Abstract: An overall encoding procedure and associated decoding procedure are presented. The encoding procedure involves at least two signal encoding processes operating on signal representations of a set of audio input channels. Local synthesis is used in connection with a first encoding process to generate a locally decoded signal, including a representation of the encoding error of the first encoding process. This locally decoded signal is applied as input to a second encoding process. The overall encoding procedure generates at least two residual encoding error signals from at least one of said encoding processes, including at least said second encoding process. The residual error signals are then subjected to compound residual encoding in a further encoding process, preferably based on correlation between the residual error signals.

Abstract: Provided is an encoding apparatus for a High Quality Multi-channel Audio Codec (HQMAC) and a decoding apparatus for the HQMAC. The encoding/decoding apparatuses for the HQMAC may perform a High Quality Multi-channel Audio Codec-Channel Based (HQMAC-CB) encoding or an HQMAC-CB decoding in accordance with characteristics of inputted audio signals to provide compatibility with a lower channel.

Abstract: An audio spatial environment engine for flexible and scalable up-mixing from an M channel audio system to an N channel audio system, where M and N are integers and N is greater than M, is provided. The input M channel audio is provided to an analysis filter bank which converts the time domain signals into frequency domain signals. Relevant inter-channel spatial cues are extracted from the frequency domain signals on a sub-band basis and are used as parameters to generate adaptive N channel filters which control the spatial placement of a frequency band element in the up-mixed sound field. The N channel filters are smoothed across both time and frequency to limit filter variability which could cause annoying fluctuation effects. The smoothed N channel filters are then applied to adaptive combinations of the frequency domain input signals and are provided to a synthesis filter bank which generates the N channel time domain output signals.

Abstract: An audio encoder receives multi-channel audio data comprising a group of plural source channels and performs channel extension coding, which comprises encoding a combined channel for the group and determining plural parameters for representing individual source channels of the group as modified versions of the encoded combined channel. The encoder also performs frequency extension coding. The frequency extension coding can comprise, for example, partitioning frequency bands in the multi-channel audio data into a baseband group and an extended band group, and coding audio coefficients in the extended band group based on audio coefficients in the baseband group. The encoder also can perform other kinds of transforms. An audio decoder performs corresponding decoding and/or additional processing tasks, such as a forward complex transform.

Abstract: A television audio signal encoder includes a matrix that sums a left channel audio signal and a right channel audio signal to produce a sum signal. The matrix also subtracts one of the left and right audio signals from the other to produce a difference signal. The encoder also includes a configurable infinite impulse response digital filter that selectively uses one or more sets of filter coefficients to filter the difference signal. Each selectable set of filter coefficients is associated with a unique filtering application to prepare the difference signal for transmission.

Abstract: There is described a method of encoding input signals (CHI to CH3; 400 to 450) in a multi-channel encoder (5; 15) to generate corresponding output data comprising down-mix output signals (610, 620) together with complementary parametric data (600). The method includes a first step of down-mixing input signals (CHI to CH3; 400 to 450) to generate the corresponding down-mix output signals (610, 620), and a second step of processing the input signals (CHI to CH3; 400 to 450) during down-mixing to generate said parametric data (600) complementary to the down-mix output signals (610, 620). Processing of the input signals (CHI to CH3; 400 to 450) involves including information in the down-mix signals (610, 620) which is useable during subsequent decoding of the down-mix output signals (610, 620) and the parametric data (600) to determine at least some parameter data and thereby enabling representations of the input signals (CHI to CH3; 400 to 450) to be subsequently regenerated.

Abstract: For a multi-channel audio signal, parametric coding is applied to different subsets of audio input channels for different frequency regions. For example, for a 5.1 surround sound signal having five regular channels and one low-frequency (LFE) channel, binaural cue coding (BCC) can be applied to all six audio channels for sub-bands at or below a specified cut-off frequency, but to only five audio channels (excluding the LFE channel) for sub-bands above the cut-off frequency. Such frequency-based coding of channels can reduce the encoding and decoding processing loads and/or size of the encoded audio bitstream relative to parametric coding techniques that are applied to all input channels over the entire frequency range.

Abstract: A system and method of remotely enabling sound enhancement techniques is disclosed. In an embodiment, a watermark is embedded in an encoded multi-channel audio stream to remotely enable an enhancement decoder portion of a multi-channel audio decoder.

Abstract: In an audio signal encoding apparatus, a first audio signal and a second audio signal are added into an addition-result signal. The first audio signal is subtracted from the second audio signal to generate a subtraction-result signal. A first difference signal is generated which represents a difference in the addition-result signal. A second difference signal is generated which represents a difference in the subtraction-result signal. A plurality of first predictors have different prediction characteristics respectively, and are responsive to the first difference signal for generating first different prediction signals for the first difference signal, respectively. A plurality of first subtracters operate for generating first prediction-error signals representing differences between the first difference signal and the first different prediction signals, respectively. A first minimum prediction-error signal representative of a smallest difference is selected from among the first prediction-error signals.

Abstract: The present invention relates to a method of processing a media signal and apparatus therefor. A media signal decoding method according to the present invention includes detecting a channel having a valid value of the multi-channels to be generated and generating the detected channel having the valid value from the downmix signal and the spatial information signal. Accordingly, the present invention is able to reduce a decoding operation quantity by detecting which one of the channels to be generated from a transferred media signal is set to a virtual value and omitting decoding for the generation of the channel set to the virtual value.

Abstract: There is provided an audio encoding device capable of generating an appropriate monaural signal from a stereo signal while suppressing the lowering of encoding efficiency of the monaural signal. In a monaural signal generation unit (101) of this device, an inter-channel prediction/analysis unit (201) obtains a prediction parameter based on a delay difference and an amplitude ratio between a first channel audio signal and a second channel audio signal; an intermediate prediction parameter generation unit (202) obtains an intermediate parameter of the prediction parameter (called intermediate prediction parameter) so that the monaural signal generated finally is an intermediate signal of the first channel audio signal and the second channel audio signal; and a monaural signal calculation unit (203) calculates a monaural signal by using the intermediate prediction parameter.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: The invention relates to methods and units supporting a multichannel audio extension. In order to allow an efficient extension requiring a low computational complexity, it is proposed that at an encoding end, at least state information is provided as side information for a provided mono audio signal (M) generated out of a multichannel audio signal. The state information indicates for each of a plurality of frequency bands how a predetermined or equally provided gain value is to be applied in the frequency domain to the mono audio signal (M) for obtaining first and a second channel signals (L,R) of a reconstructed multichannel audio signal.

Abstract: A binaural cue coding scheme in which cue codes are derived from the transmitted audio signal. In one embodiment, an encoder downmixes C input channels to generate E transmitted channels, where C>E>1. A decoder derives cue codes from the transmitted channels and uses those cue codes to synthesize playback channels. For example, in one 5-to-2 BCC embodiment, the encoder downmixes a 5-channel surround signal to generate left and right channels of a stereo signal. The decoder derives stereo cues from the transmitted stereo signal, maps those stereo cues to surround cues, and applies the surround cues to the transmitted stereo channels to generate playback channels of a 5-channel synthesized surround signal.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A new audio playback architecture may be used, which allows the use of much larger buffering than that used by a typical audio subsystem in a computing system to improve power efficiency of the system and at the same time allows to maintain the quality (e.g., fidelity and responsiveness) of the audio playback. The audio controller in the new architecture may be made to report back to the host system a more accurate indication of which audio frame is being set to the audio codec than a currently available audio controller does. Additionally, the controller is capable of re-fetching previously buffered (but not yet transmitted) data. Furthermore, buffers in both the audio controller and the main memory may be dynamically adjusted during playback of audio data and/or for different applications.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: An audio signal having at least two channels can be efficiently down-mixed into a downmixe signal and a residual signal, when the down-mixing rule used depends on a spatial parameter that is derived from the audio signal and that is post-processed by a limiter to apply a certain limit to the derived spatial parameter with the aim of avoiding instabilities during the up-mixing or down-mixing process. By having a down-mixing rule that dynamically depends on parameters describing an interrelation between the audio channels, one can assure that the energy within the down-mixed residual signal is as minimal as possible, which is advantageous in the view of coding efficiency. By post processing the spatial parameter with a limiter prior to using it in the down-mixing, one can avoid instabilities in the down- or up-mixing, which otherwise could result in a disturbance of the spatial perception of the encoded or decoded audio signal.

Abstract: An audio decoding method includes: acquiring, from encoded audio data, a reception audio signal and first auxiliary decoded audio information; calculating coefficient information from the first auxiliary decoded audio information; generating a decoded output audio signal based on the coefficient information and the reception audio signal; decoding to result in a decoded audio signal based on the first auxiliary decoded audio signal and the reception audio signal; calculating, from the decoded audio signal, second auxiliary decoded audio information corresponding to the first auxiliary decoded audio information; detecting a distortion caused in a decoding operation of the decoded audio signal by comparing the second auxiliary decoded audio information with the first auxiliary decoded audio information; correcting the coefficient information in response to the detected distortion; and supplying the corrected coefficient information as the coefficient information when generating the decoded output audio signal.

Abstract: An encoding apparatus compresses a stereo signal using a sum signal and a difference signal of a left component signal and a right component signal of the stereo signal. The encoding apparatus includes a calculating unit that calculates complexity of the sum signal and complexity of the difference signal; a setting unit that sets, based on the complexity, an allocation rate of bits to be allocated in quantizing the sum signal and the difference signal; and a quantizing unit that quantizes the sum signal and the difference signal based on the allocation rate.

Abstract: The encoder transforms the audio signals (x(n),y(n)) from the time domain to audio signal (X(k),Y(k)) in the frequency domain, and determines the cross-correlation function (Ri, Pi) in the frequency domain. A complex coherence value (Qi) is calculated by summing the (complex) cross-correlation function values (Ri, Pi) in the frequency domain. The inter-channel phase difference (IPDi) is estimated by the argument of the complex coherence value (Qi), and the inter-channel coherence (ICi) is estimated by the absolute value of the complex coherence value (Qi).

Abstract: At an audio encoder, cue codes are generated for one or more audio channels, wherein an envelope cue code is generated by characterizing a temporal envelope in an audio channel. At an audio decoder, E transmitted audio channel(s) are decoded to generate C playback audio channels, where C>E?1. Received cue codes include an envelope cue code corresponding to a characterized temporal envelope of an audio channel corresponding to the transmitted channel(s). One or more transmitted channel(s) are upmixed to generate one or more upmixed channels. One or more playback channels are synthesized by applying the cue codes to the one or more upmixed channels, wherein the envelope cue code is applied to an upmixed channel or a synthesized signal to adjust a temporal envelope of the synthesized signal based on the characterized temporal envelope such that the adjusted temporal envelope substantially matches the characterized temporal envelope.

Abstract: An acoustic signal encoding device for down-mixing at different ratios to encode a multichannel signal with a small number of channels, and an acoustic signal decoding device for decoding the signal encoded by the acoustic signal encoding device. In these devices, weighting means (103) in the acoustic signal encoding device (100) weights input signals of two channels individually according to a down-mixing coefficient thereby to calculate the level difference of the signals of two channels weighted by a level difference calculation unit (104). A separating unit (202) in the acoustic signal decoding device (200) separates the down-mixed signals into signals of two channels with the level difference information weighted.

Abstract: An on-vehicle audio/video system is provided which comprises a controller, a data source and a playing device, wherein the controller comprises at least a first transmitter, and the data source comprises at least a first receiver, a first micro-controller and a second transmitter. The controller comprises at least a first RF transmitter. The data source comprises at least a first RF receiver, a first micro-controller and a second RF transmitter. The playing device comprises at least a second RF receiver. The controller sends an RF control signal to the data source and/or playing device. The first RF receiver in the data source receives the RF control signal and sends the signal to the first micro-controller for processing. The second transmitter transmits data signal and/or control signal to the playing device under the control of the first micro-controller.

Abstract: An apparatus for providing a set of spatial cues associated with an upmix audio signal having more than two channels on the basis of a two-channel microphone signal has a signal analyzer and a spatial side information generator. The signal analyzer is configured to obtain a component energy information and a direction information on the basis of the two-channel microphone signal, such that the component energy information describes estimates of energies of a direct sound component of the two-channel microphone signal and of a diffuse sound component of the two-channel microphone signal, and such that the directional information describes an estimate of a direction from which the direct sound component of the two-channel microphone signal originates. The spatial side information generator is configured to map the component energy information and the direction information onto a spatial cue information describing the set of spatial cues associated with an upmix audio signal having more than two channels.

Abstract: A method of decoding for an audio signal comprises the step of receiving a downmix of an audio signal, an object information, and a mix information, the object information including an object level information, an object correlation information, and an object gain information, generating a downmix processing information using the object information and the mix information, and processing the downmix of the audio signal using the downmix processing information. Various embodiments of the present invention provide a method and an apparatus for decoding multi-object audio signals fast and efficiently by reducing process time, computer resource, thereby relieving the resource requirement like the wide bandwidth. The object parameters according to the embodiments of the present invention can provide backward compatibility in the view of the channel-oriented decoding process.

Abstract: Audio encoding method and device comprising the transmission, in addition to the data representing a frequency-limited signal, of information relating to a temporal filter that is to be applied to the entire enhanced signal, both in its transmitted low-frequency part and in its reconstituted high-frequency part. The application of this filter for reshaping the reconstituted high-frequency part and the correction of compression artefacts present in the transmitted low-frequency part. In this way, the application of the temporal filter, simple and inexpensive, to all or part of the reconstituted signal, makes it possible to obtain a signal of good perceived quality.

Abstract: The present invention provides improvements to prior art audio codecs that generate a stereo-illusion through post-processing of a received mono signal. These improvements are accomplished by extraction of stereo-image describing parameters at the encoder side, which are transmitted and subsequently used for control of a stereo generator at the decoder side. Furthermore, the invention bridges the gap between simple pseudo-stereo methods, and current methods of true stereo-coding, by using a new form of parametric stereo coding. A stereo-balance parameter is introduced, which enables more advanced stereo modes, and in addition forms the basis of a new method of stereo-coding of spectral envelopes, of particular use in systems where guided HFR (High Frequency Reconstruction) is employed. As a special case, the application of this stereo-coding scheme in scalable HFR-based codecs is described.

Abstract: A method for processing an audio signal, comprising: receiving a downmix signal, an object information, and a mix information; generating a multi-channel information including at least one gain modification factor using the object information and the mix information, wherein the gain modification factor corresponds to a time-subband-variant factor for controlling gain of the downmix signal is disclosed.

Abstract: Provided are a multi-channel encoding and decoding method and apparatus capable of encoding and decoding residual signal by removing redundant information between a plurality of channels without a downmixed signal. In the method and apparatus, a reference signal which is to be used for encoding from a plurality of channel signals, the phase differences between the respective channel signals and the reference signal, gains which are the ratios of the amplitudes of the respective channel signals to the amplitude of the reference signal, and residual signals which correspond to differences between predicted signals and the actual channel signals, are encoded or decoded. The predicted signals are obtained by applying the phase differences and gains to the reference signal.

Abstract: A method for processing an audio signal, comprising: receiving a downmix signal, an object information, and a mix information; generating a downmix processing information using the object information and the mix information; processing the downmix signal using the downmix processing information; and, generating a multi-channel information using the object information and the mix information, wherein the number of channel of the downmix signal is equal to the number of channel of the processed downmix signal is disclosed.

Abstract: A method of encoding input signals (l, r) to generate encoded data (100) is provided. The method involves processing the input signals (l, r) to determine first parameters (?1, ?2) describing relative phase difference and temporal difference between the signals (l, r), and applying these first parameters (?1, ?2) to process the input signals to generate intermediate signals. The method involves processing the intermediate signals to determine second parameters (?; IID, ?) describing angular rotation of the first intermediate signals to generate a dominant signal (m) and a residual signal (s), the dominant signal (m) having a magnitude or energy greater than that of the residual signal (s). These second parameters are applicable to process the intermediate signals to generate the dominant (m) and residual (s) signals.

Abstract: A method and an apparatus for encoding digital audio data with reduced bit rates, the apparatus having a provider of psycho-acoustically quantized digital audio data with a bit rate being higher than the reduced bit rate. The apparatus further has an identifier for identifying a frequency band according to a selection criterion, the selection criterion being such that an impact on the quality of the digital audio data when the data in the identified frequency band is replaced by generated noise is smaller than the impact on the quality of the digital audio data, which would arise when the data in a different frequency band is replaced by generated noise. The apparatus further has a replacer for replacing data in the identified frequency band of the digital audio data by a noise synthesis parameter.

Abstract: Generic and specific C-to-E binaural cue coding (BCC) schemes are described, including those in which one or more of the input channels are transmitted as unmodified channels that are not downmixed at the BCC encoder and not upmixed at the BCC decoder. The specific BCC schemes described include 5-to-2, 6-to-5, 7-to-5, 6.1-to-5.1, 7.1-to-5.1, and 6.2-to-5.1, where “0.1” indicates a single low-frequency effects (LFE) channel and “0.2” indicates two LFE channels.

Abstract: The present invention provides improvements to prior art audio codecs that generate a stereo-illusion through post-processing of a received mono signal. These improvements are accomplished by extraction of stereo-image describing parameters at the encoder side, which are transmitted and subsequently used for control of a stereo generator at the decoder side. Furthermore, the invention bridges the gap between simple pseudo-stereo methods, and current methods of true stereo-coding, by using a new form of parametric stereo coding. A stereo-balance parameter is introduced, which enables more advanced stereo modes, and in addition forms the basis of a new method of stereo-coding of spectral envelopes, of particular use in systems where guided HFR (High Frequency Reconstruction) is employed. As a special case, the application of this stereo-coding scheme in scalable HFR-based codecs is described.

Abstract: A multiplexing apparatus for multiplexing audio data into transport stream (TS) packets includes a first encoding section encoding the audio data by a first encoding method; a second encoding section encoding the audio data by a second encoding method, which is a variable-length encoding method and which differs from the first encoding method, for attaching a timing value indicating a timing used when audio data is decoded in units of predetermined audio data; a packetization section packetizing the audio data encoded by the first encoding section and the audio data encoded by the second encoding section into TS packets and for attaching the same ID to a plurality of packetized TS packets; a determination section determining a TS packet to be multiplexed from among the plurality of TS packets packetized by the packetization section; and a multiplexing section multiplexing the TS packet determined by the determination section.

Abstract: A method is shown for supporting a multichannel audio extension at an encoding end of a multichannel audio coding system. In order to improve the audio quality over a large frequency range, the method comprises transforming each channel of a multichannel audio signal into the frequency domain and dividing a bandwidth of the frequency domain signals into a first region of lower frequencies and at least one further region of higher frequencies. Then, the frequency domain signals are encoded in each of the frequency regions with another type of coding to obtain parametric multichannel extension information for the respective frequency region. The invention relates equally to a method for supporting in a corresponding manner a multichannel audio extension at a decoding end. Also shown are a corresponding encoder, a corresponding decoder, and corresponding devices, systems and software program products.

Abstract: There is described a multi-channel encoder (10; 600) for processing input signals conveyed in N input channels to generate corresponding output signals conveyed in M output channels together with complementary parametric data; M and N are integers wherein N>M. The encoder (10; 600) includes a down-mixer for down-mixing the input signals to generate the corresponding output signals, the encoder also comprising an analyser for processing the input signals to generate the parameter data, said parametric data describing mutual differences between the N channels of input signal to allow for regenerating during decoding one or more of the N channels of input signals from the M channels of output signal. Such an encoder (10; 600) is capable of providing highly efficient data encoding and also of being backwards compatibility with relatively simpler decoders having fewer than N decoding output channels. The invention also concerns decoders (800) compatible with such a multi-channel encoder (10; 600).

Abstract: In an audio-information encoding apparatus, in order to encode an audio signal containing a white-noise component, an index iL indicating the energy level of the white-noise component and an index iR designating the start index of a random-number table are introduced into a code train. In an audio-information decoding apparatus (20), a white-noise generating unit (25) uses the indices iL and iR contained in the code train, thereby generating a white-noise signal Sw(t) on the time axis, which has the same level as the white noise, and an adder (26) adds the white-noise signal to an audio signal Sf(t) decoded on the time axis, outputting as an output audio signal So(t).