|bit per second||bit/s||1||1|
|Decimal prefixes (SI)|
|kilobit per second||kbit/s||103||10001|
|megabit per second||Mbit/s||106||10002|
|gigabit per second||Gbit/s||109||10003|
|terabit per second||Tbit/s||1012||10004|
|Binary prefixes (IEC 80000-13)|
|kibibit per second||Kibit/s||210||10241|
|mebibit per second||Mibit/s||220||10242|
|gibibit per second||Gibit/s||230||10243|
|tebibit per second||Tibit/s||240||10244|
In telecommunications and computing, bit rate (bitrate or as a variable R) is the number of bits that are conveyed or processed per unit of time.
Computing is any activity that uses computers to manage, process, and communicate information. It includes development of both hardware and software. Computing is a critical, integral component of modern industrial technology. Major computing disciplines include computer engineering, software engineering, computer science, information systems, and information technology.
The bit is a basic unit of information in information theory, computing, and digital communications. The name is a portmanteau of binary digit.
The bit rate is quantified using the bits per second unit (symbol: "bit/s"), often in conjunction with an SI prefix such as "kilo" (1 kbit/s = 1,000 bit/s), "mega" (1 Mbit/s = 1,000 kbit/s), "giga" (1 Gbit/s = 1,000 Mbit/s) or "tera" (1 Tbit/s = 1000 Gbit/s). The non-standard abbreviation "bps" is often used to replace the standard symbol "bit/s", so that, for example, "1 Mbps" is used to mean one million bits per second.
Kilo is a decimal unit prefix in the metric system denoting multiplication by one thousand (103). It is used in the International System of Units where it has the unit symbol k, in lower case.
Mega is a unit prefix in metric systems of units denoting a factor of one million (106 or 1000000). It has the unit symbol M. It was confirmed for use in the International System of Units (SI) in 1960. Mega comes from Ancient Greek: μέγας, romanized: megas, lit. 'great'.
Giga ( or ) is a unit prefix in the metric system denoting a factor of a (short-form) billion (109 or 1000000000). It has the symbol G.
In most environments, one byte per second (1 B/s) corresponds to 8 bit/s.
When quantifying large bit rates, SI prefixes (also known as metric prefixes or decimal prefixes) are used, thus:
A metric prefix is a unit prefix that precedes a basic unit of measure to indicate a multiple or fraction of the unit. While all metric prefixes in common use today are decadic, historically there have been a number of binary metric prefixes as well. Each prefix has a unique symbol that is prepended to the unit symbol. The prefix kilo-, for example, may be added to gram to indicate multiplication by one thousand: one kilogram is equal to one thousand grams. The prefix milli-, likewise, may be added to metre to indicate division by one thousand; one millimetre is equal to one thousandth of a metre.
|1,000 bit/s||rate = 1 kbit/s (one kilobit or one thousand bits per second)|
|1,000,000 bit/s||rate = 1 Mbit/s (one megabit or one million bits per second)|
|1,000,000,000 bit/s||rate = 1 Gbit/s (one gigabit or one billion bits per second)|
Binary prefixes are sometimes used for bit rates. B/s = 8192 bit/s, and 1 MiB/s = 1024 KiB/s).The International Standard (IEC 80000-13) specifies different abbreviations for binary and decimal (SI) prefixes (e.g. 1 KiB/s = 1024
A binary prefix is a unit prefix for multiples of units in data processing, data transmission, and digital information, notably the bit and the byte, to indicate multiplication by a power of 2.
The kibibyte is a multiple of the unit byte for quantities of digital information. The binary prefix kibi means 210, or 1024; therefore, 1 kibibyte is 1024 bytes. The unit symbol for the kibibyte is KiB.
The mebibyte is a multiple of the unit byte for digital information. The binary prefix mebi means 220; therefore one mebibyte is equal to 1048576bytes, i.e., 1024 kibibytes. The unit symbol for the mebibyte is MiB.
In digital communication systems, the physical layer gross bitrate,raw bitrate, data signaling rate , gross data transfer rate or uncoded transmission rate (sometimes written as a variable Rb or fb ) is the total number of physically transferred bits per second over a communication link, including useful data as well as protocol overhead.
In the seven-layer OSI model of computer networking, the physical layer or layer 1 is the first and lowest layer. This layer may be implemented by a PHY chip.
In telecommunication, data signaling rate (DSR), also known as gross bit rate, is the aggregate rate at which data pass a point in the transmission path of a data transmission system.
In case of serial communications, the gross bit rate is related to the bit transmission time as:
In telecommunication and data transmission, serial communication is the process of sending data one bit at a time, sequentially, over a communication channel or computer bus. This is in contrast to parallel communication, where several bits are sent as a whole, on a link with several parallel channels.
The gross bit rate is related to the symbol rate or modulation rate, which is expressed in bauds or symbols per second. However, the gross bit rate and the baud value are equal only when there are only two levels per symbol, representing 0 and 1, meaning that each symbol of a data transmission system carries exactly one bit of data; for example, this is not the case for modern modulation systems used in modems and LAN equipment.
For most line codes and modulation methods:
More specifically, a line code (or baseband transmission scheme) representing the data using pulse-amplitude modulation with different voltage levels, can transfer . A digital modulation method (or passband transmission scheme) using different symbols, for example amplitudes, phases or frequencies, can transfer . This results in:
An exception from the above is some self-synchronizing line codes, for example Manchester coding and return-to-zero (RTZ) coding, where each bit is represented by two pulses (signal states), resulting in:
A theoretical upper bound for the symbol rate in baud, symbols/s or pulses/s for a certain spectral bandwidth in hertz is given by the Nyquist law:
In practice this upper bound can only be approached for line coding schemes and for so-called vestigal sideband digital modulation. Most other digital carrier-modulated schemes, for example ASK, PSK, QAM and OFDM, can be characterized as double sideband modulation, resulting in the following relation:
In case of parallel communication, the gross bit rate is given by
where n is the number of parallel channels, Mi is the number of symbols or levels of the modulation in the i-th channel, and Ti is the symbol duration time, expressed in seconds, for the i-th channel.
The physical layer net bitrate,information rate, useful bit rate, payload rate, net data transfer rate, coded transmission rate, effective data rate or wire speed (informal language) of a digital communication channel is the capacity excluding the physical layer protocol overhead, for example time division multiplex (TDM) framing bits, redundant forward error correction (FEC) codes, equalizer training symbols and other channel coding. Error-correcting codes are common especially in wireless communication systems, broadband modem standards and modern copper-based high-speed LANs. The physical layer net bitrate is the datarate measured at a reference point in the interface between the datalink layer and physical layer, and may consequently include data link and higher layer overhead.
In modems and wireless systems, link adaptation (automatic adaption of the data rate and the modulation and/or error coding scheme to the signal quality) is often applied. In that context, the term peak bitrate denotes the net bitrate of the fastest and least robust transmission mode, used for example when the distance is very short between sender and transmitter.Some operating systems and network equipment may detect the "connection speed" (informal language) of a network access technology or communication device, implying the current net bit rate. Note that the term line rate in some textbooks is defined as gross bit rate, in others as net bit rate.
The relationship between the gross bit rate and net bit rate is affected by the FEC code rate according to the following.
The connection speed of a technology that involves forward error correction typically refers to the physical layer net bit rate in accordance with the above definition.
For example, the net bitrate (and thus the "connection speed") of an IEEE 802.11a wireless network is the net bit rate of between 6 and 54 Mbit/s, while the gross bit rate is between 12 and 72 Mbit/s inclusive of error-correcting codes.
The net bit rate of ISDN2 Basic Rate Interface (2 B-channels + 1 D-channel) of 64+64+16 = 144 kbit/s also refers to the payload data rates, while the D channel signalling rate is 16 kbit/s.
The net bit rate of the Ethernet 100Base-TX physical layer standard is 100 Mbit/s, while the gross bitrate is 125 Mbit/second, due to the 4B5B (four bit over five bit) encoding. In this case, the gross bit rate is equal to the symbol rate or pulse rate of 125 megabaud, due to the NRZI line code.
In communications technologies without forward error correction and other physical layer protocol overhead, there is no distinction between gross bit rate and physical layer net bit rate. For example, the net as well as gross bit rate of Ethernet 10Base-T is 10 Mbit/s. Due to the Manchester line code, each bit is represented by two pulses, resulting in a pulse rate of 20 megabaud.
The "connection speed" of a V.92 voiceband modem typically refers to the gross bit rate, since there is no additional error-correction code. It can be up to 56,000 bit/s downstreams and 48,000 bit/s upstreams. A lower bit rate may be chosen during the connection establishment phase due to adaptive modulation –slower but more robust modulation schemes are chosen in case of poor signal-to-noise ratio. Due to data compression, the actual data transmission rate or throughput (see below) may be higher.
The channel capacity, also known as the Shannon capacity, is a theoretical upper bound for the maximum net bitrate, exclusive of forward error correction coding, that is possible without bit errors for a certain physical analog node-to-node communication link.
The channel capacity is proportional to the analog bandwidth in hertz. This proportionality is called Hartley's law. Consequently, the net bit rate is sometimes called digital bandwidth capacity in bit/s.
The term throughput , essentially the same thing as digital bandwidth consumption, denotes the achieved average useful bit rate in a computer network over a logical or physical communication link or through a network node, typically measured at a reference point above the datalink layer. This implies that the throughput often excludes data link layer protocol overhead. The throughput is affected by the traffic load from the data source in question, as well as from other sources sharing the same network resources. See also measuring network throughput.
Goodput or data transfer rate refers to the achieved average net bit rate that is delivered to the application layer, exclusive of all protocol overhead, data packets retransmissions, etc. For example, in the case of file transfer, the goodput corresponds to the achieved file transfer rate. The file transfer rate in bit/s can be calculated as the file size (in bytes) divided by the file transfer time (in seconds) and multiplied by eight.
As an example, the goodput or data transfer rate of a V.92 voiceband modem is affected by the modem physical layer and data link layer protocols. It is sometimes higher than the physical layer data rate due to V.44 data compression, and sometimes lower due to bit-errors and automatic repeat request retransmissions.
If no data compression is provided by the network equipment or protocols, we have the following relation:
for a certain communication path.
These are examples of physical layer net bit rates in proposed communication standard interfaces and devices:
|WAN modems||Ethernet LAN||WiFi WLAN||Mobile data|
See also comparison of mobile phone standards
For more examples, see list of device bit rates, spectral efficiency comparison table and OFDM system comparison table.
In digital multimedia, bitrate represents the amount of information, or detail, that is stored per unit of time of a recording. The bitrate depends on several factors:
Generally, choices are made about the above factors in order to achieve the desired trade-off between minimizing the bitrate and maximizing the quality of the material when it is played.
If lossy data compression is used on audio or visual data, differences from the original signal will be introduced; if the compression is substantial, or lossy data is decompressed and recompressed, this may become noticeable in the form of compression artifacts. Whether these affect the perceived quality, and if so how much, depends on the compression scheme, encoder power, the characteristics of the input data, the listener's perceptions, the listener's familiarity with artifacts, and the listening or viewing environment.
The bitrates in this section are approximately the minimum that the average listener in a typical listening or viewing environment, when using the best available compression, would perceive as not significantly worse than the reference standard:
In digital multimedia, bit rate refers to the number of bits used per second to represent a continuous medium such as audio or video after source coding (data compression). The encoding bit rate of a multimedia file is the size of a multimedia file in bytes divided by the playback time of the recording (in seconds), multiplied by eight.
For realtime streaming multimedia, the encoding bit rate is the goodput that is required to avoid interrupt:
The term average bitrate is used in case of variable bitrate multimedia source coding schemes. In this context, the peak bit rate is the maximum number of bits required for any short-term block of compressed data.
A theoretical lower bound for the encoding bit rate for lossless data compression is the source information rate, also known as the entropy rate.
CD-DA, the standard audio CD, is said to have a data rate of 44.1 kHz/16, meaning that the audio data was sampled 44,100 times per second and with a bit depth of 16. CD-DA is also stereo, using a left and right channel, so the amount of audio data per second is double that of mono, where only a single channel is used.
The bit rate of PCM audio data can be calculated with the following formula:
For example, the bit rate of a CD-DA recording (44.1 kHz sampling rate, 16 bits per sample and two channels) can be calculated as follows:
The cumulative size of a length of PCM audio data (excluding a file header or other metadata) can be calculated using the following formula:
The cumulative size in bytes can be found by dividing the file size in bits by the number of bits in a byte, which is eight:
Therefore, 80 minutes (4,800 seconds) of CD-DA data requires 846,720,000 bytes of storage:
The MP3 audio format provides lossy data compression. Audio quality improves with increasing bitrate:
For technical reasons (hardware/software protocols, overheads, encoding schemes, etc.) the actual bit rates used by some of the compared-to devices may be significantly higher than what is listed above. For example, telephone circuits using µlaw or A-law companding (pulse code modulation) yield 64 kbit/s.
Enhanced Data rates for GSM Evolution (EDGE) is a digital mobile phone technology that allows improved data transmission rates as a backward-compatible extension of GSM. EDGE is considered a pre-3G radio technology and is part of ITU's 3G definition. EDGE was deployed on GSM networks beginning in 2003 – initially by Cingular in the United States.
MP3 is a coding format for digital audio. Originally defined as the third audio format of the MPEG-1 standard, it was retained and further extended—defining additional bit-rates and support for more audio channels—as the third audio format of the subsequent MPEG-2 standard. A third version, known as MPEG 2.5—extended to better support lower bit rates—is commonly implemented, but is not a recognized standard.
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) possible.
MPEG-2 is a standard for "the generic coding of moving pictures and associated audio information". It describes a combination of lossy video compression and lossy audio data compression methods, which permit storage and transmission of movies using currently available storage media and transmission bandwidth. While MPEG-2 is not as efficient as newer standards such as H.264/AVC and H.265/HEVC, backwards compatibility with existing hardware and software means it is still widely used, for example in over-the-air digital television broadcasting and in the DVD-Video standard.
In telecommunication and electronics, baud is a common measure of symbol rate, one of the components that determine the speed of communication over a data channel.
Digital audio broadcasting (DAB) is a digital radio standard for broadcasting digital audio radio services, used in many countries around the world, though not North America.
G.711 is an ITU-T standard (Recommendation) for audio companding, titled Pulse code modulation (PCM) of voice frequencies. It is primarily used in telephony. The standard was released for use in 1972. Its formal name is Pulse code modulation (PCM) of voice frequencies. It is a required standard in many technologies, for example in H.320 and H.323 specifications. It can also be used for fax communication over IP networks. G.711 is a narrowband audio codec that provides toll-quality audio at 64 kbit/s. G.711 passes audio signals in the range of 300–3400 Hz and samples them at the rate of 8,000 samples per second, with the tolerance on that rate of 50 parts per million (ppm). Non-uniform (logarithmic) quantization with 8 bits is used to represent each sample, resulting in a 64 kbit/s bit rate. There are two slightly different versions: μ-law, which is used primarily in North America and Japan, and A-law, which is in use in most other countries outside North America.
MPEG-1 Audio Layer II or MPEG-2 Audio Layer II is a lossy audio compression format defined by ISO/IEC 11172-3 alongside MPEG-1 Audio Layer I and MPEG-1 Audio Layer III (MP3). While MP3 is much more popular for PC and Internet applications, MP2 remains a dominant standard for audio broadcasting.
MPEG-4 Part 3 or MPEG-4 Audio is the third part of the ISO/IEC MPEG-4 international standard developed by Moving Picture Experts Group. It specifies audio coding methods. The first version of ISO/IEC 14496-3 was published in 1999.
DVB-T is an abbreviation for "Digital Video Broadcasting — Terrestrial"; it is the DVB European-based consortium standard for the broadcast transmission of digital terrestrial television that was first published in 1997 and first broadcast in the UK in 1998. This system transmits compressed digital audio, digital video and other data in an MPEG transport stream, using coded orthogonal frequency-division multiplexing modulation. It is also the format widely used worldwide for Electronic News Gathering for transmission of video and audio from a mobile newsgathering vehicle to a central receive point. It is also used in the US by Amateur television operators.
Digital Radio Mondiale is a set of digital audio broadcasting technologies designed to work over the bands currently used for analogue radio broadcasting including AM broadcasting, particularly shortwave, and FM broadcasting. DRM is more spectrally efficient than AM and FM, allowing more stations, at higher quality, into a given amount of bandwidth, using various MPEG-4 audio coding formats.
G.722 is an ITU-T standard 7 kHz wideband audio codec operating at 48, 56 and 64 kbit/s. It was approved by ITU-T in November 1988. Technology of the codec is based on sub-band ADPCM (SB-ADPCM). The corresponding narrow-band codec based on the same technology is G.726.
Spectral efficiency, spectrum efficiency or bandwidth efficiency refers to the information rate that can be transmitted over a given bandwidth in a specific communication system. It is a measure of how efficiently a limited frequency spectrum is utilized by the physical layer protocol, and sometimes by the media access control.
Dolby Digital Plus, also known as Enhanced AC-3 is a digital audio compression scheme developed by Dolby Labs for transport and storage of multi-channel digital audio. It is a successor to Dolby Digital (AC-3), also developed by Dolby, and has a number of improvements including support for a wider range of data rates, increased channel count and multi-program support, and additional tools (algorithms) for representing compressed data and counteracting artifacts. While 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.
Perceptual audio coder (PAC) is an algorithm, like MPEG's MP3 standard, used to compress digital audio by removing extraneous information not perceived by most people. It is used by Sirius Satellite Radio for their DARS service.
MPEG Surround, also known as Spatial Audio Coding (SAC) is a lossy compression format for surround sound that provides a method for extending mono or stereo audio services to multi-channel audio in a backwards compatible fashion. The total bit rates used for the core and the MPEG Surround data are typically only slightly higher than the bit rates used for coding of the core. MPEG Surround adds a side-information stream to the core bit stream, containing spatial image data. Legacy stereo playback systems will ignore this side-information while players supporting MPEG Surround decoding will output the reconstructed multi-channel audio.
MPEG-1 Audio Layer I, commonly abbreviated to MP1, is one of three audio formats included in the MPEG-1 standard. It is a deliberately simplified version of MPEG-1 Audio Layer II, created for applications where lower compression efficiency could be tolerated in return for a less complex algorithm that could be executed with simpler hardware requirements. While supported by most media players, the codec is considered largely obsolete, and replaced by MP2 or MP3.
In signal processing, sub-band coding (SBC) is any form of transform coding that breaks a signal into a number of different frequency bands, typically by using a fast Fourier transform, and encodes each one independently. This decomposition is often the first step in data compression for audio and video signals.
Codec 2 is a low-bitrate speech audio codec that is patent free and open source. Codec 2 compresses speech using sinusoidal coding, a method specialized for human speech. Bit rates of 3200 to 450 bit/s have been successfully created. Codec 2 was designed to be used for amateur radio and other high compression voice applications.