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Compression and Decompression for Multimedia System.

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Compression and Decompression for Multimedia System

Voice mail

Voice mail systems use centralized equipment to record, store, and play back messages. Each user has access to an individual mailbox, which allows messages to be kept private. Voice-mail messaging is available to students, faculty members, and administrators. This service allows users to receive personal messages when they are away from their phones or while their phones are in use.

Compared to written notes, voice mail allows longer and more complex messages to be accurately communicated. The ability to leave detailed, private messages frequently means callers can relay information without the need for a return call. This is especially useful when you consider that half of all calls are for one-way transfers of information.

Review of compression techniques in existing voice mail system

By choosing an appropriate voice mail compression technique, you can greatly reduce transmitted data rate and the bandwidth requirements for digitally encoded voice signals.

For voice compression, it can be divided into two methods; one is through the frequency domain and the other through the time domain. The type of voice compression techniques used depends on the functionality required and affects the quality of the output. 

Compression for both speech and data is done through the removal of redundancy. In the case of speech compression, further compression can be obtained by removing irrelevancy, which is the imperceptible reconstruction error or distortion to the voice signal.

For medium rate speech coding, an analysis-synthesis method is normally used. This means the speech is represented by a set of compact parameters which are then coded. Prior to compression, the speech signal goes through an analysis stage which can either be closed-looped or open-looped.

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The decoder function is to reconstruct the signal from the quantized bit stream

Although the overall compression ratio of ADPCM is only 2:1 at 32 Kbits/s, it can be used in conjunction with DSI (Digital Speech Interpolation) to achieve compression ratio of 4:1. [1]

Sub-band Coders

In a sub-band coder, the signal is passed through a series of band-pass filters to separate the frequency bands. The output of each band is then sampled and encoded. The series of encoded data is sent to the through a multiplexer. At the receiver end the signal is demultiplexed, decoded and demodulated. The encoding process introduces quantization noises. The demodulation process introduces aliasing distortion due to the overlapping nature of the sub-bands.

Compression is achieved by allocating different bits for different sub-bands during the encoding process. The allocation of bits is done by exploiting the voice characteristics of speech. Low frequency sub-bands are encoded using more bits to preserve the critical pitch and formant information. Higher frequency sub-bands are encoded using fewer bits since high frequency losses are less perceptible.

Filter design is based on the type of signal which is to be processed. For speech, more emphasis is placed on low frequency bands to resolve the signal more accurately. Quadrature-Mirror Filter (QMF) can be used to achieve perfect reconstruction when there are no quantization noises. A brief description on AT&T sub-band coder and ITU G.722 is given below. [6]


ATELP Compression was developed by Soft Sound with the goal to provide near-CD quality at meaningful compression rates.   ATELP uses a modeling technique to discard portions of the audio data. Soft Sound claims ATELP to be superior to any MPEG 2 software implementation at any given compression rate.   ATELP yields a compression ratio of 12:1, 20:1, or 30:1. [4]


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Using MPEG audio, one may achieve a typical data reduction of still maintaining the original CD sound quality.


by Layer 1 (corresponds to 384 kbps for a stereo signal),


by Layer 2 (corresponds to 256..192 kbps for a stereo signal),


by Layer 3 (corresponds to 128..112 kbps for a stereo signal),

By exploiting stereo effects and by limiting the audio bandwidth, the coding schemes may achieve an acceptable sound quality at even lower bitrates. MPEG Layer-3 is the most powerful member of the MPEG audio coding family. For a given sound quality level, it requires the lowest bit rate - or for a given bit rate, it achieves the highest sound quality.

  1. Sound Quality

Some typical performance data of MPEG Layer-3 are:

sound quality



bit rate

reduction ratio

telephone sound

2.5 kHz


8 kbps *


better than short wave

4.5 kHz


16 kbps


better than AM radio

7.5 kHz


32 kbps


similar to FM radio

11 kHz


56...64 kbps



15 kHz


96 kbps



>15 kHz




*) Fraunhofer IIS uses a non-ISO extension of MPEG Layer-3 for enhanced performance ("MPEG 2.5")

In all international listening tests, MPEG Layer-3 impressively proved its superior performance, maintaining the original sound quality at a data reduction of 1:12 (around 64 kbit/s per audio channel). If applications may tolerate a limited bandwidth of around 10 kHz, a reasonable sound quality for stereo signals can be achieved even at a reduction of 1:24.

For the use of low bit-rate audio coding schemes in broadcast applications at bitrates of 60 kbit/s per audio channel, the ITU-R recommends MPEG Layer-3. (ITU-R doc. BS.1115)


[1] Kishan Shenoi. Digital Signal Processing in Telecommunication. Prentice Hall PTR 1995

[2] Michael P. Chin, Karl Home. Incorporating Voice Telecommunication with VSATS.Tenth International Conference on Digital Satellite Communication. IEE London UK 1994

[3] Esin Darici Haritaoglu. Wideband Speech and Audio Coding

Date accessed: 10/8/98

[4] http://www.softsound.com

[5] http://www.reed-electronics.com/ednmag/archives/1994/101394/21df4.htm

[6] http://sky.fit.qut.edu.au/~rolf/itn530/ass982/ang/voice.htm

[7] http://www.audioboxinc.com/compression.html

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