Abstract: A coding apparatus which suppresses an extreme increase in a bit rate, includes: a downmixing and coding unit (301) that downmixes audio signals that have been provided, to reduce the number of channels to be fewer than the number of the provided audio signals, and to code the downmix signals; an object parameter extracting unit (304) that extracts parameters indicating correlation between the audio signals; and a multiplexing circuit (309) that multiplexes the extracted parameters with the generated downmix coded signals. The object parameter extracting unit (304) includes: an object classifying unit (305) that classifies each of the provided audio signals into a predetermined one of types based on audio characteristics; and an object parameter extracting circuit (308) that extracts parameters using a temporal granularity and a frequency granularity each of which is determined for a corresponding one of the types.

Abstract: To provide a bandwidth extension method which allows reduction of computation amount in bandwidth extension and suppression of deterioration of quality in the bandwidth to be extended. In the bandwidth extension method: a low frequency bandwidth signal is transformed into a QMF domain to generate a first low frequency QMF spectrum; pitch-shifted signals are generated by applying different shifting factors on the low frequency bandwidth signal; a high frequency QMF spectrum is generated by time-stretching the pitch-shifted signals in the QMF domain; the high frequency QMF spectrum is modified; and the modified high frequency QMF spectrum is combined with the first low frequency QMF spectrum.

Abstract: To provide an audio signal processing apparatus which can perform, with low operation amount, audio signal processing that is either time stretch and/or compression processing or frequency modulation processing. The audio signal processing apparatus is intended to transform an input audio signal sequence using a predetermined adjustment factor. The audio signal processing apparatus includes a filter bank (2601) which transforms the input audio signal sequence into Quadrature Mirror Filter (QMF) coefficients using a filter for Quadrature Mirror Filter analysis (a QMF analysis filter) and an adjusting unit (2602) which adjusts the QMF coefficients based on a predetermined adjustment factor.

Abstract: An encoding and decoding system includes: a characteristic determining unit which determines whether a sound signal is a speech signal or an audio signal; an encoder which encodes the sound signal into an encoded signal, based on a determination by the characteristic determining unit; a transmitting unit which transmits the encoded signal; a receiving unit which receives the encoded signal; a decoder which decodes the encoded signal; and a packet loss detecting unit which detects a loss of data of the encoded signal and transmits a notification indicating the loss of the data to the characteristic determining unit. Upon receiving the notification indicating the loss of the data, the characteristic determining unit causes the encoder to encode the sound signal portion into a signal portion composed of independently decodable frames.

Abstract: Provided is an encoding device (1) including: a pitch contour analysis unit (101) which detects information, a dynamic time-warping unit (102) which generates, based on the information, pitch change ratios (Tw_ratio in FIG. 18) within a range (86) including a range (86a) of the pitch change ratios corresponding to absolute pitch differences of 42 cents or larger; a first lossless coding unit (103) which codes the generated pitch parameters (102x); a time-warping unit (104) which shifts a pitch of a signal according to the information; and a second encoding unit which codes a signal (104x) obtained by the shifting.

Abstract: An audio rendering device which uses multichannel speakers includes: a first delay computation unit which computes, based on rendering information about the multichannel speakers, a first delay corresponding to a primary wavefront which propagates in a predetermined traveling direction from a sound source that is each of speakers included in the multichannel speakers; a second delay computation unit which computes, based on the rendering information about the multichannel speakers, a second delay corresponding to a secondary wavefront which is generated by the primary wavefront and has a scattering wavefront; an addition unit which adds up the first delay and the second delay to compute a total delay; and a delay filter which applies the total delay to an input audio signal to generate a multichannel audio signal for use in rendering with the multichannel speakers, and outputs the multichannel audio signal to the multichannel speakers.

Abstract: An object decoding unit receives a plurality of pieces of coded acoustic information coded on an object basis and decodes the received coded acoustic information on an object basis to generate object signals. A rendering unit positions, for each acoustic signal resulting from synthesizing the object signals, a virtual sound source of each of the object signals resulting from the object-based decoding, in a listening space, and priority information indicating a priority of the acoustic signal is multiplexed with the coded acoustic information. The rendering unit determines, with reference to the priority indicated in the priority information or the number of object signals included in the acoustic signal, a size of an area of the listening space which is allocated to the acoustic signal, and positions, within the area, the virtual sound source of each of the object signals included in the acoustic signal.

Abstract: An audio decoding apparatus comprises: a plurality of decoding units; a band replicating unit which processes a decoded signal obtained when a corresponding decoding unit decodes a coded signal, according to a scheme specified by transmitted information; and an information transmitting unit which transmits, to a signal processing unit, information identifying the corresponding decoding unit from among the plurality of decoding units.

Abstract: A sound signal hybrid encoder includes: a signal analysis unit which determines a scheme for encoding a frame included in a sound signal; an LFD encoder which encodes a frame to generate an LFD frame; an LP encoder which encodes a frame to generate an LP frame; a switching unit which switches between the encoders according to a result of the determination by the signal analysis unit; and an AC signal generation unit which generates an AC signal according to a scheme selected from among schemes, outputs the generated AC signal, and also outputs an AC flag indicating the selected scheme.

Abstract: A hybrid sound signal decoder decodes a bitstream including audio frames encoded by an audio encoding process using a low delay filter bank and speech frames encoded by a speech encoding process using linear prediction coefficients. When a current frame to be decoded is an ith frame which is an initial speech frame after switching from an audio frame to a speech frame, the hybrid sound signal decoder generates sub-frames which are a signal corresponding to an i?1th frame before being encoded, using a sub-frame which is a signal generated using a signal of the i?1th frame before being encoded, the signal of the i?1th frame being obtained by decoding the ith frame.

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: A coding device includes: a pitch contour detection unit which detects a pitch contour of an input audio signal; a dynamic time warping unit which determines the number of pitch nodes based on the pitch contour and generates a first time warping parameter including information indicating the determined number of pitch nodes, a pitch change position, and a pitch change ratio; a first encoder which codes the first time warping parameter; a time warping unit which corrects pitch, using the information obtained from the first time warping parameter, to approximate the pitches of the number of pitch nodes to a predetermined reference value; a second encoder which codes the input audio signal at the corrected pitch; and a multiplexer which multiplexes the coded time warping parameter and the coded audio signal to generate a bitstream.

Abstract: An audio decoding device of the present invention includes: a decoding unit decoding a stream to a spectrum coefficient, and outputting stream information when a frame included in the stream cannot be decoded; an orthogonal transformation unit transforming the spectrum coefficient to a time signal; a correction unit generating a correction time signal based on an output waveform within a reference section that is in a section that overlaps between an error frame section to which the stream information is outputted and an adjacent frame section and that is a section in the middle of the adjacent frame section, when the decoding unit outputs the stream information: and an output unit generating the output waveform by synthesizing the correction time signal and the time signal.

Abstract: Provided are a new hybrid audio decoder and a new hybrid audio encoder having block switching for speech signals and audio signals. Currently, very low bitrate audio coding methods for speech and audio signal are proposed. These audio coding methods cause very long delay. Generally, in coding an audio signal, algorithm delay tends to be long to achieve higher frequency resolution. In coding a speech signal, the delay needs to be reduced because the speech signal is used for telecommunication. To balance fine coding quality for these two kinds of input signals with very low bitrate, this invention provides a combination of a low delay filter bank like AAC-ELD and a CELP coding method.

Abstract: To provide an enhanced true-to-life atmosphere enjoyed in multipoint connecting, and reduce a calculation load at a multipoint connection unit, as well.

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 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 that codes the first downmix signal; a first t-f converting unit that converts the input multi-channel audio signal into a multi-channel audio signal in a frequency domain; and a spatial information calculating unit that generates spatial information for generating a multi-channel audio signal from a downmix signal.

Abstract: A signal processing device includes a generation unit that generates a second signal from a first signal that is obtained by down mixing two signals; a mixing coefficient determination unit that determines, based on a value L and a value ?, a mixing degree for mixing the first signal and the second signal; and a mixing unit that mixes the first signal and the second signal based on the mixing degree determined by the mixing coefficient determination unit. The generation unit includes a first filter that generates a low frequency band signal in the second signal, from a low frequency band signal in the first signal; and a second filter that generates a high frequency band signal in the second signal, from a high frequency band signal in the first signal. The first filter is a filter unit which, for a complex-number signal, de-correlates an input signal and adds a reverberation component by using a delay unit and an all pass filter, and the processing unit is a filter unit different from the first filter.

Abstract: An audio decoding apparatus comprises: a plurality of decoding units; a band replicating unit which processes a decoded signal obtained when a corresponding decoding unit decodes a coded signal, according to a scheme specified by transmitted information; and an information transmitting unit which transmits, to a signal processing unit, information identifying the corresponding decoding unit from among the plurality of decoding units.

Abstract: An encoding device (200) includes an MDCT unit (202) that transforms an input signal in a time domain into a frequency spectrum including a lower frequency spectrum, a BWE encoding unit (204) that generates extension data which specifies a higher frequency spectrum at a higher frequency than the lower frequency spectrum, and an encoded data stream generating unit (205) that encodes to output the lower frequency spectrum obtained by the MDCT unit (202) and the extension data obtained by the BWE encoding unit (204). The BWE encoding unit (204) generates as the extension data (i) a first parameter which specifies a lower subband which is to be copied as the higher frequency spectrum from among a plurality of the lower subbands which form the lower frequency spectrum obtained by the MDCT unit (202) and (ii) a second parameter which specifies a gain of the lower subband after being copied.

Abstract: An encoding device (200) includes an MDCT unit (202) that transforms an input signal in a time domain into a frequency spectrum including a lower frequency spectrum, a BWE encoding unit (204) that generates extension data which specifies a higher frequency spectrum at a higher frequency than the lower frequency spectrum, and an encoded data stream generating unit (205) that encodes to output the lower frequency spectrum obtained by the MDCT unit (202) and the extension data obtained by the BWE encoding unit (204). The BWE encoding unit (204) generates as the extension data (i) a first parameter which specifies a lower subband which is to be copied as the higher frequency spectrum from among a plurality of the lower subbands which form the lower frequency spectrum obtained by the MDCT unit (202) and (ii) a second parameter which specifies a gain of the lower subband after being copied.