Abstract: Movies are produced in 24 Hz frame frequency and progressive scanning format (denoted 24p) for projection in film theatres, adhering to a worldwide standard for 35 mm film. However, the major TV systems in the world use interlaced scanning and either 50 Hz field frequency (denoted 50i) or 60 Hz field frequency (denoted 60i). Content providers would prefer providing single-picture-frequency single-audio-speed AV discs that can be replayed in most parts of the world. According to the invention, For a 50 HZ output mode, in the media player either audio signal frames are dropped adaptively or video fields or frames are repeated adaptively, depending on the current video and audio content. Thereby the less perceptible stream controls the synchronization.

Abstract: 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.

Abstract: Movies are produced in 24 Hz frame frequency and progressive scanning format (denoted 24p) for projection in film theatres, adhering to a worldwide standard for 35 mm film. However, the major TV systems in the world use interlaced scanning and either 50 Hz field frequency (denoted 50i) or 60 Hz field frequency (denoted 60i). Content providers would prefer providing single-picture-frequency single-audio-speed AV discs that can be replayed in most parts of the world. According to the invention, For a 50 HZ output mode, in the media player either audio signal frames are dropped adaptively or video fields or frames are repeated adaptively, depending on the current video and audio content. Thereby the less perceptible stream controls the synchronisation.

Abstract: With audio data reduction on the basis of ISO/IEC standard 11172-3, a frame length varying by 8 bits is used at a sampling frequency of 44.1 kHz in order to arrive, on average, at a particular fixed data rate. The lengthening of a data frame is signalled by a padding bit in the header of the frames. The invention dispenses with evaluation of the padding bit. Instead, the mean frame length L is calculated, L is rounded down to the next integer, for the subsequent frame it is first established whether the expected sync word for this frame appears, and, if this is so, this frame is decoded without taking into account the padding bit, but if the expected sync word for this frame does not appear, the decoding of the frame is started one 8-bit later without taking into account the padding bit.

Abstract: An mp3-standard bitstream is formatted into a sequence of fixed-length data frames. These include headers, side information, main information and a remaining data field without generally defined information denoted as ‘ancillary data’. The mp3PRO format is an extension of the mp3 format, wherein the additional mp3PRO data are transferred in the ancillary data fields. In various applications, e.g. Internet music search machines, a necessity arises for a fast determination of the bitstream types. Such determination is normally executed using an mp3PRO decoder. However, because the frame header does not contain a corresponding pointer to the start address of the ancillary data field, an mp3PRO decoder must first completely decode at least one data frame according to the mp3 standard in order to find the end address of the mp3 data and thereby the following start address of the mp3PRO data in that data frame.

Abstract: In the voice control of a device appertaining to consumer electronics, speech inputs of a user are converted into digitized voice signals. First features, which are characteristic of the individual sounds of the speech, and thus permit recognition of the spoken sounds, are extracted from the digitized voice signals. Furthermore, second features, which permit a characterization of the voice of the respective user and are used for distinguishing between the speech inputs of different users, are extracted from the digitized voice signals. This enables the device being operated to distinguish between speech inputs of the present user, which represent operator-control commands, and utterances by other persons, and to respond only to operator-control commands of the present user. Furthermore, the quality of an analysis of the viewing habits of different users can be improved by a distinction between the users.

Abstract: The invention specifies a system having a plurality of devices which are connected to one another via an IEEE 1394 interface and one of which contains a control unit which, when operated appropriately by a user, polls system data for devices in this system via the interface and passes this system data to an output unit of this device. The device having the output unit is, by way of example, a set-top box having a microprocessor which a user uses to poll system data for the devices, which contains, in particular the input and output characteristics of the latter, via the interface and which the user can use to store this system data on a smart card by means of a write/read device. Alternatively or at the same time, the system data can be shown on a display or transmitted to a desired address via a modem connection.

Abstract: Original digital audio signals are represented as PCM sample values wherein the distance between the values corresponds to the sampling frequency. Digital signals can have a length that is an integer multiple only of this time element. In particular coded digital audio signals are processed block-based, leading to a total length that is a multiple only of the block unit. According to the invention, information about the exact length of the original signal is transferred together with the encoded audio information. Additionally, an information value can be transferred that represents the total encoder and/or decoder delay. The decoder extracts these items of information and adjusts the total length of the decoded signal by cutting off samples from the decoded program or track.

Abstract: Voice control systems are used in diverse technical fields. In this case, the spoken words are detected by one or more microphones and then fed to a speech recognition system. In order to enable voice control even from a relatively great distance, the voice signal must be separated from interfering background signals. This can be effected by spatial separation using microphone arrays comprising two or more microphones. In this case, it is advantageous for the individual microphones of the microphone array to be distributed spatially over the greatest possible distance. In an individual consumer electronics appliance, however, the distances between the individual microphones are limited on account of the dimensions of the appliance.

Abstract: Generally, performing sampling-rate conversion from a higher sampling frequency fs1 to a lower sampling frequency fs2 results in aliasing. It is known to use a low-pass filter, known as anti-alias filter, for avoiding this alias distortion. Its effect is to remove spectral contents above fs2/2 from the digital signal. According to the invention these signal parts are suppressed at spectral decoding resulting in a bandwidth of the signal to be re-sampled which is less than half of the second sampling frequency fs2. This can be done for MPEG encoded audio signals by limiting the decoding to a certain number of subbands, for DOLBY AC-3 encoded audio signals by setting certain spectral lines to zero at decoding. The inventive method not only totally removes the processing power needed for calculating an anti-alias filter, but also limits the decoding work needed.

Abstract: Dolby AC-3 and MPEG-2 audio permit the transmission of audio signals with more than two independent audio channels. If a reproduction device has only a two-channel audio decoder (DEC1) then an external multi-channel audio decoder (DEC2) can be used for multi-channel sound reproduction. If the audio reproduction at the same time accompanies video reproduction, then a synchronization method is required in order to achieve lip synchronism between picture and sound. According to the invention, for the synchronization of a first decoder (DEC1), which merely has two-channel compatibility, with a second decoder, which has multi-channel compatibility, a counting variable is allocated a value (F) which is produced from system parameters such as the data coding method used, the transmission speed and/or the data rate. The data are received by the first decoder and output by the latter to the second decoder, the counting variable being decremented or incremented respectively for a specific volume of data.

Abstract: A magnetic tape apparatus for the recording and reproduction of signals, including a magnetic tape having plural data tracks running parallel to an edge of the tape; a multi-track reading device including a multi-track playback head configured to read simultaneously plural of the tracks and output signals corresponding to data recorded on the plural tracks; and a housing in which the tape and the reading device are permanently disposed; wherein the tape is adapted for simultaneous recording of data onto plural data tracks of the tape via external recording heads upon the apparatus being connected to an external recording facility.