In telecommunications, average bitrate (ABR) refers to the average amount of data transferred per unit of time, usually measured per second, commonly for digital music or video. An MP3 file, for example, that has an average bit rate of 128 kbit/s transfers, on average, 128,000 bits every second. It can have higher bitrate and lower bitrate parts, and the average bitrate for a certain timeframe is obtained by dividing the number of bits used during the timeframe by the number of seconds in the timeframe. Bitrate is not reliable as a standalone measure of audio or video quality, since more efficient compression methods use lower bitrates to encode material at a similar quality.
Average bitrate can also refer to a form of variable bitrate (VBR) encoding in which the encoder will try to reach a target average bitrate or file size while allowing the bitrate to vary between different parts of the audio or video. As it is a form of variable bitrate, this allows more complex portions of the material to use more bits and less complex areas to use fewer bits. However, bitrate will not vary as much as in variable bitrate encoding. [1] At a given bitrate, VBR is usually higher quality than ABR, which is higher quality than CBR (constant bitrate). [2] ABR encoding is desirable for users who want the general benefits of VBR encoding (an optimum bitrate from frame to frame) but with a relatively predictable file size. [1] Two-pass encoding is usually needed for accurate ABR encoding, as on the first pass the encoder has no way of knowing what parts of the audio or video need the highest bitrates to be encoded. [3]
LAME is a software encoder that converts digital audio into the MP3 audio coding format. LAME is a free software project that was first released in 1998 and has incorporated many improvements since then, including an improved psychoacoustic model. The LAME encoder outperforms early encoders like L3enc and possibly the "gold standard encoder" MP3enc, both marketed by Fraunhofer.
MP3 is a coding format for digital audio developed largely by the Fraunhofer Society in Germany under the lead of Karlheinz Brandenburg. It was designed to greatly reduce the amount of data required to represent audio, yet still sound like a faithful reproduction of the original uncompressed audio to most listeners; for example, compared to CD-quality digital audio, MP3 compression can commonly achieve a 75–95% reduction in size, depending on the bit rate. In popular usage, MP3 often refers to files of sound or music recordings stored in the MP3 file format (.mp3) on consumer electronic devices.
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.
Video CD is a home video format and the first format for distributing films on standard 120 mm (4.7 in) optical discs. The format was widely adopted in Southeast Asia, South Asia, East Asia, Central Asia and West Asia, superseding the VHS and Betamax systems in the regions until DVD-Video finally became affordable in the first decade of the 21st century.
Speex is an audio compression codec specifically tuned for the reproduction of human speech and also a free software speech codec that may be used on voice over IP applications and podcasts. It is based on the code excited linear prediction speech coding algorithm. Its creators claim Speex to be free of any patent restrictions and it is licensed under the revised (3-clause) BSD license. It may be used with the Ogg container format or directly transmitted over UDP/RTP. It may also be used with the FLV container format.
Variable bitrate (VBR) is a term used in telecommunications and computing that relates to the bitrate used in sound or video encoding. As opposed to constant bitrate (CBR), VBR files vary the amount of output data per time segment. VBR allows a higher bitrate to be allocated to the more complex segments of media files while less space is allocated to less complex segments. The average of these rates can be calculated to produce an average bitrate for the file.
Advanced Audio Coding (AAC) is an audio coding standard for lossy digital audio compression. It was designed to be the successor of the MP3 format and generally achieves higher sound quality than MP3 at the same bit rate.
In telecommunications and computing, bit rate is the number of bits that are conveyed or processed per unit of time.
Constant bitrate (CBR) is a term used in telecommunications, relating to the quality of service. Compare with variable bitrate.
RealVideo, also spelled as Real Video, is a suite of proprietary video compression formats developed by RealNetworks — the specific format changes with the version. It was first released in 1997 and as of 2024 was at version 15. RealVideo is supported on many platforms, including Windows, Mac, Linux, Solaris, and several mobile phones.
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.
In data compression and psychoacoustics, transparency is the result of lossy data compression accurate enough that the compressed result is perceptually indistinguishable from the uncompressed input, i.e. perceptually lossless.
If a network service wishes to use a broadband network to transport a particular kind of traffic, it must first inform the network about what kind of traffic is to be transported, and the performance requirements of that traffic. The application presents this information to the network in the form of a traffic contract.
TwinVQ is an audio compression technique developed by Nippon Telegraph and Telephone Corporation (NTT) Human Interface Laboratories in 1994. The compression technique has been used in both standardized and proprietary designs.
Α video codec is software or a device that provides encoding and decoding for digital video, and which may or may not include the use of video compression and/or decompression. Most codecs are typically implementations of video coding formats.
TooLAME is a free software MPEG-1 Layer II (MP2) audio encoder written primarily by Mike Cheng. While there are many MP2 encoders, TooLAME is well-known and widely used for its particularly high audio quality. It has been unmaintained since 2003, but is directly succeeded by the TwoLAME code fork. The name TooLAME is a play on LAME and Layer II.
MPEG-1 Audio Layer III HD was an audio compression codec developed by Technicolor, formerly known as Thomson.
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.
Fraunhofer FDK AAC is an open-source library for encoding and decoding digital audio in the Advanced Audio Coding (AAC) format. Fraunhofer IIS developed this library for Android 4.1. It supports several Audio Object Types including MPEG-2 and MPEG-4 AAC LC, HE-AAC, HE-AACv2 as well AAC-LD and AAC-ELD for real-time communication. The encoding library supports sample rates up to 96 kHz and up to eight channels.