JP5056049B2 - Audio data decoding device - Google Patents

Description

The present invention relates to a technology of a speech data decoding device, a speech data conversion device, and an error compensation method.

When audio data is transmitted using a circuit switching network or a packet network, audio signals are transmitted and received by encoding and decoding the audio data. As this audio compression method, for example, I
TU-T (International Telecommunication Union Telecommunication Standardizatio
n sector) recommendation G.711 system or CELP (Code-Excited Linear Prediction) system is known.

When audio data encoded by these compression methods is transmitted, a part of the audio data may be lost due to a radio error or network congestion. As error compensation for the missing voice data, a voice signal is generated for the missing voice data based on the information of the voice data received before the missing voice data.

Therefore, as a related technique for reducing deterioration in sound quality caused when generating a sound signal of missing sound data, Patent Document 1 updates a filter memory value using sound frame data included in a packet received late. The technology to do is disclosed. That is, when a lost packet is received with a delay, the filter memory value used in the pitch filter or the filter representing the spectral outline is updated using the audio frame data included in the packet.

Further, in Patent Document 2, in ADPCM (Adaptive Differential Pulse Code Modulation) encoding, even if correct encoded data is received after loss of encoded data, it is uncomfortable due to a state mismatch between the predictor on the encoding side and the decoding side. A technique for solving the problem of outputting abnormal sound is disclosed. That is, for a predetermined time after the packet loss transitions from “detected” to “non-detected”, the detection state control unit gradually decreases the interpolation signal generated based on the past audio data, and the coding side over time Since the state of the predictor on the decoding side gradually matches and the speech signal becomes normal, the speech signal is gradually increased. As a result, this technique has an effect of not outputting abnormal sound even immediately after recovering from the lack of encoded data.

Further, Patent Document 3 discloses a technique for calculating a linear prediction count from a speech signal and generating a speech signal from the linear prediction count.

JP 2002-268697 A JP 2005-294917 A Japanese Patent Laid-Open No. 11-305797

The conventional error compensation method for audio data is a simple method that repeats a past audio waveform. Therefore, although the above-described technique has been disclosed, there is still room for improvement.

Therefore, it is required to solve the problem that the sound quality deteriorates when an error compensation method for audio data is used.

An audio data decoding apparatus according to the present invention is an audio data decoding apparatus that outputs an interpolation signal for interpolating a loss in audio data, detects a loss, and detects that audio data corresponding to the loss has been received with a delay. A loss detection unit that, when a loss is detected, a first audio data decoder that outputs a memory such as a synthesis filter to the memory storage unit;
Decoded speech that uses speech filter data such as a synthesis filter for pre-loss speech data stored in the memory storage unit to decode the speech data corresponding to the loss, and then decodes the speech data next to the speech data corresponding to the loss A second audio data decoder that generates a signal; and an audio signal output unit that outputs the signal while changing a ratio of the decoded audio signal to all output audio signals.

The parameter may be a spectrum parameter, a delay parameter, an adaptive codebook gain, a normalized residual signal, or a normalized residual signal gain.

According to the present invention, good sound quality is expected in an interpolation signal to be compensated when a loss of audio data occurs.

Embodiments of the present invention will be described with reference to the drawings. However, such a form does not limit the technical scope of the present invention.

  Embodiment 1 of the present invention will be described below with reference to FIGS. 1 and 2.

FIG. 1 shows a configuration of a decoding apparatus for audio data encoded by a waveform encoding system typified by the G.711 system. The speech data decoding apparatus according to the first embodiment includes a loss detector 101, a speech data decoder 102, a speech data analyzer 103, a parameter correction unit 104, a speech synthesis unit 105, and a speech signal output unit 106. Here, the audio data refers to data obtained by encoding a series of audio, and also refers to audio data composed of at least one audio frame.

The loss detector 101 outputs the received audio data to the audio data decoder 102, detects whether the received audio data has been lost, and displays the loss detection result as an audio data decoder 102, a parameter correction unit 104, and an audio signal output unit 106. Output to.

The audio data decoder 102 decodes the input audio data and outputs the decoded audio signal to the audio data output unit 106 and the audio data analyzer 103.

The voice data analyzer 103 divides the decoded voice signal into frames (for example, 20 ms), and extracts a spectral parameter representing the spectral characteristics of the voice signal using linear prediction analysis on the divided signal. Next, the audio data analyzer 103 divides the frame-divided audio signal into subframes (for example, 5 ms), and delay parameters corresponding to the pitch period as parameters in the adaptive codebook based on past sound source signals for each subframe. And extract the adaptive codebook gain. Also, the audio data analyzer 103 predicts the pitch of the audio signal of the corresponding subframe using the adaptive codebook. Further, the voice data analyzer 103 normalizes the residual signal obtained by pitch prediction, and extracts the normalized residual signal and the normalized residual signal gain. And the extracted spectral parameters, delay parameters,
The adaptive codebook gain, normalized residual signal, or normalized residual signal gain (hereinafter collectively referred to as parameters) is output to the parameter correction unit 104. Here, two or more of the spectral parameters, delay parameters, adaptive codebook gain, normalized residual signal, and normalized residual signal gain may be extracted.

Based on the loss detection result input from the loss detector 101, the parameter correction unit 104 receives the spectral parameter, delay parameter, adaptive codebook gain, normalized residual signal, or normalized residual signal input from the speech data analyzer 103. The gain is used as it is, or a random number of ± 1% is added, or the gain is reduced.
Further, the parameter correction unit 104 outputs this value to the speech synthesis unit 105. The reason for correcting these values is to avoid generating an unnatural audio signal by repetition.

The speech synthesizer 105 receives the spectral parameters input from the parameter corrector 104,
A synthesized speech signal is generated using the delay parameter, adaptive codebook gain, normalized residual signal, or normalized residual signal gain, and is output to the speech signal output unit 106.

Based on the loss detection result input from the loss detector 101, the audio signal output unit 106 receives a decoded audio signal input from the audio data decoder 102, a synthesized audio signal input from the audio synthesis unit 105, or a decoded audio signal One of the signals obtained by mixing the synthesized speech signal at a certain ratio is output.

  Next, the operation of the audio data decoding apparatus according to the first embodiment will be described with reference to FIG.

First, the loss detector 101 detects whether the received audio data is lost (S
601). Here, as a method of detecting a loss, a bit error in a wireless network is detected by CRC (Cyc
A method that detects voice data loss when detected using a lic redundancy check (RF) code, or loss in an IP (Internet Protocol) network
ol for Real-Time Applications) There is a method of detecting that audio data has been lost when it is detected by missing a sequence number in the header.

If the loss detector 101 does not detect the loss of audio data, the audio data analyzer 102 decodes the audio data received and outputs it to the audio signal output unit (S602).

If the loss detector 101 detects a loss of audio data, the audio data analyzer 1
03 extracts a spectral parameter, delay parameter, adaptive codebook gain, normalized residual signal or normalized residual signal gain based on the decoded speech signal immediately before the loss (S603).
). Here, the analysis of the decoded speech signal may be performed on the decoded speech signal immediately before the loss is detected, or may be performed on all the decoded speech signals. Next, the parameter correction unit 104
Based on the loss detection result, spectral parameters, delay parameters, adaptive codebook gain, normalized residual signal or normalized residual signal gain are used as they are, or ± 1%
Is corrected by adding a random number (S604). The speech synthesizer 105 generates a synthesized speech signal using these values (S605).

Then, the audio signal output unit 106 performs audio data decoder 10 based on the loss detection result.
2, the decoded speech signal input from 2, the synthesized speech signal input from the speech synthesizer 105, or the signal obtained by mixing the decoded speech signal and the synthesized speech signal at a certain ratio is output (S 606).
. Specifically, if no loss is detected in the previous frame and the current frame, a decoded speech signal is output. If a loss is detected, a synthesized speech signal is output, and the next frame in which the loss is detected. First, the audio signal output from the audio signal output unit 106 is discontinuous by adding the audio signals such that the ratio of the synthesized audio signal is large and the ratio of the decoded audio signal increases with time. Avoid becoming.

G.71, in which parameters and the like have not been conventionally extracted by the audio data decoding apparatus of the first embodiment
In one method, parameters are extracted, and these values are used as signals for interpolating the loss of audio data, thereby improving the sound quality of the audio for interpolating the loss.

A second embodiment will be described with reference to FIGS. 3 and 4. The difference between the second embodiment and the first embodiment is that when a loss of audio data is detected, it is detected whether the next audio data after the loss is received before outputting an audio signal for interpolating the loss portion. . When the next audio data is detected, in addition to the operation of the first embodiment, the information of the next audio data is also used to generate an audio signal for the lost audio data.

FIG. 3 shows the configuration of a decoding apparatus for audio data encoded by a waveform encoding method typified by the G.711 method, as in the first embodiment. The speech data decoding apparatus according to the second embodiment includes a loss detector 201, a speech data decoder 202, a speech data analyzer 203, a parameter correction unit 204, a speech synthesis unit 205, and a speech signal output unit 206. Here, since the voice data decoder 202, the parameter correction unit 204, and the voice synthesis unit 205 perform the same operations as the voice data decoder 102, the parameter correction unit 104, and the voice synthesis unit 105 of the first embodiment, description thereof is omitted.

When the loss detector 201 detects a loss of audio data in addition to the operation of the loss detector 101 described in the first embodiment, before the audio data decoder 202 outputs an audio signal for interpolating the loss part, Detect whether audio data is received. Further, the loss detector 201 sends the detection result to the audio data decoder 202 and the audio data analyzer 20.
3, the parameter correction unit 204, and the audio signal output unit 206.

In addition to the operation of the audio data analyzer 103 described in the first embodiment, the audio data analyzer 203 generates a signal obtained by inverting the time of the audio signal for the next audio data in which loss is detected based on the detection result from the loss detector 201. Generate. Then, this signal is analyzed in the same procedure as in the first embodiment, and the extracted spectral parameter, delay parameter, adaptive codebook gain, normalized residual signal or normalized residual signal gain is converted to the parameter correction unit 20.
4 is output.

The audio signal output unit 206 is generated based on the loss detection result input from the loss detector 201 based on the decoded audio signal input from the audio data decoder 202 or the parameters of the audio data before the loss is first detected. One of the signals added to increase the ratio of the signal obtained by inverting the time of the synthesized voice signal generated by the parameter of the next voice data in which the loss is detected. Output.

  Next, the operation of the audio data decoding apparatus according to the second embodiment will be described with reference to FIG.

First, the loss detector 201 detects whether the received audio data is lost (S
701). If the loss detector 201 does not detect a loss of audio data, the first embodiment
The same operation as S602 is performed (S702).

If the loss detector 201 detects a loss of audio data, it is detected whether the loss detector 201 has received the next audio data after loss before the audio data decoder 202 outputs an audio signal for interpolating the loss part. (S703). If the next audio data has not been received, the same operations as S603 to S605 of the first embodiment are performed (S704 to S706).
. If the next audio data is received, the audio data decoder 202 decodes the next audio data (S707). Based on the decoded next voice data, the voice data analyzer 203 extracts a spectrum parameter, a delay parameter, an adaptive codebook gain, a normalized residual signal, or a normalized residual signal gain (S708). Next, the parameter correction unit 204 uses the spectrum parameter, delay parameter, adaptive codebook gain, normalized residual signal or normalized residual signal gain as it is based on the loss detection result, or uses a random number of ± 1%. It is corrected by adding it (S709). The speech synthesizer 205 uses these values to generate a synthesized speech signal (S710).

Based on the loss detection result input from the loss detector 201, the audio signal output unit 206 uses the decoded audio signal input from the audio data decoder 202 or the parameters of the audio data before the loss is initially detected. The ratio of the generated synthesized speech signal is high,
Finally, a signal added so that the ratio of the signal obtained by inverting the time of the synthesized voice signal generated by the parameter of the next voice data in which the loss is detected is increased (S711).

According to the second embodiment, VoIP (Voice over IP), which has been rapidly spreading in recent years, buffers received voice data in order to absorb fluctuations in the arrival time of voice data. When interpolating the audio signal of the selected portion, the sound quality of the interpolated signal can be improved by using the lost audio data present in the buffer.

A third embodiment will be described with reference to FIGS. 5 and 6. In this embodiment, when a loss of audio data is detected with respect to decoding of audio data encoded by the CELP method, the second embodiment
Similarly, if the first audio data decoder 302 receives the audio data after the loss before outputting the audio signal for interpolating the loss portion, the next audio is generated when generating the audio signal for the lost audio data. A configuration using data information is shown.

FIG. 5 shows a configuration of a decoding apparatus for audio data encoded by the CELP method. Example 3
The audio data decoding apparatus includes a loss detector 301, a first audio data decoder 302, a parameter interpolation unit 303, a second audio data decoder 304, and an audio signal output unit 305.

The loss detector 301 outputs the received audio data to the first audio data decoder 302 and the second audio data decoder 304, and detects whether the received audio data is lost. When the loss is detected, it is detected whether the first audio data decoder 302 receives the next audio data before outputting the audio signal for interpolating the loss part, and the detection result is compared with the first audio data decoder 302 and the first audio data decoder 302. Output to the second audio data decoder 304.

If no loss is detected, the first audio data decoder 302 decodes the input audio data and outputs the decoded audio signal to the audio data output unit, and the spectral parameters, delay parameters, and adaptive codebook at the time of decoding The gain, the normalized residual signal, or the normalized residual signal gain is output to the parameter interpolation unit 303. Also, the first audio data decoder 302 detects a loss, and when the next audio data is not received, the first audio data decoder 302 generates an audio signal that stores the loss portion using information of past audio data. For the generation method, the method described in Patent Document 1 can be used. Further, the first audio data decoder 302 generates an audio signal for the lost audio data using the parameters input from the parameter interpolation unit 303 and outputs the audio signal to the audio signal output unit 305.

The second audio data decoder 303 detects the loss, and when the first audio data decoder 302 receives the next audio data before outputting the audio signal for interpolating the loss part, the audio signal for the lost audio data Is generated using information of past audio data. Then, the second audio data decoder 304 uses the generated audio data to decode the next audio data, and uses the spectrum parameter, delay parameter, adaptive codebook gain, normalized residual signal or normalized residual signal. The gain is extracted and output to the parameter interpolation unit 303.

The parameter interpolation unit 304 generates a parameter for the lost audio data using the parameters input from the first audio data decoder 302 and the parameters input from the second audio data decoder 304, and sends them to the first audio data decoder 302. Output.

The audio signal output unit 305 outputs the decoded audio signal input from the audio data decoder 302.

  Next, the operation of the audio data decoding apparatus according to the third embodiment will be described with reference to FIG.

First, it is detected whether the audio data received by the loss detector 301 is lost (S8).
01). If there is no loss, the first audio data decoder 302 decodes the input audio data, and the spectral parameters, delay parameters, adaptive codebook gain, normalized residual signal, or normalized residual signal gain at the time of decoding are decoded. Is output to the parameter interpolation unit 303 (S
802 and S803).

If there is a loss, the loss detector 301 detects whether the next audio data after the loss is received before the first audio data decoder 302 outputs the audio signal for interpolating the loss part (S804). If the next audio data has not been received, the first audio data decoder 30
2 generates an audio signal for storing the loss part using information of past audio data (S80).
5).

If the next audio data is received, the second audio data decoder 303 generates an audio signal for the lost audio data using the information of the past audio data (S806). The second audio data decoder 304 decodes the next audio data using the generated audio data,
A spectral parameter, delay parameter, adaptive codebook gain, normalized residual signal or normalized residual signal gain at the time of decoding is generated and output to the parameter interpolation unit 303 (S80).
7). Next, the parameter interpolation unit 304 uses the parameters input from the first audio data decoder 302 and the parameters input from the second audio data decoder 304 to generate parameters for the lost audio data (S808). Then, the first audio data decoder 302 generates an audio signal for the lost audio data using the parameters generated by the parameter interpolation unit 304, and outputs the audio signal to the first audio data decoder 302 (S809).
.

The first audio data decoder 302 outputs the audio signal generated in each case to the audio signal output unit 305, and the audio signal output unit 305 outputs the decoded audio signal (S810).

According to the third embodiment, VoIP, which has been rapidly spreading in recent years, buffers received voice data in order to absorb fluctuations in the arrival time of voice data. When the signal is interpolated, the sound quality of the interpolated signal can be improved by using the lost audio data present in the buffer.

A fourth embodiment will be described with reference to FIGS. In the CELP method, if an interpolation signal is used when audio data loss occurs, the lost part can be compensated, but the interpolation signal is not generated from correct audio data. The sound quality will be degraded. Accordingly, in the fourth embodiment, in addition to the third embodiment, after outputting the interpolated audio signal for the lost portion of the audio data, when the lost portion of the audio data arrives late, by using this audio data, A technique for improving the quality of the audio signal of the next lost audio data is disclosed.

FIG. 7 shows the configuration of a decoding apparatus for audio data encoded by the CELP method, as in the third embodiment. The audio data decoding apparatus according to the fourth embodiment includes a loss detector 401, a first audio data decoder 402, a second audio data decoder 403, a memory storage unit 404, and an audio signal output unit 405.

The loss detector 401 outputs the received audio data to the first audio data decoder 402 and the second audio data decoder 403. The loss detector 401 detects whether the received audio data has been lost. When a loss is detected, it is detected whether the next audio data is received, and the detection results are detected as the first audio data decoder 402 and the second audio data decoder 403.
Or it outputs to the audio | voice signal output part 405. Further, the loss detector 401 detects whether or not the lost voice data is received with a delay.

The first audio data decoder 402 decodes the input audio data when no loss is detected. In addition, when a loss is detected, the first audio data decoder 402 generates an audio signal using past audio data information and outputs the audio signal to the audio data output unit 405. For the generation method, the method described in Patent Document 1 can be used. Further, the first audio data decoder 402 outputs a memory such as a synthesis filter to the memory storage unit 404.

The second audio data decoder 403 uses a memory such as a synthesis filter for packets immediately before loss detection stored in the memory storage unit 404 when the audio data of the loss portion arrives late. The decoded signal is output to the audio signal output unit 405.

The audio signal output unit 405, based on the loss detection result input from the loss detector 401, the decoded audio signal input from the first audio data decoder 402, the decoded audio signal input from the second audio data decoder 403, or the aforementioned An audio signal obtained by adding two signals at a certain ratio is output.

  Next, the operation of the speech data decoding apparatus according to the fourth embodiment will be described with reference to FIG.

First, the operation of the audio data decoding apparatus of the third embodiment, S801 to S810, is performed, and an audio signal for storing lost audio data is output. Here, in S805 and S806, when an audio signal is generated from past audio data, a memory such as a synthesis filter is output to the memory storage unit 404 (S903 and S904). Then, the loss detector 401 detects that the lost voice data has been received with a delay (S905). Loss detector 40
If 1 is not detected, the audio signal generated in the third embodiment is output. Ross Detector 4
If 01 is detected, the second audio data decoder 403 decodes the audio data that arrived late using a memory such as a synthesis filter for the packet immediately before loss detection stored in the memory storage unit 404 (S906). .

Then, the voice signal output unit 405, based on the loss detection result input from the loss detector 401, the decoded audio signal input from the first audio data decoder 402 and the decoded audio signal input from the second audio data decoder 403. Alternatively, an audio signal obtained by adding the two signals at a certain ratio is output (S907). Specifically, when the loss is detected and the audio data arrives late, the audio signal output unit 405 initially receives the first audio data decoder 402 as an audio signal for the audio data next to the lost audio data. Increase the ratio of the input decoded audio signal. Then, as time elapses, the audio signal output unit 405 outputs an audio signal added so as to increase the ratio of the decoded audio signal input from the second audio data decoder 403.

According to the fourth embodiment, a correct decoded speech signal can be generated by rewriting a memory such as a synthesis filter by using the speech data of the loss part that arrives late. In addition, it is possible to prevent the voice from being discontinuous by outputting the audio signal obtained by adding the correct decoded audio signal at a certain ratio without outputting it immediately. Furthermore, even if an interpolated signal is used for the lost part, the sound quality after the interpolated signal is improved by rewriting a memory such as a synthesis filter with the lost part of the audio data and generating a decoded audio signal. be able to.

Here, the fourth embodiment has been described in the form of being added to the audio conversion signal of the third embodiment, but may be added to a form in which another interpolation signal is generated.

  An audio data conversion apparatus according to the fifth embodiment will be described with reference to FIGS.

FIG. 9 shows the configuration of an apparatus for converting a speech signal encoded by a certain speech encoding method into another speech encoding method. Here, for example, a configuration of an apparatus that converts audio data encoded by a waveform encoding method typified by G.711 into audio data encoded by a CELP method is shown. The audio data conversion apparatus according to the fifth embodiment includes a loss detector 501 and an audio data decoder 50.
2. Audio data encoder 503, parameter correction unit 504, and audio data output unit 505
It is composed of

The loss detector 501 outputs the received audio data to the audio data decoder 502. Further, the loss detector 501 detects whether or not the received audio data is lost, and the detection results are displayed as an audio data decoder 502, an audio data encoder 503, and a parameter correction unit 50.
4 and the audio signal output unit 505.

If no loss is detected, the audio data decoder 502 decodes the input audio data and outputs a decoded audio signal to the audio data encoder 503.

If no loss is detected, the audio data encoder 503 determines that the audio data decoder 5
The decoded audio signal input from 02 is encoded, and the encoded audio data is output to the audio data output unit 505. Also, the audio data encoder 503 outputs a spectral parameter, a delay parameter, an adaptive codebook gain, a residual signal, or a residual signal gain, which are parameters at the time of encoding, to the parameter correction unit 504. Furthermore, the audio data encoder 503
When a loss is detected, the parameter is input from the parameter correction unit 504 and receives a parameter. The audio data encoder 503 holds a filter (not shown) used for parameter extraction, encodes the parameter received from the parameter correction unit 504, and generates audio data. At that time, the audio data encoder 503 updates a memory such as a filter. Here, the audio data encoder 503 is caused by a quantization error that occurs during encoding.
When the parameter value after encoding does not become the same value as the value input from the parameter correction unit 504, the parameter value after encoding is selected to be the closest value to the value input from the parameter correction unit 504. Further, in order to avoid the occurrence of a discrepancy with the filter memory held by the wireless communication apparatus of the communication partner, the audio data encoder 503 has a memory (see FIG. (Not shown) needs to be updated. Further, the audio data encoder 503 outputs the generated audio data to the audio data output unit 505.

The parameter correction unit 504 receives and stores a spectral parameter, a delay parameter, an adaptive codebook gain, a residual signal or a residual signal gain, which are parameters at the time of encoding, from the audio data encoder 503. Further, the parameter correction unit 504 uses the held parameter before the loss detection as it is, or makes a predetermined correction, and the loss detector 50
1 is output to the audio data encoder 503 based on the loss detection result input from 1.

The audio data output unit 505 outputs the audio signal received from the audio data encoder 503 based on the loss detection result received from the loss detector 501.

  Next, an audio data conversion apparatus according to the fifth embodiment will be described with reference to FIG.

First, the loss detector 501 detects whether the received audio data is lost (S
1001). If the loss detector 501 detects no loss, the audio data decoder 5
A decoded audio signal is generated based on the audio data received by 02 (S1002). Then, the audio data encoder 503 encodes the decoded audio signal and outputs a spectrum parameter, a delay parameter, an adaptive codebook gain, a residual signal, or a residual signal gain, which are parameters at the time of encoding (S1003).

If the loss detector 501 detects a loss, the parameter correction unit 504 maintains the pre-loss parameter held by the loss detector 501 as it is or performs a predetermined correction, and the audio data encoder 5
Output to 03. The audio data encoder 503 that has received the parameter updates the memory of the filter for extracting the parameter (S1004). Further, the audio data encoder 503 generates an audio signal based on the parameter immediately before the loss (S1005).
).

Then, the audio data output unit 505 outputs the audio signal received from the audio data encoder 503 based on the loss detection result (S1006).

According to the fifth embodiment, in an apparatus for converting data such as a gateway, for example, an interpolation signal for a loss of audio data is not generated by a waveform coding method, and an interpolation signal is interpolated using a parameter or the like. The sound quality can be improved. In addition, the amount of calculation can be reduced by interpolating the loss portion using a parameter or the like without generating an interpolation signal for the loss of audio data by the waveform encoding method.

Here, in the fifth embodiment, audio data encoded by a waveform encoding method represented by G.711 is converted to C.
In the above description, the voice data encoded by the ELP method is converted into voice data. However, the voice data encoded by the CELP method may be converted into voice data encoded by another CELP method.

Those skilled in the art can easily implement various modifications of the above-described embodiments. Therefore, the present invention is not limited to the above-described embodiments, but is interpreted in the widest range considered by the claims and their equivalents.

It is the schematic which shows the structure of the audio | voice data decoding apparatus of Example 1 of this invention. It is the schematic which shows operation | movement of the audio | voice data decoding apparatus of Example 1 of this invention. It is the schematic which shows the structure of the audio | voice data decoding apparatus of Example 2 of this invention. It is the schematic which shows operation | movement of the audio | voice data decoding apparatus of Example 2 of this invention. It is the schematic which shows the structure of the audio | voice data decoding apparatus of Example 3 of this invention. It is the schematic which shows operation | movement of the audio | voice data decoding apparatus of Example 3 of this invention. It is the schematic which shows the structure of the audio | voice data decoding apparatus of Example 4 of this invention. It is the schematic which shows operation | movement of the audio | voice data decoder of Example 4 of this invention. It is the schematic which shows the structure of the audio | voice data converter of Example 5 of this invention. It is the schematic which shows operation | movement of the audio | voice data converter of Example 5 of this invention.

Explanation of symbols

DESCRIPTION OF SYMBOLS 101 Loss detector 102 Voice data decoder 103 Voice data analyzer 104 Parameter correction part 105 Voice synthesis part 106 Voice signal output part 201 Loss detector 202 Voice data decoder 203 Voice data analyzer 204 Parameter correction part 205 Voice synthesis part 206 Voice signal output part 301 Loss Detector 302 First audio data decoder 303 Second audio data decoder 304 Parameter interpolation unit 305 Audio signal output unit 401 Loss detector 402 First audio data decoder 403 Second audio data decoder 404 Memory storage unit 405 Audio signal output unit 501 Loss detector 502 Audio data decoder 503 Audio data encoder 504 Parameter correction unit 505 Audio data output unit

Claims (2)

An audio data decoding device that outputs an interpolation signal for interpolating a loss in audio data,
A loss detector that detects the loss and detects that the audio data corresponding to the loss has been received late;
When the loss is detected, a first audio data decoder that outputs a memory such as a synthesis filter to the memory storage unit;
Using a memory such as the synthesis filter of the pre-loss audio data stored in the memory storage unit, the audio data corresponding to the loss is decoded, and then the audio data next to the audio data corresponding to the loss is obtained. A second audio data decoder that generates a decoded audio signal ;
An audio data decoding apparatus comprising: an audio signal output unit that outputs the decoded audio signal while changing a ratio of the decoded audio signal to all output audio signals.   The first audio data decoder, when the loss is not detected, decodes the input audio data and generates the decoded audio signal;
The audio signal according to claim 1, wherein the audio signal output unit outputs the audio signal while changing a ratio of the decoded audio signal generated by the first audio data decoder and the decoded audio signal generated by the second audio data decoder. Data decoding device.

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