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G.726
G.726 is an ITU-T ADPCM speech codec standard covering the transmission of voice at rates of 16, 24, 32, and 40 kbit/s. It was introduced to supersede both G.721, which covered ADPCM at 32 kbit/s, and G.723, which described ADPCM for 24 and 40 kbit/s. G.726 also introduced a new 16 kbit/s rate. The four bit rates associated with G.726 are often referred to by the bit size of a sample, which are 2, 3, 4, and 5-bits respectively. The corresponding wide-band codec based on the same technology is G.722. The most commonly used mode is 32 kbit/s, which doubles the usable network capacity by using half the rate of G.711. It is primarily used on international trunks in the phone network and is the standard codec used in DECT wireless phone systems. The principal application of 24 and 16 kbit/s channels is for overload channels carrying voice in digital circuit multiplication equipment (DCME). The principal application of 40 kbit/s channels is to carry da ... [...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 ... [...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] |
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 s ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Speech Codec
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 recons ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
DECT
Digital enhanced cordless telecommunications (Digital European cordless telecommunications), usually known by the acronym DECT, is a standard primarily used for creating cordless telephone systems. It originated in Europe, where it is the common standard, replacing earlier cordless phone standards, such as 900 MHz CT1 and CT2. Beyond Europe, it has been adopted by Australia and most countries in Asia and South America. North American adoption was delayed by United States radio-frequency regulations. This forced development of a variation of DECT called DECT 6.0, using a slightly different frequency range, which makes these units incompatible with systems intended for use in other areas, even from the same manufacturer. DECT has almost completely replaced other standards in most countries where it is used, with the exception of North America. DECT was originally intended for fast roaming between networked base stations, and the first DECT product was Net3 wireless LA ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
ITU-T
The ITU Telecommunication Standardization Sector (ITU-T) is one of the three sectors (divisions or units) of the International Telecommunication Union (ITU). It is responsible for coordinating standards for telecommunications and Information Communication Technology such as X.509 for cybersecurity, Y.3172 and Y.3173 for machine learning, and H.264/MPEG-4 AVC for video compression, between its Member States, Private Sector Members, and Academia Members. The first meeting of the World Telecommunication Standardization Assembly (WTSA), the sector's governing conference, took place on 1 March of that year. ITU-T has a permanent secretariat called the Telecommunication Standardization Bureau (TSB), which is based at the ITU headquarters in Geneva, Switzerland. The current director of the TSB is Chaesub Lee (of South Korea), whose first 4-year term commenced on 1 January 2015, and whose second 4-year term commenced on 1 January 2019. Chaesub Lee succeeded Malcolm Johnson of the Uni ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Speech Codecs
Speech is a human vocal communication using language. Each language uses phonetic combinations of vowel and consonant sounds that form the sound of its words (that is, all English words sound different from all French words, even if they are the same word, e.g., "role" or "hotel"), and using those words in their semantic character as words in the lexicon of a language according to the syntactic constraints that govern lexical words' function in a sentence. In speaking, speakers perform many different intentional speech acts, e.g., informing, declaring, asking, persuading, directing, and can use enunciation, intonation, degrees of loudness, tempo, and other non-representational or paralinguistic aspects of vocalization to convey meaning. In their speech, speakers also unintentionally communicate many aspects of their social position such as sex, age, place of origin (through accent), physical states (alertness and sleepiness, vigor or weakness, health or illness), psychologica ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Audio Codecs
An audio codec is a device or computer program capable of encoding or decoding a digital data stream (a codec) that encodes or decodes audio. In software, an audio codec is a computer program implementing an algorithm that compresses and decompresses digital audio data according to a given audio file or streaming media audio coding format. The objective of the algorithm is to represent the high-fidelity audio signal with minimum number of bits while retaining quality. This can effectively reduce the storage space and the bandwidth required for transmission of the stored audio file. Most software codecs are implemented as libraries which interface to one or more multimedia players. Most modern audio compression algorithms are based on modified discrete cosine transform (MDCT) coding and linear predictive coding (LPC). In hardware, audio codec refers to a single device that encodes analog audio as digital signals and decodes digital back into analog. In other words, it contains bot ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Mean Opinion Score
Mean opinion score (MOS) is a measure used in the domain of Quality of Experience and telecommunications engineering, representing overall quality of a stimulus or system. It is the arithmetic mean over all individual "values on a predefined scale that a subject assigns to his opinion of the performance of a system quality". Such ratings are usually gathered in a subjective quality evaluation test, but they can also be algorithmically estimated. MOS is a commonly used measure for video, audio, and audiovisual quality evaluation, but not restricted to those modalities. ITU-T has defined several ways of referring to a MOS in RecommendatioITU-T P.800.1 depending on whether the score was obtained from audiovisual, conversational, listening, talking, or video quality tests. Rating scales and mathematical definition The MOS is expressed as a single rational number, typically in the range 1–5, where 1 is lowest perceived quality, and 5 is the highest perceived quality. Other MOS ranges ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
PSQM
Perceptual Speech Quality Measure (PSQM) is a computational and modeling algorithm defined in Recommendation ITU-T P.861 that objectively evaluates and quantifies voice quality of voice-band (300 – 3400 Hz) speech codecs. It may be used to rank the performance of these speech codecs with differing speech input levels, talkers, bit rates and transcodings. P.861 was withdrawn and replaced by Recommendation ITU-T P.862 ( PESQ), which contains an improved speech assessment algorithm. Why it is used Using the PSQM standard allows automated, simulation-based test methodologies to objectively rate both speech clarity and transmitted voice quality. Various software and/or hardware products have been developed to facilitate this testing. This results in considerable savings in cost and time over the traditional practice of using large groups of people to subjectively evaluate voice signals and assess voice quality. Moreover, it yields objective results that are reliable and reprodu ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
