928 resultados para audio coding


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Scientific background: Marine mammals use sound for communication, navigation and prey detection. Acoustic sensors therefore allow the detection of marine mammals, even during polar winter months, when restricted visibility prohibits visual sightings. The animals are surrounded by a permanent natural soundscape, which, in polar waters, is mainly dominated by the movement of ice. In addition to the detection of marine mammals, acoustic long-term recordings provide information on intensity and temporal variability of characteristic natural and anthropogenic background sounds, as well as their influence on the vocalization of marine mammals Scientific objectives: The PerenniAL Acoustic Observatory in the Antarctic Ocean (PALAOA, Hawaiian "whale") near Neumayer Station is intended to record the underwater soundscape in the vicinity of the shelf ice edge over the duration of several years. These long-term recordings will allow studying the acoustic repertoire of whales and seals continuously in an environment almost undisturbed by humans. The data will be analyzed to (1) register species specific vocalizations, (2) infer the approximate number of animals inside the measuring range, (3) calculate their movements relative to the observatory, and (4) examine possible effects of the sporadic shipping traffic on the acoustic and locomotive behaviour of marine mammals. The data, which are largely free of anthropogenic noise, provide also a base to set up passive acoustic mitigation systems used on research vessels. Noise-free bioacoustic data thereby represent the foundation for the development of automatic pattern recognition procedures in the presence of interfering sounds, e.g. propeller noise.

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Protein-coding gene families are sets of similar genes with a shared evolutionary origin and, generally, with similar biological functions. In plants, the size and role of gene families has been only partially addressed. However, suitable bioinformatics tools are being developed to cluster the enormous number of sequences currently available in databases. Specifically, comparative genomic databases promise to become powerful tools for gene family annotation in plant clades. In this review, I evaluate the data retrieved from various gene family databases, the ease with which they can be extracted and how useful the extracted information is.

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We present an innovative system to encode and transmit textured multi-resolution 3D meshes in a progressive way, with no need to send several texture images, one for each mesh LOD (Level Of Detail). All texture LODs are created from the finest one (associated to the finest mesh), but can be re- constructed progressively from the coarsest thanks to refinement images calculated in the encoding process, and transmitted only if needed. This allows us to adjust the LOD/quality of both 3D mesh and texture according to the rendering power of the device that will display them, and to the network capacity. Additionally, we achieve big savings in data transmission by avoiding altogether texture coordinates, which are generated automatically thanks to an unwrapping system agreed upon by both encoder and decoder.

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Este proyecto pretende mostrar los desfases existentes entre señales de audio obtenidas de la misma fuente en distintos puntos distanciados entre sí. Para ello nos basamos en el análisis de la correlación de las señales de audio multi-microfónicas, para determinar los retrasos entre dichas señales. Durante las de tres partes diferentes que conforman este proyecto, explicaremos el dónde, cómo y por qué se produce este efecto en este tipo de señales. En la primera se presentan algunos de los conceptos teóricos necesarios para entender el desarrollo posterior, tales como la coherencia y correlación entre señales, los retardos de fase y la importancia del micro-tiempo. Además se explican diversas técnicas microfónicas que se utilizarán en la tercera parte. A lo largo de la segunda, se presenta el software desarrollado para determinar y corregir el retraso entre las señales que se deseen analizar. Para ello se ha escogido la herramienta de programación Matlab, ya que ha sido la más utilizada en la mayoría de las asignaturas que componen la titulación y por ello se posee el suficiente dominio de la misma. Además de presentar el propio software, al final de esta parte hay un manual de usuario del mismo, en el que se explica el manejo para posibles usos futuros por parte de otras personas interesadas. En la última parte se demuestra en varios casos reales, el estudio de la alineación de tomas multi-microfónicas en las cuales se produce en efecto que se intenta detectar y corregir. Aquí se realizan tres estudios de dicho fenómeno. En el primero se emplean señales digitales internas, concretamente ruido blanco, retrasando algunas muestras dichas señales unas de otras, para luego analizarlas con el software desarrollado y comprobar la eficacia del mismo. En el segundo se analizan la señales de audio obtenidas en el estudio de grabación de varios grupos de música moderna, mostrando los resultados del empleo del software en algunas de ellas, tales como las tomas de batería, bajo y guitarra. En el tercero se analizan las señales de audio obtenidas fuera del estudio de grabación, en donde no se dispone de las supuestas condiciones ideales que se tienen en el entorno que rodea a un estudio de grabación (acústicamente hablando). Se utilizan algunas de las técnicas microfónicas explicadas en el último apartado de la parte dedicada a los conceptos teóricos, para la grabación de una orquesta sinfónica, para luego analizar el efecto buscado mediante nuestro software, presentando los resultados obtenidos. De igual manera se realiza en el estudio con una agrupación coral de cuatro voces dentro de una Iglesia. ABSTRACT This project aims to show delays between audio signals obtained from the same source at diferent points spaced apart. To do this we rely on the analysis of the correlation of multi-microphonic audio signals, to determine the delay between these signals. During three diferent parts that make up this project, we will explain where, how and why this effect occurs in this type of signals. At the first part we present some of the theoretical concepts necessary to understand the subsequent development, such as coherence and correlation between signals, phase delays and the importance of micro-time. Also explains several microphone techniques to be used in the third part. During the second, it presents the software developed to determine and correct the delay between the signals that are desired to analyze. For this we have chosen the programming software Matlab , as it has been the most used in the majority of the subjects in the degree and therefore has suficient command of it. Besides presenting the software at the end of this part there is a user manual of it , which explains the handling for future use by other interested people. The last part is shown in several real cases, the study of aligning multi- microphonic sockets in which it is produced in effect trying to detect and correct. This includes three studies of this phenomenon. In the first internal digital signals are used, basically white noise, delaying some samples the signals from each other, then with software developed analyzing and verifying its efectiveness. In the second analyzes the audio signals obtained in the recording studio several contemporary bands, showing the results of using the software in some of them, such as the taking of drums, bass and guitar. In the third analyzes audio signals obtained outside the recording studio, where there are no ideal conditions alleged to have on the environment surrounding a recording studio (acoustically speaking). We use some of the microphone techniques explained in the last paragraph of the section on theoretical concepts, for the recording of a symphony orchestra, and then analyze the effect sought by our software, presenting the results. Similarly, in the study performed with a four-voice choir in a church.

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SSR es el acrónimo de SoundScape Renderer (tool for real-time spatial audio reproduction providing a variety of rendering algorithms), es un programa escrito en su mayoría en C++. El programa permite al usuario escuchar tanto sonidos grabados con anterioridad como sonidos en directo. El sonido o los sonidos se oirán, desde el punto de vista del oyente, como si el sonido se produjese en el punto que el programa decida, lo interesante de este proyecto es que el sonido podrá cambiar de lugar, moverse, etc. Todo en tiempo real. Esto se consigue sin modificar el sonido al grabarlo pero sí al emitirlo, el programa calcula las variaciones necesarias para que al emitir el sonido al oyente le llegue como si el sonido realmente se generase en un punto del espacio o lo más parecido posible. La sensación de movimiento no deja de ser el punto anterior cambiando de lugar. La idea era crear una aplicación web basada en Canvas de HTML5 que se comunicará con esta interfaz de usuario remota. Así se solucionarían todos los problemas de compatibilidad ya que cualquier dispositivo con posibilidad de visualizar páginas web podría correr una aplicación basada en estándares web, por ejemplo un sistema con Windows o un móvil con navegador. El protocolo debía de ser WebSocket porque es un protocolo HTML5 y ofrece las “garantías” de latencia que una aplicación con necesidades de información en tiempo real requiere. Nos permite una comunicación full-dúplex asíncrona sin mucho payload que es justo lo que se venía a evitar al no usar polling normal de HTML. El problema que surgió fue que la interfaz de usuario de red que tenía el programa no era compatible con WebSocket debido a un handshacking inicial y obligatorio que realiza el protocolo, por lo que se necesitaba otra interfaz de red. Se decidió entonces cambiar a JSON como formato para el intercambio de mensajes. Al final el proyecto comprende no sólo la aplicación web basada en Canvas sino también un servidor funcional y la definición de una nueva interfaz de usuario de red con su protocolo añadido. ABSTRACT. This project aims to become a part of the SSR tool to extend its capabilities in the field of the access. SSR is an acronym for SoundScape Renderer, is a program mostly written in C++ that allows you to hear already recorded or live sound with a variety of sound equipment as if the sound came from a desired place in the space. Like the web-page of the SSR says surely better explained: “The SoundScape Renderer (SSR) is a tool for real-time spatial audio reproduction providing a variety of rendering algorithms.” The application can be used with a graphical interface written in Qt but has also a network interface for external applications to use it. This network interface communicates using XML messages. A good example of it is the Android client. This Android client is already working. In order to use the application should be run it by loading an audio source and the wanted environment so that the renderer knows what to do. In that moment the server binds and anyone can use the network interface. Since the network interface is documented everyone can make an application to interact with this network interface. So the application can have as many user interfaces as wanted. The part that is developed in this project has nothing to do neither with audio rendering nor even with the reproduction of the spatial audio. The part that is developed here is about the interface used in the SSR application. As it can be deduced from the title: “Distributed Web Interface for Real-Time Spatial Audio Reproduction System”, this work aims only to offer the interface via web for the SSR (“Real-Time Spatial Audio Reproduction System”). The idea is not to make a new graphical interface for SSR but to allow more types of interfaces and communication. To accomplish the objective of allowing more graphical interfaces this project is going to use a new network interface. By now the SSR application is using only XML for data interchange but this new network interface support JSON. This project comprehends the server that launch the application, the user interface and the new network interface. It is done with these modules in order to allow creating new user interfaces that can communicate with the server or new servers that can communicate with the user interface by defining a complete network interface for data interchange.