Abstract: An apparatus for decoding an encoded audio signal having first and second portions encoded in accordance with first and second encoding algorithms, respectively, BWE parameters for the first and second portions and a coding mode information indicating a first or a second decoding algorithm, includes first and second decoders, a BWE module and a controller. The decoders decode portions in accordance with decoding algorithms for time portions of the encoded signal to obtain decoded signals. The BWE module has a controllable crossover frequency and is configured for performing a bandwidth extension algorithm using the first decoded signal and the BWE parameters for the first portion, and for performing a bandwidth extension algorithm using the second decoded signal and the bandwidth extension parameter for the second portion. The controller controls the crossover frequency for the BWE module in accordance with the coding mode information.

Abstract: An audio encoder has a first information sink oriented encoding branch such as a spectral domain encoding branch, a second information source or SNR oriented encoding branch such as an LPC-domain encoding branch, and a switch for switching between the first encoding branch and the second encoding branch, wherein the second encoding branch has a converter into a specific domain different from the spectral domain such as an LPC analysis stage generating an excitation signal, and wherein the second encoding branch furthermore has a specific domain coding branch such as LPC domain processing branch, and a specific spectral domain coding branch such as LPC spectral domain processing branch, and an additional switch for switching between the specific domain coding branch and the specific spectral domain coding branch.

Abstract: An apparatus for encoding includes a first domain converter, a switchable bypass, a second domain converter, a first processor and a second processor to obtain an encoded audio signal having different signal portions represented by coded data in different domains, which have been coded by different coding algorithms. Corresponding decoding stages in the decoder together with a bypass for bypassing a domain converter allow the generation of a decoded audio signal with high quality and low bit rate.

Abstract: An encoder for providing an audio stream on the basis of a transform-domain representation of an input audio signal includes a quantization error calculator configured to determine a multi-band quantization error over a plurality of frequency bands of the input audio signal for which separate band gain information is available. The encoder also includes an audio stream provider for providing the audio stream such that the audio stream includes information describing an audio content of the frequency bands and information describing the multi-band quantization error. A decoder for providing a decoded representation of an audio signal on the basis of an encoded audio stream representing spectral components of frequency bands of the audio signal includes a noise filler for introducing noise into spectral components of a plurality of frequency bands to which separate frequency band gain information is associated on the basis of a common multi-band noise intensity value.

Abstract: A noise filler for providing a noise-filled spectral representation of an audio signal on the basis of an input spectral representation of the audio signal has a spectral region identifier configured to identify spectral regions of the input spectral representation spaced from non-zero spectral regions of the input spectral representation by at least one intermediate spectral region, to obtain identified spectral regions, and a noise inserter configured to selectively introduce noise into the identified spectral regions to obtain the noise-filled spectral representation of the audio signal. A noise filling parameter calculator for providing a noise filling parameter on the basis of a quantized spectral representation of an audio signal has a spectral region identifier, as mentioned above, and a noise value calculator configured to selectively consider quantization errors of the identified spectral regions for a calculation of the noise filling parameter.

Abstract: For time synchronization of a data stream with multi-channel additional data and a data stream with data on at least one base channel, a fingerprint information calculation is performed on the encoder side for the at least one base channel to insert the fingerprint information into a data stream in time connection to the multi-channel additional data. On the decoder side, fingerprint information are calculated from the at least one base channel and used together with the fingerprint information extracted from the data stream to calculate and compensate a time offset between the data stream with the multi-channel additional information and the data stream with the at least one base channel, for example by means of a correlation, to obtain a synchronized multi-channel representation.

Abstract: In order to generate a multi-channel signal having a number of output channels greater than a number of input channels, a mixer is used for upmixing the input signal to form at least a direct channel signal and at least an ambience channel signal. A speech detector is provided for detecting a section of the input signal, the direct channel signal or the ambience channel signal in which speech portions occur. Based on this detection, a signal modifier modifies the input signal or the ambience channel signal in order to attenuate speech portions in the ambience channel signal, whereas such speech portions in the direct channel signal are attenuated to a lesser extent or not at all. A loudspeaker signal outputter then maps the direct channel signals and the ambience channel signals to loudspeaker signals which are associated to a defined reproduction scheme, such as, for example, a 5.1 scheme.

Abstract: A processed representation of an audio signal having a sequence of frames is generated by sampling the audio signal within first and second frames of the sequence of frames, the second frame following the first frame, the sampling using information on a pitch contour of the first and second frames to derive a first sampled representation. The audio signal is sampled within the second and third frames, the third frame following the second frame in the sequence of frames. The sampling uses the information on the pitch contour of the second frame and information on a pitch contour of the third frame to derive a second sampled representation. A first scaling window is derived for the first sampled representation, and a second scaling window is derived for the second sampled representation, the scaling windows depending on the samplings applied to derive the first sampled representations or the second sampled representation.

Abstract: The manipulation of audio data can be simplified, such as, for example, with regard to the combination of individual audio data streams to multi-channel audio data streams or the general manipulation of an audio data stream, by modifying a data block in an audio data stream divided into data blocks with determination block and data block audio data, such as by completing or adding or replacing part of the same, so that the same includes a length indicator indicating an amount or length of data, respectively, of the data block audio data or an amount or length of data, respectively, of the data block to obtain a second audio data stream with modified data blocks. Alternatively, an audio data stream with pointers in determination blocks, which point to determination block audio data associated to this determination blocks, but distributed among different data blocks, is converted into an audio data stream, wherein the determination block audio data are combined to contiguous determination block audio data.

Abstract: For synthesizing at least three output channels using two stereo input channels, the stereo input channels are analyzed to detect signal components occurring in both input channels. A signal generator is operative to introduce at least a part of the detected signal components into the second channel associated with a second speaker in an intended speaker scheme, which is positioned between a first and a third speaker in the speaker scheme. When, however, feeding of the complete detected signal components would result in a clipping situation, then only a part of the detected signal components is fed into the second channel as a real center channel and the remainder is located in the first and third channels as a phantom center channel.

Abstract: For postprocessing spectral values which are based on a first transformation algorithm for converting the audio signal into a spectral representation, first a sequence of blocks of the spectral values representing a sequence of blocks of samples of the audio signal are provided.

Abstract: In order to process a subband signal of a plurality of real subband signals which are a representation of a real discrete-time signal generated by an analysis filter bank, a weighter for weighting a subband signal by a weighting factor determined for the subband signal is provided to obtain a weighted subband signal. In addition, a correction term is calculated by a correction term determiner, the correction term determiner being implemented to calculate the correction term using at least one other subband signal and using another weighting factor provided for the other subband signal, the two weighting factors differing. The correction term is then combined with the weighted subband signal to obtain a corrected subband signal, resulting in reduced aliasing, even if subband signals are weighted to a different extent.

Abstract: In a case of transient audio input signals, in a multi-channel audio reconstruction, uncorrelated output signals are generated from an audio input signal in that the audio input signal is mixed with a representation of the audio input signal delayed by a delay time such that, in a first time interval, a first output signal corresponds to the audio input signal, and a second output signal corresponds to the delayed representation of the audio input signal, wherein, in a second time interval, the first output signal corresponds to the delayed representation of the audio input signal, and the second output signal corresponds to the audio input signal.

Abstract: A stereo signal with enhanced perceptual quality using a mid-signal and a side-signal, can be generated, when a enhanced side signal is created prior to the upmix of the stereo signal. A decorrelated representation of at least a portion of the sum signal and/or a decorrelated representation of at least a portion of the side-signal is generated. The enhanced side-signal is generated combining a representation of the side-signal with the decorrelated representation of the portion of the mid signal, with the decorrelated representation of the side-signal and the decorrelated representation of the portion of the mid-signal or with the portion of the mid-signal and the decorrelated representation of the portion of the side-signal. The stereo signal with enhanced perceptual quality is created using a representation of the mid-signal and the enhanced side-signal.

Abstract: For time synchronization of a data stream with multi-channel additional data and a data stream with data on at least one base channel, a fingerprint information calculation is performed on the encoder side for the at least one base channel to insert the fingerprint information into a data stream in time connection to the multi-channel additional data. On the decoder side, fingerprint information are calculated from the at least one base channel and used together with the fingerprint information extracted from the data stream to calculate and compensate a time offset between the data stream with the multi-channel additional information and the data stream with the at least one base channel, for example by means of a correlation, to obtain a synchronized multi-channel representation.

Abstract: A device for generating an encoded stereo signal from a multi-channel representation includes a multi-channel decoder generating three of more multi-channels from at least one basic channel and parametric information. The three or more multi-channels are subjected to headphone signal processing to generate an uncoded first stereo channel and an uncoded second stereo channel which are then supplied to a stereo encoder to generate an encoded stereo file on the output side. The encoded stereo file may be supplied to any suitable player in the form of a CD player or a hardware player such that a user of the player does not only get a normal stereo impression but a multi-channel impression.

Abstract: A device for checking a data set, wherein the data set has a plurality of partial sets and a partial control value per partial set includes a determiner for determining a control value and comparer. The determiner for determining a control value is formed so as to determine a common control value from the partial control values of the partial sets. The comparer is formed so as to compare the common control value to a comparison value provided to the comparer.

Abstract: The present invention is based on the finding that a reconstructed output channel, reconstructed with a multi-channel reconstructor using at least one downmix channel derived by downmixing a plurality of original channels and using a parameter representation including additional information on a temporal fine structure of an original channel can be reconstructed efficiently with high quality, when a generator for generating a direct signal component and a diffuse signal component based on the downmix channel is used. The quality can be essentially enhanced, if only the direct signal component is modified such that the temporal fine structure of the reconstructed output channel is fitting a desired temporal fine structure, indicated by the additional information on the temporal fine structure transmitted.

Abstract: In a multi-channel encoder generating several different parameter sets for reconstructing a multi-channel output signal using at least one transmission channel, the data stream is written such that the two parameter sets are decodable independently of each other. Thus, a multi-channel decoder is enabled to skip a parameter set which is marked as optional and/or has a higher version number when reading the data stream and still to perform a valid multi-channel reconstruction using a data set marked as mandatory or a data set having a sufficiently low version number. This achieves a flexible encoder/decoder concept suitable for future updates characterized by backward compatibility and reliability.

Abstract: For synthesizing at least three output channels using two stereo input channels, the stereo input channels are analyzed to detect signal components occurring in both input channels. A signal generator is operative to introduce at least a part of the detected signal components into the second channel associated with a second speaker in an intended speaker scheme, which is positioned between a first and a third speaker in the speaker scheme. When, however, feeding of the complete detected signal components would result in a clipping situation, then only a part of the detected signal components is fed into the second channel as a real center channel and the remainder is located in the first and third channels as a phantom center channel.