Abstract:
A method for effecting aliasing cancellation in an audio effects algorithm using a delay modulated signal, derived from interpolation of a delay modulator at an instantaneous sampling frequency, including: determining the instantaneous sampling frequency 1/Tisf and band limiting an input signal, to which the audio effects algorithm is to be applied to ½ Tisf prior to interpolation.
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. The determined encoding parameters are stored and the audio data is encoded according to those parameters after it has been determined that the encoded data will fit into the frame.
Abstract:
AC-3 is a high quality audio compression format widely used in feature films and, more recently, on Digital Versatile Disks (DVD). For consumer applications the algorithm is usually coded into the firmware of a DSP Processor, which due to cost considerations may be capable of only fixed point arithmetic. It is generally assumed that 16-bit processing is incapable of delivering the high fidelity audio, expected from the AC-3 technology. Double precision computation can be utilized on such processors to provide the high quality; but the computational burden of such implementation will be beyond the capacity of the processor to enable real-time operation. Through extensive simulation study of a high quality AC-3 encoder implementation, a multi-precision technique for each processing block is presented whereby the quality of the encoder on a 16-bit processor matches the single precision 24-bit implementation very closely without excessive additional computational complexity.
Abstract:
A method and apparatus for assigning an exponent coding strategy in a digital audio transform coder. Different coding strategies having different differential coding limits may be assigned to different set of transform exponents according to the frequency domain characteristics of the audio signal. A neural network processing system is utilised to perform an efficient mapping of each exponent set to an appropriate coding strategy.
Abstract:
A technique for computationally efficient evaluation of the Modified Discrete Cosine Transform (MDCT) using the Fast Fourier Transform (FFT) method is presented in which the input of the FFT block consists of a sequence of N complex numbers, and this complex data is evaluated using an N/2-Point FFT only, thereby descreasing computation burden almost by two.
Abstract:
A method for conversion of input audio frequency data, at an input sample frequency, to output audio frequency data, at an output sample frequency. The input data is subjected to expansion to produce expanded data at an output sample frequency. The expanded data is interpolated to produce output data. In one embodiment of the invention the interpolation is effected by a process that also filters the output data. In another embodiment, the input data is sampled by an integer factor to produce expanded data, the expanded data is then interpolated to produce the output data. Also disclosed is a method of transition of a signal output, at one frequency, to a signal output at another frequency. The signal output at said one frequency is faded out over a period, and the signal output at said other frequency is faded in over that period. Both signal outputs are combined to produce the signal output over said period. Apparatus for effecting the methods is also disclosed.
Abstract:
AC-3 is a high quality audio compression format widely used in feature films and, more recently, on Digital Versatile Disks (DVD). For consumer applications the algorithm is usually coded into the firmware of a DSP Processor, which due to cost considerations may be capable of only fixed point arithmetic. Commercial AC-3 Encoders have been successfully implemented on 20-bit and 24-bit word-length processors. However, it is generally assumed that 16-bit processing is incapable of delivering the high fidelity audio, expected from the AC-3 technology. Double precision computation can be utilised on such processors to provide the high quality; but the computational burden of such implementation will be beyond the capacity of the processor to enable real-time operation. Through extensive simulation study of a high quality AC-3 Encoder implementation, a multi-precision technique for each processing block is presented whereby the quality of the encoder on a 16-bit processor matches the single precision 24-bit implementation very closely without excessive additional computational complexity.