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Sound Quality
Sound quality is typically an assessment of the accuracy, fidelity, or Intelligibility (communication), intelligibility of sound, audio output from an electronic device. Quality can be measured objectively, such as when tools are used to gauge the accuracy with which the device reproduces an original sound; or it can be measured subjectively, such as when human listeners respond to the sound or gauge its ''perceived'' similarity to another sound. The sound quality of a reproduction or recording depends on a number of factors, including the equipment used to make it, processing and mastering done to the recording, the equipment used to reproduce it, as well as the listening environment used to reproduce it. In some cases, processing such as Equalization (audio), equalization, dynamic range compression or 3D audio effect, stereo processing may be applied to a recording to create audio that is significantly different from the original but may be perceived as more agreeable to a ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Audio System Measurements
Audio system measurements are used to quantify audio system performance. These measurements are made for several purposes. Designers take measurements to specify the performance of a piece of equipment. Maintenance engineers make them to ensure equipment is still working to specification, or to ensure that the cumulative defects of an audio path are within limits considered acceptable. Audio system measurements often accommodate psychoacoustic principles to measure the system in a way that relates to human hearing. Subjectivity and frequency weighting Subjectively valid methods came to prominence in consumer audio in the UK and Europe in the 1970s, when the introduction of compact cassette tape, dbx (noise reduction), dbx and Dolby noise reduction techniques revealed the unsatisfactory nature of many basic engineering measurements. The specification of ITU-R 468 noise weighting, weighted CCIR-468 quasi-peak noise, and flutter measurement, weighted quasi-peak wow and flutter b ... [...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] |
Lossy Audio 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] |
Monkey's Audio
Monkey's Audio is an algorithm and file format for lossless audio data compression. Lossless data compression does not discard data during the process of encoding, unlike lossy compression methods such as Advanced Audio Coding, MP3, Vorbis, and Opus. Similar to other lossless audio codecs, files encoded to Monkey's Audio are typically reduced to about half of the original size, with data transfer time and storage requirements being reduced accordingly. Comparisons Like any lossless compression scheme, Monkey's Audio format takes up several times as much space as lossy compression formats - typically, about twice as much as a 320 kbit/s bitrate MP3 file. The upside is that no data is lost compared to the input file, making lossless codecs suitable for transcoding, or simply taking up approximately half as much space as raw PCM data. Relative to FLAC, Apple Lossless Audio Codec, or WavPack, Monkey's Audio is slow to encode or decode files. While Monkey's Audio can achiev ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Apple Lossless
The Apple Lossless Audio Codec (ALAC, ), also known as Apple Lossless, or Apple Lossless Encoder (ALE), is an audio coding format, and its reference audio codec implementation, developed by Apple Inc., Apple for lossless data compression of digital music. After initially keeping it proprietary format, proprietary from its inception in 2004, in late 2011 Apple made the codec available open-source software, open source and royalty-free. Traditionally, Apple has referred to the codec as ''Apple Lossless'', though more recently it has begun to use the abbreviated term ''ALAC'' when referring to the codec. ALAC data is frequently stored within an MPEG-4 Part 14, MP4 Container format (digital), container with the filename extension ''.m4a''. This extension is also used by Apple for Advanced Audio Coding, AAC (which is a Lossy compression, lossy format) in an MP4 container (same container, different audio encoding). ALAC can also be used by the Core Audio Format, .CAF file type container ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
FLAC
FLAC (; Free Lossless Audio Codec) is an audio coding format for lossless compression of digital audio, developed by the Xiph.Org Foundation, and is also the name of the free software project producing the FLAC tools, the reference software package that includes a codec implementation. Digital audio compressed by FLAC's algorithm can typically be reduced to between 50 and 70 percent of its original size and decompresses to an identical copy of the original audio data. FLAC is an open format with royalty-free licensing and a reference implementation which is free software. FLAC supports metadata tagging, album cover art, and fast seeking. History Development was started in 2000 by Josh Coalson. The bitstream format was frozen with the release of version 0.9 of the reference implementation on 31 March 2001. Version 1.0 was released on 20 July 2001. On 29 January 2003, the Xiph.Org Foundation and the FLAC project announced the incorporation of FLAC under the Xiph.org banner. ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Lossless Codec
Lossless compression is a class of data compression that allows the original data to be perfectly reconstructed from the compressed data with no loss of information. Lossless compression is possible because most real-world data exhibits statistical redundancy. By contrast, lossy compression permits reconstruction only of an approximation of the original data, though usually with greatly improved compression rates (and therefore reduced media sizes). By operation of the pigeonhole principle, no lossless compression algorithm can shrink the size of all possible data: Some data will get longer by at least one symbol or bit. Compression algorithms are usually effective for human- and machine-readable documents and cannot shrink the size of random data that contain no redundancy. Different algorithms exist that are designed either with a specific type of input data in mind or with specific assumptions about what kinds of redundancy the uncompressed data are likely to contain. Los ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Audio Codec
An audio codec is a device or computer program capable of encoding or decoding a digital data stream (a codec) that encodes or decodes audio. In software, an audio codec is a computer program implementing an algorithm that compresses and decompresses digital audio data according to a given audio file or streaming media audio coding format. The objective of the algorithm is to represent the high-fidelity audio signal with a minimum number of bits while retaining quality. This can effectively reduce the storage space and the bandwidth required for transmission of the stored audio file. Most software codecs are implemented as libraries which interface to one or more multimedia players. Most modern audio compression algorithms are based on modified discrete cosine transform (MDCT) coding and linear predictive coding (LPC). In hardware, audio codec refers to a single device that encodes analog audio as digital signals and decodes digital back into analog. In other words, it contai ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Audio Compression (data)
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] |
Decibels
The decibel (symbol: dB) is a relative unit of measurement equal to one tenth of a bel (B). It expresses the ratio of two values of a power or root-power quantity on a logarithmic scale. Two signals whose levels differ by one decibel have a power ratio of 101/10 (approximately ) or root-power ratio of 101/20 (approximately ). The strict original usage above only expresses a relative change. However, the word decibel has since also been used for expressing an absolute value that is relative to some fixed reference value, in which case the dB symbol is often suffixed with letter codes that indicate the reference value. For example, for the reference value of 1 volt, a common suffix is " V" (e.g., "20 dBV"). As it originated from a need to express power ratios, two principal types of scaling of the decibel are used to provide consistency depending on whether the scaling refers to ratios of power quantities or root-power quantities. When expressing a power ratio, it ... [...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] |
Quantization Error
Quantization, in mathematics and digital signal processing, is the process of mapping input values from a large set (often a continuous set) to output values in a (countable) smaller set, often with a finite number of elements. Rounding and truncation are typical examples of quantization processes. Quantization is involved to some degree in nearly all digital signal processing, as the process of representing a signal in digital form ordinarily involves rounding. Quantization also forms the core of essentially all lossy compression algorithms. The difference between an input value and its quantized value (such as round-off error) is referred to as quantization error, noise or distortion. A device or algorithmic function that performs quantization is called a quantizer. An analog-to-digital converter is an example of a quantizer. Example For example, rounding a real number x to the nearest integer value forms a very basic type of quantizer – a ''uniform'' one. A typi ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
