Codec

A codec is a device or computer program for encoding or decoding a digital data stream or signal.[1][2][3] Codec is a portmanteau of coder-decoder.[4]

A coder 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 compander functions.

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 are the codecs used in the sound cards of personal computers. A video codec accomplishes the same task for video signals.

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.[5] 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.

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 permit 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.

See also

Comparisons

References

  1. ^ "Using codecs". Microsoft. Retrieved 2009-12-21.
  2. ^ "About.com - Codec". About.com. 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. ^ "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
Adaptive Multi-Rate audio codec

The Adaptive Multi-Rate (AMR, AMR-NB or GSM-AMR) audio codec is an audio compression format optimized for speech coding. AMR speech codec consists of a multi-rate narrowband speech codec that encodes narrowband (200–3400 Hz) signals at variable bit rates ranging from 4.75 to 12.2 kbit/s with toll quality speech starting at 7.4 kbit/s.AMR was adopted as the standard speech codec by 3GPP in October 1999 and is now widely used in GSM and UMTS. It uses link adaptation to select from one of eight different bit rates based on link conditions.

AMR is also a file format for storing spoken audio using the AMR codec. Many modern mobile telephone handsets can store short audio recordings in the AMR format, and both free and proprietary programs exist (see Software support) to convert between this and other formats, although AMR is a speech format and is unlikely to give ideal results for other audio. The common filename extension is .amr. There also exists another storage format for AMR that is suitable for applications with more advanced demands on the storage format, like random access or synchronization with video. This format is the 3GPP-specified 3GP container format based on ISO base media file format.

Apple Lossless

Apple Lossless, also known as Apple Lossless Audio Codec (ALAC), 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.

Asao (codec)

Asao (also known as Nellymoser audio codec) is a proprietary single-channel (mono) codec and compression format optimized for low-bitrate transmission of audio, developed by Nellymoser Inc.

Audio codec

An audio codec is a codec (a device or computer program capable of encoding or decoding a digital data stream) 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 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.

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 contains both an analog-to-digital converter (ADC) and digital-to-analog converter (DAC) running off the same clock signal. This is used in sound cards that support both audio in and out, for instance. Hardware audio codecs send and receive digital data using buses such as AC-Link, I²S, SPI, I²C, etc. Most commonly the digital data is linear PCM, and this is the only format that most codecs support, but some legacy codecs support other formats such as G.711 for telephony.

Avid DNxHD

Avid DNxHD ("Digital Nonlinear Extensible High Definition") is a lossy high-definition video post-production codec developed by Avid for multi-generation compositing with reduced storage and bandwidth requirements. It is an implementation of SMPTE VC-3 standard.

Cinepak

Cinepak is a lossy video codec developed by Peter Barrett at SuperMac Technologies, and released in 1991 with the Video Spigot, and then in 1992 as part of Apple Computer's QuickTime video suite. One of the first video compression tools to achieve full motion video on CD-ROM, it was designed to encode 320×240 resolution video at 1× (150 kbyte/s) CD-ROM transfer rates. The original name of this codec was Compact Video, which is why its FourCC identifier is CVID. The codec was ported to the Microsoft Windows platform in 1993. It was also used on first-generation and some second-generation CD-ROM game consoles, such as the Atari Jaguar CD, Sega CD, Sega Saturn, and 3DO. libavcodec includes a Cinepak decoder and an encoder, both licensed under the terms of the LGPL.

Daala

Daala is a video coding format under development by the Xiph.Org Foundation under the lead of Timothy B. Terriberry mainly sponsored by the Mozilla Corporation. Like Theora and Opus, Daala is available free of any royalties and its reference implementation is being developed as free and open-source software.

Among at least three dozen other contributors there are also engineers from Google, Inc. and Cisco Systems.

The name is taken from the female fictional character of Admiral Natasi Daala from the Star Wars universe.

The reference implementation is written in C and published, together with its source code, as free software under the terms of a BSD-like license. Software patents are being filed for techniques used in and developed for Daala. Those patents are freely licensed to everybody to use for any purpose. However, the patent holders reserve the right to use them to counter patent infringement lawsuits filed by others.

Since June 20, 2013, the development is accompanied by a series of sporadically published posts on the underlying technology on the website of the Xiph.Org Foundation. The Daala project is one of the collaborators in the IETF's NETVC project.

Dirac (video compression format)

Dirac is an open and royalty-free video compression format, specification and system developed by BBC Research & Development. Schrödinger and dirac-research (formerly just called "Dirac") are open and royalty-free software implementations (video codecs) of Dirac. Dirac format aims to provide high-quality video compression for Ultra HDTV and beyond, and as such competes with existing formats such as H.264 and VC-1.

The specification was finalised in January 2008, and further developments are only bug fixes and constraints. In September of that year, version 1.0.0 of an I-frame only subset known as Dirac Pro was released and has since been standardised by the SMPTE as VC-2. Version 2.2.3 of the full Dirac specification, including motion compensation and inter-frame coding, was issued a few days later. Dirac Pro was used internally by the BBC to transmit HDTV pictures at the Beijing Olympics in 2008.The format implementations are named in honour of the theoretical physicists Paul Dirac and Erwin Schrödinger, who shared the 1933 Nobel Prize in physics.

DivX

DivX is a brand of video codec products developed by DivX, LLC. The DivX codec gained fame for its ability to compress lengthy video segments into small sizes while maintaining relatively high visual quality.

There are three DivX codecs; the original MPEG-4 Part 2 DivX codec, the H.264/MPEG-4 AVC DivX Plus HD codec and the High Efficiency Video Coding DivX HEVC Ultra HD codec.

The most recent version of the codec itself is version 6.9.2, which is several years old. New version numbers on the packages now reflect updates to the media player, converter, etc.

FLAC

FLAC (; Free Lossless Audio Codec) is an audio coding format for lossless compression of digital audio, 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 decompress 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 has support for metadata tagging, album cover art, and fast seeking.

Internet Low Bitrate Codec

Internet Low Bitrate Codec (iLBC) is an open-source royalty-free narrowband speech audio coding format codec and reference implementation, developed by Global IP Solutions (GIPS) formerly Global IP Sound (acquired by Google Inc in 2011). It was formerly freeware with limitations on commercial use, but since 2011 it is available under a free software/open source (3-clause BSD license) license as a part of the open source WebRTC project. It is suitable for VoIP applications, streaming audio, archival and messaging. The algorithm is a version of block-independent linear predictive coding, with the choice of data frame lengths of 20 and 30 milliseconds. The encoded blocks have to be encapsulated in a suitable protocol for transport, usually the Real-time Transport Protocol (RTP).

iLBC handles lost frames through graceful speech quality degradation. Lost frames often occur in connection with lost or delayed IP packets. Ordinary low-bitrate codecs exploit dependencies between speech frames, which cause errors to propagate when packets are lost or delayed. In contrast, iLBC-encoded speech frames are independent and so this problem will not occur.

iLBC is defined in RFC 3951. It is one of the codecs used by Gizmo5, WebRTC, Ekiga, Google Talk, Maemo Recorder (on the Nokia N800/N810), Polycom IP Phone, Cisco, QuteCom, Tuenti, Yahoo! Messenger, Ooma and many others.

iLBC was submitted to IETF in 2002 and the final specification was published in 2004.

LDAC (codec)

LDAC is an audio coding technology developed by Sony, which allows streaming audio over Bluetooth connections up to 990 kbit/s at 24 bit/96 kHz (also called high-resolution audio). It is used by various Sony products, including headphones, smartphones, portable media players, active speakers and home theaters. LDAC is a lossy codec, which employs a hybrid coding scheme based on the MDCT to provide more efficient data compression. Its main competitor is Qualcomm's aptX-HD technology.Starting from Android 8.0 "Oreo", LDAC is part of the Android Open Source Project, enabling every OEM to integrate this standard into their own Android devices freely.As encoder library is open source the implementation for Linux is already present in bluez-alsa and pulseaudio-modules-bt. Now it works in Fedora 29.

Opus (audio format)

Opus is a lossy audio coding format developed by the Xiph.Org Foundation and standardized by the Internet Engineering Task Force, designed to efficiently code speech and general audio in a single format, while remaining low-latency enough for real-time interactive communication and low-complexity enough for low-end embedded processors. Opus replaces both Vorbis and Speex for new applications, and several blind listening tests have ranked it higher-quality than any other standard audio format at any given bitrate until transparency is reached, including MP3, AAC, and HE-AAC.Opus combines the speech-oriented linear predictive coding SILK algorithm and the lower-latency, MDCT-based CELT algorithm, switching between or combining them as needed for maximal efficiency. Bitrate, audio bandwidth, complexity, and algorithm can all be adjusted seamlessly in each frame. Opus has the low algorithmic delay (26.5 ms by default) necessary for use as part of a real-time communication link, permitting natural conversation, networked music performances, and live lip sync; by trading-off quality or bitrate, the delay can be reduced down to 5 ms. Its delay is exceptionally low compared to competing codecs, which require well over 100 ms, yet Opus performs very competitively with these formats in terms of quality per bitrate.As an open format standardized through RFC 6716, a reference implementation called libopus is available under the New BSD License. The reference has both fixed-point and floating-point optimizations for low- and high-end devices, with SIMD optimizations on platforms that support them. All known software patents that cover Opus are licensed under royalty-free terms.

SILK

SILK is an audio compression format and audio codec developed by Skype Limited. It was developed for use in Skype, as a replacement for the SVOPC codec. Since licensing out, it has also been used by others. It has been extended to the Internet standard Opus codec.

TTA (codec)

True Audio (TTA) is a lossless compressor for multichannel 8, 16 and 24 bits audio data.

.tta is the extension to filenames of audio files created by the True Audio codec.

Theora

Theora is a free lossy video compression format. It is developed by the Xiph.Org Foundation and distributed without licensing fees alongside their other free and open media projects, including the Vorbis audio format and the Ogg container.

The libtheora video codec is the reference implementation of the Theora video compression format being developed by the Xiph.Org Foundation.Theora is derived from the formerly proprietary VP3 codec, released into the public domain by On2 Technologies. It is broadly comparable in design and bitrate efficiency to MPEG-4 Part 2, early versions of Windows Media Video, and RealVideo while lacking some of the features present in some of these other codecs. It is comparable in open standards philosophy to the BBC's Dirac codec.

Theora is named after Theora Jones, Edison Carter's Controller on the Max Headroom television program.In 2014, a bug requesting Theora support on Android was closed "Won't Fix (Obsolete)".

Wikipedia stopped preferring Ogg Theora and now prefers WebM.

Video codec

A video codec is an electronic circuit or software that compresses or decompresses digital video. It converts uncompressed video to a compressed format or vice versa. In the context of video compression, "codec" is a concatenation of "encoder" and "decoder"—a device that only compresses is typically called an encoder, and one that only decompresses is a decoder.

The compressed data format usually conforms to a standard video compression specification. The compression is typically lossy, meaning that the compressed video lacks some information present in the original video. A consequence of this is that decompressed video has lower quality than the original, uncompressed video because there is insufficient information to accurately reconstruct the original video.

There are complex relationships between the video quality, the amount of data used to represent the video (determined by the bit rate), the complexity of the encoding and decoding algorithms, sensitivity to data losses and errors, ease of editing, random access, and end-to-end delay (latency).

Video for Windows

Video for Windows (VfW) is a multimedia framework developed by Microsoft that allows Windows to play and encode digital video.

Xvid

Xvid (formerly "XviD") is a video codec library following the MPEG-4 video coding standard, specifically MPEG-4 Part 2 Advanced Simple Profile (ASP). It uses ASP features such as b-frames, global and quarter pixel motion compensation, lumi masking, trellis quantization, and H.263, MPEG and custom quantization matrices.

Xvid is a primary competitor of the DivX Pro Codec. In contrast with the DivX codec, which is proprietary software developed by DivX, Inc., Xvid is free software distributed under the terms of the GNU General Public License. This also means that unlike the DivX codec, which is only available for a limited number of platforms, Xvid can be used on all platforms and operating systems for which the source code can be compiled.

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