Abstract: An audio encoder has a first information sink oriented encoding branch, a second information source or SNR oriented 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, and wherein the second encoding branch furthermore has a specific domain coding branch, and a specific spectral domain coding branch, and an additional switch for switching between the specific domain coding branch and the specific spectral domain coding branch. An audio decoder has a first domain decoder, a second domain decoder for decoding a signal, and a third domain decoder and two cascaded switches for switching between the decoders.

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: 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 audio encoder for encoding audio samples has a first time domain aliasing introducing encoder configured to decode audio samples in a first encoding domain and having a first framing rule, a start window and a stop window. The audio encoder further has a second encoder configured to encode samples in a second encoding domain and having a predetermined frame size number of audio samples, and a coding warm-up period number of audio samples, the second encoder having a different second framing rule, a frame of the second encoder being an encoded representation of a number of successive audio samples that is equal to the predetermined frame size number of audio samples. The audio encoder further has a controller switching from the first to the second encoder and for modifying the second framing rule or for modifying the start or the stop window of the first encoder.

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: An audio encoder has a common preprocessing stage, an information sink based encoding branch such as spectral domain encoding branch, a information source based encoding branch such as an LPC-domain encoding branch and a switch for switching between these branches at inputs into these branches or outputs of these branches controlled by a decision stage. An audio decoder has a spectral domain decoding branch, an LPC-domain decoding branch, one or more switches for switching between the branches and a common post-processing stage for post-processing a time-domain audio signal for obtaining a post-processed audio signal.

Abstract: An audio signal synthesizer generates a synthesis audio signal having a first frequency band and a second synthesized frequency band derived from the first frequency band and comprises a patch generator, a spectral converter, a raw signal processor and a combiner. The patch generator performs at least two different patching algorithms, each patching algorithm generating a raw signal. The patch generator is adapted to select one of the at least two different patching algorithms in response to a control information. The spectral converter converts the raw signal into a raw signal spectral representation. The raw signal processor processes the raw signal spectral representation in response to spectral domain spectral band replication parameters to obtain an adjusted raw signal spectral representation.

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 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: 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 audio signal synthesizer generates a synthesis audio signal having a first frequency band and a second synthesized frequency band derived from the first frequency band and comprises a patch generator, a spectral converter, a raw signal processor and a combiner. The patch generator performs at least two different patching algorithms, each patching algorithm generating a raw signal. The patch generator is adapted to select one of the at least two different patching algorithms in response to a control information. The spectral converter converts the raw signal into a raw signal spectral representation. The raw signal processor processes the raw signal spectral representation in response to spectral domain spectral band replication parameters to obtain an adjusted raw signal spectral representation.

Abstract: A watermark generator for providing a watermark signal in dependence on binary message data, the watermark generator has an information spreader configured to spread an information unit to a plurality of time-frequency-domain values, to obtain a spread information representation. The watermark generator also has a synchronization inserter configured to multiplicatively combine the spread information representation with a synchronization sequence to obtain a combined information-synchronization representation. The watermark generator also has a watermark signal provider configured to provide the watermark signal on the basis of the combined information-synchronization representation. A watermark decoder, methods and computer programs are also described.

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 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 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 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 generate 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 provide an output frame comprising a number of output values, wherein an output frame is a spectral representation of a windowed frame.

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 generate 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 provide an output frame comprising a number of output values, wherein an output frame is a spectral representation of a windowed frame.

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 generate 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 provide an output frame comprising a number of output values, wherein an output frame is a spectral representation of a windowed frame.

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 generate 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 provide an output frame comprising a number of output values, wherein an output frame is a spectral representation of a windowed frame.