The auditory system encodes sound by decomposing the amplitude signal arriving at the ear into multiple frequency bands whose center frequencies and bandwidths are approximately exponential functions of the distance from the stapes. This organization is thought to result from the adaptation of cochlear mechanisms to the animal's auditory environment. Here we report that several basic auditory nerve fiber tuning properties can be accounted for by adapting a population of filter shapes to encode natural sounds efficiently. The form of the code depends on sound class, resembling a Fourier transformation when optimized for animal vocalizations and a wavelet transformation when optimized for non-biological environmental sounds. Only for the combined set does the optimal code follow scaling characteristics of physiological data. These results suggest that auditory nerve fibers encode a broad set of natural sounds in a manner consistent with information theoretic principles.

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Efficient coding of natural sounds.

Digital coding of waveforms: Principles and applications to speech and video

Sound processing methods and apparatus are provided. A sound capture unit is configured to identify one or more sound sources. The sound capture unit generates data capable of being analyzed to determine a listening zone at which to process sound to the substantial exclusion of sounds outside the listening zone. Sound captured and processed for the listening zone may be used for interactivity with the computer program. The listening zone may be adjusted based on the location of a sound source. One or more listening zones may be pre-calibrated. The apparatus may optionally include an image capture unit configured to capture one or more image frames. The listening zone may be adjusted based on the image. A video game unit may be controlled by generating inertial, optical and/or acoustic signals with a controller and tracking a position and/or orientation of the controller using the inertial, acoustic and/or optical signal.

Methods and apparatus for targeted sound detection and characterization

There is disclosed a method, apparatus and computer program product in which at least one view component of a first type and at least one view component of a second type are obtained. The order of the texture view component and the depth view component in an access unit is determined and at least one indication of the order is encoded. The coding of the view components is adapted on the basis of the order. There is also disclosed a method, apparatus and computer program product in which at least one encoded view component of a first type and at least one encoded view component of a second type are received. Also at least one encoded indication of the order of the view components is received. The at least one encoded indication is decoded and the decoding of the view component sis adapted on the basis of the order.

Method and apparatus for video coding

Methods and apparatus for embedding digital watermarks into a digital host content are provided. A digital host content is received, e.g., at a receiver or other device incorporating a receiver. One or more watermark embedding technologies is selected. Multiple embedding opportunities are identified within the host content. A subset of the identified embedding opportunities is selected. A multiplicity of digital watermarks are then embedded into the host content in accordance with the selected subset of embedding opportunities utilizing the one or more selected watermark embedding technologies. The selecting of the subset of embedding opportunities may be adapted to provide a desired tradeoff between levels of robustness, security, and transparency of the watermark. A plurality of watermarking embedding technologies may be selected and used in the embedding step.

Security enhancements of digital watermarks for multi-media content

An apparatus for scalable encoding a spectrum of a signal including audio and/or video information, with the spectrum comprising binary spectral values, includes a means for generating a first sub-scaling layer and a second sub-scaling layer in addition to a means for forming the encoded signal, with the means for forming being implemented so as to include the first sub-scaling layer and the second sub-scaling layer into the encoded signal that the first and the second sub-scaling layer are separately decodable from each other. In contrast to a full-scaling layer, a sub-scaling layer includes only the bits of a certain order of a part of the binary spectral values in the band, so that, by additionally decoding a sub-scaling layer, a more finely controllable and a more finely scalable precision gain may be achieved.

Method and apparatus for scalable encoding and method and apparatus for scalable decoding

An Audio decoder for decoding a multi-audio-object signal having an audio signal of a first type and an audio signal of a second type encoded therein is described, the multi-audio-object signal consisting of a downmix signal (56) and side information (58), the side information comprising level information (60) of the audio signal of the first type and the audio signal of the second type in a first predetermined time/frequency resolution (42), and a residual signal (62) specifying residual level values in a second predetermined time/frequency resolution, the audio decoder comprising means (52) for computing prediction coefficients (64) based on the level information (60); and means (54) for up-mixing the downmix signal (56) based on the prediction coefficients (64) and the residual signal (62) to obtain a first up-mix audio signal approximating the audio signal of the first type and/or a second up-mix audio signal approximating the audio signal of the second type.

Audio coding using downmix

The invention relates to a method for embedding a watermark in an audio signal, according to which a spectral representation of the audio signal and a spectral representation of the watermark signal are first determined (14, 16). The spectral representation of the watermark signal is then processed, based on a psychoacoustic masking threshold (24) of the audio signal (22). The processed watermark signal is combined with the audio signal (18). The spectral representation of the watermark signal is processed iteratively in the following way: a predetermined watermark initial value (26) is first selected; the interference that has been introduced into the spectral representation by the predetermined watermark initial value after a quantization of the spectral representation of the audio signal, is determined (28). The watermark initial value is then modified until said interference introduced into the spectral representation of the audio signal by a modified watermark initial value after quantization, is less than or equal to the predetermined interference threshold (32).

Device and method for embedding a watermark in an audio signal

As the latest extension of MPEG-4 Audio coding, MPEG-4 Lossless Audio Coding includes a scalable audio coding solution (SLS) that integrates the functionalities of lossless audio coding, perceptual audio coding, and fine granular scalable audio coding into a single coder framework while providing backward compatibility to MPEG Advanced Audio Coding (AAC) at the bit-stream level. Despite its abundant functionalities, SLS still achieves a compression performance that is comparable to state-of-the-art non-scalable lossless audio coding algorithms. As a result, SLS provides a universal digital audio format for a variety of application domains including professional audio, Internet music, consumer electronics, broadcasting and others. This paper presents the structure of SLS and its latest developments during the MPEG standardization process.

/pdf/mpeg-4-scalable-to-lossless-audio-coding-3i0kvmdh99.pdf

MPEG-4 Scalable to Lossless Audio Coding

A time-discrete audio signal is processed to provide a quantization block with quantized spectral values. Furthermore, an integer spectral representation is generated from the time-discrete audio signal using an integer transform algorithm. The quantization block having been generated using a psychoacoustic model is inversely quantized and rounded to then form a difference between the integer spectral values and the inversely quantized rounded spectral values. The quantization block alone provides a lossy psychoacoustically coded/decoded audio signal after the decoding, whereas the quantization block, together with the combination block, provides a lossless or almost lossless coded and again decoded audio signal in the decoding. By generating the differential signal in the frequency domain, a simpler coder/decoder structure results.

Juergen Herre

Papers

Method and apparatus for scalable encoding and method and apparatus for scalable decoding

Audio coding using downmix

Device and method for embedding a watermark in an audio signal

MPEG-4 Scalable to Lossless Audio Coding

Apparatus and method for coding a time-discrete audio signal and apparatus and method for decoding coded audio data