Abstract: A time-domain system and method of modifying the time scale of digital audio signals includes a pre-processor. The pre-processor forms a synthesized signal for processing with minimum computation and that has optional features to give preference to certain audio channels and/or frequency bands, a mechanism of adaptively characterizing the temporal features of the synthesized signal by its normalized power and zero-crossing count, and a mechanism of identifying a segment of the synthesized signal where the time scale can be modified without introducing artifacts or losing content.

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: 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: The quality of music output from audio systems is improved by simulating the effect of low frequency signals in the human ear. This thus allows listeners to perceive the lower frequency signals, even though the speakers may be incapable of providing such low frequency outputs. A method is provided for processing enhancing bass effect in audio signals. The method also results in the bass enhancement being computationally less intensive. The bass effect enhancement techniques 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 alternatively is based on the response of an exponential transfer function.

Abstract: The invention provides for the encoding of surround sound produced by any coincident microphone techniques with coincident-to-virtual microphone signal matrixing. An encoding scheme provides significantly lower computational demand, by deriving the spatial parameters and output downmixes from the coincident microphone array signals and the coincident-to-surround channel-coefficients matrix, instead of the multi-channel signals.

Abstract: An MPEG-1 layer 3 audio encoder, including a scalefactor generator for determining first scalefactors for encoding a block of audio data if a temporal masking transient is not detected in said block of audio data; and for selecting the maximum of said scalefactors for encoding said block of audio data if a temporal masking transient is detected in said block of audio data to enable greater compression of said audio data. Increases in quantization error, due to use of the maximum scalefactor are pre-masked or post-masked by the temporal masking transient. In cases where the last portion of a block includes a temporal masking transient that masks the preceding portions of the block, the maximum scalefactor is only used to encode the block if the resulting increase in quantization error is less than 30% of the quantization error for the block.

Abstract: A method of decoding audio data, encoded in multiple DIF blocks in a Digital Video (DV) data stream, and outputting said audio data as a PCM frame, includes fetching a single Digital Interface Frame (DIF) block from the DV data stream. A first byte in the single DIF block is de-shuffled to determine its index (n) in the PCM frame. Each byte in the in the single DIF block is de-shuffled to determine its respective index (n) in the PCM frame. The de-shuffled data is written into the PCM frame for output if the present DIF block is the last in the present DV frame. Subsequent DIF blocks in the DV frame are processed in the manner described above.

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: Pitch detection of speech signals finds numerous applications in karaoke, voice recognition and scoring applications. While most of the existing techniques rely on time domain methods, the invention utilizes frequency domain methods. There is provided a method and system for determining the pitch of speech from a speech signal. The method includes the steps of: producing or obtaining the speech signal; distinguishing the speech signal into voiced, unvoiced or silence sections using speech signal energy levels; applying a Fourier Transform to the speech signal and obtaining speech signal parameters; determining peaks of the Fourier transformed speech signal; tracking the speech signal parameters of the determined peaks to select partials; and determining the pitch from the selected partials using a two-way mismatch error calculation.

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 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: 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: 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 method and audio device are presented that preserve mono energy during downmixing of a hybrid coding process of an audio signal. The method includes calculating a stereo scaling factor in a group level that is definable within a stereo band. The method may also include updating the stereo scaling factor using an update rate and synchronizing the update rate of a spatial parameter during a fast changing transient portion of the signal. A number of groups in a first stereo band may be greater than a number of groups in a second stereo band, and the first stereo band may be a lower frequency band than the second band or may be perceptually more important than the second band.

Abstract: An apparatus for adaptively suppressing noise in an input signal frequency spectrum derived from overlapping input frames is provided. The system includes a psychoacoustic power computation module configured to compute a noisy signal power in psychoacoustic bands, a voice activity scoring module configured to compute a probabilistic score for a presence of a speech, and a noise estimation module configured to estimate a noise power in the psychoacoustic bands based on information of past frames, the probabilistic score, and the computed noisy signal power. The system also includes a gain computation module configured to compute a gain for each frequency, based on a probabilistic heuristic, the probabilistic score and the information on the past frames, and a gain post-processing module configured to perform a gain time smoothing, a gain frequency smoothing, and a gain regulation for the computed gain.

Abstract: An device and method of generating environmental reverberation effects for digital audio signals is presented. The device includes a reverberation controller. The reverberation controller pre-processes one or more predetermined characteristics of a first audio signal to produce a pre-processed signal and generates a plurality of delayed outputs from the pre-processed signal, each output having a predetermined delay. The reverberation controller also produces a plurality of reflection outputs from the plurality of delayed outputs and combines the plurality of reflection outputs to produce a second audio signal having a desired reverberation response.

Abstract: A unified filter bank for use in encoding and decoding MPEG-1 audio data, wherein input audio data is encoded into coded audio data and the coded audio data is subsequently decoded into output audio data. The unified filter bank includes a plurality of filters, with each filter of the plurality of filters being a cosine modulation of a prototype filter. The unified filter bank is operational as an analysis filter bank during audio data encoding and as a synthesis filter bank during audio data decoding, wherein the unified filter bank is effective to substantially eliminate the effects of aliasing, phase distortion and amplitude distortion in the output audio data.

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: 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: 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.