Digital audio broadcasting, DAB, is the most fundamental advancement in radio technology since that introduction of FM stereo radio. It gives listeners interference â€ free reception of CD quality sound, easy to use radios, and the potential for wider listening choice through many additional stations and services.
DAB is a reliable multi service digital broadcasting system for reception by mobile, portable and fixed receivers with a simple, non-directional antenna. It can be operated at any frequency from 30 MHz to 3GHz for mobile reception (higher for fixed reception) and may be used on terrestrial, satellite, hybrid (satellite with complementary terrestrial) and cable broadcast networks.
DAB system is a rugged, high spectrum and power efficient sound and data broadcasting system. It uses advanced digital audio compression techniques (MPEG 1 Audio layer II and MPEG 2 Audio Layer II) to achieve a spectrum efficiency equivalent to or higher than that of conventional FM radio.
The efficiency of use of spectrum is increased by a special feature called Single. Frequency Network (SFN). A broadcast network can be extended virtually without limit a operating all transmitters on the same radio frequency.
EVOLUTION OF DAB
DAB has been under development since 1981 of the Institute Fur Rundfunktechnik (IRT) and since 1987 as part of a European Research Project (EUREKA-147).
Â¢ In 1987 the Eureka-147 consoritium was founded. Itâ„¢s aim was to develop and define the digital broadcast system, which later became known as DAB.
Â¢ In 1988 the first equipment was assembled for mobile demonstration at the Geneva WARC conference.
Â¢ By 1990, a small number of test receivers was manufactured. They has a size of 120 dm3
Â¢ In 1992, the frequencies of the L and S â€ band were allocated to DAB on a world wide basis.
Â¢ From mid 1993 the third generation receivers, widely used for test purposes had a size of about 25 dm3, were developed.
Â¢ The fourth generation JESSI DAB based test receivers had a size of about 3 dm3.
1995 the first consumer â€ type DAB receivers, developed for use in pilot projects, were presented at the IFA in Berlin.
1992 â€ 1995 â€ field trial period.
1996 â€ 1997 â€ introduction period
98 onwards â€ terrestrial services in full swing
For DAB via satellite 1996 â€ 2001 is planned as experimental stage 2002 â€ 2003 introduction period.
DIGITAL AUDIO DATA
The conversion of analog audio data to the digital domain begins by sampling the audio input in regular, discrete intervals of time and quantizing the sampled values into a discrete number of evenly spaced levels. The digital audio data consists of a sequence of binary values representing the number of quantizer levels for each audio sample This method of representing each sample with an independent code word is called pulse code modulation (PCM).
The digital representation of audio data offers many advantages.
Â¢ High noise immunity
Â¢ Allows the efficient implementation of many audio processing functions (i.e. mixing, filtering, equalization) though the digital computer.
According to the Shannonâ„¢s theory, a time sampled signal can faith represent signal up to half the sampling rate. The max audible frequency for humans is 20 KHz. Therefore the typical sampling rate is 48 KHz. (i.e. more than twice the signal frequency).
DIGITAL AUDIO COMPRESSION
Digital audio compression allows the efficient storage and transmission of audio data. While quantizing, the number of quantizer levels is typically a power of 2 to make full use of a fixed no: of bits per audio sample to represent the quantized values. With uniform quantizer step spacing, each additional bit has the potential of increasing the signal to noise ratio. The typical number of bits per sample used for digital audio is 8, 16, 32, 64. The audio data on a compact disc (2 channels of audio samp1. at 44.1 KHz with 32 bits per sample) requires a data rate of 32x2x44xl000( megabits per second. Ti) transfer this uncompressed data requires a large data transfer rate and a larger bandwidth. Therefore audio data need to be compressed for efficient storage and transmission.
The MPEG (Motion Picture Experts Group) audio compression algorithm is an International Standardization Organization (ISO) standard for high fidelity audio compression. The high performance of this compression algorithm is due to the exploitation of auditory masking. This masking is a perceptual weakness of the ear that occurs whenever the presence of a strong audio signal in spectral neighborhood of weaker audio signals makes it imperceptible. This noise-masking phenomenon has been observed and corroborated through a variety of psycho acoustic experiments. Due to the specific behaviour of the inner ear, the human auditory system perceives only a small part of the complex audio spectrum. Only those parts of the spectrum located above the masking threshold of a given sound contribute to its perception, where as any acoustic action occurring at the same time but with less intensity and thus situated under the masking threshold will not be heard because it is masked by the main sound event.
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