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Chapter 4 Audio and video compression

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.

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Chapter 4 Audio and video compression

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  1. Chapter 4 Audio and video compression • 4.1 Introduction • 4.2 audio compression • 4.3 Video compression

  2. 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

  3. 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. 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

  5. 4.2.1 Differential pulse code modulation –cont (figure 4.1)

  6. 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

  7. 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

  8. 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

  9. 4.2.4 Linear predictive coding -cont (figure 4.4)

  10. 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

  11. 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

  12. 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

  13. 4.2.6 Perceptual coding -cont (figure4.5)

  14. 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

  15. 4.2.6 Perceptual coding-cont (figure4.6)

  16. 4.2.6 Perceptual coding-cont (figure4.7)

  17. 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

  18. 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

  19. 4.2.7 MPEG AUDIO CODERS -cont (figure4.8) • Frame format,show figure 4.8(b)

  20. 4.2.7 MPEG AUDIO CODERS -cont table 4.2

  21. 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

  22. 4.2.8 Dolby audio coders -cont (figure4.9)

  23. 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)

  24. 4.3 Video compression • The digitization format defines the sampling rate that is used for the luminance ,Y ,and two chrominance,Cb and Cr

  25. 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

  26. 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

  27. 4.3.1 video compress principles –cont (figure4.11)

  28. 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

  29. 4.3.1 video compress principles –cont (figure4.12)

  30. 4.3.1 video compress principles –cont (figure4.13)

  31. 4.3.1 video compress principles –cont (figure4.14) • Implementation issues ,show figure4.14

  32. 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

  33. 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

  34. 4.3.2 H.261 -cont

  35. 4.3.2 H.261 -cont • H.261 video encoder principles figure 4.16(a)

  36. 4.3.2 H.261 -cont • Two threshold • Low • high

  37. 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

  38. 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

  39. 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

  40. 4.3.3 H.263 -cont (figure 4.17)

  41. 4.3.3 H.263 -cont (figure 4.18)

  42. 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)

  43. 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

  44. 4.3.4 MPEG -cont • MPEG-4 • Similar h.163 • Low bit rate range from 4.8 to 64kbps • Interactive multimedia application

  45. 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)

  46. 4.3.5 MPEG-1 -cont (figure 4.20) • Figure 4.20

  47. 4.3.5 MPEG-1 -cont (figure 4.21)

  48. 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))

  49. 4.3.6 MPEG-2 -cont (figure4.22)

  50. 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

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