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 MP EG-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, wherein the matrix transformation coefficients below -60 Decibels are neglected.

Abstract: A method for encoding multi-channel HOA audio signals for noise reduction comprises steps of decorrelating the channels using an inverse adaptive DSHT, the inverse adaptive DSHT comprising a rotation operation and an inverse DSHT, with the rotation operation rotating the spatial sampling grid of the iDSHT, perceptually encoding each of the decorrelated channels, encoding rotation information, the rotation information comprising parameters defining said rotation operation, and transmitting or storing the perceptually encoded audio channels and the encoded rotation information.

Abstract: Conventional audio compression technologies perform a standardized signal transformation, independent of the type of the content. Multi-channel signals are decomposed into their signal components, subsequently quantized and encoded. This is disadvantageous due to lack of knowledge on the characteristics of scene composition, especially for e.g. multi-channel audio or Higher-Order Ambisonics (HOA) content. An improved method for encoding pre-processed audio data comprises encoding the pre-processed audio data, and encoding auxiliary data that indicate the particular audio pre-processing. An improved method for decoding encoded audio data comprises determining that the encoded audio data had been pre-processed before encoding, decoding the audio data, extracting from received data information about the pre-processing, and post-processing the decoded audio data according to the extracted pre-processing information.

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: A method and an apparatus for storing data as a coded two-dimensional pattern of pixels on a photographic medium are described. An image representative of the data to be stored is divided into a two-dimensional array of blocks. A representative color value is determined for each block to obtain a two-dimensional array of representative color values. At least part of the data to be stored is coded such that the resulting coded two-dimensional pattern of pixels resembles the two-dimensional array of representative color values. The coded two-dimensional pattern of pixels is printed onto the photographic medium.

Abstract: A method and an apparatus for archiving an image on a photographic medium are described. The image consists of an array of pixels, where a color depth of the pixels is n bits. In addition to recording the array of pixels on the photographic medium also a subset of the n bits describing the color of the pixel is recorded on the photographic medium for each pixel.

Abstract: With quantisation index modulation QIM it is possible to achieve a very high data rate, and the capacity of the watermark transmission is mostly independent of the characteristics of the original audio signal, but the audio quality suffers from degradation with each watermark embedding-and-removal step. In order to avoid degradation of the audio quality, the inventive audio signal watermarking uses specific quantiser curves in time domain and in particular in frequency domain for embedding the watermark message into the audio signal, whereby the processing is almost perfectly reversible. Furthermore, it has embedded a power constraint in order to guarantee that the modifications of the audio signal due to the watermark embedding are inaudible.

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: Higher-order Ambisonics HOA is a representation of spatial sound fields that facilitates capturing, manipulating, recording, transmission and playback of complex audio scenes with superior spatial resolution, both in 2D and 3D. The sound field is approximated at and around a reference point in space by a Fourier-Bessel series. The invention uses space warping for modifying the spatial content and/or the reproduction of sound-field information that has been captured or produced as a higher-order Ambisonics representation. Different warping characteristics are feasible for 2D and 3D sound fields. The warping is performed in space domain without performing scene analysis or decomposition. Input HOA coefficients with a given order are decoded to the weights or input signals of regularly positioned (virtual) loudspeakers.

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: To form an audio signal, frequency components of the audio signal which are allotted to a first subband are formed by means of a subband decoder using supplied fundamental period values which respectively indicate a fundamental period for the audio signal. Frequency components of the audio signal which are allotted to a second subband are formed by exciting an audio synthesis filter using an excitation signal which is specific to the second subband. To produce this excitation signal, an excitation signal generator derives a fundamental period parameter from the fundamental period values. The fundamental period parameter is used by the excitation signal generator to form pulses with a pulse shape which is dependent on the fundamental period parameter at an interval of time which is determined by the fundamental period parameter and to mix them with a noise signal.

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: An advantage of Ambisonics representation is that the reproduction of the sound field can be adapted individually to nearly any given loudspeaker position arrangement. The invention allows systematic adaptation of the playback of spatial sound field-oriented audio to its linked visible objects, by applying space warping processing as disclosed in EP 11305845.7. The reference size (or the viewing angle from a reference listening position) of the screen used in the content production is encoded and transmitted as metadata together with the content, or the decoder knows the actual size of the target screen with respect to a fixed reference screen size. The decoder warps the sound field in such a manner that all sound objects in the direction of the screen are compressed or stretched according to the ratio of the size of the target screen and the size of the reference screen.

Abstract: An integer-reversible MDCT transformation is split into consecutive lifting steps, each introducing considerable rounding errors to the signal. Without noise shaping the rounding error noise will impact all frequency bins of the transformed signal equally. This is a particular problem for low signal level frequency bins. The invention limits the impact of rounding error noise coming with each lifting step in the integer-reversible transformation on the data rate of a lossless audio codec. The filter coefficients of an adaptive noise shaping filter for transform coefficients are adapted in individual lifting steps according to the current time domain signal characteristics. As an alternative, an auto-regressive pre-filter can be added in front of the lossless transformation, for raising the level of frequency regions with low power to decrease the dominance of rounding errors in these areas. Both processes can be combined to further improve lossless codec compression ratio.

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: At the time of encoding audio content, the finally required data rate for delivery to the customer may be unknown. A data format is disclosed that is optimized for serving as Intermediate Format for efficient and fast recoding, to obtain one or more standard complying lossy encoded data streams with flexible data rates. Encoding can be performed in two steps that are inter-coordinated for cooperating, but may be locally and/or temporally separate. Between the partial encoders encoding parameters and/or auxiliary data are transmitted in a separate parameter enhancement layer, which complements a lossy data stream and can be used by the second encoder or transcoder for fast and computationally efficient implementation of the second encoding step. An additional lossless enhancement layer allows lossless reconstruction.

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: 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: 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.