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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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 QuantizationAudio Compression - AS2010377 9
9
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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14
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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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
17
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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18
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.
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21. MPEG Audio Compression
• Four channel modes
- Monophonic
- Dual monophonic
- Stereo
- Joint- stereo
• Has 3 distinct layers for compression
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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.
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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.
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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).
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30. MPEG Audio Layer III Filter Bank
Processing
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31. MPEG Audio Layer III
enhancements
31
• 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
32
Layer Application
I Digital audio cassette
II Digital audio
broadcast
III CD quality
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34. Comparison of MPEG audio
compression standards
34
• 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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