Name : 1) Abhishek Yadav ) Prakash Giri ) Kheni Niral ) Bhadresh Langadiya Branch : Electronics & Communication Faculty : 1) V. S. Kansara

Contents Digital Audio Fundamentals Sampling and Quantizing PCM Audio Compression Disk-Based Recording Digital Audio Broadcasting Stereophony and Multichannel Sound

Digital Audio Fundamentals Digital audio is simply an alternative means of carrying an audio waveform

Digital Audio Fundamentals Digital audio is sound reproduction using pulse-code modulation and digital signals Digital audio systems include analog-to-digital conversion (ADC), digital-to-analog conversion (DAC), digital storage, processing and transmission components A primary benefit of digital audio is in its convenience of storage, transmission and retrieval Digital audio is useful in the recording, manipulation, mass-production, and distribution of sound Modern distribution of music across the Internet via on-line stores depends on digital recording and digital compression algorithms

PCM(Pulse Code Modulation) PCM consists of three steps to digitize an analog signal: 1.Sampling 2.Quantization 3.Binary encoding  Before we sample, we have to filter the signal to limit the maximum frequency of the signal as it affects the sampling rate.  Filtering should ensure that we do not distort the signal, ie remove high frequency components that affect the signal shape.

PCM(Pulse Code Modulation)

Sampling Analog signal is sampled every T S secs. T s is referred to as the sampling interval. f s = 1/T s is called the sampling rate or sampling frequency. There are 3 sampling methods: Ideal - an impulse at each sampling instant Natural - a pulse of short width with varying amplitude Flattop - sample and hold, like natural but with single amplitude value The process is referred to as pulse amplitude modulation PAM and the outcome is a signal with analog (non integer) values

Sampling

Quantization Sampling results in a series of pulses of varying amplitude values ranging between two limits: a min and a max The amplitude values are infinite between the two limits. We need to map the infinite amplitude values onto a finite set of known values This is achieved by dividing the distance between min and max into L zones, each of height   = (max - min)/L The midpoint of each zone is assigned a value from 0 to L-1 (resulting in L values) Each sample falling in a zone is then approximated to the value of the midpoint

Quantization and encoding of a sampled signal

PCM Decoder To recover an analog signal from a digitized signal we follow the following steps: We use a hold circuit that holds the amplitude value of a pulse till the next pulse arrives. We pass this signal through a low pass filter with a cutoff frequency that is equal to the highest frequency in the pre-sampled signal. The higher the value of L, the less distorted a signal is recovered.

PCM Decoder

Audio Compression In its native form, high-quality digital audio requires a high data rate, which may be excessive for certain applications One approach to the problem is to use compression, which reduces that rate significantly with a moderate loss of subjective quality While compression may achieve considerable reduction in bit rate, it must be appreciated that compression systems reintroduce the generation loss of the analog domain to digital systems

Audio Compression One of the most popular compression standards for audio and video is known as MPEG (Moving Picture Experts Group) In practice, audio and video streams of this type can be combined using multiplexing The program stream is optimized for recording and is based on blocks of arbitrary size The transport stream is optimized for transmission and is based on blocks of constant size

Audio Compression The bit stream types of MPEG-2

Audio Compression Compression and the corresponding decoding are complex processes and take time, adding to existing delays in signal paths Concealment of uncorrectable errors is also more difficult on compressed data The acceptable trade-off between loss of audio quality and transmission or storage size depends upon the application For example, one 640MB compact disc (CD) holds approximately one hour of uncompressed high fidelity music, less than 2 hours of music compressed losslessly, or 7 hours of music compressed in the MP3 format at a medium bit rate A digital sound recorder can typically store around 200 hours of clearly intelligible speech in 640MB

Disk-Based Recording The magnetic disk drive was perfected by the computer industry to allow rapid random access to data, and so it makes an ideal medium for editing Development of the optical disk was stimulated by the availability of low-cost lasers Optical disks are available in many different types, some which can only be recorded once, whereas others are erasable Optical disks have in common the fact that access is generally slower than with magnetic drives and that it is difficult to obtain high data rates, but most of them are removable and can act as interchange media

Digital Audio Broadcasting Digital Audio Broadcasting (DAB) is a digital radio technology for broadcasting radio stations Advantages of DAB Broadcasting programs with good sound quality comparable to multi-media products such as MP3 Offering stable reception and removing noises Bringing diversified program choices to the audiences Enabling transmission of text / images

DAB sender Trans- mitter Trans- mission Multi- plexer MSC Multi- plexer ODFM Packet Mux Channel Coder Audio Encoder Channel Coder DAB Signal Service Information FIC Multiplex Information Data Services Audio Services Radio Frequency FIC: Fast Information Channel MSC: Main Service Channel OFDM: Orthogonal Frequency Division Multiplexing 1.5 MHz f carriers

DAB receiver Packet Demux Audio Decoder Channel Decoder Independent Data Service Audio Service Controller Tuner ODFM Demodulator User Interface FIC Control Bus (partial) MSC

Stereophony and Multichannel Sound It is a method of sound recording in which the recording contains information about the spatial arrangement of the sound sources When a stereophonic recording is reproduced, the listener hears a more natural sound that seems to come from many separate sources and to be arranged in the same way as during the recording The listener has the impression that the sound is “three-dimensional” and possessed of an added “depth.”

Stereophony and Multichannel Sound This effect is achieved through the separate recording of electrical signals from different microphones on individual channels and through the separate reproduction of the sound on each channel by loudspeakers

Stereophony and Multichannel Sound The arrangement of the loudspeakers must be similar to that of the microphones; that is, the right and left channels must coincide The quality of stereophonic sound reproduction improves with the number of channels used However, the number of channels is usually kept within certain limits to avoid undue complexity and excessive cost

Stereophony and Multichannel Sound 5.1 channel sound is an industry standard sound format for movies and music with five main channels of sound and a sixth subwoofer channel used for special movie effects and bass for music A 5.1 channel system consists of a stereo pair of speakers, a center channel speaker placed between the stereo speakers and two surround sound speakers located behind the listener. 5.1 channel sound is found on DVD movie and music discs and some CDs

Stereophony and Multichannel Sound 6.1 channel sound is a sound enhancement to 5.1 channel sound with an additional center surround sound speaker located between the two surround sound speakers directly behind the listener. 6.1 channel sound produces a more enveloping surround sound experience. 7.1 channel sound is a further sound enhancement to 5.1 channel sound with two additional side-surround speakers located to the sides of the listener’s seating position. 7.1 channel sound is used for greater sound envelopment and more accurate positioning of sounds