WO2005106848A1 - スケーラブル復号化装置および拡張レイヤ消失隠蔽方法 - Google Patents
スケーラブル復号化装置および拡張レイヤ消失隠蔽方法 Download PDFInfo
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
- G10L19/16—Vocoder architecture
- G10L19/18—Vocoders using multiple modes
- G10L19/24—Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/038—Speech enhancement, e.g. noise reduction or echo cancellation using band spreading techniques
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/005—Correction of errors induced by the transmission channel, if related to the coding algorithm
Definitions
- the present invention relates to a scalable decoding apparatus that performs concealment processing when an enhancement layer is lost, and an enhancement layer erasure concealment method used in the apparatus.
- packet loss may occur on the transmission path, so even if part of the transmission information is lost, decoding processing can be performed from the remaining information.
- a scalable coding feature is desired.
- this scalable code there is no change in the frequency band, but only the bit rate of the signal to be coded has scalability, and in the frequency band of the signal to be coded (in the frequency axis direction) There is one that gives a certainty and makes a code (see, for example, Non-Patent Document 1).
- the latter scheme of providing scalability in the frequency band and coding will be called band-scale scalable coding.
- Such scalable code systems have been disclosed not only for audio signals but also for audio signals in a wider band (see, for example, Patent Documents 1 and 2).
- Such scalable coding is to hierarchically code I spoon acoustic signal comprising the code I spoon subject, DiffS e rv (Differentiated Services) Core (basic example using priority control on the network, such as the ⁇ ) information is transmitted preferentially. Then, depending on the status of the transmission path, discarding is performed in order of the information strength of higher enhancement layers. As a result, the probability that core information will be discarded in the communication network can be reduced, and degradation of call quality can be suppressed even if some code information is lost due to packet loss.
- DiffS e rv Differentiated Services
- Core basic example using priority control on the network, such as the ⁇
- Patent Document 3 discloses the frame erasure concealment process of ITU-T Recommendation G. 729. As disclosed in Patent Document 3, it is standard to perform concealment processing of a lost frame by extrapolation using information decoded in the past.
- Patent Document 1 Japanese Patent Application Laid-Open No. 08-263096
- Patent Document 2 Japanese Patent Application Laid-Open No. 2002-100994
- Patent Document 3 Japanese Patent Application Laid-Open No. 09-120297
- Non-Patent Document 1 T. Nomura et al, "A Bitrate and Bandwidth Scalable CELP Coder,” IEEE Proc. ICASSP 98, pp. 341-344, 1998
- Non-Patent Document 2 3GPP Standard, TS 26. 190
- decoding processing of the lost signal may be performed using the information of the core layer, but there are the following problems. That is, as described above, when not only the bit rate but also the frequency band is scalable, the decoded signal generated from the information of the core layer is a narrow band signal, whereas the information strength of both the core layer and the enhancement layer is The generated decoded signal is a wideband signal. Therefore, there is a problem that the frequency band of the decoded signal changes between the case where the decoding process is performed using only the information of the core layer and the case where the decoding process is performed using even the enhancement layer.
- the object of the present invention is to prevent the discomfort in the subjective quality without causing frequent switching of the band of the decoded signal even when the signal of the enhancement layer is lost in the band scalable coding system.
- Abstract A scalable decoding device and an enhancement layer erasure concealment method used in the device.
- the scalable decoding device is a scalable decoding device for obtaining a wideband decoded signal from code information including a core layer having scalability in the frequency axis direction and an enhancement layer, Core layer decoding means for obtaining a narrow band core layer decoding signal, core layer decoding means, a conversion means for converting the frequency band of the narrow band core layer decoding signal to a wide band and obtaining a first signal, and a core layer Compensation means for generating a wide band compensation signal based on the decoded signal obtained in the past with respect to the coding information in which the enhancement layer is lost, and a frequency component corresponding to the wide band compensation signal power core layer And adding the first signal obtained by the converting means and the second signal obtained by the removing means to obtain a second signal.
- a configuration having a, and adding means for obtaining.
- the frequency of the decoded signal may not be switched frequently so that the subjective quality may not cause discomfort.
- FIG. 1 is a block diagram showing a main configuration of a scalable decoding device according to Embodiment 1.
- FIG. 2 a block diagram showing a main configuration inside a core decoder according to Embodiment 1.
- FIG. 3 A block diagram showing the main configuration inside the extended decoder according to Embodiment 1.
- FIG. 4 A diagram showing the flow of signals in the normal state inside the extended decoder according to Embodiment 1.
- FIG. 5 Signal when the frame of the enhancement layer inside the extended decoder according to Embodiment 1 is lost Figure showing the flow of
- FIG. 6 A diagram for explaining the outline of the decoding process of the scalable decoding device according to the first embodiment.
- FIG. 7 A block diagram showing the configuration of the up-sample processing unit when the extension decoder according to Embodiment 1 is MDCT-based
- FIG. 8 A block diagram showing the main configuration of a scalable decoding device according to Embodiment 2.
- FIG. 9 When the scalable decoding device shown in Embodiment 1 or 2 is applied to a mobile communication system 10 is a block diagram showing the main configuration of the mobile station apparatus and base station apparatus of the present embodiment
- Figure 10 A block diagram showing the main configuration of the scalable decoding apparatus when combining Embodiments 1 and 2
- the core layer is a layer that performs coding and decoding of the narrowband signal.
- FIG. 1 is a block diagram showing a main configuration of a scalable decoding device according to Embodiment 1 of the present invention.
- the scalable decoding apparatus includes a packet analysis unit 101 for a core code packet, a core decoder (core decoding process unit) 102, an up-sample processing unit 103, and an extension code packet. , An extended decoder (extended decoding processing unit) 105, a high pass filter (HPF) 106, a switching switch (SW) 107, and an adder 108.
- HPF high pass filter
- SW switching switch
- Each unit of the scalable decoding device performs the following operation.
- Packet disassembling section 101 for the core code packet extracts core layer coding information from the core code packet on which the code information of the corer array input via packet network N is placed, and performs core decoding. While outputting (S 1) to the unit 102, the frame erasure information C 1 is output to the core decoder 102, the extension decoder 105, and the switching switch 107.
- the coding information refers to a coded bit stream output from a coding device (not shown) on the transmission side.
- frame loss information CI is information indicating whether a frame to be decoded is a lost frame. When the packet to be decoded is a lost packet, all frames included in this packet become a lost frame.
- the core decoder 102 performs core layer decoding processing using the frame loss information C1 and the coding information S1 output from the packet disassembly unit 101, and outputs a core layer decoded signal (narrowband signal) S3. Do.
- the specific contents of the core layer decoding process may be, for example, a decoding process based on a CELP model, or may be a decoding process based on waveform coding, or a transform code model using MDCT or the like. It may be decryption processing of Also, the core decoder 102 outputs part or all (S4) of the information obtained in the core layer decoding process to the extended decoder 105.
- the information output to the enhancement decoder 105 is used for the enhancement layer decoding process.
- the core decoder 102 outputs the signal S6 obtained in the core layer decoding process to the up-sample processing unit 103.
- the signal S6 output to the up-sample processing unit 103 may be the decoded signal of the core layer itself or, depending on the code model of the core layer, a partial decoding parameter (for example, spectrum parameter or excitation parameter). Also good.
- the up-sample processing unit 103 performs a process of increasing the Nyquist frequency on the decoded signal or a part of the decoding parameter or the decoded signal obtained in the decoding process output from the core decoder 102.
- the up-sampled signal S7 is output to the extension decoder 105. Note that this upsampling process is not limited to the process on the time axis, and depending on the scalable coding algorithm, the signal after the upsampling process is output to the extended excitation decoder 122 and used during the extended excitation decoding. It is good also as composition.
- packet decomposing section 104 for the extension code packet extracts the coding information of the enhancement layer from the extension code packet on which the coding information of the enhancement layer inputted through the packet network is carried.
- the frame loss information C 2 is output to the extension decoder 105 and the switching switch 107 as well as being output to the extension decoder 105 (S 2).
- Extended decoder 105 performs frame loss information C 2 and code information S 2 output from packet disassembly unit 104, and a core layer decoded signal S 3 output from core decoder 102 and core layer encoding processing.
- Information S4 obtained in the process and from the up-sample processing unit 103 Decoding processing of the enhancement layer is performed using the signal S7 obtained by up-sampling the decoded signal of the core layer to be output, to obtain a decoding signal (wide band signal) of the enhancement layer, and output to the HPF 106 and the adder 108 (S8, S9).
- the signal S8 output to the adder 108 and the signal S9 output to the HPF 106 may not be identical.
- the extended decoder 105 may output the signal S7 output from the up sample processing unit 103 as it is to the adder 108 or may switch conditionally with reference to the frame erasure information C2.
- the HPF 106 passes only the high frequency component (a band component not included in the narrow band decoded signal of the core layer) of the decoded signal S9 input from the extended decoder 105, and outputs it to the switching switch 107.
- the switching switch (SW) 107 turns on / off the output of the signal output from the HPF 106 to the adder 108.
- the on / off of the switch is performed by referring to the frame loss information outputted from the packet disassembling unit 101 for the core code packet and the packet disassembling unit 104 for the extension code packet. Specifically, if both the core layer and the enhancement layer are frame lost and there is a problem (a normal frame), the switch is opened and taken as an option. Also, if only the core layer is a normal frame and the enhancement layer is a lost frame, close the switch and turn it on. Furthermore, if both the core layer and the enhancement layer are lost frames, open the switch and turn it off.
- the adder 108 adds the full-band acoustic signal directly input from the extension decoder 105 and the high-band decoded signal input from the extension decoder 105 via the HPF 106, and sets the addition result as a wideband signal. Output.
- FIG. 2 is a block diagram showing a main configuration inside the above core decoder 102. As shown in FIG.
- the core decoder 102 includes a parameter decoding unit 111, a core linear prediction coefficient (LPC) decoder 112, a core excitation decoder 113, and a synthesis filter 114.
- LPC core linear prediction coefficient
- the noramator decoding unit 111 receives code information (bit stream) S 1 of the core layer code output from the packet disassembly unit 101, LPC parameter code data (including LSP code, etc.) and a sound source.
- the data is separated into parameter code / data (including pitch lag code, fixed excitation codebook code, gain code, etc.), and each data is decoded into various codes, and core (layer) LPC decoder 112 and core sound source It outputs to the decoder 113 respectively.
- the core LPC decoder 112 decodes the LPC parameter code output from the parameter decoding unit 111, and outputs the decoded LPC to the synthesis filter 114 and the extension decoder 105.
- the specific content of the decoding process is that, for example, vector quantization is used to decode coded LSP parameters and also convert powers into LPC parameters. If the frame loss information C1 output from the bucket disassembling unit 101 for the core code packet indicates that the current frame is a lost frame, the core LPC decoder 112 performs LPC compensation using frame loss compensation processing. It performs concealment processing of the parameter, and outputs LPC (compensation signal) generated by the concealment processing as decoded LPC.
- LPC compensation signal
- the core excitation decoder 113 performs decoding processing on various kinds of excitation parameter codes (pitch lag, fixed codebook, gain codebook and other codes) output from the parameter decoding unit 111, and decodes the excitation signal. Are output to the synthesis filter 114 and the up sample processing unit 103 (S6). Also, the core sound source decoder 113 outputs a part or all of the information S3 decoded by this decoding process to the extended decoder 105. Specifically, the pitch lag information and the pulse drive signal (fixed codebook excitation information) are output from the core excitation decoder 113 to the expansion decoder 105.
- excitation parameter codes pitch lag, fixed codebook, gain codebook and other codes
- the core sound source decoder 113 uses the frame loss compensation processing to generate the sound source. Parameter concealment processing is performed, and the compensated excitation signal generated by the concealment processing is output as a decoded excitation signal.
- the synthesis filter 114 drives the linear prediction filter composed of the decoded LPC output from the core LPC decoder 112 with the decoded excitation signal output from the core excitation decoder 113 to obtain the narrowband signal S5. Output.
- FIG. 3 is a block diagram showing the main configuration inside the extension decoder 105.
- This extended decoder 105 includes a parameter decoding unit 121, an extended excitation decoder 122, two switching switches (123 and 126), two synthesis filters (124 and 128), an LPC conversion unit 125, and An extended LPC decoder 127 is provided.
- the noramator decoding unit 121 receives the code information S2 of the enhancement layer from the packet disassembly unit 104, and LPC parameter code data (including LSP code etc.) and sound source parameter coded data (pitch lag) Code, fixed codebook index code, including gain code etc) , And decode to codes of various parameters, and output to the extended LPC decoder 127 and the extended excitation decoder 122, respectively.
- LPC parameter code data including LSP code etc.
- sound source parameter coded data pitch lag
- fixed codebook index code including gain code etc
- the extended LPC decoder 127 includes the decoded core LPC parameter S4 input from the core LPC decoder 112 in the core decoder 102, and the enhancement layer LPC nomometer code input from the norrometer decoding unit 111.
- the LPC parameters for use in recombining the wideband signal are decoded and output to two combining filters (output to the combining filter 124 via the switching switch 126).
- a model is used to predict an extended LSP (wideband LSP) from a decoded 1 ⁇ ? (Narrowband and SP) input from the core LPC decoder 112.
- the extended LPC decoder 127 decodes the prediction error of the wideband LSP predicted from the narrowband LSP (for example, it is encoded using MA prediction vector quantization etc.), and A series of processing is performed such as adding to the wideband LSP predicted from the narrowband LSP to decode the final wideband LSP and finally converting it to an LPC.
- the extended LPC decoder 127 uses the frame loss compensation processing.
- the concealment process of the LPC parameters is performed, and the compensated LPC generated by the concealment process is output as a decoded LPC.
- the decryption process may be another method
- the LPC conversion unit 125 converts the narrowband LPC parameter S4 into a wideband LPC parameter.
- the impulse response of the LPC synthesis filter which can obtain narrow band LSP power is up-sampled, the self-correlation is obtained from the up-sampled impulse response, and the obtained autocorrelation coefficient is made into LSP of desired order.
- the method of conversion etc. is not limited to this.
- the transformation between the autocorrelation coefficient R and the LPC parameter a can be realized by using the relation of (Equation 1) below.
- the converted LPC parameters are output to the synthesis filter 124 via the changeover switch 126. Although not shown, when using a code model that decodes the extended LPC using the converted LPC parameters, the converted LPC is also output to the extended LPC decoder 127. To do.
- the extended sound source decoder 122 receives various code information of the extended sound source parameters from the parameter decoding unit 121, and the core sound source decoder 113 decodes the core sound source parameter decoding information, the core sound source such as the decoded core sound source signal Information obtained by the decoding process is input.
- the extended sound source decoder 122 decodes the extended sound source (wide band sound source) signal and outputs the decoded signal to the synthesis filter 124 and the synthesis filter 128 (however, the output to the synthesis filter 124 is via the switch 123). To be done).
- this processing includes pitch lag decoding processing, adaptive codebook component decoding processing, fixed codebook component decoding processing, and gain parameters. Decoding processing etc. are included.
- the pitch lag decoding process is performed, for example, as follows. Since the pitch lag for the expanded sound source is differentially quantized based on the pitch lag information input from the core sound source decoder 113, the expanded sound source decoder 122 is a core sound source if it is an expansion that doubles the sampling frequency. The pitch lag for the core sound source is converted to the pitch lag for the expanded sound source by doubling the pitch lag for the sound source, while the differentially quantized pitch lag (delta lag) is decoded. Then, the extended sound source decoder 122 sets the sum of the pitch lag converted for the extended sound source and the delta lag obtained by the decoding as the decoded pitch lag for the extended sound source.
- the adaptive excitation codebook for the extended excitation decoder 122 that is, the buffer of the excitation signal generated from the extended excitation decoder 122 in the past, is used Generate adaptive codebook components and decode them.
- the one after the sampling rate conversion of the fixed codebook inputted from the core sound source decoder 113 is used as the expanded sound source decoder 122 for the fixed codebook in the expanded sound source decoding process.
- the extended sound source decoder 122 additionally has a fixed codebook in the extended sound source codebook, and decodes additional fixed codebook components by performing decoding processing. Decoded adaptive codebook component and fixed codebook component Each of these is multiplied by the decoded gain parameter and added up to obtain a decoded sound source signal.
- the extended sound source decoder 122 uses the frame loss compensation processing to generate the sound source.
- the parameter concealment process is performed, and the compensated excitation signal generated by the concealment process is output as a decoded excitation signal.
- the switching switch 123 is a switching switch that connects either the upsample processing unit 103 or the extended sound source decoder 122 and the synthesis filter 124, and the frame loss information input from the core code packet packet disassembly unit 101 It is switched based on C 1 and frame erasure information C 2 input from the extended code packet packet disassembly unit 104. Specifically, when the core layer is a normal frame and the enhancement layer is a lost frame, the input terminal of the synthesis filter 124 is connected to the output terminal of the up-sampling processing unit 103, and in the other cases, the input of the synthesis filter 124 The terminal is connected to the output terminal of the enhanced sound source decoder 122.
- the switching switch 126 is a switching switch that connects either one of the LPC converter 125 or the extended LPC decoder 127 to the second input terminal of the synthesis filter 124, and is input from the core code packet depacketizer 101. Switching is performed based on the frame loss information C1 to be transmitted and the frame loss information C2 input from the extension code packet depacketizing unit 104. Specifically, when the core layer is a normal frame and the enhancement layer is a lost frame, the second input terminal of the synthesis filter 124 is connected to the output terminal of the LPC conversion unit 125, and in the other cases, the synthesis is performed. The second input terminal of the filter 124 is connected to the output terminal of the enhanced LPC decoder 127.
- the synthesis filter 124 receives filter coefficients from the extended LPC decoder 127 or the LPC conversion unit 125 through the switch 126, and the synthesis filter is configured using these filter coefficients.
- the composed synthesis filter is driven by the sound source signal input from the enhanced sound source decoder 122 or the up-sample processing unit 103 via the switch 123, and the output signal S8 is output to the adder. Note that as long as the core layer frame is not lost, the synthesis filter 124 continues to generate an error free signal.
- the synthesis filter 128 forms a synthesis filter with the filter coefficients input from the extended LPC decoder 127, is driven by the decoded excitation signal input from the extended excitation decoder 122, and high-passes the output signal S9. Output to filter 106.
- the synthesis filter 128 always generates a wide band decoded signal regardless of the presence or absence of frame loss.
- the HPF 106 is a filter that cuts off the band of the decoded signal of the core decoder 102.
- the HPF 106 receives the output signal of the synthesis filter 128, passes only the high band component (the band expanded in the enhancement layer), and switches. Output to 107.
- the high pass filter desirably has linear phase characteristics, but is not limited thereto.
- the switching switch 107 is a switch that turns ON / OFF the signal input to the adder, and the frame loss information input from the core code packet depacketizing unit and the extension code packet demultiplexing unit power are input. Frame loss information, and can be switched based on. Specifically, if the core layer is a normal frame and the enhancement layer is a lost frame, the switch is closed and the output of the HPF 106 is input to the adder. Otherwise, the changeover switch 107 is open and the output of the HPF 106 is not input to the adder.
- Adder 108 adds the decoded signal output from synthesis filter 124 and the decoded signal having only the high frequency component input to switching switch 107, and outputs the result as a final wideband decoded signal. Do.
- the low-frequency component of the signal is important for human auditory sense, and the low-frequency component is error-free because the quality of the low-frequency component (pitch period) is largely degraded in Code LP Z decoding of CELP system. It is possible to reduce the degradation of subjective quality even if errors are mixed in the high frequency component if
- the core layer is a bit rate scalable decoder
- the core code packet can be divided into the number of layers in the bit rate scalable configuration.
- the core code packet disassembly unit is also prepared according to the number of layers.
- bit rate scalable of the core decoder 102 is the bit rate scalable of the core decoder 102. It shall be obtained only by the core decoding process. Also, if only part of the enhancement layer of the bit rate scalable enhancement layer other than the bit rate scalable core is lost, a part of the information of the bit rate scalable core and the bit rate scalable enhancement layer that has been successfully received The core decoder may be decoded using this method.
- FIG. 4 and FIG. 5 are diagrams in which the flow of signals inside the above-described extended decoder 105 is organized.
- Fig. 4 is a diagram showing the flow of signals when there is no frame loss, ie, normal
- Fig. 5 is a diagram showing the flow of signals when frames of the enhancement layer are lost.
- the NB signal indicates a narrow band signal
- the WB signal indicates a wide band signal.
- a signal S 101 indicated by a broken line indicates a signal when the frame loss has not occurred. However, if high band (extension layer) packets of this signal are lost on the transmission path, only low band packets are actually received. Therefore, in the present embodiment, upsampling processing or the like is performed on the low band packet signal to generate a signal S 102 (solid line signal) in which the sampling rate is wide and only the low band component remains. On the other hand, based on the signal S 103 of the (n ⁇ 1) th frame, the concealment processing is performed to generate the compensation signal S 104. By passing this signal S104 through the HPF, if only the high frequency component is extracted, it becomes a signal S105. In the adding section 108, only the low frequency component remains, and the high frequency component remains with only the high frequency component S 101, and the decoded signal S 106 is obtained by adding the low frequency component S 105.
- the error-free (error-free) received normally The signal obtained by using the core layer code information which is the low-pass component is upsampled to generate a signal, and this signal is a signal of the entire band generated by using the error concealment processing in the enhancement layer.
- the signals obtained by extracting only the high frequency components are added to obtain a full band decoded signal.
- the enhancement layer can support only the sound signal band supported by the core layer. Acoustic signal bands can always be generated.
- the sampling rate does not change as it is the wideband decoded signal, but the bandwidth of the output signal of the synthesis filter is narrow depending on the error condition of the extended filter. It spreads. That is, when the frame of the enhancement layer is lost, the bandwidth of the decoded signal is narrowed.
- the quality of the low frequency component does not deteriorate.
- the bandwidth of the decoded signal is lost at the decoder side. May change and you may feel uncomfortable with your hearing.
- the bandwidth of the decoded signal changes temporally by adding the high-frequency component of the decoded signal of the enhancement layer decoded using frame erasure concealment processing to the decoded signal of the core layer decoded in the error free state. This makes it possible to obtain an aurally stable quality on the decoder side.
- the configuration is such that adaptive decoding is performed on the enhancement layer code ⁇ Z decoding and frame loss concealment processing using the core layer decoding information, even if the enhancement layer information is lost, the core layer If the information of the above is correctly received, it is possible to obtain a high quality decoded signal.
- priority control in the packet network can be effectively used to realize high quality acoustic communication quality.
- the number of enhancement layers is one
- the number of enhancement layers may be two or more (two or more types of frequency bands may be output).
- the hierarchical structure in which the core layer further has bit rate scalability LE coder z scalable decoder even better.
- the algorithm of code ⁇ Z decoding ⁇ ⁇ that outputs each frequency band may have a hierarchical structure with bit rate scalability.
- the extension decoder 105 may be MDCT based.
- FIG. 7 is a block diagram showing the configuration of the up-sample processing unit 103a when the extension decoder 105 is based on MDCT.
- the up-sample processing unit 103 a includes an MDCT unit 131 and an order extension unit 132.
- Core decoder 102 outputs the core decoded signal as a narrow band decoded signal and also to MDCT section 131. This corresponds to the case where the two output signals (S3, S4) of the core decoder 102 shown in FIG. 1 are identical. Also, part or all of the information obtained in the core layer decoding process is output to the enhancement decoder 105.
- MDCT section 131 performs a modified discrete cosine transform (MDCT) process on the narrow band decoded signal output from core decoder 102, and outputs the obtained MDCT coefficients to order expanding section 132.
- MDCT modified discrete cosine transform
- the order extension unit 132 extends the order of the MDCT coefficients output from the MDCT unit 131 by zero padding (however, if upsampling is performed twice, the MDCT order is doubled and the increased part is Fill in with a factor of 0).
- the expanded MDCT coefficients are output to the expanded decoder 105.
- the extension decoder 105 generates a decoded signal of the enhancement layer by subjecting the MDCT coefficients output from the degree extension unit 132 to inverse transformation discrete cosine transform. In addition, when performing the concealment processing, the extension decoder 105 adds the extension information generated by the concealment processing to the MDCT coefficients output by the degree extension unit 132, and reversely transforms the MDCT coefficients generated by this. A cosine transform is performed to generate a decoded signal of the enhancement layer.
- FIG. 8 is a block diagram showing a main configuration of a scalable decoding device according to Embodiment 2 of the present invention. Note that this scalable decoding device has the same basic configuration as the scalable decoding device shown in the first embodiment, and the same components are identical. The symbol is attached and the description is omitted.
- the scalable decoding device includes mode judging section 201, and core decoder 102 and extended decoder 105 having an input / output interface to mode judging section 201 are the embodiments. Different from 1.
- the core decoder 102 performs core layer decoding processing using the frame loss information C1 and the coding information S1 input from the packet disassembly unit 101, and generates a core layer decoded signal (narrowband signal) S6. Output. Also, it outputs a part or all of the information obtained in the core layer decoding process to the enhancement decoder 105. The information output to the enhancement decoder 105 is used for the enhancement layer decoding process. Furthermore, the signal obtained in the core layer decoding process is output to the up-sample processing unit 103 and the mode determination unit. The signal output to the up-sample processing unit 103 may be the core layer decoded signal itself, or may be a partial decoding parameter depending on the core layer code model.
- the information output to the mode determination unit is linear prediction coefficient, pitch prediction gain, pitch lag, pitch period, signal energy, zero crossing rate, reflection coefficient, logarithmic cross section ratio, LSP parameter, normalized linear prediction residual error Etc. These parameters are generally used to classify the state of speech signal (silence, voiced steady part, noisy consonant part, rising part, transient part etc.).
- Mode determination unit 201 classifies the signal being decoded using various types of information input from core decoder 102 (eg, noise consonant part, voiced steady part, rising part, voiced transient part, silent part) , Music signal etc.), and output the classification result to the expansion decoder 105.
- information input from core decoder 102 eg, noise consonant part, voiced steady part, rising part, voiced transient part, silent part
- Music signal etc. e.g, Music signal etc.
- the extended decoder 105 is configured to receive the frame erasure information and the code information output from the packet decomposing unit 104, the information obtained in the code processing process for the core layer output from the core decoder 102, and The enhancement layer is decoded using the up-sampled core layer decoded signal input from the sample processing unit 103.
- the extension layer coder process is performed by an extension encoder (not shown) that selectively uses a code model suitable for the mode using mode information input from the mode determination unit. If it is being performed, the decoding process also performs the same process. As described above, if the configuration of the current acoustic signal is determined in the core layer and the coding model of the enhancement layer is adaptively switched, higher quality coding Z decoding can be realized.
- the decoded signal is output to the HPF 106 and the adder 108 as a decoded signal (wideband signal) of the enhancement layer.
- the signal output to the adder 108 and the signal output to the HPF 106 may not be the same.
- the signal input from the up-sample processing unit 103 may be output to the adder 108 as it is.
- the signal to be output to the adder 108 is conditionally switched by referring to the frame erasure information (for example, generated by the signal input from the up-sample processing unit 103 and the decoding process performed in the extension decoder 105 The signal may be switched.
- the extended decoder 105 performs frame erasure concealment processing.
- the concealment process suitable for the mode is performed.
- the wideband signal generated using the concealment process is output to the adder via the HPF 106 and the switch.
- the HPF 106 transforms to the frequency domain using orthogonal transformation such as the force MDCT that can be realized with digital filters in the time domain, and uses processing that returns only the high frequency component and returns to the time domain by inverse transformation. It is good.
- the core LPC decoder 112 is an acoustic parameter obtained in the LPC decoding process or an acoustic parameter that can obtain the decoded LPC force (eg, reflection coefficient, logarithmic cross section ratio, LSP, normal-linear-line prediction residual)
- the pattern is output to the mode determination unit.
- the core sound source decoder 113 is an acoustic parameter obtained in the sound source decoding process or an acoustic parameter obtained by decoding the sound source signal (for example, pitch lag, pitch period, pitch gain, pitch prediction gain, sound source signal energy, sound source Output the signal zero crossing rate etc. to the mode determination unit 201.
- an analysis unit for analyzing the zero crossing rate and energy information of the narrowband decoded signal output from the synthesis filter may be provided to input these parameters to the mode determination unit. And better then,.
- Mode determination section 201 includes core LPC decoder 112 and core sound source decoder 113 and other acoustic parameters (LSP, LPC, reflection coefficient, logarithmic cross section ratio, normalized linear prediction residual error, Pitch lag, pitch period, pitch gain, pitch prediction gain, sound source signal energy, sound source signal zero crossing rate, combined signal energy, combined signal zero crossing rate, etc. are input, and sound signal mode classification (silence part, noise characteristic) A consonant part, a voiced steady part, a rising part, a voiced transient part, an end, a music signal, etc.), and the classification result is outputted to the extended LPC decoder 127 and the extended sound source decoder 122, respectively.
- the expansion decoder 105 includes a post-processing unit such as a post filter
- the mode classification information may be output to the post-processing unit.
- the extended LPC decoder 127 may switch the decoding process according to the various modes of the acoustic signal input from the mode determination unit 201. In this case, it is assumed that the same switching process of the code model is performed even if the extended LPC encoder (not shown) has V. Also, when frame loss occurs in the enhancement layer, frame loss concealment processing corresponding to the above mode is performed to generate a decoded extension LPC.
- the extended sound source decoder 122 may switch the decoding process according to the various modes of the acoustic signal input from the mode determination unit 201. In this case, it is assumed that the same code model is switched even in the extended excitation encoder (not shown). If frame loss occurs in the extension layer, frame loss concealment processing corresponding to the above mode is performed to generate a decoded extension excitation signal.
- FIG. 9 is a block diagram showing the main configuration of a mobile station apparatus and a base station apparatus when the scalable decoding device described in Embodiment 1 or 2 is applied to a mobile communication system.
- This mobile communication system includes an audio signal transmitting device 300 and an audio signal receiving device 31.
- the scalable decoding apparatus described in the first or second embodiment is mounted on the voice signal receiving apparatus 310.
- Audio signal transmitting apparatus 300 includes an input device 301, an A / D converter 302, and a speech encoding device.
- a signal processor 304 an RF modulator 305, a transmitter 306, and an antenna 307 are provided.
- the input terminal of the AZD conversion device 302 is connected to the output terminal of the input device 301.
- the input terminal of the speech coding device 303 is connected to the output terminal of the AZD conversion device 302.
- the input terminal of the signal processing unit 304 is connected to the output terminal of the speech coding unit 303.
- the input terminal of the RF modulator 305 is connected to the output terminal of the signal processor 304.
- the input terminal of the transmitter 306 is connected to the output terminal of the RF modulator 305.
- the antenna 307 is connected to the output terminal of the transmitter 306.
- the input device 301 receives an audio signal, converts it into an analog audio signal which is an electric signal, and supplies the analog audio signal to the AZD conversion device 302.
- the AZD converter 302 converts the analog voice signal from the input device 301 into a digital voice signal, and supplies this to the voice coding device 303.
- the speech coding unit 303 codes the digital speech signal from the AZD conversion unit 302 to generate a speech code and a bit string, which are supplied to the signal processing unit 304.
- the signal processing device 304 performs channel code processing, packet processing and transmission buffer processing on the voice code and bit string from the voice coder 303, and then RF modulates the voice code and bit string. Supply to the device 305.
- the RF modulation unit 305 modulates the signal of the voice code / bit string subjected to channel code processing and the like from the signal processing unit 304 and supplies the modulated signal to the transmission unit 306.
- the transmitter 306 transmits the modulated voice code signal from the RF modulator 305 as a radio wave (RF signal) via the antenna 307.
- audio signal transmitting apparatus 300 processing is performed on a digital audio signal obtained via AZD conversion apparatus 302 in frame units of several tens of ms . If the network constituting the system is a packet network, code data of one frame or several frames are put into one packet and this packet is sent out to the packet network. If the above network is a circuit switching network, packetization processing and transmission buffer processing are unnecessary.
- the voice signal reception device 310 includes an antenna 311, a reception device 312, an RF demodulation device 313, a signal processing device 314, a voice decoding device 315, a DZA conversion device 316, and an output device 317.
- An input terminal of the receiving device 312 is connected to the antenna 311.
- the input terminal of the RF demodulator 313 is connected to the output terminal of the receiver 312.
- the input terminal of the signal processor 314 is connected to the output terminal of the RF demodulator 313.
- the input terminal of the speech decoding unit 315 is connected to the output terminal of the signal processing unit 314.
- DZA converter 316 The input terminal of is connected to the output terminal of the voice decoding device 315.
- the input terminal of the output device 317 is connected to the output terminal of the DZA converter 316.
- Receiving apparatus 312 receives a radio wave (RF signal) including voice code information via antenna 311 to generate a received voice code signal which is an analog electric signal, and outputs the RF signal as an RF signal. Feed to the demodulator 313.
- the radio wave (RF signal) received via the antenna 311 is completely the same as the radio wave (RF signal) transmitted from the audio signal transmitting apparatus 300 unless signal attenuation or noise superposition is made in the transmission path.
- the RF demodulator 313 demodulates the received speech code signal from the receiver 312 and supplies it to the signal processor 314.
- a signal processing unit 314 performs jitter absorption buffering processing of the received speech code signal from the RF demodulation unit 313, packet assembling processing and channel decoding processing, etc., and the received speech coded bit sequence is a speech decoding unit.
- the speech decoding unit 315 decodes the received speech code and bit string from the signal processing unit 314 to generate a decoded speech signal and supplies the decoded speech signal to the DZA conversion unit 316.
- the DZA converter 316 converts the digital decoded speech signal from the speech decoder 315 into an analog decoded speech signal and supplies it to the output unit 317.
- the output device 317 converts the analog decoded voice signal from the DZA converter 316 into air vibration and outputs it as sound waves so that it can be heard by the human ear.
- the scalable decoding device according to the present invention is not limited to the above embodiments.
- Embodiments 1 and 2 can be implemented in combination as appropriate.
- FIG. 10 is a block diagram showing a main configuration of a scalable decoding device when Embodiments 1 and 2 are combined.
- the core decoder 102 outputs the acoustic parameter obtained in the decoding process or the acoustic parameter obtained by analyzing the decoded signal to the mode determination unit 201.
- the acoustic parameters all the various parameters as described above can be mentioned as an example.
- Such a configuration is effective when the extension decoder 105 uses a coding algorithm using MDCT.
- various embodiments of the present invention have been described.
- the present invention has been described by way of example in the case of being configured by node software, the present invention can also be realized by software.
- the algorithm of the enhancement layer loss concealment method according to the present invention is described in a programming language, and this program is stored in memory and executed by information processing means, whereby scalable decoding according to the present invention is performed. It is possible to realize the same function as the eyebrow device.
- LSF Line Spectral Frequency
- the core layer is described as a layer that performs code ⁇ Z decoding ⁇ of the narrowband signal. If there is a layer Y for encoding and Z-decoding a signal in a wider band than the above, it is also possible to apply the contents of the present invention with X as the core layer and Y as the enhancement layer. In this case, layer X need not necessarily be the layer that performs code “Z decoding” of the narrowest band signal, and layer X itself may be a scalable structure with multiple layers of power! .
- Each function block employed in the description of each of the aforementioned embodiments may typically be implemented as an LSI constituted by an integrated circuit. These may be individually integrated into a single chip, or may be integrated into a single chip to include some or all of them.
- LSI is used to refer to “IC,” “system LSI,” “super LSI,” “unorellar LSI,” etc., depending on the difference in degree of integration.
- the method of circuit integration is not limited to LSI's, and implementation using dedicated circuitry or general purpose processors is also possible. It is also possible to use an FPGA (Field Programmable Gate Array) that can be programmed after LSI manufacture, or a reconfigurable processor that can reconfigure the connection or setting of circuit cells inside the LSI.
- FPGA Field Programmable Gate Array
- the scalable decoding device and the enhancement layer erasure concealment method according to the present invention can be applied to applications such as communication terminals in a mobile communication system.
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Abstract
Description
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EP05734140A EP1758099A1 (en) | 2004-04-30 | 2005-04-25 | Scalable decoder and expanded layer disappearance hiding method |
US11/587,964 US20080249766A1 (en) | 2004-04-30 | 2005-04-25 | Scalable Decoder And Expanded Layer Disappearance Hiding Method |
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US20080249766A1 (en) | 2008-10-09 |
EP1758099A1 (en) | 2007-02-28 |
CN1950883A (zh) | 2007-04-18 |
JPWO2005106848A1 (ja) | 2007-12-13 |
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