Abstract: An audio encoder adapted for encoding frames of a sampled audio signal to obtain encoded frames, wherein a frame includes a number of time domain audio samples. The audio encoder includes a predictive coding analysis stage for determining information on coefficients of a synthesis filter and a prediction domain frame based on a frame of audio samples. The audio encoder further includes a time-aliasing introducing transformer for transforming overlapping prediction domain frames to the frequency domain to obtain prediction domain frame spectra, wherein the time-aliasing introducing transformer is adapted for transforming the overlapping prediction domain frames in a critically-sampled way. Moreover, the audio encoder includes a redundancy reducing encoder for encoding the prediction domain frame spectra to obtain the encoded frames based on the coefficients and the encoded prediction domain frame spectra.

Abstract: An audio encoder for encoding segments of coefficients, the segments of coefficients representing different time or frequency resolutions of a sampled audio signal, the audio encoder including a processor for deriving a coding context for a currently encoded coefficient of a current segment based on a previously encoded coefficient of a previous segment, the previously encoded coefficient representing a different time or frequency resolution than the currently encoded coefficient. The audio encoder further includes an entropy encoder for entropy encoding the current coefficient based on the coding context to obtain an encoded audio stream.

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: An efficient encoded representation of a first and a second input audio signal can be derived using correlation information indicating a correlation between the first and the second input audio signals, when a signal characterization information, indicating at least a first or a second, different characteristic of the input audio signal is additionally considered. Phase information indicating a phase relation between the first and the second input audio signals is derived, when the input audio signals have the first characteristic. The phase information and a correlation measure are included into the encoded representation when the input audio signals have the first characteristic, and only the correlation information is included into the encoded representation when the input audio signals have the second characteristic.

Abstract: An embedder for embedding a watermark to be embedded into an input information representation comprises an information adder, which is implemented to provide the input information representation with the watermark and additional information to be added to obtain an input information representation provided with the watermark and the additional information. The additional information to be added comprises descriptive information describing the embedding of the at least one watermark into the input information representation.

Abstract: An embedder for embedding a watermark to be embedded into an input information representation comprises an embedding parameter determiner that is implemented to apply a derivation function once or several times to an initial value to obtain an embedding parameter for embedding the watermark into the input information representation. Further, the embedder comprises a watermark adder that is implemented to provide the input information representation with the watermark using the embedding parameter. The embedder is implemented to select how many times the derivation function is to be applied to the initial value.

Abstract: An audio encoder, an audio decoder or an audio processor includes a filter for generating a filtered audio signal, the filter having a variable warping characteristic, the characteristic being controllable in response to a time-varying control signal, the control signal indicating a small or no warping characteristic or a comparatively high warping characteristic. Furthermore, a controller is connected for providing the time-varying control signal, which depends on the audio signal. The filtered audio signal can be introduced to an encoding processor having different encoding algorithms, one of which is a coding algorithm adapted to a specific signal pattern. Alternatively, the filter is a post-filter receiving a decoded audio signal.

Abstract: An audio encoder for encoding an audio signal includes an impulse extractor for extracting an impulse-like portion from the audio signal. This impulse-like portion is encoded and forwarded to an output interface. Furthermore, the audio encoder includes a signal encoder which encodes a residual signal derived from the original audio signal so that the impulse-like portion is reduced or eliminated in the residual audio signal. The output interface forwards both, the encoded signals, i.e., the encoded impulse signal and the encoded residual signal for transmission or storage. On the decoder-side, both signal portions are separately decoded and then combined to obtain a decoded audio signal.

Abstract: An audio encoder, an audio decoder or an audio processor includes a filter (12) for generating a filtered audio signal, the filter having a variable warping characteristic, the characteristic being controllable in response to a time-varying control signal (16), the control signal indicating a small or no warping characteristic or a comparatively high warping characteristic. Furthermore, a controller (18) is connected for providing the time-varying control signal, which depends on the audio signal. The filtered audio signal can be introduced to an encoding processor (22) having different encoding algorithms, one of which is a coding algorithm adapted to a specific signal pattern. Alternatively, the filter is a post-filter receiving a decoded audio signal.

Abstract: An audio decoder for providing a decoded representation of an audio content on the basis of an encoded representation of the audio content comprises a linear-prediction-domain decoder core configured to provide a time-domain representation of an audio frame on the basis of a set of linear-prediction domain parameters associated with the audio frame and a frequency-domain decoder core configured to provide a time-domain representation of an audio frame on the basis of a set of frequency-domain parameters, taking into account a transform window out of a set comprising a plurality of different transform windows. The audio decoder comprises a signal combiner configured to overlap-and-add-time-domain representations of subsequent audio frames encoded in different domains, in order to smoothen a transition between the time-domain representations of the subsequent frames.

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 generating a signal to be transmitted original information is encoded into a main channel and a side channel, wherein the side channel is more robust against channel influences than the main channel. On the receiver side, when the receive quality is above a threshold, which is necessitated to execute a successful decoding of the main channel, the main channel is reproduced. If the receive quality falls below this threshold, however, the side channel is reproduced which may have less bits than the main channel and which is a correspondingly lower quality representation of the original information than the main channel.

Abstract: An embodiment of an apparatus for generating audio subband values in audio subband channels includes an analysis windower for windowing a frame of time-domain audio input samples being in a time sequence extending from an early sample to a later sample using an analysis window function including a sequence of window coefficients to obtain windowed samples. The analysis window function includes a first number of window coefficients derived from a larger window function including a sequence of a larger second number of window coefficients, wherein the window coefficients of the window function are derived by an interpolation of window coefficients of the larger window function. The apparatus further includes a calculator for calculating the audio subband values using the windowed samples.

Abstract: An embodiment of an apparatus for generating audio subband values in audio subband channels has an analysis windower for windowing a frame of time-domain audio input samples being in a time sequence extending from an early sample to a later sample using an analysis window function having a sequence of window coefficients to obtain windowed samples. The analysis window function has a first group of window coefficients and a second group of window coefficients. The first group of window coefficients is used for windowing later time-domain samples and the second group of window coefficients is used for windowing an earlier time-domain samples. The apparatus further has a calculator for calculating the audio subband values using the windowed samples.

Abstract: For determining a quantizer step size for quantizing a signal including audio or video information, a first quantizer step size as well as an interference threshold are provided. Then, the actual interference introduced by the first quantizer step size is determined and compared with the interference threshold. Despite the fact that the comparison reveals that the actually introduced interference exceeds the threshold, a second, coarser quantizer step size is nevertheless used, which will then be used for quantization if it turns out that the interference introduced by the coarser, second quantizer step size falls below the threshold or falls below the interference introduced by the first quantizer step size. Thus, the quantization interference is reduced while the quantization is coarsened and, thus, the compression gain is increased.

Abstract: For determining a quantizer step size for quantizing a signal including audio or video information, a first quantizer step size as well as an interference threshold are provided. Then, the actual interference introduced by the first quantizer step size is determined and compared with the interference threshold. Despite the fact that the comparison reveals that the actually introduced interference exceeds the threshold, a second, coarser quantizer step size is nevertheless used, which will then be used for quantization if it turns out that the interference introduced by the coarser, second quantizer step size falls below the threshold or falls below the interference introduced by the first quantizer step size. Thus, the quantization interference is reduced while the quantization is coarsened and, thus, the compression gain is increased.

Abstract: In a method for generating a scalable data stream, when a block of output data of a first encoder is present, this block of output data is written into the scalable data stream. If output data of a second encoder is present for a preceding period of time, this output data for the preceding section is written in transmission direction behind the block of output data of the first encoder into the data stream. When the output data of the scalable encoder for the current section is present, the output data of the second encoder is written into the bit stream subsequent to the output data of the first encoder. A determining data block is generated and written into the bit stream delayed by a period of time which corresponds to the size of the bit savings bank of the second encoder.

Abstract: In a method for generating a scalable data stream from one or several blocks of output data of a first encoder and from one or several blocks of output data of a second encoder a determining data block for a current section of an input signal is written. In addition, output data of the second encoder representing a preceding section of the input signal are written in transmission direction from an encoder to a decoder after the determining data block. When the output data of the second encoder are written for a preceding section of the input signal, the output data of the second encoder are written representing the current section of the input signal. In order to signalize where the output data of the second encoder for the preceding section end and where the output data of the second encoder for the current section begin, buffer information is written into the scalable data stream.

Abstract: In a method of producing a scalable data stream of at least two blocks of output data of a first coder and a block of output data of a second coder, wherein the at least two blocks of output data of the first coder together represent a current section of an input signal in the first coder, and wherein the block of output data of the second coder represents the same current section of the input signal, a determination data block for the current section of the input signal is written. In addition, the block of output data of the second coder, in the direction of transfer from a coding device to a decoding device, is written after the determination data block for the current section of the input signal.

Abstract: An embodiment of an analysis filterbank for filtering a plurality of time domain input frames, wherein an input frame comprises a number of ordered input samples, comprises a windower configured to generating a plurality of windowed frames, wherein a windowed frame comprises a plurality of windowed samples, wherein the windower is configured to process the plurality of input frames in an overlapping manner using a sample advance value, wherein the sample advance value is less than the number of ordered input samples of an input frame divided by two, and a time/frequency converter configured to providing an output frame comprising a number of output values, wherein an output frame is a spectral representation of a windowed frame.