Thomson Licensing

Thomson Licensing S.a.

Deutsche Thomson-Brandt Gmbh

Ulrich Schreiber

Markus Schlosser

Ulrich Gries

Mario Sieck

Peter Georg Baum

Peter Steinborn

Walter V&ouml;ssing

Markus Schneider

Stephane Chouquet

A circuit for correcting the output of an audio level meter comprises input means for generating the square or the absolute value of an input signal, a low pass filter having a predetermined attack time and release time, and output means for converting the output signal from a linear scale to a logarithmic scale. The circuit further comprises a correction means to which an information about whether the input signal to the audio level meter was subject to squaring or converting into an absolute value at the input, as well as the attack and release time of the low pass filter, are supplied as input values, and which provides, at its output, a value representing the difference between the output of the audio level meter and the true signal power of the input signal.

Audio level meter

Watermark information (denoted WM) consists of several symbols which are embedded continuously by reference sequence modulation in an audio or a video signal. At decoder site the WM is regained using correlation of the received signal with a corresponding reference sequence. The symbols form watermark data frames. The invention uses for the bit values 'zero' and 'one' in each payload symbol and for each payload symbol in a watermark data frame different reference sequences, without using synchronisation symbols. A logarithmic search is performed in the WM decoder to reduce the numbers of correlations to be calculated. The invention makes watermarking of critical sound signals much more robust.

Method and apparatus for encoding/decoding symbols carrying payload data for watermarking of an audio or video signal

The invention relates to the technical field of data transmission via a transmission link according to the TDM method, corresponding to time division multiplex. Such a method is used in an audio mixing console in order to transport the sample words of a plurality of audio channels jointly to an audio DSP processor and, after processing, to transmit them back from said processor to an output processor. Since the data output process is performed in an automated manner with the aid of a DMA unit, severe interfering noise can occur in cases of error, said noise also being output via the loudspeakers. The invention is based on the aim of avoiding such interfering effects. An error detection method based on the marking of data frames is used for this. For this purpose, a number of marking bits of the time slot of the data frame are set, the bit pattern of the marking bits being alternated in a defined sequence. Upon error detection, the outputting of the data via the loudspeakers is prevented, such that no interfering noise arises.

Data transmission method and device for carrying out the method

In audio processing technology there are interfaces of different type, the different interface types being associated with differing sample clock frequencies and/or differing data frame or data word formats. In some applications it is desired do process audio signals that stem from such interfaces of different type. According to the invention the input signals of the interfaces of different type are pre-processed for a subsequent common-format central processing, using a synchronization to the common system clock, channel decoding according to the differing channel protocols related to the different-type interfaces, and processing the resulting PCM bitstream format signals so as to form therefrom sample words that are stored in an intermediate store from which the sample words are fed to the central processing.

Method and apparatus for pre-processing in a common-format control processing input signals of, or output signals for, interfaces of different type

For TV broadcasting purposes encoding systems with related video encoders and audio encoders are used. The audio and video delay are aligned before multiplexing and transmitting the audio and video streams. According to a time stamping mechanism input time stamps are generated which become linked with data to be encoded, and are replaced before output by output time stamps, which are derived from the input time stamps by using a data delay constant. The input time stamps are used to control the delay of the encoding process and the output time stamps are indicating the output time. In order to allow a switchable out-put delay the data delay constant can be changed. Already assigned output time stamps remained unchanged. For data for which output time stamps are not already assigned, the out-put time stamps are calculated using the new delay constant.

Method and apparatus for changing the output delay of audio or video data encoding

For broadcasting purposes a multi-channel audio encoder board has been designed. Input time stamps are generated which become linked at least in one input processing stage with frames of said audio data to be encoded, wherein the input time stamps or time stamps derived from the input time stamps remain linked with the correspondingly processed frame data in the different processing stages in the processing, but are at least in the last processing stage replaced by output time stamps. In each of theses stages the corresponding time stamp information linked with current frame data to be processed is regarded in order to control the overall delay of the processing.

Method for processing and apparatus for encoding or decoding audio or video frame data

For broadcasting purposes a multi-channel audio encoder board has been designed. A requirement for such encoders is that they are able to operate with different encoding parameters. It may happen that encoding parameters change during encoding operation. In order to avoid the output of invalid data the encoding parameters required for a specific processing path are added to the input streams for the audio channels and become linked with the associated audio data and are stored in various buffers together with its audio data, i.e. the corresponding encoding parameters are kept linked with the audio data to be encoded throughout the encoding processing.

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Hohenhameln/Equord, Germany

Ulrich Schreiber