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Adaptive DPCM
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. This s ... [...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. If the input is a continuous-time analog signal, it needs to be 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, 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 these two processes, short-term redundancy (positive correlation of nearby values) of the signal is ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Codec
A codec is a device or computer program that encodes or decodes a data stream or signal. ''Codec'' is a portmanteau of coder/decoder. In electronic communications, an endec is a device that acts as both an encoder and a decoder on a signal or data stream, and hence is a type of codec. ''Endec'' is a portmanteau of encoder/decoder. A coder or encoder encodes a data stream or a signal for transmission or storage, possibly in encrypted form, and the decoder function reverses the encoding for playback or editing. Codecs are used in videoconferencing, streaming media, and video editing applications. History In the mid-20th century, a codec was a device that coded analog signals into digital form using pulse-code modulation (PCM). Later, the name was also applied to software for converting between digital signal formats, including companding functions. Examples An audio codec converts analog audio signals into digital signals for transmission or encodes them for storage. A receiv ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Pulse-code Modulation
Pulse-code modulation (PCM) is a method used to digitally represent sampled analog signals. It is the standard form of digital audio in computers, compact discs, digital telephony and other digital audio applications. In a PCM stream, the amplitude of the analog signal is sampled regularly at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps. Linear pulse-code modulation (LPCM) is a specific type of PCM in which the quantization levels are linearly uniform. This is in contrast to PCM encodings in which quantization levels vary as a function of amplitude (as with the A-law algorithm or the μ-law algorithm). Though ''PCM'' is a more general term, it is often used to describe data encoded as LPCM. A PCM stream has two basic properties that determine the stream's fidelity to the original analog signal: the sampling rate, which is the number of times per second that samples are taken; and the bit depth, which determines the ... [...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 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 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 signal. C ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Audio Coding Format
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 well as forma ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
GitHub
GitHub, Inc. () is an Internet hosting service for software development and version control using Git. It provides the distributed version control of Git plus access control, bug tracking, software feature requests, task management, continuous integration, and wikis for every project. Headquartered in California, it has been a subsidiary of Microsoft since 2018. It is commonly used to host open source software development projects. As of June 2022, GitHub reported having over 83 million developers and more than 200 million repositories, including at least 28 million public repositories. It is the largest source code host . History GitHub.com Development of the GitHub.com platform began on October 19, 2007. The site was launched in April 2008 by Tom Preston-Werner, Chris Wanstrath, P. J. Hyett and Scott Chacon after it had been made available for a few months prior as a beta release. GitHub has an annual keynote called GitHub Universe. Organizational ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
FFmpeg
FFmpeg is a free and open-source software project consisting of a suite of libraries and programs for handling video, audio, and other multimedia files and streams. At its core is the command-line ffmpeg tool itself, designed for processing of video and audio files. It is widely used for format transcoding, basic editing (trimming and concatenation), video scaling, video post-production effects and standards compliance (SMPTE, ITU). FFmpeg also includes other tools: ffplay, a simple media player and ffprobe, a command-line tool to display media information. Among included libraries are libavcodec, an audio/video codec library used by many commercial and free software products, libavformat (Lavf), an audio/video container mux and demux library, and libavfilter, a library for enhancing and editing filters through a Gstreamer-like filtergraph. FFmpeg is part of the workflow of many other software projects, and its libraries are a core part of software media players such as VLC, an ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Microsoft Knowledge Base
Microsoft Knowledge Base was (as of 2020, almost all articles now produce the message : Sorry, page not found) a repository of over 150,000 articles made available to the public by Microsoft Corporation. It contains information on many problems encountered by users of Microsoft products. Each article bears an ID number and articles are often referred to by their Knowledge Base (KB) ID. Microsoft Windows update names typically start with the letters "KB", this is in reference to the specific article on that issue. Previously, the "Q" letter was used. kbalertz.com was a website that provided email alerts of new articles, although Microsoft recently has provided a similar service. See also * MSDN Library * Diffbot * Windows Update References External links * {{Microsoft-stub Microsoft culture Microsoft websites ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Windows Sound System
Windows Sound System (WSS) is a sound card specification developed by Microsoft released at the end of 1992 for Windows 3.1. WSS featured support for up to 16-bit, 48 kHz digital sampling, beyond the capabilities of the popular contemporary Sound Blaster Pro, although it was less frequently supported than Sound Blaster and Gravis sound cards, as well as Roland sound cards, daughterboards, and sound modules. In addition, the WSS featured RCA analog audio outputs, an uncommon feature among sound cards of this era; other connections were a microphone input, a stereo line input and a stereo headphone output. The Windows Sound System was sold as a bundle which included an ISA sound card, a microphone, a pair of headphones and the software package. WSS 1.0a drivers were released in February 1993. They introduced single-mode DMA, supported games in MS-DOS, Ad Lib and Sound Blaster emulation.WSS 2.0 drivers, released in October 1993, added support for OEM sound cards (Media Visi ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Adpcm En
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. This sy ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Quadrature Mirror Filter
In digital signal processing, a quadrature mirror filter is a filter whose magnitude response is the mirror image around \pi/2 of that of another filter. Together these filters, first introduced by Croisier et al., are known as the quadrature mirror filter pair. A filter H_1(z) is the quadrature mirror filter of H_0(z) if H_1(z) = H_0(-z). The filter responses are symmetric about \Omega = \pi / 2: : \big, H_1\big(e^\big)\big, = \big, H_0\big(e^\big)\big, . In audio/voice codecs, a quadrature mirror filter pair is often used to implement a filter bank that splits an input signal into two bands. The resulting high-pass and low-pass signals are often reduced by a factor of 2, giving a critically sampled two-channel representation of the original signal. The analysis filters are often related by the following formula in addition to quadrate mirror property: : \big, H_0\big(e^\big)\big, ^2 + \big, H_1\big(e^\big)\big, ^2 = 1, where \Omega is the frequency, and the sampling rate is no ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
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, 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 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 to represent each sample, the fine ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
