3. 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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5. 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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5
6. How PCM works ?
• Sampling
• Quantizing
• Encoding
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Figure 01 : Block Diagram of Pulse Code Modulation
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7. 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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8. 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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9. 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 Quantization
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10. 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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12. 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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12
13. 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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14. 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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15. 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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15
16. 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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17. 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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18. 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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19. 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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20. 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.
20
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22. MPEG Encoder Architecture
Audio
Input Encoded
Bit Stream
Time to
Frequency
Mapping
Filter bank
Bit Allocation,
Quantize and
coding
Psychoacoustic
Model
Bit stream
formatting
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23. 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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24. 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.
24
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25. 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.
25
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26. 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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27. 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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28. 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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29. 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).
29
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30. MPEG Audio Layer III Filter Bank
Processing
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31. 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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32. MPEG Layers Applications
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Layer Application
I Digital audio cassette
II Digital audio
broadcast
III CD quality
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34. 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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35. Summary
35
• 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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36. References
36
• 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/introd
uctiontodigitalaudiocompression.pdf
• Davis Yen Pen, Digital Audio Compression,
Available:
www.ece.jcu.edu.au/subjects/ee3700/resources/
Digital_Audio_Compression.pdf
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