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Subband Coding
In signal processing, sub-band coding (SBC) is any form of transform coding that breaks a signal into a number of different frequency bands, typically by using a fast Fourier transform, and encodes each one independently. This decomposition is often the first step in data compression for audio and video signals. SBC is the core technique used in many popular lossy audio compression algorithms including MP3. Encoding audio signals The simplest way to digitally encode audio signals is pulse-code modulation (PCM), which is used on audio CDs, Digital Audio Tape, DAT recordings, and so on. Digitization transforms continuous signals into discrete ones by sampling a signal's amplitude at uniform intervals and rounding to the nearest value representable with the available Audio bit depth, number of bits. This process is fundamentally inexact, and involves two errors: ''discretization error,'' from sampling at intervals, and ''quantization error,'' from rounding. The more bits used t ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Dither
Dither is an intentionally applied form of noise used to randomize quantization error, preventing large-scale patterns such as color banding in images. Dither is routinely used in processing of both digital audio and video data, and is often one of the last stages of mastering audio to a CD. A common use of dither is converting a grayscale image to black and white, so that the density of black dots in the new image approximates the average gray level in the original. Etymology The term ''dither'' was published in books on analog computation and hydraulically controlled guns shortly after World War II. Though he did not use the term ''dither'', the concept of dithering to reduce quantization patterns was first applied by Lawrence G. Roberts in his 1961 MIT master's thesis and 1962 article. By 1964 dither was being used in the modern sense described in this article. The technique was in use at least as early as 1915, though not under the name ''dither''. In digital pro ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
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 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 for mapping data onto a sig ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
ADPCM
Adaptive differential pulse-code modulation (ADPCM) is a variant of differential pulse-code modulation (DPCM) that varies the size of the quantization step, to allow further reduction of the required data bandwidth for a given signal-to-noise ratio. Typically, the adaptation to signal statistics in ADPCM consists simply of an adaptive scale factor before quantizing the difference in the DPCM encoder. ADPCM was developed for speech coding by P. Cummiskey, Nikil S. Jayant and James L. Flanagan at Bell Labs in 1973. In telephony In telephony, a standard audio signal for a single phone call is encoded as 8000 analog samples per second, of 8 bits each, giving a 64 kbit/s digital signal known as DS0. The default signal compression encoding on a DS0 is either μ-law (mu-law) PCM (North America and Japan) or A-law PCM (Europe and most of the rest of the world). These are logarithmic compression systems where a 13- or 14-bit linear PCM sample number is mapped into an 8-bit value ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
MPEG-1 Audio Layer III
MP3 (formally MPEG-1 Audio Layer III or MPEG-2 Audio Layer III) is a coding format for digital audio developed largely by the Fraunhofer Society in Germany under the lead of Karlheinz Brandenburg. It was designed to greatly reduce the amount of data required to represent audio, yet still sound like a faithful reproduction of the original uncompressed audio to most listeners; for example, compared to CD-quality digital audio, MP3 compression can commonly achieve a 75–95% reduction in size, depending on the bit rate. In popular usage, ''MP3'' often refers to files of sound or music recordings stored in the MP3 file format (.mp3) on consumer electronic devices. Originally defined in 1991 as one of the three audio codecs of the MPEG-1 standard (along with MP2 and MP1), it was retained and further extended—defining additional bit rates and support for more audio channels—as the third audio format of the subsequent MPEG-2 standard. MP3 as a file format commonly designat ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Moving Picture Experts Group
The Moving Picture Experts Group (MPEG) is an alliance of working groups established jointly by ISO and IEC that sets standards for media coding, including compression coding of audio, video, graphics, and genomic data; and transmission and file formats for various applications.John Watkinson, ''The MPEG Handbook'', p. 1 Together with JPEG, MPEG is organized under ISO/IEC JTC 1/ SC 29 – ''Coding of audio, picture, multimedia and hypermedia information'' (ISO/IEC Joint Technical Committee 1, Subcommittee 29). MPEG formats are used in various multimedia systems. The most well known older MPEG media formats typically use MPEG-1, MPEG-2, and MPEG-4 AVC media coding and MPEG-2 systems transport streams and program streams. Newer systems typically use the MPEG base media file format and dynamic streaming (a.k.a. MPEG-DASH). History MPEG was established in 1988 by the initiative of Dr. Hiroshi Yasuda ( NTT) and Dr. Leonardo Chiariglione ( CSELT). Chiariglione was th ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Auditory System
The auditory system is the sensory system for the sense of hearing. It includes both the ear, sensory organs (the ears) and the auditory parts of the sensory system. System overview The outer ear funnels sound vibrations to the eardrum, increasing the sound pressure in the middle frequency range. The Middle ear, middle-ear ossicles further amplify the vibration pressure roughly 20 times. The base of the stapes couples vibrations into the cochlea via the oval window, which vibrates the perilymph liquid (present throughout the inner ear) and causes the round window to bulb out as the oval window bulges in. Vestibular duct, Vestibular and tympanic ducts are filled with perilymph, and the smaller cochlear duct between them is filled with endolymph, a fluid with a very different ion concentration and voltage. Vestibular duct perilymph vibrations bend organ of Corti outer cells (4 lines) causing prestin to be released in cell tips. This causes the cells to be chemically elongated an ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Auditory Masking
In audio signal processing, auditory masking occurs when the perception of one sound is affected by the presence of another sound.Gelfand, S.A. (2004) ''Hearing – An Introduction to Psychological and Physiological Acoustics'' 4th Ed. New York, Marcel Dekker Auditory masking in the frequency domain is known as simultaneous masking, frequency masking or spectral masking. Auditory masking in the time domain is known as temporal masking or non-simultaneous masking. Masked threshold The ''unmasked threshold'' is the quietest level of the signal which can be perceived without a masking signal present. The ''masked threshold'' is the quietest level of the signal perceived when combined with a specific masking noise. The amount of masking is the difference between the masked and unmasked thresholds. Gelfand provides a basic example.Gelfand, S.A. (2004) ''Hearing – An Introduction to Psychological and Physiological Acoustics'' 4th Ed. New York, Marcel Dekker Let us say that for a g ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Au File Format
The Au file format is a simple audio file format introduced by Sun Microsystems. The format was common on NeXT systems and on early Web pages. Originally it was headerless, being 8-bit μ-law algorithm, μ-law-encoded data at an 8000 Hz sample rate. Hardware from other vendors often used sample rates as high as 8192 Hz, often integer multiples of video clock signal frequencies. Newer files have a header that consists of six Signedness, unsigned 32-bit computing, 32-bit words, an optional information chunk which is always of non-zero size, and then the data (in Endianness, big-endian format). Although the format now supports many digital audio, audio encoding formats, it remains associated with the μ-law logarithmic encoding. This encoding was native to the SPARCstation 1 hardware, where SunOS exposed the encoding to application programs through the /dev/audio device file interface. This encoding and interface became a de facto standard for Unix sound. New format All fields a ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
μ-law Algorithm
The μ-law algorithm (sometimes written Mu (letter), mu-law, often abbreviated as u-law) is a companding algorithm, primarily used in 8-bit PCM Digital data, digital telecommunications systems in North America and Japan. It is one of the two companding algorithms in the G.711 standard from ITU-T, the other being the similar A-law. A-law is used in regions where digital telecommunication signals are carried on E-1 circuits, e.g. Europe. The terms PCMU, G711u or G711MU are used for G711 μ-law. Companding algorithms reduce the dynamic range of an audio signal. In analog systems, this can increase the signal-to-noise ratio (SNR) achieved during transmission; in the digital domain, it can reduce the quantization error (hence increasing the signal-to-quantization-noise ratio). These SNR increases can be traded instead for reduced Bandwidth (signal processing), bandwidth for equivalent SNR. At the cost of a reduced peak SNR, it can be mathematically shown that μ-law's non-linear q ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Kbit/s
In telecommunications, data transfer rate is the average number of bits ( bitrate), characters or symbols ( baudrate), or data blocks per unit time passing through a communication link in a data-transmission system. Common data rate units are multiples of bits per second (bit/s) and bytes per second (B/s). For example, the data rates of modern residential high-speed Internet connections are commonly expressed in megabits per second (Mbit/s). Standards for unit symbols and prefixes Unit symbol The ISQ symbols for the bit and byte are ''bit'' and ''B'', respectively. In the context of data-rate units, one byte consists of 8 bits, and is synonymous with the unit octet. The abbreviation bps is often used to mean bit/s, so that when a ''1 Mbps'' connection is advertised, it usually means that the maximum achievable bandwidth is 1 Mbit/s (one million bits per second), which is 0.125 MB/s ( megabyte per second), or about 0.1192 MiB/s ( mebibyte per second). ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Bitrate
In telecommunications and computing, bit rate (bitrate or as a variable ''R'') is the number of bits that are conveyed or processed per unit of time. The bit rate is expressed in the unit bit per second (symbol: bit/s), often in conjunction with an SI prefix such as kilo (1 kbit/s = 1,000 bit/s), mega (1 Mbit/s = 1,000 kbit/s), giga (1 Gbit/s = 1,000 Mbit/s) or tera (1 Tbit/s = 1,000 Gbit/s). The non-standard abbreviation bps is often used to replace the standard symbol bit/s, so that, for example, 1 Mbps is used to mean one million bits per second. In most computing and digital communication environments, one byte per second (symbol: B/s) corresponds roughly to 8 bit/s. However if stop bits, start bits, and parity bits need to be factored in, a higher number of bits per second will be required to achieve a throughput of the same number of bytes. Prefixes When quantifying large or small bit rates, SI ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
