Speech and audio compression has advanced rapidly in recent years spurred on by cost-effective digital technology and diverse commercial applications. Recent activity in speech compression is dominated by research and development of a family of techniques commonly described as code-excited linear prediction (CELP) coding. These algorithms exploit models of speech production and auditory perception and offer a quality versus bit rate tradeoff that significantly exceeds most prior compression techniques for rates in the range of 4 to 16 kb/s. Techniques have also been emerging in recent years that offer enhanced quality in the neighborhood of 2.4 kb/s over traditional vocoder methods. Wideband audio compression is generally aimed at a quality that is nearly indistinguishable from consumer compact-disc audio. Subband and transform coding methods combined with sophisticated perceptual coding techniques dominate in this arena with nearly transparent quality achieved at bit rates in the neighborhood of 128 kb/s per channel. >

Advances in speech and audio compression

A distributed antenna system comprises a base transceiver station, a plurality of distributed antenna units and a signal routing apparatus. The base transceiver station has a plurality of output ports that generate a plurality of downlink signals having overlapping transmit frequencies and containing different communication content. The different communication content is directed toward each of a plurality of mobile units. The base transceiver station also has at least one uplink receive port that receives an uplink signal. The uplink signal includes communication content received from at least one of the mobile units. The plurality of distributed antenna units have coverage areas that are non-overlapping or only partially overlapping. The signal routing apparatus routes the downlink signals to the plurality of distributed antennas such that each of the plurality of downlink signals are transmitted by one or more of the distributed antennas and each distributed antenna only transmits one of the downlink signals.

Distributed antenna communications system

A distributed antenna system comprises a plurality of antennas and a multi-port hub. The multi-port hub comprises an interface to a telecommunications network and a plurality of transceivers. The multi-port hub is configured to operate in a first mode ('normal' mode) in which the multi-port hub receives a downlink communications signal via the interface and distributes the downlink communications signal to the plurality of antennas using a selected downlink transmission frequency within a downlink frequency range and in which the multi-port hub receives uplink communications signals from the plurality of antennas at a selected uplink receive frequency. The multi-port hub is also configured to operate in a second mode ('listening' mode) in which the multi-port hub receives communications signals from the plurality of antennas at one or more frequencies within the downlink frequency range. The transmission frequency for the first mode ('normal' mode) may be selected based on field strength of the signals received in the second mode ('listening' mode).

System for and method of configuring distributed antenna communications system

Provided are an audio encoding method and apparatus and an audio decoding method and apparatus in which audio signals can be encoded or decoded so that sound images can be localized at any desired position for each object audio signal. The audio decoding method generating a third downmix signal by combining a first downmix signal extracted from a first audio signal and a second downmix signal extracted from a second audio signal; generating third object-based side information by combining first object-based side information extracted from the first audio signal and second object-based side information extracted from the second audio signal; converting the third object-based side information into channel-based side information; and generating a multi-channel audio signal using the third downmix signal and the channel-based side information.

Methods and apparatuses for encoding and decoding object-based audio signals

A digital radio frequency transport system that performs bi-directional simultaneous digital radio frequency distribution is provided. The transport system includes a digital host unit and at least two digital remote units coupled to the digital host unit. The bi-directional simultaneous digital radio frequency distribution is performed between the digital host unit and the at least two digital remote units.

Point-to-multipoint digital radio frequency transport

Audio sources are coded by recognizing different classes of audio such as speech and music. The classes are used to select between coding algorithms and to provide object definitions. Objects have abstract and concrete classes which may further rely on parameters produced by linear prediction and subband filters to provide a frame-based bit stream of information. Each object in the bit stream has layers of information such as basic bit rate, coding parameters and enhancement parameters. The layers of information in each object allow altering selected parameters to manipulate audio signals.

Object oriented audio coding

The method and the apparatus allow coding, manipulation and decoding of audio signals independently of the specific signal content. An encoder (AC) carries out an object-oriented coding and generates a coded signal, with variable bit rate and bandwidth, comprising a basic layer and one or more enhancement layers for each object. The encoder (AC) is able to operate according both to ad hoc algorithms and to any standardised algorithm and selects the most convenient algorithm depending on the object to be coded. Bit stream manipulation units along a transmission path towards a decoder (AD) allow scaling the bit rate or the bandwidth of the coded signal or altering parameters of the coded signal by acting on individual objects.

Method of and apparatus for coding, manipulating and decoding audio signals

The objective of the described work is to constrain the algorithmic delay of an analysis-by-synthesis coder to be 2.5 ms while maintaining the bit rate at approximately 13 kb/s, which has been shown to be adequate for the Pan European Digital Mobile Radio System. For applications which do not incorporate some redundancy for error-protection purposes, the bit rate of the speech coder increased to 16 kb/s, while the delay was reduced to as little as 2 ms. >

A 2 ms delay CELP coder

CELP coding at variable rate : Speech coding for telecommunications

A variable rate speech codec with seven operating rates ranging from 400 bit/s to 16 kbit/s and a 10 ms algorithmic delay is presented in this paper. The current rate is chosen according to an open-loop speech classification followed by a closed-loop quality evaluation. The rate selection can be source-controlled or network-controlled. The codec is based on a CUP algorithm with a variable number of excitations. The innovations have a deterministic structure allowing a very efficient search procedure and are orthogonalized to the long-term contribution. The algorithm will be implemented in the testbed of one RACE II project: CODIT (COde DIvision Testbed) [1].

Daniele Sereno

Papers

Object oriented audio coding

Method of and apparatus for coding, manipulating and decoding audio signals

A 2 ms delay CELP coder

CELP coding at variable rate : Speech coding for telecommunications

Variable rate speech coding for umts