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: 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 it 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: An apparatus, method, and computer program are capable of receiving and cross-correlating a first audio signal and a second audio signal. This produces a cross-correlated signal, which is used to identify a plurality of parameters associated with at least one of the first and second audio signals. The parameters are used to generate an indicator identifying an extent to which the first and second audio signals match.

Abstract: An audio processor, apparatus, and method use physical speakers to emulate one or more additional speakers. The physical speakers produce sounds that, from a listener's perspective, appear to come from at least one direction where a physical speaker is not present. Any number of additional speakers can be virtualized, such as three or five speakers that allow two speakers to emulate a 5.1 audio system.

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 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 for encoding frequency coefficients in an AC-3 Encoder. The method includes: representing frequency coefficients in theform of a respective exponent and mantissa; coding the exponents; and shifting the mantissas to compensate for changes in the exponent values, wherein the exponents comprise an original exponent set (e0, e1, . . . en−1) which is mapped to a new exponent set (e0′, e1′, . . . , e′n−1) after coding, so as to satisfy: ∥e′i+1−e′i∥<D, where i=0, . . . , n−1 and D is a maximum allowed difference between two consecutive exponents, and e′i≦ei.

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 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.