Abstract: Perceptual audio coder refers to audio compression schemes that exploit the properties of human auditory perception. The coder allocates the quantization noise below the masking threshold such that even with the bit rate limitation, the noise is imperceptible to the ear. These distortion and bit rate requirement makes the bit allocation-quantization process a considerable computational effort. One method includes incrementally adjusting a global gain according to a gradient. The gradient could be adjusted each time the number of bits used to represent a quantized value is counted. Another method includes limiting a rate controlling parameter to a predetermined number of loops. The method could also include deriving a global gain to ensure exit from the loop. Accordingly, embodiments of the present disclosure provide a fast and efficient method to derive the rate controlling parameter and can be applied to generic perceptual audio encoders where low computational complexity is required.

Abstract: A system and method for adaptive rate control in audio processing is provided. The process could include receiving uncompressed audio data from an input and generating MDCT spectrum for each frame of the uncompressed audio data using a filterbank. The process could also include estimating masking thresholds for current frame to be encoded based on the MDCT spectrum. The masking thresholds reflect a bit budget for the current frame. The process could also include performing quantization of the current frame based on the masking thresholds. After the quantization of the current frame, the bit budget for next frame is updated for estimating the masking thresholds of the next frame. The process could also include encoding the quantized audio data.

Abstract: The present disclosure provides a digital audio signal processing system that comprises a set of delay lines, allpass and lowpass filters to achieve the reverberation effect. The present disclosure further provides a method for generating and controlling digital reverberations for audio signals. The reverberation generated will have an increasing echo density in the time domain and a faster decay of high frequency signals than low frequency signals. The controlling mechanism of reverberation generation is realized through the extraction of the real environment characteristics.

Abstract: A method for stereo audio perceptual encoding of an input signal includes masking threshold estimation and bit allocation. The masking threshold estimation and bit allocation are performed once every two encoding processes. Another method for stereo audio perceptual encoding of an input signal includes performing a time-to-frequency transformation, performing a quantization, performing a bitstream formatting to produce an output stream, and performing a psychoacoustics analysis. The psychoacoustics analysis includes masking threshold estimation on a first of every two successive frames of the input signal.

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. 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: 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.

Abstract: Channel coupling for an AC-3 encoder, using mixed precision computations and 16-bit coupling coefficient calculations for channels with 32-bit frequency coefficients.

Abstract: The present invention provides an audio streaming system and method for transmitting audio signals with high quality. The advantages of the present invention include easy implementation, computational efficiency, and provision of better audio quality. More particularly, the present invention provides a Multi-band Time Expansion algorithm for lost packet concealment. The Multi-band Time Expansion algorithm detects the number of continuously lost packets in an audio input signal and the correctly received packets on either side of the lost packets. Then the Multi-band Time Expansion algorithm time-expands the correctly received packets that may be from either one side or both sides of the lost packets, wherein the correctly received packets are stretched to cover the length of the lost packets. Finally the Multi-band Time Expansion algorithm overlap-adds the stretched packets so that the lost packets are concealed.

Abstract: A method includes splitting and filtering a left input signal and a right input signal to produce a plurality of frequency sub-bands. Each of the frequency sub-bands includes a left sub-band signal and a right sub-band signal. The method also includes processing the left and right sub-band signals associated with each of the frequency sub-bands into a plurality of sub-band channel signals. The plurality of sub-band channel signals includes at least three sub-band channel signals. In addition, the method includes summing corresponding ones of the sub-band channel signals for reproduction in a corresponding channel of a plurality of channels. The plurality of sub-band channel signals may include a left sub-band channel signal, a right sub-band channel signal, a center sub-band channel signal, a left surround sub-band channel signal, and a right surround sub-band channel signal.

Abstract: The present invention may be used to improve the quality of music output from audio systems by simulating the effect of low frequency signals in the human ear. This thus allows listeners to perceive the lower frequencies signals, even though the speakers may be incapable of providing such low frequency outputs. The present invention provides simple methods for processing enhancing bass effect in audio signals. The simple methods also result in the bass enhancement being computationally less intensive. The bass effect enhancement techniques described in a first and second embodiment are based on the response of sine and cosine transfer functions and on the directional independence of low frequency components. The human ear is unable to resolve directions from low frequency components. The bass effect enhancement technique described in a third embodiment is based on response of a exponential transfer function.

Abstract: A method includes receiving a sequence of frames containing audio information and determining that a frame is missing in the sequence of frames. The method also includes comparing the frame that precedes the missing frame to the received frames to identify a selected frame. The method further includes identifying a replacement frame comprising the frame that follows the selected frame. In addition, the method includes inserting the replacement frame into the sequence of frames in place of the missing frame.

Abstract: A method for encoding an audio signal, including providing a masking function, representative of psychoacoustic masking; setting a quality value for data of the encoded signal, adjusting the masking function dependent upon the quality value; and allocating bits for quantization of the encoded signal based on the incremental masking function.

Abstract: An energy-based pattern recognition algorithm receives the input frames of an audio signal and a test frame sequence and returns a best match in the audio signal to the given test frame sequence. The energy of each input frame is computed and input frames for which the energy is both within a predetermined degree of closeness to the local maximum energy within the test frame sequence and a local maximum within a respective neighborhood of adjacent frames are identified as probable matches. The difference between overall energy for frames neighboring the remaining probable matches and the test frame sequence is computed as a percentage. The best match is selected based on a weighted combination of difference between local maximum energies and minimum percent deviation in overall energy. Local signal characteristic matching may be employed, with weighting, to refine matching.

Abstract: An energy-based pattern recognition algorithm receives the input frames of an audio signal and a test frame sequence and returns a best match in the audio signal to the given test frame sequence. The energy of each input frame is computed, and input frames for which the energy is within a predetermined degree of closeness to the local maximum energy within the test frame sequence are identified as probable matches. Probable matches are then eliminated if the respective probable match does not correspond to a local maximum within a respective neighborhood of adjacent frames. The difference between overall energy for frames neighboring the remaining probable matches and the test frame sequence is computed as a percentage, with the minimum percent deviation in energy from the test frame sequence being returned as the best pattern match. Local signal characteristic matching may be employed to refine matching.

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 of searching for a best-match decimation vector of decimation factors for non-uniform filter bank, the best match vector allowing perfect or near-perfect reconstruction of an input signal of the non-uniform filter bank, the method including the steps of: a) selecting a partial decimation vector having a number, l, of decimation factors, where l does not exceed a maximum number, K, of decimation factors of said best-match decimation vector; b) testing said l decimation factors to determine whether said partial decimation vector satisfies a feasibility criterion; c) testing a least common multiplier value of said l decimation factors to determine whether said least common multiplier value is greater than a predetermined value; d) testing a maximum decimation value, Dmax, of said partial decimation vector to determine whether Dmax is less than one; e) testing a minimum decimation value, Dmin, of said partial decimation vector to determine whether Dmin is greater than one; and f) if said feasibility crit

Abstract: A system and method is provided for determining whether a data frame of a coded speech signal corresponds to voice or to noise. In one embodiment, a voice activity detector determines a cross-correlation of data. If the cross-correlation is lower than a predetermined cross-correlation value, then the data frame corresponds to noise. If not, then the voice activity detector determines a periodicity of the cross-correlation and a variance of the periodicity. If the variance is less than a predetermined variance value, then the data frame corresponds to voice. In another embodiment, a method determines energy of the data frame and an average energy of the coded speech signal. If the data frame is one of a predetermined number of initial data frames, then a comparison between the average energy to the energy of the data frame is used to determine whether the data frame is noise or voice.

Abstract: An audio decoder solution is here provided where a reduction in computing power is required. The proposed method consists of forcing the multiple output channels to only one type of inverse transformation format. A format of long transform length is more suitable for input signals whose spectrum remains stationary or quasi-stationary. This provides a greater frequency resolution, improved coding performance and a reduction of computing power required. Another format of two or more short transform lengths, possessing greater time resolution, is more desirable for rapidly changing signals with time. The computer power required for two or more short transforms should be higher than for only one transformation. The time versus frequency resolution trade-off should be considered when selecting a transform block length. Advantage is taken of human hearing behaviour to reduce the computing power of a processing engine (e.g. DSP) when downmixing from an M-channel input to a P-channel output is required.