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Libgsm
Full Rate (FR or GSM-FR or GSM 06.10 or sometimes simply GSM) was the first digital speech coding standard used in the GSM digital mobile phone system. It uses linear predictive coding (LPC). The bit rate of the codec is 13 kbit/s, or 1.625 bits/audio sample (often padded out to 33 bytes/20 ms or 13.2 kbit/s). The quality of the coded speech is quite poor by modern standards, but at the time of development (early 1990s) it was a good compromise between computational complexity and quality, requiring only on the order of a million additions and multiplications per second. The codec is still widely used in networks around the world. Gradually FR will be replaced by Enhanced Full Rate (EFR) and Adaptive Multi-Rate (AMR) standards, which provide much higher speech quality with lower bit rate. Technology ''GSM-FR'' is specified in ETSI 06.10 (ETS 300 961) and is based on RPE-LTP ( Regular Pulse Excitation - Long Term Prediction) speech coding paradigm. Like many other linear predict ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Comparison Of Audio Coding Formats
The following tables compare general and technical information for a variety of audio coding formats. For listening tests comparing the perceived audio quality of audio formats and codecs, see the article Codec listening test. General information Notes # The 'Music' category is merely a guideline on commercialized uses of a particular format, not a technical assessment of its capabilities. (For example, in terms of marketshare, MP3 and AAC dominate the personal audio market, though many other formats are comparably well suited to fill this role from a purely technical standpoint.) # First public release date is first of either specification publishing or source releasing, or in the case of closed-specification, closed-source codecs, is the date of first binary releasing. Many developing codecs have pre-releases consisting of pre-1.0 versions and perhaps 1.0 release candidates (RCs), although 1.0 may not necessarily be the release version. # Latest stable version is that o ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Speech Coding
Speech coding is an application of data compression of digital audio signals containing speech. Speech coding uses speech-specific parameter estimation using audio signal processing techniques to model the speech signal, combined with generic data compression algorithms to represent the resulting modeled parameters in a compact bitstream. Some applications of speech coding are mobile telephony and voice over IP (VoIP). The most widely used speech coding technique in mobile telephony is linear predictive coding (LPC), while the most widely used in VoIP applications are the LPC and modified discrete cosine transform (MDCT) techniques. The techniques employed in speech coding are similar to those used in audio data compression and audio coding where knowledge in psychoacoustics is used to transmit only data that is relevant to the human auditory system. For example, in voiceband speech coding, only information in the frequency band 400 to 3500 Hz is transmitted but the reco ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Technische Universität Berlin
The Technical University of Berlin (official name both in English and german: link=no, Technische Universität Berlin, also known as TU Berlin and Berlin Institute of Technology) is a public research university located in Berlin, Germany. It was the first German university to adopt the name "Technische Universität" (Technical University). The university alumni and professor list includes several US National Academies members, two National Medal of Science laureates and ten Nobel Prize laureates. TU Berlin is a member of TU9, an incorporated society of the largest and most notable German institutes of technology and of the Top International Managers in Engineering network, which allows for student exchanges between leading engineering schools. It belongs to the Conference of European Schools for Advanced Engineering Education and Research. The TU Berlin is home of two innovation centers designated by the European Institute of Innovation and Technology. The university is labele ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
RTP Audio Video Profile
The Real-time Transport Protocol (RTP) specifies a general-purpose data format and network protocol for transmitting digital media streams on Internet Protocol (IP) networks. The details of media encoding, such as signal sampling rate, frame size and timing, are specified in an RTP payload format. The format parameters of the RTP payload are typically communicated between transmission endpoints with the Session Description Protocol (SDP), but other protocols, such as the Extensible Messaging and Presence Protocol (XMPP) may be used. Audio and video payload types RFC 3551, entitled RTP Profile for Audio and Video (RTP/AVP), specifies the technical parameters of payload formats for audio and video streams. The standard also describes the process of registering new payload types with IANA; additional payload formats and payload types are defined in the following specifications: * , Standard 65, ''RTP Profile for Audio and Video Conferences with Minimal Control'' * , ''Media Type Regi ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Extended Adaptive Multi-Rate - Wideband
Extension, extend or extended may refer to: Mathematics Logic or set theory * Axiom of extensionality * Extensible cardinal * Extension (model theory) * Extension (predicate logic), the set of tuples of values that satisfy the predicate * Extension (semantics), the set of things to which a property applies * Extension by definitions * Extensional definition, a definition that enumerates every individual a term applies to * Extensionality Other uses * Extension of a polyhedron, in geometry * Exterior algebra, Grassmann's theory of extension, in geometry * Homotopy extension property, in topology * Kolmogorov extension theorem, in probability theory * Linear extension, in order theory * Sheaf extension, in algebraic geometry * Tietze extension theorem, in topology * Whitney extension theorem, in differential geometry * Group extension, in abstract algebra and homological algebra Music * Extension (music), notes that fit outside the standard range * ''Extended'' (Solar Field ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Adaptive Multi-Rate Wideband
Adaptive Multi-Rate Wideband (AMR-WB) is a patented wideband speech audio coding standard developed based on Adaptive Multi-Rate encoding, using a similar methodology to algebraic code-excited linear prediction (ACELP). AMR-WB provides improved speech quality due to a wider speech bandwidth of 50–7000 Hz compared to narrowband speech coders which in general are optimized for POTS wireline quality of 300–3400 Hz. AMR-WB was developed by Nokia and VoiceAge and it was first specified by 3GPP. AMR-WB is codified as G.722.2, an ITU-T standard speech codec, formally known as ''Wideband coding of speech at around 16 kbit/s using Adaptive Multi-Rate Wideband (AMR-WB)''. G.722.2 AMR-WB is the same codec as the 3GPP AMR-WB. The corresponding 3GPP specifications are TS 26.190 for the speech codec and TS 26.194 for the Voice Activity Detector. The AMR-WB format has the following parameters: * Frequency bands processed: 50–6400 Hz (all modes) plus 6400–7000 Hz (23 ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Half Rate
Half Rate (HR or GSM-HR or GSM 06.20) is a speech coding system for GSM, developed in the early 1990s. Since the codec, operating at 5.6 kbit/s, requires half the bandwidth of the Full Rate codec, network capacity for voice traffic is doubled, at the expense of audio quality. The sampling rate is 8 kHz with resolution 13 bit, frame length 160 samples (20 ms) and subframe length 40 samples (5 ms). GSM Half Rate is specified in ETSI EN 300 969 (GSM 06.20), and uses a form of the VSELP algorithm. Previous specification was in ETSI ETS 300 581–2, which first edition was published in December 1995. Retrieved on 2009-07-11 For some Nokia phones one can configure the use of this codec: * To ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Ventrilo
Ventrilo (or Vent for short) is a proprietary VoIP software that includes text chat. The Ventrilo client and server are both available as freeware for use with up to 8 people on the same server. Rented servers can maintain up to 400 people. The Ventrilo server is available under a limited license for Microsoft Windows and macOS and is accessible on FreeBSD Kopi, Solaris and NetBSD. The client is available for Windows and macOS. However, the macOS client is still unable to properly use most servers because of a lack of support for the sparsely used GSM codec. Flagship Industries does not offer a Linux Ventrilo client. Third party Ventrilo clients are available for mobile devices, such as Ventrilode for iPhone and Ventriloid for Android. Ventrilo supports the GSM Full Rate codec and used to support the Speex codec, which Ventrilo 4.0.0 replaced with the Opus codec. Usage Ventrilo is sometimes used by gamers who use the software to communicate with other players on the s ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Asterisk (PBX)
Asterisk is a software implementation of a private branch exchange (PBX). In conjunction with suitable telephony hardware interfaces and network applications, Asterisk is used to establish and control telephone calls between telecommunication endpoints, such as customary telephone sets, destinations on the public switched telephone network (PSTN), and devices or services on voice over Internet Protocol (VoIP) networks. Its name comes from the asterisk (*) symbol for a signal used in dual-tone multi-frequency (DTMF) dialing. Asterisk was created in 1999 by Mark Spencer of Digium, which since 2018 is a division of Sangoma Technologies Corporation. Originally designed for Linux, Asterisk runs on a variety of operating systems, including NetBSD, OpenBSD, FreeBSD, macOS, and Solaris, and can be installed in embedded systems based on OpenWrt. Features The Asterisk software includes many features available in commercial and proprietary PBX systems: voice mail, conference callin ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Linphone
__NOTOC__ Linphone (contraction of ''Linux phone'') is a free voice over IP softphone, SIP client and service. It may be used for audio and video direct calls and calls through any VoIP softswitch or IP-PBX. Linphone also provides the possibility to exchange instant messages. It has a simple multilanguage interface based on Qt for GUI and can also be run as a console-mode application on Linux. The softphone is currently developed by Belledonne Communications in France. Linphone was initially developed for Linux but now supports many additional platforms including Microsoft Windows, macOS, and mobile phones running Windows Phone, iOS or Android. It supports ZRTP for end-to-end encrypted voice and video communication. Linphone is licensed under the GNU GPL-3.0-or-later and supports IPv6. Linphone can also be used behind network address translator (NAT), meaning it can run behind home routers. It is compatible with telephony by using an Internet telephony service provider (ITS ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
QuteCom
QuteCom (previously called WengoPhone) was a free-software SIP-compliant VoIP client developed by the QuteCom (previously OpenWengo) community under the GPL-2.0-or-later license. It allows users to speak to other users of SIP-compliant VoIP software at no cost. It also allows users to call landlines and cell phones, send SMS and make video calls. None of these functions are tied to a particular provider, allowing users to choose among any SIP provider. History Development on WengoPhone began in September 2004. The first published version was released as version 0.949. A next-generation version of WengoPhone (WengoPhone NG) began development in 2005. The last release of WengoPhone before the change to QuteCom was version 2.1.2. A WengoPhone Firefox plug-in has also been published, which is currently available on Mac OS X and Microsoft Windows, with a Linux version under development. On 28 January 2008, Wengo, the original sponsor of WengoPhone, transferred the sponsorship of ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Ekiga
Ekiga (formerly called GnomeMeeting) is a VoIP and video conferencing application for GNOME and Microsoft Windows. It is distributed as free software under the terms of the GNU GPL-2.0-or-later. It was the default VoIP client in Ubuntu until October 2009, when it was replaced by Empathy. Ekiga supports both the SIP and H.323 (based on OPAL) protocols and is fully interoperable with any other SIP compliant application and with Microsoft NetMeeting. It supports many high-quality audio and video codecs. Ekiga was initially written by Damien Sandras in order to graduate from the University of Louvain (UCLouvain). It is currently developed by a community-based team led by Sandras. The logo was designed based on his concept by Andreas Kwiatkowski. Ekiga.net was also a free and private SIP registrar, which enabled its members to originate and terminate (receive) calls from and to each other directly over the Internet. The service was discontinued at the end of 2018. Features Feature ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
