Telefonaktiebolaget Lm Ericsson (publ)

Erik Ekudden

Roar Hagen

Karl Hellwig

Anders K Eriksson

Ingemar Johansson

Jim Sundqvist

Willem Baastian Kleijn

Stefan Bruhn

Jonas Svedberg

Tomas Frankkila

Fredrik Klas Jansson

Tomas Svensson

H&aring;kan Gunnar Olofsson

Tor Bjorn Minde

Anders Tomas Uvliden

Yngve Kenneth Wallstedt

Bastiaan Kleijn

Johan Nystrom

Bjorn Stig Johansson

Bj&ouml;rn Stig Erik Johansson

Daniel Brighenti

Karin Agneta Bengtsson

The present invention advantageously provides a manner by which to further suppress noise superimposed upon an information signal without increasing distortion to the signal, e.g., speech. By distributing the noise suppression, the quality of the information signal provided to a listener is improved. In one embodiment, a first noise suppressor is employed at the transmitter to suppress noise superimposed upon an information signal prior to its transmission by the transmitter, and a second noise suppressor is employed at the receiver to suppress the noise component of a communication signal received at the receiver.

System and method for distributed noise suppression

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

A system and method is disclosed to remove cross talk terms from speech signals in a multi-party conference system using an estimator to first estimate the respective cross talk terms for each of the said plurality of users. In one embodiment, these terms are then multiplied with a filter based on the estimated cross talk terms. Source speech signals may thereby be recovered from the plurality of multi-party conference users without the cross talk.

System, apparatus, and method for improving speech quality in multi-party devices

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 decoder improves delayed packet concealment in a packet network by using two decoder sections. A first decoder section bases its decoding during the concealment phase on erroneous filter states and a set of speech parameters, whereas a second decoder section bases its decoding on saved and updated filter states and the same speech parameters. The outputs of the two decoder sections are thereafter combined to form the final speech signal. This decoding strategy produces a speech signal with smooth transitions from delayed to non-delayed packets and uses information from the most recent packets for speech generation.

Delayed packet concealment method and apparatus

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

Perceptually relevant non-speech information can be preserved during encoding of an audio signal by determining whether the audio signal includes such information. If so, a speech/noise classification of the audio signal is overriden to prevent misclassification of the audio signal as noise.

Complex signal activity detection for improved speech/noise classification of an audio signal

Method and systems for reporting and analyzing link quality for radiocommunications are described. Use of frame erasure information (e.g., frame erasure rate (FER)) is highlighted. Presentation of frame erasure information can be segregated on a per-connection or per-cell basis, in addition to system-wide output. Moreover, frame erasure information can be correlated with geographical location of a mobile, the mobile's handover status or the equipment type of the mobile station to provide the network operator with a better understanding of quality issues within the network.

Link quality reporting using frame erasure rates

Low rate data signals, such as standard TTY signals and standard DTMF signals, can be effectively analyzed and encoded by speech encoders employed in conventional digital cellular system. This is accomplished by providing a low rate data encoder in the transmission path of the low rate data signal, just prior to the speech encoder used by the digital cellular system. In so doing, the standard, low rate data signal is transformed (i.e., modified) by the low rate data encoder into a signal that is more compatible with the speech encoder. Similarly, a low rate data decoder is provided at the receiving end of the transmission path to decode the modified, low rate data signal.

Low-rate speech coder for non-speech data transmission

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

A source/channel encoding mode control method in a TDMA radio communication system determines the current type of source signal to be encoded and transmitted, restricts encoding to a class of source/channel encoding modes compatible with the determined type of source signal, determines a quality measure for previously transmitted signals that have been received and decoded, and selects based on the quality measure, the most suitable source/channel encoding mode for the determined class.

Source/channel encoding mode control method and apparatus

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

The present invention relates to a method and an arrangement for reconstruction of a received speech signal (r), which has been transmitted over a radio channel that has been subjected to disturbances, such as, e.g., noise, interference or fading. A speech signal (r.sub.rec), where the effects from these disturbances are minimized, is generated by an estimated speech signal (r), corresponding to expected future values of the received speech signal (r), produced according to a linear predictive reconstruction model in a signal modelling circuit. The received speech signal (r) and the estimated speech signal (r) are combined in a signal combination circuit according to a variable ratio, which ratio is determined by a quality parameter (q). The quality parameter (q) may be based on measured power level of a received power level of the desired ratio signal in proportion to an interfering radio signal or a bit error rate signal or bad frame indicator, which has been calculated from data signal that has been transmitted via a certain radio channel and which represents the received speech signal.

Method and arrangement for reconstruction of a received speech signal

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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Åkersberga, Sweden

Erik Ekudden