Digital signal processing (DSP) is the use of digital processing, such as by computers or more specialized digital signal processors, to perform a wide variety of signal processing operations. The digital signals processed in this manner are a sequence of numbers that represent samples of a continuous variable in a domain such as time, space, or frequency. In digital electronics, a digital signal is represented as a pulse train, which is typically generated by the switching of a transistor.
MP3 is a coding format for digital audio developed largely by the Fraunhofer Society in Germany under the lead of Karlheinz Brandenburg, with support from other digital scientists in other countries. 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.
Vorbis is a free and open-source software project headed by the Xiph.Org Foundation. The project produces an audio coding format and software reference encoder/decoder (codec) for lossy audio compression. Vorbis is most commonly used in conjunction with the Ogg container format and it is therefore often referred to as Ogg Vorbis.
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.
Software-defined radio (SDR) is a radio communication system where components that conventionally have been implemented in analog hardware are instead implemented by means of software on a personal computer or embedded system. While the concept of SDR is not new, the rapidly evolving capabilities of digital electronics render practical many processes which were once only theoretically possible.
A software modem, commonly referred to as a softmodem, is a modem with minimal hardware that uses software running on the host computer, and the computer's resources, in place of the hardware in a conventional modem.
SETI@home is a project of the Berkeley SETI Research Center to analyze radio signals with the aim of searching for signs of extraterrestrial intelligence. Until March 2020, it was run as an Internet-based public volunteer computing project that employed the BOINC software platform. It is hosted by the Space Sciences Laboratory at the University of California, Berkeley, and is one of many activities undertaken as part of the worldwide SETI effort.
Ultra low frequency (ULF) is the ITU designation for the frequency range of electromagnetic waves between 300 hertz and 3 kilohertz, corresponding to wavelengths between 1,000 to 100 km. In magnetosphere science and seismology, alternative definitions are usually given, including ranges from 1 mHz to 100 Hz, 1 mHz to 1 Hz, and 10 mHz to 10 Hz.
Communication with submarines is a field within military communications that presents technical challenges and requires specialized technology. Because radio waves do not travel well through good electrical conductors like salt water, submerged submarines are cut off from radio communication with their command authorities at ordinary radio frequencies. Submarines can surface and raise an antenna above the sea level, or float a tethered buoy carrying an antenna, then use ordinary radio transmissions; however, this makes them vulnerable to detection by anti-submarine warfare forces.
ReplayGain is a proposed technical standard published by David Robinson in 2001 to measure and normalize the perceived loudness of audio in computer audio formats such as MP3 and Ogg Vorbis. It allows media players to normalize loudness for individual tracks or albums. This avoids the common problem of having to manually adjust volume levels between tracks when playing audio files from albums that have been mastered at different loudness levels.
Altacast is a free and open-source audio encoder that can be used to create Internet streams of varying types. Many independent and commercial broadcasters use Altacast to create Internet radio stations, such as those listed on the Icecast, Loudcaster and Shoutcast station directories.
Gapless playback is the uninterrupted playback of consecutive audio tracks, such that relative time distances in the original audio source are preserved over track boundaries on playback. For this to be useful, other artifacts at track boundaries should not be severed either. Gapless playback is common with compact discs, gramophone records, or tapes, but is not always available with other formats that employ compressed digital audio. The absence of gapless playback is a source of annoyance to listeners of music where tracks are meant to segue into each other, such as some classical music, progressive rock, concept albums, electronic music, and live recordings with audience noise between tracks.
Sound eXchange (SoX) is a cross-platform audio editing software. It has a command-line interface, and is written in standard C. It is free software, licensed under GPL-2.0-or-later, with libsox licensed under LGPL-2.1-or-later, and distributed by Chris Bagwell through SourceForge.
Digital room correction is a process in the field of acoustics where digital filters designed to ameliorate unfavorable effects of a room's acoustics are applied to the input of a sound reproduction system. Modern room correction systems produce substantial improvements in the time domain and frequency domain response of the sound reproduction system.
Underwater acoustic communication is a technique of sending and receiving messages below water. There are several ways of employing such communication but the most common is by using hydrophones. Underwater communication is difficult due to factors such as multi-path propagation, time variations of the channel, small available bandwidth and strong signal attenuation, especially over long ranges. Compared to terrestrial communication, underwater communication has low data rates because it uses acoustic waves instead of electromagnetic waves.
Constrained Energy Lapped Transform (CELT) is an open, royalty-free lossy audio compression format and a free software codec with especially low algorithmic delay for use in low-latency audio communication. The algorithms are openly documented and may be used free of software patent restrictions. Development of the format was maintained by the Xiph.Org Foundation and later coordinated by the Opus working group of the Internet Engineering Task Force (IETF).
Smaart is a suite of audio and acoustical measurements and instrumentation software tools introduced in 1996 by JBL's professional audio division. It is designed to help the live sound engineer optimize sound reinforcement systems before public performance and actively monitor acoustical parameters in real time while an audio system is in use. Most earlier analysis systems required specific test signals sent through the sound system, ones that would be unpleasant for the audience to hear. Smaart is a source-independent analyzer and therefore will work effectively with a variety of test signals including speech or music.
An audio analyzer is a test and measurement instrument used to objectively quantify the audio performance of electronic and electro-acoustical devices. Audio quality metrics cover a wide variety of parameters, including level, gain, noise, harmonic and intermodulation distortion, frequency response, relative phase of signals, interchannel crosstalk, and more. In addition, many manufacturers have requirements for behavior and connectivity of audio devices that require specific tests and confirmations.
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.
David W. Gunness is an American audio engineer, electrical engineer and inventor. He is known for his work on loudspeaker design, especially high-output professional horn loudspeakers for public address, studio, theater, nightclub, concert and touring uses.
