Abstract: A camera module includes a substrate with a cavity therein. A processor is located in the cavity, and wire bonding is for connecting the processor to the substrate. An imaging module is adapted to overlay the processor in the cavity and rest on at least part of the edge of the cavity. Wire bonding is for connecting the imaging module to the substrate and the processor. The cavity includes a longitudinal cutout section adapted to accommodate at least some wire bonding for connecting the processor to the substrate or associated surface mount components.

Abstract: A method and apparatus control bit rates used in a moving pictures encoder, such as an MPEG standard encoder. A sequence of moving pictures is divided into segments each of which comprises one or more groups of pictures. A constant overall bit rate is specified for the sequence of pictures, but variable bit rate encoding is used within each segment. A difference between the number of bits allocated for encoding the segment and the actual bits used for encoding is determined, and the difference is distributed over one or more subsequent segments.

Abstract: A method and apparatus control bit rates used in a moving pictures encoder, such as an MPEG standard encoder. A sequence of moving pictures is divided into segments each of which comprises one or more groups of pictures. A constant overall bit rate is specified for the sequence of pictures, but variable bit rate encoding used within each segment. A difference between the number of bits allocated for encoding the segment and the actual bits used for encoding is determined, and the difference distributed over one or more subsequent segments.

Abstract: A method and apparatus for encoding pictures of a moving pictures sequence according to an overall target bit-rate, such as in a MPEG video encoder. Each picture has an assigned picture coding type for which a quality factor is adaptively determined according to past bit usages, so that bits can be adaptively allocated amongst picture types for optimizing visual quality of the encoded pictures.

Abstract: In a transform encoder for audio data, encoded data in the form of mantissas, exponents and coupling data is packed into fixed length frames in an output bitstream. The fields within the frame for carrying the different forms of data are variable in length, and apace within the frame must be allocated between them to fit all of the required information into the frame. The space required by the various data types depends on certain encoding parameters, which are calculated for a particular frame before the data is encoded, thus ensuring that the encoded data will fit into the frame before the computationally expensive encoding process is carried out. Information in relation to, for example, transform length, coupling parameters and exponent strategy are determined, which allows the space required for the coupling and exponent data to be calculated. The mantissa encoding parameters can then be iteratively determined so that the encoded mantissas will fit into the frame with the other encoded data.

Abstract: A method and apparatus for coding audio data in a frequency transform digital audio coder employing differential frequency coefficient exponent coding. Differential coding of exponents places constraints on possible values an exponent can take, which can lead to distortion in the decoded and reconstructed audio signal. The method and apparatus herein can overcome this restriction by mapping the input exponent set to a new set of values which satisfy the differential constraint as well as reducing information loss, thereby minimizing overall signal distortion due to coding restrictions.

Abstract: A method of parametrically encoding a transient audio signal, including the steps of: determining a set V of the N largest frequency components of the transient audio signal, where N is a predetermined number; determining an approximate envelope of the transient audio signal; and determining a predetermined number P of samples W of the approximate envelope for use in generating a spline approximation of the approximate envelope, whereby a parametric representation of the transient audio signal is given by parameters including V, N, P and W, such that a decoder receiving the parametric representation can reproduce a received approximation of the transient audio signal.

Abstract: An efficient architecture for a rake combiner is disclosed, for constructively combining the desired multi-path signals from a Code-Division Multiple-Access (CDMA) based system, such as a Third-Generation Partnership Project (3GPP) Frequency Division Duplex (FDD) mode Wideband CDMA (W-CDMA) system, or an IS-95 CDMA system. The described rake combiner employs a single M-stage tap-delay line, an N+1 input adder, an arrangement of index offsets, pass gates, comparators and an M-stage counter to perform the combination, where M represents the delay spread in terms of symbol duration and N represents the number of rake fingers to be combined. The rake combiner architecture facilitates lowered resource requirements through use of a single tap-delay line in contrast to a conventional rake combiner which uses a series of M-stage tap-delay lines and an N input adder to perform the combination.

Abstract: A method of dividing, in a micro computer unit (MCU), a first binary number (N), having a first number of significant bits, by a second binary number (D), having a second number of significant bits, produces an integer result (Y). The method includes: determining the difference (K) between the first and second numbers of significant bits; aligning the most significant bits (MSBs) of N and D by shifting the bits of D, by K bit positions, such that its MSB occupies the same relative bit position as the MSB of N; repeating K times: multiplying Y by 2; dividing D by 2; and, if N is greater than or equal to D: increasing Y by 1; setting N equal to N−D.

Abstract: A method and apparatus for determining parameters for coupling of channels in a digital audio encoder. A frequency range of two audio channels is coupled together in a coupling channel, and a systematic method of determining optimum coupling parameters is employed. Sub-bands of the channels are processed individually, and a measure of the power of each sub-band is used to determine a coupling coefficient generation scheme for each individual sub-band. Adjacent ones of the sub-bands using the same coupling scheme are combined to form bands in the coupling channel, which dictate the generation of the coupling coordinates for the audio channels. The arrangement of the sub-bands in bands also facilitates the generation of phase flags for each band, on the basis of the coupling scheme used in the band.

Abstract: A method and apparatus for subband phase flag determination for coupling of channels in a dual channel audio encoder is based on a psychoacoustic model of the human auditory system. The method and apparatus are applicable to audio encoders which utilize a coupling channel to combine certain frequency components of the input audio signals. The method ensures a least square error between the original channel frequency coefficients at the encoder and the estimated coefficients at the decoder by determining the sign of the dot product of the coefficients for one of the channels and the coupling coefficients. No restriction is placed on the strategy utilized for generating the coupling channel coefficients or the coupling coordinates.

Abstract: A method and apparatus for decoding a bitstream (100) of transform coded multi-channel audio data. The bitstream is subjected to a block decoding process (101) to obtain for each input audio channel within the multi-channel audio data a corresponding block of frequency coefficients (102). Each block of frequency coefficients (102) is assigned a higher precision inverse transform or a lower precision inverse transform according to predetermined characteristics of the audio data represented by the block. The blocks of frequency coefficients are subsequently subjected to the assigned transform (105, 106) and an output audio signal (108) is generated in response to each of the higher and lower precision inverse transform processes.

Abstract: A method and apparatus for decoding a multi-channel audio bitstream in which adaptive frequency domain downmixer (3) is used to downmix, according to long and shorter transform block length information (17), the decoded frequency coefficients of the multi-channel audio (12,13,14,15) such that the long and shorter transform block information is maintained separately within the mixed down left and right channels. In this way, the long and shorter transform block coefficients of the mixed down let and right channels can be inverse transformed adaptively (4,5,6,7) according to the long and shorter transform block information, and the results of the inverse transform of the long and short block of each the left and right channel added together (8,9) to form the total mixed down output of the left and right channel.

Abstract: This invention relates to the color matching function used in connection with the luminance or white stretch function in a television video processor. Automatic adjustment in the amplitude of the color difference signals is provided to compensate for the effect of the stretch of the luminance signal in a provision called color matching. The general principal of color matching being, any percentage change in the amplitude of the luminance signal due to the white stretch effect, must be balanced by the same percentage changes in the color difference signal so that the ratio of the color signals can be maintained after matrixing. This accomplished by compensating the color difference signals by a varying amount that decreases with increasing the input luminance signal level when the level of the input luminance signal is above said selected threshold.