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Video Transmission System. Heejune AHN Embedded Communications Laboratory Seoul National Univ. of Technology Fall 200 8 Last updated 2008. 11. 23. Agenda . QoS Requirement and Constraints Impacts on Video Coding Representative Video Transmission Systems MPEG-2 PS & TS System
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Video Transmission System Heejune AHN Embedded Communications Laboratory Seoul National Univ. of Technology Fall 2008 Last updated 2008. 11. 23
Agenda • QoS Requirement and Constraints • Impacts on Video Coding • Representative Video Transmission Systems • MPEG-2 PS & TS System • Internet-based Video Transmission System
1. Transmission & Video coding • Transmission constraints • Impact on the video coding system: cross-layer concepts • e.g. rate-control, un-equal FEC, scalable coding etc • QoS & QoE • QoS (Quality of Service) • network layer performance measures • Bit-rate (mean, variation), delay (mean, jitter), loss (bit, packet) • QoE (Quality of Experience) • Application (user) layer performance measures network/storage Bit-rate Input Video Input Video decoder encoder Delay Loss
QoS: Data rate • Offered • Circuit Switched (constant) • 384kbps (WCDMA) • DMB, DTV • Packet-switched (variable) • Internet • LAN • ADSL connected to Internet • Required • Mean Rate • Required Quality • TV (2~5Mbps) • Rate-Variation • Scene complexity (inherent) • Coding scheme
QoS: Error (distortion) • Offered • Circuit switched • low random bit error • very seldom burst errors • Wired Packet-switched • almost no bit error • low packet loss • some burst packet loss • Due to network Congestion • Wireless packet-switched • Some bit loss • Some packet loss • Some burst packet loss • Due to channel fading • Required • Low transmission distortion
QoS: Delay • Offered • Circuit Switched • usu. low transmission delay • Not in sattlelite • Packet-switched • Variable delay • Due to congestion and re-routing in Internet • Due to ARQ in wireless comm. • Required • One-way application • Constant delay & low variation • e.g. DTV • Two-way application • Low delay & low variation • e.g. Videophone (< 400ms) • Interactive • Low delay and low feedback delay • e.g. VoD TV
2. Impact of trans. on Video coding • Data Rate • Rate control & output buffering • constant bit-rate for circuit-switching network • Smoothed bit-rate for packet switching network • High activity scene has lower quality • Error • Error propagation • VLC error • Bit error => the corresponding VLC decoding error • Spatial error propagation • VLC error => the successive VLC decoding error • Resync. marker for every slices, piictures, GOPs. • Temporal error propagation • Wrong motion-compensation of Blocks in successive frame
Error-Concealment • Use Reversible VLC • Use Spatial domain smoothing • POCS (projection onto convex set) • Use temporal domain: MV estimation • Use zero vector from previous frame • Use median vector • Re-estimation using boundary pixels
Feedback based control • Due to low delay requirement • Error tracking & intra-coding (H.263 Annex N) • Reference picture selection (H.263 Annex U) intracoding
Delay • Delay components • Capture delay • one frame, but can reduce it • Encoder delay • Depends on encoder performance, less than one frame • B picture introduces extra delay • Output buffer • Depends on smoothing and rate control • Max. out buffer delay = buffer size / tx rate • Network delay • Depends on network types and network conditions • Input buffer • Decoding delay • Display buffer • Low delay case • Select appropriate network and QoS negotiation • no B picture, low output buffer
3. Video Transport Systems • MPEG-2 system • Provides Multiplexing and synchronization mechanism • MPEG-2 = System + Video + Audio • Build PS & TS from ES • Application environment • Fixed, guaranteed bit-rate, predictable delay and predictable errors • Used in Digital Cable TV, T/S-DTV, DVD etc • Internet Multimedia transport System • H.324-based system • ITU-T’s Internet conference system • Used in Serom’s Dialpad etc • RTSP (Real-time Streaming protocol) • VCR remocon DESC, SET-UP, PLAY, STOP, TEAR-DOWN • SIP • simple Session Initiation Protocol, current VoIP • RTP • transport protocol for multimedia data
PS stream • ES (elementary steam) : video bytes, audio bytes streams • PES (packetized ES) : timestamped ES packet • Program Stream = MPEG-1 system • One program (video, audio, etc), no loss assumption • Variable and long Packet (called PACK) • Pack header includes “SCR (system clock reference) • PS stream encoded audio encoded audio PES-1 PES-1 PACK PACK encoded video encoded video PES-2 PES-2 DVD PES-3 PES-3 etc etc
RS, Conv PES-1 Program (KBS) PES-2 TS tx Program (MBC) PES-3 Program map • TS (Transport stream) • Time stamp and clock info is supported • Multiple program is muxed • Fixed size (188 Byte) TS packet • inner coding (Reed-soloman) and outer coding (convolutional) • Program table info is added RS, Conv PES-1 TS PES-2 rx Program (MBC) PES-3 Program map
RTP based transmission • H.323 system components • H.263 terminal • Gateway : to PSTN • Gatekeeper : call and BW broker • MCU: media mixer and trsnscoder • H.263 terminal protocol architectures • Signaling • Data transport
packet 1 packet 2 Internet packet 3 00.00 00.01 재생버퍼 00.10 00.08 packet 1 00.14 cam 00.18 00.20 packet 2 00.27 00.28 00.32 packet 3 00.39 00.40 전송 시간 도착 시간 재생 시간 RTP • Real time protocol • Supports time-stamp, not guarantees the real time transmission • “He………llo” is different from “Hell ……..o”
Ver P X CC M PT Sequence Number Time-Stamp Synchronization Source Identifier (SSRC) Contributing Source Identifier (CSRC) . : Contributing Source Identifier (CSRC) RTP 의 위상 IP/UDP header • RTP Format • V: Version (현재 버전은 ‘2’) • P: Padding 유/ 무 (‘1’/’0’) • 패딩의 마지막 바이트는 패딩의 길이 • X: 확장 헤더 유/ 무 (‘1’/’0’) • CC: Contributor(CSRC ID)의 수 ( 0~15 ) • M: Marker bit : frame end/ silent period • PT: Payload Type • Fixed : 0: PCMu Audio, 33: MPEG2 Video • Dynamic 97+ (payload header) Video/audio data
End system SSRC=‘9’ PCMu Audio Translater H.261 Mixer SSRC=’5’ SSRC=’5’ 203.246.81.51 CSRC=’9’ / ’15’ G.721 Audio End system SSRC=’15’ 203.246.81.10 203.246.81.50 203.246.81.54 • SN (sequence number) • 첫번째 패킷- random하게 설정 , 전송시마다 1 씩 증가 • TS (time-stamp) • 첫번째 패킷- random/negotiated value • 이전 timestamp값 + 재생되어야 하는 시간 (클럭틱에 의존) • SSRC & CSRC • Mixing 시에 사용
200 Sender Report 201 Receiver Report RTCP Message types 202 Source Description Message 203 Bye Message 204 Application Specific Message <Type> • RTCP (Real-time control protocol) • Help RTP function • Use rtp port + 1
End system SSRC=‘9’ PCMu Audio Translater H.261 Mixer SSRC=’5’ SSRC=’5’ 203.246.81.51 CSRC=’9’ / ’15’ G.721 Audio End system SSRC=’15’ 203.246.81.10 203.246.81.50 203.246.81.54 RTCP
Conclusion • Congratulation ! • You have finished “From basics, through standard, to application on video coding” • The course was not 100% perfect, but ok. • I hope you could find any interesting the video coding and information theoretic topics. • Any further research cooperation is welcomed.