. Data-driven clock gating aims to disable these. To reduce the hardware overhead involved, flip-flops (FFs) are grouped so that they share a common clock enabling signal. We propose a practical solution based on the toggling activity correlations of FFs and their physical position proximity constraints in the layout.
I INTRODUCTION
Radio broadcasting is one of the most widespread electronic mass media comprising of hundreds of programme providers, thousands of HF transmitters and billions of radio receivers worldwide. Since the broadcasting began in the early 1920s, the market was widely covered by the AM services. Today with the invent of FM we live in a world of digital communication systems and services because of its advantages over analog systems like storage capacity, reliability, quality of service, miniaturization and many more.
The new digital radio system Digital Audio Broadcasting (DAB) has the capability to replace the existing AM and FM audio broadcast services in many parts of the World in near future. Thiswas developed in the 1990s by the Eureka 147 DAB project. DAB is very well suited for mobile receivers and provides very high tolerance against multipath reception and inter symbol interference (ISI). It allows use of single frequency networks (SFNs) for high frequency efficiency. In several countries in Europe and overseas, broadcasting
Manuscript received March, 2015
Sheik ibrahims,PGStudent, SRM University, Chennai. KasthuriBhaJ.K., Assistant Professor, SRM University, Chennai
organizations, network providers and receiver manufacturers are already implementing digital broadcasting services using the DAB system. Perceptual audio coding (MPEG-2), Coded Orthogonal Frequency Division Multiplexing (COFDM), provision for the multiplex of several programs and data transmission protocols, are the new concepts of digital radio broadcasting Guglielmo Marconi got interested in Hertzian waves and started experimenting in his father‟s villa near Bologna. He transmitted wireless signals over tens of kilometres in 1895, seven years after the first demonstration of electromagnetic waves in space by Hertz. In the meanwhile several other inventions were being made which ultimately led Marconi to develop long distance wireless telegraphy. It was the forerunner of radio and Marconi is wrongly called the „inventor of radio‟. Radio as we know it today, that is, audio broadcast by wireless was the culmination of many other inventions starting with the work of Thomas Alva Edison. Edison observed that if plates sealed in an incandescent lamp, which he himself had developed, were connected to the positive end of the filament through a galvanometer, current would flow. If the galvanometer lead was connected to the negative end of the filament, no current flowed. This phenomenon was known as the Edison effect (1883). J J Thompson showed that this current was due to the travel of electrons from the filament to the plate (1899). Later J A Fleming patented two electrode vacuum tube as a detector of high frequency oscillations through the rectifying action of the device. These discoveries laid the foundation of telecommunication and broadcasting. Marconi was not only a good scientist and inventor but was also good at exploiting the business potential of telecommunication. At the turn of the nineteenth century Marconi conceived the idea of wireless telegraphy and formed in 1897, the Wireless Telegraph and Signal Company for manufacturing wireless apparatus. Marconi‟s system was adopted for ship to shore communication after he was successful in receiving a prearranged
Morse code signal, which was sent from the far side of Atlantic in December 1901. The year 2001 was celebrated as the centenary of the first transatlantic wireless transmission. Marconi type of „transmitters‟ used a spark between two electrodes, which produced a small-scale discharge of energy resulting in crackles in headphones of a distant receiver. The spark between the electrodes was produced by a telegrapher‟s key making it possible to communicate by a code. The transmitters were called the „spark transmitters‟.
Design of Digital Audio Broadcasting (DAB)
and DAB+ Adaptive Channel Decoder with
Clock Gating Technique
Around the same time R A Fessenden devised another type of wireless transmitter equipment employing „alternator‟ – an electromechanical device which produced continuous waves (CW) of
a single frequency rather than burst of energy as in the case of Marconi type of transmitters. The „CW transmitters‟ could cover longer distances with lesser power. They could transmit and receive Morse code better. The first ever 1 KW continuous wave transmitter operating at 42 KHz was amplitude modulated with the Morse code signal in 1902. J A Fleming‟s invention of two element vacuum tube with rectifying properties in 1904 and the development of three element vacuum tube triode by Lee De Forest which could perform the function of an amplifier.
In the evolution of new and innovative systems in today‟s technology creating various measures to reduce power consumption and increase the efficiency of systems, we are constantly researching in the field of transmission and reception with added benefits overcoming their drawbacks. Its an endless area of development in the field of Audio Broadcasting. The advancement from the analog broadcasting we have reached the phase of Digital Broadcasting of Audio Signals. But the drawback lies in the development of Receivers forthis Digital Audio Broadcasting (DAB). The Receivers used till date is far from consumer‟s reach as they are not power efficient. So we propose a Receiver system which will be power efficient, economical and consumes less space. Thus we modify the existing system into a benefiting Receiver system
II THESIS OBJECTIVES
The work reported in this thesis evaluates the performance of reconfigurable DAB/DAB+ architecture. Performance studies have been carried out for DAB+ employing maximum likelihood path decoding scheme . Performance of MLP DAB/DAB in multipath fading channels with Rayleigh and Rician fading have been analyzed. Bit error rate (BER) has been considered as the performance index in all analysis. The analysis has been carried out with simulation studies under Modelsim.
DIGITAL AUDIO BROADCASTING
Digital Audio Broadcasting (DAB) is a digital radio technology for broadcasting radio stations, used in several countries; DAB may offer more radio programs over a specific spectrum than analogue FM radio.
DAB is more robust with regard
to noise and multipath fading for mobile listening, since DAB reception quality first degrades rapidly when the signal strength falls below a critical threshold, whereas FM reception quality degrades slowly with the decreasing signal.
DAB gives substantially higher spectral efficiency, measured in programs per MHz and per transmitter site, than analogue communication. This has led to an increase in the number of stations available to listeners, especially outside of the major urban areas. The DAB standard integrates features to reducemultipath fading and signal noise, which afflict existing analogue systems.Also, as DAB transmits digital
audio, there is no hiss with a weak signal, which can happen on FM.It is common belief that DAB is more expensive to transmit than FM. DAB uses higher frequencies than FM; Higher radiated powers, or a combination, to achieve the same coverage. A DAB network is also more expensive than an FM network.
As DAB requires digital signal processing techniques to convert from the received digitally encoded signal to the analogue audio content, the complexity of the electronic circuitry required to do this is high. This translates into needing more power to effect this conversion than compared to an analogue FM to audio conversion, meaning that portable receiving equipment will tend to have a shorter battery life, or require higher power (and hence more bulk). This means that they use more energy than analogue Band II VHF receivers. As an indicator of this increased power consumption, some radio manufacturers quote the length of time their receivers can play on a single charge.
Digital audio signal compression has improved to a point where the digital audio signal occupies less bandwidth than an analog signal for the same quality, much lower bandwidth than the CD standard. Current ISO MPEG 2 Audio layer 3 (MP3) provides quasi-stereo CD quality at 64 kbps, that is reduced by more than 15 fold compared to CD. New ISO MPEG-2 AAC/AAC+ audio coding provides CD quality at 48 kbps.Digital signals are more robust with respect to external interference thanks to dedicated error correcting digital signal processing.Digital audio signals may be multiplexed with any other signal, such as image, test or files, and offers an enriched multimedia content. Also interfaces with other applications are eased. Digital signal processing is cheaper than analog processing thanks to silicon large scale integration.
Existing works
DAB+ is the upgraded version of digital audio broadcasting (DAB). DAB and DAB+ coexist in many countries, so receivers are required to be compatible with both standards. In this paper, a solution integrating an MPEG1-LayerII (MP2) decoder and an advanced audio coding (AAC) low-complexity (AAC LC) decoder is proposed to provide basic audio decoding for both DAB and DAB+. It also utilizes simple methods to improve high frequencies and stereo quality instead of complicated spectrum band replication and parametric stereo. A highly integrated low-power audio decoder design compatible with DAB/DAB+ and using a purely ASIC approach is presented. As a result of the system structure optimization and hardware sharing, the audio decoder is fabricated in 1P4M 0.18-μm CMOS technology using only 3.2 mm2 silicon area (including 147 456 bits RAM and 170 496 bits ROM).
backward compatible with MP2. So in order to decode both DAB and DAB+ programs, the receiver must have both MP2 and HE AAC V2 audio decoders, as shown in Fig. 1. Several solutions for DAB+ receivers have been published
[6]–[8]. All these solutions use embedded CPU or DSP cores and run software to decode MP2 and HE AAC V2 audio. The advantages of such solutions are flexibility and fast time to market. However, the architectural components of the DSP are usually not dedicated to MP2 and HE AAC V2 decoding, making such solutions costly and highly power consuming for implementing complicated decoding algorithms. So, though these solutions can be implemented to produce DAB/DAB+ compatible receivers, their cost and power consumption are high. In order to lower cost and power consumption, the best choice is to design the DAB and DAB+ decoders as an ASIC chip. ASIC designs have less flexibility than the CPU or DSP based designs, but can achieve the optimal hardware design for the dedicated algorithm. There have been ASIC design approaches for separated MP2 and AAC decoders. We previously designed a pure hardware MP2 decoder with only 10-mW power consumption using 0.18-μm CMOS technology in 2007 [9]. Liu et al. [10] reported a VLSI implementation for portable application-oriented AAC audio codec. Tsai [11], [12] reported two low-power-consumption AAC audio decoder ASIC chips with 2.45 mW, 3 × 3 mm using 0.18-μm CMOS technology, and 0.21 mW, 2.7 × 2.7 mm
MP2 decoder structure
AAC LC decoder structure.
But it has some drawbacks such as it used only off chip memory and off chip ADC/DAC. Support only low bit rate. Hardware complexity is increased. The external interface is need to more and it‟s difficult to debug. Proposed works
DAB+ is the upgraded version of digital audio broadcasting (DAB). DAB and DAB+ coexist in many countries, so receivers are required to be compatible with both standards. DAB+ aims to achieve higher spectrum efficiency by adopting HE AAC V2. Other than powerful AAC, the special techniques of SBR and PS are used to improve the sound quality for low bit rate programs. DAB+ increases spectrum efficiency by two factors. One is using powerful AAC instead of MP2. The other is using SBR and PS to ignore high-frequency components and stereo information and then improving the high-frequency quality and stereo effect, to some extent, by special techniques but not to the original quality. In practice, HE AAC V2 is mainly used in low data rate applications, where audio quality and bandwidth are both critical. Due to high throughput, DAB+ has serious effect of single tone and multi tone frequency distortions. To raise the signal quality, we proposed a maximum likelihood decoding scheme to make detector itself to eliminate error occurred from both single tone and multi tone distortions.
The proposed system explains the design of area efficient high throughput multimedia processor which is going to be designed using single chip based on system on chip design. Existing methods need many external interfaces such that ADC, DAC, external driver circuit etc .which is increase the hardware complexity and difficult to debug. To overcome this issue our proposed system is I²C based audio codec and implemented in FPGA which is verilog HDL. It does not need any external interfaces or off chip memories where design complete modules which could be dump in single chip. The proposed system used I²C controller to connect master and slave that provides faster communication and support multimedia rate audio input. And also I²C used to reduce design complexity compared to AMBE. These two are transmitting of signal as master and receiving the signals of slave. The memory is used for static RAM. as it‟s required less power and fast compared to dynamic ram.
Finite state machine is used for proposed system like as linear prediction. as FSM receives data from master, where the data‟s are considered as training sequence.
During training mode, FSM calculate frequency threshold which is considered as normal frequency level of the input audio. If the incoming audio signal have frequency at in range or out of range FSM cut off exceeded.
stream at the receiving node can be quite complex. Dr. A. J. Viterbi first described the classic decoder, currently in use, in the late 1960‟s. This
“maximum likelihood” technique greatly improved upon the earlier highly complex methods. Convolutional encoding of data combined with Viterbi decoding at the receiving node is an accepted industry standard for digital channels. Although the Viterbi decoder greatly simplified the process of reconstructing the original data stream, its operation is too complex a subject to be dealt with in this note. The discussion here will be limited to the convolutional encoder.
Maximum likelihood Viterbi Decoding
A Viterbi decoder uses the Viterbi algorithm for decoding a bit stream that has been encoded using a convolutional code.There are other algorithms for decoding a convolutional encoded stream (for example, the Fano algorithm). The Viterbi algorithm is the most resource-consuming, but it does the maximumlikelihood decoding. It is most often used for decoding convolutional codes with constraint lengths k<=10, but values up to k=15 is used in practice.Here we propose a new approach to design Viterbi decoder using finite state machines instead of using shift registers and adders. FSM reduces both static and dynamic power. The Viterbi algorithm is the optimal solution for theconvolutional codes .The nonlinear and recursive nature limits the maximum achievable throughput rate.
Proposed architecture
The most common solution to develop a high throughput Viterbi decoder is fully parallel approach, However, as the constraint length rises, thehardware complexity increases exponentially, and so does the power consumption. The trace back algorithm for survivor memory management uses the k-pointer algorithm which divides the memory into banks, and accesses them concurrently to achieve the demanded data bandwidth. Power consumption is a key factor in mobile or battery-powered systems. This paper proposes a modified trace back scheme that reduces the memory access based on the path merging property which will be discussed with mathematical formulation
III TYPES OF VITERBI UNITS
Branch metric unit (BMU)
A branch metric unit's function is to calculate branch metrics, which are normed distances between every possible symbol in the code alphabet, and the received symbol.There are hard decision and soft decision Viterbi decoders. A hard decision Viterbi decoder receives a simple bit stream on its input, and a Hamming distance is used as a metric. A soft decision Viterbi decoder receives a bit stream containing information about the reliability of each received symbol.
Path metric unit (PMU)
A path metric unit summarizes branch metrics to get metrics
for paths, where K is the constraint length of the code, one of which can eventually be chosen as optimal. Every
clock it makes decisions, throwing off wittingly non optimal paths. The results of these decisions are written to the memory of a trace back unit.
The core elements of a PMU are ACS (Add-Compare-Select) units. The way in which they are connected between themselves is defined by a specific code's trellis
diagram.Since branch metrics are always , there must be an additional circuit preventing metric counters from overflow. An alternate method that eliminates the need to monitor the path metric growth is to allow the path metrics to "roll over",
It is possible to monitor the noise level on the incoming bit stream by monitoring the rate of growth of the "best" path metric. A simpler way to do this is to monitor a single location or "state" and watch it pass "upward" through say four discrete levels within the range of the accumulator. As it passes upward through each of these thresholds, a counter is
incremented that reflects the "noise" present on the incoming signal.
Trace back Unit (TBU)
Back-trace unit restores an (almost) maximum-likelihood path from the decisions made by PMU. Since it does it in inverse direction, a Viterbi decoder comprises a FILO (first-in-last-out) buffer to reconstruct a correct order.
IV HARDWARE SIMULATION
Hardware simulation It can be seen from the ModelSim simulation waveform diagram shown in Fig..
When inputting test signal (0-255 cycle), the first output (data out) in parallel of the synchronous bytes is 0,12,24,36... .Allowed, when the input data are enough and greater than or equal to the delay length, it will be completely interleaved and the data out (convolution de-interleaver) completely restored the input test signal . When the input clock frequency is 50MHz, the total time delay of convolution interleaver is 44880us.
Above Simulation waveforms shows DAB and DAB+ signal. It clearly shows DAB+ has higher data rate compare to DAB signal. Experimental result shows that our proposed approach makes better reconfigurable architecture with maximum signal to noise ratio compared to existing system.
V CONCLUTION
