차세대 자동차용 카메라 데이터 통신을 위한 비대칭 동시 양방향 송수신기의 설계

학위논문(박사) -- 서울대학교대학원 : 공과대학 전기·정보공학부, 2022.2. 정덕균.본 학위 논문에서는 차세대 자동차용 카메라 링크를 위해 높은 속도의 4레벨 펄스 진폭 변조 신호와 낮은 속도의 2레벨 펄스 진폭 변조 신호를 통신하는 비대칭 동시 양방향 송수신기의 설계 기술에 대해 제안하고 검증되었다. 첫번째 프로토타입 설계에서는, 10B6Q 직류 밸런스 코드를 탑재한 4레벨 펄스 진폭 변조 송신기와 고정된 데이터와 참조 레벨을 가지는 4레벨 펄스 진폭 변조 적응형 수신기에 대한 내용이 기술되었다. 4레벨 펄스 진폭 변조 송신기에서는 교류 연결 링크 시스템에 대응하기 위한 면적 및 전력 효율성이 좋은 10B6Q 코드가 제안되었다. 이 코드는 직류 밸런스를 맞추고 연속적으로 같은 심볼을 가지는 길이를 6개로 제한 시킨다. 비록 여기서는 입력 데이터 길이 10비트를 사용하였지만, 제안된 기술은 카메라의 다양한 데이터 타입에 대응할 수 있도록 입력 데이터 길이에 대한 확장성을 가진다. 반면, 4레벨 펄스 진폭 변조 적응형 수신기에서는, 샘플러의 옵셋을 최적으로 제거하여 더 낮은 비트에러율을 얻기 위해서, 기존의 데이터 및 참조 레벨을 조절하는 대신, 이 레벨들은 고정시키고 가변 게인 증폭기를 적응형으로 조절하도록 하였다. 상기 10B6Q 코드 및 고정 데이터 및 참조레벨 기술을 가진 프로토타입 칩들은 40 나노미터 상호보완형 메탈 산화 반도체 공정으로 제작되었고 칩 온 보드 형태로 평가되었다. 10B6Q 코드는 합성 게이트 숫자는 645개와 함께 단 0.0009 mm2 의 면적 만을 차지한다. 또한, 667 MHz 동작 주파수에서 단 0.23 mW 의 전력을 소모한다. 10B6Q 코드를 탑재한 송신기에서 8-Gb/s 4레벨 펄스 진폭 변조 신호를 고정 데이터 및 참조 레벨을 가지는 적응형 수신기로 12-m 케이블 (22-dB 채널 로스) 을 통해서 보낸 결과 최소 비트 에러율 108 을 달성하였고, 비트 에러율 105 에서는 아이 마진이 0.15 UI x 50 mV 보다 크게 측정되었다. 송수신기를 합친 전력 소모는 65.2 mW (PLL 제외) 이고, 성과의 대표수치는 0.37 pJ/b/dB 를 보여주었다. 첫번째 프로토타입 설계을 포함하여 개선된 두번째 프로토타입 설계에서는, 12-Gb/s 4레벨 펄스 진폭 변조 정방향 채널 신호와 125-Mb/s 2레벨 펄스 진폭 변조 역방향 채널 신호를 탑재한 비대칭 동시 양방향 송수신기에 대해 기술되고 검증되었다. 제안된 넓은 선형 범위를 가지는 하이브리드는 gmC 저대역 통과 필터와 에코 제거기와 함께 아웃바운드 신호를 24 dB 이상 효율적으로 감소시켰다. 또한, 넓은 선형 범위를 가지는 하이브리드와 함께 게인 감소기를 형성하게 되는 선형 범위 증폭기를 통해 4레벨 펄스 진폭 변조 신호의 선형성과 진폭의 트레이드 오프 관계를 깨는 것이 가능하였다. 동시 양방향 송수신기 칩은 40 나노미터 상호보완형 메탈 산화 반도체 공정으로 제작되었다. 상기 설계 기술들을 이용하여, 4레벨 펄스 진폭 변조 및 2레벨 펄스 진폭 변조 송수신기 모두 5m 채널 (채널 로스 15.9 dB) 에서 1E-12 보다 낮은 비트 에러율을 달성하였고, 총 78.4 mW 의 전력 소모를 기록하였다. 종합적인 송수신기는 성과 대표지표로 0.41 pJ/b/dB 와 함께 동시 양방향 통신 아래에서 4레벨 펄스 진폭 변조 신호 및 2레벨 펄스 진폭 변조 신호 각각에서 아이 마진 0.15 UI 와 0.57 UI 를 달성하였다. 이 수치는 성과 대표지표 0.5 이하를 가지는 기존 동시 양방향 송수신기와의 비교에서 최고의 아이 마진을 기록하였다.In this dissertation, design techniques of a highly asymmetric simultaneous bidirectional (SB) transceivers with high-speed PAM-4 and low-speed PAM-2 signals are proposed and demonstrated for the next-generation automotive camera link. In a first prototype design, a PAM-4 transmitter with 10B6Q DC balance code and a PAM-4 adaptive receiver with fixed data and threshold levels (dtLevs) are presented. In PAM-4 transmitter, an area- and power-efficient 10B6Q code for an AC coupled link system that guarantees DC balance and limited run length of six is proposed. Although the input data width of 10 bits is used here, the proposed scheme has an extensibility for the input data width to cover various data types of the camera. On the other hand, in the PAM-4 adaptive receiver, to optimally cancel the sampler offset for a lower BER, instead of adjusting dtLevs, the gain of a programmable gain amplifier is adjusted adaptively under fixed dtLevs. The prototype chips including above proposed 10B6Q code and fixed dtLevs are fabricated in 40-nm CMOS technology and tested in chip-on-board assembly. The 10B6Q code only occupies an active area of 0.0009 mm2 with a synthesized gate count of 645. It also consumes 0.23 mW at the operating clock frequency of 667 MHz. The transmitter with 10B6Q code delivers 8-Gb/s PAM-4 signal to the adaptive receiver using fixed dtLevs through a lossy 12-m cable (22-dB channel loss) with a BER of 1E-8, and the eye margin larger than 0.15 UI x 50 mV is measured for a BER of 1E-5. The proto-type chips consume 65.2 mW (excluding PLL), exhibiting an FoM of 0.37 pJ/b/dB. In a second prototype design advanced from the first prototypes, An asymmetric SB transceivers incorporating a 12-Gb/s PAM-4 forward channel and a 125-Mb/s PAM-2 back channel are presented and demonstrated. The proposed wide linear range (WLR) hybrid combined with a gmC low-pass filter and an echo canceller effectively suppresses the outbound signals by more than 24dB. In addition, linear range enhancer which forms a gain attenuator with WLR hybrid breaks the trade-off between the linearity and the amplitude of the PAM-4 signal. The SB transceiver chips are separately fabricated in 40-nm CMOS technology. Using above design techniques, both PAM-4 and PAM-2 SB transceivers achieve BER less than 1E-12 over a 5-m channel (15.9 dB channel loss), consuming 78.4 mW. The overall transceivers achieve an FoM of 0.41 pJ/b/dB and eye margin (at BER of 1E-12) of 0.15 UI and 0.57 UI for the forward PAM-4 and back PAM-2 signals, respectively, under SB communication. This is the best eye margin compared to the prior art SB transceivers with an FoM less than 0.5.CHAPTER 1 INTRODUCTION 1 1.1 MOTIVATION 1 1.2 DISSERTATION ORGANIZATION 4 CHAPTER 2 BACKGROUND ON AUTOMOTIVE CAMERA LINK 6 2.1 OVERVIEW 6 2.2 SYSTEM REQUIREMENTS 10 2.2.1 CHANNEL 10 2.2.2 POWER OVER DIFFERENTIAL LINE (PODL) 12 2.2.3 AC COUPLING AND DC BALANCE CODE 15 2.2.4 SIMULTANEOUS BIDIRECTIONAL COMMUNICATION 18 2.2.4.1 HYBRID 18 2.2.4.2 ECHO CANCELLER 20 2.2.5 ADAPTIVE RECEIVE EQUALIZATION 22 CHAPTER 3 AREA AND POWER EFFICIENT 10B6Q ENCODER FOR DC BALANCE 25 3.1 INTRODUCTION 25 3.2 PRIOR WORKS 28 3.3 PROPOSED AREA- AND POWER-EFFICIENT 10B6Q PAM-4 CODER 30 3.4 DESIGN OF THE 10B6Q CODE 33 3.4.1 PAM-4 DC BALANCE 35 3.4.2 PAM-4 TRANSITION DENSITY 35 3.4.3 10B6Q DECODER 37 3.5 IMPLEMENTATION AND MEASUREMENT RESULTS 40 CHAPTER 4 PAM-4 TRANSMITTER AND ADAPTIVE RECEIVER WITH FIXED DATA AND THRESHOLD LEVELS 45 4.1 INTRODUCTION 45 4.2 PRIOR WORKS 47 4.3 ARCHITECTURE AND IMPLEMENTATION 49 4.2.1 PAM-4 TRANSMITTER 49 4.2.2 PAM-4 ADAPTIVE RECEIVER 52 4.3 MEASUREMENT RESULTS 62 CHAPTER 5 ASYMMETRIC SIMULTANEOUS BIDIRECTIONAL TRANSCEIVERS USING WIDE LINEAR RANGE HYBRID 68 5.1 INTRODUCTION 68 5.2 PRIOR WORKS 70 5.3 WIDE LINEAR RANGE (WLR) HYBRID 75 5.3 IMPLEMENTATION 78 5.3.1 SERIALIZER (SER) DESIGN 78 5.3.2 DESERIALIZER (DES) DESIGN 79 5.4 HALF CIRCUIT ANALYSIS OF WLR HYBRID AND LRE 82 5.5 MEASUREMENT RESULTS 88 CHAPTER 6 CONCLUSION 97 BIBLIOGRAPHY 99 초 록 106박

Adaptation algorithms for data echo cancellation using nonquadratic cost functions

Adaptation algorithms for data echo cancellation using nonquadratic cost function

Estimation and detection of transmission line characteristics in the copper access network

The copper access-network operators face the challenge of developing and maintaining cost-effective digital subscriber line (DSL) services that are competitive to other broadband access technologies. The way forward is dictated by the demand of ever increasing data rates on the twisted-pair copper lines. To meet this demand, a relocation of the DSL transceivers in cabinets closer to the customers are often necessary combined with a joint expansion of the accompanying optical-fiber backhaul network. The equipment of the next generation copper network are therefore becoming more scattered and geographically distributed, which increases the requirements of automated line qualification with fault detection and localization. This scenario is addressed in the first five papers of this dissertation where the focus is on estimation and detection of transmission line characteristics in the copper access network. The developed methods apply model-based optimization with an emphasis on using low-order modeling and a priori information of the given problem. More specifically, in Paper I a low-order and causal cable model is derived based on the Hilbert transform. This model is successfully applied in three contributions of this dissertation. In Paper II, a class of low-complexity unbiased estimators for the frequency-dependent characteristic impedance is presented that uses one-port measurements only. The so obtained characteristic impedance paves the way for enhanced time domain reflectometry (a.k.a. TDR) on twisted-pair lines. In Paper III, the problem of estimating a nonhomogeneous and dispersive transmission line is investigated and a space-frequency optimization approach is developed for the DSL application. The accompanying analysis shows which parameters are of interest to estimate and further suggests the introduction of the concept capacitive length that overcomes the necessity of a priori knowledge of the physical line length. In Paper IV, two methods are developed for detection and localization of load coils present in so-called loaded lines. In Paper V, line topology identification is addressed with varying degree of a priori information. In doing so, a model-based optimization approach is employed that utilizes multi-objective evolutionary computation based on one/two-port measurements. A complement to transceiver relocation that potentially enhances the total data throughput in the copper access network is dynamic spectrum management (DSM). This promising multi-user transmission technique aims at maximizing the transmission rates, and/or minimizing the power consumption, by mitigating or cancelling the dominating crosstalk interference between twisted-pair lines in the same cable binder. Hence the spectral utilization is improved by optimizing the transmit signals in order to minimize the crosstalk interference. However, such techniques rely on accurate information of the (usually) unknown crosstalk channels. This issue is the main focus of Paper VI and VII of this dissertation in which Paper VI deals with estimation of the crosstalk channels between twisted-pair lines. More specifically, an unbiased estimator for the square-magnitude of the crosstalk channels is derived from which a practical procedure is developed that can be implemented with standardized DSL modems already installed in the copper access network. In Paper VII the impact such a non-ideal estimator has on the performance of DSM is analyzed and simulated. Finally, in Paper VIII a novel echo cancellation algorithm for DMT-based DSL modems is presented

Resource-constrained optimal scheduling of SDF graphs via timed automata (extended version)

Synchronous dataflow (SDF) graphs are a widely used formalism for modelling, analysing and realising streaming applications, both on a single processor and in a multiprocessing context. Efficient schedules are essential to obtain maximal throughput under the constraint of available number of resources. This paper presents an approach to schedule SDF graphs using a proven formalism of timed automata (TA). TA maintain a good balance between expressiveness and tractability, and are supported by powerful verification tools, e.g. Uppaal. We describe a compositional translation of SDF graphs to TA, and analysis and verification in the Uppaal state-of-the-art tool. This approach does not require any transformation of SDF graphs and helps to find schedules with a compromise between the number of processors required and the throughput. It also allows quantitative model checking and verification of user-defined properties such as the absence of deadlocks, safety, liveness and throughput analysis. This translation also forms the basis for future work to extend this analysis of SDF graphs with new features such as stochastics, energy consumption and costs

Full-duplex wireless communications: challenges, solutions and future research directions

The family of conventional half-duplex (HD) wireless systems relied on transmitting and receiving in different time-slots or frequency sub-bands. Hence the wireless research community aspires to conceive full-duplex (FD) operation for supporting concurrent transmission and reception in a single time/frequency channel, which would improve the attainable spectral efficiency by a factor of two. The main challenge encountered in implementing an FD wireless device is the large power difference between the self-interference (SI) imposed by the device’s own transmissions and the signal of interest received from a remote source. In this survey, we present a comprehensive list of the potential FD techniques and highlight their pros and cons. We classify the SI cancellation techniques into three categories, namely passive suppression, analog cancellation and digital cancellation, with the advantages and disadvantages of each technique compared. Specifically, we analyse the main impairments (e.g. phase noise, power amplifier nonlinearity as well as in-phase and quadrature-phase (I/Q) imbalance, etc.) that degrading the SI cancellation. We then discuss the FD based Media Access Control (MAC)-layer protocol design for the sake of addressing some of the critical issues, such as the problem of hidden terminals, the resultant end-to-end delay and the high packet loss ratio (PLR) due to network congestion. After elaborating on a variety of physical/MAC-layer techniques, we discuss potential solutions conceived for meeting the challenges imposed by the aforementioned techniques. Furthermore, we also discuss a range of critical issues related to the implementation, performance enhancement and optimization of FD systems, including important topics such as hybrid FD/HD scheme, optimal relay selection and optimal power allocation, etc. Finally, a variety of new directions and open problems associated with FD technology are pointed out. Our hope is that this treatise will stimulate future research efforts in the emerging field of FD communication

Orthogonal transmultiplexers : extensions to digital subscriber line (DSL) communications

An orthogonal transmultiplexer which unifies multirate filter bank theory and communications theory is investigated in this dissertation. Various extensions of the orthogonal transmultiplexer techniques have been made for digital subscriber line communication applications. It is shown that the theoretical performance bounds of single carrier modulation based transceivers and multicarrier modulation based transceivers are the same under the same operational conditions. Single carrier based transceiver systems such as Quadrature Amplitude Modulation (QAM) and Carrierless Amplitude and Phase (CAP) modulation scheme, multicarrier based transceiver systems such as Orthogonal Frequency Division Multiplexing (OFDM) or Discrete Multi Tone (DMT) and Discrete Subband (Wavelet) Multicarrier based transceiver (DSBMT) techniques are considered in this investigation. The performance of DMT and DSBMT based transceiver systems for a narrow band interference and their robustness are also investigated. It is shown that the performance of a DMT based transceiver system is quite sensitive to the location and strength of a single tone (narrow band) interference. The performance sensitivity is highlighted in this work. It is shown that an adaptive interference exciser can alleviate the sensitivity problem of a DMT based system. The improved spectral properties of DSBMT technique reduces the performance sensitivity for variations of a narrow band interference. It is shown that DSBMT technique outperforms DMT and has a more robust performance than the latter. The superior performance robustness is shown in this work. Optimal orthogonal basis design using cosine modulated multirate filter bank is discussed. An adaptive linear combiner at the output of analysis filter bank is implemented to eliminate the intersymbol and interchannel interferences. It is shown that DSBMT is the most suitable technique for a narrow band interference environment. A blind channel identification and optimal MMSE based equalizer employing a nonmaximally decimated filter bank precoder / postequalizer structure is proposed. The performance of blind channel identification scheme is shown not to be sensitive to the characteristics of unknown channel. The performance of the proposed optimal MMSE based equalizer is shown to be superior to the zero-forcing equalizer

Efficient Algorithms for Immersive Audio Rendering Enhancement

Il rendering audio immersivo è il processo di creazione di un’esperienza sonora coinvolgente e realistica nello spazio 3D. Nei sistemi audio immersivi, le funzioni di trasferimento relative alla testa (head-related transfer functions, HRTFs) vengono utilizzate per la sintesi binaurale in cuffia poiché esprimono il modo in cui gli esseri umani localizzano una sorgente sonora. Possono essere introdotti algoritmi di interpolazione delle HRTF per ridurre il numero di punti di misura e per creare un movimento del suono affidabile. La riproduzione binaurale può essere eseguita anche dagli altoparlanti. Tuttavia, il coinvolgimento di due o più gli altoparlanti causa il problema del crosstalk. In questo caso, algoritmi di cancellazione del crosstalk (CTC) sono necessari per eliminare i segnali di interferenza indesiderati. In questa tesi, partendo da un'analisi comparativa di metodi di misura delle HRTF, viene proposto un sistema di rendering binaurale basato sull'interpolazione delle HRTF per applicazioni in tempo reale. Il metodo proposto mostra buone prestazioni rispetto a una tecnica di riferimento. L'algoritmo di interpolazione è anche applicato al rendering audio immersivo tramite altoparlanti, aggiungendo un algoritmo di cancellazione del crosstalk fisso, che considera l'ascoltatore in una posizione fissa. Inoltre, un sistema di cancellazione crosstalk adattivo, che include il tracciamento della testa dell'ascoltatore, è analizzato e implementato in tempo reale. Il CTC adattivo implementa una struttura in sottobande e risultati sperimentali dimostrano che un maggiore numero di bande migliora le prestazioni in termini di errore totale e tasso di convergenza. Il sistema di riproduzione e le caratteristiche dell'ambiente di ascolto possono influenzare le prestazioni a causa della loro risposta in frequenza non ideale. L'equalizzazione viene utilizzata per livellare le varie parti dello spettro di frequenze che compongono un segnale audio al fine di ottenere le caratteristiche sonore desiderate. L'equalizzazione può essere manuale, come nel caso dell'equalizzazione grafica, dove il guadagno di ogni banda di frequenza può essere modificato dall'utente, o automatica, la curva di equalizzazione è calcolata automaticamente dopo la misurazione della risposta impulsiva della stanza. L'equalizzazione della risposta ambientale può essere applicata anche ai sistemi multicanale, che utilizzano due o più altoparlanti e la zona di equalizzazione può essere ampliata misurando le risposte impulsive in diversi punti della zona di ascolto. In questa tesi, GEQ efficienti e un sistema adattativo di equalizzazione d'ambiente. In particolare, sono proposti e approfonditi tre equalizzatori grafici a basso costo computazionale e a fase lineare e quasi lineare. Gli esperimenti confermano l'efficacia degli equalizzatori proposti in termini di accuratezza, complessità computazionale e latenza. Successivamente, una struttura adattativa in sottobande è introdotta per lo sviluppo di un sistema di equalizzazione d'ambiente multicanale. I risultati sperimentali verificano l'efficienza dell'approccio in sottobande rispetto al caso a banda singola. Infine, viene presentata una rete crossover a fase lineare per sistemi multicanale, mostrando ottimi risultati in termini di risposta in ampiezza, bande di transizione, risposta polare e risposta in fase. I sistemi di controllo attivo del rumore (ANC) possono essere progettati per ridurre gli effetti dell'inquinamento acustico e possono essere utilizzati contemporaneamente a un sistema audio immersivo. L'ANC funziona creando un'onda sonora in opposizione di fase rispetto all'onda sonora in arrivo. Il livello sonoro complessivo viene così ridotto grazie all'interferenza distruttiva. Infine, questa tesi presenta un sistema ANC utilizzato per la riduzione del rumore. L’approccio proposto implementa una stima online del percorso secondario e si basa su filtri adattativi in sottobande applicati alla stima del percorso primario che mirano a migliorare le prestazioni dell’intero sistema. La struttura proposta garantisce un tasso di convergenza migliore rispetto all'algoritmo di riferimento.Immersive audio rendering is the process of creating an engaging and realistic sound experience in 3D space. In immersive audio systems, the head-related transfer functions (HRTFs) are used for binaural synthesis over headphones since they express how humans localize a sound source. HRTF interpolation algorithms can be introduced for reducing the number of measurement points and creating a reliable sound movement. Binaural reproduction can be also performed by loudspeakers. However, the involvement of two or more loudspeakers causes the problem of crosstalk. In this case, crosstalk cancellation (CTC) algorithms are needed to delete unwanted interference signals. In this thesis, starting from a comparative analysis of HRTF measurement techniques, a binaural rendering system based on HRTF interpolation is proposed and evaluated for real-time applications. The proposed method shows good performance in comparison with a reference technique. The interpolation algorithm is also applied for immersive audio rendering over loudspeakers, by adding a fixed crosstalk cancellation algorithm, which assumes that the listener is in a fixed position. In addition, an adaptive crosstalk cancellation system, which includes the tracking of the listener's head, is analyzed and a real-time implementation is presented. The adaptive CTC implements a subband structure and experimental results prove that a higher number of bands improves the performance in terms of total error and convergence rate. The reproduction system and the characteristics of the listening room may affect the performance due to their non-ideal frequency response. Audio equalization is used to adjust the balance of different audio frequencies in order to achieve desired sound characteristics. The equalization can be manual, such as in the case of graphic equalization, where the gain of each frequency band can be modified by the user, or automatic, where the equalization curve is automatically calculated after the room impulse response measurement. The room response equalization can be also applied to multichannel systems, which employ two or more loudspeakers, and the equalization zone can be enlarged by measuring the impulse responses in different points of the listening zone. In this thesis, efficient graphic equalizers (GEQs), and an adaptive room response equalization system are presented. In particular, three low-complexity linear- and quasi-linear-phase graphic equalizers are proposed and deeply examined. Experiments confirm the effectiveness of the proposed GEQs in terms of accuracy, computational complexity, and latency. Successively, a subband adaptive structure is introduced for the development of a multichannel and multiple positions room response equalizer. Experimental results verify the effectiveness of the subband approach in comparison with the single-band case. Finally, a linear-phase crossover network is presented for multichannel systems, showing great results in terms of magnitude flatness, cutoff rates, polar diagram, and phase response. Active noise control (ANC) systems can be designed to reduce the effects of noise pollution and can be used simultaneously with an immersive audio system. The ANC works by creating a sound wave that has an opposite phase with respect to the sound wave of the unwanted noise. The additional sound wave creates destructive interference, which reduces the overall sound level. Finally, this thesis presents an ANC system used for noise reduction. The proposed approach implements an online secondary path estimation and is based on cross-update adaptive filters applied to the primary path estimation that aim at improving the performance of the whole system. The proposed structure allows for a better convergence rate in comparison with a reference algorithm

Seamless virtual network for international business continuity in presence of intentional blocks

東京電機大学201

FFT and FIR Filter implementations for the DSL MODEMS

Broad band digital communication that operates over a standard copper wires. It requires the DSL modems which splits the transmissions into 2 frequency bands. The lower frequencies for voice and the higher frequencies for digital data (internet) in order to transmit the data to larger distances through a copper cable we need modulation techniques. Generally in this DSL modems modulation used is QAM technique. The output of the QAM is complex data this complex data we cannot transfer directly through a copper cable because the data should be in time domain or otherwise the phase of the data which is in frequency domain can be lost, in copper cable so this data should be converted in time domain by using IDFT technique. As IDFT requires more number of complex multiplications and more number of complex additions in comparison to IFFT so to reduce the additions and multiplications IFFT technique is used. At the receiver side we can retrieve the same data by using FFT technique. In this section the implemented FFT architecture is fully efficient and this architecture will require less area. And before we have to transmit through the copper line we have to do interpolation or decimation by using the Filtering operation. The implemented poly phase architecture for the filtering is fully efficient, symmetrical and it requires less number of multipliers

Channelized voice over digital subscriber line

The objective of this article is to present a promising voice over digital subscriber line (VoDSL) solution: an alternative method that uses physical layer transportation to provide channelized VoDSL (CVoDSL). This article also presents the advantages that the CVoDSL provides to the customers in terms of high-quality services and lower cost, and service providers in terms of market growth and success. It also offers a set of requirements, architectural design, and implementation scenarios that provide end-to-end solutions for CVoDSL