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XCS703 GRAPHICS AND MULTIMEDIA SEP REPORT AUDIO COMPRESSION

Submitted by VENKATA LAKSHMI.K 08011012170 IV CSE C

AUDIO COMPRESSION
INTRODUCTION

What is Compression?
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It is a process of deriving more compact (i.e., smaller) representations of data Significant reduction in the data size to reduce the storage/bandwidth requirements

Goal of Compression
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Constraints on Compression
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Perfect or near-perfect reconstruction (lossless/lossy) Reducing redundancies Exploiting the characteristics of human vision

Strategies for Compression

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MOTIVATION FOR COMPRESSION

Massive Amounts of Data Involved in Storage/Transmission of Text, Sound, Images, and Videos in Many Applications Applications
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Medical imaging Teleradiology Space/Satellite imaging Multimedia Digital Video: entertainment, home use Digital photography A typical hospital generates close to 1 terabits per year NASA's EOS will generate 1 terabytes per day One 2-hour video = 1.3 terabits Video transmission speed = 180Mb/sec With MPEG1 (1.5Mb/s), need compression ratio of 120 With MPEG2 (4-10Mb/s), need comp. ratio of 18-45

Concrete Figures
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BASIC DEFINITIONS IN COMPRESSION/CODING Coding: Compression 2

 Codeword: A binary string representing either the whole coded data or one coded data symbol Coded Bitstream: the binary string representing the whole coded data. Lossless Compression: 100% accurate reconstruction of the original data Lossy Compression: The reconstruction involves errors which may or may not be tolerable Bit Rate: Average number of bits per original data element after compression BASIC SOUND DEFINITIONS A sound/audio (digital) signal is a sequence of values, called samples, where every sample is the intensity of the recorded sound at the corresponding moment in time. The sampling rate of a sound is the number of samples per second.

The CD quality sampling rate is 44.1K samples per second (or 44.1 KHz), usually at 16 bits per sample, though 24 bits per sample is now common. In digital sound used for miniDV, digital TV, and DVD, the sampling rate is 48 KHz. In DVD-Audio and in Blu-ray audio tracks, the sampling rate is 96 KHz or 192 KHz.

When the sampling rate is infinity, the signal becomes what is called "analog signal".

When signal is captured as an analog signal, it can be converted to a digital signal by an analog-to-digital converter (also called A/D converter or digitizer) If a digital signal is fed into a digital-to-analog converter (also called D/A converter), the output is obviously an analog signal. Modulators are D/A converters, and demodulators are A/D converters. So, a modem is both an A/D and D/A converter.

STRATEGIES FOR COMPRESSION: REDUNDANCY REDUCTION

Symbol-Level Representation Redundancy

Different symbols occur with different frequencies Variable-length codes vs. fixed-length codes Frequent symbols are better coded with short codes Infrequent symbols are coded with long codes Example Techniques: Huffman Coding

 Block-Level Representation Redundancy

Different blocks of data occur with varying frequencies Better then to code blocks than individual symbols The block size can be fixed or variable The block-code size can be fixed or variable Frequent blocks are better coded with short codes Example techniques: Block-oriented Huffman, Run-Length Encoding (RLE), Arithmetic Coding, Lempil-Ziv (LZ)

Inter-Pixel Spatial Redundancy

Neighboring pixels tend to have similar values Neighboring pixels tend to exhibit high correlations Techniques: Decorrelation and/or processing in the frequency domain Spatial decorrelation converts correlations into symbol- or block-redundancy Frequency domain processing addresses visual redundancy (see the next slide)

Inter-Pixel Temporal Redundancy (in Video)

Often, the majority of corresponding pixels in successive video-frames are identical over long spans of frames Due to motion, blocks of pixels change in position but not in values between successive frames Thus, block-oriented motion-compensated redundancy reduction techniques are used for video compression.

Visual Redundancy

The human visual system (HVS) has certain limitations that make many image contents invisible. Those contents, termed visually redundant, are the target of removal in lossy compression.

In fact, the HVS can see within a small range of spatial frequencies: 1-60 cycles/arcdegree

Approach for reducing visual redundancy in lossy compression 1. Transform: Convert the data to the frequency domain 2. Quantize: Under-represent the high frequencies 3. Losslessly compress the quantized data Audio compression
Audio compression is a form of data compression designed to reduce the transmission bandwidth requirement of digital audio streams and the storage size of audio files. Audio compression algorithms are implemented in computer software as audio codecs. Generic data compression algorithms perform poorly with audio data, seldom reducing data size much below 87% from the original, [citation needed] and are not designed for use in real time applications. Consequently, specifically optimized audio lossless and lossy algorithms have been created. Lossy algorithms provide greater compression rates and are used in mainstream consumer audio devices.In both lossy and lossless compression, information redundancy is reduced, using methods such as coding, pattern recognition and linear prediction to reduce the amount of information used to represent the uncompressed data.

Audio compression can mean two things: 1. Audio data compression - in which the amount of data in a recorded waveform is reduced for transmission. This is used in CD and MP3 encoding, internet radio, and the like. 2. Audio level compression - in which the dynamic range (difference between loud and quiet) of an audio waveform is reduced. This is used in guitar effects racks, recording studios, etc. Audio compression consist of analog signal of different frequencies The audio signals are converted to digital form and then processed,stored or transmitted Compression allows storing the audio in significantly less space and if there is need to transmit the signal ADAPTIVE DIFFERENTIAL PULSE CODE MODULATION ADPCM provides a form of compression by encoding and storing in the data stream only the difference between the values of successive samples. This provides a wide dynamic range while reducing the size of the data stream.

 For instance,normal speed has long sections of silence that can be compressed because there are no changes across a number of samples. Audio compression is a form of data compression designed to reduce the size of audio files. Audio compression algorithms are implemented in computer software as audio codecs. Generic data compression algorithms work with audio data but the compression ratios are low (around 50-60% of original size) and they do not work in real time and are therefore not practical. Specific audio "lossless" and "lossy" algorithms have been created. Lossy algorithms provide far greater compression ratios and are used in mainstream consumer audio devices. As with image compression, both lossy and lossless compression algorithms are used in audio compression, lossy being the most practical for everyday use. In both lossy and lossless compression, information redundancy is reduced, using methods such as coding, pattern recognition and linear prediction to reduce the amount of information used to describe the data. For example, suppose you wanted to record twenty house numbers along one side of a street, each of which goes up by 2. If the first address was 14461, or five digits, the uncompressed stream would require 20 times 5 bytes, or 100 bytes, to store. You could recode that to take advantage of the repetition and simply say begin at 14461, increase by 2, repeat 19 times. Now the data are losslessly captured in a smaller space. In practice the pattern recognition for lossy and lossless compression is far more complex than that. The trade-off of slightly reduced audio quality is clearly outweighed for most practical audio applications where users cannot perceive any difference and space requirements are substantially reduced. For example, on one CD, one can fit an hour of high fidelity music, less than 2 hours of music compressed losslessly, or 7 hours of music compressed in MP3 format. Lossless audio codecs have no quality issues, so the usability can be estimated by Speed of compression and decompression Degree of compression Software and hardware support Robustness and error correction

Lossy audio compression is used in an extremely wide range of applications. In addition to the direct applications (mp3 players or computers), digitally compressed audio streams are used in most video DVDs; digital television; streaming media on the internet; satellite and 6

cable radio; and increasingly in terrestrial radio broadcasts. Lossy compression typically achieves far greater compression than lossless compression (data of 5 percent to 20 percent of the original stream, rather than 50 percent to 60 percent), by discarding less-critical data. The innovation of lossy audio compression was to use psychoacoustics to recognize that not all data in an audio stream can be perceived by the human auditory system. Most lossy compression reduces perceptual redundancy by first identifying sounds which are considered perceptually irrelevant, that is, sounds that are very hard to hear. Typical examples include high frequencies, or sounds that occur at the same time as other louder sounds. Those sounds are coded with decreased accuracy or not coded at all.

References 1) http://en.wikipedia.org/wiki/Audio_compression_(data) 2) http://www.bdti.com/MyBDTI/pubs/audio_icspat00.pdf 3) www.monkeysaudio.com/ 4) http://computing.unn.ac.uk/staff/cgpv1/cm613/lecture_material.htm 5) http://computing.unn.ac.uk/staff/cgpv1/cm613/files/Audio%20Compression.pdf 6) http://computing.unn.ac.uk/staff/cgpv1/cm613/files/mpegaud.pdf 7) http://www.bdti.com/MyBDTI/pubs/audio_icspat00.pdf