Abstract: The invention relates to audio signal processing apparatuses and methods, such as an audio signal downmixing apparatus (105) for processing an input audio signal comprising a plurality of input channels (113) into an output audio signal comprising a plurality of primary output channels (123) and at least one auxiliary output channel (125) using a downmix matrix D, wherein the downmix matrix D comprises a primary downmix matrix DU providing the plurality of primary output channels (123) and an auxiliary downmix matrix DW providing the at least one auxiliary output channel (125). The audio signal downmixing apparatus (105) comprises an auxiliary downmix matrix determiner (107) configured to determine the auxiliary downmix matrix DW, and a processor (109) configured to process the input audio signal into the output audio signal using the downmix matrix D.

Abstract: An audio signal encoding method is provided. The method comprises: collecting audio signal samples, determining sinusoidal components in subsequent frames, estimation of amplitudes and frequencies of the components for each frame, merging thus obtained pairs into sinusoidal trajectories, splitting particular trajectories into segments, transforming particular trajectories to the frequency domain by means of a digital transform performed on segments longer than the frame duration, quantization and selection of transform coefficients in the segments, entropy encoding, outputting the quantized coefficients as output data, wherein segments of different trajectories starting within a particular time are grouped into Groups of Segments (GOS), and the partitioning of trajectories into segments is synchronized with the endpoints of a Group of Segments).

Abstract: The invention relates to audio signal processing apparatuses and methods, such as an audio signal downmixing apparatus (105) for processing an input audio signal comprising a plurality of input channels (113) into an output audio signal comprising a plurality of primary output channels (123) and at least one auxiliary output channel (125) using a downmix matrix D, wherein the downmix matrix D comprises a primary downmix matrix DU providing the plurality of primary output channels (123) and an auxiliary downmix matrix DW providing the at least one auxiliary output channel (125). The audio signal downmixing apparatus (105) comprises an auxiliary downmix matrix determiner (107) configured to determine the auxiliary downmix matrix DW, and a processor (109) configured to process the input audio signal into the output audio signal using the downmix matrix D.

Abstract: An apparatus for processing soundfield data is provided. The soundfield data defines a soundfield within a spatial reproduction region comprising at least one bright zone and at least one quiet zone. The apparatus comprises an applicator configured to apply a spatially continuously varying weighting function to the soundfield data in order to obtain weighted soundfield data defining a weighted soundfield, wherein the spatially continuously varying weighting function is configured to enhance the soundfield in at least one of the bright zone and the quiet zone.

Abstract: An input audio signal comprises a plurality of input audio channels. A KLT-based pre-processor transforms the plurality of input audio channels into a plurality of eigenchannels and provides metadata associated with the plurality of eigenchannels. Each eigenchannel is associated with an eigenvalue and an eigenvector. The metadata allows reconstructing the plurality of input audio channels on the basis of the plurality of eigenchannels. A selector selects a subset of the plurality of eigenvectors corresponding to a plurality of selected eigenchannels on the basis of a geometric mean of the eigenvalues. An eigenchannel encoder encodes the plurality of selected eigenchannels. A metadata encoder encodes the metadata.

Abstract: An apparatus and a method for encoding an input audio signal are disclosed. The input audio signal comprises a plurality of input audio channels. Metadata associated with a plurality of KLT eigenchannels allows reconstructing the plurality of input audio channels on the basis of the plurality of eigenchannels. A subset of the plurality of eigenchannels is encoded. The metadata is encoded and provided in a quantized form. The plurality of input audio channels is transformed into the plurality of eigenchannels on the basis of the metadata in the quantized form.

Abstract: The disclosure relates to an apparatus for encoding an input audio signal, wherein the input audio signal comprises a plurality of input audio channels. The apparatus comprises a KLT-based pre-processor configured to transform the plurality of input audio channels into a plurality of eigenchannels and to provide metadata in the form of a plurality of metadata elements, wherein the metadata allows reconstructing the plurality of input audio channels on the basis of the plurality of eigenchannels, a metadata re-arrangement unit configured to re-arrange the plurality of metadata elements on the basis of a re-arrangement scheme into one or more metadata blocks, wherein each of the one or more metadata blocks is a multi-dimensional array, and a metadata encoder configured to encode each of the one or more metadata blocks.

Abstract: Audio signal processing apparatuses and methods are provided, such as an audio signal downmixing apparatus for processing an input audio signal into an output audio signal, wherein the input audio signal comprises a plurality of input channels recorded at a plurality of spatial positions and the output audio signal comprises a plurality of primary output channels. The audio signal downmixing apparatus comprises a downmix matrix determiner configured to determine for each frequency bin j of a plurality of frequency bins a downmix matrix DU with j being an integer in the range from 1 to N, and a processor configured to process the input audio signal using the downmix matrix DU into the output audio signal.

Abstract: The disclosure relates to an apparatus for processing soundfield data, the soundfield data defining a soundfield within a spatial reproduction region comprising at least one bright zone and at least one quiet zone. The apparatus comprises an applicator configured to apply a spatially continuously varying weighting function to the soundfield data in order to obtain weighted soundfield data defining a weighted soundfield, wherein the spatially continuously varying weighting function is configured to enhance the soundfield in at least one of the bright zone and the quiet zone.

Abstract: A method for encoding an input signal comprising signal frames into quantized bits is disclosed, the method comprises generating, for each frame of the input signal, a signal matrix comprising matrix coefficients obtained from that frame, grouping the matrix coefficients of each signal matrix into a plurality of partition vectors, and for each partition vector, selecting one vector quantization scheme from among a plurality of vector quantization schemes and quantizing that partition vector according to the selected vector quantization scheme to obtain the quantized bits. In an adaptive mode, the method comprises grouping differently the matrix coefficients obtained from different frames, and/or selecting different vector quantization schemes for partition vectors obtained from different frames.

Abstract: Embodiments provide an audio signal encoding method comprising the steps of: collecting audio signal samples (114), determining sinusoidal components (312) in subsequent frames, estimation of amplitudes (314) and frequencies (313) of the components for each frame, merging thus obtained pairs into sinusoidal trajectories, splitting particular trajectories into segments, transforming (318, 319) particular trajectories to the frequency domain by means of a digital transform performed on segments longer than the frame duration, quantization (320, 321) and selection (322, 323) of transform coefficients in the segments, entropy encoding (328), outputting the quantized coefficients as output data (115), wherein segments of different trajectories starting within a particular time are grouped into Groups of Segments (GOS), and the partitioning of trajectories into segments is synchronized with the endpoints of a Group of Segments (GOS).

Abstract: An apparatus for estimating an overall mixing time, where the apparatus comprises a processing element configured to determine differences between energy profiles of a first room impulse response of the first pair of room impulse responses and a second room impulse response of the first pair of room impulse responses at a plurality of different sample times of the first pair of room impulse responses, set a sample time of the plurality of sample times as a mixing time for the first pair of room impulse responses at which the difference between the energy profiles of the first room impulse response and the second room impulse response of the first pair of room impulse responses is equal to or below a threshold value, and determine the overall mixing time based on the mixing time for the first pair of room impulse responses.

Abstract: The invention relates to audio signal processing apparatuses and methods, such as an audio signal downmixing apparatus (105) for processing an input audio signal comprising a plurality of input channels (113) into an output audio signal comprising a plurality of primary output channels (123) and at least one auxiliary output channel (125) using a downmix matrix D, wherein the downmix matrix D comprises a primary downmix matrix DU providing the plurality of primary output channels (123) and an auxiliary downmix matrix DW providing the at least one auxiliary output channel (125). The audio signal downmixing apparatus (105) comprises an auxiliary downmix matrix determiner (107) configured to determine the auxiliary downmix matrix DW, and a processor (109) configured to process the input audio signal into the output audio signal using the downmix matrix D.

Abstract: Audio signal processing apparatuses and methods are provided, such as an audio signal downmixing apparatus for processing an input audio signal into an output audio signal, wherein the input audio signal comprises a plurality of input channels recorded at a plurality of spatial positions and the output audio signal comprises a plurality of primary output channels. The audio signal downmixing apparatus comprises a downmix matrix determiner configured to determine for each frequency bin j of a plurality of frequency bins a downmix matrix DU with j being an integer in the range from 1 to N, and a processor configured to process the input audio signal using the downmix matrix DU into the output audio signal.

Abstract: The invention provides an audio decoder being configured for decoding a bitstream so as to produce therefrom an audio output signal, the bitstream including at least an active phase followed by at least an inactive phase, wherein the bitstream has encoded therein at least a silence insertion descriptor frame which describes a spectrum of a background noise, the audio decoder including: a silence insertion descriptor decoder configured to decode the silence insertion descriptor frame; a decoding device configured to reconstruct the audio output signal from the bitstream during the active phase; a spectral converter configured to determine a spectrum of the audio output signal; a noise estimator device configured to determine a first spectrum of the noise of the audio output signal; a resolution converter configured to establish a second spectrum of the noise of the audio output signal; a comfort noise spectrum estimation device; and a comfort noise generator.

Abstract: An apparatus for estimating an overall mixing time, where the apparatus comprises a processing element configured to determine differences between energy profiles of a first room impulse response of the first pair of room impulse responses and a second room impulse response of the first pair of room impulse responses at a plurality of different sample times of the first pair of room impulse responses, set a sample time of the plurality of sample times as a mixing time for the first pair of room impulse responses at which the difference between the energy profiles of the first room impulse response and the second room impulse response of the first pair of room impulse responses is equal to or below a threshold value, and determine the overall mixing time based on the mixing time for the first pair of room impulse responses.

Abstract: The invention relates to a method and means for encoding background noise information during voice signal encoding methods. A basic idea of the invention is to provide the scalability known for transmitting voice information in a similar manner when forming an SID frame. The invention provides encoding of a narrowband first component and of a broadband second component of a piece of background noise information and formation of an SID frame which describes the background noise with separate areas for the first and second components.

Abstract: The invention provides an audio decoder being configured for decoding a bitstream so as to produce therefrom an audio output signal, the bitstream including at least an active phase followed by at least an inactive phase, wherein the bitstream has encoded therein at least a silence insertion descriptor frame which describes a spectrum of a background noise, the audio decoder including: a silence insertion descriptor decoder configured to decode the silence insertion descriptor frame; a decoding device configured to reconstruct the audio output signal from the bitstream during the active phase; a spectral converter configured to determine a spectrum of the audio output signal; a noise estimator device configured to determine a first spectrum of the noise of the audio output signal; a resolution converter configured to establish a second spectrum of the noise of the audio output signal; a comfort noise spectrum estimation device; and a comfort noise generator.

Abstract: A parametric background noise estimate is continuously updated during an active or non-silence phase so that the noise generation may immediately be started with upon the entrance of an inactive phase following the active phase. In accordance with another aspect, a spectral domain is very efficiently used in order to parameterize the background noise thereby yielding a background noise synthesis which is more realistic and thus leads to a more transparent active to inactive phase switching.

Abstract: The inventive method provides for an encoder in a voice codec to be designed such that after a particular idle time (“Idle Period”) it recalculates the averaged energy and the autocorrelation function. Administrative points in the network inform the encoder about the idle time which has been set in the transmission network.