Unit 2 - Audio and Video Compression
Unit 2 - Audio and Video Compression
Unit 2 - Audio and 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
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
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
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
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
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:
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
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
The Sensitivity of the ear varies with the frequency of the signal,the perception threshold of the ear that is, its minimum level of sensitivityas 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
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
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
The 12 set of 32 PCM samples are first transformed into an equivalent set of frequency components using a mathematical technique Using the known hearing thresholds and masking properties of each subband,the model determines the various masking effects of this set of signals
Signal-to-mask ratios(SMRs)
32192kbps
40ms
64kbps
60ms
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
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 AC3 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)
The digitization format defines the sampling rate that is used for the luminance ,Y ,and two chrominance,Cb and Cr
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
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
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
4.3.3 H.263
Over wireless and public switched telephone networks(PSTN) Include video telephony videoconferencing , security surveillance ,interactive game Low bit rates Digitization formats
Frame types:
I-frame P-frame B-frame PB-frame:because of the much reduced encoding overhead To overcome this limitation ,for those pixels of a potential close-match macroblock that fall outsize of the frame boundary
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 Prevent these errors from affecting neighboring GOBs in succeeding frames Show figure 4.18
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
Four level
MPEG-2
LOW MAIN
High 1440
high
MPEG-4
Similar h.163 Low bit rate range from 4.8 to 64kbps Interactive multimedia application
4.3.5 MPEG-1
NTSC PAL
Frame type:I,P,B-frame,(figure 4.20) Based on the h.261,there are two main differences:
Figure 4.20
4.3.6 MPEG-2
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))
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)
4.3.7 MPEG-4
-cont (figure4.23)
4.3.7 MPEG-4
-cont (figure4.24)
4.3.7 MPEG-4
-cont
4.3.7 MPEG-4
-cont (figure4.25)
4.3.7 MPEG-4
-cont
4.3.7 MPEG-4
-cont (figure4.26)
4.3.7 MPEG-4
-cont
4.3.7 MPEG-4
-cont (figure4.27)