Abstract: An apparatus for providing improved data compression may include an encoder comprising a quantizer for encoding input data and a side model. The quantizer may be trained with respect to high priority data among the input data and may be configured to partially encode the input data by encoding the high priority data. The side model may be trained jointly with the training of the quantizer and is configured to model low priority data among the input data.

Abstract: An apparatus for providing improved data compression may include an encoder comprising a quantizer for encoding input data and a side model. The quantizer may be trained with respect to high priority data among the input data and may be configured to partially encode the input data by encoding the high priority data. The side model may be trained jointly with the training of the quantizer and is configured to model low priority data among the input data.

Abstract: An apparatus for providing improved data compression may include an encoder comprising a quantizer for encoding input data and a side model. The quantizer may be trained with respect to high priority data among the input data and may be configured to partially encode the input data by encoding the high priority data. The side model may be trained jointly with the training of the quantizer and is configured to model low priority data among the input data.

Abstract: An apparatus for providing improved data compression may include an encoder comprising a quantizer for encoding input data and a side model. The quantizer may be trained with respect to high priority data among the input data and may be configured to partially encode the input data by encoding the high priority data. The side model may be trained jointly with the training of the quantizer and is configured to model low priority data among the input data.

Abstract: An apparatus for providing improved data compression may include an encoder comprising a quantizer for encoding input data and a side model. The quantizer may be trained with respect to high priority data among the input data and may be configured to partially encode the input data by encoding the high priority data. The side model may be trained jointly with the training of the quantizer and is configured to model low priority data among the input data.

Abstract: An apparatus for providing improved data compression may include an encoder comprising a quantizer for encoding input data and a side model. The quantizer may be trained with respect to high priority data among the input data and may be configured to partially encode the input data by encoding the high priority data. The side model may be trained jointly with the training of the quantizer and is configured to model low priority data among the input data.

Abstract: A system and method for modeling speech in such a way that both voiced and unvoiced contributions can co-exist at certain frequencies. In various embodiments, three spectral bands (or bands of up to three different types) are used. In one embodiment, the lowest band or group of bands is completely voiced, the middle band or group of bands contains both voiced and unvoiced contributions, and the highest band or group of bands is completely unvoiced. The embodiments of the present invention may be used for speech coding and other speech processing applications.

Abstract: 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.

Abstract: A method and system are introduced that provide dynamic quantizer structures which are configurable during run time. A quantizer configuration and data are stored in a binary format. The dynamic quantizer data is represented as a bitstream, and the bitstream in turn is used as additional input during initialization (or re-initialization/re-configuration) of a speech coder. A configuration header fully specifies the structure and configuration of the dynamic quantizer for each quantized parameter, and the dynamic quantizer data and configurations are fully and dynamically allocated into the speech coder memory. This enables easy re-configuration of a codec associated with the quantizer structures for different scenarios. The use of dynamic quantizer structures in turn enhances compression efficiency of an input signal. The dynamic quantizer structures can also be applied to other compression applications that allow lossy compression.

Abstract: 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.

Abstract: An apparatus for providing improved data compression may include an encoder comprising a quantizer for encoding input data and a side model. The quantizer may be trained with respect to high priority data among the input data and may be configured to partially encode the input data by encoding the high priority data. The side model may be trained jointly with the training of the quantizer and is configured to model low priority data among the input data.

Abstract: 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.

Abstract: 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.

Abstract: The present invention provides, methods, computer-readable media, and apparatuses for tuning and adjusting the computational complexity of algorithm that is executed by a signal encoder. The signal encoder may comprise a speech encoder. When a resource shortage on a computer platform is detected, a degree of the resource shortage and a corresponding complexity adjustment for a speech encoder are determined. The speech encoder is then tuned to adjust the computational complexity of an executed speech processing algorithm. The resource shortage may correspond to a computational capability, audio buffer memory, or battery of a mobile device. A speech process being executed by the mobile device is tuned to adjust the computational demands in accordance with a complexity adjustment. A number of iteration rounds may be adjusted while the speech encoder is executing a speech processing algorithm. The iterations may correspond to an algebraic codebook search.

Abstract: A system and method for modeling speech in such a way that both voiced and unvoiced contributions can co-exist at certain frequencies. In various embodiments, three spectral bands (or bands of up to three different types) are used. In one embodiment, the lowest band or group of bands is completely voiced, the middle band or group of bands contains both voiced and unvoiced contributions, and the highest band or group of bands is completely unvoiced. The embodiments of the present invention may be used for speech coding and other speech processing applications.

Abstract: A method and system are introduced that provide dynamic quantizer structures which are configurable during run time. A quantizer configuration and data are stored in a binary format. The dynamic quantizer data is represented as a bitstream, and the bitstream in turn is used as additional input during initialization (or re-initialization/re-configuration) of a speech coder. A configuration header fully specifies the structure and configuration of the dynamic quantizer for each quantized parameter, and the dynamic quantizer data and configurations are fully and dynamically allocated into the speech coder memory. This enables easy re-configuration of a codec associated with the quantizer structures for different scenarios. The use of dynamic quantizer structures in turn enhances compression efficiency of an input signal. The dynamic quantizer structures can also be applied to other compression applications that allow lossy compression.

Abstract: The invention relates to a support of a concatenative TTS synthesis. In order to generate a speech database as a basis for the TTS synthesis, first, a speech processing including a segmental parametric speech encoding of speech data based on a parametric modeling of speech is performed, which results in compressed parameterized speech segments. Then, the compressed parameterized speech segments are assembled in a speech database. In order to synthesize output speech, compressed parameterized speech segments are selected from the speech database based on an available text and decompressed to regain parameterized speech segments. The parameterized speech segments are then concatenated in a parameter domain. The output speech is synthesized based on these concatenated parametric speech segments.

Abstract: 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.

Abstract: The present invention provides a new methodology for reusing codebooks for a multistage vector quantization of parameter quantizers of signals. Prior art multistage vector quantization is done in such a way that each stage has different optimized codebooks. The prior art codebooks, thus, use quite a lot of a memory storage space. Using the same codebook stages several times, according to the present invention, reduces the memory usage and a codebook structure maintains good quality by using optimized codebooks for the most important (first) stages in the quantization. The number of codebooks is reduced by reusing the same codebooks in the refining stages. Additionally, according to the present invention, using many predictors is space-wise efficient since they need only a few of coefficients instead of larger codebooks.

Abstract: 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.