A framework for the analysis and optimization of delay in multiview video coding schemes

Esta tesis presenta un novedoso marco de referencia para el análisis y optimización del retardo de codificación y descodificación para vídeo multivista. El objetivo de este marco de referencia es proporcionar una metodología sistemática para el análisis del retardo en codificadores y descodificadores multivista y herramientas útiles en el diseño de codificadores/descodificadores para aplicaciones con requisitos de bajo retardo. El marco de referencia propuesto caracteriza primero los elementos que tienen influencia en el comportamiento del retardo: i) la estructura de predicción multivista, ii) el modelo hardware del codificador/descodificador y iii) los tiempos de proceso de cuadro. En segundo lugar, proporciona algoritmos para el cálculo del retardo de codificación/ descodificación de cualquier estructura arbitraria de predicción multivista. El núcleo de este marco de referencia consiste en una metodología para el análisis del retardo de codificación/descodificación multivista que es independiente de la arquitectura hardware del codificador/descodificador, completada con un conjunto de modelos que particularizan este análisis del retardo con las características de la arquitectura hardware del codificador/descodificador. Entre estos modelos, aquellos basados en teoría de grafos adquieren especial relevancia debido a su capacidad de desacoplar la influencia de los diferentes elementos en el comportamiento del retardo en el codificador/ descodificador, mediante una abstracción de su capacidad de proceso. Para revelar las posibles aplicaciones de este marco de referencia, esta tesis presenta algunos ejemplos de su utilización en problemas de diseño que afectan a codificadores y descodificadores multivista. Este escenario de aplicación cubre los siguientes casos: estrategias para el diseño de estructuras de predicción que tengan en consideración requisitos de retardo además del comportamiento tasa-distorsión; diseño del número de procesadores y análisis de los requisitos de velocidad de proceso en codificadores/ descodificadores multivista dado un retardo objetivo; y el análisis comparativo del comportamiento del retardo en codificadores multivista con diferentes capacidades de proceso e implementaciones hardware. ABSTRACT This thesis presents a novel framework for the analysis and optimization of the encoding and decoding delay for multiview video. The objective of this framework is to provide a systematic methodology for the analysis of the delay in multiview encoders and decoders and useful tools in the design of multiview encoders/decoders for applications with low delay requirements. The proposed framework characterizes firstly the elements that have an influence in the delay performance: i) the multiview prediction structure ii) the hardware model of the encoder/decoder and iii) frame processing times. Secondly, it provides algorithms for the computation of the encoding/decoding delay of any arbitrary multiview prediction structure. The core of this framework consists in a methodology for the analysis of the multiview encoding/decoding delay that is independent of the hardware architecture of the encoder/decoder, which is completed with a set of models that particularize this delay analysis with the characteristics of the hardware architecture of the encoder/decoder. Among these models, the ones based in graph theory acquire special relevance due to their capacity to detach the influence of the different elements in the delay performance of the encoder/decoder, by means of an abstraction of its processing capacity. To reveal possible applications of this framework, this thesis presents some examples of its utilization in design problems that affect multiview encoders and decoders. This application scenario covers the following cases: strategies for the design of prediction structures that take into consideration delay requirements in addition to the rate-distortion performance; design of number of processors and analysis of processor speed requirements in multiview encoders/decoders given a target delay; and comparative analysis of the encoding delay performance of multiview encoders with different processing capabilities and hardware implementations.

Depth perceptual video coding for free viewpoint video based on H.264/AVC

A novel scheme for depth sequences compression, based on a perceptual coding algorithm, is proposed. A depth sequence describes the object position in the 3D scene, and is used, in Free Viewpoint Video, for the generation of synthetic video sequences. In perceptual video coding the human visual system characteristics are exploited to improve the compression efficiency. As depth sequences are never shown, the perceptual video coding, assessed over them, is not effective. The proposed algorithm is based on a novel perceptual rate distortion optimization process, assessed over the perceptual distortion of the rendered views generated through the encoded depth sequences. The experimental results show the effectiveness of the proposed method, able to obtain a very considerable improvement of the rendered view perceptual quality.

NAMA3DS1-COSPAD1: Subjective video quality assessment database on coding conditions introducing freely available high quality 3D stereoscopic sequences

Research in stereoscopic 3D coding, transmission and subjective assessment methodology depends largely on the availability of source content that can be used in cross-lab evaluations. While several studies have already been presented using proprietary content, comparisons between the studies are difficult since discrepant contents are used. Therefore in this paper, a freely available dataset of high quality Full-HD stereoscopic sequences shot with a semiprofessional 3D camera is introduced in detail. The content was designed to be suited for usage in a wide variety of applications, including high quality studies. A set of depth maps was calculated from the stereoscopic pair. As an application example, a subjective assessment has been performed using coding and spatial degradations. The Absolute Category Rating with Hidden Reference method was used. The observers were instructed to vote on video quality only. Results of this experiment are also freely available and will be presented in this paper as a first step towards objective video quality measurement for 3DTV.

Unidad de efectos de audio en Simulink

La tecnología moderna de computación ha permitido cambiar radicalmente la investigación tecnológica en todos los ámbitos. El proceso general utilizado previamente consistía en el desarrollo de prototipos analógicos, creando múltiples versiones del mismo hasta llegar al resultado adecuado. Este es un proceso costoso a nivel económico y de carga de trabajo. Es por ello por lo que el proceso de investigación actual aprovecha las nuevas tecnologías para lograr el objetivo final mediante la simulación. Gracias al desarrollo de software para la simulación de distintas áreas se ha incrementado el ritmo de crecimiento de los avances tecnológicos y reducido el coste de los proyectos en investigación y desarrollo. La simulación, por tanto, permite desarrollar previamente prototipos simulados con un coste mucho menor para así lograr un producto final, el cual será llevado a cabo en su ámbito correspondiente. Este proceso no sólo se aplica en el caso de productos con circuitería, si bien es utilizado también en productos programados. Muchos de los programas actuales trabajan con algoritmos concretos cuyo funcionamiento debe ser comprobado previamente, para después centrarse en la codificación del mismo. Es en este punto donde se encuentra el objetivo de este proyecto, simular algoritmos de procesado digital de la señal antes de la codificación del programa final. Los sistemas de audio están basados en su totalidad en algoritmos de procesado de la señal, tanto analógicos como digitales, siendo estos últimos los que están sustituyendo al mundo analógico mediante los procesadores y los ordenadores. Estos algoritmos son la parte más compleja del sistema, y es la creación de nuevos algoritmos la base para lograr sistemas de audio novedosos y funcionales. Se debe destacar que los grupos de desarrollo de sistemas de audio presentan un amplio número de miembros con cometidos diferentes, separando las funciones de programadores e ingenieros de la señal de audio. Es por ello por lo que la simulación de estos algoritmos es fundamental a la hora de desarrollar nuevos y más potentes sistemas de audio. Matlab es una de las herramientas fundamentales para la simulación por ordenador, la cual presenta utilidades para desarrollar proyectos en distintos ámbitos. Sin embargo, en creciente uso actualmente se encuentra el software Simulink, herramienta especializada en la simulación de alto nivel que simplifica la dificultad de la programación en Matlab y permite desarrollar modelos de forma más rápida. Simulink presenta una completa funcionalidad para el desarrollo de algoritmos de procesado digital de audio. Por ello, el objetivo de este proyecto es el estudio de las capacidades de Simulink para generar sistemas de audio funcionales. A su vez, este proyecto pretende profundizar en los métodos de procesado digital de la señal de audio, logrando al final un paquete de sistemas de audio compatible con los programas de edición de audio actuales. ABSTRACT. Modern computer technology has dramatically changed the technological research in multiple areas. The overall process previously used consisted of the development of analog prototypes, creating multiple versions to reach the proper result. This is an expensive process in terms of an economically level and workload. For this reason actual investigation process take advantage of the new technologies to achieve the final objective through simulation. Thanks to the software development for simulation in different areas the growth rate of technological progress has been increased and the cost of research and development projects has been decreased. Hence, simulation allows previously the development of simulated protoypes with a much lower cost to obtain a final product, which will be held in its respective field. This process is not only applied in the case of circuitry products, but is also used in programmed products. Many current programs work with specific algorithms whose performance should be tested beforehand, which allows focusing on the codification of the program. This is the main point of this project, to simulate digital signal processing algorithms before the codification of the final program. Audio systems are entirely based on signal processing, both analog and digital systems, being the digital systems which are replacing the analog world thanks to the processors and computers. This algorithms are the most complex part of every system, and the creation of new algorithms is the most important step to achieve innovative and functional new audio systems. It should be noted that development groups of audio systems have a large number of members with different roles, separating them into programmers and audio signal engineers. For this reason, the simulation of this algorithms is essential when developing new and more powerful audio systems. Matlab is one of the most important tools for computer simulation, which has utilities to develop projects in different areas. However, the use of the Simulink software is constantly growing. It is a simulation tool specialized in high-level simulations which simplifies the difficulty of programming in Matlab and allows the developing of models faster. Simulink presents a full functionality for the development of algorithms for digital audio processing. Therefore, the objective of this project is to study the posibilities of Simulink to generate funcional audio systems. In turn, this projects aims to get deeper into the methods of digital audio signal processing, making at the end a software package of audio systems compatible with the current audio editing software.

Desarrollo de una biblioteca de efectos digitales de audio para ser usada en un laboratorio docente

Desarrollo de una librería de efectos de audio en lenguaje nativo de Matlab con procesamiento a tiempo no real. Incluye una interfaz de usuario sencilla y auto explicativa, y ofrece un control libre de los parámetros del efecto, elección y visualización del audio de entrada, reproducción y visualización del audio de salida, y representación característica del procesamiento que se está realizando. El objetivo principal de la librería es que sea usada por alumnos en un laboratorio docente, permitiendo la experimentación con diversos parámetros y entradas de audio facilitando, de esta forma, la comprensión de los diferentes procesamientos que se están realizando. El proyecto incluye una extensa documentación y una plantilla con el objetivo de que se puedan añadir en un futuro más programas de efectos, puesto que la intención del proyecto es ofrecer una librería a largo plazo y facilitar el mantenimiento y las modificaciones futuras.

Implementación de efectos de audio en la tarjeta Raspberry Pi B+

Este proyecto consiste en el diseño e implementación de un procesador digital de efectos de audio en tiempo real orientado a instrumentos eléctricos tales como guitarras, bajos, teclados, etc. El procesador está basado en la tarjeta Raspberry Pi B+, ordenador de placa reducida de bajo coste, desarrollado en Reino unido y cuyo lanzamiento tuvo lugar en el año 2012. En primer lugar, ha sido necesario lograr que la tarjeta asuma la funcionalidad de un procesador de audio en tiempo real. Para ello se ha instalado un sistema operativo Linux orientado a Raspberry (Raspbian) y se ha hecho uso de Pure Data (Pd): lenguaje de programación gráfico que fue desarrollado en los años 90 por Miller Puckette con intención de ser enfocado a la creación de eventos multimedia y de música por computador. El papel que desempeña Pd es de capa intermedia entre el hardware y el software ya que se encarga de tomar bloques de N muestras del convertidor analógico/digital y encaminarlas a través del flujo de señal diseñado gráficamente. En segundo lugar, se han implementado diferentes efectos de audio de distintas características. Así pues, se encuentran efectos basados en retardos, filtros digitales y procesadores de dinámica. Concretamente, los efectos implementados son los siguientes: delay, flanger, vibrato, reverberador de Schroeder, filtros (paso bajo, paso alto y paso banda), ecualizador paramétrico y compresor y expansor de dinámica. Estos efectos han sido implementados en lenguaje C de acuerdo con la API de Pd. Con esto se ha conseguido obtener un objeto por cada efecto, el cual es “instanciado” en Pd pudiendo ejecutarlo en tiempo real. En este proyecto se expone la problemática que supone cada paso del diseño proponiendo soluciones válidas. Además se incluye una guía paso a paso para configurar la tarjeta y lograr realizar un bypass de señal y un efecto simple partiendo desde cero. ABSTRACT. This project involves the design and implementation of a digital real-time audio processor for electrical instruments (guitars, basses, keyboards, etc.). The processor is based on the Raspberry Pi B + card: low cost computer, developed in UK in 2012. First, it was necessary to make the cards assume the functionality of a real time audio processor. A Linux operating system called Raspberry (Raspbian) was installed. In this Project is used Pure Data (Pd): a graphical programming language developed in the 90s by Miller Puckette intending to be focused on creating multimedia and computer music events. The role of Pd is an intermediate layer between the hardware and the software. It is responsible for taking blocks of N samples of the analog/digital converter and route it through the signal flow. Secondly, it is necessary to implemented the different audio effects. There are delays based effects, digital filter and dynamics effects. Specifically, the implemented effects are: delay, flanger, vibrato, Schroeder reverb, filters (lowpass, highpass and bandpass), parametric equalizer and compressor and expander dynamics. These effects have been implemented in C language according to the Pd API. As a result, it has been obtained an object for each effect, which is instantiated in Pd. In this Project, the problems of every step are exposed with his corresponding solution. It is inlcuded a step-by-step guide to configure the card and achieve perform a bypass signal process and a simple effect.

Evolución de los sistemas técnicos de reproducción musical

Con el fin de analizar la caracterización del proceso de evolución de los sistemas técnicos de reproducción musical se han planteado dos vías principales de análisis: en primer lugar, el estudio de las características técnicas de cuatro sistemas de reproducción que se han considerado representativos de dicho avance (sistemas basados en disco de vinilo, casete compacto, disco compacto y formato de codificación perceptual) y, en segundo lugar, el funcionamiento de dichos sistemas dentro del concepto que se ha denominado como cadena de transmisión de información de tipo musical. A través de la evaluación del grado de evolución de cada sistema y de su posterior comparación se procederá a realizar dicha caracterización. Para ello se tendrán en cuenta factores objetivos como, por ejemplo, la calidad de la señal de audio pero también cualitativos como, por ejemplo, la portabilidad que pueda proporcionar un tipo de sistema u otro. Por otra parte, se plantea un segundo objetivo que es actualizar la definición de reproductor, si así se desprende de los análisis, con la aparición de reproductores basados en codificación perceptual. ABSTRACT. In order to analyze the characterization of the process of evolution of technical systems of music playback considered two main ways of analysis: first, the study of the technical characteristics of four playback systems that have been considered representative of the advance (based on vinyl, compact cassette, compact disc and perceptual formats coding systems) and, second, the operation of these systems within the concept that has been called chain of information transmission of music. Through the evaluation of the degree of development of each system and its subsequent comparison will proceed to make such characterization. For this objective kept in mind factors such as the quality of the audio signal but also qualitative, for example, portability that can provide a particular device type will be considered. Moreover, a second objective is to update the definition of the player, if it is apparent from the analysis, with the appearance of players based on perceptual coding arises.

Diseño de un sistema networking audio para instalación audiovisual

Una vez presentada la tecnología de Networking audio (redes de datos, protocolos actuales, etc.) se realizará un diseño de la instalación del sistema de audio, en el que el punto de partida es la parte creativa de la actividad en dicha instalación: un juego en el que la comunicación auditiva es lo fundamental. La instalación se compondrá de una sala central, tres salas de grupos, tres salas de cabinas de actores y ocho salas de pasaje. Esta actividad tan particular hará plantearse configuraciones, equipamiento y formas de trabajar especiales que, mediante la tecnología de audio vía red de datos y el equipamiento auxiliar a esta red, podría realizarse de la una forma óptima cumpliendo con todos los objetivos de la actividad, tanto técnicos como relativos al juego. El libro se dividirá en dos partes: La primera parte consistirá en una explicación de lo que son las redes de datos y los aspectos básicos para entenderlas desde un punto de vista práctico: qué es Ethernet, los componentes de una red... Una vez explicada la terminología específica de redes, se expondrán los protocolos que se usan para transmitir audio profesional a día de hoy. En la segunda parte, se empezará presentando la actividad que se realizará en nuestra instalación: un juego de rol. A continuación se conocerá el flujo de señales existentes para después, poner en práctica lo aprendido en la primera parte: diseñaremos una instalación audiovisual mediante networking audio. Un sistema de estas características necesita además de dispositivos en red, sistemas convencionales de audio. Durante el diseño y debido a las necesidades tan específicas de la instalación, se verá que ha sido necesario pensar en sistemas especiales para hacer posible la actividad para la que ha sido ideada nuestra instalación. Los objetivos de este proyecto son, desarrollar los puntos que tendría que tener en cuenta un integrador que se proponga diseñar un sistema de audio networking para una instalación audiovisual para, a continuación, poner en práctica estos conocimientos con la exposición del diseño de una instalación en la que se llevará a cabo una actividad lúdica y de aprendizaje en la que una óptima transmisión de señal de audio a tiempo real, es lo fundamental. ABSTRACT. Once introduced the Networking technology (data networks, current protocols, etc.), the audio installation design is being done. In which the starting point is the creative part of the activity will be made: one game in which the auditory communication is fundamental. The installation will consist of a central room, three meeting groups, three actor cabins rooms and eight passage rooms. This particular activity will consider configurations, equipment and forms of special working that through audio technology via data network and auxiliary equipment to this network, it could be done in an optimal way to meet all the goals of the activity, both technical and relative to the game. The book is divided into two parts: The first part consists of an explanation of what the data networks and the basics to understand from a practical point of view: what Ethernet is, the network components... Once specific network terminology is explained, the current protocols used to transmit professional audio are being showed. In the second part, it is introducing the activity to be made in our installation: a game. Then, the flow of existing signals are being known, we practice what I learned in the first part: we will design an audiovisual installation by audio networking. A system like this besides networked devices, it needs conventional audio systems. During the design and due to the very specific needs of the installation, you will see that it was necessary to think of special systems for this special activity. The goals of this project are to develop the points that an system integrator would have to consider to design a system of networking audio for an audiovisual installation, then put this knowledge into practice with the installation design where it will take place a fun and learning activity in which an optimal transmission of audio signal in real time, is basic.

The U5 RNA of trypanosomes deviates from the canonical U5 RNA: The Leptomonas collosoma U5 RNA and its coding gene

Fractionation of the abundant small ribonucleoproteins (RNPs) of the trypanosomatid Leptomonas collosoma revealed the existence of a group of unidentified small RNPs that were shown to fractionate differently than the well-characterized trans-spliceosomal RNPs. One of these RNAs, an 80-nt RNA, did not possess a trimethylguanosine (TMG) cap structure but did possess a 5′ phosphate terminus and an invariant consensus U5 snRNA loop 1. The gene coding for the RNA was cloned, and the coding region showed 55% sequence identity to the recently described U5 homologue of Trypanosoma brucei [Dungan, J. D., Watkins, K. P. & Agabian, N. (1996) EMBO J. 15, 4016–4029]. The L. collosoma U5 homologue exists in multiple forms of RNP complexes, a 10S monoparticle, and two subgroups of 18S particles that either contain or lack the U4 and U6 small nuclear RNAs, suggesting the existence of a U4/U6⋅U5 tri-small nuclear RNP complex. In contrast to T. brucei U5 RNA (62 nt), the L. collosoma homologue is longer (80 nt) and possesses a second stem–loop. Like the trypanosome U3, U6, and 7SL RNA genes, a tRNA gene coding for tRNACys was found 98 nt upstream to the U5 gene. A potential for base pair interaction between U5 and SL RNA in the 5′ splice site region (positions −1 and +1) and downstream from it is proposed. The presence of a U5-like RNA in trypanosomes suggests that the most essential small nuclear RNPs are ubiquitous for both cis- and trans-splicing, yet even among the trypanosomatids the U5 RNA is highly divergent.

Musical Listening and Kinesthesis : Is There an Audio-Vocal Tuning System?

Peer reviewed

In silico identification and characterisation of 17 polymorphic anonymous non-coding sequence markers (ANMs) for red grouse (Lagopus lagopus scotica)

Peer reviewed

Exome-wide analysis of rare coding variation identifies novel associations with COPD and airflow limitation in MOCS3, IFIT3 and SERPINA12

Funding: British Women’s Heart and Health Study is funded by the Department of Health grant no. 90049 and the British Heart Foundation grant no. PG/09/022. British Regional Heart Study is supported by the British Heart Foundation (grant RG/ 13/16/30528). CB (COPDBEAT) received funding from the Medical Research Council UK (grant no. G0601369), CB (COPDBEAT) and AJW (UKCOPD) were supported by the National Institute for Health Research (NIHR Leicester Biomedical Research Unit). MB (COPDBEAT) received funding from the NIHR (grant no. PDF-2013-06-052). Hertfordshire Cohort Study received support from the Medical Research Council, Arthritis Research UK, the International Osteoporosis Foundation and the British Heart Foundation; NIHR Biomedical Research Centre in Nutrition, University of Southampton; NIHR Musculoskeletal Biomedical Research Unit, University of Oxford. Generation Scotland: Scottish Family Health Study is funded by the Chief Scientist Office, Scottish Government Health Directorates, grant number CZD/16/6 and the Scottish Funding Council grant HR03006. EU COPD Gene Scan is funded by the European Union, grant no. QLG1-CT-2001-01012. English Longitudinal Study of Aging is funded by the Institute of Aging, NIH grant No. AG1764406S1. GoDARTs is funded by the Wellcome Trust grants 072960, 084726 and 104970. MDT has been supported by MRC fellowship G0902313. UK Biobank Lung Exome Variant Evaluation study was funded by a Medical Research Council strategic award to MDT, IPH, DPS and LVW (MC_PC_12010)

A herpes simplex virus 1 recombinant lacking the glycoprotein G coding sequences is defective in entry through apical surfaces of polarized epithelial cells in culture and in vivo

During infection of a new host, the first surfaces encountered by herpes simplex viruses are the apical membranes of epithelial cells of mucosal surfaces. These cells are highly polarized, and the protein composition of their apical and basolateral membranes are very different, so that different viral entry pathways have evolved for each surface. To determine whether the viral glycoprotein G (gG) is specifically required for efficient infection of a particular surface of polarized cells, apical and basal surfaces were infected with wild-type virus or a gG deletion mutant. After infection of polarized cells in culture, the gG− virus was deficient in infection of apical surfaces but was able to infect cells through basal membranes, replicate, and spread into surrounding cells. The gG-dependent step in apical infection was a stage beyond attachment. After in vivo infection of apical surfaces of epithelial cells of nonscarified mouse corneas, infection by glycoprotein C− or gG− virus was considerably reduced as compared with that observed after infection with wild-type virus. In contrast, when corneas were scarified, allowing virus access to other cell surfaces, the gG and glycoprotein C deletion mutants infected eyes as efficiently as wild-type viruses. A secondary mutation allowing infection of apical surfaces by gG− virus arose readily during passage of the virus in nonpolarized cells, indicating that either the gG-dependent step of apical infection can be bypassed or that another viral protein can acquire the same function.

Complex promoter and coding region β2-adrenergic receptor haplotypes alter receptor expression and predict in vivo responsiveness

The human β2-adrenergic receptor gene has multiple single-nucleotide polymorphisms (SNPs), but the relevance of chromosomally phased SNPs (haplotypes) is not known. The phylogeny and the in vitro and in vivo consequences of variations in the 5′ upstream and ORF were delineated in a multiethnic reference population and an asthmatic cohort. Thirteen SNPs were found organized into 12 haplotypes out of the theoretically possible 8,192 combinations. Deep divergence in the distribution of some haplotypes was noted in Caucasian, African-American, Asian, and Hispanic-Latino ethnic groups with >20-fold differences among the frequencies of the four major haplotypes. The relevance of the five most common β2-adrenergic receptor haplotype pairs was determined in vivo by assessing the bronchodilator response to β agonist in asthmatics. Mean responses by haplotype pair varied by >2-fold, and response was significantly related to the haplotype pair (P = 0.007) but not to individual SNPs. Expression vectors representing two of the haplotypes differing at eight of the SNP loci and associated with divergent in vivo responsiveness to agonist were used to transfect HEK293 cells. β2-adrenergic receptor mRNA levels and receptor density in cells transfected with the haplotype associated with the greater physiologic response were ≈50% greater than those transfected with the lower response haplotype. The results indicate that the unique interactions of multiple SNPs within a haplotype ultimately can affect biologic and therapeutic phenotype and that individual SNPs may have poor predictive power as pharmacogenetic loci.

How the owl resolves auditory coding ambiguity

The barn owl (Tyto alba) uses interaural time difference (ITD) cues to localize sounds in the horizontal plane. Low-order binaural auditory neurons with sharp frequency tuning act as narrow-band coincidence detectors; such neurons respond equally well to sounds with a particular ITD and its phase equivalents and are said to be phase ambiguous. Higher-order neurons with broad frequency tuning are unambiguously selective for single ITDs in response to broad-band sounds and show little or no response to phase equivalents. Selectivity for single ITDs is thought to arise from the convergence of parallel, narrow-band frequency channels that originate in the cochlea. ITD tuning to variable bandwidth stimuli was measured in higher-order neurons of the owl’s inferior colliculus to examine the rules that govern the relationship between frequency channel convergence and the resolution of phase ambiguity. Ambiguity decreased as stimulus bandwidth increased, reaching a minimum at 2–3 kHz. Two independent mechanisms appear to contribute to the elimination of ambiguity: one suppressive and one facilitative. The integration of information carried by parallel, distributed processing channels is a common theme of sensory processing that spans both modality and species boundaries. The principles underlying the resolution of phase ambiguity and frequency channel convergence in the owl may have implications for other sensory systems, such as electrolocation in electric fish and the computation of binocular disparity in the avian and mammalian visual systems.