Abstract: Embodiments of the present invention provide an encoder comprising a quantization stage, an entropy encoder, a residual quantization stage and a coded signal former. The quantization stage is configured to quantize an input signal using a dead zone in order to obtain a plurality of quantized values. The entropy encoder is configured to encode the plurality of quantized values using an entropy encoding scheme in order to obtain a plurality of entropy encoded values. The residual quantization stage is configured to quantize a residual signal caused by the quantization stage, wherein the residual quantization stage is configured to determine at least one quantized residual value in dependence on the dead zone of the quantization stage. The coded signal former is configured to form a coded signal from the plurality of entropy encoded values and the at least one quantized residual value.

Abstract: An apparatus for downmixing three or more audio input channels to obtain two or more audio output channels is provided. The apparatus includes a receiving interface for receiving the three or more audio input channels and for receiving side information. Moreover, the apparatus includes a downmixer for downmixing the three or more audio input channels depending on the side information to obtain the two or more audio output channels. The number of the audio output channels is smaller than the number of the audio input channels. The side information indicates a characteristic of at least one of the three or more audio input channels, or a characteristic of one or more sound waves recorded within the one or more audio input channels, or a characteristic of one or more sound sources which emitted one or more sound waves recorded within the one or more audio input channels.

Abstract: An audio encoder for encoding an audio signal, includes: a first encoding processor for encoding a first audio signal portion in a frequency domain, wherein the first encoding processor includes: a time frequency converter for converting the first audio signal portion into a frequency domain representation having spectral lines up to a maximum frequency of the first audio signal portion; a spectral encoder for encoding the frequency domain representation; a second encoding processor for encoding a second different audio signal portion in the time domain; a cross-processor for calculating, from the encoded spectral representation of the first audio signal portion, initialization data of the second encoding processor, so that the second encoding processing is initialized to encode the second audio signal portion immediately following the first audio signal portion in time in the audio signal.

Abstract: An apparatus for processing an encoded signal, the encoded signal having an encoded audio signal having information on a pitch delay or a pitch gain, and a bass post-filter control parameter, has: an audio signal decoder for decoding the encoded audio signal using the information on the pitch delay or the pitch gain to obtain a decoded audio signal; a controllable bass post-filter for filtering the decoded audio signal to obtain a processed signal, wherein the controllable bass post-filter has the variable bass post-filter characteristic controllable by the bass post-filter control parameter; and a controller for setting the variable bass post-filter characteristic in accordance with the bass post-filter control parameter included in the encoded signal.

Abstract: An apparatus for generating four or more audio output signals has a panning gain determiner and a signal processor. The panning gain determiner is configured to determine a proper subset from a set of five or more loudspeaker positions, so that the proper subset has four or more of the five or more loudspeaker positions. Moreover, the panning gain determiner is configured to determine the proper subset depending on a panning position and on the five or more loudspeaker positions, and to determine a panning gain for each of the four or more audio output signals by determining the panning gain depending on the panning position and on the four or more loudspeaker positions of the proper subset. The signal processor is configured to generate each of the four or more audio output signals depending on the panning gain for the audio output signal and on an audio input signal.

Abstract: An apparatus for processing an audio signal is provided. The apparatus has a signal processor and a configurator. The configurator is adapted to configure the signal processor based on configuration information such that a configurable upsampling factor is equal to a first upsampling value when a first ratio of the second configurable number of samples to a first configurable number of samples has a first ratio value. Moreover, the configurator is adapted to configure the signal processor such that the configurable upsampling factor is equal to a different second upsampling value, when a different second ratio of the second configurable number of samples to the first configurable number of samples has a different second ratio value. The first or the second ratio value is not an integer value.

Abstract: An apparatus for playing back an audio object associated with a position includes a distance calculator for calculating distances of the position to speakers or for reading the distances of the position to the speakers. The distance calculator is configured to take a solution with a smallest distance. The apparatus is configured to play back the audio object using the speaker corresponding to the solution.

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: In an embodiment, bitstream elements of sub-frames are encoded differentially to a global gain value so that a change of the global gain value results in an adjustment of an output level of the decoded representation of the audio content. Concurrently, the differential coding saves bits. Even further, the differential coding enables the lowering of the burden of globally adjusting the gain of an encoded bitstream. In another embodiment, a global gain control across CELP coded frames and transform coded frames is achieved by co-controlling the gain of the codebook excitation of the CELP codec, along with a level of the transform or inverse transform of the transform coded frames. In another embodiment, the gain value determination in CELP coding is performed in the weighted domain of the excitation signal.

Abstract: An apparatus for generating a plurality of audio channels for a first speaker setup is characterized by an imaginary speaker determiner, an energy distribution calculator, a processor and a renderer. The imaginary speaker determiner is configured to determine a position of an imaginary speaker not contained in the first speaker setup to obtain a second speaker setup containing the imaginary speaker. The energy distribution calculator is configured to calculate an energy distribution from the imaginary speaker to the other speakers in the second speaker setup. The processor is configured to repeat the energy distribution to obtain a downmix information for a downmix from the second speaker setup to the first speaker setup. The renderer is configured to generate the plurality of audio channels using the downmix information.

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: In an embodiment, bitstream elements of sub-frames are encoded differentially to a global gain value so that a change of the global gain value results in an adjustment of an output level of the decoded representation of the audio content. Concurrently, the differential coding saves bits. Even further, the differential coding enables the lowering of the burden of globally adjusting the gain of an encoded bitstream. In another embodiment, a global gain control across CELP coded frames and transform coded frames is achieved by co-controlling the gain of the codebook excitation of the CELP codec, along with a level of the transform or inverse transform of the transform coded frames. In another embodiment, the gain value determination in CELP coding is performed in the weighted domain of the excitation signal.

Abstract: A decoder for generating an audio output signal having one or more audio output channels is provided. The decoder includes a receiving interface for receiving an audio input signal including a plurality of audio object signals, for receiving loudness information on the audio object signals, and for receiving rendering information indicating whether one or more of the audio object signals shall be amplified or attenuated. Moreover, the decoder includes a signal processor for generating the one or more audio output channels of the audio output signal. The signal processor is configured to determine a loudness compensation value depending on the loudness information and depending on the rendering information. Furthermore, the signal processor is configured to generate the one or more audio output channels of the audio output signal from the audio input signal depending on the rendering information and depending on the loudness compensation value. Moreover, an encoder is provided.

Abstract: A method is described which decodes a downmix matrix for mapping a plurality of input channels of audio content to a plurality of output channels, the input and output channels being associated with respective speakers at predetermined positions relative to a listener position, wherein the downmix matrix is encoded by exploiting the symmetry of speaker pairs of the plurality of input channels and the symmetry of speaker pairs of the plurality of output channels. Encoded information representing the encoded downmix matrix is received and decoded for obtaining the decoded downmix matrix.

Abstract: A watermark generator for providing a watermark signal in dependence on binary message data includes an information processor configured to provide, in dependence on information units of the binary message data, a first time-frequency domain representation, values of which represent the binary message data. The watermark generator also includes a differential encoder configured to derive a second time-frequency domain representation from the first time-frequency-domain representation, such that the second time-frequency-domain representation includes a plurality of values, wherein a difference between two values of the second time-frequency-domain representation represents a corresponding value of the first time-frequency-domain representation, in order to obtain a differential encoding of the values of the first time-frequency-domain representation.

Abstract: A watermark decoder includes a time-frequency-domain representation provider, a memory unit, a synchronization determiner and a watermark extractor. The time-frequency-domain representation provider provides a frequency-domain representation of the watermarked signal for a plurality of time blocks. The memory unit stores the frequency-domain representation of the watermarked signal for a plurality of time blocks. Further, the synchronization determiner identifies an alignment time block based on the frequency-domain representation of the watermarked signal of a plurality of time blocks. The watermark extractor provides binary message data based on stored frequency-domain representations of the watermarked signal of time blocks temporally preceding the identified alignment time block considering a distance to the identified alignment time block.

Abstract: An apparatus for selecting one of a first encoding algorithm and a second encoding algorithm includes a filter configured to receive the audio signal, to reduce the amplitude of harmonics in the audio signal and to output a filtered version of the audio signal. First and second estimators are provided for estimating first and second quality measures in the form of SNRs of segmented SNRs associated with the first and second encoding algorithms without actually encoding and decoding the portion of the audio signal using the first and second encoding algorithms. A controller is provided for selecting the first encoding algorithm or the second encoding algorithm based on a comparison between the first quality measure and the second quality measure.

Abstract: A watermark signal provider comprises a time-frequency-domain waveform provider to provide time-domain waveforms for a plurality of frequency subbands. The time-frequency-domain waveform provider is configured to map a given value of a time-frequency-domain representation onto a bit shaping function, a temporal extension of which is longer than a bit interval, such that there is a temporal overlap between bit shaped functions provided for temporally subsequent values of the time-frequency-domain representation of the same frequency subband. A time-domain waveform of a given frequency subband contains a plurality of bit shaped functions provided for temporally subsequent values of the time-frequency-domain representation. The water mark signal provider further has a time-domain waveform combiner.

Abstract: A decoder for generating an audio output signal having one or more audio output channels is provided, having a receiving interface for receiving an audio input signal having a plurality of audio object signals, for receiving loudness information on the audio object signals, and for receiving rendering information indicating whether one or more of the audio object signals shall be amplified or attenuated, further having a signal processor for generating the one or more audio output channels of the audio output signal, configured to determine a loudness compensation value depending on the loudness information and depending on the rendering information, and configured to generate the one or more audio output channels of the audio output signal from the audio input signal depending on the rendering information and depending on the loudness compensation value. One or more by-pass audio object signals are employed for generating the audio output signal. Moreover, an encoder is provided.

Abstract: The invention provides an audio encoder including a combination of a linear predictive coding filter having a plurality of linear predictive coding coefficients and a time-frequency converter, wherein the combination is configured to filter and to convert a frame of the audio signal into a frequency domain in order to output a spectrum based on the frame and on the linear predictive coding coefficients; a low frequency emphasizer configured to calculate a processed spectrum based on the spectrum, wherein spectral lines of the processed spectrum representing a lower frequency than a reference spectral line are emphasized; and a control device configured to control the calculation of the processed spectrum by the low frequency emphasizer depending on the linear predictive coding coefficients of the linear predictive coding filter.