Abstract: A combination device (305) according to the present invention includes: a detection unit (501) that detects active coded bitstreams that are effective coded bitstreams from a plurality of coded bitstreams (116) within a predetermined time period; a first combining unit (504) that combines, from a plurality of downmix sub-streams (115) included in the coded bitstreams (116), only downmix sub-streams (115) included in the active coded bitstreams so as to generate a combined downmix sub-stream (121); and a second combining unit (506) that combines, from a plurality of parameter sub-streams (113) included in the coded bitstreams (116), only parameter sub-streams (113) included In the active coded bitstreams so as to generate a combined parameter sub-stream (122).

Abstract: Provided is an encoding device which improves the sound quality of a stereo signal while maintaining a low bit rate. The encoding device includes: an LP inverse filter (121) which LP-inverse-filterS a left signal L(n) by using an inverse quantization linear prediction coefficient AdM(z) of a monaural signal; a T/F conversion unit (122) which converts the left sound source signal Le(n) from a temporal region to a frequency region; an inverse quantizer (123) which inverse-quantizes encoded information Mqe; spectrum division units (124, 125) which divide a high-frequency component of the sound source signal Mde(f) and the left signal Le(f) into a plurality of bands; and scale factor calculation units (126, 127) which calculate scale factors ai and ssi by using a monaural sound source signal Mdeh,i(f), a left sound source signal Leh,i(f), Mdeh,i(f), and right sound source signal Reh,i(f) of each divided band.

Abstract: There is provided an encoder capable of improving inter-channel prediction (ICP) performance in scalable stereo sound encoding using an ICP. In the encoder, ICP analysis units (113, 114, 115) use, as reference signal candidates, a frequency coefficient (sL?(f)) in the low-band portion of a side residual signal, a frequency coefficient (mM,i(f)) in each sub-band portion of a monaural residual signal, and a frequency coefficient (mL(f)) in the low-band portion of the monaural residual signal, respectively, and perform an ICP analysis between the respective these candidates and a frequency coefficient (sM,i(f)) in each sub-band portion of the side residual signal to generate first, second, and third ICP coefficients.

Abstract: Provided is an audio decoder which can reduce an amount of arithmetic operations while suppressing occurrence of aliasing noise. The audio decoder includes: a decoder (102) and an analysis filter bank (110) which generate, from a coded down-mixed signal, the first frequency band signal (x) corresponding to a down-mixed signal (M); a channel expansion unit (130) which converts the first frequency band signal (x) generated by the analysis filter bank (110) into output signals (y) corresponding to respective audio signals of N channels, using BC information; an synthesis filter bank (140) which performs band synthesis for the output signals (y) generate by the channel expansion unit (130) and thereby converts the output signals (y) into the respective audio signals of the N channels on a time axis; and an aliasing noise detection unit (120) which detects occurrence of aliasing noise in the first frequency band signal (x).

Abstract: The delay in a multi-channel audio coding apparatus and a multi-channel audio decoding apparatus is reduced. The audio coding apparatus includes: a downmix signal generating unit (410) that generates, in a time domain, a first downmix signal that is one of a 1-channel audio signal and a 2-channel audio signal from an input multi-channel audio signal; a downmix signal coding unit (404) that codes the first downmix signal; a first t-f converting unit (401) that converts the input multi-channel audio signal into a multi-channel audio signal in a frequency domain; and a spatial information calculating unit (409) that generates spatial information for generating a multi-channel audio signal from a downmix signal.

Abstract: In the conventional art inventions for coding multi-channel audio signals, three of the major processes involved are: generation of a reverberation signal using an all-pass filter; segmentation of a signal in the time and frequency domains for the purpose of level adjustment; and mixing of a coded binaural signal with an original signal coded up to a fixed crossover frequency. These processes pose the problems mentioned in the present invention. The present invention proposes the following three embodiments: to control the extent of reverberations by dynamically adjusting all-pass filter coefficients with the inter-channel coherence cues; to segment a signal in the time domain finely in the lower frequency region and coarsely in the higher frequency region; and to control a crossover frequency used for mixing based on a bit rate, and if the original signal is coarsely quantized, to mix a downmix signal with an original signal in proportions determined by an inter-channel coherence cue.

Abstract: Provided is a stereo audio encoding device which can improve the ICP (Inter-channel Prediction) performance of a stereo audio signal while suppressing the bit rate. The device (100) includes: a QMF analysis unit (101) which divides two channel signals constituting a stereo audio signal into a plurality of frequency band signals; a monaural signal generation unit (104) which generates a monaural signal by averaging the two channel signals of the divided frequency bands; parameter band constituting units (102, 105) each of which collects one or more of the continuous frequency bands to constitute a parameter band in such a manner that less bands are contained in a lower frequency for the two channel signals and monaural signals of the divided frequency bands; and an ICP analysis unit (106) which performs inter-channel prediction by using the channel signal and the monaural signal of the divided frequency bands.

Abstract: Provided is a stereo audio encoding device which can improve ICP accuracy of a stereo audio signal having a low inter-channel correlation while suppressing a bit rate. The device (100) includes: a monaural signal generation unit (101) which generates an average value of a left channel signal L and a right channel signal R as a monaural signal M; an adaptive synthesis unit (103) which generates a synthesis signal L2 of the left channel signal L and the right channel signal R by using a synthesis ratio a inputted from a synthesis ratio adjusting unit (105); LPC analysis units (102, 104) which perform LPC analysis on the monaural signal M and the synthesis signal L2 so as to generate linear prediction residual signals Me, L2e, respectively; a synthesis ratio adjusting unit (105) which firstly initializes the synthesis ratio a to 1.

Abstract: Provided is an encoding device which can achieve both of highly effective encoding/decoding and high-quality decoding audio when executing a scalable stereo audio encoding by using MDCT and ICP. In the encoding device, an MDCT conversion unit (111) executes an MDCT conversion on a residual signal of left channel/right channel subjected to window processing. An MDCT conversion unit (112) executes an MDCT conversion on the monaural residual signal which has been subjected to the window processing. An ICP analysis unit (117) executes an ICP analysis by using the correlation between a frequency coefficient of a high-band portion of the left channel/right channel and a frequency coefficient of a high-band portion of the monaural residual signal so as to generate an ICP parameter of the left channel/right channel residual signal. An ICP parameter quantization unit (118) quantizes each of the ICP parameters.

Abstract: Disclosed is a stereo encoding device which can improve critical channel encoding accuracy without increasing the encoding information amount. The device includes: a monaural signal synthesis unit (101) which combines a left channel signal L(n) and a right channel signal R(n) so as to generate a monaural signal M(n); a correlation coefficient calculation unit (102) which calculates a correlation coefficient CML between M(n) and L(n) and a correlation coefficient CMR between M(n) and R(n); a critical channel judging unit (103) which decides one of the L(n) and R(n) having a smaller correlation with M(n) as the critical channel if the ratio of CML against CMR is not within a predetermined range from 90% to 111%, for example; and an ICP encoding unit (104) which performs ICP encoding by adjusting the degree of the ICP parameter of the critical channel to be higher than the degree of the ICP parameter of the non-critical channel.

Abstract: According to the present invention, it is possible to calculate appropriate chirp factor and noise component amount with a little processing amount. Input subband signal is segmented into a plurality of ranges by a range segmentation unit 101. The range segmentation is performed for energy value calculation, chirp factor calculation, noise component calculation, and tone component calculation, respectively, and determined range segmentation information ei, bi, qi, and hi are outputted. Respective processing for the energy calculation, the chirp factor calculation, the tone component calculation, and the noise component calculation are performed sequentially for the respective corresponding ranges. By using linear prediction processing, it is possible to obtain an parameter having higher accuracy with a little operation amount.

Abstract: Disclosed is a stereo audio encoding device capable of improving a spatial image of a decoded audio in stereo audio encoding. In this device, an original cross correlation calculation unit (101) calculates a mutual relationship coefficient (C1) between the original L channel signal and the original R channel signal. A stereo audio reconfiguration unit (104) subjects the inputted L channel signal and the R channel signal to encoding and decoding so as to generate an L channel reconfigured signal (L?) and an R channel reconfigured signal (R?). A reconfiguration cross correlation calculation unit (105) calculates a cross correlation coefficient (C2) between the L channel reconfigured signal (L?) and the R channel reconfigured signal (R?). A cross correlation comparison unit (106) calculates and outputs a comparison result &agr; between the cross correlation coefficient (C1) and the cross correlation coefficient (C2).

Abstract: This method and apparatus extract symbolic high-level musical structure resembling that of a music score. Humming or the like is converted with this invention into a sequence of notes that represent the melody that the user (usually human, but potentially animal) is trying to express. These retrieved notes each contain information such as a pitch, the start time and duration and the series contains the relative order of each note. A possible application of the invention is a music retrieval system whereby humming forms the query to some search engine.

Abstract: Provided is a multi-channel acoustic signal processing device by which loads of arithmetic operations are reduced. The multi-channel acoustic signal processing device (100) includes: a decorrelated signal generation unit (181), and a matrix operation unit (187) and a third arithmetic unit (186). The decorrelated signal generation unit (181) generates a decorrelated signal w? indicating a sound which includes a sound indicated by an input signal x and reverberation, by performing reverberation processing on the input signal x. The matrix operation unit (187) and the third arithmetic unit (186) generate audio signals of m channels, by performing arithmetic operation on the input signal x and the decorrelated signal w? generated by the decorrelated signal generation unit (181), using a matrix R3 which indicates distribution of a signal intensity level and distribution of reverberation.

Abstract: An energy corrector (105) for correcting a target energy for high-frequency components and a corrective coefficient calculator (106) for calculating an energy corrective coefficient from low-frequency subband signals are newly provided. These processors perform a process for correcting a target energy that is required when a band expanding process is performed on a real number only. Thus, a real subband combining filter and a real band expander which require a smaller amount of calculations can be used instead of a complex subband combining filter and a complex band expander, while maintaining a high sound-quality level, and the required amount of calculations and the apparatus scale can be reduced.

Abstract: To provide an acoustic signal processing apparatus which can reduce the amount of calculation in matrix arithmetic. An acoustic signal processing apparatus (24) converts down-mixed acoustic signals of NI channels to acoustic signals of NO channels, where NO>NI.

Abstract: A temporal processing apparatus (energy shaping apparatus) (600a) includes: a splitter (601) splitting an audio signal, included in the sub-band domain, which are obtained through a hybrid time and frequency transformation into diffuse signals indicating reverberating components and direct signals indicating non-reverberating components; a downmix unit (604) generating a downmix signal by downmixing the direct signals; BPFs (605 and 606) respectively generating a bandpass downmix signal and bandpass diffuse signals, by performing bandpass processing on the downmix signal and the diffuse signals on a sub-band-to-sub-band basis, which are split on the sub-band basis; normalization processing units (607 and 608) respectively generating a normalized downmix signal and normalized diffuse signals by normalizing the bandpass downmix signal and the bandpass diffuse signals with regard to respective energy; a scale computation processing unit (609) computing, on a predetermined time slot basis, a scale factor indicati

Abstract: A method (100) and apparatus (200) are disclosed for transcribing a humming signal into a sequence of musical notes. The method begins by grouping (305) the signal into frames of data samples. Each frame is then processed to derive (320) a frequency distribution for each frames. The frequency distributions are processed to derive (410) a Harmonic Product Energy (HPE) distribution over the frames. The MPE distribution is then segmented (115, 120) to obtain boundaries of musical notes. The frequency distributions of the frames are also processed to derive (412) a fundamental frequency distribution. A pitch for each note is determined (125) from the fundamental frequency distribution.

Abstract: An energy corrector (105) for correcting a target energy for high-frequency components and a corrective coefficient calculator (106) for calculating an energy corrective coefficient from low-frequency subband signals are newly provided. These processors perform a process for correcting a target energy that is required when a band expanding process is performed on a real number only. Thus, a real subband combining filter and a real band expander which require a smaller amount of calculations can be used instead of a complex subband combining filter and a complex band expander, while maintaining a high sound-quality level, and the required amount of calculations and the apparatus scale can be reduced.

Abstract: A method for use in identifying an audio input, comprising the steps of: deriving a signature code from the audio input; subjecting the signature code to Correlation Matrix Memory (CMM) processing; and identifying the audio input based on an output of the CMM processing.