JP3566931B2 - Method and apparatus for assembling packet of audio signal code string and packet disassembly method and apparatus, program for executing these methods, and recording medium for recording program - Google Patents

Description

[0001]
TECHNICAL FIELD OF THE INVENTION
The present invention relates to a method and an apparatus for assembling a packet of a code string of an audio signal, and a method and an apparatus for disassembling the packet, which are used in a technique for suppressing and compensating for the quality deterioration of an audio signal lost due to packet loss that may occur when transmitting an audio signal in packets. The present invention relates to a program that executes these methods and a storage medium that stores the program.
[0002]
[Prior art]
As represented by mobile communication and VoIP (Voice over IP), voice and data can be integratedly handled by packet communication. In many cases, a plurality of audio frames are packed and transmitted in one packet in order to reduce the overhead of the header of the normal packet.
In voice communication, in general, it is essential that the order in which a signal sequence forming an audio signal is generated is maintained even when the signal is reproduced after transmission. On the other hand, in packet communication, transmission of a packet transmitted at regular time intervals is required. Since the delay varies, the arrival time may fluctuate. A fluctuation absorbing buffer is used to reproduce the audio signal by the code stored in the packet in the order in which the fluctuations are emitted and transmitted.
[0003]
One of the problems in packet voice communication is packet loss. By increasing the bandwidth and speed of the communication channel, the degradation and delay due to coding are eliminated. On the other hand, packet loss is a problem that cannot be avoided even if the communication capacity increases.
The causes of packet loss include the following. First, when the number of packets is large, packets may be completely lost due to collision between the packets. If the code bit error reaches about 50% in the course of transmission, it is assumed that all the packet information has been lost, and it may be determined that the packet has been lost. Furthermore, if the arrival delay of a packet is larger than compensated by the fluctuation absorbing buffer, it may be determined that the packet has been lost and that the packet has been lost. These causes the voice quality to deteriorate.
[0004]
In order not to make the hearing uncomfortable due to the deterioration of quality, the lost packet part needs to be compensated by some other signal. Depending on the encoding method, encoding is performed using information before and after the buffer. Therefore, once a packet is lost, quality may deteriorate for a while after restoration. Suppressing the deterioration of the quality in terms of audibility is also included in the packet loss compensation.
A conventional technique for suppressing and compensating for the quality deterioration of a voice signal lost due to packet loss will be described with reference to FIG.
[0005]
The missing speech interpolation device includes a demultiplexing circuit 102, a residual signal power decoder 119, an inverse quantizer 120, a long-term prediction coefficient decoding / selection unit 118, a short-term prediction coefficient decoding unit 107, and a long-time synthesis / interpolation filter 117. The short cut filter 121 and the input / output terminals 101, 115, and 116 are provided, and when the loss of the voice coded information is detected, that is, when the loss detection signal is input, the connection cutoff switch 121 is opened. Along with the long-term prediction coefficient decoder / selector 118, a long-term prediction coefficient for interpolation (1. A numerical value is set in advance and is always a constant value, or 2. the length of the previous frame obtained from the long-term prediction coefficient decoder A long-term prediction coefficient for interpolation according to the temporal prediction coefficient) is output to the long-term predictor 112. In the short-term predictor 114, the short-term prediction coefficient in the previous frame is set as it is. The output signal is input to the short-time synthesis filter 110 by self-driving without any input to the long-time synthesis / interpolation filter 117. The short-time synthesis filter 110 performs normal decoding processing to interpolate the reproduced digital audio signal. (See JP-A-5-88697)
This conventional technique is a method in which a decoder analyzes a pitch period from a past signal, extracts an appropriate waveform, and repeats the waveform to generate a pseudo signal. The most likely cause of the deterioration in the pitch cycle repetition compensation is the discontinuity of the waveform. The discontinuity of the waveform is likely to occur in a portion where the compensation signal during packet loss and the signal after the recovery from the packet loss are joined. In order to make this discontinuity inconspicuous, the pitch cycle is adjusted so as to be continuous from disappearance to that after restoration, or the synthesized signal and the signal after restoration are gradually changed by OLA (Overlap add). And a method of gradually attenuating the power of the combined signal has been proposed.
[0006]
In packet elimination compensation of CELP (Code Excited Linear Prediction) used for low bit rate audio coding, an audio signal in a packet is classified into periodicity and aperiodicity in advance, and erasure is performed. If the pitch frequency of the packet is periodic, a method of using an excitation signal of an adaptive codebook is used, and if the packet is non-periodic, a method of randomly using white noise is often used.
Still other methods include, as characteristic processes, a method of repeating a synthesis filter coefficient, attenuating an adaptive / fixed codebook gain, and attenuating a gain prediction.
[0007]
These methods were effective methods for suppressing an unpleasant signal for hearing. However, in many cases, it is difficult to always reproduce a sound close to the original sound because the reproduction is a pseudo synthetic signal. Between packets, the pitch or power may fluctuate rapidly, or the sound quality may be significantly degraded due to waveform discontinuity or unreasonable adjustment due to pitch interval mismatch. Further, in the case of the compression codec, there is a problem that a rising portion is deteriorated after recovery from disappearance.
[0008]
[Problems to be solved by the invention]
An object of the present invention is to solve the drawbacks of the conventional packet loss compensation technology and improve voice quality deterioration due to packet loss, assuming that the capacity of the communication path is sufficiently large and some auxiliary information can be added. .
In the related art, when the pitch period, the power, and the like change in the section where the packet has been lost, the deterioration becomes remarkable. The present invention provides an audio signal while a packet is lost, an encoding method and a decoding method capable of suppressing deterioration of an audio signal after recovery from loss, even if the audio data length included in the packet is large, and It is an object to provide means for realizing these methods.
[0009]
[Means for Solving the Problems]
In order to solve the above problem, the present invention provides a method of assembling a voice packet, comprising the steps of: coding a voice signal of a current frame by a first coding method of high quality to generate a first code string; A decoded signal is generated by decoding the code sequence, and the audio signals of N frames from one to N (N: an arbitrary integer equal to or more than 1) from the current frame are encoded by the second encoding method. An internal state necessary for encoding is calculated from the decoded signal, and using the calculated internal state, audio signals of N frames from one to N ahead of the current frame are highly compressed. Encoding is performed by a second encoding method to generate a second code string, the first code string and the second code string are combined and stored in a packet at the current frame time, and in the decomposition, to determine the presence or absence of packet loss in each frame, current frame When it is determined that the packet at the time has not been lost, the first decoding corresponding to the first encoding method from the first code sequence of the current frame among the code sequences stored in the packet at the current frame time When the audio signal is decoded by the encoding method, and it is determined that the packet at the current frame time has been lost, the first code sequence of the past frame among the code sequences stored in the packet at the past frame time is used. The internal state for decoding the audio signals of N frames from one to N ahead of the decoded audio signal by the second decoding method corresponding to the highly-compressed second encoding method is as follows. Then, the speech signal is decoded by the second decoding method corresponding to the second coding method of the current frame using the calculated internal state.

[0010]
BEST MODE FOR CARRYING OUT THE INVENTION
As shown in FIG. 1, in a voice over packet (VoP), a voice packet is received by the network module 1, decomposed for each packet and output to the fluctuation absorbing buffer 3, and a packet loss is determined and a packet loss flag is determined. Output to the compensator 2.
The decomposed packets are accumulated in the fluctuation absorbing buffer 3, and after waiting for a while, the packets are reproduced.
If the packet loss is not determined, the main bit stream of the current frame of the packet stored in the fluctuation absorbing buffer 3 is output to the audio decoder 4. When packet loss is determined, efficient packet loss compensation can be performed using the main and sub bit streams of the packets before and after reaching the fluctuation absorbing buffer 3. For example, when the packet (3) to be reproduced does not arrive, the audio decoder 4 creates an analysis coefficient and outputs the analysis coefficient to the packet erasure compensator 2. The packet erasure compensator 2 reaches the analysis coefficient and the fluctuation absorbing buffer 3. The erasure compensation data is output to the audio decoder 4 using the sub-bit stream (3) of the past packet (2), and efficient packet erasure compensation can be performed. By performing parameter interpolation and volume control, the audio decoder 4 can perform processing to minimize deterioration and output audio.
[0011]
In the present invention, one main codec that is normally used is provided with resistance to packet loss by combining a plurality of sub-codecs as a compensating means when a packet is lost.
The main codec uses a high-quality coding method with a relatively low compression rate, and the sub-codec needs to select a coding method with higher compression and sufficiently higher quality than the main codec. By doing so, an increase in the data amount can be suppressed.
(Encoder)
The encoder of the present invention will be described with reference to FIGS.
[0012]
The input audio data is divided into frames (data units), and a signal to be normally reproduced on the decoder side is encoded by a main codec (encoder) 11 by a first encoding method to create a main bit stream. For example, the G.10 is a pulse code modulation (PCM) system that transmits an audio band signal of 8 kHz sampling at 64 kb / s in the audio encoding system recommended by the ITU-T. Encode according to the 711 encoding standard.
The sub-codec (encoder) 14 encodes the audio data one packet ahead by the second encoding method to create a sub-bit stream. For example, G.264, which is an LD-CELP method for 16 kb / s telephone band voice coding recommended by the ITU-T. Encode according to the 728 coding standard. When two or more sub-bitstreams are included, it is preferable to include a bitstream obtained by encoding a signal that is prefetched further than that.
[0013]
Then, the main bit stream and the sub-bit stream one packet ahead are combined by the combining unit 15, and the packetizing unit 16 performs a process of adding a chronological sequence number, a bit error detection code, and the like by a known serial number assigning circuit or the like. Output as packets.
In the above-mentioned manner, each packet has a main bit stream in which the signal stored in the main buffer is encoded by the main codec, and a sub-bit in which the signal ahead of the main buffer stored in the sub buffer is encoded by the sub codec. Have a stream.
[0014]
As shown in FIG. 4, when the decoding side determines that a packet has been lost (packet {circle around (3)}) by creating a bit stream in this way, it is included in the packet {right arrow over (2)} immediately before the loss. It is possible to generate a sound (decoded sound signal [3]) in a section lost according to the information of the sub-bit stream [3]. This is because the sub-bit stream includes a signal obtained by encoding a pre-read signal by a sub-codec. Also, in order to reduce the overhead of the header of a packet, a plurality of voice frames are often packed in one packet and transmitted. In the conventional packet erasure compensation, a pseudo speech signal is generated by repeating a past signal using a synthesized speech, so that the longer the speech data included in the packet, the more the degradation becomes remarkable.
[0015]
According to the present invention, since the sub codec includes audio data one packet ahead of the main codec, it is possible to perform packet loss compensation with little deterioration regardless of the length of audio data included in each packet.
Care must be taken depending on the compression coding used for the sub-codec.
This is because a codec that decodes the analysis coefficients required for encoding and decoding (depending on the codec, for example, synthetic signals, filter coefficients, prediction coefficients, etc.) from the previous packet may cause packet loss. Then, the analysis coefficients of the predictor, the quantizer, and the like in the normal encoder and the decoder are different. Even in such a case, in order to match the analysis coefficients, it is necessary for the encoder to transmit initial information of the analysis coefficients as encoded information.
[0016]
In the present invention, a set of a main bit stream encoded by a main codec which is high quality encoding and a sub bit stream encoded by a sub codec which is high compression encoding exists in a packet. Therefore, if the analysis coefficient of the sub codec is created by the main codec (decoder) 12 and the sub codec (analysis coefficient calculation) 13 from the signal obtained by decoding the main codec, the information need not be transmitted.
For example, G. When the analysis coefficients of the encoder are generated from past synthesized signals as in 728, it is possible to replace the synthesized signal portion with a high-quality signal decoded by the main codec. Similarly, in the decoder, it is necessary to replace the synthesized signal portion with high quality coding. The decoding can be performed correctly by matching the internal states of the encoder and the decoder. Further, the quality of the decoded signal can be improved by replacing the analysis coefficient with a high-quality signal instead of the synthesized signal.
[0017]
The analysis coefficient is, for example, G. 728: Refers to a synthesis filter coefficient obtained by a backward synthesis filter adaptor of an LD-CELP encoder (not shown) and an auditory weighting filter coefficient obtained by an auditory weighting filter.
The synthetic inverted filter _{e n = x n + a 1} x n over _{1 +} ··· + a n _{x 0}
a: Filter coefficient x: Synthetic signal e: Residual signal Similarly, it is possible to replace the auditory weighting filter with a high quality signal.
[0018]
Perceptual weighting filter _{w n = a 0 x n +} a 1 x n over _{1 + ··· + a n x o} - (b 0 w n + b 1 w n over _{1 + ··· + b n w o} )
a, b: filter coefficient x: synthetic signal w: auditory weighting signal In this way, high-quality decoding can be performed by replacing the signal with a high-quality signal.
The calculated analysis coefficients are transferred to the sub codec (decoder) 14, that is, 728: Shape code vector (waveform) stored in a codebook (codebook) as an output from an optimum codebook data selector of an LD-CELP encoder (not shown) and an optimum gain level The corresponding code is selected and a sub-bitstream is output.
(decoder)
The decoder will be described with reference to FIGS.
[0019]
On the decoder side, first, the received signal is depacketized by the depacketizing section 21, and the main codec / sub-codec distribution section outputs a main bit stream (code according to the G.711 voice coding standard of the current frame) among the voice packets at the current frame time. Stream) and a sub-bitstream (code stream according to the G.728 speech coding standard of the previous frame).
The main bit stream is decoded by the main codec (decoder) 23 into an audio signal by the first decoding method.
[0020]
Then, the decoded signal is calculated by the sub codec (analysis coefficient calculation) 24 to generate an internal state of the sub codec (decoder) 25. Alternatively, as described above, the internal state of the sub codec is created directly from the main codec.
Finally, with the internal state being inherited, the sub bit stream is decoded by the sub codec (decoder) 25 according to the second decoding method, and an audio signal is output.
[0021]
Specifically, the LD.-CELP decoder (not shown) supplies the G. A code for a shape code vector (waveform) and a code for a gain level are input as a coded sequence according to the 728 speech coding standard, and a shape code vector and a gain vector are selected from a codebook (excitation VW codebook). The analysis coefficient calculated as the filter coefficient is transferred to use the decoded sound.
Detection of the presence / absence of a voice packet is performed at a stage prior to the depacketizing unit 21 shown in FIG. 5, and is performed by detecting a disorder of a sequence number or a bit error by a widely used packet loss detection circuit.
[0022]
If it is not determined that there is a packet loss signal, the changeover switch 27 is switched to the main codec (decoder) 23 to output an audio signal. When it is determined that a packet loss signal is present, the switch 27 is switched to the sub codec (decoder) 25 side.
If the main codec uses a codec that requires past information, such as ADPCM (Adaptive Pulse Code Modulation), that is, a codec that takes over the internal state, if a past packet is lost, loss compensation is performed. Degradation occurs in the connection between the decoded sub-codec used in the above and the audio signal decoded from the main codec. In such a case, by creating necessary information from the signal reproduced by the sub codec, it is possible to suppress the deterioration of the reproduction of the main codec after the compensation.
[0023]
When there is only one sub-codec and packets are continuously lost, if there are not as many sub-codecs as the number of lost packets, loss compensation by the sub-codec cannot be performed, and it is considered that voice deteriorates. In such a case, the compensation shown in FIG. 6 is performed by using the waveform repetition compensator 26 according to the conventional method shown in FIG. Only when the sub-codec cannot be used, a waveform is created using a synthesized signal such as the past pitch frequency repetition erasure compensation.
As a countermeasure for the case of burst erasure (two or more packet erasures), an example is shown in FIG. 5 in which compensation is performed by a conventional method. It is sufficient to store a code string based on a signal, and decode a speech signal using a code string in a preceding frame from a code string extending over two or more frames in packet decomposition.
[0024]
When there is no great difference in the degree of distortion of the quantization noise between the main codec and the sub codec, a signal-to-quantization noise ratio (SNR) can be increased by synchronizing and adding each other. . That is why quantization noise is often uncorrelated for different codecs, and the sum of the powers of the correlated and uncorrelated noise is considered to be greater in the acoustic part when added together. .
As the number of sub-codecs increases, the more pre-read information is provided for the main codec. As a result, it is possible to provide resistance even when packets are continuously lost.
[0025]
However, as shown in FIG. 7, as the number of sub-codecs is increased, pre-read information is required, and as a result, the delay is increased. Also. The amount of information increases by the number of sub-codecs.
The cause of the delay in VoIP is, in addition to the above, a large delay caused by a fluctuation absorbing buffer that absorbs fluctuations such as packet arrival delay. In addition, a large delay occurs on a PC (Personal Computer) due to a buffer of a network card, a sound card, or the like, but the problem is solved by introducing dedicated hardware and improving the performance of the PC. In a real-time conversation, it is desirable that the total delay time in one direction is within 200 milliseconds. Therefore, it is necessary to adjust the total number of sub-codecs, fluctuation absorption, and other delays to meet the standard.
[0026]
In mobile communication and VoIP, the communication speed is not always constant, and it is considered that the amount of information that can be used for voice communication changes depending on the amount of information used for an application. The present invention is characterized in that a combination suitable for a network is made possible by flexibly changing the quality of a sub-codec and the number of sub-codecs according to a communication speed and a calculation speed of a computer.
FIG. 8 shows a schematic diagram of a waveform when this method is used.
Referring to this figure, it can be seen that the present method is closer to the original sound than the conventional method.
[0027]
Also, the packet assembling apparatus and the packet disassembling apparatus of the present invention have a computer having a CPU, a memory, and the like, a user terminal used by a user who is an access subject, and a machine readable medium such as a CD-ROM, a magnetic disk device, and a semiconductor memory. It can be composed of various recording media.
A control program for causing the computer to execute the above-described operation is stored in a recording medium, and the control program is read by the computer, and the operation of the computer is controlled to realize each element in the above-described embodiment on the computer. it can.
[0028]
【The invention's effect】
According to the present invention, as compared with the conventional method, deterioration in quality due to packet loss is suppressed as much as possible, discontinuous portions of the waveform are eliminated, and a lost portion faithful to the original sound can be compensated. Further, since the code sequence of the current frame and the code sequence of the previous frame are combined as the packet at the current frame time, even if the previous packet is lost, it can be easily compensated.
[Brief description of the drawings]
FIG. 1 is a diagram showing a basic configuration of a VoP.
FIG. 2 is an explanatory diagram of processing of an encoder.
FIG. 3 is a schematic configuration diagram of an encoder.
FIG. 4 is an explanatory diagram of processing when a packet is lost.
FIG. 5 is a schematic configuration diagram of a decoder.
FIG. 6 is an explanatory diagram of processing when a burst is lost.
FIG. 7 is a diagram showing the structure of one packet when a plurality of sub-codecs are provided.
FIG. 8 is a comparison diagram of the waveforms of the conventional technique and the technique of the present invention for the original sound.
FIG. 9 is a configuration diagram of a conventional missing voice interpolation device.
[Explanation of symbols]
DESCRIPTION OF SYMBOLS 1 Network module 2 Packet loss compensation part 3 Fluctuation absorption buffer 4 Audio decoder 11 Main codec (encoder)
12, 23 Main codec (decoder)
13, 24 Sub codec (calculation of analysis coefficient)
14 Sub Codec (Encoder)
15 Coupling unit 16 Packetization unit 21 Depacketization unit 22 Main codec / sub-codec distribution unit 25 Sub-codec (decoder)
26 Waveform repetition compensator 27 Selector switch

Claims (9)

A packet assembling method for encoding an audio signal for each frame to generate a code sequence, storing the generated code sequence, and generating a packet,
Encoding the audio signal of the current frame with a first encoding method of high quality to generate a first code sequence;
The first code string is decoded to generate a decoded signal, and the audio signals of N frames from one to N (N: any integer equal to or greater than 1) from the current frame are converted to a second code. Calculating the internal state required for encoding by the decoding method from the decoded signal,
Using the calculated internal state, audio signals of N frames from one to N ahead of the current frame are encoded by a highly-compressed second encoding method to generate a second code sequence. The process of
Combining the first code string and the second code string and storing them in a packet at the current frame time;
A packet assembling method comprising: A packet assembling apparatus that encodes an audio signal for each frame to generate a code sequence, stores the generated code sequence, and generates a packet,
A first encoder that encodes the audio signal of the current frame with a high-quality first encoding method to generate a first code string;
The first code string is decoded to generate a decoded signal, and the audio signals of N frames from one to N (N: any integer equal to or greater than 1) from the current frame are converted to a second code. Means for calculating, from the decoded signal, an internal state necessary for encoding by the encoding method, and using the calculated internal state, audio of N frames from one to N ahead of the current frame. Means for encoding the signal with a highly compressed second encoding method to generate a second code sequence ; and
A combining unit that combines the first code string and the second code string and stores the first code string and the packet at the current frame time;
A packet assembling apparatus comprising: A first code string generated by the packet assembling method according to claim 1 and coded by the first high-quality coding method of the current frame and N (N: 1 or more arbitrary random numbers from the next one) A step of inputting a packet at the current frame time obtained by combining the second code string encoded by the second encoding method with high compression of N frames up to (integer) ahead;
Determining the presence or absence of packet loss for each frame;
If it is determined that the packet at the current frame time has not been lost, the first code sequence of the current frame among the code sequences stored in the packet at the current frame time is converted to the first code sequence corresponding to the first encoding method. A first step of decoding the audio signal by the decoding method of
If it is determined that the packet at the current frame time has been lost, one of the code strings stored in the packet at the previous frame time is ahead of the audio signal decoded from the first code string of the past frame . To calculate the internal state for decoding the audio signals of N frames from N to N ahead by the second decoding method corresponding to the high-compression second encoding method, and calculating the calculated internal state. Using the second decoding method corresponding to the second coding method of the current frame to decode the audio signal using the second coding method ;
A packet disassembly method comprising: The packet disassembly method according to claim 3,
A packet disassembly method comprising, if it is determined that the packet at the current frame time has not been lost, adding a current frame signal and a current frame signal included in a past packet to generate an audio signal; Method. A first code string encoded by the high-quality first encoding method of the current frame generated by the packet assembling apparatus according to claim 2 and N (N: an arbitrary integer equal to or more than 1) from the first code string. Means for inputting a packet at the current frame time obtained by combining a second code string coded by the second compression method with high compression of N frames up to the next frame;
Packet loss determining means for determining the presence or absence of packet loss for each frame;
If it is determined that the packet at the current frame time has not been lost, the first code sequence of the current frame among the code sequences stored in the packet at the current frame time is converted to the first code sequence corresponding to the first encoding method. When the audio signal is decoded by the decoding method of (1) and it is determined that the packet at the current frame time has been lost, the first code of the past frame in the code string stored in the packet at the past frame time An internal part for decoding the audio signals of N frames from one to N ahead of the audio signal decoded from the column by the second decoding method corresponding to the high-compression second encoding method Decoding means for calculating a state and decoding the audio signal by a second decoding method corresponding to the second encoding method of the current frame using the calculated internal state ,
A packet decomposer comprising: The packet decomposer according to claim 5,
The decryption means
A packet decomposer for generating an audio signal by adding a current frame signal and a current frame signal included in a past packet when it is determined that a packet at a current frame time has not been lost. Encoding the audio signal of the current frame with a first encoding method of high quality to generate a first code string;
The first code string is decoded to generate a decoded signal, and the audio signals of N frames from one to N (N: any integer equal to or greater than 1) from the current frame are converted to a second code. Calculating from the decoded signal the internal state required for encoding by the encoding method,
Using the calculated internal state , audio signals of N frames from one to N ahead of the current frame are encoded by a highly-compressed second encoding method to generate a second code sequence. Steps to
Combining the first code string and the second code string and storing them in a packet at the current frame time;
Computer-readable recording medium on which a program for causing a computer to execute the program is recorded. Encoding the audio signal of the current frame by the first encoding method of high quality to generate a first code string; decoding the first code string to generate a decoded signal; Calculating, from the decoded signal, an internal state necessary for encoding the audio signals of N frames from the destination to N (N: an arbitrary integer equal to or greater than N) frames by the second encoding method; Using the calculated internal state, the audio signals of N frames from one to N ahead of the current frame are encoded by a highly-compressed second encoding method to form a second code sequence. Generating the current frame time packet from the generating step, combining the first code string and the second code string into a packet at the current frame time, and inputting the packet generated at the current frame time.
A procedure for determining the presence or absence of packet loss for each frame;
If it is determined that the packet at the current frame time has not been lost, the first code sequence of the current frame among the code sequences stored in the packet at the current frame time is converted to the first code sequence corresponding to the first encoding method. A first procedure of decoding an audio signal by the decoding method of
If it is determined that the packet at the current frame time has been lost, one of the code strings stored in the packet at the previous frame time is ahead of the audio signal decoded from the first code string of the past frame . To calculate the internal state for decoding the audio signals of N frames from N to N ahead by the second decoding method corresponding to the high-compression second encoding method, and calculating the calculated internal state. A second procedure for decoding a speech signal using a second decoding method corresponding to a second encoding method for the current frame using the second coding method ;
Computer-readable recording medium on which a program for causing a computer to execute the program is recorded. A computer-readable recording medium recording the program according to claim 8,
When it is determined that the packet at the current frame time has not been lost, a program for causing a computer to execute a procedure of generating an audio signal by adding the current frame signal included in the past frame and the current frame is recorded. Computer readable recording medium.

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