Codec

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A codec is a device or computer program that encodes or decodes a data stream or signal. [1] [2] [3] Codec is a portmanteau of coder/decoder. [4]

Contents

In electronic communications, an endec is a device that acts as both an encoder and a decoder on a signal or data stream, [5] 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 receiving device converts the digital signals back to analog form using an audio decoder for playback. An example of this is the codecs used in the sound cards of personal computers. A video codec accomplishes the same task for video signals.

An Emergency Alert System unit is usually an endec, but sometimes just a decoder.

When implementing the Infrared Data Association (IrDA) protocol, an endec may be used between the UART and the optoelectronic systems. [6]

Compression

In addition to encoding a signal, a codec may also compress the data to reduce transmission bandwidth or storage space. Compression codecs are classified primarily into lossy codecs and lossless codecs.

Lossless codecs are often used for archiving data in a compressed form while retaining all information present in the original stream. If preserving the original quality of the stream is more important than eliminating the correspondingly larger data sizes, lossless codecs are preferred. This is especially true if the data is to undergo further processing (for example editing) in which case the repeated application of processing (encoding and decoding) on lossy codecs will degrade the quality of the resulting data such that it is no longer identifiable (visually, audibly or both). Using more than one codec or encoding scheme successively can also degrade quality significantly. The decreasing cost of storage capacity and network bandwidth has a tendency to reduce the need for lossy codecs for some media.

Many popular codecs are lossy. They reduce quality in order to maximize compression. Often, this type of compression is virtually indistinguishable from the original uncompressed sound or images, depending on the codec and the settings used. [7] The most widely used lossy data compression technique in digital media is based on the discrete cosine transform (DCT), used in compression standards such as JPEG images, H.26x and MPEG video, and MP3 and AAC audio. Smaller data sets ease the strain on relatively expensive storage sub-systems such as non-volatile memory and hard disk, as well as write-once-read-many formats such as CD-ROM, DVD and Blu-ray Disc. Lower data rates also reduce cost and improve performance when the data is transmitted, e.g. over the internet.

Media codecs

Two principal techniques are used in codecs, pulse-code modulation and delta modulation. Codecs are often designed to emphasize certain aspects of the media to be encoded. For example, a digital video (using a DV codec) of a sports event needs to encode motion well but not necessarily exact colors, while a video of an art exhibit needs to encode color and surface texture well.

Audio codecs for cell phones need to have very low latency between source encoding and playback. In contrast, audio codecs for recording or broadcast can use high-latency audio compression techniques to achieve higher fidelity at a lower bit rate.

There are thousands of audio and video codecs, ranging in cost from free to hundreds of dollars or more. This variety of codecs can create compatibility and obsolescence issues. The impact is lessened for older formats, for which free or nearly-free codecs have existed for a long time. The older formats are often ill-suited to modern applications, however, such as playback in small portable devices. For example, raw uncompressed PCM audio (44.1 kHz, 16-bit stereo, as represented on an audio CD or in a .wav or .aiff file) has long been a standard across multiple platforms, but its transmission over networks is slow and expensive compared with more modern compressed formats, such as Opus and MP3.

Many multimedia data streams contain both audio and video, and often some metadata that permits synchronization of audio and video. Each of these three streams may be handled by different programs, processes, or hardware; but for the multimedia data streams to be useful in stored or transmitted form, they must be encapsulated together in a container format.

Lower bitrate codecs allow more users, but they also have more distortion. Beyond the initial increase in distortion, lower bit rate codecs also achieve their lower bit rates by using more complex algorithms that make certain assumptions, such as those about the media and the packet loss rate. Other codecs may not make those same assumptions. When a user with a low bitrate codec talks to a user with another codec, additional distortion is introduced by each transcoding.

Audio Video Interleave (AVI) is sometimes erroneously described as a codec, but AVI is actually a container format, while a codec is a software or hardware tool that encodes or decodes audio or video into or from some audio or video format. Audio and video encoded with many codecs might be put into an AVI container, although AVI is not an ISO standard. There are also other well-known container formats, such as Ogg, ASF, QuickTime, RealMedia, Matroska, and DivX Media Format. MPEG transport stream, MPEG program stream, MP4, and ISO base media file format are examples of container formats that are ISO standardized.

Malware

Fake codecs are used when an online user takes a type of codec and installs viruses and other malware into whatever data is being compressed and uses it as a disguise. This disguise appears as a codec download through a pop-up alert or ad. When a user goes to click or download that codec, the malware is then installed on the computer. Once a fake codec is installed it is often used to access private data, corrupt an entire computer system or to keep spreading the malware. One of the previous most used ways to spread malware was fake AV pages and with the rise of codec technology, both have been used in combination to take advantage of online users. [8] This combination allows fake codecs to be automatically downloaded to a device through a website linked in a pop-up ad, virus/codec alerts or articles as well.

See also

Related Research Articles

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.

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.

MPEG-1 is a standard for lossy compression of video and audio. It is designed to compress VHS-quality raw digital video and CD audio down to about 1.5 Mbit/s without excessive quality loss, making video CDs, digital cable/satellite TV and digital audio broadcasting (DAB) practical.

Windows Media Audio (WMA) is a series of audio codecs and their corresponding audio coding formats developed by Microsoft. It is a proprietary technology that forms part of the Windows Media framework. WMA consists of four distinct codecs. The original WMA codec, known simply as WMA, was conceived as a competitor to the popular MP3 and RealAudio codecs. WMA Pro, a newer and more advanced codec, supports multichannel and high resolution audio. A lossless codec, WMA Lossless, compresses audio data without loss of audio fidelity. WMA Voice, targeted at voice content, applies compression using a range of low bit rates. Microsoft has also developed a digital container format called Advanced Systems Format to store audio encoded by WMA.

Audio Video Interleave is a proprietary multimedia container format and Windows standard introduced by Microsoft in November 1992 as part of its Video for Windows software. AVI files can contain both audio and video data in a file container that allows synchronous audio-with-video playback. Like the DVD video format, AVI files support multiple streaming audio and video, although these features are seldom used.

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">Video codec</span> Digital video processing

A video codec is software or hardware that compresses and decompresses digital video. In the context of video compression, codec is a portmanteau of encoder and decoder, while a device that only compresses is typically called an encoder, and one that only decompresses is a decoder.

<span class="mw-page-title-main">FLAC</span> Lossless digital audio coding format

FLAC 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.

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. Therefore, it may be decompressed to a file that is identical to the source material.

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. for lossless data compression of digital music. After initially keeping it proprietary from its inception in 2004, in late 2011 Apple made the codec available 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.

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.

A container format or metafile is a file format that allows multiple data streams to be embedded into a single file, usually along with metadata for identifying and further detailing those streams. Notable examples of container formats include archive files and formats used for multimedia playback. Among the earliest cross-platform container formats were Distinguished Encoding Rules and the 1985 Interchange File Format.

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">MPEG-4 SLS</span> Extension to the MPEG-4 Audio standard

MPEG-4 SLS, or MPEG-4 Scalable to Lossless as per ISO/IEC 14496-3:2005/Amd 3:2006 (Scalable Lossless Coding), is an extension to the MPEG-4 Part 3 (MPEG-4 Audio) standard to allow lossless audio compression scalable to lossy MPEG-4 General Audio coding methods (e.g., variations of AAC). It was developed jointly by the Institute for Infocomm Research (I2R) and Fraunhofer, which commercializes its implementation of a limited subset of the standard under the name of HD-AAC. Standardization of the HD-AAC profile for MPEG-4 Audio is under development (as of September 2009).

OptimFROG is a proprietary lossless audio data compression codec developed by Florin Ghido. OptimFROG is optimized for very high compression at the expense of encoding and decoding speed, and consistently measures among the highest compressing lossless codecs. OptimFROG comes with three compressors: one lossless codec for integer LPCM format WAVE files, one for IEEE_754 floating-point WAVE files, and a DualStream format with a lossy part and a correction file for losslessness.

A video coding format is a content representation format of digital video content, such as in a data file or bitstream. It typically uses a standardized video compression algorithm, most commonly based on discrete cosine transform (DCT) coding and motion compensation. A specific software, firmware, or hardware implementation capable of compression or decompression in a specific video coding format is called a video codec.

<span class="mw-page-title-main">Audio coding format</span> Digitally coded format for audio signals

An audio coding format is a content representation format for storage or transmission of digital audio. 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.

References

  1. ↑ "Using codecs". Microsoft. Archived from the original on September 6, 2010. Retrieved 2009-12-21.
  2. ↑ Siegchrist, Gretchen. "About.com - Codec". About.com. Archived from the original on April 5, 2015. Retrieved 2009-12-21.
  3. ↑ "Ubuntu Documentation - What is a codec?". Ubuntu Documentation Team. Archived from the original on February 19, 2012. Retrieved 2009-12-21.
  4. ↑ "Codec – Definition of Codec by Merriam-Webster". Merriam-Webster. Retrieved 2019-01-15.
  5. ↑ Rouse, Margaret (24 October 2016). "Encoder/Decoder". Techopedia. Retrieved 27 December 2023.
  6. ↑ "IrDA SIR ENDEC functional description".
  7. ↑ "Audio quality of aac vs. mp3 vs. wma vs. ogg encoders". SoundExpert. Retrieved 2010-07-25. above 5.0 – all sound artifacts will be beyond threshold of human perception with corresponding perception margin
  8. ↑ "Fake Video Codecs Still Going Strong" . Retrieved 2022-01-09.
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