Nokia Corporation

Nokia Mobile Phones Limited

Nokia Telecommunications Oy

Nokia Networks Oy

Nokia Technologies Oy

Janne Vainio

Hannu J Mikkola

Jani Rotola-Pukkila

Kari Juhani Jarvinen

Pasi Sakari Ojala

Jari Mäkinen

Matti Kajala

Petri J Haavisto

Jari M Makinen

Pekka Kapanen

Ari Lakaniemi

Hannu Korhonen

Sakari Himanen

Tuomas Vaittinen

Janne Petteri Bergman

Tero Honkanen

Juha Marila

Arto Saikanm&auml;ki

Toni P Nieminen

Toni Nieminen

A system and method for locating a preferable playback start location after a winding or rewinding action in an audio playing device. In response to an adjustment of the playing location for audio content to a desired playing position, the system determines whether at least one non-speech or silent period of at least a predetermined duration exists within the vicinity of the desired playing position. If at least one such non-speech or silent period exists within the vicinity of the desired playing position, the system adjusts the playing position to fall within one of the at least one non-speech period or silent period.

System and method for winding audio content using a voice activity detection algorithm

A method including causing, at least in part, display of selectable objects on a graphical user interface, where each of the selectable objects corresponds to data or an application accessible via the graphical user interface. The method further includes causing, at least in part, display of the selectable objects in motion travelling across the graphical user interface based on a category of the selectable object or context dependent data, and allowing user selection and manipulation of the selectable objects displayed on the graphical user interface.

Method and apparatus for fluid graphical user interface

An encoder comprising an input for inputting frames of an audio signal in a frequency band, at least a first excitation block for performing a first excitation for a speech like audio signal, and a second excitation block for performing a second excitation for a non-speech like audio signal. The encoder further comprises a filter for dividing the frequency band into a plurality of sub bands each having a narrower bandwidth than the frequency band. The encoder also comprises an excitation selection block for selecting one excitation block among the at least first excitation block and the second excitation block for performing the excitation for a frame of the audio signal on the basis of the properties of the audio signal at least at one of the sub bands. The invention also relates to a device, a system, a method and a storage medium for a computer program.

Classification of audio signals

A method and corresponding codec for (channel) encoding speech or other data bits for transmission via a wireless communication channel, the method providing unequal error protection (UEP) using only a single encoder, and including: a step of determining how many bits to puncture in each of typically two protection classes (CC) so as to achieve either a predetermined or iterated desired level of error protection; and a step of identifying which bits to puncture for each class so as to provide relatively strong and uniform protection for all bits in the first class (C), but protection that decreases in the same manner as the subjective importance decreases from the beginning to the end of the other classes. The method also accounts for so-called soft puncturing by modulators transmitting multiple bits per symbol with weaker protection for some of the bits of each symbol.

Method and apparatus for puncturing with unequal error protection in a wireless communication system

A method of encoding speech in a communications system includes the steps of receiving a speech signal including voice signals and background signals, and detecting voice activity and providing an indicator when no voice activity is detected. The speech signal is encoded to generate a plurality of parameters representing the signal. When the indicator is not present, a first parametric representation of the speech signal is output, including the plurality of parameters. When the indicator is present, at least one of the plurality of parameters is modified and a second parametric representation of the speech signal, including the modified parameter is output.

Method and apparatus for encoding speech

A method for use by a speech decoder in handling bad frames received over a communications channel a method in which the effects of bad frames are concealed by replacing the values of the spectral parameters of the bad frames (a bad frame being either a corrupted frame or a lost frame) with values based on an at least partly adaptive mean of recently received good frames, but in case of a corrupted frame (as opposed to a lost frame), using the bad frame itself if the bad frame meets a predetermined criterion. The aim of concealment is to find the most suitable parameters for the bad frame so that subjective quality of the synthesized speech is as high as possible.

Spectral parameter substitution for the frame error concealment in a speech decoder

According to an aspect of the invention, an enhanced audible feedback solution has been invented for electronic devices using an input device facilitating navigation though a plurality of available user interface input options and confirmation of a selected input option. The electronic device is arranged to define, as a response to detecting a selection of a character on the basis of a detection of a first input to an input device of the electronic device, an audio segment specific to the character. The electronic device is arranged to output the defined audio segment via the audio output means prior to a confirmation by a second input to the input device, the second input being associated with a function adding the character as part of a character sequence entered by the user.

Method for inputting characters in electronic device

A speech decoder comprises a decoder () for converting a linear prediction encoded speech signal into a first sample stream having a first sampling rate and representing a first frequency band. Additionally it comprises a vocoder () for converting an input signal into a second sample stream having a second sampling rate and representing a second frequency band, and combination means () for combining the first and second sample streams in processed form. It comprises also means () for generating a second linear prediction filter, to be used by the vocoder () on the second frequency band, on the basis of a first linear prediction filter used by the decoder () on the first frequency band. Extrapolation through an infinite impulse response filter is the preferable method of generating the second linear prediction filter.

Speech decoder and a method for decoding speech

Methods and arrangements are disclosed for digitally encoding and decoding sound&#x2014;An input signal is split () into a primary frequency band and at least one secondary frequency band. The parts of the input signal in the frequency bands are separately encoded. Certain characteristics of the input signal in the primary frequency band and corresponding characteristics of the input signal in at least one secondary frequency band are examined () in order to find out, whether there is certain resemblance therebetween. Alternatively certain characteristic features of the process applied to encoding the primary frequency band extracted () and used () in encoding the secondary frequency band, or such extracted characteristic features are replaced () with a locally generated, independent set of corresponding features.

Excitation for higher band coding in a codec utilising band split coding methods

A wireless telecommunications system comprises a mobile station (MS) and a network. The mobile station has a multi-rate speech encoder which produces an encoded speech signal which is transmitted to the network. The network has a multi-rate speech decoder which decodes the encoded speech signal to produce a decoded speech signal. The network also comprises a signal analyser which measures speech characteristics of the decoded speech signal to produce speech characteristics information and an up-link mode control unit which receives the speech characteristics information and produces a mode command. The mode command is transmitted by the network to the mobile station where it is used to control the speech encoding bit rate of the multi-rate speech encoder.

Method and apparatus providing adaptive multi-rate speech coding

A method and arrangements are provided for transmitting frames of digital information over a wireless communication connection between a transmitter and a receiver. In the transmitter, a certain sequence of bits within each frame of digital information is convolutionally encoded and punctured before transmitting the frame over a wireless communication connection. The receiver decodes and depunctures the sequence of bits within each frame of digital information, that was convolutionally encoded and punctured, after receiving the frame over a wireless communication connection. The transmitter rearranges the sequence of bits within each frame of digital information that is to be convolutionally encoded and punctured, before convolutionally encoding and puncturing it. The rearranged order is one that has been found to produce, during the course of convolutionally encoding with a certain convolutional code and puncturing with a certain puncturing pattern, a convolutionally encoded and punctured sequence where the statistical probability of transmission errors exhibits a predefined behavior. In the receiver, the sequence of bits within each frame of digital information that was so rearranged in the transmitter is inversely rearranged so that the effect of said rearranging in the transmitter on the mutual order of the bits of the sequence is cancelled, after decoding and depuncturing the sequence of bits.

Method and arrangement for providing optimal bit protection against transmission errors

An apparatus and method for detecting and controlling audible errors in a sound communication system at the receiver utilizes channel quality data and also iterative synthesis. Errors occurring in synthesized speech are detected by searching for atypical sound with a stringency dependent upon channel quality. The greater the channel quality deficiency is, the higher the typicality standards will be. Errors are controlled by either re-synthesizing the signal in an iterative way using typicality standards which vary with channel quality deficiency, or by modifying the output signal using typicality standards which vary with channel quality deficiency, or both.

Audible error detector and controller utilizing channel quality data and iterative synthesis

A method and corresponding apparatus for encoding a sequence of bits for transmission as symbols, some of the bit positions of the symbols having a higher bit error rate than other bit positions. A plurality of sequences of bits is provided using a convolutional encoder, in response to a sequence of input bits, each sequence of bits being defined by a predetermined generator polynomial having a predetermined level of sensitivity to puncturing. Then the bits of each sequence of bits are mapped to symbol positions based on the level of sensitivity of the generator polynomial defining the sequence of bits. With interleaving, the mapping of bits of each sequence of bits to symbol positions can precede a symbol interleaving step, or it can follow a bit interleaving step.

Method and system for allocating convolutional encoded bits into symbols before modulation for wireless communication

A method and system for concealing errors in one or more bad frames in a speech sequence as part of an encoded bit stream received in a decoder. When the speech sequence is voiced, the LTP-parameters in the bad frames are replaced by the corresponding parameters in the last frame. When the speech sequence is unvoiced, the LTP-parameters in the bad frames are replaced by values calculated based on the LTP history along with an adaptively-limited random term.

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Lempäälä, Finland

Janne Vainio