Audio compression

Audio Compression

M.K.H.Gunasekara – AS2010377J.A.S. Fernando – AS2010362

CSC 362 1.5 Seminar IDepartment of Statistics and Computer Science

Overview

• Introduction• Pulse Code Modulation• µ - law• A - law• Differential Pulse Code Modulation• Adaptive Differential Pulse Code Modulation• Application• MPEG Audio Compression• MPEG Encoder Architecture• MPEG Audio Compression Algorithm• MPEG Layers• MPEG Layers Applications• Comparison of MPEG audio compression standards• Summary• References

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Introduction

• A process that is useful for conserving both transmission bandwidth and storage space.

• The various audio compression techniques offer different levels of complexity, compressed audio quality, and amount

of data compression.

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Audio Compression

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Lossy Audio Compression• AAC• ADPCM• ATRAC• Dolby AC-3• MP2• MP3

Lossless Audio Compression• Apple Lossless • apt-X Lossless• Audio Lossless Coding • Direct Stream Transfer (DST)• Dolby TrueHD• DTS-HD Master Audio• Free Lossless Audio Codec

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What is PCM ?

• PCM is a method used to digitally represent sampled analog signals.

• It is the standard form of digital audio in computers, Compact Discs, digital telephony and other digital audio applications.

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How PCM works ?

• Sampling

• Quantizing

• Encoding

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Figure 01 : Block Diagram of Pulse Code Modulation

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Sampling

• Sampling theory (Nyquist)– If input signal has maximum frequency (bandwidth) f,

sampling frequency must be at least 2f

– With a low-pass filter to interpolate between samples, the input signal can be fully reconstructed

Example : highest frequency of telephone voice channel is 3.4 kHz

Sampling rate ≥ 2 x 3.4

≥ 6.8 kHz

Hence sample rate is 8 kHz

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Sampling

• 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

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Figure 02 : Sampling Methods

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Quantization

• The samples are divided into many discrete levels. Then each sample is numbered according to their corresponding level.

• Quantization error - the coded signal is an approximation of the actual amplitude value.

Figure 03 : Linear Quantization Figure 04 : Non Linear QuantizationAudio Compression - AS2010377 9

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Encoding

• After quantizing the corresponding level it is to be represented in some manner.

• We represent levels in binary format.

• Representation of level 50 = 110010

Figure 05 : Representation of sample signal

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Figure 07- Circuit diagram of pulse code modulation

Figure 06- Block diagram of pulse code modulation

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What is µ law ?

• µ -law encoding is a form of logarithmic quantization or companding.

• It's based on the observation that many signals are statistically more likely to be near a low signal level than a high signal level.

• Most quality codecs (including the Sparcstation's audio codec) use µ -law encoded samples.

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What is A law ?

• A-Law compression is extremely similar to Mu-Law compression

• If the absolute value of the source sample is less than 256, the 16-bit sample is simply shifted down 4 bits and converted to an 8-bit value, thus losing the top 4 bits in the process.

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A law and µ law

Both methods compress 2:1 ratio.

• µ -Law has a larger dynamic range compared to A-law

• µ -Law has worse distortion with small signals compared to A-law

• µ -Law is used in North-America and Japan while A-law is commonly used in Europe• A-law takes precedence over µ -law with international calls

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Differential PCM

• Differential pulse-code modulation is a signal encoder that uses the baseline of pulse-code modulation but adds some functionalities based on the prediction of the samples of the signal.

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Adaptive Differential PCM

• ADPCM, commonly termed as a form of compression, is a more efficient way of storing waveforms than 16-bit or 8-bit PCM.

• ADPCM stores the value differences between two adjacent PCM samples and makes some assumptions that allow data reduction.

• Because of these assumptions low frequencies are properly reproduced

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Adaptive Differential PCM

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• An ADPCM algorithm is used to map a series of 8 bit µ-law (or a-law) PCM samples into a series of 4 bit ADPCM samples. In this way, the capacity of the line is doubled.

• Some ADPCM techniques are used in Voice over IP communications.

• ADPCM was also used by Interactive Multimedia Association for development of legacy audio codec known as ADPCM DVI, IMA ADPCM or DVI4

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Applications

Audio Compression

Format

Algorithm Bit Rate Bits per sample Implementations

G.711 Companding A-law ; µ-law ; PCM

64 kbit/s 8 bit FFmpeg, Ekiga, Asterisk (PBX)

G.711.1 A-law ; µ-law 64, 80, 96 kbit/s 16 bit VoIP

G.721 ADPCM 32 kbit/s 13 bit

G.722 ADPCM 64 kbit/s 14 bit QuickTime, RealPlayer

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Wrap - up

• Techniques to compress general digital audio signals include µ -law and adaptive differential pulse code modulation.

• These approaches apply low-complexity, low-compression, and medium audio quality algorithms to audio signals.

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MPEG Audio Compression

• Motion Picture Experts Group

• An ISO standard for high-fidelity audio compression.

• A lossy compression.

• But can achieve transparent, perceptually lossless compression.

• Compresses without regard to the source of the audio data.

• Removes distortion and other features that are imperceptible to the human ear.

• 6-to-1 compression ratio.

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MPEG Audio Compression

• Four channel modes

- Monophonic

- Dual monophonic

- Stereo

- Joint- stereo

• Has 3 distinct layers for compression

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MPEG Encoder Architecture

Audio

Input Encoded

Bit Stream

Time to Frequency Mapping

Filter bank

Bit Allocation, Quantize and

coding

PsychoacousticModel

Bit stream formatting

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MPEG Audio Compression Algorithm

• The input audio stream passes through a filter bank.

Filter Bank

- Divides the input into multiple subbands.

- Relatively simple and provide good time

resolution

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MPEG Audio Compression Algorithm …

• The input audio stream simultaneously passes

through a psychoacoustic model.

The Psychoacoustic Model

- The key component of the MPEG encoder

that enables its high performance.

- Analyzes the audio signal and computes the

amount of noise masking that is available as a

function of frequency.

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MPEG Audio Compression Algorithm...

• The bit or noise allocation block uses the signal-to-mask ratios to decide how to apportion the total number of code bits available for the quantization of the subband signals to minimize the audibility of the quantization noise.

• Finally, the last block takes the representation of the quantized audio samples and formats the data into a decodable bit stream.

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MPEG Layers

• Offers three compatible layers.

- Layer I

- Layer II

- Layer III

• Each succeeding layer able to understand the lower layers.

• Each succeeding layer offers more complexity

and better compression for a given level of audio quality.

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MPEG Audio Layer I

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• Uses the basic filter bank found in all layers.

• This filter bank divides the audio signal into 32

constant-width frequency bands.

• Bit rates greater than 128 kbits/s per channel.

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MPEG Audio Layer II

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• A simple enhancement of Layer I.

• It improves compression performance by

coding data in larger groups.

• Bit rates of 128 kbits/s per channel.

• Improves performance by representing the bit

allocation, the scale factor values, and the

quantized samples.

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MPEG Audio Layer III

– Widely popular under the name Mp3

– The most complex with the best audio quality

– Used for bit rates of 64 kbits/s per channel.

• Compensates for some filter bank deficiencies

by

processing the filter outputs with a modified

discrete cosine transform (MDCT).

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MPEG Audio Layer III Filter Bank Processing

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MPEG Audio Layer III enhancements

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• Alias reduction

• Nonuniform quantization

• Entropy coding of data values

• Use of a bit reservoir

• Noise allocation instead of bit allocation

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MPEG Layers Applications

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Layer Application

I Digital audio cassette

II Digital audio broadcast

III CD quality

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Comparison of MPEG audio compression standards

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Standard Audio Sampling rate(kHz)

Compressed bit rate (kbits/sec)

Channels

MPEG - 1 Layer I 32, 44.1, 48 32- 448 1- 2 channels

MPEG - 1 Layer I 32, 44.1, 48 32 – 384 1-2 channels

MPEG - 1 Layer III 32, 44.1, 48 32 – 320 1-2 channels

MPEG - 2 Layer I 32, 44.1, 48 32 - 448 1 – 5.1 channels

16, 22.05, 24 32 - 256

MPEG - 2 Layer II 32, 44.1, 48 32 - 384 1 – 5.1 channels

16, 22.05, 24 8 - 160

MPEG - 2 Layer III 32, 44.1, 48 32 - 384 1 – 5.1 channels

16, 22.05, 24 8 - 160

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Comparison of MPEG audio compression standards

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• MPEG 2 AAC

- more efficient than the previous MPEG audio standards

- supports from 1 to 48 channels at sampling rates of 8 to 96 kHz

• MPEG 4 AAC

- similar to the MPEG-2 AAC standard, with some minor changes

- supports the compression of natural audio at bit rates ranging from 2 up to 64 kb/s.

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Summary

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• The MPEG audio standard has three layers of successive complexity for improve compression performance.

• MPEG audio has been extended in following ways

- Multichannel audio support

- Lower, compressed audio rates

- Lower audio sampling rates

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References

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• Introduction To Data Compression ,Third Edition ,Khalid Sayood

• B. Cavagndo , J. Bier, Introduction to Digital Audio Compression,Available: http://rizal.blog.undip.ac.id/files/2009/07/introductiontodigitalaudiocompression.pdf

• Davis Yen Pen, Digital Audio Compression, Available: www.ece.jcu.edu.au/subjects/ee3700/resources/Digital_Audio_Compression.pdf

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THANK YOU

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