Sound quality

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Microphone covers are occasionally used to improve sound quality by reducing noise from wind. Microphone and cover.JPG
Microphone covers are occasionally used to improve sound quality by reducing noise from wind.

Sound quality is typically an assessment of the accuracy, fidelity, or intelligibility of 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. [1]

Contents

The sound quality of a reproduction or recording depends on a number of factors, including the equipment used to make it, [2] processing and mastering done to the recording, the equipment used to reproduce it, as well as the listening environment used to reproduce it. [3] In some cases, processing such as equalization, dynamic range compression or 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 listener. In other cases, the goal may be to reproduce audio as closely as possible to the original.

When applied to specific electronic devices, such as loudspeakers, microphones, amplifiers or headphones sound quality usually refers to accuracy, with higher quality devices providing higher accuracy reproduction. When applied to processing steps such as mastering recordings, absolute accuracy may be secondary to artistic or aesthetic concerns. In still other situations, such as recording a live musical performance, audio quality may refer to proper placement of microphones around a room to optimally use room acoustics.

Digital audio

Digital audio is stored in many formats. The simplest form is uncompressed PCM, where audio is stored as a series of quantized audio samples spaced at regular intervals in time. [4] As samples are placed closer together in time, higher frequencies can be reproduced. According to the sampling theorem, any bandwidth-limited signal (that does not contain a pure sinusoidal component), bandwidth B, can be perfectly described by more than 2B samples per second, allowing perfect reconstruction of the bandwidth-limited analog signal. [5] For example, for human hearing bandwidth between 0 and 20 kHz, audio must be sampled at above 40 kHz. Due to the need for filtering out ultrasonic frequencies resulting from the conversion to an analog signal, in practice slightly higher sample rates are used: 44.1 kHz (CD audio) or 48 kHz (DVD).

In PCM, each audio sample describes the sound pressure at an instant in time with a limited precision. The limited accuracy results in quantization error, a form of noise that is added to the recording. To reduce quantization error, more precision can be used in each measurement at the expense of larger samples (see audio bit depth). With each additional bit added to a sample, quantization error is reduced by approximately 6 dB. For example, CD audio uses 16 bits per sample, and therefore it will have quantization noise approximately 96 dB below the maximum possible sound pressure level (when summed over the full bandwidth)

The amount of space required to store PCM depends on the number of bits per sample, the number of samples per second, and the number of channels. For CD audio, this is 44,100 samples per second, 16 bits per sample, and 2 channels for stereo audio leading to 1,411,200 bits per second. However, this space can be greatly reduced using audio compression. In audio compression, audio samples are processed using an audio codec. In a lossless codec audio samples are processed without discarding information by packing repetitive or redundant samples into a more efficiently stored form. A lossless decoder then reproduces the original PCM with no change in quality. Lossless audio compression typically achieves a 30-50% reduction in file size. Common lossless audio codecs include FLAC, ALAC, Monkey's Audio and others.

If additional compression is required, lossy audio compression such as MP3, Ogg Vorbis or AAC can be used. In these techniques, lossless compression techniques are enhanced by processing audio to reduce the precision of details that are unlikely or impossible for human hearing to perceive using principles from psychoacoustics. After the removal of these details, lossy compression can be applied to the remainder to greatly reduce the file size. Lossy audio compression therefore allows a 75-95% reduction in file size, but runs the risk of potentially reducing audio quality if important information is mistakenly discarded.

See also

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An audio file format is a file format for storing digital audio data on a computer system. The bit layout of the audio data is called the audio coding format and can be uncompressed, or 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.

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 simply 8-bit 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 unsigned 32-bit words, an optional information chunk which is always of non-zero size, and then the data.

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 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.

<span class="mw-page-title-main">Lossy compression</span> Data compression approach that reduces data size while discarding or changing some of it

In information technology, lossy compression or irreversible compression is the class of data compression methods that uses inexact approximations and partial data discarding to represent the content. These techniques are used to reduce data size for storing, handling, and transmitting content. The different versions of the photo of the cat on this page show how higher degrees of approximation create coarser images as more details are removed. This is opposed to lossless data compression which does not degrade the data. The amount of data reduction possible using lossy compression is much higher than using lossless techniques.

Adaptive Transform Acoustic Coding (ATRAC) is a family of proprietary audio compression algorithms developed by Sony. MiniDisc was the first commercial product to incorporate ATRAC, in 1992. ATRAC allowed a relatively small disc like MiniDisc to have the same running time as CD while storing audio information with minimal perceptible loss in quality. Improvements to the codec in the form of ATRAC3, ATRAC3plus, and ATRAC Advanced Lossless followed in 1999, 2002, and 2006 respectively.

<span class="mw-page-title-main">Digital audio</span> Technology that records, stores, and reproduces sound

Digital audio is a representation of sound recorded in, or converted into, digital form. In digital audio, the sound wave of the audio signal is typically encoded as numerical samples in a continuous sequence. For example, in CD audio, samples are taken 44,100 times per second, each with 16-bit sample depth. 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 analog audio technology in many areas of audio engineering, record production and telecommunications in the 1990s and 2000s.

<span class="mw-page-title-main">G.711</span> ITU-T recommendation

G.711 is a narrowband audio codec originally designed for use in telephony that provides toll-quality audio at 64 kbit/s. It is an ITU-T standard (Recommendation) for audio encoding, titled Pulse code modulation (PCM) of voice frequencies released for use in 1972.

Sound can be recorded and stored and played using either digital or analog techniques. Both techniques introduce errors and distortions in the sound, and these methods can be systematically compared. Musicians and listeners have argued over the superiority of digital versus analog sound recordings. Arguments for analog systems include the absence of fundamental error mechanisms which are present in digital audio systems, including aliasing and associated anti-aliasing filter implementation, jitter and quantization noise. Advocates of digital point to the high levels of performance possible with digital audio, including excellent linearity in the audible band and low levels of noise and distortion.

In telecommunications and computing, bit rate is the number of bits that are conveyed or processed per unit of time.

Transcoding is the direct digital-to-digital conversion of one encoding to another, such as for video data files, audio files, or character encoding. This is usually done in cases where a target device does not support the format or has limited storage capacity that mandates a reduced file size, or to convert incompatible or obsolete data to a better-supported or modern format.

WavPack is a free and open-source lossless audio compression format and application implementing the format. It is unique in the way that it supports hybrid audio compression alongside normal compression which is similar to how FLAC works. It also supports compressing a wide variety of lossless formats, including various variants of PCM and also DSD as used in SACDs, together with its support for surround audio.

Generation loss is the loss of quality between subsequent copies or transcodes of data. Anything that reduces the quality of the representation when copying, and would cause further reduction in quality on making a copy of the copy, can be considered a form of generation loss. File size increases are a common result of generation loss, as the introduction of artifacts may actually increase the entropy of the data through each generation.

Dolby Digital Plus, also known as Enhanced AC-3, is a digital audio compression scheme developed by Dolby Labs for the transport and storage of multi-channel digital audio. It is a successor to Dolby Digital (AC-3), and has a number of improvements over that codec, including support for a wider range of data rates, an increased channel count, and multi-program support, as well as additional tools (algorithms) for representing compressed data and counteracting artifacts. Whereas Dolby Digital (AC-3) supports up to five full-bandwidth audio channels at a maximum bitrate of 640 kbit/s, E-AC-3 supports up to 15 full-bandwidth audio channels at a maximum bitrate of 6.144 Mbit/s.

<span class="mw-page-title-main">DTS-HD Master Audio</span> Lossless audio codec for home theater

DTS-HD Master Audio is a multi-channel, lossless audio codec developed by DTS as an extension of the lossy DTS Coherent Acoustics codec. Rather than being an entirely new coding mechanism, DTS-HD MA encodes an audio master in lossy DTS first, then stores a concurrent stream of supplementary data representing whatever the DTS encoder discarded. This gives DTS-HD MA a lossy "core" able to be played back by devices that cannot decode the more complex lossless audio. DTS-HD MA's primary application is audio storage and playback for Blu-ray Disc media; it competes in this respect with Dolby TrueHD, another lossless surround format.

<span class="mw-page-title-main">Audio bit depth</span> Number of bits of information recorded for each digital audio sample

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.

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.

aptX Family of proprietary audio codecs owned by Qualcomm

aptX is a family of proprietary audio codec compression algorithms owned by Qualcomm, with a heavy emphasis on wireless audio applications.

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 at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps.

References

  1. ↑ "Sound Quality or Timbre". hyperphysics.phy-astr.gsu.edu. Retrieved 2017-04-13.
  2. ↑ "Quality of sound and the tech behind it: What to look for when choosing a speaker - Pocket-lint". www.pocket-lint.com. Retrieved 2017-04-13.
  3. ↑ "Pitch, Loudness and Quality of Musical Notes - Pass My Exams: Easy exam revision notes for GSCE Physics". www.passmyexams.co.uk. Retrieved 2017-04-13.
  4. ↑ "What is pulse code modulation (PCM)? - Definition from WhatIs.com". SearchNetworking. Retrieved 2017-04-13.
  5. ↑ "The Sampling Theorem". www.dspguide.com. Retrieved 2017-04-13.
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