Bandwidth Scaling for AI Interconnect - More Wavelengths versusMore Fiber?

High-speed networks with large delays present a unique environment where TCP may have a problem utilizing the full bandwidth. Several congestion control proposals have been suggested to remedy this problem. The existing protocols consider mainly two properties: TCP friendliness and bandwidth scalability. That is, a protocol should not take away too much bandwidth from standard TCP flows while utilizing the full bandwidth of high-speed networks. This work presents another important constraint, namely, RTT (round trip time) unfairness where competing flows with different RTTs may consume vastly unfair bandwidth shares. Existing schemes have a severe RTT unfairness problem because the congestion window increase rate gets larger as the window grows ironically the very reason that makes them more scalable. RTT unfairness for high-speed networks occurs distinctly with drop tail routers for flows with large congestion windows where packet loss can be highly synchronized. After identifying the RTT unfairness problem of existing protocols, This work presents a new congestion control scheme that alleviates RTT unfairness while supporting TCP friendliness and bandwidth scalability. The proposed congestion control algorithm uses two window size control policies called additive increase and binary search increase. When the congestion window is large, additive increase with a large increment ensures square RTT unfairness as well as good scalability. Under small congestion windows, binary search increase supports TCP friendliness. The simulation results confirm these properties of the protocol.

The co-packaging of optics and electronics provides a potential path forward to achieving beyond 50 Tbps top of rack switch packages. In a co-packaged design, the scaling of bandwidth, cost, and energy is governed by the number of optical transceivers (TxRx) per package as opposed to transistor shrink. Due to the large footprint of optical components relative to their electronic counterparts, the vertical stacking of optical TxRx chips in a co-packaged optics design will become a necessity. As a result, development of efficient, dense, and wide alignment tolerance chip-to-chip optical couplers will be an enabling technology for continued TxRx scaling. In this paper, we propose a novel scheme to vertically couple into standard 220 nm silicon on insulator waveguides from 220 nm silicon nitride on glass waveguides using overlapping, inverse double tapers. Simulation results using Lumerical’s 3D Finite Difference Time Domain solver are presented, demonstrating insertion losses below -0.13 dB for an inter-chip spacing of 1 μm; 1 dB vertical and lateral alignment tolerances of approximately 2.6 μm and ± 2.8 μm, respectively; a greater than 300 nm 1 dB bandwidth; and 1 dB twist and tilt tolerances of approximately ± 2.3 degrees and 0.4 degrees, respectively. These results demonstrate the potential of our coupler for use in co-packaged designs requiring high performance, high density, CMOS compatible out of plane optical connections.

A brushless DC motor (BLDCM) is a typical nonlinear and strong coupling plant. Unknown, complex, and time-varying external disturbances make the control of BLDCM be a big challenge. PID control, passive receiving tracking errors, cannot satisfy the requirements of high-precision speed control. To address it, a scalable observer bandwidth based active disturbance rejection control (SOADRC) is proposed. By adjusting observer bandwidth, high-precision speed control can be achieved. A numerical model of BLDCM is built to verify the SOADRC. Simulation results show that SOADRC is able to get better speed control accuracy.

Future bandwidth needs force the development of optical interconnects for data transmission at board level. The aim is to develop technology solutions for embedded optical architectures in data center and network systems to allow significant reduction in power consumption, increased energy efficiency, system density and bandwidth scalability, which is currently unfeasible in today's copper driven systems. A passive optical backplane comprising a printed circuit board (PCB) with integrated optical waveguides and standardized pluggable optical coupling interfaces to the line cards is proposed for future data communication systems. Our activities are focused on embedded glass-based electro-optical circuit boards (EOCB) with integrated gradient-index multimode waveguides, which exhibit very low propagation loss at key optical communication wavelengths. A pre-processed glass panel with planar integrated optical waveguides is embedded into a backplane by using proven industrial PCB processes. A pluggable optical connector and receptacle system is designed for connecting the optical fiber links of the line cards with the optical waveguide circuit of the backplane.

The increase in video traffic is causing considerable load on the already-limited cellular network resources. This poses a new landscape of challenges to service and network providers in distributing bandwidth-hungry services such as Multi-View Video (MVV). A paradigm to circumvent this challenge is to allow direct communication between closely located User Equipment (UE) without data routing via the radio and core network components. Device-to-Device (D2D) capable networks promise scalability and performance improvements in utilizing scarce resources. In this work, a mode selection strategy at the Base Station (BS) is proposed. It considers segment availability information at the UE to decide whether the Mobile User (MU) receives dedicated resources via a traditional Content Delivery Network (CDN) or reuse existing resources via the D2D communications channel. A novel architecture whereby the BS in the Radio Access Network (RAN) has video buffers, with caching maps which are aware of the video preferences of users in cell sites is analyzed. This presents the foundation to build a D2D-enhanced live MVV streaming system with no additional infrastructure for mobile devices. Simulation results provide guidelines for maximizing view and user diversity as well as bandwidth scalability in Long-Term Evolution (LTE) networks in order to achieve high system resource utilization whilst mitigating load on the network.

The congestion control mechanisms of the current standard TCP can encounter problems in high-speed wide area networks due to its slow response with a large congestion window. Several congestion control proposals have already been suggested to solve this problem and mainly consider two properties: TCP friendliness and bandwidth scalability, to ensure that a protocol does not take away too much bandwidth from TCP, while utilizing the full bandwidth of high speed networks. However, further studies on the TCP friendliness of high-speed TCP are still needed. Recent studies have pointed out that existing schemes have a severe RTT unfairness problem, where competing flows with different RTTs can consume considerable unfair bandwidth shares. Burstiness is one of the main reasons behind such problems. As the congestion window achieved by a high-speed TCP connection can be quite large, there is a strong possibility that the sender will transmit a large burst of packets. As such, the current congestion control mechanisms of high-speed TCP can lead to bursty traffic flows in high speed networks, with a negative impact on both TCP friendliness and RTT unfairness. The proposed solution to these problems is to evenly space, or pace packets sent into the network over an entire round-trip time, so that packets are not sent in a burst. Accordingly, the current paper evaluates this approach with a high bandwidth-delay product network and shows that pacing offers better TCP friendliness and RTT fairness without degrading the bandwidth scalability.

One of the recent trends in communication is Free Space optics (FSO). These systems provide solution to the bottleneck problems of the RF communication systems, like Bandwidth scalability, cost effectiveness and portability. But, few of the challenging tasks in implementing this system are Line of sight, atmospheric conditions like fog, snow…etc. Actually the Coherent Optical OFDM orthogonal Frequency Division Multiplexing (CO OFDM) system in FSO is capable of transmitting data upto 0.22 km in dense fog weather conditions. This can be improved by reducing the fading due to free space. One of the efficient ways to reduce fading effects is spatial diversity. Spatial diversity will improve the received power and ensures the low Bit Error Rate (BER). The aim of this paper is to incorporate this Diversity technique into the FSO OFDM, so that an enhanced FSO link is established. The performance analysis is carried out in terms of BER, Constellation diagram, received power.

TCP-friendly rate control (TFRC), an equation-based congestion control protocol, has been a promising alternative to TCP for streaming multimedia applications. However, TFRC using the TCP response function has the same bandwidth scalability problem as TCP in high-speed long-distance networks. In this paper, we propose high-speed equation-based rate control (HERC) for streaming multimedia transfer over high-speed long-distance networks, as an extension of TFRC by replacing the TCP response function with a high-speed response function. Our result indicates that while HERC achieves better bandwidth scalability than TFRC, it has worse smoothness than TFRC with the same loss history size. We prove that the smoothness index measured by the coefficient of variation of sending rates is approximately equal to 1.09d//spl radic/LCoV[/spl theta/] where d is the exponent parameter of the response function, L is the loss history size, and CoV[/spl theta/] is the coefficient of variation of loss intervals. Furthermore, we show that by setting L to 32d/sup 2/, HERC with any value of d can achieve the same smoothness as TFRC under the same loss interval distribution.

A new look-up table based behavioral model for dynamic nonlinear power amplifiers is proposed. This model labelled as hybrid look-up tables model is based on the combination of a memoryless look-up table sub-model and a nested look-up tables one. It is demonstrated that the proposed model circumvents the computational complexity associated with the parameters identification in analytically defined behavioral models. Moreover, the proposed model reduces the size of the standalone nested look-up tables model by approximately 80% while maintaining its accuracy. Furthermore, a novel slew-rate based trimming and indexing technique to reduce the nested look-up tables model size is developed and corroborated experimentally. Additionally, the two-box structure of the hybrid look-up tables model makes it suitable for bandwidth scalability. Experimental validation using LTE-advanced test signals with up to 120MHz bandwidth demonstrates the ability of the proposed hybrid look-up tables model to be bandwidth scalable with less than 0.5dB degradation in the normalized mean-squared error.

High speed networks with high delays present a unique environment where TCP can not efficiently utilize the full bandwidth. In this paper, we present a simple and novel Time Based Congestion Control scheme (TBCC for short) that alleviates RTT unfairness while supporting TCP friendliness and bandwidth scalability. We provide the general framework for TBCC protocol modifications. Then, using a simplified design, we illustrate the scalability of TBCC for improving the efficiency, without sacrificing the fairness of TCP. We assess the RTT unfairness of TBCC and show that the RTT unfairness of TBCC can be firmly guaranteed to be no worse than that of unmodified TCP. Compared with similar protocols such as HSTCP and STCP, TBCC can guarantee the bandwidth scalability and achieve better fairness. The flexible framework of TBCC provides a whole class of design options for improving the performance of TCP in high speed networks.KeywordsCongestion ControlCongestion WindowHigh Speed NetworkFeedback FunctionCongestion AvoidanceThese keywords were added by machine and not by the authors. This process is experimental and the keywords may be updated as the learning algorithm improves.

A low-power dual-standard video decoder has been developed for mobile applications. It supports MPEG-2 SP@ML and H.264/AVC BL@L4 video decoding in a single chip and features a scalable architecture to reach area/power efficiency. This chip integrates diverse algorithms of MPEG-2 and H.264/AVC to reduce silicon area. Three low-power techniques are proposed. First, a domain-pipelined scalability (DPS) technique is used to optimize the pipelined structure according to the number of processing cycles. Second, bandwidth scalability is implemented via a line-pixel-lookahead (LPL) scheme to improve the external bandwidth and reduce the internal memory size, leading to 51% of memory power reduction compared to a conventional design. Third, low-power motion compensation and deblocking filter are designed to reduce the operating frequency without degrading system performance. A test chip is fabricated in a 0.18mum one-poly six-metal CMOS technology with an area of 15.21 mm <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">2</sup> . For mobile applications, H.264/AVC and MPEG-2 video decoding of quarter-common intermediate format (QCIF) sequences at 15 frames per second are achieved at 1.15 MHz clock frequency with power dissipation of 125 muW and 108 muW, respectively, at 1V supply voltage

This paper proposes a bandwidth-scalable coding scheme based on the G.729 standard as a base layer coder. In the scheme, according to the channel conditions, the output speech of the decoder can be selected to be narrowband (4-kHz bandwidth) or wideband (8-kHz bandwidth). The proposed scheme consists of two layers: base and enhancement. The base coder uses the G.729 algorithm to encode narrowband speech. The enhancement coder is based on a full-band CELP model and it encodes wideband speech while making use of the available base layer information. Two bandwidth-scalable coders are designed: one is scalable with the 8 kbit/s G.729 base coder and another with the 6.4 kbit/s G.729 (Annex D) base coder. Subjective tests show that, for wideband speech, the proposed coders at 16 kbit/s achieve better performance than the 16 kbit/s MPEG-4 CELP with bandwidth scalability.

We introduce the concept of that fully exploit the potential of edge computing and have a deeply symbiotic relationship with it. Such an application is custom-designed to take advantage of one or more of the unique attributes of edge computing such as (a) bandwidth scalability, (b) low-latency offload, (c) privacy-preserving denaturing, and (d) WAN-failure resiliency. The application may also contribute to scalability through adaptation to reduce offered load. We contrast edge-native applications with shallower uses of edge computing in a taxonomy that spans edge-enhanced, device-native applications, edge-accelerated, cloud-native applications, and device-only applications. We close with a case study that illustrates these concepts in the context of cognitive assistance for automotive safety.

The past two decades have witnessed a rapid expansion within the telecommunications industry. This growth has been primarily motivated by the proliferation of digital communication systems and services which have become easily available through wired and wireless systems. Current research trends involve the integration of speech, audio, video and data channels into true multimedia communications over fixed and mobile networks. However, while the available bandwidth in wired terrestrial networks is relatively cheap and expandable, it becomes a limited resource in satellite and cellular-radio systems. In order to accommodate an ever growing number of users while maintaining high quality and low operational costs, it is necessary to maximise spectral efficiency. This has given rise to the development of high rate compression techniques with the ability to adapt to a broad class of input signals and to varying network resources. The research carried out in this thesis has mainly focused on the design of a single algorithm for compressing speech and audio signals sampled at different rates. The algorithms are based on the analysis-by-synthesis linear prediction coding (AbS-LPC) scheme, which has been widely employed in various speech coding standards. However, this bit rate reduction technique is based on the speech production mechanism and as such provides a rigid structure which presents a major limitation for audio coding. In order to improve the audio quality at low rates and to compensate for the errors incurred by the linear prediction during segments of high transitions, the algorithms employ an efficient pulse excitation structure which represents the short innovation sequences with sparse unit magnitude pulses. The scheme proposed for the compression of telephone bandwidth speech and audio signals at 12kb/s achieves similar quality to the G.728 coder at 16kb/s and higher audio quality than the GSM-EFR standard at 12.2kb/s. Wideband speech and audio coding schemes have been designed using both the fullband approach at bit rates of 17 and 19kb/s and also the split band technique at a bit rate of 20kb/s. The perceptual quality is comparable to the G.722 coder operating at 48kb/s. The subband decomposition technique is also adapted to code speech and audio signals sampled at 32kHz. The quality of the coder at 28kb/s is similar to the quality achieved by the MP3 coder at 32kb/s. The algorithm also provides bandwidth and bit rate scalability ranging from 12 to 64kb/s, making it ideal for deployment in rate-adaptive communication systems.

This paper proposes physical channel structures and a cell search method for OFDM based radio access in the evolved UTRA (UMTS Terrestrial Radio Access) downlink, which supports multiple scalable transmission bandwidths from 1.25 to 20 MHz, in the proposed physical channel structures, the central sub-carrier of the OFDM signal is located on the frequency satisfying the 200-kHz raster condition regardless of the transmission bandwidth of the cell site. Moreover, the synchronization channel (SCH) and broadcast channel (BCH), which are necessary for cell search, are transmitted from the central part of the entire transmission spectrum with a fixed bandwidth. In the proposed cell search method, a user equipment (UE) acquires the target cell in the cell search process in the initial or connected mode employing the SCH and possibly the reference signal, which are transmitted from the central part of the given transmission bandwidth. Alter detecting the target cell, the UE decodes the common control information through the BCH, which is transmitted at the same frequency as the SCH, and identifies the transmission bandwidth of the cell to be connected. Computer simulations show the fast cell search time performance using the proposed SCH structure and the cell search method.