Unit 2 - Audio and Video Compression Ppt

Chapter 4 Audio and video compression

4.1 Introduction 4.2 audio compression 4.3 Video compression

4.1 introduction

Both audio and most video signals are continuously varying analog signals

The compression algorithms associated with digitized audio and video are different from close

4.2 Audio compress Pulse code modulation(PCM) Bandlimited signal The bandwidth of the communication

channels that are available dictate rates that are less than these.This can be achieved in one of two ways: Audio signal is sampled at a lower

rate A compression algorithm is used

4.2.1 Differential pulse code modulation

DPCM is a derivative of standard PCM and exploits the fact that,for most audio signals, the range of the differences in amplitude between successive samples of the audio waveform is less than the range of the actual sample amplitudes.

Figure4.1

4.2.1 Differential pulse code modulation –cont (figure 4.1)

4.2.2 Adaptive differential PCM

Additional savings in bandwidth –or improved quality –can be obtained by varying the number of bits used for the difference signal depending on its amplitude

A second ADPCM standard ,which is G.722.It added subband coding.

A third standard based on ADPCM is also available.this is defined in G.726.This also uses subband coding but with a speech bandwidth of 3.4kHz

4.2.3 Adaptive Predictive Coding(APC)

Even higher levels of compression-but at higher levvels of complexity-can be obtained by also making the predictor coefficients adaptive.This is the principle of adaptive of adaptive predictive coding

4.2.4 Linear predictive coding

There are then quantizized and sent and the destination uses them,together with a sound synthesizer,to regenerate a sound that is perceptually comparable with the source audio signal.this is LPC technique.

Three feature which determine the perception of a signal by the ear are its: Pitch Period Loudness

Basic feature of an LPC encoder/decoder: figure 4.4

4.2.4 Linear predictive coding -cont (figure 4.4)

4.2.5 Code-excited LPC

Code-excited LPC The synthesizers used in most LPC decoders

are based on a very basic model of the vocal tract

In the CELP model,instead of treating each digitized segment independently for encoding purpose

All coders of this type have a delay associated with them which is incurred while each block of digitized samples is analyzed by the encoder and the speech is reconstructed at the decoder

4.2.6 Perceptual coding

Perceptual encoders have been designed for the compression of general audio

Perceptual coding since its role is to exploit a number of the limitation of the human ear.

Sensitivity of the ear A strong signal may reduce the level of

sensitivity of the ear to other signals which are near to it in frequency

4.2.6 Perceptual coding -cont

The Sensitivity of the ear varies with the frequency of the signal,the perception threshold of the ear – that is, its minimum level of sensitivity-as a function of frequency is show in figure 4.5(a)

Most sensitive to signals in the range 2-5kHz

Shown 4.5(b) shows how the the sensitivity of the ear changes in the vicinity of a loud signal

4.2.6 Perceptual coding -cont (figure4.5)

4.2.6 Perceptual coding -cont

The masking effect also varies with frequency as show in figure 4.6

Critical bandwidth Temporal masking:

When the ear hears a loud sound,it takes a short but finite time before it can hear a quieter sound

SHOW 4.7

4.2.6 Perceptual coding-cont (figure4.6)

4.2.6 Perceptual coding-cont (figure4.7)

4.2.7 MPEG AUDIO CODERS

ENCODING Input signal is first sampled and

quantized using PCM The bandwidth that is available for

transmission is divided into a number of frequency subbands using a bank of analysis filters

Scaling factor: THE analysis filter band also determines

the maximum amplitude of the 12 subband samples in each subband

4.2.7 MPEG AUDIO CODERS -cont

Discrete Fourier transform(DFT) The 12 set of 32 PCM samples are first

transformed into an equivalent set of frequency components using a mathematical technique

Signal-to-mask ratios(SMRs) Using the known hearing thresholds and

masking properties of each subband,the model determines the various masking effects of this set of signals

4.2.7 MPEG AUDIO CODERS -cont (figure4.8)

Frame format,show figure 4.8(b)

4.2.7 MPEG AUDIO CODERS -cont table 4.2

1

Layer ApllicationCompressed

bit rate QualityExample

input-to-output delay

2

3

Digital audio cassette

Digital audio and digital video broadcasting

CD-quality

32-448kbps

32-192kbps

64kbps

Hi-fi quality at 192 kbps per channel

Near CD-quality at 128 kbps per

channel

CD-quality of 64kbps per

channel

20ms

40ms

60ms

Table 4.2 Summary of MPEG layer1,2 and 3 perceptual encoders

4.2.8 Dolby audio coders

MPEG V.S Dolby AC-1 ,show figure 4.9 MPEG:

Advantage: psychoacoustic model is required only in the encoder

Disadvantage:a significant portion of each encoded frame contains bit allocation information

Dolby AC-1: Use a fixed bit allocation strategy for

each subband which is then used by both the encoder and decoder

4.2.8 Dolby audio coders -cont (figure4.9)

4.2.8 Dolby audio coders -cont

Dolby AC-2 standard which is utilized in many applications including the compression associated with the audio of a number of PC sound cards

The hybrid approach is used in the Dolby AC-3 standard which has been defined for use in a similar range of applications as the MPEG audio standards including the audio associated with advanced television(ATV)

4.3 Video compression

The digitization format defines the sampling rate that is used for the luminance ,Y ,and two chrominance,Cb and Cr

4.3.1 video compress principles

Frame type I-frame:

I-frames are encoded without reference to any other frames

GOP:The number of frame between I-frames

P-frame: encoding of a p-frame is relative to the

contents of either a preceding I-frame or a preceding P-frame

4.3.1 video compress principles -cont

The number of P-frames between I-frame is limited since any errors present in the first P-frame will be propagated to the next

B-frame:their contents are predicted using search regions in both past and future frames

PB-frame:this does not refer to a new frame type as such but rather the way two neighboring P- and B-frame are encoded as if they were a single frame

D-frame:only used in a specific type of application. It has been defined for use in movie/video-on-demand application

4.3.1 video compress principles –cont (figure4.11)

4.3.1 video compress principles -cont

Motion estimation and compensation P-frame Macroblock structure ,show figure

4.12(a) P-frame Encoding procedure,show figure

4.12(b) Best match macroblock Motion vector DCT+ Quantization +run-length & V Huffman

B-frame encoding procedure,show figure 4.13


Implementation issues ,show figure4.14

4.3.1 video compress principles –cont

Performance - Compression ratio I-frame:10:1 – 20:1 P-frame:20:1-30:1 B-frame:30:1-50:1

4.3.2 H.261

For the provision of video telephony and videoconferencing services over an ISDN

Transmission channels multiples of 64kbps Digitization format used is either the

common intermediate format(CIF) or the quarter CIF(QCIF) CIF:Y=352X288, Cb=Cr=176X144 QCIF:Y=176X144, Cb=Cr=88X72

H.261 encoding format show figure 4.15

4.3.2 H.261 -cont

4.3.2 H.261 -cont

H.261 video encoder principles figure 4.16(a)

4.3.2 H.261 -cont

Two threshold Low high

4.3.3 H.263

Over wireless and public switched telephone networks(PSTN)

Include video telephony videoconferencing , security surveillance ,interactive game

Low bit rates Digitization formats

QCIF:Y=176X144 , Cb=Cr=88X72 S-QCIF:Y=128X96, Cb=Cr=64X68

4.3.3 H.263 -cont

Frame types: I-frame P-frame B-frame PB-frame:because of the much reduced

encoding overhead Unrestricted motion vectors

To overcome this limitation ,for those pixels of a potential close-match macroblock that fall outsize of the frame boundary

4.3.3 H.263 -cont

Error resilience Cause error propagation,show figure4.17(a) Error tracking and resilience,show

figure4.17(b) When an error is detected , decoder send

NAK to encoder Independent segment decoding

Prevent these errors from affecting neighboring GOBs in succeeding frames

Show figure 4.18

4.3.3 H.263 -cont (figure 4.17)

4.3.3 H.263 -cont (figure 4.18)

4.3.3 H.263 -cont (figure 4.19)

Reference picture selection(figure 4.19 ) NAK mode ,show figure 4.19(a) ACK mode,show figure 4.19(b)

4.3.4 MPEG

MPEG-1 Source intermediate digitization format(SIF) Resolution:352X288 VHS-quality audio Video on CD-ROM at bit rates up to 1.5Mbps

MPEG-2 Four level

LOW MAIN High 1440 high

4.3.4 MPEG -cont

MPEG-4 Similar h.163 Low bit rate range from 4.8 to 64kbps Interactive multimedia application

4.3.5 MPEG-1

Support two type spatial resolutions NTSC PAL

Frame type:I,P,B-frame,(figure 4.20) Based on the h.261,there are two main

differences: Temporal B-frame was increased

Video bitstream structure (figure 4.21)

4.3.5 MPEG-1 -cont (figure 4.20)

Figure 4.20

4.3.5 MPEG-1 -cont (figure 4.21)

4.3.6 MPEG-2

Support four levels and five profiles MP@ML

For digital television broadcasting Resolution of either 720X480 pixels at 30Hz

or 720X576 pixels at 25Hz Bit rate from 4Mbps – 15Mbps Use interlaced scanning,show 4.22(a) Field mode(figure 4.22(b)) Frame mode(figure 4.22(c))

4.3.6 MPEG-2 -cont (figure4.22)

4.3.6 MPEG-2 -cont

HDTV(Grand Alliance) ITU-R HDTV

16/9 ASPECT RATIO MP@HL Audio: Dolby AC-3

DVB HDTV 4/3 ASPECT RATIO SSP@H1440-SPATIALLY-SCALEABLE

PROFILE AT HIGH 1440 MPEG audio layer 2

4.3.7 MPEG-4

Scene composition Content-based functionalities Audio-visual object(AVOs) Object descriptor Binary format for scenes Scene descriptor Video object planes(VOPs)(figure 4.23)

Audio and video compression(figure 4.24)

4.3.7 MPEG-4 -cont

Transmission format(figure 4.25) Transport stream Packetized elementary Elementary stream(ES) FlexMux layer Synchronization layer Elementary stream descriptor(ESD) Composition and rendering block

4.3.7 MPEG-4 -cont

Error resilience techniques (figure 4.26) Use of fixed-length Based on reversible VLCs Error occur

macroblock header

4.3.7 MPEG-4 -cont

Reversible VLCs (figure 4.27) The associated set of RVLCs is then

produced by adding a fixed—length prefix and suffix to each of the corresponding VLCs

Forward direction scan Reverse direction scan The error at difference points in the

bitstream resulting in an overlap region

Unit 2 - Audio and Video Compression Ppt

Documents

Transcript of Unit 2 - Audio and Video Compression Ppt