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PCM
Pulse-code modulation (PCM) is a method used to Digital signal (signal processing), digitally represent 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 (computing), stream, the amplitude of the analog signal is Sampling (signal processing), sampled at uniform intervals, and each sample is Quantization (signal processing), quantized to the nearest value within a range of digital steps. Alec Reeves, Claude Shannon, Barney Oliver and John R. Pierce are credited with its invention. 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 ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Audio Bit Depth
In digital audio using pulse-code modulation (PCM), bit depth is the number of bits of information in each sample, and it directly corresponds to the resolution of each sample. Examples of bit depth include Compact Disc Digital Audio, which uses 16 bits per sample, and DVD-Audio and Blu-ray Disc, which can support up to 24 bits per sample. In basic implementations, variations in bit depth primarily affect the noise level from quantization error—thus the signal-to-noise ratio (SNR) and dynamic range. However, techniques such as dithering, noise shaping, and oversampling can mitigate these effects without changing the bit depth. Bit depth also affects bit rate and file size. Bit depth is useful for describing PCM digital signals. Non-PCM formats, such as those using lossy compression, do not have associated bit depths. Binary representation A PCM signal is a sequence of digital audio samples containing the data providing the necessary information to reconst ... [...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] |
Audio File Format
An audio file format is a file format for storing digital audio data on a computer system. The bit layout of the audio data (excluding metadata) is called the audio coding format and can be uncompressed, or audio compression (data), compressed to reduce the file size, often using lossy compression. The data can be a raw bitstream in an audio coding format, but it is usually embedded in a container format or an audio data format with defined storage layer. Format types It is important to distinguish between the audio coding format, the Container format (digital), container containing the Raw audio format, raw audio data, and an audio codec. A codec performs the encoding and decoding of the raw audio data while this encoded data is (usually) stored in a container file. Although most audio file formats support only one type of audio coding data (created with an audio codec, audio coder), a multimedia container format (as Matroska or Audio Video Interleave, AVI) may support multiple ... [...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] |
Red Book (audio CD Standard)
Compact Disc Digital Audio (CDDA or CD-DA), also known as Digital Audio Compact Disc or simply as Audio CD, is the standard format for audio compact discs. The standard is defined in the '' Red Book'' technical specifications, which is why the format is also dubbed ''"Redbook audio"'' in some contexts. CDDA utilizes pulse-code modulation (PCM) and uses a 44,100 Hz sampling frequency and 16-bit resolution, and was originally specified to store up to 74 minutes of stereo audio per disc. The first commercially available audio CD player, the Sony CDP-101, was released in October 1982 in Japan. The format gained worldwide acceptance in 1983–84, selling more than a million CD players in its first two years, to play 22.5 million discs, before overtaking records and cassette tapes to become the dominant standard for commercial music. Peaking around year 2000, the audio CD contracted over the next decade due to rising popularity and revenue from digital downloading, and duri ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Telephony
Telephony ( ) is the field of technology involving the development, application, and deployment of telecommunications services for the purpose of electronic transmission of voice, fax, or data, between distant parties. The history of telephony is intimately linked to the invention and development of the telephone. Telephony is commonly referred to as the construction or operation of telephones and telephonic systems and as a system of telecommunications in which telephonic equipment is employed in the transmission of speech or other sound between points, with or without the use of wires. The term is also used frequently to refer to computer hardware, software, and computer network systems, that perform functions traditionally performed by telephone equipment. In this context the technology is specifically referred to as Internet telephony, or voice over Internet Protocol (VoIP). Overview The first telephones were connected directly in pairs: each user had a separate telephone wire ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Sampling Rate
In signal processing, sampling is the reduction of a continuous-time signal to a discrete-time signal. A common example is the conversion of a sound wave to a sequence of "samples". A sample is a value of the signal at a point in time and/or space; this definition differs from the term's usage in statistics, which refers to a set of such values. A sampler is a subsystem or operation that extracts samples from a continuous signal. A theoretical ideal sampler produces samples equivalent to the instantaneous value of the continuous signal at the desired points. The original signal can be reconstructed from a sequence of samples, up to the Nyquist limit, by passing the sequence of samples through a reconstruction filter. Theory Functions of space, time, or any other dimension can be sampled, and similarly in two or more dimensions. For functions that vary with time, let s(t) be a continuous function (or "signal") to be sampled, and let sampling be performed by measuring t ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Sampling (signal Processing)
In signal processing, sampling is the reduction of a continuous-time signal to a discrete-time signal. A common example is the conversion of a sound wave to a sequence of "samples". A sample is a value of the signal at a point in time and/or space; this definition differs from the term's usage in statistics, which refers to a set of such values. A sampler is a subsystem or operation that extracts samples from a continuous signal. A theoretical ideal sampler produces samples equivalent to the instantaneous value of the continuous signal at the desired points. The original signal can be reconstructed from a sequence of samples, up to the Nyquist limit, by passing the sequence of samples through a reconstruction filter. Theory Functions of space, time, or any other dimension can be sampled, and similarly in two or more dimensions. For functions that vary with time, let s(t) be a continuous function (or "signal") to be sampled, and let sampling be performed by measuring ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Audio Interchange File Format
Audio Interchange File Format (AIFF) is an audio file format standard used for storing sound data for personal computers and other electronic audio devices. The format was developed by Apple Inc. in 1988 based on Electronic Arts' Interchange File Format (IFF, widely used on Amiga systems) and is most commonly used on Apple Macintosh computer systems. The audio data in most AIFF files is uncompressed pulse-code modulation (PCM). This type of AIFF file uses much more disk space than lossy formats like MP3—about 10 MB for one minute of stereo audio at a sample rate of 44.1 kHz and a bit depth of 16 bits. There is also a compressed variant of AIFF known as AIFF-C or AIFC, with various defined compression codecs. In addition to audio data, AIFF can include loop point data and the musical note of a sample, for use by hardware samplers and musical applications. The file extension for the standard AIFF format is .aiff or .aif. For the compressed format the preferred suffi ... [...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] |
A-law Algorithm
An A-law algorithm is a standard companding algorithm, used in European 8-bit PCM digital communications systems to optimize, i.e. modify, the dynamic range of an analog signal for digitizing. It is one of the two companding algorithms in the G.711 standard from ITU-T, the other being the similar μ-law, used in North America and Japan. For a given input x, the equation for A-law encoding is as follows: F(x) = \sgn(x) \begin \dfrac, & , x, < \dfrac, \\ ex \dfrac, & \dfrac \leq , x, \leq 1, \end where is the compression parameter. In Europe, . A-law expansion is given by the inverse function: The reason for this encoding is that the wide |
