Telefonaktiebolaget Lm Ericsson (publ)

Anders Tomas Uvliden

Jonas Svedberg

Ingemar Johansson

Per Anders Karlsson

Tor Bjorn Minde

Magnus Westerlund

Jim Sundqvist

Erik Ekudden

Anders Nohlgren

Tor Björn Minde

Tor Bj&ouml;rn Minde

Arne Steinarson

Per Gunnar Heikkil&auml;

Per Gunnar Heikkilang

Gunnar Heikkila

Per Gunnar Heikkila

Ingemar Johannson

Anders Karlsson

A multi-part fixed codebook includes both individual fixed codebooks for each channel and a shared fixed codebook. Although the shared fixed codebook is common to all channels, the channels are associated with individual lags. Furthermore, the individual fixed codebooks are associated with individual gains, and the individual lags are also associated with individual gains. The excitation from each individual fixed codebook is added to the corresponding excitation (a shared codebook vector, but individual lags and gains for each channel) from the shared fixed codebook.

Multi-channel signal encoding and decoding

A multi-codebook fixed bitrate CELP signal block encoder/decoder includes a codebook selector () for selecting, for each signal block, a corresponding codebook identification in accordance with a deterministic selection procedure that is independent of signal type. Included are also means for encoding/decoding each signal block by using a codebook having the selected codebook identification.

CELP encoding/decoding method and apparatus

An improved forward error correction (FEC) technique for coding speech data provides an encoder module which primary-encodes an input speech signal using a primary synthesis model to produce primary-encoded data, and redundant-encodes the input speech signal using a redundant synthesis model to produce redundant-encoded data. A packetizer combines the primary-encoded data and the redundant-encoded data into a series of packets and transmits the packets over a packet-based network, such as an Internet Protocol (IP) network. A decoding module primary-decodes the packets using the primary synthesis model, and redundant-decodes the packets using the redundant synthesis model. The technique provides interaction between the primary synthesis model and the redundant synthesis model during and after decoding to improve the quality of a synthesized output speech signal. Such &ldquo;interaction,&rdquo; for instance, may take the form of updating states in one model using the other model.

Forward error correction in speech coding

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

In producing an approximation of an original speech signal from encoded information about the original speech signal, current parameters associated with a current segment of the original speech signal are determined from the encoded information. Reproduction of a noise component of the original speech signal is improved by using at least one of the current parameters and corresponding previous parameters respectively associated with previous segments of the original speech signal to produce a modified parameter. The modified parameter is then used to produce an approximation of the current segment of the original speech signal.

Speech coding with improved background noise reproduction

An improved method and system of measuring the perceived speech quality in mobile telecommunications networks is disclosed herein. In an embodiment of the invention, the method uses both radio link parameters and an objective measuring technique performed on received signals to estimate the speech quality perceived by the end-user. A radio link processing stage extracts temporal information from a set of available radio link parameters such as the BER, FER, RxLev, handover statistics, soft information, and speech energy. Concurrently, a speech processing stage is used to process a sequence of original signals and received signals, obtained from the output of a telecommunications system. The signal sequences are processed by an objective measuring technique such as Perceptual Speech Quality Measure (PSQM). The outputs from the radio link processing and speech processing stages are utilized to calculate an estimate for speech quality. Furthermore, a weight may be given to radio link processing and speech processing in accordance with their performance under various conditions such that the overall speech quality is calculated with respect to the best approach.

Speech quality measurement based on radio link parameters and objective measurement of received speech signals

A method and system for measuring the speech quality in a mobile cellular telecommunications network using available radio link parameters is disclosed herein. In a preferred embodiment, the method includes receiving a set of radio link parameters, as defined in a standard or otherwise available, such as the BER, FER, RxLev, handover statistics, soft information, and speech energy. Temporal information is obtained from the radio link parameters to create a set of temporal parameters which can be statistically analyzed, for example, for the maximum and minimum, mean, standard deviation, and autocorrelation values for a time interval. The temporal parameters are combined to yield a set of correlated parameters that are more closely related to the speech quality. An estimator then uses the correlated parameters to calculate an estimate for the speech quality. The method of the present invention takes advantage of temporal information and correlated relationships from the transmitted parameters. Furthermore, the method is inherently simple and reliable as compared to prior art methods. Still further, the technique provides a method that allows operators to monitor quality conditions throughout the network.

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Luleå, Sweden

Anders Tomas Uvliden