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G.722
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. G.722 provides improved speech quality due to a wider speech bandwidth of 50–7000 Hz compared to narrowband speech coders like G.711 which in general are optimized for POTS wireline quality of 300–3400 Hz. G.722 sample audio data at a rate of 16 kHz (using 14 bits), double that of traditional telephony interfaces, which results in superior audio quality and clarity. Other ITU-T 7 kHz wideband codecs include G.722.1 and G.722.2. These codecs are not variants of G.722 and they use different patented compression technologies. G.722.1 is based on Siren codecs and offers lower bit-rate compressions (24 kbit/s or 32 kbit/s). It uses a modified discrete cosine transform (MDCT) audio cod ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Adaptive Multi-Rate Wideband
Adaptive Multi-Rate Wideband (AMR-WB) is a patented Wideband audio, wideband speech coding, speech audio coding standard developed based on Adaptive Multi-Rate audio codec, 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 Plain old telephone service, 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 paramete ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Wideband Audio
Wideband audio, also known as wideband voice or HD voice, is high definition voice quality for telephony audio, contrasted with standard digital telephony "toll quality". It extends the frequency range of audio signals transmitted over telephone lines, resulting in higher quality speech. The range of the human voice extends from 100 Hz to 17 kHz but traditional, voiceband or narrowband telephone calls limit audio frequencies to the range of 300 Hz to 3.4 kHz. Wideband audio relaxes the bandwidth limitation and transmits in the audio frequency range of 50 Hz to 7 kHz. In addition, some wideband codecs may use a higher audio bit depth of 16 bits to encode samples, also resulting in much better voice quality. Wideband codecs have a typical sample rate of 16 kHz. For superwideband codecs the typical value is 32 kHz. History In 1987, the International Telecommunication Union (ITU) standardized a version of wideband audio known as G.722. Radi ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Wideband Audio
Wideband audio, also known as wideband voice or HD voice, is high definition voice quality for telephony audio, contrasted with standard digital telephony "toll quality". It extends the frequency range of audio signals transmitted over telephone lines, resulting in higher quality speech. The range of the human voice extends from 100 Hz to 17 kHz but traditional, voiceband or narrowband telephone calls limit audio frequencies to the range of 300 Hz to 3.4 kHz. Wideband audio relaxes the bandwidth limitation and transmits in the audio frequency range of 50 Hz to 7 kHz. In addition, some wideband codecs may use a higher audio bit depth of 16 bits to encode samples, also resulting in much better voice quality. Wideband codecs have a typical sample rate of 16 kHz. For superwideband codecs the typical value is 32 kHz. History In 1987, the International Telecommunication Union (ITU) standardized a version of wideband audio known as G.722. Radi ... [...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 of ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
List Of Codecs
The following is a list of compression formats and related codecs. Audio compression formats Non-compression * Linear pulse-code modulation (LPCM, generally only described as PCM) is the format for uncompressed audio in media files and it is also the standard for CD-DA; note that in computers, LPCM is usually stored in container formats such as WAV, AIFF, or AU, or as raw audio format, although not technically necessary. ** FFmpeg * Pulse-density modulation (PDM) ** Direct Stream Digital (DSD) is standard for Super Audio CD *** foobar2000 Super Audio CD Decoder (based on MPEG-4 DST reference decoder) *** FFmpeg (based on dsd2pcm) * Pulse-amplitude modulation (PAM) Lossless compression * Actively used ** Most popular *** Free Lossless Audio Codec (FLAC) **** libFLAC **** FFmpeg *** Apple Lossless Audio Codec (ALAC) **** Apple QuickTime **** libalac **** FFmpeg **** Apple Music *** Monkey's Audio (APE) **** Monkey's Audio SDK **** FFmpeg (decoder only) *** OptimFROG (OFR) *** T ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Siren (codec)
Siren is a family of patented, transform-based, wideband audio coding formats and their audio codec implementations developed and licensed by PictureTel Corporation (acquired by Polycom, Inc. in 2001). There are three Siren codecs: Siren 7, Siren 14 and Siren 22. Editions Siren 7 (or Siren7 or simply Siren) provides 7 kHz audio, bit rates 16, 24, 32 kbit/s and sampling frequency 16 kHz. Siren is derived from PictureTel's PT716plus algorithm. In 1999, ITU-T approved G.722.1 recommendation, which is based on Siren 7 algorithm. It was approved after a four-year selection process involving extensive testing. G.722.1 provides only bit rates 24 and 32 kbit/s and does not support Siren 7's bit rate 16 kbit/s. The algorithm of Siren 7 is identical to its successor, G.722.1, although the data formats are slightly different. Siren 14 (or Siren14) provides 14 kHz audio, bit rates 24, 32, 48 kbit/s for mono, 48, 64, 96 kbit/s for stereo and sampling frequency 32 kHz. Sire ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Sub-band ADPCM
Adaptive differential pulse-code modulation (ADPCM) is a variant of differential pulse-code modulation (DPCM) that varies the size of the quantization step, to allow further reduction of the required data bandwidth for a given signal-to-noise ratio. Typically, the adaptation to signal statistics in ADPCM consists simply of an adaptive scale factor before quantizing the difference in the DPCM encoder. ADPCM was developed for speech coding by P. Cummiskey, Nikil S. Jayant and James L. Flanagan at Bell Labs in 1973. In telephony In telephony, a standard audio signal for a single phone call is encoded as 8000 analog samples per second, of 8 bits each, giving a 64 kbit/s digital signal known as DS0. The default signal compression encoding on a DS0 is either μ-law (mu-law) PCM (North America and Japan) or A-law PCM (Europe and most of the rest of the world). These are logarithmic compression systems where a 13- or 14-bit linear PCM sample number is mapped into an 8-bit value. This sy ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Voice Over IP
Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for the delivery of speech, voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. The terms Internet telephony, broadband telephony, and broadband phone service specifically refer to the provisioning of communications services (voice, fax, Short Message Service, SMS, voice-messaging) over the Internet, rather than via the public switched telephone network (PSTN), also known as plain old telephone service (POTS). Overview The steps and principles involved in originating VoIP telephone calls are similar to traditional digital telephony and involve signaling, channel setup, digitization of the analog voice signals, and encoding. Instead of being transmitted over a circuit-switched network, the digital information is packetized and transmission occurs as IP packets over a packet-switched network. They transport media streams using spec ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Modified Discrete Cosine Transform
The modified discrete cosine transform (MDCT) is a transform based on the type-IV discrete cosine transform (DCT-IV), with the additional property of being lapped transform, lapped: it is designed to be performed on consecutive blocks of a larger dataset, where subsequent blocks are overlapped so that the last half of one block coincides with the first half of the next block. This overlapping, in addition to the energy-compaction qualities of the DCT, makes the MDCT especially attractive for signal compression applications, since it helps to avoid compression artifact, artifacts stemming from the block boundaries. As a result of these advantages, the MDCT is the most widely used lossy compression technique in audio data compression. It is employed in most modern audio coding standards, including MP3, Dolby Digital (AC-3), Vorbis (Ogg), Windows Media Audio (WMA), ATRAC, Cook codec, Cook, Advanced Audio Coding (AAC), High-Definition Coding (HDC), LDAC (codec), LDAC, Dolby AC-4, and MP ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
ACELP
Algebraic code-excited linear prediction (ACELP) is a speech coding algorithm in which a limited set of pulses is distributed as excitation to a linear prediction filter. It is a linear predictive coding (LPC) algorithm that is based on the code-excited linear prediction (CELP) method and has an algebraic structure. ACELP was developed in 1989 by the researchers at the Université de Sherbrooke in Canada. The ACELP method is widely employed in current speech coding standards such as AMR, EFR, AMR-WB (G.722.2), VMR-WB, EVRC, EVRC-B, SMV, TETRA, PCS 1900, MPEG-4 CELP and ITU-T G-series standards G.729, G.729.1 (first coding stage) and G.723.1. The ACELP algorithm is also used in the proprietary ACELP.net codec. Audible Inc. use a modified version for their speaking books. It is also used in conference-calling software, speech compression tools and has become one of the 3GPP formats. The ACELP patent expired in 2018 and is now royalty-free. Features The main advantage of ACE ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Request For Comments
A Request for Comments (RFC) is a publication in a series from the principal technical development and standards-setting bodies for the Internet, most prominently the Internet Engineering Task Force (IETF). An RFC is authored by individuals or groups of engineers and computer scientists in the form of a memorandum describing methods, behaviors, research, or innovations applicable to the working of the Internet and Internet-connected systems. It is submitted either for peer review or to convey new concepts, information, or, occasionally, engineering humor. The IETF adopts some of the proposals published as RFCs as Internet Standards. However, many RFCs are informational or experimental in nature and are not standards. The RFC system was invented by Steve Crocker in 1969 to help record unofficial notes on the development of ARPANET. RFCs have since become official documents of Internet specifications, communications protocols, procedures, and events. According to Crocker, the doc ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Internet Standard
In computer network engineering, an Internet Standard is a normative specification of a technology or methodology applicable to the Internet. Internet Standards are created and published by the Internet Engineering Task Force (IETF). They allow interoperation of hardware and software from different sources which allows internets to function. As the Internet became global, Internet Standards became the lingua franca of worldwide communications. Engineering contributions to the IETF start as an Internet Draft, may be promoted to a Request for Comments, and may eventually become an Internet Standard. An Internet Standard is characterized by technical maturity and usefulness. The IETF also defines a Proposed Standard as a less mature but stable and well-reviewed specification. A Draft Standard was an intermediate level, discontinued in 2011. A Draft Standard was an intermediary step that occurred after a Proposed Standard but prior to an Internet Standard. As put in RFC 2026: In ge ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
