Nokia Corporation

Nokia Mobile Phones Limited

Ari Heikkinen

Anssi Sakari Rämö

Ari Lakaniemi

Jani Kristian Nurminen

Sakari Himanen

Vesa T Ruoppila

Pekka Mellanen

Alejandro Monge-Navarro

Mikko Tapio Tammi

Samuli Pietil&auml;

Anssi RÄMÖ

A method and device for improving coding efficiency in audio coding. From the pitch values of a pitch contour of an audio signal, a plurality of simplified pitch contour segments are generated to approximate the pitch contour, based on one or more pre-selected criteria. The contour segments can be linear or non-linear with each contour segment represented by a first end point and a second end point. If the contour segments are linear, then only the information regarding the end points, instead of the pitch values, are provided to a decoder for reconstructing the audio signal. The contour segment can have a fixed maximum length or a variable length, but the deviation between a contour segment and the pitch values in that segment is limited by a maximum value.

Method and system for pitch contour quantization in audio coding

A device is disclosed that makes packetized and encoded speech data audible to a listener, as is a method for operating the device. The device includes a unit for generating a synchronization request for reducing an amount of synchronization delay, and further includes a speech decoder that is responsive to the synchronization delay adjustment request for executing a time-warping operation for one of lengthening or shortening a duration of a speech frame. In one embodiment the speech decoder comprises a code excited linear prediction (CELP) speech decoder, and the CELP decoder time-warping operation is applied to a reconstructed excitation signal u(k) to derive a time-warped reconstructed signal u(k). The time-warped reconstructed signal u(k) is input to a Linear Predictor (LP) synthesis filter to derive a CELP decoder time-warped output signal y^(k) In another embodiment the speech decoder comprises a parametric speech decoder, and where an adaptation of the frame length N in the parametric speech decoder results in the use of a modified frame length N.

Decoder synchronization adjustment

The invention relates to a method for use in parametric speech coding. In order to enable an improved parametric coding of speech signals, the method comprises a first step of pre-processing a to be encoded speech based signal such that a phase structure of the to be encoded speech based signal is approached to a phase structure which is obtained when the to be encoded speech based signal is parametrically encoded and decoded again. Only in a second step, a parametric encoding is applied to this pre-processed to be encoded speech based signal. The invention relates equally to a corresponding device, to a corresponding coding module, to a corresponding system and to a corresponding software program product.

Speech coding method, device, coding module, system and software program product for pre-processing the phase structure of a to be encoded speech signal to match the phase structure of the decoded signal

A device is disclosed that makes packetized and encoded speech data audible to a listener, as is a method for operating the device. The device includes a unit for generating a synchronization request for reducing an amount of synchronization delay, and further includes a speech decoder that is responsive to the synchronization delay adjustment request for executing a time-warping operation for one of lengthening or shortening a duration of a speech frame. In one embodiment the speech decoder comprises a code excited linear prediction (CELP) speech decoder, and the CELP decoder time-warping operation is applied to a reconstructed excitation signal u(k) to derive a time-warped reconstructed signal u(k). The time-warped reconstructed signal u(k) is input to a Linear Predictor (LP) synthesis filter to derive a CELP decoder time-warped output signal y(k). In another embodiment the speech decoder comprises a parametric speech decoder, and where an adaptation of the frame length N in the parametric speech decoder results in the use of a modified frame length N.

Method and apparatus for reducing synchronization delay in packet switched voice terminals using speech decoder modification

The present invention discloses a method of improving the coded speech quality in low bit rate analysis-by-synthesis (AbS) speech coders. This is accomplished by relaxing the waveform matching constraints for non-stationary plosive speech segments of speech signals by suitably shifting pulse locations of the coded excitation signal. The shifting results in the coded signal having phase information that does not exactly match original signal in places where it is perceptually insignificant to the listener. Furthermore, a technique for adaptive phase dispersion is introduced to the coded excitation signal to efficiently preserve important signal characteristics such as the energy spread of the original signal.

Method and device for coding speech in analysis-by-synthesis speech coders

This invention presents a voicing determination algorithm for classification of a speech signal segment as voiced or unvoiced. The algorithm is based on a normalized autocorrelation where the length of the window is proportional to the pitch period. The speech segment to be classified is further divided into a number of sub-segments, and the normalized autocorrelation is calculated for each sub-segment if a certain number of the normalized autocorrelation values is above a predetermined threshold, the speech segment is classified as voiced. To improve the performance of the voicing determination algorithm in unvoiced to voiced transients, the normalized autocorrelations of the last sub-segments are emphasized. The performance of the voicing decision algorithm can be enhanced by utilizing also the possible lookahead information.

Method and apparatus for speech coding with voiced/unvoiced determination

A method and system for waveform interpolation speech coding. The method comprises the steps of decomposing the speech signal into a slowly evolving waveform component and a rapidly evolving waveform component in the encoder and determining the power ratio of these surface components so that the power ratio can be used to determine the bit allocation when the surface components are quantized. The power ratio can also be used to modify the phases of the slowly evolving waveform component when the surface components are reconstructed in the decoder in order to improve the speech quality.

Speech coding exploiting the power ratio of different speech signal components

A method and apparatus for coding successive pitch periods of a speech signal. Based on a priori knowledge of statistical properties of successive speech periods, a shaped lattice structure is designed to cover the most probable points in the pitch space. The codebook index search starts with finding an open-loop estimate in the pitch space considering all dimensions and refining the open-loop estimate in a closed-loop search separately in each dimension based on the shaped lattice structure. The closed-loop search for the first subframe is for obtaining an absolute pitch period or a delta pitch while the closed-loop search for each of the other subframes is for obtaining a delta pitch for the respective subframe.

Method and apparatus for coding successive pitch periods in speech signal

A receiving device, the signal received by which has at least one digital data channel, demodulates ( ) and decodes ( ) the digital data included in the data channel and measures ( ) its quality as well on the frequency to be used and at least on one further frequency. The parameter describing the bit error ratio and/or the quantity of correctly received code words will be delivered to a controller ( ) that makes the decision whether the tuning frequency of the radio receiving ( ) part included in the device should be changed.

Method for improving the quality of reception in a radio receiver and a radio receiver

In order to tune a radio receiver (10) to an optimal tuning frequency it is found the critical frequency closest to the current tuning frequency (60) at which the radio receiver loses the locking to the clock frequency of the received signal, and, if necessary (62), the tuning frequency is corrected away from the critical frequency found. The control block (13) of the radio receiver comprises means for adjusting the tuning of the reception part (11) in response to the information (17) indicating whether the locking to the clock frequency is on. The search for the critical frequency may be performed in on or two stages and alternately on both sides of the current tuning frequency or first on one side and then on the other.

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Tampere, Finland

Ari Heikkinen