This document discusses three methods for reducing the bit-rate of transmitted video streams: 1) Time-shifting of MPEG-2 packets, which smooths out variable bit-rates without changing individual encoding rates; 2) Open loop transrating, which uses encoding tools to recompress streams at lower rates in a non-reversible way; 3) Closed loop transrating, which iteratively adjusts rates using feedback to maintain quality. These techniques help network operators optimize bandwidth usage and revenues by controlling streaming rates to match infrastructure limits and service pricing models.
This document discusses three methods for reducing the bit-rate of transmitted video streams: 1) Time-shifting of MPEG-2 packets, which smooths out variable bit-rates without changing individual encoding rates; 2) Open loop transrating, which uses encoding tools to recompress streams at lower rates in a non-reversible way; 3) Closed loop transrating, which iteratively adjusts rates using feedback to maintain quality. These techniques help network operators optimize bandwidth usage and revenues by controlling streaming rates to match infrastructure limits and service pricing models.
This document analyzes the RC4 encryption algorithm and examines how its performance is affected by changing parameters like encryption key length and file size. Experimental tests were conducted to measure encryption time for different key lengths and file types. The results show encryption time increases with longer keys and larger files, and are modeled mathematically. The document also provides background on encryption methods, how RC4 works, and compares stream and block ciphers.
This document provides a European standard for a second generation digital transmission system for cable systems, known as DVB-C2. It defines the system architecture, input processing, bit-interleaved coding and modulation, data slice packet generation, layer 1 part 2 signalling generation and coding, frame builder functions, and OFDM generation for the DVB-C2 system. The standard specifies the frame structure, coding, modulation, and other technical aspects to enable digital video and audio broadcasting over cable networks.
1) The document describes a modification to the Huffman coding used in JPEG image compression. It proposes pairing each non-zero DCT coefficient with the run-length of subsequent (rather than preceding) zero coefficients.
2) This allows using separate optimized Huffman code tables for each DCT coefficient position, improving compression by 10-15% over standard JPEG coding.
3) The decoding procedure is not changed and no end-of-block marker is needed, providing advantages with no increase in complexity.
This document provides an introduction to digital television. It discusses analog TV standards and the conversion to digital with ITU-BT.601 and BT.709 defining digital video formats. It describes MPEG-2 transport streams and tables for encoding digital TV signals. Standards for digital terrestrial, satellite and cable broadcasting networks are also summarized.
This document discusses intra-AS and inter-AS routing. It begins by explaining how routers are organized into autonomous systems (AS) and how gateway routers run both intra-AS routing protocols within their AS and inter-AS routing protocols with gateway routers in other ASes. It then provides examples of intra-AS and inter-AS routing, explaining how packets flow between hosts in different ASes. The document also covers IP addressing schemes including classful addressing and CIDR, and how addresses are allocated hierarchically to allow for efficient routing.
This document discusses quality of service (QoS) provisioning in wireless multimedia networks. It describes QoS challenges in wireless networks due to limited bandwidth, unreliable links, and varying channel conditions. It also discusses the characteristics of multimedia services and traffic modeling challenges. The document outlines IEEE 802.11 MAC layer enhancements including the distributed coordination function, point coordination function, and IEEE 802.11e standard for supporting QoS through enhanced distributed channel access and hybrid coordination function. It emphasizes the need for end-to-end QoS, adaptive frameworks, and call admission control for wireless multimedia networks.
This document proposes a bi-level/full-color video combination scheme to enable video communication across a wide range of bandwidths. Bi-level video uses 1 bit per pixel and works well below 56 Kbps, while full-color video has higher quality but requires over 33.6 Kbps. The scheme uses bandwidth estimation to switch between the two formats in the 33.6-56 Kbps range for smooth adaptation. It estimates available bandwidth from receiver feedback on packet loss and round-trip time, then adjusts the video format accordingly.
This document discusses three methods for reducing the bit-rate of transmitted video streams: 1) Time-shifting of MPEG-2 packets, which smooths out variable bit-rates without changing individual encoding rates; 2) Open loop transrating, which uses encoding tools to recompress streams at lower rates in a non-reversible way; 3) Closed loop transrating, which iteratively adjusts rates using feedback to maintain quality. These techniques help network operators optimize bandwidth usage and revenues by controlling streaming rates to match infrastructure limits and service pricing models.
This document analyzes the RC4 encryption algorithm and examines how its performance is affected by changing parameters like encryption key length and file size. Experimental tests were conducted to measure encryption time for different key lengths and file types. The results show encryption time increases with longer keys and larger files, and are modeled mathematically. The document also provides background on encryption methods, how RC4 works, and compares stream and block ciphers.
This document provides a European standard for a second generation digital transmission system for cable systems, known as DVB-C2. It defines the system architecture, input processing, bit-interleaved coding and modulation, data slice packet generation, layer 1 part 2 signalling generation and coding, frame builder functions, and OFDM generation for the DVB-C2 system. The standard specifies the frame structure, coding, modulation, and other technical aspects to enable digital video and audio broadcasting over cable networks.
1) The document describes a modification to the Huffman coding used in JPEG image compression. It proposes pairing each non-zero DCT coefficient with the run-length of subsequent (rather than preceding) zero coefficients.
2) This allows using separate optimized Huffman code tables for each DCT coefficient position, improving compression by 10-15% over standard JPEG coding.
3) The decoding procedure is not changed and no end-of-block marker is needed, providing advantages with no increase in complexity.
This document provides an introduction to digital television. It discusses analog TV standards and the conversion to digital with ITU-BT.601 and BT.709 defining digital video formats. It describes MPEG-2 transport streams and tables for encoding digital TV signals. Standards for digital terrestrial, satellite and cable broadcasting networks are also summarized.
This document discusses intra-AS and inter-AS routing. It begins by explaining how routers are organized into autonomous systems (AS) and how gateway routers run both intra-AS routing protocols within their AS and inter-AS routing protocols with gateway routers in other ASes. It then provides examples of intra-AS and inter-AS routing, explaining how packets flow between hosts in different ASes. The document also covers IP addressing schemes including classful addressing and CIDR, and how addresses are allocated hierarchically to allow for efficient routing.
This document discusses quality of service (QoS) provisioning in wireless multimedia networks. It describes QoS challenges in wireless networks due to limited bandwidth, unreliable links, and varying channel conditions. It also discusses the characteristics of multimedia services and traffic modeling challenges. The document outlines IEEE 802.11 MAC layer enhancements including the distributed coordination function, point coordination function, and IEEE 802.11e standard for supporting QoS through enhanced distributed channel access and hybrid coordination function. It emphasizes the need for end-to-end QoS, adaptive frameworks, and call admission control for wireless multimedia networks.
This document proposes a bi-level/full-color video combination scheme to enable video communication across a wide range of bandwidths. Bi-level video uses 1 bit per pixel and works well below 56 Kbps, while full-color video has higher quality but requires over 33.6 Kbps. The scheme uses bandwidth estimation to switch between the two formats in the 33.6-56 Kbps range for smooth adaptation. It estimates available bandwidth from receiver feedback on packet loss and round-trip time, then adjusts the video format accordingly.
This document discusses cellular network planning and optimization, specifically for WCDMA radio resource management (RRM). It covers several key topics:
Quality of Service (QoS) in UMTS is achieved through a system of bearers that negotiate bandwidth and latency requirements between network elements. Radio access bearers connect the user equipment to the core network.
RRM functions like admission control, power control, handover control, and packet scheduling work to guarantee QoS, maintain coverage, and optimize cell capacity in WCDMA networks. Power control is a critical RRM mechanism that uses fast and outer loop techniques to control transmission power and mitigate interference.
Latency Considerations in LTE: Implications to Security GatewayTerry Young
This white paper discusses latency considerations for LTE and LTE-Advanced networks. Latency requirements are becoming more stringent over time. LTE-A targets latency of 10ms or less, with 1ms or less required for the X2 interface to support new optimization techniques. Higher latency can negatively impact user experience through slower page loads and reduced throughput. It can also result in lost revenue for online businesses. Any network element must minimize its contribution to overall latency in order to meet budgets. Low-latency solutions like the Stoke Security eXchange are important for meeting stringent LTE-A requirements.
Probabilistic Approach to Provisioning of ITV - By Amos_KohnAmos Kohn
This white paper discusses a probabilistic approach to provisioning network and computing resources for delivering interactive TV. It develops a proprietary spreadsheet model to estimate the costs and benefits of deploying an interactive TV streaming processor. The model is based on analyzing user behavior, data packaging into MPEG streams, required bit rates, forward and return network paths, processing needs, and financial projections to calculate return on investment.
Minimizing network delay or latency is a critical factor in delivering mobile broadband services; businesses and users expect network response will be close to instantaneous. Excess latency can have a profound effect on user experience—from excess delay during a simple phone conversation, reducing throughput at edge of cell coverage areas by reducing effectiveness of RAN optimization techniques, to slow- loading webpages and delays with streaming video. Response delays negatively impact revenue. In financial institutions, low latency networks have become a competitive advantage where even a few extra microseconds, can enable trades to execute ahead of the competition.
The direct correlation between delay and revenue in the web browsing experience is well documented. Amazon famously claimed that every 100 millisecond reduction in delay led to a one percent increase in sales. Google also stated that for every half second delay, it saw a 20 percent reduction in traffic.
For LTE network operators, control of latency is growing in importance as both an operational and business issue. Low latency is not only critical to maintaining the quality user experience (and therefore, the operator competitive advantage) of growing social, M2M, and real-time services, but latency reduction is fundamental to meeting the capacity expectations of LTE-A, where latency budgets will be cut in half and X2 will need to perform at microsecond speed.
Total network latency is the sum of delay from all the network components, including air interface, the processing, switching, and queuing of all network elements (core and RAN) along the path, and the propagation delay in the links. With ever tightening latency expectations, the relative contribution of any individual network element, such as a security gateway, must be minimized. For example, when latency budgets were targeting 150ms, a network node providing packet processing at 250μs was only adding 0.17% to the budget. However, in LTE-A, with latency targets slashed to 10ms, that same network node will consume almost 15x more of the budget. More important, when placed on the S1 with a target of only 1ms, 250 μs is 25% of the entire S1 latency allocation, and endangers meeting the microsecond latency needed at the X2. Clearly, operators need to apply stringent latency requirements for all network nodes, when designing LTE and LTE-A networks.
This document provides guidance on optimizing the paging success rate key performance indicator (KPI) in GSM base station subsystem (BSS) networks. It defines paging success rate as the ratio of paging responses received to paging requests sent. The document outlines potential causes of low paging success rates such as hardware faults, transmission problems, parameter misconfiguration, interference, coverage issues, and uplink/downlink imbalance. It then provides detailed analysis and optimization procedures to address each cause. Case studies demonstrate how to resolve real-world paging success rate issues.
The document discusses key performance indicators (KPIs) for GSM base station subsystem (BSS) networks, including the paging success rate KPI. It defines paging success rate, describes factors that affect it such as coverage, interference, and traffic volume. The document also discusses network parameters that impact paging success rate, such as paging times/intervals, paging based on location area versus all cells, and mobility management parameters like T3212. The goal is to understand KPI measurement points and constraints in order to optimize network performance.
This document discusses radio network planning for WCDMA networks. It describes the planning process, which involves initial planning, detailed radio network planning, and network operation and optimization. The initial planning phase involves estimating site density and configurations through radio link budgeting and coverage analysis. Key aspects of WCDMA link budgeting include interference degradation margin, fast fading margin, transmit power increase, and soft handover gain. Detailed planning then refines site locations and configurations based on propagation modeling and traffic forecasts. Network performance is then analyzed and optimized through monitoring key performance indicators.
This paper proposes an adaptive energy management policy for wireless video streaming between a battery-powered client and server. It models the energy consumption of the server and client based on factors like CPU frequency, transmission power, and channel bandwidth. The paper formulates an optimization problem to assign optimal energy to each video frame. This maximizes system lifetime while meeting a minimum video quality requirement. Experimental results show the proposed policy increases overall system lifetime by 20% on average.
Wcdma Radio Network Planning And OptimizationPengpeng Song
The document discusses WCDMA radio network planning and optimization, including key topics such as:
1) Fundamentals of WCDMA link budget analysis and radio interface protocol architecture.
2) Radio resource utilization techniques like power control, handover control, and congestion control.
3) Issues of coverage and capacity planning as well as enhancement methods.
4) The process of WCDMA radio network planning including dimensioning, detailed planning, and optimization aspects to address interference.
The document discusses quality of service (QoS) in multimedia communication networks, including QoS parameters and classes, deterministic and predictive QoS parameters, guaranteed and best effort QoS, QoS-aware service models, scheduling and policing mechanisms like priority scheduling and weighted fair queueing, and QoS architectures like Integrated Services and Differentiated Services.
This document provides guidelines for optimizing LTE radio frequency (RF) networks. It describes the network optimization process, including single site verification and RF optimization. RF optimization aims to control pilot pollution while optimizing coverage, signal quality, and handover success rates. The document discusses LTE RF optimization objectives such as RSRP, SINR, and handover success rate. It also covers troubleshooting coverage issues like weak coverage, lack of a dominant cell, and cross coverage. Optimization methods include adjusting antenna parameters, transmit power, and network configuration parameters.
The document provides definitions and analysis methods for optimizing the Call Setup Success Rate (CSSR) in GSM networks. It defines the CSSR, lists influencing factors, and describes a three-step analysis and optimization process. The process involves identifying causes of low CSSR related to assignment success rate, immediate assignment success rate, or SDCCH drop rate. An optimization case from Vietnam addresses a difference in core network mechanisms that lowered the CSSR.
This document summarizes recent research on video streaming over Bluetooth networks. It discusses three key areas: intermediate protocols, quality of service (QoS) control, and media compression. For intermediate protocols, it evaluates streaming via HCI, L2CAP, and IP layers and their tradeoffs. For QoS control, it describes how error control mechanisms like link layer FEC, retransmission, and error concealment can improve video quality over Bluetooth. It also discusses congestion control. For media compression, it notes the importance of compression to achieve efficiency over limited Bluetooth bandwidths.
Survey paper on Virtualized cloud based IPTV Systemijceronline
International Journal of Computational Engineering Research(IJCER) is an intentional online Journal in English monthly publishing journal. This Journal publish original research work that contributes significantly to further the scientific knowledge in engineering and Technology.
1. The document discusses planning a WCDMA network, including dimensioning the network, estimating coverage and capacity, and accounting for uncertainties.
2. Dimensioning involves initially estimating the number of sites and equipment needed based on factors like traffic load and distribution. Coverage is estimated using link budget calculations and propagation models. Capacity is estimated based on load factor calculations that account for interference.
3. Planning must consider uncertainties from factors like user locations, speeds, and data rates that impact coverage and capacity in real networks. Both static and dynamic simulations are used to optimize the network plan.
Probabilistic Approach to Provisioning of ITV - Amos K.Amos Kohn
This white paper discusses a probabilistic approach to provisioning network and computing resources for delivering interactive TV. It develops a proprietary spreadsheet model to estimate the costs and benefits of deploying an interactive TV streaming processor. The model is based on analyzing user behavior, data packaging into MPEG streams, required bit rates, transport of data over the forward and return paths, necessary processing power, and financial projections to calculate return on investment.
IBM VideoCharger and Digital Library MediaBase.docVideoguy
This document provides an overview of video streaming over the internet. It discusses video compression standards like H.261, H.263, MJPEG, MPEG1, MPEG2 and MPEG4. It also covers internet transport protocols like TCP and UDP, and challenges like firewall penetration. Both commercial streaming products and research projects aiming to improve streaming are reviewed, with limitations of current approaches outlined. The SuperNOVA research project is evaluated against other work seeking to make high quality video streaming over the internet practical.
An SDN Based Approach To Measuring And Optimizing ABR Video Quality Of Experi...Cisco Service Provider
Reprinted with permission of NCTA, from the 2014 Cable Connection Spring Technical Forum Conference Proceedings. For more information on Cisco video solutions, visit: http://www.cisco.com/c/en/us/products/video/index.html
International Journal of Engineering Research and DevelopmentIJERD Editor
The document summarizes an emerging VP8 video codec that is designed for mobile devices. It aims to significantly reduce computational complexity through several techniques while maintaining good video quality. The key techniques include a predictive algorithm for motion estimation that reduces computation by 18.5-20x compared to full search, using integer discrete cosine transform instead of floating point to achieve 2.6-3.5x speed improvement, and skipping DCT and quantization for some macroblocks to reduce computations. Experimental results on test sequences show negligible quality degradation of 0.2-0.5dB for integer DCT and 0.5dB on average for the full codec, while achieving real-time encoding rates on mobile devices. The proposed low-complexity
Multicasting Of Adaptively-Encoded MPEG4 Over Qos-Cognizant IP NetworksEditor IJMTER
we propose a novel architectural planning for multicasting of adaptively-encoded
layered MPEG4 over a QoS-aware IP network. We re-quire a QoS-aware IP network in this case to
(1) Support priority dropping of packets in time of congestion. (2) Provide congestion notification to
the multicast sender. For the first requirement, we use RED's extension for service differentiation. It
recognizes the priority of packets when they need to be dropped and drops lower priority packets
first. We couple RED with our proposal for the second requirement which is the adoption of
Backward Explicit Congestion Notification (BECN) for use with IP multicast. BECN will provide
early congestion notification at the IP layer level to the video sender. BECN detects upcoming
congestion based on size of the RED queue in the routers. The MPEG4 adaptive-encoder can change
the sending rate and also can divide the video packets into lower priority packets and high priority
packets. Based on BECN messages from the routers, a simple flow controller at the sender sets the
rate for the adaptive MPEG4 encoder and also sets the ratio between the high priority and low
priority packets within the video stream. We use a TES model for generating the MPEG4 traffic that
is based on real video traces. Simulation results show that combining priority dropping, MPEG4
adaptive encoding, and multicast BECN: (1) Improves bandwidth utilization (2) Reduces time to
react to congestion and hence improves the received video quality (3) Maintains graceful degradation
in quality with congestion and provides minimum quality even if congestion persists.
An Overview on Multimedia Transcoding Techniques on Streaming Digital Contentsidescitation
The current IT infrastructure as well as various
commercial applications are directly formulated based on
deployment in multimedia system e.g. education, marketing,
risk management, tele-medicines, military etc. One of the
challenges found in using such application is to deliver
uninterrupted stream of video between multiple terminals
e.g. smart-phone, PDAs, laptops, IPTV etc. The research shows
that there is a stipulated need of designing novel mechanism
of bit rate adjustment as well as format conversion policy so
that the source stream may stream well in diverse end devices
with multiple configuration of processor, memory, decoding
etc. This paper discusses various eminent points from
literature that will throw better highlights in understanding
a schema of direct digital-to-digital data conversion of one
encoding to another termed as transcoding. Although
multimedia transcoding has covered more than a decade in
the area of research, but unfortunately, there is a huge trade-
off between the application, service, resource constraint, and
hardware design that gives rise to QoS issues.
This document discusses network provisioning for multimedia services using traffic aggregation. It covers topics like network provisioning, packet aggregation, traffic engineering, dimensioning, traffic analysis and aggregation. Methods are proposed for optimizing network resource reservations to guarantee delay bounds for aggregated multimedia traffic, including using real video traces and generating synthetic aggregates. Network provisioning scenarios are described for provisioning using real traces, dynamic aggregates, traffic patterns, and optimizing bandwidth utilization.
This document discusses cellular network planning and optimization, specifically for WCDMA radio resource management (RRM). It covers several key topics:
Quality of Service (QoS) in UMTS is achieved through a system of bearers that negotiate bandwidth and latency requirements between network elements. Radio access bearers connect the user equipment to the core network.
RRM functions like admission control, power control, handover control, and packet scheduling work to guarantee QoS, maintain coverage, and optimize cell capacity in WCDMA networks. Power control is a critical RRM mechanism that uses fast and outer loop techniques to control transmission power and mitigate interference.
Latency Considerations in LTE: Implications to Security GatewayTerry Young
This white paper discusses latency considerations for LTE and LTE-Advanced networks. Latency requirements are becoming more stringent over time. LTE-A targets latency of 10ms or less, with 1ms or less required for the X2 interface to support new optimization techniques. Higher latency can negatively impact user experience through slower page loads and reduced throughput. It can also result in lost revenue for online businesses. Any network element must minimize its contribution to overall latency in order to meet budgets. Low-latency solutions like the Stoke Security eXchange are important for meeting stringent LTE-A requirements.
Probabilistic Approach to Provisioning of ITV - By Amos_KohnAmos Kohn
This white paper discusses a probabilistic approach to provisioning network and computing resources for delivering interactive TV. It develops a proprietary spreadsheet model to estimate the costs and benefits of deploying an interactive TV streaming processor. The model is based on analyzing user behavior, data packaging into MPEG streams, required bit rates, forward and return network paths, processing needs, and financial projections to calculate return on investment.
Minimizing network delay or latency is a critical factor in delivering mobile broadband services; businesses and users expect network response will be close to instantaneous. Excess latency can have a profound effect on user experience—from excess delay during a simple phone conversation, reducing throughput at edge of cell coverage areas by reducing effectiveness of RAN optimization techniques, to slow- loading webpages and delays with streaming video. Response delays negatively impact revenue. In financial institutions, low latency networks have become a competitive advantage where even a few extra microseconds, can enable trades to execute ahead of the competition.
The direct correlation between delay and revenue in the web browsing experience is well documented. Amazon famously claimed that every 100 millisecond reduction in delay led to a one percent increase in sales. Google also stated that for every half second delay, it saw a 20 percent reduction in traffic.
For LTE network operators, control of latency is growing in importance as both an operational and business issue. Low latency is not only critical to maintaining the quality user experience (and therefore, the operator competitive advantage) of growing social, M2M, and real-time services, but latency reduction is fundamental to meeting the capacity expectations of LTE-A, where latency budgets will be cut in half and X2 will need to perform at microsecond speed.
Total network latency is the sum of delay from all the network components, including air interface, the processing, switching, and queuing of all network elements (core and RAN) along the path, and the propagation delay in the links. With ever tightening latency expectations, the relative contribution of any individual network element, such as a security gateway, must be minimized. For example, when latency budgets were targeting 150ms, a network node providing packet processing at 250μs was only adding 0.17% to the budget. However, in LTE-A, with latency targets slashed to 10ms, that same network node will consume almost 15x more of the budget. More important, when placed on the S1 with a target of only 1ms, 250 μs is 25% of the entire S1 latency allocation, and endangers meeting the microsecond latency needed at the X2. Clearly, operators need to apply stringent latency requirements for all network nodes, when designing LTE and LTE-A networks.
This document provides guidance on optimizing the paging success rate key performance indicator (KPI) in GSM base station subsystem (BSS) networks. It defines paging success rate as the ratio of paging responses received to paging requests sent. The document outlines potential causes of low paging success rates such as hardware faults, transmission problems, parameter misconfiguration, interference, coverage issues, and uplink/downlink imbalance. It then provides detailed analysis and optimization procedures to address each cause. Case studies demonstrate how to resolve real-world paging success rate issues.
The document discusses key performance indicators (KPIs) for GSM base station subsystem (BSS) networks, including the paging success rate KPI. It defines paging success rate, describes factors that affect it such as coverage, interference, and traffic volume. The document also discusses network parameters that impact paging success rate, such as paging times/intervals, paging based on location area versus all cells, and mobility management parameters like T3212. The goal is to understand KPI measurement points and constraints in order to optimize network performance.
This document discusses radio network planning for WCDMA networks. It describes the planning process, which involves initial planning, detailed radio network planning, and network operation and optimization. The initial planning phase involves estimating site density and configurations through radio link budgeting and coverage analysis. Key aspects of WCDMA link budgeting include interference degradation margin, fast fading margin, transmit power increase, and soft handover gain. Detailed planning then refines site locations and configurations based on propagation modeling and traffic forecasts. Network performance is then analyzed and optimized through monitoring key performance indicators.
This paper proposes an adaptive energy management policy for wireless video streaming between a battery-powered client and server. It models the energy consumption of the server and client based on factors like CPU frequency, transmission power, and channel bandwidth. The paper formulates an optimization problem to assign optimal energy to each video frame. This maximizes system lifetime while meeting a minimum video quality requirement. Experimental results show the proposed policy increases overall system lifetime by 20% on average.
Wcdma Radio Network Planning And OptimizationPengpeng Song
The document discusses WCDMA radio network planning and optimization, including key topics such as:
1) Fundamentals of WCDMA link budget analysis and radio interface protocol architecture.
2) Radio resource utilization techniques like power control, handover control, and congestion control.
3) Issues of coverage and capacity planning as well as enhancement methods.
4) The process of WCDMA radio network planning including dimensioning, detailed planning, and optimization aspects to address interference.
The document discusses quality of service (QoS) in multimedia communication networks, including QoS parameters and classes, deterministic and predictive QoS parameters, guaranteed and best effort QoS, QoS-aware service models, scheduling and policing mechanisms like priority scheduling and weighted fair queueing, and QoS architectures like Integrated Services and Differentiated Services.
This document provides guidelines for optimizing LTE radio frequency (RF) networks. It describes the network optimization process, including single site verification and RF optimization. RF optimization aims to control pilot pollution while optimizing coverage, signal quality, and handover success rates. The document discusses LTE RF optimization objectives such as RSRP, SINR, and handover success rate. It also covers troubleshooting coverage issues like weak coverage, lack of a dominant cell, and cross coverage. Optimization methods include adjusting antenna parameters, transmit power, and network configuration parameters.
The document provides definitions and analysis methods for optimizing the Call Setup Success Rate (CSSR) in GSM networks. It defines the CSSR, lists influencing factors, and describes a three-step analysis and optimization process. The process involves identifying causes of low CSSR related to assignment success rate, immediate assignment success rate, or SDCCH drop rate. An optimization case from Vietnam addresses a difference in core network mechanisms that lowered the CSSR.
This document summarizes recent research on video streaming over Bluetooth networks. It discusses three key areas: intermediate protocols, quality of service (QoS) control, and media compression. For intermediate protocols, it evaluates streaming via HCI, L2CAP, and IP layers and their tradeoffs. For QoS control, it describes how error control mechanisms like link layer FEC, retransmission, and error concealment can improve video quality over Bluetooth. It also discusses congestion control. For media compression, it notes the importance of compression to achieve efficiency over limited Bluetooth bandwidths.
Survey paper on Virtualized cloud based IPTV Systemijceronline
International Journal of Computational Engineering Research(IJCER) is an intentional online Journal in English monthly publishing journal. This Journal publish original research work that contributes significantly to further the scientific knowledge in engineering and Technology.
1. The document discusses planning a WCDMA network, including dimensioning the network, estimating coverage and capacity, and accounting for uncertainties.
2. Dimensioning involves initially estimating the number of sites and equipment needed based on factors like traffic load and distribution. Coverage is estimated using link budget calculations and propagation models. Capacity is estimated based on load factor calculations that account for interference.
3. Planning must consider uncertainties from factors like user locations, speeds, and data rates that impact coverage and capacity in real networks. Both static and dynamic simulations are used to optimize the network plan.
Probabilistic Approach to Provisioning of ITV - Amos K.Amos Kohn
This white paper discusses a probabilistic approach to provisioning network and computing resources for delivering interactive TV. It develops a proprietary spreadsheet model to estimate the costs and benefits of deploying an interactive TV streaming processor. The model is based on analyzing user behavior, data packaging into MPEG streams, required bit rates, transport of data over the forward and return paths, necessary processing power, and financial projections to calculate return on investment.
IBM VideoCharger and Digital Library MediaBase.docVideoguy
This document provides an overview of video streaming over the internet. It discusses video compression standards like H.261, H.263, MJPEG, MPEG1, MPEG2 and MPEG4. It also covers internet transport protocols like TCP and UDP, and challenges like firewall penetration. Both commercial streaming products and research projects aiming to improve streaming are reviewed, with limitations of current approaches outlined. The SuperNOVA research project is evaluated against other work seeking to make high quality video streaming over the internet practical.
An SDN Based Approach To Measuring And Optimizing ABR Video Quality Of Experi...Cisco Service Provider
Reprinted with permission of NCTA, from the 2014 Cable Connection Spring Technical Forum Conference Proceedings. For more information on Cisco video solutions, visit: http://www.cisco.com/c/en/us/products/video/index.html
International Journal of Engineering Research and DevelopmentIJERD Editor
The document summarizes an emerging VP8 video codec that is designed for mobile devices. It aims to significantly reduce computational complexity through several techniques while maintaining good video quality. The key techniques include a predictive algorithm for motion estimation that reduces computation by 18.5-20x compared to full search, using integer discrete cosine transform instead of floating point to achieve 2.6-3.5x speed improvement, and skipping DCT and quantization for some macroblocks to reduce computations. Experimental results on test sequences show negligible quality degradation of 0.2-0.5dB for integer DCT and 0.5dB on average for the full codec, while achieving real-time encoding rates on mobile devices. The proposed low-complexity
Multicasting Of Adaptively-Encoded MPEG4 Over Qos-Cognizant IP NetworksEditor IJMTER
we propose a novel architectural planning for multicasting of adaptively-encoded
layered MPEG4 over a QoS-aware IP network. We re-quire a QoS-aware IP network in this case to
(1) Support priority dropping of packets in time of congestion. (2) Provide congestion notification to
the multicast sender. For the first requirement, we use RED's extension for service differentiation. It
recognizes the priority of packets when they need to be dropped and drops lower priority packets
first. We couple RED with our proposal for the second requirement which is the adoption of
Backward Explicit Congestion Notification (BECN) for use with IP multicast. BECN will provide
early congestion notification at the IP layer level to the video sender. BECN detects upcoming
congestion based on size of the RED queue in the routers. The MPEG4 adaptive-encoder can change
the sending rate and also can divide the video packets into lower priority packets and high priority
packets. Based on BECN messages from the routers, a simple flow controller at the sender sets the
rate for the adaptive MPEG4 encoder and also sets the ratio between the high priority and low
priority packets within the video stream. We use a TES model for generating the MPEG4 traffic that
is based on real video traces. Simulation results show that combining priority dropping, MPEG4
adaptive encoding, and multicast BECN: (1) Improves bandwidth utilization (2) Reduces time to
react to congestion and hence improves the received video quality (3) Maintains graceful degradation
in quality with congestion and provides minimum quality even if congestion persists.
An Overview on Multimedia Transcoding Techniques on Streaming Digital Contentsidescitation
The current IT infrastructure as well as various
commercial applications are directly formulated based on
deployment in multimedia system e.g. education, marketing,
risk management, tele-medicines, military etc. One of the
challenges found in using such application is to deliver
uninterrupted stream of video between multiple terminals
e.g. smart-phone, PDAs, laptops, IPTV etc. The research shows
that there is a stipulated need of designing novel mechanism
of bit rate adjustment as well as format conversion policy so
that the source stream may stream well in diverse end devices
with multiple configuration of processor, memory, decoding
etc. This paper discusses various eminent points from
literature that will throw better highlights in understanding
a schema of direct digital-to-digital data conversion of one
encoding to another termed as transcoding. Although
multimedia transcoding has covered more than a decade in
the area of research, but unfortunately, there is a huge trade-
off between the application, service, resource constraint, and
hardware design that gives rise to QoS issues.
This document discusses network provisioning for multimedia services using traffic aggregation. It covers topics like network provisioning, packet aggregation, traffic engineering, dimensioning, traffic analysis and aggregation. Methods are proposed for optimizing network resource reservations to guarantee delay bounds for aggregated multimedia traffic, including using real video traces and generating synthetic aggregates. Network provisioning scenarios are described for provisioning using real traces, dynamic aggregates, traffic patterns, and optimizing bandwidth utilization.
H2B2VS (HEVC hybrid broadcast broadband video services) – Building innovative...Raoul Monnier
Broadcast and broadband networks continue to be separate worlds in the video consumption business. Some initiatives such as HbbTV have built a bridge between both worlds, but its application is almost limited to providing links over the broadcast channel to content providers’ applications such as Catch-up TV services. When it comes to reality, the user is using either one network or the other.
H2B2VS is a Celtic-Plus project aiming at exploiting the potential of real hybrid networks by implementing efficient synchronization mechanisms and using new video coding standard such as High Efficiency Video Coding (HEVC). The goal is to develop successful hybrid network solutions that enable value added services with an optimum bandwidth usage in each network and with clear commercial applications. An example of the potential of this approach is the transmission of Ultra-HD TV by sending the main content over the broadcast channel and the required complementary information over the broadband network. This technology can also be used to improve the life of handicapped persons: Deaf people receive through the broadband network a sign language translation of a programme sent over the broadcast channel; the TV set then displays this translation in an inset window.
One of the most important contributions of the project is developing and testing synchronization methods between two different networks that offer unequal qualities of service with significant differences in delay and jitter.
In this paper, the main technological project contributions are described, including SHVC, the scalable extension of HEVC and a special focus on the synchronization solution adopted by MPEG and DVB. The paper also presents some of the implemented practical use cases, such as the sign language translation described above, and their performance results so as to evaluate the commercial application of this type of solution.
High Efficiency of Media Processing Amos K.Amos Kohn
This document discusses challenges with STB-based media personalization for cable operators and proposes a network-based alternative. STB-based personalization is problematic due to the variety of legacy STBs in homes with limited capabilities, the high costs of more powerful STBs, insufficient infrastructure to support advanced features, bandwidth overload on the access network, and threats to retaining subscribers. A network-based approach could address these issues by performing media processing before content reaches STBs, allowing operators to reuse existing infrastructure for a unified experience across devices while lowering costs and retaining customers. The document outlines coding tools like object-based structures and scalable encoding that could enable such a network-based personalization solution.
This document discusses post-processing and rate distortion algorithms for the VP8 video codec. It first provides background on the need for post-processing algorithms to reduce blocking artifacts in compressed video, and for rate control algorithms to regulate bitrates and achieve high video quality within bandwidth constraints. It then summarizes existing in-loop deblocking filters and post-processing algorithms. A novel optimal post-processing/in-loop filtering algorithm is described that can achieve better performance than H.264/AVC or VP8 by computing optimal filter coefficients. Finally, a proposed rate distortion optimization algorithm for VP8 is discussed to improve its rate control and coding efficiency.
Set-top boxes integrate video and audio decoding with a multimedia application environment to provide personalized multimedia services and cable TV through a user-friendly interface. While multimedia computers are more versatile and expensive, set-top boxes are inexpensive limited-functionality devices primarily for entertainment. Digital video networks need high bandwidth delivery to homes with low bandwidth bidirectional communication for interaction between users and providers. Set-top box hardware and software architectures integrate components to decode video, run applications, and provide a uniform interface for interactivity and access to services.
Review on Data Traffic in Real Time for MANETSIRJET Journal
This document discusses different types of data traffic in mobile ad hoc networks (MANETs). It begins by introducing MANETs and some of the key challenges in building them, including dynamic topology and limited bandwidth. It then reviews related work analyzing self-similar variable bit rate (VBR) video traffic and congestion control mechanisms for real-time traffic in MANETs. The main types of data traffic discussed are constant bit rate (CBR), variable bit rate (VBR), and bursty traffic. CBR provides consistent bandwidth but may waste storage, while VBR can optimize quality but requires variable bandwidth. High data traffic loads that exceed network capacity can cause data loss and increased delay.
The document discusses proposed extensions to the DVB-S2 digital video broadcasting standard over satellites. The extensions aim to improve spectrum efficiency and allow for higher data rates. Key points include:
1) Reducing roll-off factors and side lobes of carriers allows transponders to be placed closer together, utilizing bandwidth.
2) Using wider bandwidth transponders that are 3 times the size of typical ones further increases efficiency.
3) A new 64-APSK modulation scheme and additional coding options provide more flexibility to optimize throughput.
4) Overall, the extensions could provide up to 20% higher data rates for broadcasting and 64% for professional services.
Set top boxes allow users to access multimedia services and cable television through a user-friendly interface. They integrate video and audio decoding with an application execution environment. While multimedia computers are more versatile and expensive, set top boxes have limited functionality and lower cost, targeting entertainment applications. Digital video networks use set top boxes to deliver high-bandwidth video and low-bandwidth interactivity between the user and service provider over various connection types like cable, satellite, and internet. Set top box hardware and software architectures integrate components like an MPEG decoder and microprocessor to control devices and enable downloading and running of applications.
The impact of jitter on the HEVC video streaming with Multiple CodingHakimSahour
This document discusses the impact of jitter on video quality when streaming HEVC encoded video over wireless networks. It presents a study evaluating the effects of quantization parameter (QP) values, video content, and jitter on quality of experience (QoE). The study finds that using higher QP values, which lowers bitrate and increases compression, degrades video quality as measured by PSNR. It also finds that different video content results in varying PSNR values for the same encoding settings. Additionally, the results show that adjusting the QP value can help recover from the negative effects of jitter on received video quality. The document proposes using multiple description coding (MDC) to further improve transmission over error-prone wireless channels.
This document evaluates the performance of IPTV video streaming over WiMAX networks under different terrain environments, including free space, outdoor to indoor, and pedestrian environments. It uses OPNET simulations to analyze network statistics such as packet loss, path loss, delay, and throughput. The results show that free space terrain has the lowest path loss and packet delay, while outdoor to indoor and pedestrian environments have higher path loss and delay. Specifically, free space path loss was around 100dB while outdoor environments was around 145dB. Additionally, packet loss was highest for outdoor scenarios due to lower signal to noise ratios in those environments. In general, more obstructed environments led to worse performance for IPTV video streaming over WiMAX networks.
Comparative study of_digital_modulation (1)Bindia Kumari
This document compares different digital modulation techniques that can be used in orthogonal frequency division multiplexing (OFDM) and WiMAX networks. It simulates BPSK, QPSK, 16-QAM and 64-QAM modulation in MATLAB and measures their performance in terms of bit error rate and throughput. The results show that higher order modulations like 64-QAM provide much higher throughput but also higher bit error rates compared to lower order modulations at a given signal-to-noise ratio. The best configuration balances low bit error rates and high throughput.
The Optimization of IPTV Service Through SDN In A MEC Architecture, Respectiv...CSCJournals
The aim of this paper is to present the ‘Power’ of SDN Technology and MEC Technic in improving the delivering of IPTV Service. Those days, the IPTV end –users are tremendous increased all over the world , but in the same time also the complains for receiving these prepaid real time multimedial services like; high latency, high bandwidth, low performance and low QoE/QoS. On the other end, IPTV Distributors need a new system, technics, network solutions to distribute content continuesly and simultaneously to all active end-users with high-quality, lowlatency and high Performance, thus monitoring and re-configuring this ‘Big Data’ require high Bandwidth by causing difficult problems by offering it affecting in the same time the price and QoE/QoSperformance of delivered service.
For this reason, we have achieved to optimize the IPTV service by applying SDN solution in a MEC Architecture (Multiple-Access Edge Computing). In this way , through MEC Technology and SDN, it is possible to receive an IPTV service with Low Latency, High Performance and Low Bandwidth by solving successfully all the problems faced by the actual IPTV Operators. These improvements of delivering IPTV service through MEC will be demonstrated by using the OMNet +++ simulator in an LTE-A mobile network. The results show clearly that by applying the MEC technique in the LTE-A network for receiving IPTV Service through SDN Network, the service was delivered with latency decreased by >90% (compared to the cases when the MEC technique is not applied), with PacketLoss of almost 0 and with high performance QoE. In addition these strong Contributions, the ‘Big’ innovation achieved in this work through simulations is that the quality of delivered IPTV Service did not change according to the increasing of the end-users.This latency of delivering the video streaming services did not change. This means that the IPTV Service providers will increase their benefits by ensuring in the same time also the delivering of service with high quality and performance toward innumerous end users. Consequently, MEC Technology and SDN solution will be the two right and "smart" network choices that will boost the development of the 5th Mobile generation and will significantly improve the benefit of Video Streaming services offered by current providers worldwide (Netflix, HULU, Amazon Prime, YouTube, etc).
This document summarizes a research paper that proposes a framework to improve quality of experience and energy efficiency for heterogeneous wireless multimedia broadcast receivers. The framework groups users based on their device capabilities and channel conditions. It broadcasts scalable video streams that are encoded with different layers to support different groups. Time slicing is used to allow discontinuous reception and energy savings by turning radios off between bursts. A game theoretic model is used to optimize source encoding, transmission scheduling, and modulation/coding to maximize reception quality and network capacity while balancing energy usage. Evaluation shows the approach enables 75-95% energy savings.
The document discusses techniques for constant bit rate video streaming over packet switching networks. It proposes adapting variable bit rate video to a constant bit rate by controlling the video encoder's output rate based on buffer level feedback. This allows transporting video over networks using constant bit rate channels while avoiding network congestion issues. The key techniques involve bit allocation to each coding unit based on buffer status, and adjusting encoder quantization parameters to encode units with the allocated bits. Simulation results show the approach maintains constant compression ratio and peak signal-to-noise ratio while varying group of picture size and quality settings.
The document provides an overview of MPEG-4, a standard that offers both advanced audio and video codecs as well as tools for combining multimedia such as audio, video, graphics and interactivity. It was developed through an open international process to select the best technologies. MPEG-4 codecs like AVC and AAC provide high compression efficiency, having been adopted for HDTV, mobile video, and digital music. Its rich media tools allow interactive experiences combining different media types.
This document provides an overview of Codan's 6700/6900 series block up converter (BUC) systems and components. It describes the BUC, low-noise block converter (LNB), and redundancy systems. It also covers installation, operation, and troubleshooting of the systems. The document contains information on frequency bands, conversion plans, interfaces, cable connections, monitor/control, commands, maintenance procedures, and compliance standards.
This document discusses digital set-top boxes (STBs) and related standards. It covers:
1) The DVB standards for digital TV broadcasting via different transmission media, including DVB-T for terrestrial, DVB-S for satellite, and DVB-C for cable. These share source coding/compression and service multiplexing standards.
2) STBs will be needed until integrated digital TVs are cheaper. Affordable STBs are key for digital TV adoption. Common standards help lower STB costs through economies of scale.
3) "Open architecture" and "interoperability" mean the STB functionality is defined by public standards and can receive services across networks, respectively. The
The document discusses DCT/IDCT concepts and applications. It provides an introduction to DCT and IDCT, explaining that they are used widely in video and audio compression. It describes the DCT and IDCT functions and how they work to transform signals between spatial and frequency domains. Examples of one-dimensional and two-dimensional DCT/IDCT equations are also given. Finally, common applications of DCT/IDCT compression techniques are listed, such as in DVD players, cable TV, graphics cards, and medical imaging systems.
This document discusses image compression using the discrete cosine transform (DCT). It develops simple Mathematica functions to compute the 1D and 2D DCT. The 1D DCT transforms a list of real numbers into elementary frequency components. It is computed via matrix multiplication or using the discrete Fourier transform with twiddle factors. The 2D DCT applies the 1D DCT to rows and then columns of an image, making it separable. These functions illustrate how Mathematica can be used to prototype image processing algorithms.
DVB-S2 is the second-generation specification for satellite broadcasting developed by DVB in 2003. It uses more advanced channel coding (LDPC codes) and modulation formats (QPSK, 8PSK, 16APSK, 32APSK) for a 30% increase in transmission capacity over DVB-S. DVB-S2 allows for adaptive coding and modulation to optimize transmission for each user. It is designed for broadcast, interactive, and professional applications with flexibility to handle different transponder characteristics and content formats.
The STi7167 is an integrated system-on-chip that combines a configurable DVB-T or DVB-C demodulator with STB decoding and display functions. It provides advanced HD and SD video decoding, audio decoding, graphics processing, and connectivity options. The chip's integrated features allow for low cost and small size STB designs for cable or terrestrial networks.
This document provides an overview of service information (SI) in digital video broadcasting (DVB) systems, including sections like the network information section (NIT), service description section (SDT), bouquet association section (BAT), program association section (PAT), conditional access section (CAT), transport stream description section (TSDT), event information section (EIT), and running status section (RST). It includes syntax diagrams and details for each section, such as table IDs, section lengths, descriptors, and other fields. It also provides the PID and refresh interval requirements for each table type.
1) The document describes a modification to the Huffman coding used in JPEG image compression. It proposes pairing each non-zero DCT coefficient with the run-length of subsequent (rather than preceding) zero coefficients.
2) This allows using separate optimized Huffman code tables for each DCT coefficient position, improving compression by 10-15% over standard JPEG coding.
3) The decoding procedure is not changed and no end-of-block marker is needed, providing advantages with no increase in complexity.
Dani Pedrosa won the MotoGP race at Laguna Seca, finishing just 0.344 seconds ahead of Valentino Rossi in second and 1.926 seconds ahead of Jorge Lorenzo in third. Casey Stoner finished fourth, over 12 seconds behind Pedrosa. There were several crashes during the race, with Andrea Dovizioso, Sete Gibernau, and Gabor Talmacsi all falling out of contention. James Toseland received a ride through penalty for a jump start.
The document provides implementation guidelines for using the DVB Simulcrypt standard, including describing the architecture and protocols, clarifying differences between protocol versions, explaining state diagrams and behaviors, and providing recommendations for error handling, redundancy management, and custom signaling profiles to facilitate reliable and efficient Simulcrypt headend implementation.
1) The document discusses quantization and pulse code modulation (PCM) in voice signal encoding. PCM assigns 256 possible values to digitally represent analog voice samples, divided into chords and steps on a linear scale.
2) A logarithmic quantization scale is better than a linear one for voice signals, as it allocates more quantization steps to lower amplitudes prevalent in speech. This "compressed encoding" improves fidelity.
3) Quantization error occurs when samples with different amplitudes are assigned the same digital value, distorting the reconstructed waveform. Compression helps maintain a higher signal-to-noise ratio especially for low amplitudes.
This document provides implementation guidelines for the DVB Simulcrypt standard. It describes the architecture and protocols involved in simulcrypt systems, including the ECMG protocol between the security client system and conditional access modules, and the EMMG/PDG protocol between conditional access modules and multiplex equipment. The document outlines differences between version 1 and 2 of the standards, and provides recommendations for compliance. It also includes detailed state diagrams and descriptions of the protocols involved.
The Event Logger monitors and logs Digital Program Insertion (DPI) messages to verify correct transmission of signals via satellite. It watches for configured GPI state changes that indicate an expected DPI message. If the message is received on time, it is logged as a matched event. If not received on time, it is flagged as missed. The Event Logger also decodes DPI messages to help diagnose issues, and is compatible with various encoding systems. It has 6 ASI inputs, 108 GPI sensors, and logs data in real-time and for archiving.
This document discusses the basics of BISS scrambling. It describes BISS mode 1, which uses a session word, and BISS mode E, which encrypts the session word using an identifier and encryption algorithm. BISS mode E provides an additional layer of protection for transmitting the session word. The document also covers calculating the encrypted session word, using buried and injected identifiers, and how to operate scramblers in the different BISS modes.
Euler's theorem states that for any plane graph, the number of vertices (v) minus the number of edges (e) plus the number of faces (f) equals 2. The document proves this theorem by considering a minimal tree (T) within the graph and its dual tree (D), showing that the number of edges of T and D sum to the total edges (e) of the original graph. Some applications of the theorem are that any plane graph contains an edge of degree 5 or higher and any finite set of points not all on a line contains a line with exactly two points.
This document provides an overview of satellite communications fundamentals. It discusses how satellites provide capabilities not available through landlines, such as mobility and quick implementation. However, satellites are not always the most cost effective solution due to limited frequency spectrum and spatial capacity. The document describes different types of satellite services and configurations, including geostationary and non-geostationary satellites. It also covers topics like frequency reuse, earth station antennas, and satellite link delays.
The document discusses quantization in analog-to-digital conversion. It describes the three processes of A/D conversion as sampling, quantization, and binary encoding. Quantization involves mapping amplitude values into a set of discrete values using a quantization interval or step size. The document discusses uniform quantization and how the quantization levels are determined. It also covers non-uniform quantization and provides examples and MATLAB code demonstrations of audio signal quantization.
1) Reed-Solomon codes are a type of error-correcting code invented in 1960 that can detect and correct multiple symbol errors. They work by encoding data into redundant symbols that can be used to detect and locate errors.
2) Reed-Solomon codes are particularly good at correcting burst errors, where a block of symbols are corrupted together by noise. Even if an entire block of bits is corrupted, the code can still correct the errors by replacing the corrupted symbol.
3) The error correction capability of Reed-Solomon codes increases with larger block sizes, as noise is averaged over more symbols. However, implementing Reed-Solomon codes also becomes more complex with higher redundancy.
This document describes the head-end architecture and synchronization for digital video broadcasting using SimulCrypt. It outlines the system components including an event information scheduler, SimulCrypt synchronizer, entitlement control message generator, entitlement management message generator, and multiplexer. It also describes the interfaces between these components, covering processes like channel and stream establishment and closure, as well as bandwidth allocation and status reporting.
AppSec PNW: Android and iOS Application Security with MobSFAjin Abraham
Mobile Security Framework - MobSF is a free and open source automated mobile application security testing environment designed to help security engineers, researchers, developers, and penetration testers to identify security vulnerabilities, malicious behaviours and privacy concerns in mobile applications using static and dynamic analysis. It supports all the popular mobile application binaries and source code formats built for Android and iOS devices. In addition to automated security assessment, it also offers an interactive testing environment to build and execute scenario based test/fuzz cases against the application.
This talk covers:
Using MobSF for static analysis of mobile applications.
Interactive dynamic security assessment of Android and iOS applications.
Solving Mobile app CTF challenges.
Reverse engineering and runtime analysis of Mobile malware.
How to shift left and integrate MobSF/mobsfscan SAST and DAST in your build pipeline.
Essentials of Automations: Exploring Attributes & Automation ParametersSafe Software
Building automations in FME Flow can save time, money, and help businesses scale by eliminating data silos and providing data to stakeholders in real-time. One essential component to orchestrating complex automations is the use of attributes & automation parameters (both formerly known as “keys”). In fact, it’s unlikely you’ll ever build an Automation without using these components, but what exactly are they?
Attributes & automation parameters enable the automation author to pass data values from one automation component to the next. During this webinar, our FME Flow Specialists will cover leveraging the three types of these output attributes & parameters in FME Flow: Event, Custom, and Automation. As a bonus, they’ll also be making use of the Split-Merge Block functionality.
You’ll leave this webinar with a better understanding of how to maximize the potential of automations by making use of attributes & automation parameters, with the ultimate goal of setting your enterprise integration workflows up on autopilot.
Taking AI to the Next Level in Manufacturing.pdfssuserfac0301
Read Taking AI to the Next Level in Manufacturing to gain insights on AI adoption in the manufacturing industry, such as:
1. How quickly AI is being implemented in manufacturing.
2. Which barriers stand in the way of AI adoption.
3. How data quality and governance form the backbone of AI.
4. Organizational processes and structures that may inhibit effective AI adoption.
6. Ideas and approaches to help build your organization's AI strategy.
HCL Notes and Domino License Cost Reduction in the World of DLAUpanagenda
Webinar Recording: https://www.panagenda.com/webinars/hcl-notes-and-domino-license-cost-reduction-in-the-world-of-dlau/
The introduction of DLAU and the CCB & CCX licensing model caused quite a stir in the HCL community. As a Notes and Domino customer, you may have faced challenges with unexpected user counts and license costs. You probably have questions on how this new licensing approach works and how to benefit from it. Most importantly, you likely have budget constraints and want to save money where possible. Don’t worry, we can help with all of this!
We’ll show you how to fix common misconfigurations that cause higher-than-expected user counts, and how to identify accounts which you can deactivate to save money. There are also frequent patterns that can cause unnecessary cost, like using a person document instead of a mail-in for shared mailboxes. We’ll provide examples and solutions for those as well. And naturally we’ll explain the new licensing model.
Join HCL Ambassador Marc Thomas in this webinar with a special guest appearance from Franz Walder. It will give you the tools and know-how to stay on top of what is going on with Domino licensing. You will be able lower your cost through an optimized configuration and keep it low going forward.
These topics will be covered
- Reducing license cost by finding and fixing misconfigurations and superfluous accounts
- How do CCB and CCX licenses really work?
- Understanding the DLAU tool and how to best utilize it
- Tips for common problem areas, like team mailboxes, functional/test users, etc
- Practical examples and best practices to implement right away
Monitoring and Managing Anomaly Detection on OpenShift.pdfTosin Akinosho
Monitoring and Managing Anomaly Detection on OpenShift
Overview
Dive into the world of anomaly detection on edge devices with our comprehensive hands-on tutorial. This SlideShare presentation will guide you through the entire process, from data collection and model training to edge deployment and real-time monitoring. Perfect for those looking to implement robust anomaly detection systems on resource-constrained IoT/edge devices.
Key Topics Covered
1. Introduction to Anomaly Detection
- Understand the fundamentals of anomaly detection and its importance in identifying unusual behavior or failures in systems.
2. Understanding Edge (IoT)
- Learn about edge computing and IoT, and how they enable real-time data processing and decision-making at the source.
3. What is ArgoCD?
- Discover ArgoCD, a declarative, GitOps continuous delivery tool for Kubernetes, and its role in deploying applications on edge devices.
4. Deployment Using ArgoCD for Edge Devices
- Step-by-step guide on deploying anomaly detection models on edge devices using ArgoCD.
5. Introduction to Apache Kafka and S3
- Explore Apache Kafka for real-time data streaming and Amazon S3 for scalable storage solutions.
6. Viewing Kafka Messages in the Data Lake
- Learn how to view and analyze Kafka messages stored in a data lake for better insights.
7. What is Prometheus?
- Get to know Prometheus, an open-source monitoring and alerting toolkit, and its application in monitoring edge devices.
8. Monitoring Application Metrics with Prometheus
- Detailed instructions on setting up Prometheus to monitor the performance and health of your anomaly detection system.
9. What is Camel K?
- Introduction to Camel K, a lightweight integration framework built on Apache Camel, designed for Kubernetes.
10. Configuring Camel K Integrations for Data Pipelines
- Learn how to configure Camel K for seamless data pipeline integrations in your anomaly detection workflow.
11. What is a Jupyter Notebook?
- Overview of Jupyter Notebooks, an open-source web application for creating and sharing documents with live code, equations, visualizations, and narrative text.
12. Jupyter Notebooks with Code Examples
- Hands-on examples and code snippets in Jupyter Notebooks to help you implement and test anomaly detection models.
Programming Foundation Models with DSPy - Meetup SlidesZilliz
Prompting language models is hard, while programming language models is easy. In this talk, I will discuss the state-of-the-art framework DSPy for programming foundation models with its powerful optimizers and runtime constraint system.
What is an RPA CoE? Session 1 – CoE VisionDianaGray10
In the first session, we will review the organization's vision and how this has an impact on the COE Structure.
Topics covered:
• The role of a steering committee
• How do the organization’s priorities determine CoE Structure?
Speaker:
Chris Bolin, Senior Intelligent Automation Architect Anika Systems
Your One-Stop Shop for Python Success: Top 10 US Python Development Providersakankshawande
Simplify your search for a reliable Python development partner! This list presents the top 10 trusted US providers offering comprehensive Python development services, ensuring your project's success from conception to completion.
For the full video of this presentation, please visit: https://www.edge-ai-vision.com/2024/06/temporal-event-neural-networks-a-more-efficient-alternative-to-the-transformer-a-presentation-from-brainchip/
Chris Jones, Director of Product Management at BrainChip , presents the “Temporal Event Neural Networks: A More Efficient Alternative to the Transformer” tutorial at the May 2024 Embedded Vision Summit.
The expansion of AI services necessitates enhanced computational capabilities on edge devices. Temporal Event Neural Networks (TENNs), developed by BrainChip, represent a novel and highly efficient state-space network. TENNs demonstrate exceptional proficiency in handling multi-dimensional streaming data, facilitating advancements in object detection, action recognition, speech enhancement and language model/sequence generation. Through the utilization of polynomial-based continuous convolutions, TENNs streamline models, expedite training processes and significantly diminish memory requirements, achieving notable reductions of up to 50x in parameters and 5,000x in energy consumption compared to prevailing methodologies like transformers.
Integration with BrainChip’s Akida neuromorphic hardware IP further enhances TENNs’ capabilities, enabling the realization of highly capable, portable and passively cooled edge devices. This presentation delves into the technical innovations underlying TENNs, presents real-world benchmarks, and elucidates how this cutting-edge approach is positioned to revolutionize edge AI across diverse applications.
Digital Banking in the Cloud: How Citizens Bank Unlocked Their MainframePrecisely
Inconsistent user experience and siloed data, high costs, and changing customer expectations – Citizens Bank was experiencing these challenges while it was attempting to deliver a superior digital banking experience for its clients. Its core banking applications run on the mainframe and Citizens was using legacy utilities to get the critical mainframe data to feed customer-facing channels, like call centers, web, and mobile. Ultimately, this led to higher operating costs (MIPS), delayed response times, and longer time to market.
Ever-changing customer expectations demand more modern digital experiences, and the bank needed to find a solution that could provide real-time data to its customer channels with low latency and operating costs. Join this session to learn how Citizens is leveraging Precisely to replicate mainframe data to its customer channels and deliver on their “modern digital bank” experiences.
Dandelion Hashtable: beyond billion requests per second on a commodity serverAntonios Katsarakis
This slide deck presents DLHT, a concurrent in-memory hashtable. Despite efforts to optimize hashtables, that go as far as sacrificing core functionality, state-of-the-art designs still incur multiple memory accesses per request and block request processing in three cases. First, most hashtables block while waiting for data to be retrieved from memory. Second, open-addressing designs, which represent the current state-of-the-art, either cannot free index slots on deletes or must block all requests to do so. Third, index resizes block every request until all objects are copied to the new index. Defying folklore wisdom, DLHT forgoes open-addressing and adopts a fully-featured and memory-aware closed-addressing design based on bounded cache-line-chaining. This design offers lock-free index operations and deletes that free slots instantly, (2) completes most requests with a single memory access, (3) utilizes software prefetching to hide memory latencies, and (4) employs a novel non-blocking and parallel resizing. In a commodity server and a memory-resident workload, DLHT surpasses 1.6B requests per second and provides 3.5x (12x) the throughput of the state-of-the-art closed-addressing (open-addressing) resizable hashtable on Gets (Deletes).
Generating privacy-protected synthetic data using Secludy and MilvusZilliz
During this demo, the founders of Secludy will demonstrate how their system utilizes Milvus to store and manipulate embeddings for generating privacy-protected synthetic data. Their approach not only maintains the confidentiality of the original data but also enhances the utility and scalability of LLMs under privacy constraints. Attendees, including machine learning engineers, data scientists, and data managers, will witness first-hand how Secludy's integration with Milvus empowers organizations to harness the power of LLMs securely and efficiently.
For the full video of this presentation, please visit: https://www.edge-ai-vision.com/2024/06/how-axelera-ai-uses-digital-compute-in-memory-to-deliver-fast-and-energy-efficient-computer-vision-a-presentation-from-axelera-ai/
Bram Verhoef, Head of Machine Learning at Axelera AI, presents the “How Axelera AI Uses Digital Compute-in-memory to Deliver Fast and Energy-efficient Computer Vision” tutorial at the May 2024 Embedded Vision Summit.
As artificial intelligence inference transitions from cloud environments to edge locations, computer vision applications achieve heightened responsiveness, reliability and privacy. This migration, however, introduces the challenge of operating within the stringent confines of resource constraints typical at the edge, including small form factors, low energy budgets and diminished memory and computational capacities. Axelera AI addresses these challenges through an innovative approach of performing digital computations within memory itself. This technique facilitates the realization of high-performance, energy-efficient and cost-effective computer vision capabilities at the thin and thick edge, extending the frontier of what is achievable with current technologies.
In this presentation, Verhoef unveils his company’s pioneering chip technology and demonstrates its capacity to deliver exceptional frames-per-second performance across a range of standard computer vision networks typical of applications in security, surveillance and the industrial sector. This shows that advanced computer vision can be accessible and efficient, even at the very edge of our technological ecosystem.
"Choosing proper type of scaling", Olena SyrotaFwdays
Imagine an IoT processing system that is already quite mature and production-ready and for which client coverage is growing and scaling and performance aspects are life and death questions. The system has Redis, MongoDB, and stream processing based on ksqldb. In this talk, firstly, we will analyze scaling approaches and then select the proper ones for our system.
1. TRANSRATING EFFICIENCY — BIT-RATE REDUCTION
METHODS TO ENABLE MORE SERVICES OVER EXISTING
BANDWIDTH
L. Lastein and B. Reul
BarcoNet – a Scientific Atlanta Company, Denmark
ABSTRACT
This paper explains the efficiency of three transrating and statistical re-
multiplexing techniques, examines how they work and describes the
combinations most suitable for different real-life applications in Digital
Headends, Play-outs and regional hubs in Broadcast, Terrestrial, Satellite
and Cable Networks.
The three techniques are known as:
• Time-shifting of MPEG-2 packets
• Open loop transrating
• Closed loop transrating
INTRODUCTION
The introduction of MPEG-2 audio and video encoding has created a multitude of new
potential revenue streams. One of the major benefits for a network operator is the
compression of both video and audio to fit into a significantly lower bandwidth. The higher
the cost of the accessible bandwidth, the greater the benefit gain from compression. But the
technology has also meant new challenges for Digital TV (DTV) networks, be it on cable,
satellite or wireless platforms. The parameters used to encode services by the originator
may not fit the business model of the DTV operator. Also, technical issues arising from more
efficient encoding technologies (such as statistical multiplexing) create a demand for
controlling the most important MPEG-2 parameter of them all: The video encoding rates.
What is the problem?
As MPEG-2 compression technology has matured and improved its bandwidth usage
efficiency, one of the major breakthroughs for encoding has been the deployment of
statistical multiplexing. Video programs can now be encoded using a variable bit-rate
according to the actual needs of the specific program compared to other programs in the
same multiplex. When combined with the typical business scenario from a DTV-operator, a
problem arises. The operator has to aggregate existing content to a digital network, reaching
the subscribers. When this content is acquired from different networks and re-multiplexed to
new service bouquets, the total bit-rate of the newly created multiplexes are out of control.
The operator needs to build in a margin to allow for the fluctuating bit-rate and avoid a
network breakdown in the event that the total bandwidth capacity is exceeded.
A relatively new group of players in this market are broadband operators, who target
alternative broadband access methods to the home. Very rapid deployments of xDSL
connections in different regions of the world have enabled new players to enter the market
of other network operators. To increase the number of potential customers they are offering
video-services to be delivered on a broadband connection. However, the bandwidth
limitations of the “last-mile” to the subscriber is a challenge – only single services at a time
2. can be streamed and connected to the subscriber PC or Ethernet-based Set Top Box (STB).
Often the access-network allows only a limited bandwidth, meaning that re-compression of
the individual program stream is necessary. The purpose is to limit all programs to a fixed,
defined bit-rate that may have no relation to the bit-rate used when these services are
presented to the broadband provider.
Optimize the revenue on the existing bandwidth
Network operators offer different content based on unique business models. The
prioritization of the scarce bandwidth does not necessarily match the priorities already
allocated by the content providers using MPEG-2 compression for their distribution. The
operator must allocate the bandwidth resource to the services where he makes money. This
has particular relevance for bandwidth-scarce applications like DVB-T and DVB-S where
clear prioritizations are needed. For high-revenue earning services, the perceived quality by
the subscriber should be high, and for the low-revenue earners there must be compromises
when selecting the quality level.
Using Transrating to create the optimized Business-model for digital services
Controlling a video rate is conceptually simple. It can be achieved by, first decoding, and
then again re-encoding each service. The encoding can be set at a lower rate.. A decoder
and encoder for each service can quickly runs up to a large investment.
In addition, re-encoding does not always result in the best video quality. MPEG-2 encoding
is based on reducing the details in the picture that are less visible to the human eye.
However, when this technology is re-applied on a previously encoded video, artifacts
introduced to the video will be enhanced. Re-encoding will, in these applications, not be in a
position to take advantage of information about previous encoding processes in the
transmission chain.
A solution to this problem is the technology of transrating. This technology results in a
continued high video quality level by effectively re-using information about previous
encoding whilst at the same time delivering a more cost effective solution to the problem. .
The following types of applications may benefit from using transrating.
- Re-multiplexing VBR streams: Relevant for both DVB-T, DVB-S and cable
environments. Single MPEG-2 programs encoded as a part of a statistical multiplex
may vary from 2 to 10 Mbit/s. Transrating will ensure that the operator does not have
to reserve expensive mainly unused overhead in the downstream network.
- Reducing Constant Bit Rate (CBR) streams. Still some services are distributed in
high CBR-rates that will not be cost-efficient to aggregate and re-transmit in a digital
network without lowering the rates. Relevant for all DTV applications.
- A mix of the two applications mentioned above (all applications), where VBR-services
are re-multiplexed with CBR-services. All rates are reduced and statistically
multiplexed in order to achieve better network utilization.
- IP-Streaming of MPEG-2 services: Rate-limiting for single service, enabling a CBR on
a specified bit-rate, typically significantly lower than both average rates of incoming
CBR or VBR-programs.
- Ensuring compliance with Service Level Agreements between network operators and
content providers by limiting/controlling the rates of single programs re-transmitted
over a DTV-network.
3. THE TECHNOLOGIES OF TRANSRATING
Transrating focuses on reducing the rate of a single service and/or a multiplex of services. A
service consists of many elements, such as SI/PSI data, audio, VBI, other data and finally
video. Video is seen from a bit-rate perspective as the largest consumer of bandwidth.
Reducing video-rates is the only real alternative that will have a significant effect on the total
service bandwidth. Transrating works specifically by reducing the video encoding rates and
it does affect the rest of the components in a service. A transrating device typically makes
use of one of two core processes. The first one, “time-shifting of MPEG-2 packets,” involves
the “smoothing” of the total Transport Stream rate without altering the actual video encoding
rates of the individual services. The second process is where rate reduction occurs for
individual services.
Time-shifting of MPEG-2 data packets
One of the main problems faced is the re-multiplexing of
services, previously encoded using Variable Bit-Rates (VBR). A
relatively easy way around this is to advance and delay MPEG-
2 packets in order to “equalize” the bit-rate level of the total
stream. Figure 1 shows five different services, encoded using
variable bit-rates. Horizontal arrows show how MPEG-2
packets can be advanced or delayed in the time-domain to
decrease the peaks.
Of course, there are limits as to how much such a process can
reduce the rate-peaks. These limits are defined by the Virtual
Buffer Verifier specifications (VBV) (1) in the MPEG-2
compression standard. The VBV specifies how much data the Figure 1 - Time-shifting of
decoder of the programs must be able to store in the local data-packets in an
memory before it converts the data into base band video. It is MPEG-2 VBR stream
based on the fact that data must arrive at the decoder before it
has to be removed from the buffer for decoding.
This type of technology is only really effective in two types of applications. Single programs
must be VBR encoded, otherwise there is no point in shuffling the data packets. If a number
of programs use high rates simultaneously, shifting does not help. Time shifting only lowers
the maximum rate, not the average rate. Because the transrater must have finite delay and
cannot send packets it has not received, the maximum time that a packet can be advanced
is limited.
Time-shifting technology does not guarantee any output bandwidth. Also it does not help the
operator to adjust the bandwidth-consumption of the services to where the potential profit is
most likely to be found. Nevertheless it works well for the preparation of VBR-feeds for the
process of transrating, which will be explained next.
Transrating – Open Loop
The tools of rate-reduction are revealed when looking into the fundamentals of the encoding
process. A typical encoder can compress the video to a digital bit-rate specified by the
operator. The same tools can be used for transrating and therefore the basic processes of
encoding need to be explained.
4. The basics of MPEG-2 video encoding
The major steps in MPEG-2 encoding are:
- Dividing the pixels into macro-blocks of 8x8 pixels
- Perform motion compensation by identifying temporal
redundancy.
- Find spatial redundancy with DCT-encoding. Until now no
reduction of the video information has been made and the
process is therefore reversible.
- The reduction of the information so far generated is done
with the quantization. This process removes details in the
picture, which is often in the high frequencies. In areas
with high spatial activity (pencil drawings, letters in a book
or other high contrast areas) the details are both high and
low frequency, and low frequencies will be reduced. The
logic of this is based on the limited perception of the
human brain, which filters out this information. The
quantification removes content in the picture, which will
give the user none or limited perception of degradation
anyway.
- Variable Length Encoding is the process of reducing the
mathematical redundancy.
The output of the variable length encoding is the core content of
the video in the transport stream. In addition to these principles, Figure 2 - The MPEG-2
each picture is treated as either an I, P or B-frame which will be Encoding process
explained below.
The schematics of open loop transrating
The obvious way to obtain actual rate-reductions on a video-service is to perform re-
quantization. The process of quantization is (as seen in Figure 2) integral to MPEG-2
encoding. When a transrater device performs this process, it partially inverse encodes,
including the process of inverse quantization, and then decides on an increased level of
quantization, resulting in higher compression rates. On a flow chart this can be explained
like Figure 3 below.
Figure 3 - The Open Loop Algorithm
Such a transrating process reduces the bit-rate of the video, but not without trade-offs in the
video quality. The problem is this relatively simple algorithm works in an “open loop” without
any information about what kind of effect it has on the image while transrating the individual
pictures in the stream. When making a transrater device using this algorithm, a defensive
rate-reduction strategy must be followed to avoid a picture breakdown from visual artifacts in
the video. This will mean limitations to the device, since all streams carry different content
5. encoded by different encoders. It will only be capable of a certain degree of rate-reduction.
The consequences for exceeding the maximum level for quantification will be very easy to
see for the end-consumer – the subscriber. The complex referenced structure of the MPEG
stream may cause programs to break up and show severe artifacts for up to a half second.
The risks when reducing the rates
The problems of open loop transrating can be explained based on the basic terminology of a
transport stream, the so-called Group of Pictures (GOP). The stream consists of I, B and P-
pictures, typically in a syntax as seen below on Figure 4, showing a 15,2 GOP (15 pictures
in total, 2 B-pictures between each P or I frame). The arrows indicate how the P-pictures are
related to either the previous P-picture or the I-picture. The B-pictures are related to the I or
P picture before it and, if present, the P-picture after.
Figure 4 - The structure of the GOP-sequence anchor picture references
As mentioned previously, the re-quantification removes information and compromises the
picture quality. This will be referred to as introducing an error. Depending on the GOP-
sequence, there is a difference where this error is introduced. Errors introduced on B-
pictures will have little effect. It will only be shown for 1 or 2 frames, out of 25 frames during
one second. If the error is a block-artifact, the visual impact will be a short pulse. The
situation is however more serious if errors are applied on the P or I-pictures.
The I and the P-pictures are the anchor-pictures of the GOP, since the following P-pictures
always refer to the previous I or P-picture. So, if there is an error applied to one of the early
P-pictures in the GOP – or even the I-picture, which is starting the GOP – the error
continues throughout the GOP-sequence. If introduced on an I-picture, the block artifact will
be present in the video for approximate half a second, depending on the length of the GOP.
This problem for the video quality, can also been seen in other ways.
If an error on the I-picture is introduced due to re-quantization, it is easy to imagine that
errors also can be applied to the following P-pictures. The result is shown in Figure 5.
Figure 5 - The development of the error throughout the GOP (error-drifting)
Simply put, one bad decision at the start of the GOP is compounded by another one. The
sum of these errors, “breathing”, is perceived as degradation of picture quality throughout
the GOP, until a new GOP starts with an I-frame. The cycle repeats every half a second.
Other factors that influence the quality level of Open Loop Transrating are mainly related to
how the actual decisions on the level of re-quantification are made, mainly on the B-pictures.
6. How to apply open loop transrating
Open loop transrating is a method of actually lowering the bit-rate of video services, that
should be followed by the time-shifting of the MPEG-2 data packets.
Since I and P pictures are risky to re-quantify, the transrater device could limit its processing
range to only include B-pictures. That will prevent any major introduction of artifacts in the
decoded video. It will have a rate-reducing effect since most of the pictures in the stream are
B-pictures. On the other hand, the biggest pictures by far in the GOP – the ones carrying
most information – are the I and P pictures. Only very limited re-quantification can be done
on these pictures in order to avoid severe block artifacts in the decoded video.
In conclusion, the typical rate-reduction made by transrater devices utilizing open loop
transrating is app 10-20%, depending on how the GOP previously has been encoded. As an
example, better quality encoders have better motion-estimation performance, meaning that
the B-pictures already will be small in size. In that case, the benefit of transrating is reduced.
Since this does not fulfill all the business-objectives mentioned in the introduction, another
type of transrating can be applied, solving these problems by enabling a higher level of rate-
reduction. This is in this paper referred to as “Closed Loop Transrating”.
TRANSRATING - CLOSED LOOP
The name of this kind of transrating indicates that the main differentiator from the open loop
type is a “learning” loop. This algorithm aims to overcome the problems of the Open Loop
Transrating, which in effect means that the total rate-reductions of a stream will be greater.
The schematics of Closed Loop Transrating
Closed Loop is built on the Open Loop schematics, but it has two “learning” loops added:
- First of all, the re-quantification of the single macro-blocks is done based on
measurements of the output picture quality for the single picture. In order to optimize
the level of re-quantification, the first step is to actually measure the error applied.
Secondly, the error detection must have a valid basis. It is very difficult to determine.
Figure 6 - The Closed Loop Transrating algorithm. Based on Assuncao et al (2)
7. Compared to the Open Loop Transrating algorithm it can be seen in Figure 6 that the basic
process is unchanged, but that several loops have been added. The box named “REF” is
the reference picture comparison between the previous I or P picture and the picture in
question. The function of REF is to store the error, which is feed back into the transrating
process for the next anchor picture.
Note that this process also makes use of both DCT encoding and decoding (DCT encoding
and inverse DCT encoding). According to the physics of the video encoding scheme
described previously, this takes the MPEG-2 stream almost back to the pixel-domain. This is
done to ensure use of the reference-picture quality measurements. The error applied on the
actual anchor picture is known and transferred into the process of re-quantifying the next
anchor picture. In this way the re-quantification of the anchor picture will be able to
compensate for the error.
Although this is more complicated, it does solve some of the problems occurring when Open
Loop Transrating is applied. It cannot change the fact, that reducing the picture size will
reduce the quality. This process is however capable of, first of all, knowing the exact level of
error that has been introduced on each frame and second , compensating for this error in
the next anchor picture.
Figure 7 shows how the Closed Loop Transrating algorithm overcomes the risk of breathing,
since it monitors the error included on a frame-by-frame reference.
Figure 7 - Eliminating the error developing throughout the GOP
Even though errors still are introduced in each picture, the error will be compensated for in
the next anchor picture. As a consequence, the total error throughout the GOP-sequence
will not exceed the error introduced in the first anchor-picture. That has basically been
determined by the rate-reductions requested by the operator.
In short, the advantage of the Closed Loop Transrating is built on these two processes:
- The level of re-quantifications on the individual anchor picture is being applied while
monitoring the actual errors applied on the picture.
- All errors applied as a natural consequence of the re-quantification are eliminated in
the next anchor-picture, thereby eliminating the risk of an exponentially developing
error throughout the GOP, which could cause video quality problems.
How to apply Closed Loop Transrating
The main advantage of this algorithm compared to the Open Loop Transrating is that it
allows significant rate-reductions of the I and P pictures, without risking severe impairments
in the video quality. The algorithm will, based on the inputs from the user itself, determine
how large the rate-reductions on the I and P pictures can be.
Based on simulations with different types of stream, the Closed Loop Algorithm has shown
up to 50% video bit-rate reductions, without causing significant blocking artifacts on the
8. decoded video image. Compared to the Open Loop algorithm, the further reduction is
achieved on the I and P pictures, while the B pictures remain as compressed as they would
be using conventional Open Loop transrating.
A particular benefit of the Closed Loop algorithm is that it will be able to achieve significant
rate-reductions also on high-quality encoders, which outputs smaller B pictures than
conventional encoders.
It is also clear that the complexity of the algorithm is greater than Open Loop. From an
implementation standpoint that means that there is a need for more powerful hardware. The
“learning” loops require extra processing compared to the Open Loop Transrating. Tests
have shown that if a Closed Loop Transrating process carried out on I and P pictures takes
100% processing power, the Open Loop needs approximately 40-50%. In effect, Closed
Loop Transrating requires twice as much processing hardware as the Open Loop
Transrating algorithm.
CONCLUSION
This paper has presented three different processes relevant to transrating. The conclusion is
as follows:
- Time-shifting of MPEG-2 data packets is a necessary tool for preparing all transrating
processes, as long as the incoming video-streams are in Variable Bit-Rate mode
(VBR). This will reduce the potential peaks when re-multiplexing independently
encoded VBR programs, but it will not guarantee a specific bit-rate and will still
require a margin of error to prevent overflow in the streams coming out of the
headend.
- Open Loop Transrating is based on re-quantification of the video and results in a
rate-reduction of the encoded video streams. Due to the linear process of re-
quantification, the process can only output a certain bit-rate reduction, varying from
10-20% max on common streams.
- Closed Loop Transrating is similar to the Open Loop transrating, but it uses a double
“learning” loop, based on inputs from measurement of the video quality of the
specific, transrated frame and the frames before it. This complex algorithm enables
up to 50% video bit-rate reductions.
Transrating is a strong alternative to decoding/re-encoding programs, which offers cost-
effective rate-reductions with a minimum loss of video quality, especially when Closed Loop
Transrating is applied.
REFERENCES
1. ISO/IEC 13818-2 (1995) Generic Coding of Moving Pictures and Associated Audio
Information: Video. Appendix C: Video Buffering Verifier
2. A.A. Assuncao, Pedro and Ghanbari, Mohammed; Transcoding of MPEG-2 video in the
frequency domain; Department of Electronic Systems Engineering, University of Essex,
IEEE 1997