Abstract: Multimedia application data formats often use data packets and may contain packetized hierarchical layers with a base layer (BL) and one or more enhancement layers (EL). Packets that comprise variable length coded data typically provide padding bits, which can be used for adding checksum information. The checksum information refers to both, the decoded base layer and decoded enhancement layer packet data, and is used for detecting synchronization loss between the layers. Though the packets provide different individual amounts of padding bits, this variable amount of bits per packet is sufficient for adding checksum information for synchronization related purposes. This allows e.g. out-of-sync detection even if the enhancement layer encoding provides no overhead information for synchronization, in particular no packet headers and no packet synchronization words.

Abstract: Spherical microphone arrays capture a three-dimensional sound field (P(?c, t)) for generating an Ambisonics representation (Anm(t)), where the pressure distribution on the surface of the sphere is sampled by the capsules of the array. The impact of the microphones on the captured sound field is removed using the inverse microphone transfer function. The equalisation of the transfer function of the microphone array is a big problem because the reciprocal of the transfer function causes high gains for small values in the transfer function and these small values are affected by transducer noise.

Abstract: Spherical microphone arrays capture a three-dimensional sound field for generating an Ambisonics representation, where the pressure distribution on the surface of the sphere is sampled by the capsules of the array. The impact of the microphones on the captured sound field is removed using the inverse microphone transfer function. The equalisation of the transfer function of the microphone array is a big problem because the reciprocal of the transfer function causes high gains for small values in the transfer function and these small values are affected by transducer noise. The invention minimises that noise by using a Wiener filter processing in the frequency domain, which processing is automatically controlled per wave number by the signal-to-noise ratio of the microphone array.

Abstract: In frame-based bit stream formats the data required for decoding a current frame are usually stored within the data section for that frame. One exception is the mp3 bit stream where data for a current frame is stored in previous frames. If the decoder did not receive the required previous frame, decoding of the current mp3 frame is skipped. The invention can be applied for such bit streams, in an archival mode, a streaming mode and a sample-exact cutting of an archival mode. In the streaming and cutting modes, new headers are established. The number of frames required for initializing the decoder status is signalized in the header, as well as a consistency check value in the streaming mode. These frames are used for decoder initialization but not for decoding samples or coefficients. For a sample-exact cutting, for the frame at which the cut shall occur, the number of samples or coefficients to be muted is also indicated in the header.

Abstract: Filter banks may have different structures and different individual output signal domains. Often a translation between different filter bank domains is desirable. Usually, mapping matrices are used that, however, vary over frequency. This requires a significant amount of lookup tables. A method for transforming first data frames of a first filter bank domain to second data frames of a different second filter bank domain, comprises steps of transcoding sub-bands of the first filter bank domain into sub-bands of an intermediate domain that corresponds to said second filter bank domain but has warped phase, and transcoding the sub-bands of the intermediate domain to sub-bands of the second filter bank domain, wherein a phase correction is performed on the sub-bands of the intermediate domain.

Abstract: An audio signal may have a BL and an EL, wherein the EL represents additional information for enhancing the quality of the BL audio content. Decoding of such dual-layer signals usually comprises partial decoding of the BL data, wherein frequency bins of the BL are restored, mapping the restored frequency bins to the MDCT domain, adding them to the decoded EL and performing inverse Integer MDCT. A low-complexity method for decoding comprises reverse mapping of the decoded EL data, adding the reverse mapped EL data to the partially decoded BL data and filtering the sum, using the inverse BL filter bank.

Abstract: The invention is related to a data structure for Higher Order Ambisonics HOA audio data, which data structure includes 2D or 3D spatial audio content data for one or more different HOA audio data stream descriptions. The HOA audio data can have on order of greater than ‘3’, and the data structure in addition can include single audio signal source data and/or microphone array audio data from fixed or time-varying spatial positions.

Abstract: Multimedia application data formats often use data packets and may contain packetized hierarchical layers with a base layer (BL) and one or more enhancement layers (EL). Packets that comprise variable length coded data typically provide padding bits, which can be used for adding checksum information. The checksum information refers to both, the decoded base layer and decoded enhancement layer packet data, and is used for detecting synchronization loss between the layers. Though the packets provide different individual amounts of padding bits, this variable amount of bits per packet is sufficient for adding check-sum information for synchronization related purposes. This allows e.g. out-of-sync detection even if the enhancement layer encoding provides no overhead information for synchronization, in particular no packet headers and no packet synchronization words.

Abstract: In lossy based lossless coding a PCM audio signal passes through a lossy encoder to a lossy decoder. The lossy encoder provides a lossy bit stream. The difference signal between the PCM signal and the lossy decoder output is lossless encoded, providing an extension bit stream. The invention facilitates enhancing a lossy perceptual audio encoding/decoding by an extension that enables mathematically exact reproduction of the original waveform using enhanced de-correlation, and provides additional data for reconstructing at decoder site an intermediate-quality audio signal. The lossless extension can be used to extend the widely used mp3 encoding/decoding to lossless encoding/decoding and superior quality mp3 encoding/de-coding.

Abstract: A two-layer hierarchical audio bit stream can have a frame-based structure for the base layer bit stream and can be decoded independently from a higher layer and the decoding can start following every sync header. In the extension layer bit stream the frame structure may not be reflected on bit stream level. To facilitate seek operations with such highly compressed extension-layer data, the header of the extension layer bit stream comprises an FAT table with seek target positions. Because there are fewer entry points in the enhancement layer than sync headers in the base layer, a re-synchronisation and some base layer frames are required to start decoding of the enhancement layer and to generate the full audio quality. Three seeking ways of seeking are described, of which each one offers a different compromise between seeking accuracy, re-synchronisation latency and audio quality.

Abstract: Advanced solutions for encrypting multi-layer audio data are required, ie. audio data that comprise a base layer and one or more enhancement layers. A method for encrypting such an encoded audio signal comprises separating the base layer into two sections, encrypting the side information within frames of the second section of the base layer, and encrypting at least a part of the data of the enhancement layer, wherein the encrypted section of the base layer and the encrypted enhancement layer require different decryption keys for decryption. Thus, free preview zones are possible to implement.

Abstract: Lossless audio coding performs decorrelation and encodes the transformed signal. The encoded bit stream comprises de-correlation parameters and the lossless representation data of the transformed signal. However, in the case of lossy based lossless coding, the additional amount of information exceeds the base layer amount of data. Therefore the additional data cannot be packed completely into the base layer e.g. as ancillary data. The data streams resulting from the combination of lossy coding format with a lossless coding extension are the base layer containing the lossy coding information and the enhancement data stream for rebuilding the mathematically lossless original input signal. Every higher layer depends on the lower layers and can only be reasonably decoded in combination with these lower layers. According to the invention, a special combination of one-time header information with repeated header information in a block structure is used. Assignment information data identify the different layers.

Abstract: In lossy based lossless coding a PCM audio signal passes through a lossy encoder to a lossy decoder. The lossy encoder provides a lossy bit stream. The lossy decoder also provides side information that is used to control the coefficients of a prediction filter that de-correlates the difference signal between the PCM signal and the lossy decoder output. The de-correlated difference signal is lossless encoded, providing an extension bit stream. Instead of, or in addition to, de-correlating in the time domain, a de-correlation in the frequency domain using spectral whitening can be performed. The lossy encoded bit stream together with the lossless encoded extension bit stream form a lossless encoded bitstream. The invention facilitates enhancing a lossy perceptual audio encoding/decoding by an extension that enables mathematically exact reproduction of the original waveform, and provides additional data for reconstructing at decoder site an intermediate-quality audio signal.

Abstract: Representations of spatial audio scenes using higher-order Ambisonics HOA technology typically require a large number of coefficients per time instant. This data rate is too high for most practical applications that require real-time transmission of audio signals. According to the invention, the compression is carried out in spatial domain instead of HOA domain. The (N+1)2 input HOA coefficients are transformed into (N+1)2 equivalent signals in spatial domain, and the resulting (N+1)2 time-domain signals are input to a bank of parallel perceptual codecs. At decoder side, the individual spatial-domain signals are decoded, and the spatial-domain coefficients are transformed back into HOA domain in order to recover the original HOA representation.

Abstract: Perceptual audio codecs make use of filter banks and MDCT in order to achieve a compact representation of the audio signal, by removing redundancy and irrelevancy from the original audio signal. During quasi-stationary parts of the audio signal a high frequency resolution of the filter bank is advantageous in order to achieve a high coding gain, but this high frequency resolution is coupled to a coarse temporal resolution that becomes a problem during transient signal parts by producing audible pre-echo effects. The invention achieves improved coding/decoding quality by applying on top of the output of a first filter bank a second non-uniform filter bank, i.e. a cascaded MDCT. The inventive codec uses switching to an additional extension filter bank (or multi-resolution filter bank) in order to re-group the time-frequency representation during transient or fast changing audio signal sections.

Abstract: Multimedia application data formats often use data packets and may contain packetized hierarchical layers with a base layer (BL) and one or more enhancement layers (EL). Packets that comprise variable length coded data typically provide padding bits, which can be used for adding checksum information. The checksum information refers to both, the decoded base layer and decoded enhancement layer packet data, and is used for detecting synchronization loss between the layers. Though the packets provide different individual amounts of padding bits, this variable amount of bits per packet is sufficient for adding check-sum information for synchronization related purposes. This allows e.g. out-of-sync detection even if the enhancement layer encoding provides no overhead information for synchronization, in particular no packet headers and no packet synchronization words.

Abstract: An audio signal may have a BL and an EL, wherein the EL represents additional information for enhancing the quality of the BL audio content. Decoding of such dual-layer signals usually comprises partial decoding of the BL data, wherein frequency bins of the BL are restored, mapping the restored frequency bins to the MDCT domain, adding them to the decoded EL and performing inverse Integer MDCT. A low-complexity method for decoding comprises reverse mapping of the decoded EL data, adding the reverse mapped EL data to the partially decoded BL data and filtering the sum, using the inverse BL filter bank.

Abstract: Filter banks may have different structures and different individual output signal domains. Often a translation between different filter bank domains is desirable. Usually, mapping matrices are used that, however, vary over frequency. This requires a significant amount of lookup tables. A method for transforming first data frames of a first filter bank domain to second data frames of a different second filter bank domain, comprises steps of transcoding sub-bands of the first filter bank domain into sub-bands of an intermediate domain that corresponds to said second filter bank domain but has warped phase, and transcoding the sub-bands of the intermediate domain to sub-bands of the second filter bank domain, wherein a phase correction is performed on the sub-bands of the intermediate domain.

Abstract: Lossless compression algorithms can only exploit redundancies of the original audio signal to reduce the data rate, but not irrelevancies as identified by psycho-acoustics. Lossless audio coding schemes apply a filter or transform for decorrelation and then encode the transformed signal. The encoded bit stream comprises the parameters of the transform or filter, and the lossless representation of the transformed signal. However, in case of lossy based lossless coding the additional amount of information exceeds the amount of data for the base layer by a multiple of the base layer data amount. Therefore the additional data cannot be packed completely into the base layer data stream e.g. as ancillary data. The at least two data streams resulting from the combination of lossy coding format with a lossless coding extension are the base layer containing the lossy coding information and the enhancement data stream for rebuilding the mathematically lossless original input signal.

Abstract: The present invention provides method and device for transcoding between audio coding formats with different time-frequency analysis domains, as used for example by MPEG-AAC and mp3, particularly for facilitated and faster transcoding between such audio signals. A method for transcoding a framed audio signal from a first parameter domain into a second parameter domain comprises linearly transforming two or more parameters of the first parameter domain to at least one parameter of the second parameter domain, wherein the two or more parameters of the first parameter domain come from different frames of the audio signal in the first parameter domain. The linear transformation can be described as a matrix and implemented as a look-up table.