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Audio Signal Processing
Audio signal processing is a subfield of signal processing that is concerned with the electronic manipulation of audio signals. Audio signals are electronic representations of sound waves—longitudinal waves which travel through air, consisting of compressions and rarefactions. The energy contained in audio signals or sound power level is typically measured in decibels. As audio signals may be represented in either Digital signal (signal processing), digital or analog signal, analog format, processing may occur in either domain. Analog processors operate directly on the electrical signal, while digital processors operate mathematically on its digital representation. History The motivation for audio signal processing began at the beginning of the 20th century with inventions like the telephone, phonograph, and radio that allowed for the transmission and storage of audio signals. Audio processing was necessary for early radio broadcasting, as there were many problems with stud ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Signal Processing
Signal processing is an electrical engineering subfield that focuses on analyzing, modifying and synthesizing ''signals'', such as audio signal processing, sound, image processing, images, Scalar potential, potential fields, Seismic tomography, seismic signals, Altimeter, altimetry processing, and scientific measurements. Signal processing techniques are used to optimize transmissions, Data storage, digital storage efficiency, correcting distorted signals, improve subjective video quality, and to detect or pinpoint components of interest in a measured signal. History According to Alan V. Oppenheim and Ronald W. Schafer, the principles of signal processing can be found in the classical numerical analysis techniques of the 17th century. They further state that the digital refinement of these techniques can be found in the digital control systems of the 1940s and 1950s. In 1948, Claude Shannon wrote the influential paper "A Mathematical Theory of Communication" which was publis ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Nyquist–Shannon Sampling Theorem
The Nyquist–Shannon sampling theorem is an essential principle for digital signal processing linking the frequency range of a signal and the sample rate required to avoid a type of distortion called aliasing. The theorem states that the sample rate must be at least twice the Bandwidth (signal processing), bandwidth of the signal to avoid aliasing. In practice, it is used to select band-limiting filters to keep aliasing below an acceptable amount when an analog signal is sampled or when sample rates are changed within a digital signal processing function. The Nyquist–Shannon sampling theorem is a theorem in the field of signal processing which serves as a fundamental bridge between continuous-time signals and discrete-time signals. It establishes a sufficient condition for a sample rate that permits a discrete sequence of ''samples'' to capture all the information from a continuous-time signal of finite Bandwidth (signal processing), bandwidth. Strictly speaking, the theorem ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Nippon Telegraph And Telephone
(NTT) is a Japanese telecommunications holding company headquartered in Tokyo, Japan. Ranked 55th in ''Fortune'' Global 500, NTT is the fourth largest telecommunications company in the world in terms of revenue, as well as the third largest publicly traded company in Japan after Toyota and Sony, as of June 2022. In 2023, the company was ranked 56th in the Forbes Global 2000. NTT was the world's largest company by market capitalization in the late 1980s, and remained among the world's top 10 largest companies by market capitalization until the burst of the Dot-com bubble in the early 2000s. The company traces its origin to the national telegraph service established in 1868, which came under the purview of the Ministry of Communications in the 1880s. In 1952, the telegraph and telephone services were spun off as the government-owned . Under Prime Minister Yasuhiro Nakasone, the company was privatised in 1985 along with the Japanese National Railways and the Japan Tobacco a ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Nagoya University
, abbreviated to or NU, is a Japanese national research university located in Chikusa-ku, Nagoya. It was established in 1939 as the last of the nine Imperial Universities in the then Empire of Japan, and is now a Designated National University. The university is the birthplace of the Sakata School of physics and the Hirata School of chemistry. As of 2021, seven Nobel Prize winners have been associated with Nagoya University, the third most in Japan and Asia behind Kyoto University and the University of Tokyo. History Nagoya Imperial University was established as the last of the Imperial Universities in 1939 and was later renamed Nagoya University in 1947. Although relatively new as a university, it can trace its roots back to a Temporary Medical School/Public Hospital opened in 1871. Renowned for its contributions in physics and chemistry, the university has been the birthplace of notable scientific advancements such as the Sakata model, the PMNS matrix, the Okazak ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Fumitada Itakura
is a Japanese scientist. He did pioneering work in statistical signal processing, and its application to speech analysis, synthesis and coding, including the development of the linear predictive coding (LPC) and line spectral pairs (LSP) methods. Biography Itakura was born in Toyokawa, Aichi Prefecture, Japan. He received undergraduate and graduate degrees from Nagoya University in 1963 and 1965, respectively. In 1966, while studying his PhD at Nagoya, he developed the earliest concepts for what would later become known as linear predictive coding (LPC), along with Shuzo Saito from Nippon Telegraph and Telephone (NTT). They described an approach to automatic phoneme discrimination that involved the first maximum likelihood approach to speech coding. In 1968, he joined the NTT Musashino Electrical Communication Laboratory in Tokyo. The same year, Itakura and Saito presented the Itakura–Saito distance algorithm. The following year, Itakura and Saito introduced partial correl ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Linear Predictive Coding
Linear predictive coding (LPC) is a method used mostly in audio signal processing and speech processing for representing the spectral envelope of a digital signal of speech in compressed form, using the information of a linear predictive model. LPC is the most widely used method in speech coding and speech synthesis. It is a powerful speech analysis technique, and a useful method for encoding good quality speech at a low bit rate. Overview LPC starts with the assumption that a speech signal is produced by a buzzer at the end of a tube (for voiced sounds), with occasional added hissing and popping sounds (for voiceless sounds such as sibilants and plosives). Although apparently crude, this Source–filter model is actually a close approximation of the reality of speech production. The glottis (the space between the vocal folds) produces the buzz, which is characterized by its intensity (loudness) and frequency (pitch). The vocal tract (the throat and mouth) forms the tube, whi ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Differential Pulse-code Modulation
Differential pulse-code modulation (DPCM) is a signal encoder that uses the baseline of pulse-code modulation (PCM) but adds some functionalities based on the prediction of the samples of the signal. The input can be an analog signal or a Digital signal (signal processing), digital signal. If the input is a continuous-time analog signal, it needs to be sampling (signal processing), sampled first so that a discrete-time signal is the input to the DPCM encoder. * Option 1: take the values of two consecutive samples; if they are analog samples, quantization (signal processing), quantize them; calculate the difference between the first one and the next; the output is the difference. * Option 2: instead of taking a difference relative to a previous input sample, take the difference relative to the output of a local model of the decoder process; in this option, the difference can be quantized, which allows a good way to incorporate a controlled loss in the encoding. Applying one of thes ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Audio Data Compression
In information theory, data compression, source coding, or bit-rate reduction is the process of encoding information using fewer bits than the original representation. Any particular compression is either lossy or lossless. Lossless compression reduces bits by identifying and eliminating Redundancy (information theory), statistical redundancy. No information is lost in lossless compression. Lossy compression reduces bits by removing unnecessary or less important information. Typically, a device that performs data compression is referred to as an encoder, and one that performs the reversal of the process (decompression) as a decoder. The process of reducing the size of a data file is often referred to as data compression. In the context of data transmission, it is called source coding: encoding is done at the source of the data before it is stored or transmitted. Source coding should not be confused with channel coding, for error detection and correction or line coding, the means ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Audio Coding
An audio coding format (or sometimes audio compression format) is a content representation format for storage or transmission of digital audio (such as in digital television, digital radio and in audio and video files). Examples of audio coding formats include MP3, AAC, Vorbis, FLAC, and Opus. A specific software or hardware implementation capable of audio compression and decompression to/from a specific audio coding format is called an '' audio codec''; an example of an audio codec is LAME, which is one of several different codecs which implements encoding and decoding audio in the MP3 audio coding format in software. Some audio coding formats are documented by a detailed technical specification document known as an audio coding specification. Some such specifications are written and approved by standardization organizations as technical standards, and are thus known as an audio coding standard. The term "standard" is also sometimes used for ''de facto'' standards as wel ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Digital Audio
Digital audio is a representation of sound recorded in, or converted into, digital signal (signal processing), digital form. In digital audio, the sound wave of the audio signal is typically encoded as numerical sampling (signal processing), samples in a continuous sequence. For example, in CD audio, samples are taken 44,100 Hertz, times per second, each with 16-bit audio bit depth, resolution. Digital audio is also the name for the entire technology of sound recording and reproduction using audio signals that have been encoded in digital form. Following significant advances in digital audio technology during the 1970s and 1980s, it gradually replaced comparison of analog and digital recording, analog audio technology in many areas of audio engineering, record production and telecommunications in the 1990s and 2000s. In a digital audio system, an analog signal, analog electrical signal representing the sound is converted with an analog-to-digital converter (ADC) into a digital ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Computer Music
Computer music is the application of computing technology in music composition, to help human composers create new music or to have computers independently create music, such as with algorithmic composition programs. It includes the theory and application of new and existing computer software technologies and basic aspects of music, such as sound synthesis, digital signal processing, sound design, sonic diffusion, acoustics, electrical engineering, and psychoacoustics. The field of computer music can trace its roots back to the origins of electronic music, and the first experiments and innovations with electronic instruments at the turn of the 20th century. History Much of the work on computer music has drawn on the relationship between music and mathematics, a relationship that has been noted since the Ancient Greeks described the " harmony of the spheres". Musical melodies were first generated by the computer originally named the CSIR Mark 1 (later renamed CSIRAC) in Austra ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Computer
A computer is a machine that can be Computer programming, programmed to automatically Execution (computing), carry out sequences of arithmetic or logical operations (''computation''). Modern digital electronic computers can perform generic sets of operations known as Computer program, ''programs'', which enable computers to perform a wide range of tasks. The term computer system may refer to a nominally complete computer that includes the Computer hardware, hardware, operating system, software, and peripheral equipment needed and used for full operation; or to a group of computers that are linked and function together, such as a computer network or computer cluster. A broad range of Programmable logic controller, industrial and Consumer electronics, consumer products use computers as control systems, including simple special-purpose devices like microwave ovens and remote controls, and factory devices like industrial robots. Computers are at the core of general-purpose devices ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
