Bit rate in the context of Low-power wide-area network


Bit rate in the context of Low-power wide-area network

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⭐ Core Definition: Bit rate

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.

The bit rate is expressed in the unit bit per second (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 = 1,000 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.

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Bit rate in the context of Lossless compression

Lossless compression is a class of data compression that allows the original data to be perfectly reconstructed from the compressed data with no loss of information. Lossless compression is possible because most real-world data exhibits statistical redundancy. By contrast, lossy compression permits reconstruction only of an approximation of the original data, though usually with greatly improved compression rates (and therefore reduced media sizes).

By operation of the pigeonhole principle, no lossless compression algorithm can shrink the size of all possible data: Some data will get longer by at least one symbol or bit.

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Bit rate in the context of DVD-Video

DVD-Video is a consumer video format used to store digital video on DVDs. DVD-Video was the dominant consumer home video format in most of the world in the 2000s. As of 2025, it continues to compete with its high-definition Blu-ray Disc counterpart, while both receive competition as the collective delivery method of physical media by streaming services such as Netflix and Disney+. Discs using the DVD-Video specification require a DVD drive and an MPEG-2 decoder (e.g., a DVD player, or a computer DVD drive with a software DVD player). Commercial DVD movies are encoded using a combination of MPEG-2 compressed video and audio of varying formats (often multi-channel formats as described below). Typically, the data rate for DVD movies ranges from 3 to 9.5 Mbit/s, and the bit rate is usually adaptive. DVD-Video was first available in Japan on October 19, 1996 (with major releases beginning December 20, 1996), followed by a release on March 24, 1997, in the United States.

The DVD-Video specification was created by the DVD Forum and was not publicly available. Certain information in the DVD Format Books is proprietary and confidential and Licensees and Subscribers were required to sign a non-disclosure agreement. The DVD-Video Format Book could be obtained from the DVD Format/Logo Licensing Corporation (DVD FLLC) for a fee of $5,000. FLLC announced in 2024 that "On December 31, 2024, the current DVD Format/Logo License ("License") will expire. On the same date, our Licensing program, which originally started from 2000, will be terminated. There will be no new License program available and thus no License renewal is required."

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Bit rate in the context of MP3

MP3 (formally MPEG-1 Audio Layer III or MPEG-2 Audio Layer III) is an audio coding format 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 Audio Layer III was originally defined in 1991 as one of the three possible audio codecs of the MPEG-1 standard (along with MPEG-1 Audio Layer I and MPEG-1 Audio Layer II). All three options were retained and further extended—defining additional bit rates and support for more audio channels (supporting surround sound—in the subsequent MPEG-2 standard).

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Bit rate in the context of Audio bit depth

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.

In basic implementations, variations in bit depth primarily affect the noise level from quantization error—thus the signal-to-noise ratio (SNR) and dynamic range. However, techniques such as dithering, noise shaping, and oversampling can mitigate these effects without changing the bit depth. Bit depth also affects bit rate and file size.

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Bit rate in the context of Ethernet

Ethernet (/ˈθərnɛt/ EE-thər-net) is a family of wired computer networking technologies commonly used in local area networks (LAN), metropolitan area networks (MAN) and wide area networks (WAN). It was commercially introduced in 1980 and first standardized in 1983 as IEEE 802.3. Ethernet has since been refined to support higher bit rates, a greater number of nodes, and longer link distances, but retains much backward compatibility. Over time, Ethernet has largely replaced competing wired LAN technologies such as Token Ring, FDDI and ARCNET.

The original 10BASE5 Ethernet uses a thick coaxial cable as a shared medium. This was largely superseded by 10BASE2, which used a thinner and more flexible cable that was both less expensive and easier to use. More modern Ethernet variants use twisted pair and fiber optic links in conjunction with switches. Over the course of its history, Ethernet data transfer rates have been increased from the original 2.94 Mbit/s to the latest 800 Gbit/s, with rates up to 1.6 Tbit/s under development. The Ethernet standards include several wiring and signaling variants of the OSI physical layer.

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Bit rate in the context of Quality of service

Quality of service (QoS) is the description or measurement of the overall performance of a service, such as a telephony or computer network, or a cloud computing service, particularly the performance seen by the users of the network. To quantitatively measure quality of service, several related aspects of the network service are often considered, such as packet loss, bit rate, throughput, transmission delay, availability, jitter, etc.

In the field of computer networking and other packet-switched telecommunication networks, quality of service refers to traffic prioritization and resource reservation control mechanisms rather than the achieved service quality. Quality of service is the ability to provide different priorities to different applications, users, or data flows, or to guarantee a certain level of performance to a data flow.

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Bit rate in the context of SONET

Synchronous Optical Networking (SONET) and Synchronous Digital Hierarchy (SDH) are standardized protocols that transfer multiple digital bit streams synchronously over optical fiber using lasers or highly coherent light from light-emitting diodes (LEDs). At low transmission rates, data can also be transferred via an electrical interface. The method was developed to replace the plesiochronous digital hierarchy (PDH) system for transporting large amounts of telephone calls and data traffic over the same fiber without the problems of synchronization.

SONET and SDH, which are essentially the same, were originally designed to transport circuit mode communications, e.g. DS1, DS3, from a variety of different sources. However, they were primarily designed to support real-time, uncompressed, circuit-switched voice encoded in PCM format. The primary difficulty in doing this prior to SONET/SDH was that the synchronization sources of these various circuits were different. This meant that each circuit was actually operating at a slightly different rate and with different phase. SONET/SDH allowed for the simultaneous transport of many different circuits of differing origin within a single framing protocol. SONET/SDH is not a complete communications protocol in itself, but a transport protocol (not a "transport" in the OSI Model sense).

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Bit rate in the context of Advanced Audio Coding

Advanced Audio Coding (AAC) is an audio coding standard for lossy digital audio compression. It was developed by Dolby, AT&T, Fraunhofer and Sony, originally as part of the MPEG-2 specification but later improved under MPEG-4. AAC was designed to be the successor of the MP3 format (MPEG-2 Audio Layer III) and generally achieves higher sound quality than MP3 at the same bit rate. AAC encoded audio files are typically packaged in an MP4 container most commonly using the filename extension .m4a.

The basic profile of AAC (both MPEG-4 and MPEG-2) is called AAC-LC (Low Complexity). It is widely supported in the industry and has been adopted as the default or standard audio format on products including Apple's iTunes Store, Nintendo's Wii, DSi and 3DS and Sony's PlayStation 3. It is also further supported on various other devices and software such as iPhone, iPod, PlayStation Portable and Vita, PlayStation 5, Android and older cell phones, digital audio players like Sony Walkman and SanDisk Clip, media players such as VLC, Winamp and Windows Media Player, various in-dash car audio systems, and is used on Spotify, Google Nest, Amazon Alexa. Apple Music, YouTube and also YouTube Music streaming services. AAC has been further extended into HE-AAC (High Efficiency, or AAC+), which improves efficiency over AAC-LC. Another variant is AAC-LD (Low Delay).

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Bit rate in the context of H.264/MPEG-4 AVC

Advanced Video Coding (AVC), also referred to as H.264 or MPEG-4 Part 10, is a video compression standard based on block-oriented, motion-compensated coding. It is by far the most commonly used format for the recording, compression, and distribution of video content, used by 79% of video industry developers as of December 2024. It supports a maximum resolution of 8K UHD.

The intent of the H.264/AVC project was to create a standard capable of providing good video quality at substantially lower bit rates than previous standards (i.e., half or less the bit rate of MPEG-2, H.263, or MPEG-4 Part 2), without increasing the complexity of design so much that it would be impractical or excessively expensive to implement. This was achieved with features such as a reduced-complexity integer discrete cosine transform (integer DCT), variable block-size segmentation, and multi-picture inter-picture prediction. An additional goal was to provide enough flexibility to allow the standard to be applied to a wide variety of applications on a wide variety of networks and systems, including low and high bit rates, low and high resolution video, broadcast, DVD storage, RTP/IP packet networks, and ITU-T multimedia telephony systems. The H.264 standard can be viewed as a "family of standards" composed of a number of different profiles, although its "High profile" is by far the most commonly used format. A specific decoder decodes at least one, but not necessarily all profiles. The standard describes the format of the encoded data and how the data is decoded, but it does not specify algorithms for encoding—that is left open as a matter for encoder designers to select for themselves, and a wide variety of encoding schemes have been developed. H.264 is typically used for lossy compression, although it is also possible to create truly lossless-coded regions within lossy-coded pictures or to support rare use cases for which the entire encoding is lossless.

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Bit rate in the context of Asymmetric digital subscriber line

Asymmetric digital subscriber line (ADSL) is a type of digital subscriber line (DSL) technology, a data communications technology that enables faster data transmission over copper telephone lines than a conventional voiceband modem can provide. ADSL differs from the less common symmetric digital subscriber line (SDSL). In ADSL, bandwidth and bit rate are said to be asymmetric, meaning greater toward the customer premises (downstream) than the reverse (upstream). Providers usually market ADSL as an Internet access service primarily for downloading content from the Internet, but not for serving content accessed by others.

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Bit rate in the context of Data-rate units

In telecommunications, data rate units are commonly multiples of bits per second (bit/s) and bytes per second (B/s). For example, the data rates of modern residential high-speed Internet connections are commonly expressed in megabits per second (Mbit/s).They are used as units of measurement for expressing data transfer rate, the average number of bits (bit rate), characters or symbols (symbol rate), or data blocks per unit time passing through a communication link in a data-transmission system.

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Bit rate in the context of Windows Media Audio

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. Audio encoded in WMA is stored in a digital container format called Advanced Systems Format (ASF).

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 (the regular WMA format is lossy). WMA Voice, targeted at voice content, applies compression using a range of low bit rates.

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Bit rate in the context of Compression artifact

A compression artifact (or artefact) is a noticeable distortion of media (including images, audio, and video) caused by the application of lossy compression. Lossy data compression involves discarding some of the media's data so that it becomes small enough to be stored within the desired disk space or transmitted (streamed) within the available bandwidth (known as the data rate or bit rate). If the compressor cannot store enough data in the compressed version, the result is a loss of quality, or introduction of artifacts. The compression algorithm may not be intelligent enough to discriminate between distortions of little subjective importance and those objectionable to the user.

The most common digital compression artifacts are DCT blocks, caused by the discrete cosine transform (DCT) compression algorithm used in many digital media standards, such as JPEG, MP3, and MPEG video file formats. These compression artifacts appear when heavy compression is applied, and occur often in common digital media, such as DVDs, common computer file formats such as JPEG, MP3 and MPEG files, and some alternatives to the compact disc, such as Sony's MiniDisc format. Uncompressed media (such as on Laserdiscs, Audio CDs, and WAV files) or losslessly compressed media (such as FLAC or PNG) do not suffer from compression artifacts.

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Bit rate in the context of High Efficiency Video Coding

High Efficiency Video Coding (HEVC), also known as H.265 and MPEG-H Part 2, is a proprietary video compression standard designed as part of the MPEG-H project as a successor to the widely used Advanced Video Coding (AVC, H.264, or MPEG-4 Part 10). The standard was published in 2013. In comparison to AVC, HEVC offers from 25% to 50% better data compression at the same level of video quality, or substantially improved video quality at the same bit rate. It supports resolutions up to 8192×4320, including 8K UHD, and unlike the primarily eight-bit AVC, HEVC's higher-fidelity Main 10 profile has been incorporated into nearly all supporting hardware. The High Efficiency Image Format (HEIF) is a container format whose default codec is HEVC.

While AVC uses the integer discrete cosine transform (DCT) with 4×4 and 8×8 block sizes, HEVC uses both integer DCT and discrete sine transform (DST) with varied block sizes between 4×4 and 32×32.

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Bit rate in the context of Linear predictive coding

Linear predictive coding (LPC) is a method used mostly in audio signal processing and speech processing for representing the spectral envelope of a digital signal of speech in compressed form, using the information of a linear predictive model.

LPC is the most widely used method in speech coding and speech synthesis. It is a powerful speech analysis technique, and a useful method for encoding good quality speech at a low bit rate.

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