Patent Description:
In transmission of a coded and packetized audio signal through an Internet network with an IP (Internet Protocol) phone, a packet can be lost because of a network congestion or the like (this phenomenon will be referred to hereinafter as "packet loss"). With an occurrence of a packet loss, necessary audio codes are lost resulting in a failure in decoding of audio, thereby causing an audio discontinuity. A technology for preventing an audio discontinuity caused by a packet loss is an audio packet loss concealment technology. The audio packet loss concealment technology is designed to detect a packet loss and generate a pseudo audio signal corresponding to the lost packet (which will be referred to hereinafter as "concealment signal").

When an audio encoding technique used is a technique of performing audio encoding while updating internal states of encoder/decoder, encoding parameters to be originally received are not obtained and thus the audio packet loss concealment technology includes performing an update of the internal states of the decoder by use of artificially-generated parameters as well.

The CELP (Code Excited Linear Prediction) encoding is widely used as a technique for performing the audio encoding while updating the internal states of encoder/decoder. In the CELP encoding, an autoregressive model is assumed, and an excitation signal e(n) is filtered by an all-pole synthesis filter a(i) to synthesize an audio signal. Namely, the audio signal s(n) is synthesized according to the below equation. In the equation below, a(i) represents linear prediction coefficients (LP (Linear Prediction) coefficients) and the degree to be used is a value such as P=<NUM>.

In the CELP encoding, the internal states stored include ISF (Immittance Spectral Frequency) parameters as mathematically equivalent representation of the linear prediction coefficients, and a past excitation signal. With an occurrence of a packet loss, these are artificially generated, and there arises a deviation from the original parameters that would be obtained by decoding. An inconsistency of a synthesized audio caused by a deviation of the parameters is perceived as a noise by a listener, which significantly degrades the subjective quality.

The paragraphs below will describe a configuration and an operation of an audio decoder to perform the audio packet loss concealment, using an example where the CELP encoding is used as the audio encoding technique.

A configuration diagram and an operation of the audio decoder are shown in <FIG> and <FIG>. As shown in <FIG>, an audio decoder <NUM> has a packet loss detector <NUM>, an audio code decoder <NUM>, a concealment signal generator <NUM>, and an internal state buffer <NUM>.

The packet loss detector <NUM>, when receiving an audio packet correctly , sends a control signal, and audio codes included in the audio packet, to the audio code decoder <NUM> (normal reception: YES in step S100 in <FIG>). Thereafter, the audio code decoder <NUM> performs decoding of the audio codes and updating of the internal states as described below (steps S200 and S400 in <FIG>). On the other hand, the packet loss detector <NUM>, when failing to receive an audio packet correctly, sends a control signal to the concealment signal generator <NUM> (packet loss: NO in step S100 in <FIG>). Thereafter, the concealment signal generator <NUM> generates a concealment signal and updates the internal states as described below (steps S300 and S400 in <FIG>). The processes of steps S100 to S400 in <FIG> are repeated to the end of communication (or until step S500 results in a determination of YES).

The audio codes include at least encoded ISF parameters <MAT> encoded pitch lags Tjp of the first to fourth subframes, encoded adaptive codebook gains gjp of the first to fourth subframes, encoded fixed codebook gains gjc of the first to fourth subframes, and encoded fixed codebook vectors cj(n) of the first to fourth subframes. The ISF parameters may be replaced by LSF (line spectral frequency) parameters which are mathematically equivalent representation thereof. Although the discussion below uses the ISF parameters, the same discussion may also be true for the case using the LSF parameters.

The internal state buffer includes past ISF parameters <MAT> and, as equivalent representation of <MAT> ISP (Immittance Spectral Pair) parameters <MAT> ISF residual parameters <MAT> past pitch lags Tjp, past adaptive codebook gains gjp, past fixed codebook gains gjc, and an adaptive codebook u(n). It is determined, depending upon a design principle, how many subframes of the past parameters should be included. It is assumed in the present specification that one frame includes four subframes, but another value may be adopted depending upon the design principle.

<FIG> shows an exemplary functional configuration of the audio code decoder <NUM>. As shown in this <FIG>, the audio code decoder <NUM> has an ISF decoder <NUM>, a stability processor <NUM>, an LP coefficient calculator <NUM>, an adaptive codebook calculator <NUM>, a fixed codebook decoder <NUM>, a gain decoder <NUM>, an excitation vector synthesizer <NUM>, a post-filter <NUM>, and a synthesis filter <NUM>. It should be noted, however, that the post-filter <NUM> is not an indispensable constitutive element. In <FIG>, for convenience of explanation, the internal state buffer <NUM> is indicated by a double-dot line inside the audio code decoder <NUM>. However, the internal state buffer <NUM> is not included inside the audio code decoder <NUM>, but is indeed the internal state buffer <NUM> itself shown in <FIG>. The same is also true in the configuration diagrams of the audio code decoder hereinafter.

A configuration diagram of the LP coefficient calculator <NUM> is shown in <FIG> and a processing flow of calculation of LP coefficients from the encoded ISF parameters is shown in <FIG>. As shown in <FIG>, the LP coefficient calculator <NUM> has an ISF-ISP converter 122A, an ISP interpolator 122B, and an ISP-LPC converter 122C.

First described are a functional configuration and its operation associated with the process of calculating the LP coefficients from the encoded ISF parameters (<FIG>).

The ISF decoder <NUM> decodes the encoded ISF parameters to obtain the ISF residual parameters <MAT> and calculates the ISF parameters <MAT> in accordance with the following equation (step S1 in <FIG>). Here, meani represents mean vectors obtained in advance by learning or the like.

The example of using an MA prediction for the calculation of the ISF parameters is described herein, but it is also possible to adopt a configuration to perform calculation of the ISF parameters using an AR prediction as described below. Here, the ISF parameters of the immediately preceding frame are denoted by <MAT> and weight factors of the AR prediction by ρi.

The stability processor <NUM> performs a process according to the below equation so as to place a distance of not less than <NUM> between elements of the ISF parameters in order to secure stability of the filter (step S2 in <FIG>). The ISF parameters are indicative of a line spectrum representing the shape of an audio spectrum envelope, and as the distance between them becomes shorter, peaks of the spectrum become larger, causing resonance. For this reason, the process for securing stability becomes necessary to prevent gains from becoming too large at the peaks of the spectrum. Here, min_dist represents a minimum ISF distance, and isf_min represents a minimum of ISF necessary for securing the distance of min_dist. isf_min is successively updated by adding the distance of min_dist to a value of neighboring ISF. On the other hand, isf_max represents a maximum of ISF necessary for securing the distance of min_dist. isf_max is successively updated by subtracting the distance of min_dist from a value of neighboring ISF.

The ISF-ISP converter 122A in the LP coefficient calculator <NUM> converts <MAT> into ISP parameters <MAT> in accordance with the following equation (step S3 in <FIG>). Here, C is a constant determined in advance.

The ISP interpolator 122B calculates the ISP parameters for the respective subframes from the past ISP parameters <MAT> included in the internal state buffer <NUM> and the foregoing ISP parameters <MAT> in accordance with the below equation (step S4 in <FIG>). Other coefficients may be used for the interpolation.

The ISP-LPC converter 122C converts the ISP parameters for the respective subframes into LP coefficients <MAT> (step S5 in <FIG>). A specific conversion procedure to be used can be the processing procedure described in Non Patent Literature <NUM>. The number of subframes included in a look-ahead signal is assumed to be <NUM> herein, but the number of subframes may differ, depending upon the design principle.

Next described are other configurations and operations in the audio code decoder <NUM>.

The adaptive codebook calculator <NUM> decodes encoded pitch lags to calculate the pitch lags TjP of the first to fourth subframes. Then, the adaptive codebook calculator <NUM> uses the adaptive codebook u(n) to calculate adaptive codebook vectors for the respective subframes in accordance with the below equation. The adaptive codebook vectors are calculated by interpolating the adaptive codebook u(n) by a FIR filter Int(i). Here, the length of the adaptive codebook is denoted by Nadapt. The filter Int(i) used for the interpolation is an FIR filter with a predetermined length <NUM>l + <NUM>, and L' presents the sample number of the subframes. By using the interpolation filter Int(i), the pitch lags can be utilized to the accuracy of decimal places. For the details of the interpolation filter, the method described in Non Patent Literature <NUM> can be referred to.

The fixed codebook decoder <NUM> decodes the encoded fixed codebook vectors to acquire the fixed codebook vectors cj(n) of the first to fourth subframes.

The gain decoder <NUM> decodes the encoded adaptive codebook gains and the encoded fixed codebook gains to acquire the adaptive codebook gains and fixed codebook gains of the first to fourth subframes. For example, the decoding of the adaptive codebook gains and the fixed codebook gains can be carried out, for example, by the below technique described in Non Patent Literature <NUM>. Since the below technique described in Non Patent Literature <NUM> does not use the interframe prediction as used in gain encoding of AMR-WB, it can enhance packet loss resistance.

For example, the gain decoder <NUM> acquires the fixed codebook gain in accordance with the below processing flow.

First, the gain decoder <NUM> calculates the power of the fixed codebook vector. Here, the length of the subframe is defined as Ns.

Next, the gain decoder <NUM> decodes the vector-quantized gain parameter to acquire the adaptive codebook gain <MAT> and the quantized fixed codebook gain <MAT>.

It then calculates a predictive fixed codebook gain as described below from the quantized fixed codebook gain and the aforementioned power of the fixed codebook vector.

Finally, the gain decoder <NUM> decodes the prediction coefficient <MAT> and multiplies it to the prediction gain to acquire the fixed codebook gain.

The excitation vector synthesizer <NUM> multiplies the adaptive codebook vector by the adaptive codebook gain and multiplies the fixed codebook vector by the fixed codebook gain and calculates a sum of them to acquire an excitation signal, as expressed by the following equation.

The post-filter <NUM> subjects the excitation signal vectors, for example, to post-processes such as processes of pitch enhancement, noise enhancement, and low-frequency enhancement. The pitch enhancement, the noise enhancement, and the low-frequency enhancement can be effected by use of the techniques described in Non Patent Literature <NUM>.

The synthesis filter <NUM> synthesizes a decoded signal with the excitation signal as a drive audio source, by linear prediction inverse filtering.

If a pre-emphasis is done in the encoder, a de-emphasis is carried out.

On the other hand, if a pre-emphasis is not done in the encoder, a de-emphasis is not carried out.

The paragraphs below will describe the operation concerning an internal state update.

In order to interpolate parameter upon an occurrence of packet loss, the LP coefficient calculator <NUM> updates the internal states of the ISF parameters by vectors calculated by the following equation.

Here, ωi(-j) represents the ISF parameters j frames prior, which are stored in the buffer. ωiC represents the ISF parameters in speech intervals obtained in advance by learning or the like. β is a constant and can be a value of, e.g., <NUM>, to which the value is not necessarily limited. ωiC and β may be varied by an index to express a property of an encoding target frame, for example, as in the ISF concealment described in Non Patent Literature <NUM>.

Furthermore, the LP coefficient calculator <NUM> also updates the internal states of the ISF residual parameters in accordance with the following equation.

The excitation vector synthesizer <NUM> updates the internal states by the excitation signal vectors in accordance with the below equation.

Furthermore, the excitation vector synthesizer <NUM> updates the internal states of the gain parameters by the following equation.

The adaptive codebook calculator <NUM> updates the internal states of the parameters of the pitch lags by the following equation. <MAT> The range of j is defined as (-<NUM> ≤ j < Mla) but different values may be selected as the range of j, depending upon the the design principle.

<FIG> shows an exemplary functional configuration of the concealment signal generator <NUM>. As shown in this <FIG>, the concealment signal generator <NUM> has an LP coefficient interpolator <NUM>, a pitch lag interpolator <NUM>, a gain interpolator <NUM>, a noise signal generator <NUM>, a post-filter <NUM>, a synthesis filter <NUM>, an adaptive codebook calculator <NUM>, and an excitation vector synthesizer <NUM>. It should be noted, however, that the post-filter <NUM> is not an indispensable constitutive element.

The LP coefficient interpolator <NUM> calculates <MAT> by the following equation. In this respect, ωi(-j) represents the ISF parameters j frames prior, which are stored in the buffer. <MAT> In this equation, <MAT> represents the internal states of the ISF parameters calculated upon normal reception of a packet. α is also a constant and can be a value of, e.g., <NUM> to which the value is not necessarily limited. α may be varied by an index to express a property of an encoding target frame, for example, as in the ISF concealment described in Non Patent Literature <NUM>.

The procedure of obtaining the LP coefficients from the ISF parameters is the same as performed in the case of normal reception of a packet.

The pitch lag interpolator <NUM> uses the internal state parameters about the pitch lags <MAT> to calculate predictive values of the pitch lags <MAT> A specific processing procedure to be used can be the technique disclosed in Non Patent Literature <NUM>.

In order to interpolate the fixed codebook gains, the gain interpolator <NUM> can use the technique according to the below equation as described in Non Patent Literature <NUM>.

The noise signal generator <NUM> generates white noise for the same length as the fixed codebook vectors and uses the resultant noise for the fixed codebook vectors.

The operations of the post-filter <NUM>, the synthesis filter <NUM>, the adaptive codebook calculator <NUM>, and the excitation vector synthesizer <NUM> are the same as those in the aforementioned case of normal reception of a packet.

The internal state update is the same as performed in the case of normal reception of a packet, except for an update of the ISF residual parameters. The updating of the ISF parameters is carried out in accordance with the following equation by the LP coefficient interpolator <NUM>.

As described above, since the CELP encoding involves the internal states, a degradation of audio quality occurs because of a deviation between the parameters obtained by interpolations implemented upon a packet loss and the parameters that would have been used for decoding. Particularly, as to the ISF parameters, intraframe/interframe predictive encoding is carried out, and thus there is the problem that an influence by a packet loss continues even after recovery from the packet loss.

More specifically, a problem of a sudden increase of power is identified in the first frame after recovery from a packet loss occurring in the vicinity of an audio start portion. This is caused for the following reason: That is, in the audio start portion where the power of the excitation signal becomes high, the impulse response of the LP coefficients calculated from the ISF coefficients obtained by the interpolation process upon a packet loss has a higher gain than the one that would have been originally expected for the decoder. This is perceived, according to the subjective quality standard, as an unpleasant discontinuity of audio.

The method described in Patent Literature <NUM> generates the interpolated ISF coefficients for a lost frame. However, since the ISF parameters are generated by a normal decoding process for the first frame after recovery from the loss, it fails to suppress the sudden increase of power.

On the other hand, the method described in Patent Literature <NUM> transmits a gain adjustment parameter (normalized prediction residual power) obtained on the encoding side and uses it for a power adjustment on the decoding side, thereby controlling the power of the excitation signal of a lost packet frame and enabling prevention of the sudden increase of power.

<FIG> shows an exemplary functional configuration of an audio decoder 1X implemented by the technology of Patent Literature <NUM>, and <FIG> shows an exemplary functional configuration of a concealment signal generator 13X. In Patent Literature <NUM>, an audio packet includes auxiliary information of at least a normalized prediction residual power in addition to the parameters described in the conventional technique.

A normalized prediction residual power decoder <NUM> provided in the audio signal generator 1X decodes the auxiliary information of the normalized prediction residual power from a received audio packet to calculate a reference normalized prediction residual power, and outputs it to the concealment signal generator 13X.

Since the constitutive elements of the concealment signal generator 13X, other than normalized prediction residual adjuster <NUM>, are the same as those in the aforementioned conventional technology, only the normalized prediction residual adjuster <NUM> will be described below.

The normalized prediction residual adjuster <NUM> calculates the normalized prediction residual power from the LP coefficients output by the LP coefficient interpolator <NUM>. Next, the normalized prediction residual adjuster <NUM> calculates a synthesis filter gain adjustment coefficient, using the normalized prediction residual power and the reference normalized prediction residual power. Finally, the normalized prediction residual adjuster <NUM> multiplies the excitation signal by the synthesis filter gain adjustment coefficient and output the result to the synthesis filter <NUM>.

The above-described technology of Patent Literature <NUM> can control the power of the concealment signal upon an occurrence of a packet loss in the same manner as performed in the normal reception. However, it is difficult to secure a bit rate necessary for transmission of the foregoing gain adjustment parameter in the process of low-bit-rate audio encoding. In addition, since it is the processing in the concealment signal generator, it is difficult to deal with a sudden change of power caused by a disagreement of the ISF parameters in a recovery frame.

An object of the present invention is therefore to reduce a discontinuity of audio which can occur upon recovery from a packet loss at the audio start point, and thereby improve the subjective quality.

The present invention as described above can reduce a discontinuity of audio possibly occurring subsequent to recovery from a packet loss at the audio start point and thus improve the subjective quality.

Preferred embodiments of an audio signal processing device, an audio signal processing method, and an audio signal processing program according to the present invention will be described below in detail using the drawings. The same elements will be denoted by similar reference signs in the description of the drawings to avoid duplicate descriptions.

The audio signal processing device in the first embodiment has the same configuration as the aforementioned audio decoder <NUM> shown in <FIG> and has a novel feature in the audio code decoder, and thus the audio code decoder will be described below.

<FIG> is a diagram showing a functional configuration of an audio code decoder 12A in the first embodiment, and <FIG> shows a flowchart of the LP coefficient calculation process. The audio code decoder 12A shown in <FIG> is configured by adding a discontinuity detector <NUM> to the aforementioned configuration of <FIG>. Since the present embodiment differs from the conventional technology only in the LP coefficient calculation process, the operations of respective parts associated with the LP coefficient calculation process will be described below.

A discontinuity detector <NUM> refers to a fixed codebook gain gc<NUM> acquired by decoding and a fixed codebook gain gc-<NUM> included in the internal states and compares a change of the gain with a threshold in accordance with the following equation (step S11 in <FIG>).

When the gain change exceeds the threshold, the detector detects an occurrence of a discontinuity (also referred to hereinafter simply as "detects a discontinuity") and outputs a control signal indicating a detection result of a discontinuity occurrence to the stability processor <NUM>.

The following equation may be used for the comparison between the gain change and the threshold.

Furthermore, the comparison between the gain change and the threshold may be made by the following equation, where a gc(c) represents the maximum among the fixed codebook gains of the first to fourth subframes included in the current frame and a gc(p) represents the minimum among the fixed codebook gains included in the internal states.

The flowing equation can also be used.

The above example of the first embodiment shows an example in which a discontinuity detection is conducted using the fixed codebook gain gc-<NUM> of the fourth subframe of the immediately preceding frame (lost frame) and the fixed codebook gain gc<NUM> of the first subframe of the current frame. However, comparison between the gain change and the threshold may be made using averages calculated from the fixed codebook gains included in the internal states and the fixed codebook gains included in the current frame.

The ISF decoder <NUM> performs the same operation as in the conventional technology (step S12 in <FIG>).

The stability processor <NUM> corrects the ISF parameters by the following process when the discontinuity detector <NUM> detects a discontinuity (step S13 in <FIG>).

First, the stability processor <NUM> subjects the ISF parameters <MAT> stored in the internal state buffer <NUM> to a process of expanding a distance between two adjacent element to become M-<NUM> times wider than the ordinary distance. The process of placing a very wide distance than the ordinary distance provides an effect to suppress excessive peaks and dips in the spectrum envelope. Here, min_dist represents the minimum ISF distance, and isf_min represents the minimum of ISF necessary for securing the distance of min_dist. isf_min is successively updated by adding the distance of min_dist to a value of neighboring ISF. On the other hand, isf_max is the maximum of ISF necessary for securing the distance of min_dist. isf_max is successively updated by subtracting the distance of min_dist from a value of neighboring ISF.

Next, a stability processor <NUM> subjects the ISF parameters of the current frame to a process of expanding a distance between two adjacent element to become M<NUM> times wider than the ordinary distance. <NUM> < M<NUM> < M-<NUM> is assumed herein, but it is also possible to set one of r M-<NUM> and M<NUM> to <NUM> and the other to a value larger than <NUM>.

Furthermore, the stability processor <NUM> performs the following process in the same manner as carried out in the ordinary decoding process, when the discontinuity detector detects no discontinuity.

The minimum distance placed between elements when a discontinuity is detected may be varied depending upon the frequency of ISF. The minimum distance placed between elements when a discontinuity is detected needs only to be different from the minimum distance placed between elements in the ordinary decoding process.

The ISF-ISP converter 122A in the LP coefficient calculator <NUM> converts the ISF parameters <MAT> into the ISP parameters <MAT> respectively, in accordance with the following equation (step S14 in <FIG>). Here, C is a constant determined in advance.

The ISP interpolator 122B calculates the ISP parameters for the respective subframes from the past ISP parameters <MAT> and the foregoing ISP parameters <MAT> in accordance with the following equation (step S15 in <FIG>). Other coefficients may be used for the interpolation.

The ISP-LPC converter 122C converts the ISP parameters for the respective subframes into the LP coefficients <MAT> (step S16 in <FIG>). Here, the number of subframes included in a look-ahead signal was assumed to be <NUM>, but the number of subframes may differ depending upon the design principle. A specific conversion procedure to be used can be the processing procedure described in Non Patent Literature <NUM>.

Furthermore, the ISF-ISP converter 122A updates the ISF parameters stored in the internal state buffer <NUM> <MAT> in accordance with the following equation. <MAT> At this time, even when a discontinuity is detected, the ISF-ISP converter 122A may carry out the below procedure to update the ISF parameters <MAT> stored in the internal state buffer, using the calculation result of the ISF parameters.

As in the above first embodiment, a discontinuity of decoded audio can be determined with the quantized codebook gains used in the calculation of the excitation signal and the ISF/LSF parameters (e.g., the distance between elements of the ISF/LSF parameters given for ensuring stability of the synthesis filter) can be corrected according to a result of the determination for a discontinuity. This reduces the discontinuity of audio which can occur upon recovery from a packet loss at the audio start point, and thereby improves the subjective quality.

<FIG> is a diagram showing a functional configuration of an audio code decoder <NUM> according to a modification example of the first embodiment. Since it differs from the configuration of the conventional technology shown in <FIG> only in the discontinuity detector <NUM> and the second stability processor <NUM>, the operations of these will be described. The second stability processor <NUM> has a gain adjustor 121X and a gain multiplier 121Y, and a processing flow of the second stability processor <NUM> is shown in <FIG>.

The discontinuity detector <NUM> refers to the fixed codebook gain gc<NUM> obtained by decoding and the fixed codebook gain gc-<NUM> included in the internal states and compares the gain change with a threshold, in the same manner as performed by the discontinuity detector <NUM> in the first embodiment. Then, the discontinuity detector <NUM> sends to the gain adjustor 121X, a control signal including information about whether the gain change exceeds the threshold.

The gain adjustor 121X reads from the control signal the information about whether the gain change exceeds the threshold, and, when the gain change exceeds the threshold, it outputs a predetermined gain gon to the gain multiplier 121Y. On the other hand, when the gain change does not exceed the threshold, the gain adjustor 121X outputs a predetermined gain goff to the gain multiplier 121Y. This operation of the gain adjustor 121X corresponds to step S18 in <FIG>.

The gain multiplier 121Y multiplies the synthesized signal output from the synthesis filter <NUM> by the foregoing gain gon or gain goff (step S19 in <FIG>) and outputs the resultant decoded signal.

Here, the audio code decoder may be configured such that the LP coefficient calculator <NUM> outputs the LP coefficients or the ISF parameters to feed them to the second stability processor <NUM> (as indicated by a dotted line from the LP coefficient calculator <NUM> to the gain adjustor 121X in <FIG>). In this case, the gains to be multiplied are determined using the LP coefficients or the ISF parameters calculated by the LP coefficient calculator <NUM>.

By adding the second stability processor <NUM> to the audio code decoder <NUM> and adjusting the gain, depending upon whether the gain change exceeds the threshold as described in the above modification example, an appropriate decoded signal can be obtained.

The second stability processor <NUM> may be configured to multiply the excitation signal by the foregoing calculated gain and output the result to the synthesis filter <NUM>.

An audio signal processing device according to the second embodiment has the same configuration as that of the aforementioned audio decoder <NUM> in <FIG> and has a novel feature in an audio code decoder, and thus the audio code decoder will be described below. <FIG> shows an exemplary functional configuration of the audio code decoder 12B, <FIG> shows an exemplary functional configuration associated with the calculation process of the LP coefficients, and <FIG> shows a flow of the calculation process of the LP coefficients. The audio code decoder 12B in <FIG> is configured by adding the discontinuity detector <NUM> to the aforementioned configuration shown in <FIG>.

The ISF decoder <NUM> calculates the ISF parameters in the same manner as performed in the conventional technology (step S21 in <FIG>).

The stability processor <NUM> performs the process of placing a distance of not less than <NUM> between elements of the ISF parameters <MAT> in order to secure the stability of the filter in the same manner as performed in the conventional technology (step S22 in <FIG>).

The ISF-ISP converter 122A converts the ISF parameters output by the stability processor <NUM> into the ISP parameters in the same manner as performed in the first embodiment (step S23 in <FIG>).

The ISP interpolator 122B, in the same manner as performed in the first embodiment (step S24 in <FIG>), calculates the ISP parameters for the respective subframes from the past ISP parameters <MAT> and the ISP parameters <MAT> obtained by the conversion by the ISF-ISP converter 122A.

The ISP-LPC converter 122C, in the same manner as performed in the first embodiment (step S25 in <FIG>), converts the ISP parameters for the respective subframes into the LP coefficients <MAT>. Here, the number of subframes included in the look-ahead signal is assumed to be <NUM>, but the number of subframes may differ depending upon the design principle.

The internal state buffer <NUM> updates the ISF parameters stored in the past with the new ISF parameters.

The discontinuity detector <NUM> reads the LP coefficients of the fourth subframe in the lost packet frame from the internal state buffer <NUM> and calculates the power of the impulse response of the LP coefficients of the fourth subframe in the lost packet frame. The LP coefficients of the fourth subframe in the lost packet frame to be used can be the coefficients output by the LP coefficient interpolator <NUM> included in the concealment signal generator <NUM> shown in <FIG> and accumulated in the internal state buffer <NUM> upon the packet loss.

Then, the discontinuity detector <NUM> detects a discontinuity, for example, by the below equation (step S26 in <FIG>).

When the gain change does not exceed the threshold ( NO in step S27 of <FIG>), the discontinuity detector <NUM> does not detect an occurrence of a discontinuity, and the ISP-LPC converter 122C outputs the LP coefficients and ends the processing. On the other hand, when the gain change exceeds the threshold ( YES in step S27 of <FIG>), the discontinuity detector <NUM> detects an occurrence of a discontinuity and sends a control signal indicative of a result of the detection for an occurrence of a discontinuity to the stability processor <NUM>. When receiving the control signal, the stability processor <NUM> corrects the ISP parameters in the same manner as performed in the first embodiment (step S28 in <FIG>). The subsequent operations of the ISF-ISP converter 122A, ISP interpolator 122B, and ISP-LPC converter 122C (steps S29, S2A, and S2B in <FIG>) are the same as above.

As discussed in the above second embodiment, a discontinuity of decoded audio can be determined by the power of the excitation signal, and the discontinuous audio is reduced to improve the subjective quality in the same manner as performed in the first embodiment.

Upon a detection of discontinuity, the ISF parameters may be corrected by another method. The third embodiment differs from the first embodiment only in the stability processor <NUM>, and thus only the operation of the stability processor <NUM> will be described.

When the discontinuity detector <NUM> detects a discontinuity, the stability processor <NUM> performs the following process to correct the ISF parameters.

With respect to the ISF parameters stored in the internal state buffer <NUM>, <MAT> the stability processor <NUM> replaces the ISF parameters up to a low-order P' dimension (<NUM> < P' ≤ P) in accordance with the below equation. Here, the following definition is adopted. <MAT> <MAT>.

The stability processor <NUM> may overwrite the ISF parameters of the low-order P' dimensions with P'-dimension vectors obtained in advance by learning as follows.

Next, as to the ISF parameters of the current frame, the stability processor <NUM> may, as performed in the first embodiment, perform the process of expanding the distance between elements to becomeM<NUM> times wider than the ordinary distance or may determine them in accordance with the below equation. Here, the following definition is adopted. <MAT> <MAT>.

The stability processor <NUM> may overwrite them with P'-dimensional vectors learned in advance.

Furthermore, the foregoing P'-dimensional vectors may be learned in the decoding process or may be defined, for example, as follows. <MAT> In a frame at the start of decoding, however, ωi-<NUM> may be defined as predetermined P'-dimensional vector ωiinit.

As discussed in the above third embodiment, the distance obtained by equally dividing the ISF/LSF parameters into those of a predetermined dimension can be used as the distance between elements of the ISF/LSF parameters given for ensuring the stability of the synthesis filter, whereby the discontinuous audio is reduced to improve the subjective quality as performed in the first and second embodiments.

A fourth embodiment will be described in which the encoding side detects an occurrence of a discontinuity and transmits a discontinuity determination code (indicative of a detection result) as included in audio codes to the decoding side and also in which the decoding side determines the operation of the stability process, based on the discontinuity determination code included in the audio codes.

<FIG> shows an exemplary functional configuration of the encoder <NUM>, and <FIG> is a flowchart showing the processes performed in the encoder <NUM>. As shown in <FIG>, the encoder <NUM> has an LP analyzer/encoder <NUM>, a residual encoder <NUM>, and a code multiplexer <NUM>.

An exemplary functional configuration of the LP analyzer/encoder <NUM> among them is shown in <FIG>, and a flowchart showing the processes performed in the LP analyzer/encoder <NUM> is shown in <FIG>. As shown in <FIG>, the LP analyzer/encoder <NUM> has an LP analyzer <NUM>, an LP-ISF converter <NUM>, an ISF encoder <NUM>, a discontinuity determiner <NUM>, an ISF concealer <NUM>, an ISF-LP converter <NUM>, and an ISF buffer <NUM>.

In the LP analyzer/encoder <NUM>, the LP analyzer <NUM> performs a linear prediction analysis on an input signal to obtain linear prediction coefficients (step T41 in <FIG> and step U41 in <FIG>). For the calculation of linear prediction coefficients, an autocorrelation function is first calculated from the audio signal, and then the Levinson-Durbin algorithm or the like can be applied.

The LP-ISF converter <NUM> converts the calculated linear prediction coefficients into the ISP parameters in the same manner as performed in the first embodiment (steps T42, U42). The conversion from linear prediction coefficients into ISF parameters may be implemented by use of the method described in the Non Patent Literature.

The ISF encoder <NUM> encodes the ISF parameters using a predetermined method to calculate ISF codes (steps T43, U43) and outputs quantized ISF parameters obtained in the process of encoding to the discontinuity determiner <NUM>, the ISF concealer <NUM>, and the ISF-LP converter <NUM> (step U47). Here, the quantized ISF parameters are equal to the ISF parameters obtained by an inverse quantization of the ISF codes. A method of encoding may be vector-encoding, or encoding by a vector quantization or the like of error vectors from ISFs of the immediately preceding frame and mean vectors determined in advance by learning.

The discontinuity determiner <NUM> encodes a discontinuity determination flag stored in an internal buffer (not shown) built in the discontinuity determiner <NUM> and outputs a resultant discontinuity determination code (step U47). In addition, the discontinuity determiner <NUM> uses concealment ISF parameters <MAT> read from the ISF buffer <NUM> and the quantized ISF parameters <MAT> to make a determination on a discontinuity in accordance with the below equation (steps T44, U46). Here, Thresω represents a threshold determined in advance, and P' an integer satisfying the following equation (<NUM> < P' ≤ P).

The example is described above in which the discontinuity determination is made using the Euclidean distances between the ISF parameters. However, the discontinuity determination may be made by other methods.

The ISF concealer <NUM> calculates the concealment ISF parameters from the quantized ISF parameters by the same process as performed by the decoder-side ISF concealer and outputs the resultant concealment ISF parameters to the ISF buffer <NUM> (steps U44, U45). The operation of the ISF concealment process may be performed by any method as long as it is the same process as that of the decoder-side packet loss concealer.

The ISF-LP converter <NUM> calculates quantized linear prediction coefficients by converting the foregoing quantized ISF parameters and outputs a resultant quantized linear prediction coefficients to the residual encoder <NUM> (step T45). A method used for converting the ISF parameters into the quantized linear prediction coefficients may be the method described in the Non Patent Literature.

The residual encoder <NUM> filters the audio signal by use of the quantized liner prediction coefficients to calculate residual signals (step T46).

Next, the residual encoder <NUM> encodes the residual signals by encoding means using CELP or TCX (Transform Coded Excitation) or by encoding means switchably using CELP and TCX and outputs resultant residual codes (step T47). Since the operation of the residual encoder <NUM> is less relevant to the present invention, description thereof is omitted herein.

The code multiplexer <NUM> assembles the ISF codes, the discontinuity determination code and the residual codes in a predetermined order and outputs resultant audio codes (step T48).

An audio signal processing device according to the fourth embodiment has the same configuration as that of the aforementioned audio decoder <NUM> in <FIG> and has a novel feature in the audio code decoder, and thus the audio code decoder will be described below. <FIG> shows an exemplary functional configuration of an audio code decoder 12D, and <FIG> is a flowchart showing the process of calculating the LP coefficients. The audio code decoder 12D shown in <FIG> is configured by adding the discontinuity detector <NUM> to the aforementioned configuration shown in <FIG>.

The ISF decoder <NUM> decodes the ISF codes and outputs resultant codes to the stability processor <NUM> and the internal state buffer <NUM> (step S41 in <FIG>).

The discontinuity detector <NUM> decodes the discontinuity determination code and outputs a resultant discontinuity detection result to the stability processor <NUM> (step S42 in <FIG>).

The stability processor <NUM> performs the stability process according to the discontinuity detection result (step S43 in <FIG>). The processing procedure of the stability processor to be used can be the same method as executed in the first embodiment and the third embodiment.

The stability processor <NUM> may perform the stability process as described below, on the basis of other parameters included in the audio codes, in addition to the discontinuity detection result acquired from the discontinuity determination code. For example, the stability processor <NUM> may be configured to perform the stability process in such a manner that an ISF stability stab is calculated in accordance with the below equation and that when the ISF stability exceeds a threshold, even if the discontinuity determination code shows a detection of a discontinuity, the process is performed as if no discontinuity is detected. Here, C is a constant determined in advance.

The ISF-ISP converter 122A in the LP coefficient calculator <NUM> converts the ISF parameters into the ISP parameters by the same processing procedure as performed in the first embodiment (step S44 in <FIG>).

The ISP interpolator 122B calculates the ISP parameters for the respective subframes by the same processing procedure as performed in the first embodiment (step S45 in <FIG>).

The ISP-LPC converter 122C converts the ISP parameters calculated for the respective subframes into the LPC parameters by the same processing procedure as performed in the first embodiment (step S46 in <FIG>).

In the fourth embodiment as described above, the encoding side performs the discontinuity determination (the discontinuity determination using the Euclidian distances between concealment ISF parameters and quantized ISF parameters, as an example) encodes auxiliary information about a result of the determination and outputs encoded information to the decoding side, and the decoding side determine a discontinuity using the auxiliary information obtained by decoding. In this manner, the appropriate processing can be executed according to the discontinuity determination result made by the encoding side while the encoding side and the decoding side work in concert with each other.

The functional configuration of the encoder is the same as that of the fourth embodiment shown in <FIG>, and the processing flow of the encoder is the same as the processing flow of the fourth embodiment shown in <FIG>. The below will describe the LP analyzer/encoder according to the fifth embodiment which is different from that in the fourth embodiment.

<FIG> shows an exemplary functional configuration of the LP analyzer/encoder, and <FIG> shows a flow of the processes performed by the LP analyzer/encoder. As shown in <FIG>, the LP analyzer/encoder <NUM> has the LP analyzer <NUM>, the LP-ISF converter <NUM>, the ISF encoder <NUM>, the discontinuity determiner <NUM>, the ISF concealer <NUM>, the ISF-LP converter <NUM>, and the ISF buffer <NUM>.

In this LP analyzer/encoder <NUM>, the LP analyzer <NUM> performs the linear prediction analysis on the input signal by the same process as performed in the fourth embodiment to obtain the linear prediction coefficients (step U51 in <FIG>).

The LP-ISF converter <NUM> converts the calculated linear prediction coefficients into the ISF parameters by the same process as performed in the fourth embodiment (step U52 in <FIG>). The method described in the Non Patent Literature may be used for the conversion from the linear prediction coefficients into the ISF parameters.

The ISF encoder <NUM> reads the discontinuity determination flag stored in the internal buffer (not shown) of the discontinuity determiner <NUM> (step U53 in <FIG>).

The ISF encoder <NUM> calculates the ISF codes by vector-quantization of ISF residual parameters ri calculated by the below equation (step U54 in <FIG>). Here, the ISF parameters calculated by the LP-ISF converter are denoted by ωi and mean vectors, which are meani, obtained in advance by learning.

Next, the ISF encoder <NUM> uses the quantized ISF residual parameters <MAT> obtained by quantization of the ISF residual parameters ri to update the ISF residual parameter buffer in accordance with the following equation (step U55 in <FIG>).

The ISF encoder <NUM> calculates the ISF codes by vector-quantization of the ISF residual parameters ri calculated by the below equation (step U54 in <FIG>). Here, the ISF residual parameters obtained by decoding in the immediately preceding frame are denoted as follows. <MAT> <MAT>.

By the above procedure, the ISF encoder <NUM> calculates the ISF codes and outputs quantized ISF parameters obtained in the process of encoding to the discontinuity determiner <NUM>, the ISF concealer <NUM>, and the ISF-LP converter <NUM>.

The ISF concealer <NUM> calculates the concealment ISF parameters from the quantized ISF parameters by the same process as performed by the decoder-side ISF concealer in the same manner as executed in the fourth embodiment and outputs them to the ISF buffer <NUM> (steps U56, U58 in <FIG>). The operation of the ISF concealment process may be performed by any method as long as it is the same process as that of the decoder-side packet loss concealer.

The discontinuity determiner <NUM> performs a determination of a discontinuity by the same process as performed in the fourth embodiment and stores a determination result in the internal buffer (not shown) of the discontinuity determiner <NUM> (step U57 in <FIG>).

The ISF-LP converter <NUM> converts the quantized ISF parameters, in the same manner as performed in the fourth embodiment, to calculate the quantized linear prediction coefficients and outputs them to the residual encoder <NUM> (<FIG>) (step U58 in <FIG>).

An audio signal processing device according to the fifth embodiment has the same configuration as that of the aforementioned audio decoder <NUM> in <FIG> and has a novel feature in the audio code decoder, and thus the audio code decoder will be described below. <FIG> shows an exemplary functional configuration of the audio code decoder 12E, and <FIG> shows a flow of the calculation process performed by the LP coefficients. The audio code decoder 12E shown in <FIG> is configured by adding the discontinuity detector <NUM> to the aforementioned configuration shown in <FIG>.

The discontinuity detector <NUM> decodes the discontinuity determination code and outputs the resultant discontinuity determination flag to the ISF decoder <NUM> (step S51 in <FIG>).

The ISF decoder <NUM> calculates the ISF parameters as follows, depending upon the value of the discontinuity determination flag, and outputs the ISF parameters to the stability processor <NUM> and the internal state buffer <NUM> (step S52 in <FIG>).

The ISF decoder <NUM> uses the quantized ISF residual parameters <MAT> obtained by decoding of the ISF codes, and the mean vectors mean; obtained in advance by learning to obtain the quantized ISF parameters <MAT> in accordance with the following equation.

Next, the ISF decoder <NUM> updates the ISF residual parameters stored in the internal state buffer <NUM> in accordance with the following equation.

The ISF decoder <NUM> reads, from the internal state buffer <NUM>, the ISF residual parameters <MAT> obtained by decoding of the immediately preceding frame and uses the resultant ISF residual parameters <MAT> the mean vectors meani obtained in advance by learning and the quantized ISF residual parameters <MAT> obtained by decoding of the ISF codes to calculate the quantized ISF parameters <MAT> in accordance with the following equation.

The stability processor <NUM> performs the same process as performed in the first embodiment (step S53 in <FIG>) when a discontinuity is not detected.

The ISF-ISP converter 122A in the LP coefficient calculator <NUM> converts the ISF parameters into the ISP parameters by the same processing procedure as described in the first embodiment (step S54 in <FIG>).

The ISP interpolator 122B calculates the ISP parameters for the respective subframes by the same processing procedure as performed in the first embodiment (step S55 in <FIG>).

The ISP-LPC converter 122C, by the same processing procedure as performed in the first embodiment (step S56 in <FIG>), converts the ISP parameters calculated for the respective subframes into the LPC parameters.

In the fifth embodiment as described above, the encoding side is configured as follows: When the discontinuity determination flag does not indicate a detection of a discontinuity, the vector quantization of the ISF residual parameters is carried out using the ISF residual parameters obtained by decoding of the immediately preceding frame. On the other hand, when the discontinuity determination flag indicates a detection of a discontinuity, the encoder avoids using the ISF residual parameters obtained by decoding of the immediately preceding frame. Similarly, the decoding side is configured as follows: When the discontinuity determination flag does not indicate a detection of a discontinuity, the quantized ISF parameters are calculated using the ISF residual parameters obtained by decoding of the immediately preceding frame. On the other hand, when the discontinuity determination flag indicates a detection of discontinuity, the decoder avoids using the ISF residual parameters obtained by decoding of the immediately preceding frame. In this manner, the appropriate processing according to a discontinuity determination result can be executed while the encoding side and the decoding side work in concert with each other.

The above first to fifth embodiments may be applied in combination. For example, as described in the fourth embodiment, the decoding side decodes the discontinuity determination code included in the audio codes from the encoding side to detect a discontinuity. When a discontinuity is detected, it may carry out the subsequent operation as follows.

For the ISF parameters <MAT> stored in the internal state buffer, the ISF parameters up to the low-degree P' dimension (<NUM> < P' ≤ P) are replaces in accordance with the following equation as described in the third embodiment.

On the other hand, the ISF parameters of the current frame are calculated in accordance with the following equation as described in the fifth embodiment.

Thereafter, using the ISF parameters obtained as described above, the LP coefficients are obtained by the processes of the ISF-ISP converter 122A, the ISP interpolator 122B, and the ISP-LPC converter 122C as performed in the first embodiment.

It is also effective to adopt optional combinations of the first to fifth embodiments as described above.

It may be considered in the decoding operation according to the above first to sixth embodiments and their modifications, how the frame is lost (e.g., whether a single frame is lost or consecutive frames are lost). In the seventh embodiment, it suffices that a discontinuity detection is made using, for example, the result of decoding of the discontinuity determination code included in the audio codes, and the method of how it should be performed is not limited to the above.

An audio signal processing device according to the seventh embodiment has the same configuration as that of the aforementioned audio decoder <NUM> in <FIG> and has a novel feature in the audio code decoder, and thus the audio code decoder will be described below.

<FIG> shows an exemplary configuration of the audio decoder <NUM> according to the seventh embodiment, and <FIG> shows a flowchart of the processes performed in the audio decoder. As shown in <FIG>, in addition to the aforementioned audio code decoder <NUM>, the concealment signal generator <NUM> and the internal state buffer <NUM>, the audio decoder <NUM> has a reception state determiner <NUM> that determines packet reception states in some past frames and stores a packet loss history.

The reception state determiner <NUM> determines a packet reception state and updates the packet loss history information, based on a determination result (step S50 in <FIG>).

When a packet loss is detected (NO in step S100), the reception state determiner <NUM> outputs a packet loss detection result of the pertinent frame to the concealment signal generator <NUM>, and the concealment signal generator <NUM> generates the concealment signal as described above and updates the internal states (steps S300, S400). The concealment signal generator <NUM> may also utilize the packet loss history information for interpolation of parameters or the like.

On the other hand, when no packet loss is detected (YES in step S100), the reception state determiner <NUM> outputs the packet loss history information including a packet loss detection result of the pertinent frame and the audio codes included in the received packet to the audio code decoder <NUM>, and the audio code decoder <NUM> decodes the audio codes as described before and updates the internal states (steps S200, S400).

Thereafter, the processes of steps S50 to S400 are repeated until the communication ends (or until step S500 results in a determination of YES).

<FIG> shows an exemplary functional configuration of the audio code decoder <NUM>, and <FIG> shows a flowchart of the calculation processes performed by the LP coefficients. An example will be described below using the packet loss history information only for the LP coefficient calculator <NUM>, but the audio code decoder may be configured to use the packet loss history information for other constitutive elements.

Since the audio code decoder <NUM> has the same configuration as described in the first embodiment, except for the configuration associated with the calculation process of LP coefficients, the below will describe the configuration and its operation associated with the calculation process of LP coefficients.

The ISF decoder <NUM> decodes the ISF codes in the same manner as performed in the first embodiment and outputs the ISF parameters to the stability processor <NUM> (step S71 in <FIG>).

The discontinuity detector <NUM> refers to the packet loss history information to determine the reception state (step S72). The discontinuity detector <NUM> may be designed, for example, as follows: It stores a specific reception pattern which indicates, for example, a packet loss occurred three frames prior, a normal reception occurred two frames prior, and a packet loss occurred one frame prior. When the reception pattern is recognized which has been looked for, it sets a reception state flag to off and, otherwise, it sets the reception state flag to on.

Furthermore, the discontinuity detector <NUM> detects a discontinuity in the same manner as described in one of the first to sixth embodiments.

Then, the stability processor <NUM> performs the stability process according to the reception state flag and a result of the discontinuity detection, for example, as described below (step S73).

When the reception state flag is off, the stability processor <NUM> performs the same process as performed when a discontinuity is not detected, regardless of a result of the discontinuity detection.

On the other hand, when the reception flag is on and when the result of the discontinuity detection indicates that a discontinuity is not detected, the stability processor <NUM> performs the same process as performed when a discontinuity is not detected.

Furthermore, when the reception flag is on and when the result of the discontinuity detection is detection of discontinuity, the stability processor <NUM> performs the same process as performed when a discontinuity is detected.

Thereafter, the operations (steps S74 to S76) of the ISF-ISP converter 122A, the ISP interpolator 122B, and the ISP-LPC converter 122C in the LP coefficient calculator <NUM> are performed in the same manners as performed in the first embodiment.

In the seventh embodiment as described above, the stability process is carried out depending upon a result of the discontinuity detection and the state of the reception state flag, whereby more accurate processing can be executed while it is considered how the frame is lost (e.g., whether a single frame is lost or consecutive frames are lost).

The below will describe audio signal processing programs that program a computer to operate as an audio signal processing device according to the present invention.

<FIG> is a drawing showing various exemplary configurations of the audio signal processing programs. <FIG> is an exemplary hardware configuration of the computer, and <FIG> shows a schematic view of a computer. Audio signal processing programs P1-P4 (which will be referred to hereinafter generally as "audio signal processing program P") shown in <FIG>, respectively, can program the computer C10 shown in <FIG> and <FIG> to operate as an audio signal processing device. It should be noted that the audio signal processing program P described in the present specification can be implemented not only on the computer as shown in <FIG> and <FIG> but also on any information processing device such as a cell phone, a personal digital assistance, or a portable personal computer.

The audio signal processing program P can be provided in a form stored in a recording medium M. Examples of the recording medium M include recording media such as flexible disc, CD-ROM, DVD, or ROM, semiconductor memories, and so on.

As shown in <FIG>, the computer C10 has a reading device C12 such as a flexible disc drive unit, a CD-ROM drive unit, or a DVD drive unit, a working memory (RAM) C14, a memory C16 for storing a program stored in the recording medium M, a display C16, a mouse C20 and a keyboard C22 as input devices, a communication device C24 for executing transmission/reception of data or the like, and a central processing unit (CPU) C26 for controlling execution of the program.

When the recording medium M is put into the reading device C12, the computer C10 becomes accessible to the audio signal processing program P stored in the recording medium M through the reading device C12 and becomes able to operate as an audio signal processing device programmed by the audio signal processing program P.

The audio signal processing program P may be one provided as computer data signal W superimposed on a carrier wave, as shown in <FIG>, transmitted through a network. In this case, the computer C10 stores the audio signal processing program P received by the communication device C24 into the memory C16 and then can execute the audio signal processing program P.

The audio signal processing program P can be configured by adopting the various configurations shown in <FIG>. These correspond to the configurations recited in claims <NUM> to <NUM> associated with the audio signal processing programs as set forth in the scope of claims. For example, the audio signal processing program P1 shown in <FIG> has a discontinuity detection module P11 and a discontinuity correction module P12. The audio signal processing program P2 shown in <FIG> has an ISF/LSF quantization module P21, an ISF/LSF concealment module P22, a discontinuity detection module P23, and an auxiliary information encoding module P24. The audio signal processing program P3 shown in <FIG> has a discontinuity detection module P31, an auxiliary information encoding module P32, and an ISF/LSF quantization module P33. The audio signal processing program P4 shown in <FIG> has an auxiliary information decoding module P41, a discontinuity correction module P42, and an ISF/LSF decoding module P43.

By implementing the various embodiments described above, the subjective quality can be improved while reducing a discontinuous audio which can occur in the recovery from a packet loss at the audio start point.

The stability processor, which is the first feature of the invention, is configured so that when a discontinuity is detected in the first packet which is received correctly after a packet loss occurs, for example, a distance between elements of the ISF parameters is set wider than normal, whereby it can prevent the gain of the LP coefficients from becoming too large. Since it can prevent both the gain of the LP coefficient and the power of the excitation signal from increasing, a discontinuity of the synthesized signal is reduced, whereby a degradation of the subjective quality can be suppressed. Furthermore, the stability processor may reduce a discontinuity of the synthesized signal by multiplying the synthesized signal by the gain calculated by using the LP coefficients or the like.

The discontinuity detector, which is the second feature of the invention, monitors the gain of the excitation signal included in the first packet which is received correctly after a packet loss occurs, and determines a discontinuity for a packet whose gain of the excitation signal increased more than a certain level.

Claim 1:
An audio signal processing device comprising:
an immittance spectral frequency/line spectral frequency, ISF/LSF, quantizer configured to quantize ISF/LSF parameters;
an ISF/LSF concealer (<NUM>) configured to generate concealment ISF/LSF parameters representative of concealment information about the ISF/LSF parameters;
a discontinuity detector (<NUM>, <NUM>) configured to determine an occurrence of a discontinuity which occurs in a first audio packet which is received correctly after an occurrence of a packet loss, the discontinuity detector (<NUM>, <NUM>) being configured to use distances between the quantized ISF/LSF parameters obtained in a quantization process of the ISF/LSF quantizer and the concealment ISF/LSF parameters generated by the ISF/LSF concealer (<NUM>) to determine an occurrence of a discontinuity; and
an auxiliary information encoder configured to encode auxiliary information for determination on an occurrence of a discontinuity.