Abstract: An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information, includes a domain converter for converting a first domain representation of the audio signal into a second domain representation of the audio signal; and a harmonic post-filter for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function including a numerator and a denominator, wherein the numerator includes a gain value indicated by the gain information, and wherein the denominator includes an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag.

Abstract: Disclosed herein is an automatic beverage can opener. As an example, provided is an automatic beverage can opener comprising: a lever, the lever being automated and movable about a lever pivot axis; a wedge mechanism that is rotatably interconnected to the lever, the wedge mechanism comprising a tab engagement finger; and a can holder for positioning a beverage can in a position where a tab thereof can be engaged by the tab engagement finger upon automated movement of the lever.

Abstract: Embodiments according to the invention are related to methods and apparatuses for spectrotemporally improved spectral gap filling in audio coding using a filtering. Embodiments according to the invention are related to methods and apparatuses for spectrotemporally improved spectral gap filling in audio coding using different noise filling methods. Embodiments according to the invention are related to methods and apparatuses for spectrotemporally improved spectral gap filling in audio coding using a tilt.

Abstract: Embodiments according to the invention are related to methods and apparatuses for spectrotemporally improved spectral gap filling in audio coding using a filtering. Embodiments according to the invention are related to methods and apparatuses for spectrotemporally improved spectral gap filling in audio coding using different noise filling methods. Embodiments according to the invention are related to methods and apparatuses for spectrotemporally improved spectral gap filling in audio coding using a tilt.

Abstract: Embodiments according to the invention are related to methods and apparatuses for spectrotemporally improved spectral gap filling in audio coding using a filtering. Embodiments according to the invention are related to methods and apparatuses for spectrotemporally improved spectral gap filling in audio coding using different noise filling methods. Embodiments according to the invention are related to methods and apparatuses for spectrotemporally improved spectral gap filling in audio coding using a tilt.

Abstract: A method and an apparatus for encoding an audio signal. The apparatus includes a converter converting the audio signal into a spectral representation; a scale parameter calculator calculating scale parameters; a spectral processor processing the spectral representation using the scale parameters; and a scale parameter encoder generating an encoded representation of the scale parameters. The scale parameter calculator is calculates an amplitude-related measure for each band to obtain a set of amplitude-related measures. A pre-emphasis operation is performed to the amplitude-related measures, so that low frequency amplitudes are emphasized with respect to high frequency amplitudes according to a tilt value, or a pre-emphasis factor. The scale parameter calculator controls the tilt value, or the pre-emphasis factor, based on a harmonicity measure of the audio signal.

Abstract: Encoder for encoding a spectral representation of audio signal divided into a plurality of sub-bands, wherein the spectral representation consists of frequency bins or of frequency coefficients and wherein at least one sub-band contains more than one frequency bin, the encoder having: a quantizer configured to generate a quantized representation of the spectral representation of audio signal divided into the plurality sub-bands; a band-wise parametric coder configured to provide a coded parametric representation of the spectral representation depending on the quantized representation, wherein the coded parametric representation consists of parameters describing the spectral representation in the sub-bands or coded versions of the parameters; wherein there are at least two sub-bands being different and parameters describing the spectral representation in the at least two sub-bands being different.

Abstract: A processor for processing an (encoded) audio signal, the processor comprising: an LTP buffer configured to receive samples derived from a frame of the encoded audio signal; an interval splitter configured to divide a time interval associated with a subsequent frame of the encoded audio signal into sub-intervals depending on the encoded pitch parameter; calculation means configured to derive sub-interval parameters from the encoded pitch parameter dependent on a position of the sub-intervals within the time interval associated with the subsequent frame of the encoded audio signal; a predictor configured for generating a prediction signal from the LTP buffer dependent on the sub-interval parameters; and a frequency domain transformer configured for generating a prediction spectrum (XP) based on the prediction signal.

Abstract: An audio encoder for encoding an audio signal comprising an pulse portion and a stationary portion, comprises: a pulse extractor configured for extracting the pulse portion from the audio signal, further comprising a pulse coder for encoding the extracted pulse portion to acquire an encoded pulse portion; wherein the pulse extractor is configured to determine a spectrogram of the audio signal to extract the pulse portion, wherein the spectrogram has higher time resolution than the signal encoder; a signal encoder configured for encoding a residual signal derived from the audio signal to acquire an encoded residual signal, the residual signal being derived from the audio signal so that the pulse portion is reduced or eliminated from the audio signal; and an output interface configured for outputting the encoded pulse portion and the encoded residual signal to provide an encoded signal.

Abstract: An apparatus for encoding a first channel and a second channel of an audio input signal including two or more channels to obtain an encoded audio signal according to an embodiment includes a normalizer configured to determine a normalization value for the audio input signal depending on the first channel of the audio input signal and depending on the second channel of the audio input signal. Moreover, the apparatus includes an encoding unit configured to generate a processed audio signal having a first channel and a second channel. The encoding unit is configured to encode the processed audio signal to obtain the encoded audio signal.

Abstract: An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information, includes a domain converter for converting a first domain representation of the audio signal into a second domain representation of the audio signal; and a harmonic post-filter for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function including a numerator and a denominator, wherein the numerator includes a gain value indicated by the gain information, and wherein the denominator includes an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag.

Abstract: An apparatus for generating an encoded signal includes: a window sequence controller for generating a window sequence information for windowing an audio or image signal, the window sequence information indicating a first window for generating a first frame of spectral values, a second window function and at least one third window function for generating a second frame of spectral values, wherein the first window function, the second window function and the one or more third window functions overlap within a multi-overlap region; a preprocessor for windowing a second block of samples corresponding to the second window function and the at least one third window functions using an auxiliary window function to acquire a second block of windowed samples, a spectrum converter for applying an aliasing-introducing transform; and a processor for processing the first frame and the second frame to acquire encoded frames of the audio or image signal.

Abstract: In an aspect, a toy assembly is provided, and includes a housing, an inner object (which may, in some embodiments, represent a character) inside the housing, a tether, and a motor. The tether is driven by the motor to pull at least one removable housing portion from the housing.

Abstract: An audio quantizer for quantizing a plurality of audio information items has: a first stage vector quantizer for quantizing the plurality of audio information items to determine a first stage vector quantization result and a plurality of intermediate quantized items corresponding to the first stage vector quantization result; a residual item determiner for calculating a plurality of residual items from the plurality of intermediate quantized items and the plurality of audio information items; and a second stage vector quantizer for quantizing the plurality of residual items to obtain a second stage vector quantization result, wherein the first stage vector quantization result and the second stage vector quantization result are a quantized representation of the plurality of audio information items.

Abstract: Audio decoder for decoding an encoded audio signal having multi-channel audio data having data for two or more audio channels, and information on jointly encoded scale parameters, having: a scale parameter decoder for decoding the information on the jointly encoded scale parameters to obtain a first and a second set of scale parameters for a first channel and a second channel, respectively, of a decoded audio signal; and a signal processor for applying the first and second sets of scale parameters to a first and second channel representation, respectively, derived from the multi-channel audio data to obtain the first and second channels of the decoded audio signal, wherein the jointly encoded scale parameters have information on a first group and on a second group of jointly encoded scale parameters, and wherein the scale parameter decoder is configured to combine a jointly encoded scale parameter of the first group and one of the second group using a first and a second combination rule, respectively, to obta

Abstract: An apparatus for estimating an inter-channel time difference between a first channel signal and a second channel signal, includes: a calculator for calculating a cross-correlation spectrum for a time block from the first channel signal in the time block and the second channel signal in the time block; a spectral characteristic estimator for estimating a characteristic of a spectrum of the first channel signal or the second channel signal for the time block; a smoothing filter for smoothing the cross-correlation spectrum over time using the spectral characteristic to obtain a smoothed cross-correlation spectrum; and a processor for processing the smoothed cross-correlation spectrum to obtain the inter-channel time difference.

Abstract: In an aspect, a toy assembly is provided, and includes a housing, an inner object (which may, in some embodiments, represent a character) inside the housing, a tether, and a motor. The tether is driven by the motor to pull at least one removable housing portion from the housing.

Abstract: The invention provides an audio encoder for encoding an audio signal so as to produce therefrom an encoded signal, the audio encoder including: a framing device configured to extract frames from the audio signal; a quantizer configured to map spectral lines of a spectrum signal derived from the frame of the audio signal to quantization indices, wherein the quantizer has a dead-zone, in which the input spectral lines are mapped to quantization index zero; and a control device configured to modify the dead-zone; wherein the control device includes a tonality calculating device configured to calculate at least one tonality indicating value for at least one spectrum line or for at least one group of spectral lines, wherein the control device is configured to modify the dead-zone for the at least one spectrum line or the at least one group of spectrum lines depending on the respective tonality indicating value.

Abstract: An apparatus for decoding an encoded audio signal to obtain a reconstructed audio signal is provided. The apparatus includes a receiving interface, a delay buffer and a sample processor for processing the selected audio signal samples to obtain reconstructed audio signal samples of the reconstructed audio signal. The sample selector is configured to select, if a current frame is received by the receiving interface and if the current frame being received by the receiving interface is not corrupted, the plurality of selected audio signal samples from the audio signal samples being stored in the delay buffer depending on a pitch lag information being included by the current frame.

Abstract: An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information, includes a domain converter for converting a first domain representation of the audio signal into a second domain representation of the audio signal; and a harmonic post-filter for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function including a numerator and a denominator, wherein the numerator includes a gain value indicated by the gain information, and wherein the denominator includes an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag.