JP5111430B2 - Speech coding apparatus, speech decoding apparatus, and methods thereof - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To suppress quality degradation of a decoded voice signal caused by disappearance of a frame from being propagated in a communication system using a CELP type voice encoder. <P>SOLUTION: This voice encoder or the like is provided with an adaptive code book 208 for storing a first voice source signal; a pitch peak position detecting part 151 for generating a second voice source signal corresponding to the first voice source signal, by applying the first voice source signal to preset processing, for detecting a sample satisfying a preset condition as a specified pulse, out of the samples forming the second voice source signal, and for obtaining a position of the specified pulse; and a pitch peak information embedding part for encoding redundant information including information showing the last position of the specified pulse in the frame. <P>COPYRIGHT: (C)2009,JPO&amp;INPIT

Description

The present invention relates to a speech coding apparatus, a speech decoding apparatus, and a method thereof that encode speech signals to generate speech coding information, packetize and transmit the information .

  In packet communication typified by Internet communication, erasure compensation (concealment) processing is generally performed when encoded information cannot be received at the decoder side due to loss of packets (or frames) in the transmission path. (For example, see Patent Document 1 and Patent Document 2).

  A conventional audio signal transmission system is shown in FIG. As shown in FIG. 25, the conventional audio signal transmission system includes an audio signal transmission device 1 and an audio signal reception device 10. The audio signal transmitter 1 includes an input device 2, an A / D (analog / digital) converter 3, an audio encoder 4, a signal processor 5, an RF modulator 6, a transmitter 7, and an antenna 8. . The input device 2 receives the audio signal, converts it into an analog audio signal, which is an electrical signal, and provides it to the A / D conversion device 3. The A / D conversion device 3 converts the analog speech signal from the input device 2 into a digital speech signal and gives it to the speech coding device 4. The speech encoding device 4 encodes the digital speech signal from the A / D conversion device 3 to generate speech encoded information, and provides it to the signal processing device 5. The signal processing device 5 performs channel coding processing, multiplexing processing, packetization processing, transmission buffering processing, and the like on the speech coding information from the speech coding device 4, and then converts the speech coding information to RF (Radio). Frequency) to the modulation device 6. The RF modulation device 6 modulates the voice encoded signal from the signal processing device 5 and gives the modulated signal to the transmission device 7. The transmission device 7 transmits the voice encoded signal from the RF modulation device 6 as a radio wave (RF signal) via the antenna 8.

  The audio signal receiver 10 includes an antenna 9, a receiver 11, an RF demodulator 12, a signal processor 13, an audio decoder 14, a D / A (digital / analog) converter 15, and an output device 16. .

  The receiving device 11 receives a radio wave (RF signal) that is a voice encoded signal via the antenna 9 to generate a received voice signal that is an analog electric signal, and gives this to the RF demodulator 12. The radio wave (RF signal) received by the antenna 9 is exactly the same as the radio wave (RF signal) transmitted from the audio signal transmitting apparatus 1 if there is no signal attenuation or noise superposition in the transmission path.

  The RF demodulator 12 demodulates the received audio signal from the receiver 11 and provides it to the signal processor 13. The signal processing device 13 performs jitter absorption buffering processing, packet assembling processing, demultiplexing processing, channel decoding processing, and the like of the received voice signal from the RF demodulator 12, and then decodes the received voice signal. To device 14.

  If the packet does not arrive within a predetermined time, the signal processing device 13 notifies the speech decoding device 14 that packet loss has occurred. The audio decoding device 14 decodes the received audio signal from the signal processing device 13 to generate a decoded audio signal, and supplies this to the D / A conversion device 15.

  When receiving the packet loss information from the signal processing device 13, the speech decoding device 14 cannot receive the received speech signal of the corresponding packet, and thus performs a frame loss compensation process to generate a speech signal. The D / A conversion device 15 converts the digital decoded speech signal from the speech decoding device 14 into an analog decoded speech signal and gives it to the output device 16. The output device 16 converts the analog decoded audio signal from the D / A conversion device 15 into air vibration and outputs it as a sound wave so that it can be heard by the human ear.

  The speech decoding apparatus 14 includes a speech decoding unit 56 and a frame erasure compensation unit 57. The speech decoding unit 56 has three input terminals, two of which are connected to two output terminals of the signal processing device, respectively. The remaining one input terminal is connected to the output terminal of the frame loss compensation unit 57. The audio decoding unit 56 has two output terminals, which are connected to the D / A converter 15 and the frame loss compensation unit 57, respectively. The input terminal and output terminal of the frame erasure compensation unit 57 are connected to the output terminal and input terminal of the speech decoding unit 56, respectively. The frame erasure compensation unit 57 receives the parameter information decoded in the past by the speech decoding unit 56, generates speech parameters required when the frame of the received speech signal is lost, and creates the speech decoding unit 56. Output to.

  The voice decoding unit 56 performs a normal decoding process on the received voice signal from the other output terminal of the signal processing device 13 when the frame loss signal is not received from the one output terminal of the signal processing device 13 and performs decoding. Generate an audio signal. Also, when receiving a frame loss signal, the speech decoding unit 56 performs a decoding process using the speech parameter input from the frame loss compensation unit 57. There are various types of frame erasure compensation processing depending on the audio coding method. 729 and the like are defined as part of the decoding algorithm.

Japanese Patent Laid-Open No. 09-120297 Japanese Patent Laid-Open No. 09-120297

However, in the conventional audio signal transmission system, when a transmission frame (or packet) is lost on the transmission path, sound Koefuku Goka device 14 by using the encoding information received in the past frames (or packets) Perform erasure compensation processing. Influence for the synchronization of the internal states will not be taken, the packet lost decoding of frame erasure portion later becoming not frame erasure only frames between the time the speech encoder 4 and the sound Koefuku Goka 14 There is a problem that the quality of the decoded speech signal may be greatly deteriorated by propagation.

  For example, ITU-T Recommendation G. When the CELP (Code Excited Linear Prediction) scheme shown in 729 is used, the encoder and the decoder are subjected to frame erasure compensation processing by performing speech encoding and decoding processing using a past decoded driving excitation signal. If different driving sound source signals are combined with each other, the internal states of the encoder and decoder do not match for a while, and the quality of the decoded speech signal may be greatly degraded. Even in the internal state, the quality degradation due to the mismatch of the contents of the adaptive codebook which is a buffer of the excitation signal generated in the past is remarkable.

The present invention has been made in view of such a point, and provides a speech coding apparatus, a speech decoding apparatus, and these methods capable of improving the quality of a frame erasure unit and a decoded speech signal immediately after the frame erasure. The purpose is to provide.

A speech coding apparatus according to the present invention is a CELP (Code Excited Linear Prediction) type speech coding apparatus, and adds an adaptive codebook storing a first excitation signal and a plurality of samples of the first excitation signal. and processing means for generating a second sound source signal and, during the last one pitch period of the second sound source signal, detects a sample absolute value of the amplitude is maximum as a particular pulse, the position of the specific pulse The specific pulse position detecting means for obtaining the information and the encoding means for generating the specific pulse position information by encoding the position of the specific pulse are employed.

The speech decoding apparatus according to the present invention is a CELP (Code Excited Linear Prediction) speech decoding apparatus, which decodes specific pulse position information encoded and transmitted by a communication partner to generate a first specific pulse. Decoding means for obtaining a position; adaptive codebook for storing a first excitation signal ; processing means for adding a plurality of samples of the first excitation signal to generate a second excitation signal; between the end 1 pitch period of the sound source signal, detects a sample absolute value of the amplitude is maximum as a particular pulse, and the particular pulse position detecting means for determining the position of said specific pulses as a second specific pulse position, lost Frame erasure compensation means for performing processing for adjusting the second specific pulse position to the first specific pulse position on the second sound source signal. The take.

The speech encoding method of the present invention is a CELP (Code Excited Linear Prediction) speech encoding method, which adds a plurality of samples of the first excitation signal stored in the adaptive codebook and adds the second excitation signal. a processing step of generating, the between the trailing one pitch period of the second sound source signal, detects a sample absolute value of the amplitude is maximum as a particular pulse, particular pulse position detection step for determining the position of said specific pulses And an encoding step of generating specific pulse position information by encoding the position of the specific pulse.

The speech decoding method of the present invention is a CELP (Code Excited Linear Prediction) speech decoding method, which decodes specific pulse position information encoded and transmitted by a communication partner to obtain a first specific pulse position. A decoding step of obtaining, a processing step of adding a plurality of samples of the first excitation signal stored in the adaptive codebook to generate a second excitation signal, and a last one pitch period of the second excitation signal A specific pulse position detecting step in which a sample having the maximum absolute value of the amplitude is detected as a specific pulse and the position of the specific pulse is determined as a second specific pulse position, and in the lost frame, the second specific A frame erasure compensation step for performing processing for adjusting the pulse position to the first specific pulse position on the second sound source signal.

According to the present invention, without increasing the number or in bits increase of no small number of bits, it is possible to improve decoded speech quality degradation immediately erased frame and a frame erasure.

1 is a block diagram showing a speech signal transmitting apparatus including a speech coding apparatus and a speech signal receiving apparatus including a speech decoding apparatus according to Embodiment 1 of the present invention. The block diagram which shows the structure of the audio | voice coding apparatus which concerns on Embodiment 1 of this invention. Signal diagram for explaining the operating principle of the pitch peak position detector FIG. 2 is a schematic diagram showing a packet transmitted from the speech encoding apparatus according to Embodiment 1 of the present invention. The block diagram which shows the structure of the speech decoding apparatus which concerns on Embodiment 1 of this invention. Signal waveform diagram for explaining the operation of the speech decoding apparatus according to Embodiment 1 of the present invention. Signal waveform diagram for explaining the operation of the speech decoding apparatus according to Embodiment 1 of the present invention. Signal waveform diagram for explaining the operation of the speech decoding apparatus according to Embodiment 1 of the present invention. The flowchart which shows the audio | voice coding processing procedure which concerns on Embodiment 1 of this invention. The flowchart which shows the speech decoding processing procedure which concerns on Embodiment 1 of this invention. FIG. 3 is a block diagram showing a speech signal transmitting apparatus including a speech coding apparatus and a speech signal receiving apparatus including a speech decoding apparatus according to Embodiment 2 of the present invention. FIG. 3 is a block diagram showing the configuration of a speech encoding apparatus according to Embodiment 2 of the present invention. Outline diagram showing packet transmitted from speech coding apparatus according to embodiment 2 of the present invention. The block diagram which shows the structure of the speech decoding apparatus which concerns on Embodiment 2 of this invention. A block diagram showing a speech signal transmitting apparatus including a speech encoding apparatus and a speech decoding apparatus including a speech encoding apparatus according to Embodiment 3 of the present invention. The block diagram which shows the structure of the speech decoding apparatus which concerns on Embodiment 3 of this invention. Outline diagram for explaining the operation of the speech decoding apparatus according to Embodiment 3 of the present invention. FIG. 7 is a block diagram showing a speech signal transmitting apparatus including a speech encoding apparatus and a speech decoding apparatus including a speech encoding apparatus according to Embodiment 4 of the present invention. Block diagram showing the configuration of a speech encoding apparatus according to Embodiment 4 of the present invention. Signal waveform diagram for explaining the method of detecting the pitch peak position of the speech coding apparatus according to Embodiment 4 of the present invention. The flowchart which shows the pitch peak position detection processing procedure of the speech coder according to Embodiment 4 of the present invention. Block diagram showing the configuration of a speech decoding apparatus according to Embodiment 4 of the present invention. The flowchart which shows the speech-encoding processing procedure which concerns on Embodiment 4 of this invention. The flowchart which shows the speech decoding processing procedure which concerns on Embodiment 4 of this invention. A block diagram showing a configuration of a speech signal transmitting apparatus including a conventional speech encoding apparatus and a speech signal receiving apparatus including a speech decoding apparatus

According to a first aspect of the present invention, an audio encoding device includes an audio signal encoding unit that encodes an input audio signal, an audio signal transmission unit that transmits the encoded audio signal in a predetermined data unit, Pitch peak position detecting means for detecting the pitch peak position of the transmitted audio signal; and pitch peak position information transmitting means for transmitting information representing the detected pitch peak position together with the encoded audio signal. The structure to comprise is taken.

According to this configuration, since the pitch peak position of the encoded excitation signal is transmitted as information in addition to the conventional excitation encoding information, the receiver side transmits the information by examining the pitch peak position of the decoded excitation signal. It is possible to confirm whether or not there is a discrepancy between the sound source signal on the side and the sound source signal on the reception side.

According to a second aspect of the present invention, in the speech encoding apparatus according to the above configuration, the speech signal encoding means performs a linear prediction analysis of the input speech signal to calculate a linear prediction coefficient, and An encoding unit that performs quantization and encoding of the linear prediction coefficient, a linear prediction filter that is configured by the quantized linear prediction coefficient, and an excitation code that encodes and generates an excitation signal that drives the linear prediction filter And a CELP-type speech encoding means including a conversion unit.

According to this configuration, since the pitch peak position of the encoded excitation signal is transmitted as information in addition to the conventional excitation encoding information, transmission is performed by examining the pitch peak position of the excitation signal decoded on the receiver side. It is possible to confirm whether or not there is a discrepancy between the sound source signal on the side and the sound source signal on the reception side.

According to a third aspect of the present invention, in the speech coding apparatus according to the above configuration, the pitch peak position information sending means multiplexes and sends the detected pitch peak position information and the coded speech signal. The structure to do is taken.

According to this configuration, the encoded audio signal and pitch peak position information can be transmitted as a set. Therefore, the decoder receiving the multiplexed information can obtain the encoded speech signal and pitch peak position information by separating them.

According to a fourth aspect of the present invention, in the speech coding apparatus according to the above-described configuration, the pitch peak position information sending unit uses a part of the information coded by the coding unit as the detected pitch peak position. A configuration is adopted in which information is transmitted and transmitted.

According to this configuration, in order to send out the pitch peak position information of the encoded sound source signal, by checking the pitch peak position of the sound source signal decoded on the receiver side, It is possible to check whether there is a mismatch between the two. In addition, the pitch peak position information is transmitted in place of the low error sensitivity portion of the LPC encoded information, so that it is possible to suppress the quality deterioration of the decoded speech without increasing the bit rate. It is.

According to a fifth aspect of the present invention, in the speech coding apparatus according to the above configuration, the pitch peak position detection unit detects a pitch peak position located at the rearmost position in the frame of the speech signal as the pitch peak position information. Take the configuration.

According to this configuration, the excitation signal (contents of the adaptive codebook) decoded by comparing the last pitch peak position in the frame with the received pitch peak position in the excitation signal decoded on the decoder side It is possible to confirm whether it is correct. If the excitation signal (contents of the adaptive codebook) is not correct, the excitation signal (adaptive codebook) is obtained by matching the pitch peak position of the decoded excitation signal (adaptive codebook) to the received pitch peak position. Can be modified.

The sixth aspect of the present invention employs a configuration in which the speech coding apparatus multiplexes and transmits the pitch peak position information detected in the previous frame together with the coding information in the current frame in the above configuration.

According to this configuration, when the previous frame is an erasure frame, the decoder can confirm the validity of the excitation signal (adaptive codebook) generated by the frame erasure compensation of the previous frame on the decoder side. To do.

The seventh aspect of the present invention employs a configuration in which the speech encoding apparatus multiplexes and transmits the pitch peak position information detected in the next frame together with the encoding information in the current frame in the above configuration.

According to this configuration, when the current frame is an erasure frame, it is possible to perform more accurate frame erasure compensation processing using the pitch peak position information received by the decoder in the previous frame.

According to an eighth aspect of the present invention, an audio decoding device includes: an audio signal decoding unit that decodes an encoded audio signal; and information on a pitch peak position of the audio signal, Compensation means for performing compensation processing for quality degradation caused by the lost frame when the lost frame exists is adopted.

According to this configuration, compensation for the lost frame itself and compensation for the frame after the lost frame can be performed based on the pitch peak position information as compensation for the quality degradation caused by the lost frame. In particular, in an audio signal, by adjusting the pitch peak position, it is possible to perform compensation processing for a lost frame and compensation that avoids a shift in the pitch of the frame following the lost frame, and quality degradation can be reduced. .

According to a ninth aspect of the present invention, in the speech decoding apparatus according to the above configuration, the speech signal decoding unit includes a linear prediction coefficient decoding unit that performs a decoding process of a linear prediction coefficient, and a decoded linear prediction coefficient. A linear prediction filter that controls the gains of the adaptive codebook and the fixed codebook based on the decoded gain parameter, and the linear prediction filter based on the contents of the controlled adaptive codebook and fixed codebook. The structure is a CELP speech decoding unit including a driving sound source decoding unit.

According to this configuration, since it is possible to generate an adaptive codebook so that the correct pitch peak position is obtained, the mismatch of the contents of the adaptive codebook between the encoder side and the decoder side in the normal frame after frame erasure Can be reduced.

According to a tenth aspect of the present invention, in the audio decoding device according to the above configuration, the pitch peak position information is multiplexed with the encoded audio signal, and the multiplexed pitch peak position information is A configuration including separation means for separating the encoded audio signal is adopted.

According to an eleventh aspect of the present invention, in the speech decoding apparatus according to the above configuration, a part of the encoded speech signal is replaced with the pitch peak position information, and the replaced pitch peak position information Is separated from the encoded speech signal.

According to these configurations, since the speech decoding apparatus can be applied to a system that transmits pitch peak position information instead of a low error sensitivity portion of LPC encoded information, without increasing the bit rate, In addition, it is possible to suppress quality degradation of decoded speech.

According to a twelfth aspect of the present invention, in the speech decoding apparatus according to the above configuration, the speech signal decoding unit includes a linear prediction coefficient decoding unit that performs a decoding process of a linear prediction coefficient, and a decoded linear prediction coefficient. A linear prediction filter that controls the gains of the adaptive codebook and the fixed codebook based on the decoded gain parameter, and the linear prediction filter based on the contents of the controlled adaptive codebook and fixed codebook. A CELP speech decoding unit including a driving sound source decoding unit, and the compensation unit includes a frame immediately before the current frame among the speech signals decoded by the speech signal decoding unit. In such a case, the content of the adaptive codebook is modified based on the pitch peak position information.

According to this configuration, since the position of the pitch peak in the immediately preceding frame can be accurately reflected in the adaptive codebook, the contents of the adaptive codebook on the encoder side and the decoder side in the normal frame after the frame disappearance do not match. Can be reduced.

According to a thirteenth aspect of the present invention, in the speech decoding apparatus according to the above configuration, the speech signal decoding means performs a decoding process on the encoding parameter received in the immediately preceding frame in the current frame, and the compensation means Adopts a configuration in which, when the immediately preceding frame is a lost frame, compensation processing for the lost frame is performed using pitch peak position information received in the current frame.

According to this configuration, since the frame erasure compensation is performed using the pitch peak position information of the erasure frame, the decoding information of the subsequent normal frame, and the decoding information of the immediately preceding normal frame, it is possible to perform a more accurate frame erasure compensation process. It becomes possible.

According to a fourteenth aspect of the present invention, in the speech decoding apparatus according to the above configuration, when the current frame is an erasure frame among the speech signals decoded by the speech signal decoding unit, A configuration is adopted in which the lost frame compensation processing is performed using the pitch peak position information received in the immediately preceding frame.

According to this configuration, when the frame loss occurs, the pitch peak position information received in the immediately preceding normal frame can be used, so that more accurate frame loss compensation processing can be performed and the sound source is set so that the correct pitch peak position is obtained. Since it is possible to generate a signal (adaptive codebook), it is possible to reduce inconsistencies in the contents of the adaptive codebook between the encoder side and the decoder side in a normal frame after frame loss. In particular, in the speech encoding apparatus, when the pitch peak position information detected in the next frame is transmitted together with the encoded information (encoded speech signal) in the current frame, the pitch peak position information of the lost frame is It can be received in the immediately preceding normal frame, and the lost frame compensation process can be performed based on the pitch peak position information of the lost frame itself.

According to a fifteenth aspect of the present invention, a mobile station apparatus encodes an audio signal encoding unit that encodes an input audio signal, audio signal transmission means that transmits the encoded audio signal in a predetermined data unit, Pitch peak position detecting means for detecting the pitch peak position of the transmitted audio signal; and pitch peak position information transmitting means for transmitting information representing the detected pitch peak position together with the encoded audio signal. And the structure which performs radio | wireless communication between base station apparatuses is taken.

According to this configuration, since the pitch peak position of the encoded excitation signal is transmitted as information in addition to the conventional excitation encoding information, the base station side transmits the information by examining the pitch peak position of the decoded excitation signal. It is possible to confirm whether or not there is a discrepancy between the sound source signal on the side and the sound source signal on the reception side.

According to a sixteenth aspect of the present invention, the mobile station apparatus decodes the received audio signal based on audio signal decoding means for decoding the received encoded audio signal, and information indicating the pitch peak position of the audio signal. And a compensation means for performing compensation processing for quality degradation caused by the lost frame when a lost frame exists in the network, and adopts a configuration for performing wireless communication with the base station apparatus.

According to this configuration, compensation for the lost frame itself and compensation for the frame after the lost frame can be performed based on the pitch peak position information as compensation for the quality degradation caused by the lost frame. In particular, in an audio signal, by adjusting the pitch peak position, it is possible to perform compensation processing for a lost frame and compensation that avoids a shift in the pitch of the frame following the lost frame, and quality degradation can be reduced. .

According to a seventeenth aspect of the present invention, a base station apparatus encodes an audio signal encoding unit that encodes an input audio signal, audio signal transmitting means that transmits the encoded audio signal in a predetermined data unit, Pitch peak position detecting means for detecting the pitch peak position of the transmitted audio signal; and pitch peak position information transmitting means for transmitting information representing the detected pitch peak position together with the encoded audio signal. And the structure which performs radio | wireless communication between mobile station apparatuses is taken.

According to this configuration, since the pitch peak position of the encoded excitation signal is transmitted as information in addition to the conventional excitation encoding information, the mobile station transmits the information by examining the pitch peak position of the decoded excitation signal. It is possible to confirm whether or not there is a discrepancy between the sound source signal on the side and the sound source signal on the reception side.

According to an eighteenth aspect of the present invention, the base station apparatus decodes the received audio signal based on the audio signal decoding means for decoding the received encoded audio signal, and information indicating the pitch peak position of the audio signal. And a compensation means for performing compensation processing for quality degradation caused by the lost frame when a lost frame exists in the mobile station apparatus, and adopts a configuration for performing wireless communication with the mobile station apparatus.

According to this configuration, compensation for the lost frame itself and compensation for the frame after the lost frame can be performed based on the pitch peak position information as compensation for the quality degradation caused by the lost frame. In particular, in an audio signal, by adjusting the pitch peak position, it is possible to perform compensation processing for a lost frame and compensation that avoids a shift in the pitch of the frame following the lost frame, and quality degradation can be reduced. .

According to a nineteenth aspect of the present invention, an audio signal transmission method includes: an audio signal encoding step for encoding an input audio signal; an audio signal transmission step for transmitting the encoded audio signal in a predetermined data unit; A pitch peak position detecting step of detecting a pitch peak position of the transmitted audio signal; a pitch peak position information transmitting step of transmitting information representing the detected pitch peak position together with the encoded audio signal; Deterioration of quality caused by a lost frame when a lost frame exists in the audio signal based on an audio signal decoding step for decoding the encoded audio signal and information indicating a pitch peak position of the audio signal And a compensation process for performing the compensation process.

According to this method, since the pitch peak position of the encoded excitation signal is transmitted as information in addition to the conventional excitation encoding information, the receiver side transmits the information by examining the pitch peak position of the decoded excitation signal. It is possible to confirm whether or not there is a discrepancy between the sound source signal on the side and the sound source signal on the reception side. Further, as compensation for quality degradation caused by a lost frame, compensation for the lost frame itself and compensation for a frame after the lost frame can be performed based on the pitch peak position information. In particular, in an audio signal, by adjusting the pitch peak position, it is possible to perform compensation processing for a lost frame and compensation that avoids a shift in the pitch of the frame following the lost frame, and quality degradation can be reduced. .

According to a twentieth aspect of the present invention, there is provided an audio signal encoding step of encoding an input audio signal, an audio signal transmission step of transmitting the encoded audio signal in a predetermined data unit, and the transmission A voice peak encoding step for detecting a pitch peak position of a voice signal to be detected and a pitch peak position information sending step for sending information representing the detected pitch peak position together with the encoded voice signal. Let the device run.

According to this program, since the pitch peak position of the encoded excitation signal is sent as information in addition to the conventional excitation encoding information, the receiver side transmits the information by examining the pitch peak position of the decoded excitation signal. It is possible to confirm whether or not there is a discrepancy between the sound source signal on the side and the sound source signal on the reception side.

In a twenty-first aspect of the present invention, an audio frame includes an erasure frame in the audio signal based on an audio signal decoding step in which the program decodes the encoded audio signal and information indicating a pitch peak position of the audio signal. A speech decoding apparatus is caused to execute a compensation process for compensating for quality degradation caused by the lost frame if present.

According to this program, compensation for the lost frame itself and compensation for the frame after the lost frame can be performed based on the pitch peak position information as compensation for quality degradation caused by the lost frame. In particular, in an audio signal, by adjusting the pitch peak position, it is possible to perform compensation processing for a lost frame and compensation that avoids a shift in the pitch of the frame following the lost frame, and quality degradation can be reduced. .

  Hereinafter, embodiments of the present invention will be described in detail with reference to the drawings.

(Embodiment 1)
FIG. 1 is a block diagram showing a configuration of an audio signal transmission system according to Embodiment 1 of the present invention.

  The audio signal transmission system includes an audio signal transmitting apparatus 100 and an audio signal receiving apparatus 199.

  The audio signal transmission device 100 includes an input device 102, an A / D conversion device 103, an audio encoding device 104, a signal processing device 105, an RF modulation device 106, a transmission device 107, and an antenna 108. The A / D conversion device 103 is connected to the input device 102.

  The input terminal of the speech encoding device 104 is connected to the output terminal of the A / D conversion device 103. The input terminal of the signal processing device 105 is connected to the output terminal of the speech encoding device 104. The input terminal of the RF modulation device 106 is connected to the output terminal of the signal processing device 105. The input terminal of the transmitter 107 is connected to the output terminal of the RF modulator 106. The antenna 108 is connected to the output terminal of the transmission device 107.

  The input device 102 receives the audio signal, converts it into an analog audio signal, which is an electrical signal, and gives it to the A / D conversion device 103. The A / D conversion device 103 converts an analog voice signal from the input device 102 into a digital voice signal, and gives this to the voice coding device 104. The audio encoding device 104 encodes the digital audio signal from the A / D conversion device 103 to generate audio encoding information, and provides the generated signal to the signal processing device 105.

  The signal processing device 105 performs channel coding processing, packetization processing, transmission buffer processing, and the like on the speech coding information from the speech coding device 104, and then provides the speech coding information to the RF modulation device 106. The RF modulation device 106 modulates the voice encoded signal from the signal processing device 105 and provides the modulated signal to the transmission device 107. The transmission device 107 transmits the audio encoded signal from the RF modulation device 106 as a radio wave (RF signal) via the antenna 108.

  In the audio signal transmitting apparatus 100, the digital audio signal obtained through the A / D conversion apparatus 103 is processed in units of several tens of milliseconds, and one frame or several frames of encoded data is processed into one packet. And send this packet to the packet network. In this embodiment, it is assumed that one frame is transmitted in one packet in order to minimize transmission delay. Therefore, packet loss corresponds to frame loss.

  The present invention can be applied not only to a packet switching network but also to a circuit switching network. In this case, packetization processing, jitter absorption buffering processing, and packet assembly processing are unnecessary.

  The audio signal receiving apparatus 199 includes an antenna 110, a receiving apparatus 111, an RF demodulating apparatus 112, a signal processing apparatus 113, an audio decoding apparatus 114, a D / A conversion apparatus 115, and an output apparatus 116. An input terminal of the receiving device 111 is connected to the antenna 110. The input terminal of the RF demodulator 112 is connected to the output terminal of the receiver 111. The input terminal of the signal processing device 113 is connected to the output terminal of the RF demodulation device 112. The two input terminals of the speech decoding apparatus 114 are connected to the two output terminals of the signal processing apparatus 113 on a one-to-one basis. The input terminal of the D / A conversion device 115 is connected to the output terminal of the speech decoding device 114. The input terminal of the output device 116 is connected to the output terminal of the D / A converter 115.

  The receiving device 111 receives a radio wave (RF signal) that is voice encoded information via the antenna 110, generates a received voice signal that is an analog electric signal, and supplies this to the RF demodulator 112. The radio wave (RF signal) received via the antenna 110 is exactly the same as the radio wave (RF signal) transmitted by the audio signal transmitting apparatus 100 if there is no signal attenuation or noise superposition in the transmission path.

  The RF demodulator 112 demodulates the received audio signal from the receiver 111 and supplies it to the signal processor 113. The signal processing device 113 performs jitter absorption buffering processing, packet assembling processing, packet loss detection processing, demultiplexing processing, channel decoding processing, and the like of the received voice signal from the RF demodulation device 112, and performs reception voice signal and packet processing. The erasure information is provided to the speech decoding device 114, respectively.

  The audio decoding device 114 decodes the received audio signal from the signal processing device 113 to generate a decoded audio signal, and provides the decoded audio signal to the D / A conversion device 115. The D / A conversion device 115 converts the digital decoded speech signal from the speech decoding device 114 into an analog decoded speech signal and provides it to the output device 116. The output device 116 converts the analog decoded audio signal from the D / A converter 115 into air vibrations and outputs the sound as a sound wave to the human ear.

  Next, the speech encoding apparatus 104 will be described in detail with reference to FIGS. FIG. 2 is a block diagram showing the configuration of the speech encoding apparatus 104. FIG. 3 is a schematic diagram for explaining the operation of the pitch peak position detector.

  As shown in FIG. 1, the speech encoding apparatus 104 includes a pitch peak position detection unit 151, a multiplexing unit 152, a speech encoding unit 153, and a 1-frame delay unit 154. The input terminal of the speech encoding unit 153 is connected to the output terminal of the A / D conversion device 103. The input terminal of the pitch peak position detector 151 is connected to one of the two output terminals of the speech encoder 153. The input terminal of the 1-frame delay unit 154 is connected to the other of the two output terminals of the speech encoding unit 153. The multiplexing unit 152 is connected between the output terminals of the pitch peak position detection unit 151 and the one-frame delay unit 154 and the input terminal of the signal processing device 105.

  The audio encoding unit 153 performs encoding processing on the digital audio signal input from the A / D conversion apparatus 103 and outputs encoding parameter information to the 1-frame delay unit 154. At the same time, speech encoding section 153 outputs a pitch parameter (quantization pitch period), which will be described later, and a sound source signal held in the adaptive codebook to pitch peak position detection section 151. The pitch peak position detection unit 151 detects the pitch peak position in the sound source signal of the past one pitch period length using the quantized pitch period information that is the pitch parameter and the past sound source signal sequence, and outputs the detected position to the multiplexing unit 152. To do.

  It is assumed that the pitch peak position is closest to the end of the frame, which exists in the range of one pitch period in the past from the end of the current frame. Therefore, when one frame is divided into a plurality of subframes, pitch peak position detection is performed in the last subframe. Even if the pitch period is long and there is no pitch peak in the current frame, the pitch peak position in the previous frame section is detected by searching for the pitch peak position within the range of one pitch period from the end of the current frame. I will do it.

  The multiplexing unit 152 multiplexes the pitch peak position information in the current frame detected by the pitch peak position detection unit 151 and the speech coding information in the previous frame output from the 1-frame delay unit 154, and the signal processing device 105. Output to.

  Next, the speech encoding unit 153 will be described in more detail with reference to FIG. As shown in FIG. 2, the speech encoding unit 153 includes a preprocessing unit 201, a linear prediction analyzer 202 for obtaining a linear prediction coefficient, an LPC quantizer 203 that performs quantization and encoding of the linear prediction coefficient, an auditory weight Filter 204, auditory weight filter 205, LPC synthesis filter 206 as a linear prediction filter composed of quantized linear prediction coefficients, adder 207, adaptive codebook 208, multiplier 209, fixed codebook 210, multiplier 211 , Gain quantizer 212, adder 213, excitation parameter determination unit 214, and encoding unit 215. The adaptive codebook 208, fixed codebook 210, and gain quantizer 212 constitute a sound source encoding unit, and the LPC synthesis filter 206 is driven by this sound source encoding unit.

  The preprocessing unit 201 inputs a digital audio signal from the A / D conversion device 103 (FIG. 1), and cuts processing for improving the quality of the audio and DC components such as background noise suppression processing and pre-emphasis processing. High-pass filter processing is performed for output to the linear prediction analyzer 202 and the auditory weight filter 204. The linear prediction analyzer 202 performs linear prediction analysis of the preprocessed digital speech signal input from the preprocessing unit 201 to calculate a linear prediction coefficient, and an LPC quantizer 203, auditory weight filter 204, and auditory weight filter 205. And output respectively.

  The LPC quantizer 203 quantizes and encodes the linear prediction coefficient input from the linear prediction analyzer 202, outputs the quantized linear prediction coefficient to the LPC synthesis filter 206, and sets the encoding result as a parameter L. Output. The parameter L is input to the encoding unit 215 and encoded together with other encoded excitation parameters. The auditory weight filter 204 and the auditory weight filter 205 are ARMA type digital filters using the linear prediction coefficient calculated by the linear prediction analyzer 202, and human auditory characteristics with respect to a quantization error by a speech encoding unit to be described later. The two auditory weighting filters 204 and 205 have the same filter characteristics.

  The auditory weight filter 204 receives the pre-processed digital audio signal from the pre-processing unit 201, performs an ARMA filter process for performing auditory weighting, and outputs the result to the adder 213. The auditory weight filter 205 receives the digital audio signal synthesized by the LPC synthesis filter 206, performs an ARMA filter process for performing the same auditory weighting, and outputs the result to the adder 213.

  The LPC synthesis filter 206 is an AR type digital filter configured using the linear prediction coefficient quantized by the LPC quantizer 203, and generates a synthesized speech signal using the sound source signal output from the adder 207. And output to the auditory weight filter 205. The adder 207 performs vector addition of the adaptive codebook vector input from the adaptive codebook 208 via the multiplier 209 and the fixed codebook vector input from the fixed codebook 210 via the multiplier 211 to generate a sound source A vector is generated and output to the LPC synthesis filter 206. The generated excitation vector is fed back to adaptive codebook 208, and the contents of adaptive codebook 208 are updated. The updated excitation signal buffer of the adaptive codebook is output to pitch peak position detector 151.

  The adaptive codebook 208 is a memory that stores and holds the excitation vectors generated in the past by the adder 207, and is sequentially updated by the excitation vector output from the adder 207. Further, adaptive codebook 208 cuts out a vector from an appropriate position and outputs the vector to multiplier 209. In the case of a voiced signal, since the sound source signal has periodicity, it is possible to efficiently encode a sound source signal using a sound source signal generated in the past. Therefore, such an adaptive codebook is generally used. The extraction position of the adaptive codebook vector is determined by the pitch parameter P. The pitch parameter P is determined by the sound source parameter determination unit.

  Fixed codebook 210 generates an arbitrary vector based on a noise sequence or a combination of a small number of pulses, and can store or generate a predetermined number of vectors. Each vector has a unique number. A fixed codebook vector having a corresponding shape is generated by designating the number. The number is determined by the sound source parameter determination unit 214 as a fixed codebook index C.

  Although not shown in FIG. 2, the fixed codebook is generally composed of a plurality of channels and a plurality of subsets, or pitch periodicization processing is performed on the fixed codebook vector.

  Multiplier 209 multiplies the adaptive codebook gain (pitch gain) quantized by gain quantizer 212 by the vector output from adaptive codebook 208 and outputs the result to adder 207. Multiplier 211 multiplies the vector output from fixed codebook 210 by the fixed codebook gain quantized by gain quantizer 212 and outputs the result to adder 207.

  Gain quantizer 212 outputs quantized adaptive codebook gain and quantized fixed codebook gain indicated by excitation gain parameter G to multipliers 209 and 211, respectively. The sound source gain parameter G is determined by the sound source parameter determination unit 214. The sound source parameter determination unit 214 outputs an error between the input audio signal that is auditory weighted by the auditory weighting filter 204 and the synthesized voice signal of the LPC synthesis filter 206 that is auditory weighted by the auditory weighting filter 205, which are output from the adder 213. The adaptive codebook parameter P, the fixed codebook parameter C, and the gain parameter G are determined so as to minimize.

  The adder 213 calculates a difference vector between the output vector from the auditory weight filter 205 and the output vector from the auditory weight filter 204 and outputs the difference vector to the sound source parameter determination unit 214. The adaptive codebook parameter P, the fixed codebook parameter C, the gain codebook parameter G determined by the excitation parameter determination unit, and the linear prediction parameter L encoded by the LPC quantizer 203 are collectively processed by the encoding unit 215. Are combined into one code and output to the one-frame delay unit 154. The 1-frame delay unit 154 holds the speech coding information input from the coding unit 215 for a time of 1 frame, and then outputs the speech coding information to the multiplexing unit 152.

  Next, the operation of the pitch peak position detector 151 will be described in more detail with reference to FIG.

  As shown in FIG. 3, pitch peak position detection section 151 inputs pitch parameter P and the latest adaptive codebook content (sound signal sequence generated in the past) at least one pitch period length or longer, and is the most temporal. The pitch peak position behind is detected. The simplest method for detecting the pitch peak position is the sample (FIG. 3) whose absolute value is the maximum during the period from the end of the adaptive codebook (latest sample: 1101 in FIG. 3) to one pitch period P in the past. Is a method of detecting 1102) as the pitch peak position. If the current input audio signal has no pitch periodicity (if it is a voiceless part or a noise part), the pitch peak position is not set excessively and a code indicating that there is no pitch periodicity is used. It is assigned separately and the information is output as pitch peak position information.

  Next, the speech decoding apparatus 114 will be described in detail with reference to FIGS. 1, 4, and 5 to 7. FIG. 4 is a schematic diagram of a transmission packet, FIG. 5 is a block diagram showing the configuration of the speech decoding apparatus 114, and FIGS. 6 to 8 are conceptual diagrams for explaining frame erasure compensation processing.

  As shown in FIG. 1, the speech decoding apparatus 114 includes a demultiplexing unit 155, a speech decoding unit 156, a frame delay unit 157, and a frame erasure compensation unit 158.

  The input terminal of the demultiplexing unit 155 is connected to one of the two output terminals of the signal processing device 113. The speech decoding unit 156 has three input terminals, the first input terminal is one output terminal of the demultiplexing unit 155, the second input terminal is one output terminal of the signal processing device 113, and the third input terminal The input terminals are connected to the frame loss compensation unit 158, respectively. Speech decoding section 156 has two output terminals, one connected to one of the two input terminals of frame erasure compensation section 158 and the other connected to D / A converter 115. The input terminal of the 1-frame delay unit 157 is connected to one of the output terminals of the demultiplexing unit 155. Frame erasure compensation unit 158 has two input terminals, one connected to the output terminal of one frame delay unit 157 and the other connected to one output terminal of speech decoding unit 156.

  The demultiplexing unit 155 demultiplexes the multiplexed information input from the signal processing device 113 into audio encoded information and pitch peak position information, and outputs them to the audio decoding unit 156 and the 1-frame delay unit 157, respectively. Further, the signal processing device 113 outputs the frame erasure information to the speech decoding unit 156. The audio decoding unit 156 performs decoding processing using the audio encoding information input from the demultiplexing unit 155 and outputs the decoded audio signal to the D / A conversion device 115. In addition, parameters that need to be updated in the frame erasure compensation process are output to the frame erasure compensation unit 158.

  In addition, in the speech decoding unit 156, when the frame erasure compensation information output from the signal processing device 113 indicates that “the current frame is lost”, the information from the demultiplexing unit 155 is not input. An audio signal is generated using the parameter information input from the frame erasure compensation unit 158 and output to the D / A converter 115. Also at this time, parameters that need to be updated in the frame erasure compensation process are output to the frame erasure compensation unit 158. The one-frame delay unit 157 holds the pitch peak position information input from the demultiplexing unit 155 for a time corresponding to one frame, and then outputs the information to the frame erasure compensation unit 158. The frame erasure compensator 158 outputs the current frame (pitch peak position information sent one frame before the pitch peak position information one frame ahead of the current frame) output from the one frame delay unit 157. The pitch peak position information in the frame (peak peak position information) is input, and the frame erasure compensation process is performed so that the pitch peak position in the current frame matches the position indicated by the input pitch peak position information. The frame erasure compensation process is performed using the last pitch peak position of the current frame input from the 1-frame delay unit 157 and the speech coding parameters decoded up to the previous frame input from the speech decoding unit 156. .

  Incidentally, the reason why the pitch peak position information output from the one-frame delay unit 157 in the speech decoding apparatus 114 is the pitch peak position in the current frame will be described with reference to FIG.

  FIG. 4 is a schematic diagram for explaining the case where the speech encoding information and pitch peak position information of each frame encoded by the speech encoding device 104 are packetized and transmitted. As shown in FIG. 4, in the speech encoding device 104, the encoded speech encoded information is delayed by one frame, and the pitch peak position information is multiplexed without being delayed.

  Accordingly, in FIG. 4, for example, the pitch peak position information 1001p of the frame f2 is transmitted by the first packet 1001 without delay, whereas the speech coding information 1002a of the frame f2 is delayed by one frame, The packet 1002 is transmitted together with the pitch peak position information 1002p of the frame f3.

  The packetized speech coding information and pitch peak position information transmitted from speech coding apparatus 104 (speech signal transmitting apparatus 100) in this way are decoded by speech decoding section 156 of speech signal receiving apparatus 199. . In this case, for example, the pitch peak position information 1001p of the frame f2 transmitted by the packet 1001 waits for the voice encoded information 1002a of the frame f2 transmitted after being delayed by one frame on the transmission side to arrive by the packet 1002. It is processed.

  Next, the speech decoding unit 156 will be described in more detail with reference to FIG. As shown in FIG. 5, the speech decoding unit 156 includes a parameter decoding unit 301, a gain decoder 302, a changeover switch 303, an adaptive codebook 304, a fixed codebook 305, and an LPC decoder that performs linear prediction coefficient decoding processing. 306, a multiplier 307, a multiplier 308, an adder 309, an LPC synthesis filter 310 that is a linear prediction filter constituted by the linear prediction coefficients decoded in the LPC decoder 306, and a post-processing unit 311. The adaptive codebook 304, fixed codebook 305, and gain decoder 302 constitute a sound source decoding unit, and this sound source decoding unit drives the LPC synthesis filter 310.

  The input terminal of the parameter decoder 301 is connected to one of the output terminals of the demultiplexing unit 155. The input terminal of the gain decoder 302 is connected to one of the output terminals of the parameter decoding unit 301. The input terminal of the LPC decoder 306 is connected to one of the output terminals of the parameter decoding unit 301. The input terminal of the changeover switch 303 is connected to the output terminal of the parameter decoding unit 301, the output terminal of the gain decoder, the output terminal of the LPC decoder 306, and the output terminal of the frame erasure compensation unit. In addition, a terminal for receiving frame loss information of the changeover switch 303 is connected to the signal processing device 113 (FIG. 1). The input terminal of the adaptive codebook 304 is connected to the output terminal of the switch 303 and the output terminal of the adder 309.

  The input terminal of the fixed codebook 305 is connected to the output terminal of the changeover switch 303. One of the two input terminals of the multiplier 307 is connected to the output terminal of the adaptive codebook 304, and the other is connected to the output terminal of the changeover switch 303. One of the two input terminals of the multiplier 308 is connected to the fixed codebook 305 and the other is connected to the output terminal of the changeover switch 303. One of the two input terminals of the adder 309 is connected to the output terminal of the multiplier 307, and the other is connected to the output terminal of the multiplier 308. One of the two input terminals of the LPC synthesis filter 310 is connected to the adder 309 and the other is connected to the changeover switch 303. The input terminal of the post-processing unit 311 is connected to the output terminal of the LPC synthesis filter 310, and outputs the digital decoded speech signal to the D / A converter 115.

  The parameter decoding unit 301 converts speech encoding parameters (pitch (adaptive codebook) parameter P, fixed codebook parameter C, linear prediction parameter L, gain parameter G) from speech encoding information (bitstream) input from the demultiplexing unit 155. ), The gain parameter G is output to the gain decoder 302, the linear prediction coefficient parameter L is output to the LPC decoder 306, and the other parameters are output to the changeover switch 303.

  The gain decoder 302 decodes the adaptive codebook gain Gp and the fixed codebook gain Gc from the gain parameter G input from the parameter decoding unit 301, and outputs them to the changeover switch 303. The LPC decoder 306 decodes the decoded quantized linear prediction coefficient α from the linear prediction coefficient parameter L input from the parameter decoding unit 301 and outputs the decoded quantized linear prediction coefficient α to the changeover switch 303. The changeover switch 303 is a switch for switching between a parameter group input from the parameter decoder 301, the gain decoder 302, and the LPC decoder 306 and a parameter group input from the frame erasure compensation unit 158. When the frame erasure information received from the device 113 (FIG. 1) indicates “the current frame is a erasure frame”, the switch is switched to the parameter side generated by the frame erasure compensation processing unit 158; A switch is connected to the parameter side output from the unit 301, the gain decoder 302 and the LPC decoder 306.

  The changeover switch 303 sets the adaptive codebook parameter (pitch) P or P ′ to the adaptive codebook 304, the adaptive codebook gain Gp or Gp ′ to the multiplier 307, and the fixed codebook parameter C or C ′ to the fixed codebook 305. The fixed codebook gain Gc or Gc ′ is output to the multiplier 308, and the decoded quantized linear prediction coefficient α or α ′ is output to the LPC decoder 306, respectively. The changeover switch 303 also outputs the parameters output to the adaptive codebook 304, fixed codebook 305, multiplier 307, multiplier 308, and LPC synthesis filter 310 to the frame erasure compensation unit 158 simultaneously.

The adaptive codebook 304 buffers the excitation signal generated in the past, and is updated each time the latest excitation signal is input from the adder 309. The buffered signal length of the adaptive codebook 304 is equal to or greater than the maximum pitch period length + frame length. An adaptive codebook vector is cut out from the position specified by the adaptive codebook parameter (pitch) P or P ′ input from the changeover switch 303 and output to the multiplier 307, and the sound source signal before one frame is subjected to frame erasure compensation processing. To the unit 158. Multiplier 307 multiplies the adaptive code vector output from adaptive codebook 304 by adaptive codebook gain Gp or Gp ′ input from changeover switch 303 and outputs the result to adder 309.

  The fixed codebook 305 generates a fixed codebook vector designated by the fixed codebook parameter C or C ′ input from the changeover switch 303, and outputs it to the multiplier 308. Multiplier 308 multiplies the fixed codebook vector input from fixed codebook 305 by fixed codebook gain Gc or Gc ′ input from changeover switch 303 and outputs the result to adder 309. Adder 309 adds the adaptive codebook vector input from multiplier 307 and the fixed codebook vector input from multiplier 308, and outputs the result to LPC synthesis filter 310 and adaptive codebook 304.

  The LPC decoder 306 outputs the decoded quantized linear prediction coefficient α from the linear prediction parameter L decoded by the parameter decoding unit 301 to the changeover switch 303. The LPC synthesis filter 310 drives an AR type digital filter constituted by the decoded quantized linear prediction coefficient α or α ′ input from the changeover switch 303 by a sound source signal input from the adder 309, and post-processes the synthesized speech signal To the unit 311. The post-processing unit 311 is a formant emphasis post-filter process for improving the subjective quality of the audio signal, a pitch emphasis post-filter process, a slope correction post-filter process, and a noise post-process for improving the subjective quality of the background noise signal. , And outputs the digital decoded speech signal subjected to these processes to the D / A converter 115.

  Next, details of the frame erasure compensation unit 158 will be described with reference to FIGS. 5 and 6 to 8. The frame erasure compensation unit 158 includes a parameter buffer 312, a pitch peak position detector 313, a pitch period calculator 314, and a parameter generation unit 315.

  An input terminal of the parameter buffer 312 is connected to the changeover switch 303 in the speech decoding unit 156. One of the two input terminals of the pitch peak position detector 313 is connected to the adaptive codebook 304 of the speech decoding unit, and the other is connected to one of the output terminals of the parameter buffer 312 (input pitch information). .

  One of the three input terminals of the pitch period calculator 314 is the pitch peak position detector 313, the other is one frame delay unit 157, and the other is one of the output terminals of the parameter buffer 312 (input pitch information). ) Are connected to each other. Five input terminals of the parameter generation unit 315 are connected to five output terminals of the parameter buffer 312, and the remaining one is connected to the pitch period calculator 314.

  The parameter buffer 312 is a buffer that stores various speech coding parameters used in the past by the speech decoding unit 156 to generate decoded speech. The decoded adaptive codebook parameter (pitch) P, Noise codebook parameter C, adaptive codebook gain Gp, fixed codebook gain Gc, and decoded quantized linear prediction coefficient α are input from speech decoding section 156. In the erasure frame, parameters generated by the parameter generation unit (adaptive codebook parameter (pitch) P ′, fixed codebook parameter C ′, adaptive codebook gain Gp ′, fixed codebook gain Gc ′, decoded quantization linear The prediction coefficient α ′) is input.

  The various parameters buffered are output to the parameter generation unit 315 and used when generating various parameters in the lost frame. The pitch peak position detector 313 uses the excitation signal input from the adaptive codebook 304 in the speech decoding unit 156 and the pitch period of the previous frame input from the parameter buffer 312 to use the pitch peak position detection unit 151 (see FIG. In the same manner as in 1), the pitch peak position is detected and output to the pitch period calculator 314. However, the range where the pitch peak position is searched is the part of the sound source signal generated before the previous frame, and the range where the pitch peak position exists is the pitch of the previous frame from the end point of the sound source signal generated in the previous frame. Up to a point that goes back by the period.

  The pitch period calculator 314 outputs the final pitch peak position of the previous frame input from the pitch peak position detector 313 and the final pitch peak position of the current frame input from the 1 frame delay unit 157 (output from the 1 frame delay unit). Is the pitch peak position information received one frame before, and the pitch peak position information sent one frame before is the pitch corresponding to the voice frame that is currently decoded (with erasure compensation processing). And the pitch period decoded by the speech decoding unit in the previous frame input from the parameter buffer 312 and the current pitch period is obtained. Specifically, the pitch period of the current frame can be determined by (Equation 1).

Pc = (N (PPc) + Nf-N (PPp)) / Int [0.5+ (N (PPc) + Nf-N (PPp)) / Pp] (Formula 1)
However, PPc is the final pitch peak position in the current frame (obtained from information input from the one-frame delay unit 157), and PPp is the final pitch peak position in the previous frame (obtained from information input from the pitch peak position detector 313). , Pp is the pitch period in the previous frame (input from the parameter buffer 312), N (PPx) is a numerical value when the pitch peak position PPx in the frame X is represented by the distance from the head of the frame X, and Nf is the frame length. Show. Int [] indicates the maximum integer that does not exceed the operation result.

  For example, consider the case of frame loss as shown in FIG. FIG. 6 shows a case where the current frame is lost, and the loss compensation process for repeating the waveform of the previous frame is performed using the pitch period of the previous frame. At this time, the final pitch peak position in the waveform of the current frame generated by the frame erasure compensation is PPc ′.

  Incidentally, as shown in FIG. 7, when the pitch period is actually short in the current frame, the actual pitch peak position PPc and the pitch peak position PPc ′ when the frame erasure compensation process as shown in FIG. Deviation occurs between the two. Such a shift causes not only the lost current frame but also the pitch peak position of the adaptive codebook to shift between the encoder side and the decoder side in the subsequent normal reception frame, so that the degradation of the decoded speech quality increases. .

  However, if the pitch peak position information in the lost frame is transmitted together with the speech coding parameter of the previous frame, the pitch period Pc is set as in (Equation 1) so that the pitch peak position PPc in the lost current frame matches. Can be determined. Depending on the quantization accuracy of the pitch, the pitch peak position may not be matched even if it is repeated at the same pitch period, so stepwise using the pitch quantization value in the vicinity of the pitch period Pc obtained by (Equation 1). It is also possible to adjust the pitch peak position by changing the pitch period Pc.

  In addition, when it is a problem that the polarity of the peak value at the pitch peak position does not match, the polarity information is also transmitted and used as the pitch peak position information.

FIG. 8 shows an example in which the frame loss compensation unit 158 obtains the pitch period Pc in this way and performs the frame loss interpolation process using the pitch period Pc. Compared with the decoded signal (dashed line) when there is no frame loss, the pitch level position can be matched although it does not match at the waveform level, so there is a mismatch between the adaptive codebook on the encoder side and the decoder side in the subsequent frame. The influence can be reduced. Further, since only the pitch peak position information is transmitted, the increase in bit rate is small. For example, compared to transmitting primary codec information and secondary codec information as defined in IETF standard RFC2198 (Perkins et al., “RTP Payload for Redundant Audio data ”, Sept. 1997), etc., at a lower bit rate. FEC (Forward Error Correction) can be realized.

  When the pitch peak position information of the current frame indicates that “the current frame is a frame having no pitch periodicity”, the pitch period calculator 314 calculates the pitch period as described above. Without performing, the information which shows that the present frame is a frame which does not have pitch periodicity is output to a parameter generation part.

  The parameter generation unit 315 generates a sound source parameter of a lost frame using the pitch period calculated by the pitch period calculator 314 and various decoding parameters in the previous frame input from the parameter buffer 312. For example, when the information input from the pitch period calculator 314 indicates that “the current frame is a frame having no pitch periodicity”, the parameter generation unit 315 sets the adaptive codebook gain Gp ′ to 0.0. The randomly generated fixed codebook parameter C ′ and the value obtained by attenuating the fixed codebook gain used in the previous frame are output to the changeover switch 303 as Gc ′. On the other hand, when the information input from the pitch period calculator 314 indicates that “the current frame has pitch periodicity and the pitch period is Pc ′”, the parameter generation unit 315 sets the pitch parameter. Set to Pc 'and set adaptive codebook gain Gp' to the value of adaptive codebook gain Gp or Gp that was decoded in the previous frame (eg, a value multiplied by 0.9) or the average value of Gp decoded in the past Then, the fixed codebook gain Gc ′ is set to 0.0 and output to the changeover switch 303. As the decoded quantized linear prediction coefficient α ′, the decoded quantized linear prediction coefficient in the previous frame is used repeatedly, or the bandwidth is expanded to flatten the spectral characteristics.

  Next, operations of speech encoding apparatus 104 and speech decoding apparatus 114 will be described with reference to FIGS. FIG. 9 is a flowchart for explaining the operation of speech encoding apparatus 104, and FIG. 10 is a flowchart for explaining the operation of speech decoding apparatus 114.

  As shown in FIG. 9, speech encoding apparatus 104 first performs CELP speech encoding processing for one frame by speech encoding section 153 in step ST101, and then obtained in step ST101 in step ST102. The encoded speech information is stored by the 1-frame delay unit 154 until the processing of the next frame is completed.

  Next, in step ST103, the speech encoding apparatus 104 uses the pitch peak position detection unit 151 to detect the pitch peak position at the rearmost position in the current frame.

  Next, in step ST104, the speech encoding apparatus 104 extracts the speech encoding information of the previous frame stored in step ST102 (speech encoding information of the previous frame) from the 1-frame delay unit 154, and then performs step ST105. Then, the speech encoding information of the previous frame extracted in step ST104 and the pitch peak position information detected in step ST103 are multiplexed by the multiplexing unit 152, and this is used as the output of the speech encoding device 104. Output.

  Speech encoding apparatus 104 repeats the series of encoding processes in steps ST101 to ST105 described above.

  Next, the operation of the speech decoding apparatus 114 will be described. As shown in FIG. 10, the speech decoding apparatus 114 first determines whether or not the current frame is lost in step ST110 by the signal processing apparatus 113. If it has not disappeared, the process proceeds to step ST111, and if it has disappeared, the process proceeds to step ST114.

  Next, when the frame is not lost, in step ST111, the demultiplexing unit 155 demultiplexes the multiplexed information received.

  Subsequently, in step ST112, the parameter decoding unit 301 in the speech decoding unit 156 decodes speech coding parameters from the separated speech coding information. Next, in step ST113, the remaining part of speech decoding section 156 performs a process of re-synthesizing the speech signal from the decoded speech encoding parameter, and the decoded speech signal is output to D / A converter 115. .

  On the other hand, when it is determined in step ST110 that the frame is lost, the speech decoding apparatus 114 moves to step ST114, and the pitch period calculator 314 determines that the current frame (erasure frame) is the pitch period. It is checked whether or not the frame has sex. Information regarding whether or not there is pitch periodicity is included in the received pitch peak position information. The speech decoding apparatus 114 proceeds to step ST115 if there is pitch periodicity, and proceeds to step ST118 if there is no pitch periodicity. If the current frame (erased frame) has pitch periodicity, the speech decoding apparatus 114 moves to step ST115 and uses the pitch peak position detector 313 to obtain the pitch information of the previous frame and the decoded excitation signal. Read from the parameter buffer 312 and the adaptive codebook 304, respectively.

  Next, the speech decoding apparatus 114 moves to step ST116, and the pitch peak position detector 313 detects the rearmost pitch peak position in the previous frame. When the parameter buffer 312 holds the pitch peak position information received two times before, the pitch peak position in the previous frame may be obtained using the information.

  Next, the speech decoding apparatus 114 moves to step ST117 and uses the pitch period calculator 314 to use the pitch peak position of the current frame (erased frame), the pitch peak position of the previous frame, and the pitch period of the previous frame, The pitch period is calculated by the above (Equation 1).

  Next, the speech decoding apparatus 114 proceeds to step ST118, and the parameter generation unit 315 generates various speech encoding parameters in the lost frame.

  If it is determined that the current frame is a frame having pitch periodicity, the fixed codebook gain is set to zero, and a speech signal is generated using only the adaptive codebook. The pitch period (adaptive codebook parameter) used at this time is the one calculated by the pitch period calculator 314. If it is determined that the current frame is a frame having no pitch periodicity, speech decoding apparatus 114 sets the adaptive codebook gain to zero and generates a speech signal using only the fixed codebook. In this case, fixed codebook parameters are determined randomly. As the linear prediction coefficient, regardless of whether or not the current frame has a pitch periodicity, the previous frame is repeatedly used or gradually whitened while performing bandwidth expansion.

  Finally, in step ST119, the speech decoding apparatus 114 updates the contents of the parameter buffer 312 and ends the decoding process for one frame.

  The speech decoding apparatus 114 repeats a series of decoding processes in steps ST110 to ST119.

  As described above, according to the audio signal transmitting apparatus 100 and the audio signal receiving apparatus 199 of the present embodiment, it is possible to perform the frame erasure compensation process with high accuracy and to adjust the pitch peak position by adding a small amount of redundant information. Thus, it is possible to avoid the shift of the pitch peak position from being propagated after the lost frame, and as a result, it is possible to reduce the influence of error propagation after the lost frame.

(Embodiment 2)
FIG. 11 is a block diagram showing a configuration of an audio signal transmission system according to Embodiment 2 of the present invention.

  The audio signal transmission system illustrated in FIG. 11 includes an audio signal transmission device 400 and an audio signal reception device 499.

  Audio signal transmitting apparatus 400 is obtained by replacing audio encoding apparatus 104 in audio signal transmitting apparatus 100 of Embodiment 1 with audio encoding apparatus 404. Components that are the same as those in the first embodiment and that perform the same operation are denoted by the same reference numerals as those in the first embodiment, and detailed description thereof is omitted.

  The audio encoding device 404 generates audio encoding information by encoding the digital audio signal from the A / D conversion device 103, and provides the generated audio encoding information to the signal processing device 105.

  Audio signal receiving apparatus 499 is obtained by replacing audio decoding apparatus 114 in audio signal receiving apparatus 199 of Embodiment 1 with audio decoding apparatus 414. Components that are the same as those in the first embodiment and that perform the same operation are denoted by the same reference numerals as those in the first embodiment, and detailed description thereof is omitted. The audio decoding device 414 generates a decoded audio signal by decoding the received audio signal from the signal processing device 113 and supplies the decoded audio signal to the D / A conversion device 115.

  Next, the details of speech encoding apparatus 404 will be described with reference to FIGS. 11 and 12.

  In speech encoding apparatus 404 shown in FIGS. 11 and 12, the same components that perform the same operations as speech encoding apparatus 104 of Embodiment 1 shown in FIGS. 1 and 2 are the same as those in Embodiment 1. Reference numerals are assigned and detailed description is omitted.

The input terminal of the 1-frame delay unit 452 is connected to the output terminal of the pitch peak position detection unit 151. One of the two input terminals of the multiplexing unit 454 is connected to the output terminal of the one- frame delay unit 452, and the other is connected to the output terminal of the speech encoding unit 153.

  The 1-frame delay unit 452 holds the most recent pitch peak position information in the current frame, which is output from the pitch peak position detection unit 151, and outputs the information to the multiplexing unit 454 after holding for one frame time. The multiplexing unit 454 multiplexes the pitch peak position information of the previous frame input from the 1-frame delay unit 452 and the encoded speech information of the current frame input from the speech encoding unit 153 to be combined into one. The encoded information is output to the D / A converter 115.

  That is, the difference between the first embodiment and the second embodiment is a difference in which one of the pitch peak position information and the voice encoded information is delayed by one frame and multiplexed. The second embodiment delays the pitch peak position information, and the first embodiment delays the speech coding information. All other operations in the speech coding apparatus according to Embodiment 1 and Embodiment 2 are the same.

  Incidentally, FIG. 13 is a schematic diagram for explanation when the speech coding apparatus 404 packetizes and transmits the speech coding information and pitch peak position information of each encoded frame. As shown in FIG. 13, in the speech encoding device 404, the pitch peak position information is delayed by one frame, and the encoded speech information is multiplexed without being delayed.

  Therefore, in FIG. 13, for example, the audio coding information 2001a of the frame f2 is transmitted by the first packet 2001 without delay, whereas the pitch peak position information 2002p of the frame f2 is delayed by one frame, The packet 2002 is transmitted together with the audio coding information 2002a of the frame f3.

  Thus, the packetized speech coding information and pitch peak position information transmitted from speech coding apparatus 404 (speech signal transmitting apparatus 400) are decoded by speech decoding section 456 of speech signal receiving apparatus 499. . In this case, for example, the pitch peak position information 2002p of the frame f2 transmitted by the packet 2002 is transmitted after being delayed by one frame on the transmission side. This is the pitch peak position information of the frame. In the audio decoding unit 456, when the audio encoding information of the frame to be currently decoded is lost based on the frame erasure information, the frame erasure compensation unit uses the audio encoding information decoded in the previous frame. 457 performs general frame loss compensation. If the previous frame is a lost frame, the lost frame is delayed by delaying the frame lost information of the previous frame by one frame by a one-frame delay unit 601 (described later) of the audio decoding unit 456. In the current frame that follows, the adaptive codebook corrector 603 (described later) is operated by this frame erasure information. Thereby, adaptive codebook 304 generated based on the parameter compensated by frame erasure compensation unit 457 in the previous frame is corrected.

  Next, speech decoding apparatus 414 according to Embodiment 2 will be described in detail with reference to FIG. 11 and FIG.

  In the speech decoding apparatus 414 shown in FIG. 14, the same components as those in the speech decoding apparatus 114 according to the first embodiment shown in FIG. Description is omitted.

  As illustrated in FIG. 14, the speech decoding apparatus 414 includes a demultiplexing unit 455, a speech decoding unit 456, and a frame erasure compensation unit 457.

  The input terminal of the demultiplexing unit 455 is connected to one of the output terminals of the signal processing device 113. The eight input terminals of the speech decoding unit 456 are one each for the two output terminals of the demultiplexing unit 455, one output terminal of the signal processing device 113, and five output terminals of the frame erasure compensation unit 457. Each is connected. One of the six input terminals of the frame erasure compensation unit 457 is connected to one output terminal of the demultiplexing unit 455, and the remaining five are connected to the five output terminals of the speech decoding unit 456. The remaining one output terminal of the speech decoding unit is connected to the D / A converter 115.

  The demultiplexing unit 455 separates the speech coding parameter information and the pitch peak position information from the coding information output from the signal processing device 113 (FIG. 1), and outputs both to the speech decoding unit 456. The demultiplexing unit 455 also outputs pitch peak position information (PP) to the frame erasure compensation unit 457. The audio decoding unit 456 receives the frame erasure information from the signal processing device 113, and when the frame to be decoded (current frame) is an erasure frame, the audio decoding unit 456 generates the frame erasure compensation unit 457 by switching the changeover switch 303. A speech signal is synthesized using the speech coding parameters to be output to the D / A converter 115. If the frame to be decoded is not an erasure frame, decoding processing is performed using the audio encoding parameter output from the demultiplexing unit 455, an audio signal is synthesized, and output to the D / A converter 115. However, when the previous frame is an erasure frame, the content of the adaptive codebook generated in the previous frame is corrected using the pitch peak position (of the previous frame) output from the demultiplexing unit 455, and then the decoding process is performed. I do.

  The details of the speech decoding unit 456 will be described with reference to FIG. In FIG. 14, the speech decoding unit 414 is different from the speech decoding unit 114 in the first embodiment shown in FIG. 5 in that a 1-frame delay unit 601, a changeover switch 602, and an adaptive codebook modifier 603 are new. It is a point added to. Since the other components perform the same operation with the same configuration, the same reference numerals as those in FIG.

  The input terminal of the 1-frame delay unit 601 is connected to one of the output terminals of the signal processing unit 113. One of the two input terminals of the changeover switch 602 is connected to the adaptive codebook 304 and the other is connected to the adaptive codebook modifier 603. In addition, switch changeover control information is input from the 1-frame delay unit 601 to the changeover switch 602. One of the two input terminals of the adaptive codebook corrector 603 is connected to one of the input terminals of the changeover switch 602, and the other is connected to one of the output terminals of the demultiplexing unit 455.

  The 1-frame delay unit 601 receives frame erasure information from the signal processing device 113, holds it for one frame time, and then outputs it to the changeover switch 602. The changeover switch 602 is turned on (connected) when the information input from the one-frame delay unit 601 indicates that the frame (= previous frame) is lost, and is turned off when the frame is not lost. (Released). The two switches are linked and are turned ON / OFF at the same time. When the changeover switch 602 is turned on, the contents of the adaptive codebook 304 are read and output to the adaptive codebook modifier 603. The read contents of the adaptive codebook 304 are corrected by the adaptive codebook modifier 603, and then output to the adaptive codebook 304 via the changeover switch 602, so that the contents of the adaptive codebook are rewritten. The voice decoding process is performed after the adaptive codebook 304 is rewritten.

  Adaptive codebook modifier 603 receives from the demultiplexing unit 455 the pitch peak position information at the rearmost of the previous frame to be decoded, and the sound source generated in the previous frame stored in adaptive codebook 304 The content of the excitation signal buffer stored in the adaptive codebook 304 is corrected so that the pitch peak position of the portion matches the pitch peak position. Specifically, in the same manner as the method described in the first embodiment, the final pitch peak position PPp ′ in the second previous frame (using the sound source signal and the pitch period Pp ′ in the second previous frame) And the final pitch peak position PPp in the previous frame (obtained from the pitch peak position information received in the current frame) in the previous frame. (Expression 2) calculates a pitch period Pp in the previous frame, and generates a modified adaptive codebook by redoing the frame erasure compensation process in the previous frame using Pp.

Pp = (N (PPp) + Nf−N (PPp ′)) ÷ Int [0.5+ (N (PPp) + Nf−N (PPp ′)) / Pp ′] (Formula 2)
Alternatively, the pitch peak position PPp in the previous frame is simply obtained using the excitation signal generated in the previous frame stored in the adaptive codebook 304 before correction, and the obtained pitch peak position and A method of performing alignment by shifting the contents of the adaptive codebook by the difference from the pitch peak position obtained from the pitch peak position information received in the current frame is also possible.

  The adaptive codebook corrected in this way is output to the adaptive codebook 304 via the changeover switch 602, and the contents of the adaptive codebook 304 are corrected. After correcting the contents of adaptive codebook 304, the speech signal is decoded by the same operation as speech decoding section 156 described in the first embodiment.

  Next, the frame loss compensation unit 457 will be described in detail with reference to FIG. The frame erasure compensation unit 457 includes a parameter buffer 312 and a parameter generation unit 604. Since the operation of the parameter buffer 312 is the same as that of the first embodiment, the description thereof is omitted. The operation of the parameter generation unit 604 is basically the same as that of the parameter generation unit 315 of the first embodiment. That is, for example, when the pitch peak position information PP indicates that “the previous frame is a frame having no pitch periodicity”, the adaptive codebook gain Gp ′ is set to 0.0, and a randomly generated fixed code A value obtained by attenuating the book parameter C ′ and the fixed codebook gain used in the previous frame is output to the changeover switch 303 as Gc ′. On the contrary, when the pitch peak position information PP indicates that “the previous frame has pitch periodicity”, the adaptive codebook gain Gp or Gp obtained by decoding the adaptive codebook gain Gp ′ in the previous frame is attenuated. (For example, a value multiplied by 0.9) or an average value of Gp decoded in the past, the fixed codebook gain Gc ′ is set to 0.0, and each is output to the changeover switch 303.

  For the pitch period information, the pitch period of the previous frame stored in the parameter buffer 312 or a pitch period corresponding thereto (added random fluctuation or incremented by one sample) is used, and for the fixed codebook parameter, a random number is used. The codebook index generated in step 1 is used. Note that as the decoded quantized linear prediction coefficient α ′, the decoded quantized linear prediction coefficient in the previous frame is repeatedly used, or the bandwidth is expanded to flatten the spectral characteristics.

  Thus, according to audio signal transmitting apparatus 400 and audio signal receiving apparatus 499 of the present embodiment, the effect of error propagation after frame loss can be reduced without an increase in delay.

(Embodiment 3)
FIG. 15 is a block diagram showing a configuration of an audio signal transmission system according to Embodiment 3 of the present invention.

  The audio signal transmission system includes an audio signal transmitter 700 and an audio signal receiver 799.

  Since audio signal transmitting apparatus 700 is the same as audio signal transmitting apparatus 400 in the second embodiment, the same reference numerals as those in the first and second embodiments are given to the respective components, and detailed description thereof is omitted.

  Audio signal receiving apparatus 799 is obtained by replacing audio decoding apparatus 414 in audio signal receiving apparatus 499 of Embodiment 2 with audio decoding apparatus 714. Components having the same operations as those of the first and second embodiments are denoted by the same reference numerals as those of the first and second embodiments, and the description thereof is omitted. The audio decoding device 714 generates a decoded audio signal by decoding the received audio signal from the signal processing device 113 and supplies the decoded audio signal to the D / A conversion device 115.

  Next, speech decoding apparatus 714 according to Embodiment 3 will be described in detail with reference to FIG. 15 and FIG.

  The speech decoding apparatus 714 shown in FIGS. 15 and 16 performs the same operation as the speech decoding apparatus 114 of the first embodiment shown in FIG. 5 and the speech decoding apparatus 414 of the second embodiment shown in FIG. The same constituent elements are denoted by the same reference numerals as those in the first and second embodiments, and detailed description thereof is omitted.

  As shown in FIGS. 15 and 16, speech decoding apparatus 714 includes demultiplexing section 755, speech decoding section 756, and frame erasure compensation section 757.

  The input terminal of the demultiplexing unit 755 is connected to one of the output terminals of the signal processing device 113. The three input terminals of the speech decoding unit 756 are connected to the output terminal of the demultiplexing unit 755, one output terminal of the signal processing device 113, and the output terminal of the frame erasure compensation unit 757, respectively. One of two input terminals of the frame erasure compensation unit 757 is connected to one output terminal of the demultiplexing unit 755, and the other is connected to one of the two output terminals of the speech decoding unit 756. The other output terminal of the speech decoding unit 756 is connected to the D / A conversion device 115.

  The demultiplexing unit 755 separates the speech coding parameter information and the pitch peak position information, and outputs the speech coding parameter information to the speech decoding unit 756 and the pitch peak position information to the frame erasure compensation unit 757, respectively. The audio decoding unit 756 receives the frame erasure information from the signal processing device 113, and is generated by the frame erasure compensation unit 757 when the frame to be decoded (the frame before the currently received frame) is an erasure frame. A speech signal is synthesized using speech coding parameters and output to the D / A converter 115. If the frame to be decoded is not an erasure frame, decoding processing is performed using the audio encoding parameter output from the demultiplexing unit 755, the audio signal is synthesized, and output to the D / A converter 115. The difference from Embodiment 2 is that the frame to be decoded is received one frame before. That is, decoding is performed after waiting for one frame on the decoder side. In this way, for example, when a frame to be decoded is lost and frames before and after the lost frame are received, it is possible to perform lost frame compensation processing using the preceding and following frame information, It is possible to perform erasure compensation processing so as to smoothly change between the previous and next frames.

  With reference to FIG. 16, the operation of speech decoding apparatus 714 will be described in detail. As shown in FIG. 16, in speech decoding section 756, parameter decoding section 301 in speech decoding section 156 of Embodiment 1 is replaced with parameter decoding section 801 and 1-frame delay section 802. A 1-frame delay unit 803 is inserted between the signal processing device 113 and the changeover switch 303. These two points are different from the first embodiment. Since the other components perform the same operation with the same configuration, the same reference numerals as those in the first embodiment are given, and detailed description thereof is omitted.

  In FIG. 16, the parameter decoding unit 801 receives speech coding information from the demultiplexing unit 755, separates each speech coding parameter, and outputs it to the 1-frame delay unit 802. At the same time, parameter decoding section 801 outputs adaptive codebook parameter Pn (adaptive codebook parameter in the frame next to the frame to be decoded) to pitch period calculator 814 in frame erasure compensation section 757.

  The 1-frame delay unit 802 is a pitch (adaptive codebook) parameter P (pitch (adaptive codebook) parameter of a frame to be decoded from now on) after holding each parameter output from the parameter decoding unit 801 for one frame time, 1 frame before Pn) and fixed codebook parameter C are output to the selector switch 303, respectively. At the same time, the 1-frame delay unit 802 outputs the gain parameter G to the gain decoder 302. At the same time, the 1-frame delay unit 802 outputs the linear prediction coefficient parameter L to the LPC decoder 306. The 1-frame delay unit 803 receives the frame erasure information output from the signal processing device 113, holds it for one frame time, and then outputs it to the changeover switch 303.

  Next, the frame loss compensation unit 757 will be described in detail with reference to FIG. The frame erasure compensation unit 757 includes a pitch peak position detector 313, a parameter buffer 812, a pitch period calculator 814, and a parameter generation unit 815.

  Pitch peak position detector 313 performs the same operation as pitch peak position detector 313 in the first embodiment. That is, the final pitch peak position in the previous frame of the frame being decoded is detected using the pitch period input from the parameter buffer 812 and the excitation signal input from the adaptive codebook 304 and output to the pitch period calculator 814. To do. If the pitch peak position is normally received in the frame preceding the frame being decoded, the information may be buffered and used. The parameter buffer 812 performs the same operation as the parameter buffer 312 of the first and second embodiments. However, the only difference is that the buffering parameter is traced back by one frame in the past due to the presence of the one-frame delay unit 802.

  The pitch period calculator 814 sets the pitch peak position information PPc of the decoded frame from the demultiplexing unit 755, the pitch period Pn of the next frame of the decoded frame from the parameter decoding unit 801, and the pitch period information Pp before the decoded frame as parameters. The pitch peak position information PPp of the frame before the decoded frame is input from the buffer 812 from the pitch peak position detector 313, and the pitch period of the lost frame (decoded frame) is calculated and output to the parameter generation unit 815.

  When the pitch peak position information PPc input from the demultiplexing unit 755 indicates that “the frame does not have pitch periodicity”, the pitch cycle calculator 814 sends the information to the parameter generation unit. Output.

  The parameter generation unit 815 performs the same operation as the parameter generation unit 315 of the first embodiment. That is, the sound source parameter of the lost frame is generated using the pitch period calculated by the pitch period calculator 814 and various decoding parameters in the previous frame input from the parameter buffer 812.

  For example, if the information input from the pitch period calculator 814 indicates that “the frame to be decoded is a frame that does not have pitch periodicity”, the adaptive codebook gain Gp ′ is set to 0.0 and randomly generated. The fixed codebook parameter C ′ and the value obtained by attenuating the fixed codebook gain used in the previous frame are output to the changeover switch 303 as Gc ′.

  On the other hand, when the information input from the pitch period calculator 814 indicates that “the frame to be decoded has pitch periodicity and the pitch period is Pc ′”, the pitch period is converted into the pitch period calculator 814. Is set to the value P ′ obtained in step 1, and the adaptive codebook gain Gp ′ obtained by decoding the adaptive codebook gain Gp ′ in the previous frame is attenuated (for example, a value obtained by multiplying 0.9) or the average of Gp decoded in the past The fixed codebook gain Gc ′ is set to 0.0, and output to the changeover switch 303.

  Note that as the decoded quantized linear prediction coefficient α ′, the decoded quantized linear prediction coefficient in the previous frame is repeatedly used, or the bandwidth is expanded to flatten the spectral characteristics.

  FIG. 17 shows the pitch period Pn of the frame next to the decoded frame, the pitch peak position information N (PPc) of the decoded frame, the pitch period Pp of the frame before the decoded frame, and the pitch of the frame before the decoded frame. FIG. 5 shows a principle diagram for obtaining a pitch period Pc of a decoded frame using a period Pp and pitch peak position information N (PPp) of a frame before the decoded frame (obtained from the excitation signal of the adaptive codebook).

  Information on the pitch peak position PPc of the decoded frame is supplied from the demultiplexing unit 755. Information on the pitch peak position PPp of the previous frame is supplied from the pitch peak position detector 313. The pitch period Pp of the previous frame is supplied from the parameter buffer 812. The pitch period Pn of the next frame is supplied from the parameter decoding unit 801. Using the pitch peak position PPc of the decoded frame, the pitch peak position PPp of the previous frame, the pitch period Pp of the previous frame, and the pitch period Pn of the next frame, the pitch period Pc of the decoded frame is calculated from (Equation 3). Ask. N (PPx) represents the distance from the beginning of the frame X to the pitch peak position PPx, Nf represents the subframe length, and Int [] represents the maximum integer value less than the calculation result in [].

Pc = (N (PPc) + Nf -N (PPp)) /
Int [((N (PPc) + Nf−N (PPp)) / Pp + (N (PPc) + Nf−N (PPp)) / Pn) × 0.5 + 0.5] (Formula 3)
For example, in the case of FIG. 17, the distance between PPc and PPp is 3.1 periods in the period Pp and 2.8 periods in the period Pn, so (3.1 + 2.8) × 0.5 = 2.95. Therefore, the denominator is 3 periods, and the value obtained by dividing N (PPc) + Nf-N (PPp) by 3 is the pitch period.

  By obtaining the pitch period of the decoded frame, which is an erasure frame, by such a method, it is possible to perform frame erasure compensation processing in which the pitch continuously changes using an intermediate pitch period of the preceding and succeeding frames. Further, since the pitch peak position in the lost frame can be matched as in the second embodiment, it is possible to reduce the influence of the mismatch between the adaptive codebooks on the encoder side and the decoder side in the subsequent normal frame. .

  As described above, according to audio signal transmitting apparatus 700 and audio signal receiving apparatus 799 of the present embodiment, it is possible to realize an accurate frame erasure compensation method by adding a small amount of redundant information, and to perform error propagation after an erasure frame. Can reduce the effects of

(Embodiment 4)
FIG. 18 is a block diagram showing a configuration of an audio signal transmission system according to Embodiment 4 of the present invention.

  The audio signal transmission system illustrated in FIG. 18 includes an audio signal transmission apparatus 1000 and an audio signal reception apparatus 1199.

  Audio signal transmitting apparatus 1000 is obtained by replacing audio encoding apparatus 104 in audio signal transmitting apparatus 100 of Embodiment 1 with audio encoding apparatus 1104. Components that are the same as those in the first embodiment and that perform the same operation are denoted by the same reference numerals as those in the first embodiment, and detailed description thereof is omitted.

  The speech encoding device 1104 generates speech encoding information by encoding the digital speech signal from the A / D conversion device 103, and provides the generated speech encoding information to the signal processing device 105.

  Audio signal receiving apparatus 1199 is obtained by replacing audio decoding apparatus 114 in audio signal receiving apparatus 199 of Embodiment 1 with audio decoding apparatus 1114. Components that are the same as those in the first embodiment and that perform the same operation are denoted by the same reference numerals as those in the first embodiment, and detailed description thereof is omitted. The audio decoding device 1114 generates a decoded audio signal by decoding the received audio signal from the signal processing device 113 and supplies the decoded audio signal to the D / A conversion device 115.

  Next, the speech encoding apparatus 1104 will be described in detail with reference to FIGS. 18, 19, 20 and 21. FIG. FIG. 19 is a block diagram showing the configuration of the speech coding apparatus 1104, FIG. 20 is a schematic diagram for explaining the operation of the pitch peak position detection unit, and FIG. 21 is a flowchart showing the pitch peak position detection processing procedure. is there. However, in the speech encoding apparatus 1104 shown in FIG. 18 and FIG. 19, the same components as those in the speech encoding apparatus 104 of Embodiment 1 shown in FIG. 1 and FIG. The same reference numerals are assigned and detailed description is omitted.

  As illustrated in FIG. 18, the speech encoding apparatus 1104 includes a pitch peak position detection unit 151, a pitch peak position information embedding unit 1152, a speech encoding unit 153, and a 1-frame delay unit 154. The input terminal of the speech encoding unit 153 is connected to the output terminal of the A / D conversion device 103. The input terminal of the pitch peak position detector 151 is connected to one of the two output terminals of the speech encoder 153. The input terminal of the 1-frame delay unit 154 is connected to the other of the two output terminals of the speech encoding unit 153. The pitch peak position information embedding unit 1152 is connected between the output terminals of the pitch peak position detection unit 151 and the one-frame delay unit 154 and the input terminal of the signal processing device 105.

  The audio encoding unit 153 performs encoding processing on the digital audio signal input from the A / D conversion apparatus 103 and outputs encoding parameter information to the 1-frame delay unit 154. At the same time, speech encoding section 153 outputs a pitch parameter (quantization pitch period), which will be described later, and a sound source signal held in the adaptive codebook to pitch peak position detection section 151.

  The pitch peak position detection unit 151 detects a pitch peak position in a sound source signal having a past pitch period length using the quantized pitch period information that is a pitch parameter and a past sound source signal sequence, and a pitch peak position information embedding unit Output to 1152.

  It is assumed that the pitch peak position is closest to the end of the frame, which exists in the range of one pitch period in the past from the end of the current frame. Therefore, when one frame is divided into a plurality of subframes, pitch peak position detection is performed in the last subframe. Even if the pitch period is long and there is no pitch peak in the current frame, the pitch peak position in the previous frame section is detected by searching for the pitch peak position within the range of one pitch period from the end of the current frame. I will do it.

  The pitch peak position information embedding unit 1152 embeds the pitch peak position information in the current frame detected by the pitch peak position detection unit 151 in the speech encoded information in the previous frame output from the 1 frame delay unit 154, and The data is output to the processing device 105. By this processing, the specific information of the speech encoded information in the previous frame is replaced with the pitch peak position information. Therefore, there is no increase in the amount of information due to the embedding of pitch peak position information. It is assumed that speech coding information to be replaced has been selected in advance to have a small influence on the quality of decoded speech. Incidentally, the pitch peak position information may be embedded intermittently. In this way, the pitch peak position information is transmitted at the minimum necessary frequency or frame, and quality degradation due to a decrease in the bit rate of the encoded audio signal can be suppressed.

  As shown in FIG. 19, in speech encoding section 153, adaptive codebook parameter P, fixed codebook parameter C, gain codebook parameter G determined by excitation parameter determining section, and encoded by LPC quantizer The linear prediction parameter L is collectively collected as one code by the encoding unit 215 and output to the 1-frame delay unit 154. The 1-frame delay unit 154 holds the speech coding information input from the coding unit 215 for one frame time, and then outputs the speech coding information to the pitch peak position information embedding unit 152.

  Next, the operation of the pitch peak position detector 151 will be described in more detail with reference to FIG.

  As shown in FIG. 20, pitch peak position detection section 151 inputs pitch parameter P and the latest adaptive codebook content (sound signal sequence generated in the past) at least one pitch period length or longer. The pitch peak position behind is detected. The simplest method for detecting the pitch peak position is traced back to one pitch period P in the past from the end of the adaptive codebook (latest sample: sample 1101 in FIG. 20, which is e [-1]). In this method, the sample having the maximum absolute value (sample 1102 in FIG. 20) is detected as the pitch peak position.

  However, with such a simple method, a peak different from the original pitch peak may be detected as a pitch peak due to the influence of sampling or the like. In such a case, as shown in (Equation 4), by calculating c [i] taking into account the point one pitch backward, the point where the absolute value of the calculated value is the largest is detected as the pitch peak position. The pitch peak position detection accuracy can be improved (step ST1601 to step ST1606 in FIG. 21, peak1 is the pitch peak position).

c [i] = e [i] + e [iP], i = -1,…, -P (Formula 4)
When the pitch period is given with fractional accuracy, the accuracy is further increased by using the value obtained by interpolating the second term of (Equation 4) using fractional accuracy.

  Further, a point at which the amplitude becomes maximum before and after the point at which (Equation 4) is maximized (for example, about plus or minus 5 samples (at 8 kHz sampling)) is re-searched to obtain the final pitch peak position (FIG. 21). Step ST1607 to step ST1611, a is a constant (for example, about 5), and the accuracy is further improved. FIG. 21 is a flowchart of such a pitch peak position detection processing algorithm.

  If the current input audio signal has no pitch periodicity (if it is a voiceless part or a noise part), the pitch peak position is not set excessively and a code indicating that there is no pitch periodicity is used. It may be assigned separately and the information may be output as pitch peak position information.

  The pitch peak position information embedding unit 1152 includes the pitch peak obtained by the pitch peak position detection unit 151 in a portion of the speech coding information input via the one-frame delay unit 154 that has a relatively small influence on decoding. Write location information.

  For example, according to ITU-T recommendation G.729, among the LSP encoded information, the 5 bits allocated to the higher order side in the second stage of vector quantization can improve the quality of decoded speech even if erroneous on the transmission path. It is known that the effect is relatively small (Kataoka, Hayashi, “Bit error sensitivity of ITU-T standard 8 kbit / s speech coding”, 1995 IEICE Information and Systems Society Conference, D-251) Therefore, it can be considered that these 5 bits are used to represent the pitch peak position.

  Furthermore, if it is not necessary to consider measures against bit errors on the transmission path, it is conceivable to use 6 bits including 1 bit of pitch parity. It should be noted that the embedding of the pitch peak position information can be performed every fixed number of frames instead of every frame, thereby reducing the influence on the decoded speech.

  Next, speech decoding apparatus 1114 will be described in detail with reference to FIGS. 18, 22, 23 and 24, and FIGS. 6 to 8 used in the description of the first embodiment. FIG. 22 is a block diagram showing a configuration of the speech decoding apparatus 1114, and FIGS. 23 and 24 are flowcharts showing speech encoding and decoding processing procedures. However, in the speech decoding apparatus 1114 shown in FIGS. 18 and 22, the same constituent elements that perform the same operations as those of the speech decoding apparatus 114 of Embodiment 1 shown in FIGS. The same reference numerals are assigned and detailed description is omitted.

  As illustrated in FIG. 18, the speech decoding apparatus 1114 includes a pitch peak position information separating unit 1155, a speech decoding unit 156, a frame delay unit 157, and a frame erasure compensation unit 158.

  The input terminal of the pitch peak position information separation unit 1155 is connected to one of the two output terminals of the signal processing device 113. The speech encoding unit 156 has three input terminals, one for one output terminal of the pitch peak position information separation unit 1155, one for one output terminal of the signal processing device 113, and one for frame erasure compensation. The unit 158 is connected to each. Speech encoding section 156 has two output terminals, one connected to one of the two input terminals of frame erasure compensation section 158, and the other connected to D / A converter 115. The input terminal of the 1-frame delay unit 157 is connected to one of the output terminals of the pitch peak position information separation unit 1155. Frame erasure compensation unit 158 has two input terminals, one connected to the output terminal of one frame delay unit 157 and the other connected to one output terminal of speech decoding unit 156.

  The pitch peak position information separation unit 1155 separates the pitch peak position information from the encoded information in which the pitch peak position information input from the signal processing device 113 is embedded, and sends the pitch peak position information to the one-frame delay unit 157. The other speech coding information is output to speech decoding section 156, respectively.

  Further, the signal processing device 113 outputs the frame erasure information to the speech decoding unit 156. Speech decoding section 156 performs decoding processing using the speech encoded information input from pitch peak position information separation section 1155 and outputs the decoded speech signal to D / A conversion device 115. Speech decoding section 156 also outputs parameters that need to be updated in the frame erasure compensation process to frame erasure compensation section 158.

  When the frame erasure compensation information input from the signal processing device 113 indicates that “the current frame has been lost”, the information from the pitch peak position information separation unit 1155 is not input, so the frame erasure compensation unit An audio signal is generated using the parameter information input from 158 and output to the D / A converter 115. Also at this time, parameters necessary for the frame erasure compensation process are output to the frame erasure compensation unit 158. The 1-frame delay unit 157 holds the pitch peak position information input from the pitch peak position information separation unit 155 for a time corresponding to one frame, and then outputs the information to the frame loss compensation unit 158.

  The frame erasure compensator 158 inputs the current frame (the pitch peak position information sent one frame before is the pitch peak position information one frame ahead of the previous frame, which is input from the one frame delay unit 157. Pitch peak position information) is input, and the frame erasure compensation process is performed so that the pitch peak position in the current frame matches the position indicated by the input pitch peak position information. The frame erasure compensation process is performed using the last pitch peak position of the current frame input from the 1-frame delay unit 157 and the speech coding parameters decoded up to the previous frame input from the speech decoding unit 156. .

  Also, as shown in FIG. 22, in the speech decoding unit 156, the parameter decoding unit 301 converts speech coding parameters (pitch (adaptive code) from speech coding information (bitstream) input from the pitch peak position information separating unit 1155. Book) parameter P, fixed codebook parameter C, linear prediction parameter L, gain parameter G), gain parameter G to gain decoder 302, linear prediction coefficient parameter L to LPC decoder 306, and other parameters Each is output to the changeover switch 303. The subsequent processing is the same as that of the speech decoding unit 156 described above with reference to FIG.

  Next, details of the frame erasure compensation unit 158 will be described with reference to FIGS. 22 and 6 to 8. The frame erasure compensation unit 158 includes a parameter buffer 312, a pitch peak position detector 313, a pitch period calculator 314, and a parameter generation unit 315.

  The parameter buffer is a buffer for storing various speech coding parameters used in the past by the speech decoding unit 156 to generate decoded speech. The decoded adaptive codebook parameter (pitch) P, noise code Book parameter C, adaptive codebook gain Gp, fixed codebook gain Gc, and decoded quantized linear prediction coefficient α are input from speech coding section 156. In the erasure frame, parameters generated by the parameter generation unit (adaptive codebook parameter (pitch) P ′, fixed codebook parameter C ′, adaptive codebook gain Gp ′, fixed codebook gain Gc ′, decoded quantization linear The prediction coefficient α ′) is input. The various parameters buffered are output to the parameter generation unit 315 and used when generating various parameters in the lost frame.

  The pitch peak position detector 313 uses the excitation signal input from the adaptive codebook 304 in the speech decoding unit 156 and the pitch period of the previous frame input from the parameter buffer 312, similarly to the pitch peak position detection unit 151. The pitch peak position is detected and output to the pitch period calculator 314. However, the range where the pitch peak position is searched is the part of the sound source signal generated before the previous frame, and the range where the pitch peak position exists is the pitch of the previous frame from the end point of the sound source signal generated in the previous frame. Up to a point that goes back by the period.

  The pitch period calculator 314 outputs the final pitch peak position of the previous frame input from the pitch peak position detector 313 and the final pitch peak position of the current frame input from the 1 frame delay unit 157 (output from the 1 frame delay unit). Is the pitch peak position information received one frame before, and the pitch peak position information sent one frame before is the pitch corresponding to the voice frame that is currently decoded (with erasure compensation processing). And the pitch period decoded by the speech decoding unit in the previous frame input from the parameter buffer 312 and the current pitch period is obtained. Specifically, the pitch period of the current frame can be determined by (Equation 5).

Pc = (peak_c−peak_p) ÷ Int [0.5+ (peak_c−peak_p) / Pp] (Formula 5)
However, peac_c is obtained from the final pitch peak position in the current frame (obtained from information inputted from the one-frame delay unit 157), and peak_p is obtained from the final pitch peak position in the previous frame (information inputted from the pitch peak position detector 313). Pp indicates the pitch period (input from the parameter buffer 312) in the previous frame, and peak_p and peak_c are expressed as positions with the start of the current frame as a reference (for example, 0). The Int [] indicates the maximum integer that does not exceed the operation result.

  For example, consider the case of frame loss as shown in FIG. FIG. 6 shows a case where the current frame is lost, and the loss compensation process for repeating the waveform of the previous frame is performed using the pitch period of the previous frame. At this time, the final pitch peak position in the waveform of the current frame generated by the frame erasure compensation is PPc ′.

  By the way, as shown in FIG. 7, when the pitch period is actually short in the current frame (thick line), when the actual pitch peak position PPc and the frame erasure compensation processing as shown in FIG. A deviation occurs from the pitch peak position PPc ′. Such a shift causes not only the lost current frame but also the pitch peak position of the adaptive codebook to shift between the encoder side and the decoder side in the subsequent normal reception frame, so that the degradation of the decoded speech quality increases. .

  However, if the pitch peak position information in the lost frame is transmitted together with the speech coding parameter of the previous frame, the pitch period Pc is set as in (Equation 5) so that the pitch peak position PPc in the lost current frame matches. Can be determined. Depending on the quantization accuracy of the pitch, the pitch peak position may not be matched even if it is repeated at the same pitch period, so stepwise using the pitch quantization value near the pitch period Pc obtained by (Equation 5). It is also possible to adjust the pitch peak position by changing the pitch period Pc.

  In addition, when it is a problem that the polarity of the peak value at the pitch peak position does not match, the polarity information is also transmitted and used as the pitch peak position information. FIG. 8 (thick line) shows an example in which the pitch period Pc is obtained in this way and the frame erasure interpolation process is performed using the pitch period Pc.

  Although the decoded signal (dashed line) does not match the waveform level when there is no frame loss, the pitch peak position can be matched, so the influence of the mismatch between the adaptive codebook on the encoder side and the decoder side in the subsequent frame It becomes possible to reduce. In addition, since the pitch peak position information is embedded in the speech encoded information (pitch peak position information is transmitted instead of speech encoded information that has little influence on the decoded speech), there is no increase in bit rate.

  When transmitting two or more frames in one packet, it is only necessary to send only the pitch peak position information related to the last frame in one packet, so there is no need to embed pitch peak position information for the other frames. The influence on the decoded audio signal is further reduced. When the pitch peak position information of the current frame indicates that “the current frame is a frame having no pitch periodicity”, the pitch period is not calculated as described above, and the current frame has a pitch. Information indicating that the frame does not have periodicity may be output to the parameter generation unit.

  The parameter generation unit 315 generates a sound source parameter of a lost frame using the pitch period calculated by the pitch period calculator 314 and various decoding parameters in the previous frame input from the parameter buffer 312. For example, when the information input from the pitch period calculator 314 indicates that “the current frame is a frame having no pitch periodicity”, the adaptive codebook gain Gp ′ is set to 0.0 and randomly generated. A value obtained by attenuating the fixed codebook parameter C ′ and the fixed codebook gain used in the previous frame is output to the changeover switch 303 as Gc ′.

  On the other hand, if the information input from the pitch period calculator 314 indicates that “the current frame has pitch periodicity and the pitch period is Pc ′”, the pitch parameter is set to Pc ′, The adaptive codebook gain Gp ′ is set to an adaptive codebook gain Gp decoded in the previous frame, an attenuated Gp value (for example, a value multiplied by 0.9) or an average value of Gp decoded in the past, and the fixed codebook gain Gc ′ is set to 0.0 and output to the changeover switch 303.

  Note that as the decoded quantized linear prediction coefficient α ′, the decoded quantized linear prediction coefficient in the previous frame is repeatedly used, or the bandwidth is expanded to flatten the spectral characteristics.

  Next, operations of speech encoding apparatus 1104 and speech decoding apparatus 1114 will be described with reference to FIGS. FIG. 23 is a flowchart for explaining the operation of speech encoding apparatus 1104, and FIG. 24 is a flowchart for explaining the operation of speech decoding apparatus 1114.

  As shown in FIG. 23, speech encoding apparatus 1104 first performs CELP speech encoding processing for one frame by speech encoding section 153 in step ST901, and then obtains in step ST901 in step ST902. The encoded speech information is stored by the 1-frame delay unit 154 until the processing of the next frame is completed.

  Next, in step ST903, speech coding apparatus 1104 uses pitch pitch position detection section 151 to detect the pitch peak position that is the last in the current frame.

  Next, in step ST904, the speech encoding apparatus 1104 extracts the speech encoding information of the previous frame (speech encoding information of the previous frame) saved in step ST902 from the 1-frame delay unit 154, and then performs step ST905. Then, the pitch peak position information detected in step ST903 is embedded in the extracted speech encoding information of the previous frame and output as the output of the speech encoding device 1104. Speech encoding apparatus 1104 repeats the series of encoding processes in steps ST901 to ST905 described above.

  Next, the operation of the speech decoding apparatus 1114 will be described. As shown in FIG. 24, the speech decoding apparatus 114 first determines in step ST910 whether or not the current frame has been lost by the signal processing apparatus 113. If it has not disappeared, the process proceeds to step ST911, and if it has disappeared, the process proceeds to step ST914.

  Next, when the frame is not lost, in step ST911, the pitch peak position information is separated from the encoded information received by the pitch peak position information separating section 1155.

  Subsequently, in step ST912, the parameter decoding unit 301 in the speech decoding unit 156 decodes speech encoding parameters from the separated speech encoding information. Next, in step ST913, the remaining part of speech decoding section 156 performs a process of re-synthesizing the speech signal from the decoded speech encoding parameter, and the decoded speech signal is output to D / A converter 115. .

  On the other hand, if it is determined in step ST910 that the frame is lost, the speech decoding apparatus 1114 moves to step ST914 and the pitch period calculator 314 determines that the current frame (erasure frame) is the pitch period. It is checked whether or not the frame has sex. Information regarding whether or not there is pitch periodicity is included in the received pitch peak position information. If there is pitch periodicity, speech decoding apparatus 1114 moves to step ST915, and if there is no pitch periodicity, it moves to step ST918. If the current frame (erased frame) has pitch periodicity, the speech decoding apparatus 1114 moves to step ST915, and the pitch peak position detector 313 uses the pitch information of the previous frame and the decoded excitation signal. Read from the parameter buffer 312 and the adaptive codebook 304, respectively.

  Next, the speech decoding apparatus 1114 moves to step ST916, and the pitch peak position detector 313 detects the pitch peak position that is the rearmost in the previous frame. When the parameter buffer 312 holds the pitch peak position information received two times before, the pitch peak position in the previous frame may be obtained using the information.

  Next, the speech decoding apparatus 1114 moves to step ST917, and the pitch period calculator 314 uses the pitch peak position of the current frame (erased frame), the pitch peak position of the previous frame, and the pitch period of the previous frame, The pitch period is calculated by the above (Formula 5).

  Next, the speech decoding apparatus 1114 moves to step ST918, and the parameter generating unit 315 generates various speech encoding parameters in the lost frame.

  If it is determined that the current frame is a frame having pitch periodicity, the fixed codebook gain is set to zero, and a speech signal is generated using only the adaptive codebook. The pitch period (adaptive codebook parameter) used at this time is the one calculated by the pitch period calculator 314. If it is determined that the current frame is a frame having no pitch periodicity, speech decoding apparatus 1114 sets the adaptive codebook gain to zero and generates a speech signal using only the fixed codebook. In this case, fixed codebook parameters are determined randomly. As the linear prediction coefficient, regardless of whether or not the current frame has a pitch periodicity, the previous frame is repeatedly used or gradually whitened while performing bandwidth expansion.

  Finally, in step ST919, the speech decoding apparatus 1114 updates the contents of the parameter buffer 312 and ends the decoding process for one frame.

  The speech decoding apparatus 1114 repeats a series of decoding processes in steps ST910 to ST919.

  As described above, according to the present embodiment, it is possible to perform accurate frame erasure compensation processing while suppressing quality degradation without increasing the bit rate, and to reduce the influence of error propagation after the erasure frame. Can do.

100, 400, 700, 1000 Audio signal transmitter 199, 499, 799, 1199 Audio signal receiver 102 Input device 103 A / D converter 104, 404, 1104 Audio encoder 105, 113 Signal processor 106 RF modulator Reference Signs List 107 transmitter 108, 110 antenna 111 receiver 112 RF demodulator 114, 414, 714, 1114 speech decoder 115 D / A converter 116 output device 151 pitch peak position detector 152, 454 multiplexer 153 speech encoding 154, 157, 601 1 frame delay unit 155, 455, 755 demultiplexing unit 156, 456, 756 speech decoding unit 158, 457, 757 frame erasure compensation unit 202 linear prediction analyzer 203 LPC quantizer 206 LPC synthesis filter 208, 304 Adaptive codebook 210, 305 Fixed codebook 212 Gain quantizer 301 Parameter decoder 302 Gain decoder 306 LPC decoder 303 Switch 313 Pitch peak position detector 314, 814 Pitch period calculator 312, 812 Parameter buffer 315 , 815 Parameter generator 603 Adaptive codebook modifier 1152 Pitch peak position information embedding unit

Claims (5)

An adaptive codebook for storing the first excitation signal;
Processing means for adding a plurality of samples of the first sound source signal to generate a second sound source signal;
Between the end 1 pitch period of the second sound source signal, a particular pulse position detecting means absolute value of the amplitude is detected as a specific pulse samples which maximizes obtains the position of said specific pulses,
Encoding means for encoding the position of the specific pulse to generate specific pulse position information;
A CELP (Code Excited Linear Prediction) type speech coding apparatus. Decoding means for decoding the specific pulse position information encoded and transmitted at the communication partner to obtain the first specific pulse position;
An adaptive codebook for storing the first excitation signal;
Processing means for adding a plurality of samples of the first sound source signal to generate a second sound source signal;
Between the end 1 pitch period of the second sound source signal, detects a sample absolute value of the amplitude is maximum as a particular pulse, particular pulse position detecting means for determining the position of said specific pulses as a second specific pulse position When,
Frame erasure compensation means for performing processing for applying the second specific pulse position to the first specific pulse position on the second sound source signal in the lost frame;
CELP (Code Excited Linear Prediction) type speech decoding apparatus. The first specific pulse position is obtained by decoding the specific pulse position information included in reception information of a frame next to the lost frame.
The CELP speech decoding apparatus according to claim 2 . A processing step of adding a plurality of samples of the first excitation signal stored in the adaptive codebook to generate a second excitation signal;
Between the end 1 pitch period of the second sound source signal, a particular pulse position detecting step of absolute value of the amplitude is detected as a specific pulse samples which maximizes obtains the position of said specific pulses,
An encoding step of encoding the position of the specific pulse to generate specific pulse position information;
CELP (Code Excited Linear Prediction) type speech coding method. A decoding step of decoding the specific pulse position information encoded and transmitted at the communication partner to obtain a first specific pulse position;
A processing step of adding a plurality of samples of the first excitation signal stored in the adaptive codebook to generate a second excitation signal;
Between the end 1 pitch period of the second sound source signal, detects a sample absolute value of the amplitude is maximum as a particular pulse, particular pulse position detection step for determining the position of said specific pulses as a second specific pulse position When,
In a lost frame, a frame loss compensation step for performing processing for adjusting the second specific pulse position to the first specific pulse position on the second sound source signal;
CELP (Code Excited Linear Prediction) type speech decoding method.

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