Thomson Licensing

Deutsche Thomson-Brandt Gmbh

Jürgen Schmidt

Jens Spille

Andreas Aust

Ernst F Schröder

Johannes Böhm

Klaus Eilts-Grimm

Imre Varga

A method for decoding a data stream containing audio/video substreams and control substreams comprises buffering nodes having the possibility to buffer multiple data packets in the same buffer. This may be achieved by having separate parameters for the allocated buffer size and any stored packet. Thus, not only multiple packets may be stored in the buffering node, but also such node may exist while its buffer is empty, so that the node may be reused later. This is particularly useful for buffering and selectively accessing multiple audio packets in MPEG-4 audio nodes or sound nodes.

Method and apparatus for decoding a data stream in audio video streaming systems

Method for describing the composition of audio signals, which are encoded as separate audio objects. The arrangement and the processing of the audio objects in a sound scene is described by nodes arranged hierarchically in a scene description. A node specified only for spatialization on a 2D screen using a 2D vector describes a 3D position of an audio object using said 2D vector and a 1D value describing the depth of said audio object. In a further embodiment a mapping of the coordinates is performed, which enables the movement of a graphical object in the screen plane to be mapped to a movement of an audio object in the depth perpendicular to said screen plane.

Method for describing the composition of audio signals

A parametric description describing the wideness of a non-point sound source is generated and linked with the audio signal of said sound source. A presentation of said non-point sound source by multiple decorrelated point sound sources at different positions is defined. Different diffuseness algorithms are applied for ensuring a decorrelation of the respective outputs. According to a further embodiment primitive shapes of several distributed uncorellated sound sources are defined, e.g. a box, a sphere and a cylinder. The width of a sound source can also be defined by an opening-angle relative to the listener. Furthermore, the primitive shapes can be combined to do more complex shapes.

Method for coding and decoding the wideness of a sound source in an audio scene

In the MPEG-4 standard ISO/IEC 14496:2001 several audio objects that can be coded with different MPEG-4 format coding types can together form a composed audio system representing a single soundtrack from the several audio substreams. In a receiver the multiple audio objects are decoded separately, but not directly played back to a listener. Instead, transmitted instructions for mixdown are used to prepare a single soundtrack. Mixdown conflicts can occur in case the audio signals to be combined have different channel numbers or configurations. According to the invention an additional audio channel configuration node is used that tags the correct channel configuration information items to the decoded audio data streams to be presented. The invention enables the content provider to set the channel configuration in such a way that the presenter at receiver side can produce a correct channel presentation under all circumstances. An escape code value in the channel configuration data facilitates correct handling of not yet defined channel combinations.

Method and apparatus for processing two or more initially decoded audio signals received or replayed from a bitstream

The transmission and use of real, i. e. of measured, room impulse responses for the reproduction of sound signals with this room characteristic compatible to the MPEG-4 standard is made possible by inserting impulse responses in multiple successive control parameter fields, especially the params[128] array. A first control parameter field contains information about the number and content of the following fields. For presentation of the sound signals the content of the successive control parameter fields is separated, stored in an additional memory of a node and used during the calculation of the room characteristic.

Method for coding and decoding impulse responses of audio signals

A method for improving timing behavior of a processing unit in a multithreading environment is disclosed, wherein the processing unit generates data frames for an output unit by combining data from a plurality of input units, and the processed data are buffered in an output buffer between the processing unit and the output unit. The method comprises sending from the output unit to the processing unit a value corresponding to the filling of the output buffer, calculating a timer value, setting a timer with the timer value, wherein the timer calls the processing unit thread after the specified time. The timer value depends on the value corresponding to the averaged filling of the output buffer. As a result, the average filling of the output buffer is lower compared to conventional thread management, and thus the system is more flexible and reacts quicker.

Method and apparatus for processing data in a processing unit being a thread in a multithreading environment

The reproduction of mono audio signals leads to an unpleasant sound impression because of the lack of any spatial character. It is therefore desirable to produce a two-channel signal electrically from the single-channel input signal. Some of the known single-band methods are very costly and provide only an inadequate impression of a spatial character. In the case of the invention, a plurality of signals of different types are first of all formed from the mono input signal by filtering, and virtual single-band stereo signals are then generated separately for each of these signals of different types. These stereo signals are subsequently combined to form two output signals.

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Wunstorf, Germany

Jürgen Schmidt