Telefonaktiebolaget Lm Ericsson (publ)

Google Inc.

Global IP Sound Ab

Global IP Solutions, Inc.

Global IP Solutions (gips) Ab

Global IP Solutions Ab

Roar Hagen

Erik Ekudden

Bastiaan Kleijn

Soren Vang Andersen

Niklas Enbom

W Bastiaan Kleijn

Tina Abrahamsson

Willem Baastian Kleijn

Jan K Skoglund

Ingemar Johansson

Tomas Svensson

Soren Andersen

Johan Nystr&ouml;m

Tor Bj&ouml;rn Minde

Bj&ouml;rn Stig Erik Johansson

Bjorn Stig Johansson

The present invention relates to improvements of predictive encoding/decoding operations performed on a signal which is transmitted over a packet switched network. The signal is encoded on a block by block basis in such way that a block A-B is predictive encoded independently of any preceding blocks. A start statelocated somewhere between the end boundaries A and B of the block is encoded using any applicable coding method. Both block parts surrounding the start state is then predictive encoded based on the start state and in opposite directions with respect to each other, thereby resulting in a full encoded representationof the block A-B. At the decoding end, corresponding decoding operations are performed.

Low bit rate codec

A method and an apparatus for indicating presence of a transient noise in a call are provided. The method comprises the steps of determining activity at an endpoint of the call by monitoring presence of a signal input from the endpoint into the call and monitoring presence of a potential source of transient noise at the endpoint. Further, based on the activity determination and the monitoring of the presence of a potential source of transient noise, a signal representative of the presence of a transient noise in the call is sent. The present invention is advantageous in that it enables improvement of the quality of the call.

Method of indicating presence of transient noise in a call and apparatus thereof

The invention relates to a method and an apparatus for mixing audio signals. The invention is based on the idea to base selection of what audio channels to mix on the dynamic behavior of the audio channels in terms of whether they are active or not, rather than basing the selection on quantative measures of the audio channels. According to the invention, a set of audio channels are arranged in a mixing stack and a mixed audio signal is produced from the topmost channels above a predetermined threshold level in the stack. Whenever a channel becomes active, it is placed at the top of the stack.

Audio mixing

The present invention relates to improvements of predictive encoding/decoding operations performed on a signal which is transmitted over a packet switched network. The signal is encoded on a block by block basis in such way that a block A-B is predictive encoded independently of any preceding blocks. A start state () located somewhere between the end boundaries A and B of the block is encoded using any applicable coding method. Both block parts surrounding the start state is then predictive encoded based on the start state and in opposite directions with respect to each other, thereby resulting in a full encoded representation () of the block A-B. At the decoding end, corresponding decoding operations are performed.

The quality of comfort noise generated by a speech decoder during non-speech periods is improved by modifying comfort noise parameter values normally used to generate the comfort noise. The comfort noise parameter values are modified in response to variability information associated with a background noise parameter. The modified comfort noise parameter values are then used to generate the comfort noise.

Speech coding with comfort noise variability feature for increased fidelity

The invention relates to methods for encoding/decoding a digitized sound signal which is transmitted over a packet switched network in such a way that diversity is provided. Each sound segment of the digitized sound signal is encoded into at least two different segment descriptions, i.e. into different representations of the sound signal segment. The segment descriptions differ from each other in that they provide different quantization values for a digital sound signal sample. The different segment descriptions are transmitted in separate data packets over the network at different points of time.

Transmission over packet switched networks

The invention relates to methods for encoding/decoding of a digital signal which is transmitted over a packet switched network. Prediction samples are generated at the transmitting and receiving end. The digital signal is lossless encoded at the transmitting end, and lossless decoded at the receiving end, based on the quantizations of generated prediction samples. During encoding, the generated prediction samples are quantized separately from the quantization of the digital samples. The predictions are used in the index domain in the form of quantized indices during encoding/decoding of the digital signal.

Encoding and decoding of a digital signal

A linear predictive analysis-by-synthesis encoder includes a search algorithm block ( ) and a vector quantizer ( ) for vector quantizing optimal gains from a plurality of subframes in a frame. The internal encoder states are updated ( ) using the vector quantized gains.

Linear predictive analysis-by-synthesis encoding method and encoder

A speech encoding/decoding apparatus. A speech encoding apparatus has a coding portion for receiving input information related to an uncoded signal representative of an original speech signal, the coding portion including a fixed coding portion for receiving the input information and producing a first coded signal estimate, and an adaptive coding portion for receiving the input information and producing a second coded signal estimate. A controller is connected to the fixed coding portion and the adaptive coding portion for receiving information indicative of speech characteristics of the uncoded signal and generates a control signal; and a code modifier receives the first coded signal estimate from the fixed coding portion and the control signal from the controller and produces a modified signal estimate.

Speech coding including soft adaptability feature

A channel optimized vector quantization apparatus includes a device for weighting a sample vector x by a weighting matrix A and a device for weighting a set of code book vectors &ccirc;,by a weighting matrix B. Device form a set of distance measures &lcub;d,(Ax,B&ccirc;,)&rcub; representing the distance between the weighted sample vector Ax and each weighted code book vector B&ccirc;,. Other device form a set of distortion measures &lcub;,(x)&rcub; by multiplying each distance measure by a channel transition probability P,that an index r has been received at a decoder when an index i has been sent from an encoder and adding together these multiplied distance measures for each possible index r. Finally device determine an index i,corresponding to the smallest distortion measure,(x) and represents the sample vector by this index i,.

Distance and distortion estimation method and apparatus in channel optimized vector quantization

An apparatus and method for reducing sparseness in a coded speech signal. Sparse codebook values are generated from a codebook. An anti-sparseness operation is performed on the sparse codebook values to produce output codebook values having a greater density of non-zero values than the sparse codebook values. The output codebook values are processed by a speech processor to generate an encoded speech signal during an encoding operation or a decoded speech signal during a decoding operation.

Reducing sparseness in coded speech signals

In producing from an original speech signal a plurality of parameters from which an approximation of the original speech signal can be reconstructed, a coded signal of the original speech signal is generated. At least one of the parameters is determined using first and second differences between the original speech signal and the coded signal. The first difference is a difference between a waveform associated with the original speech signal and a waveform associated with the coded signal, and the second difference is a difference between an energy parameter derived from the original speech signal and a corresponding energy parameter associated with the coded signal.

Adaptive combining of multi-mode coding for voiced speech and noise-like signals

At a transmitter of a communication system, a target signal and a primary coded signal are produced in response to an input signal. The primary coded signal is intended to match the target signal. Also produced is encoded enhancement information indicative of how closely the primary coded signal matches the target signal. At a receiver, the primary coded signal is reconstructed, the encoded enhancement information is decoded, and an enhanced reconstructed signal is produced by applying the decoded enhancement information to the reconstructed primary coded signal.

Enhanced coding to improve coded communication signals

A post-processing method for a speech decoder which outputs a decoded speech signal in the time domain provides high frequency resolution based on a frequency spectrum having non-harmonic and noise deficiencies. This is obtained by transforming the decoded time domain signal to a frequency domain signal by using a high frequency resolution transform (FFT). Then an analysis of the energy distribution of the frequency domain signal is made throughout its frequency area (4 kHz) to find the disturbing frequency components and to prioritize such frequency components which are situated in the higher part of the frequency spectrum. Next, the suppression degree for the disturbing frequency components is found based on prioritizing. Finally the steps of controlling a post-filtering of the transform in dependence of the finding, and inverse transforming the post-filtered transform in order to obtain a post-filtered decoded speech signal in the time domain are performed.

High resolution post processing method for a speech decoder

Adaptive speech coding includes receiving an original speech signal, performing on the original speech signal a current coding operation, and adapting the current coding operation in response to information used in the current coding operation. Adaptive speech decoding includes receiving coded information, performing a current decoding operation on the coded information, and adapting the current decoding operation in response to information used in the current decoding operation.

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Stockholm, Sweden

Roar Hagen