Codecs
Source coding is associated with using variable-duration
codes for lowering symbols to a bare minimum with the intention to represent
the records inside the transmitting message. The lossless records compression
reduces bits by means of identifying and getting rid of statistical redundancy.
However, data compression is depending on space-time complexity tradeoff (like the
degree of compression, the amount of distortion delivered whilst the usage of
lossy information compression, computational sources required to compress and
decompress the statistics). The most desirable layout principle of a codec,
therefore, lies in meticulous bargaining of the aforesaid trade-off, which aims
to maintain the QoS necessities. nanobiztech
Specification of speech and audio formats has been one in
all ETSI's (and 3GPP’s) many fulfillment memories. globalmarketingbusiness
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Enhanced Voice Services (EVS codec
The codec for EVS, the successor of the modern cellular
excessive definition (HD) voice codec AMR-WB, was standardized by way of the
third Generation Partnership Project (3GPPTM) in September 2014. The EVS codec
addresses 3GPP's needs for present-day generation enabling the operation of
3GPPTM cellular verbal exchange structures in the most competitive approach in
phrases of communication great and performance. allinonetechs
The EVS codec enhances coding efficiency and first-class for
NarrowBand (NB) and Wide Band (WB) for a large bit charge range, starting from
5.9 kbps variable bit charge (VBR). It similarly offers an enormous step in
great over these traditional telephony bandwidths with Super Wide Band (SWB)
and Full Band (FB) operation starting from nine. Six and sixteen.Four kbps,
respectively. The maximum bit charge is 128 kbps. The ability to switch the bit
fee at each 20-ms. leadmarketingbusiness
A mistakes concealment mechanism mitigates the quality
effect of channel mistakes ensuing in lost packets. The codec additionally
consists of a machine for jitter buffer management (JBM). Furthermore, it
capabilities a unique channel-conscious mode accomplishing increased robustness
in especially unfavorable channel conditions. Enhanced interoperation with
AMR-WB is supplied overall 9 bit quotes between 6.6 kbps and 23. Eighty-five
kbps. technologycompanians
Audio formats
Mobile streaming audio and messaging services might also
comprise speech best, tune handiest, or speech combined with music on the background.
For this predicted blended streaming content material, the codecs described so
far have difficulties in constantly acting properly for each speech and track
at low bit-charges (i.E. Well under 32 kbit/s). answerhop
Radio sources and channel capability set in addition
barriers on data charges available for streaming. Streamed audio content has to
be made available at a low bit-fee well under 32 kbit/s, corresponding to the
bit-rate variety already used inside the AMR-WB codec. If video is covered
within the content material, the records fee ought to be as little as feasible.
Improved aacPlus codec
It is an extended and advanced version based totally on the
advocated Release five Audio codec AAC-LC. It is optimized for high audio
satisfactory at low bitrates and is therefore well appropriate for offerings
including Packed-Switched Streaming carrier (PSS), Multimedia Messaging Service
(MMS), Multimedia Broadcast Multicast Service (MBMS), and Presence. The
Enhanced aacPlus codec offers the following competencies:
·
Excellent (CD-like) audio excellent at bitrates
properly beneath 64 kbit/s
·
Efficient stereo modes, allowing high
high-quality stereo starting at bitrates below 24 kbit/s
·
Music excellent across the whole bit-rate
exceeding that of any other audio codec regarded today
·
Flexible configuration permitting the use of any
precise bit-fee starting from 8 kbit/s
·
The low computational complexity for decoder and
encoder
·
Fully laid out in 3GPPTM, along with optimized
floating-point and stuck-factor supply code
Extended Adaptive Multi-Rate - Wideband (AMR-WB+) codec
The Extended AMR-WB codec (AMR-WB+) began to start with
centered on wideband programs; it extends the AMR-WB codec (with new modes) for
use in packet-switched streaming and messaging offerings, as well as for MBMS,
IMS Messaging, and Presence. As this codec simply brings additional modes to
the prevailing AMR-WB codec, there are no service or architectural influences.
The work, therefore, contained in attractive the AMR-WB
codec for audio packages with the aid of growing an audio extension based
totally at the 3GPPTM AMR-WB speech codec. The audio extension is commonly
intended for non-conversational offerings. Among the principal goals of the
audio extension have been:
·
High perceptual excellent with speech, music,
and combined content material
·
Music overall performance comparable to the
exceptional of today's audio formats
·
Speech overall performance at the least as
suitable as that of AMR-WB
·
Similar bit-fees as the AMR-WB codec to make
sure green use of radio sources
·
Mono and stereo coding
Adaptive Multi-Rate Wideband (AMR-WB) codec
In March 2001, 3GPPTM accepted the technical specifications
for the Adaptive Multi-Rate Wideband (AMR-WB) coding set of rules as a part of
3GPPTM Release five. The International Cable Union (ITU-T) Study Group 16
permitted the same wideband coding set of rules as Recommendation G.722.2 and
its Annexes in January 2002.
The AMR-WB codec affords a bandwidth of 50 Hz up to 7 kHz,
in comparison to the conventional three.1 kHz of the conventional telephony
(300-3400 Hz). The codec contains Voice Activity Detection (VAD), Broken
Transmission (DTX), and Comfort Noise Generation (CNG) operations together with
nine modes (bit fees) among 6.60 and 23. Eighty-five kbit/s. The coding scheme
is known as 'Multi-Rate Algebraic Code Excited Linear Prediction.'
The variety of bit-charges lets in the application of the
AMR-WB codec for GSM Full Rate channels, GERAN (Enhanced Data charge for GSM
Evolution, EDGE) 8-Phase Shift Keying (eight-PSK) channels, and 3G UMTSTM
Terrestrial Radio Access Network Wideband Code Division Multiple Access (UTRAN
WCDMA) channels. In GSM, link edition is used to optimize the perceived
transmission best based totally on size reports of the radio channel quality.
AMR-WB is needed in 3GPPTM for Multimedia Messaging Service (MMS), Packet-switched
Streaming Service (PSS), Multi Broadcast Multicast Services (MBMS), and
Packet-Switched Conversational Services, while sixteen kHz sampled speech is
used.
In addition to 3GPPTM wireless packages, further
applications had been focused through ITU-T standardization, together with
Voice over IP (VoIP), Internet programs, Public Switched Telephone Network
(PSTN), and Integrated Services Digital Network (ISDN) wideband telephony, and
audio/video teleconferencing.
Adaptive Multi-Rate (AMR) codec
Between the encoding and the interpreting strategies, which
take location within the transmitting and receiving ends of a verbal exchange
over a digital network, another crucial feature takes region. This is the
'channel coding' procedure described in 3GPPTM (GSM) Technical Specification
45.003: this technique is crucial to shield the encoded speech signal in
opposition to interference inside the radio link. They want to stabilize speech
coding and channel coding to optimize community ability brought about the Adaptive
Multi-Rate (AMR) speech coder, which appeared in GSM Release ninety-eight.
The AMR coder balances the share of available GSM radio
channel bit price (22800 for the whole-rate or 11400 bit/s for the half of-fee)
among the Speech coding and the Channel coding, permitting the handiest use of
the radio resources.
For the version of the uplink codec style, the network has
to estimate the channel pleasant, become aware of the exceptional codec mode
for the present propagation conditions, and ship this information to the Mobile
Station (handset, etc.) over the air interface. For the downlink codec version,
the Mobile Station ought to estimate the downlink channel exceptional and ship
great records to the community. This record is used to define a 'cautioned'
codec mode.
Each link may use a distinct codec mode, but it is mandatory
for both links to apply the identical channel mode (either full-rate or 1/2
price). The channel mode is chosen by using the Radio Resource management
function in the community: it's far achieved at call installation or after a
handover between cells. The channel type can similarly be changed all through a
name as a function of the channel situations.
Enhanced Full Rate (EFR) codec
In the mid-1990s, the qualitative drawbacks proven by way of
the HR codec, together with the advent of greater advanced and effective
digital signal processing technologies, pushed the GSM Association to request
the Speech Experts Group (SEG) of ETSI to provide a new and better sounding
speech coding algorithm, known as Enhanced Full Rate (EFR), running at 12.2
kbit/s and leaving 10.6 kbit/s for the channel coding, which assured bigger
mistakes safety and avoided losing the call in bad interference conditions.
After 30 years, that is nevertheless the nice (narrowband) codec used for
speech communications over cellular phones in 2G and 3G networks!
GSM Half Rate codec
At the crowning glory of the GSM Full Rate exercise, the
Half Rate (HR) speech coding exercising turned into started out, with the goal
of assembly the equal fundamental great of the GSM full price by using 1/2 the
bit fee (the GSM complete charge speech codec requires thirteen kbit/s,
introduced to nine. eight kbit/s to the channel coder, making a complete fee
for the GSM speech channel of 22.8 kbit/s).
The ensuing new set of rules produced the standardized GSM
Half Rate codec that used the most effective 5600 bit/s, leaving 5800 bit/s to
the associated channel coder, making a total fee for the GSM half speech
channel of eleven. Four kbit/s. Unfortunately, the HR codec showed that it could
suffer in terms of perceived pleasant in excessive situations (e.G. With
certain background noises, cellular-to-mobile communications (tandeming), or
positive languages).
GSM Full Rate codec
When GSMTM was first being specified, the mission changed to
prove that the limited to be had spectrum may be exploited greater efficiently
than with the prevailing analog structures. The potential of structures (i.E.
Wide variety of customers the mobile community can help for a given quantity of
certified frequency allocation) will be maximized while keeping or even
enhancing the speech first-class as perceived by using the user. The paintings
ended in a virtual 'complete-price speech' coding set of rules.
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