WO2006129615A1 - スケーラブル符号化装置およびスケーラブル符号化方法 - Google Patents
スケーラブル符号化装置およびスケーラブル符号化方法 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
- 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
- G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
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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
- G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
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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
- G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
- G10L19/12—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
Definitions
- the present invention relates to a scalable code encoding device and a scalable code encoding method for applying code encoding to a stereo signal.
- monaural communication is expected to reduce communication costs because it has a low bit rate, and mobile phones that support only monaural communication are less expensive because of their smaller circuit scale.
- mobile phones that support only monaural communication are less expensive because of their smaller circuit scale.
- users who do not want high-quality voice communication will purchase a mobile phone that supports only monaural communication.
- mobile phones that support stereo communication and mobile phones that support monaural communication are mixed in a single communication system, and the communication system needs to support both stereo communication and monaural communication. Arise.
- communication data is exchanged by radio signals, so some communication data may be lost depending on the propagation path environment. Thus, it is very useful if the mobile phone has a function that can restore the remaining communication data based on the received data even if a part of the communication data is lost.
- Non-Patent Literature 1 Ramprashad S. A., “Stereophonic and ELP coding using cross channel p rediction,, Proc. IEEE Workshop on Speech Codings Pages: 136-138, (17-20 Sept. 2000)
- Non-Patent Document 2 ISO / IEC 14496-3: 1999 (B.14 Scalable AAC with core coder) Invention Disclosure
- Non-Patent Document 1 has an adaptive codebook, a fixed codebook, and the like for two-channel audio signals, and each channel separately. A sound source signal is generated and a composite signal is generated. That is, the CELP code of the audio signal is performed for each channel, and the obtained code information of each channel is output to the decoding side. Therefore, there are problems that code parameters are generated for the number of channels, the coding rate increases, and the circuit scale of the code device increases. If the number of adaptive codebooks, fixed codebooks, etc. is reduced, the code rate is reduced and the circuit scale is reduced, but conversely, the sound quality of the decoded signal is greatly degraded.
- an object of the present invention is to provide a scalable coding apparatus and a scalable coding method capable of reducing the code rate and reducing the circuit scale while preventing sound quality deterioration of the decoded signal. It is.
- a scalable coding apparatus includes a monaural code encoding means for encoding a monaural signal, and a driving sound source obtained by the encoding code of the monaural code encoding means.
- 1st prediction means for predicting 1 channel driving excitation
- 1st channel code encoding means for encoding the first channel using the driving excitation predicted by the first prediction means
- the monaural code And second prediction means for predicting the second channel driving sound source included in the stereo signal from the driving sound sources obtained by the encoding means and the first channel coding means, and the second prediction means.
- a second channel encoding means for encoding the second channel using a driving excitation source.
- FIG. 1 is a block diagram showing a main configuration of a scalable code base device according to Embodiment 1.
- FIG. 2 is a block diagram showing a main configuration inside a stereo code base unit according to Embodiment 1.
- FIG. 3 is a flowchart for explaining a procedure of prediction processing performed in the sound source prediction unit according to Embodiment 1.
- FIG. 4 is a flowchart for explaining the procedure of prediction processing performed in the sound source prediction unit according to Embodiment 1.
- FIG. 5 is a block diagram illustrating in more detail the internal configuration of the stereo code key unit according to Embodiment 1.
- FIG. 6 is a block diagram showing the main configuration of the enhancement layer of the scalable coding apparatus according to Embodiment 2
- FIG. 7 is a block diagram showing the main configuration inside the stereo code key unit according to Embodiment 3.
- FIG. 8 is a block diagram illustrating the configuration inside the stereo code key unit according to Embodiment 3 in more detail.
- FIG. 9 is a flowchart showing a procedure of bit allocation processing in the codebook selection unit according to the third embodiment.
- FIG. 10 is a flowchart showing another procedure of bit allocation processing in the codebook selection unit according to the third embodiment.
- FIG. 1 is a block diagram showing the main configuration of scalable coding apparatus 100 according to Embodiment 1 of the present invention.
- a case where a stereo audio signal having two-channel power is encoded will be described as an example, and the first channel and the second channel shown below are respectively an L channel and an R channel, or vice versa. This indicates the channel.
- Scalable code input device 100 includes adder 101, multiplier 102, monaural code input unit 103, and stereo code input unit 104.
- Adder 101, multiplier 102, and monaural code input unit 100 Unit 103 constitutes the base layer, and stereo code key unit 104 constitutes the enhancement layer.
- Each part of the scalable coding apparatus 100 performs the following operations.
- Adder 101 adds first channel signal CH1 and second channel signal CH2 input to scalable coding apparatus 100, and generates a sum signal.
- Multiplier 102 multiplies this sum signal by 1Z2 to halve the scale to generate monaural signal M. That is, the adder 101 and the multiplier 102 obtain an average signal of the first channel signal CH1 and the second channel signal CH2 and set it as the monaural signal M.
- the monaural code key unit 103 encodes the monaural signal M and outputs the obtained encoding parameters.
- the coding parameters are, for example, CEPC codes, LPC (LSP) parameters, adaptive codebook index, adaptive excitation gain, fixed codebook index, and fixed excitation gain.
- the monaural code key unit 103 outputs a driving sound source signal obtained at the time of code keying to the stereo code key unit 104.
- the stereo code key unit 104 is a first channel input to the scalable code key device 100.
- the signal CHI and the second channel signal CH2 are subjected to later-described encoding using the driving excitation signal output from the monaural encoding unit 103, and the resulting stereo signal encoding parameters are output.
- the basic layer outputs a monaural signal code parameter
- the enhancement layer outputs a stereo signal code parameter. It is to be done.
- the stereo signal code parameter is obtained by decoding the stereo signal together with the base layer (monaural signal) code signal parameter in the decoding apparatus. That is, the scalable coding apparatus according to the present embodiment realizes a scalable coding that includes a monaural signal and a stereo signal.
- a decoding device that has acquired base layer and enhancement layer coding parameters cannot obtain enhancement layer coding parameters due to deterioration of the transmission path environment, and can obtain only base layer coding parameters. Even if it works well, it can decode monaural signals, albeit with low quality. Further, if the decoding apparatus can acquire both the base layer and enhancement layer code parameters, a high-quality stereo signal can be decoded using them.
- FIG. 2 is a block diagram showing a main configuration inside the stereo code key unit 104 described above.
- Stereo encoding section 104 includes LPC inverse filter 111, excitation prediction section 112, multiplier 113, CELP code section 114, excitation prediction section 115, multiplier 116, and CELP code section 117.
- a system for processing the first channel signal (LPC inverse filter 111, excitation prediction unit 112, multiplier 113, CELP code unit 114), and a system for processing the second channel signal (sound source prediction unit 115, It is roughly divided into a multiplier 116 and a CELP code section 117).
- the sound source prediction unit 112 predicts the driving sound source signal of the first channel from the driving signal of the monaural signal output from the monaural code unit 103 of the base layer, and multiplies the predicted driving sound source signal by a multiplier. In addition to outputting to 113, information (prediction parameter) P1 regarding this prediction is output. This prediction method will be described later.
- Multiplier 113 multiplies the drive excitation signal of the first channel obtained by excitation prediction section 112 by the predicted excitation gain fed back from CELP code section 114 and outputs the result to CELP code section 114.
- CELP code 114 Using the first channel driving sound source signal output from the multiplier 113, the CELP code of the first channel signal is obtained, and the obtained LPC quantum index P2 and codebook index for the first channel are obtained. P3 is output. CELP code section 114 also outputs quantized LPC coefficients of the first channel signal obtained by LPC analysis and LPC quantization to LPC inverse filter 111. The LPC inverse filter 111 performs inverse filtering processing on the first channel signal using this quantized LPC coefficient, and outputs the obtained driving sound source signal of the first channel signal to the sound source prediction unit 112.
- the sound source prediction unit 115 includes a monaural signal driving sound source signal output from the monaural code unit 103 of the base layer, and a first channel signal driving sound source signal output from the CELP code unit 114. Then, the driving sound source signal of the second channel is predicted, and the predicted driving sound source signal is output to the multiplier 116. This prediction method will also be described later.
- Multiplier 116 multiplies the second channel driving excitation signal obtained by excitation prediction section 115 by the predicted excitation gain fed back from CELP encoding section 117 and outputs the result to CELP encoding section 117.
- the CELP code input unit 117 performs CELP code input of the second channel signal using the second channel driving excitation signal output from the multiplier 116, and obtains the LPC quantization for the second channel obtained. Outputs indepth P4 and codebook index P5.
- FIG. 3 is a flowchart for explaining the procedure of the prediction process performed in the sound source prediction unit 112.
- the sound source prediction unit 112 has a monaural drive sound source signal EXC and a first channel signal.
- Excitation signal EXC is input No. (ST 1010) o sound source prediction unit 112, these
- a delay time difference that maximizes the value of the cross-correlation function between the driving sound source signals is calculated (ST1020).
- the cross-correlation function ⁇ of EXC and EXC follows the following equation (1).
- the sound source prediction unit 112 obtains the amplitude ratio as follows (ST1030). First, EXC
- EXC (n) and EXC (n) are each a monaural driving sound source signal.
- the square root C of the energy ratio between the driving signal of the monaural signal and the driving sound signal of the first channel signal is found according to the following equation (4), and this is used as the amplitude ratio.
- the sound source prediction unit 112 quantizes the calculated delay time difference M and amplitude ratio C with a predetermined number of bits, and uses the quantized delay time difference M and amplitude ratio C to obtain a monaural signal.
- FIG. 4 is a flowchart for explaining the procedure of the prediction process performed in the sound source prediction unit 115.
- the sound source prediction unit 115 converts the driving sound source signal EXC of the second channel into a monaural signal drive.
- this equation (6) is an equation when the monaural signal is an average of the first channel signal and the second channel signal.
- FIG. 5 is a block diagram illustrating the internal configuration of stereo code key unit 104 in more detail.
- stereo code input section 104 includes first channel adaptive codebook 127 and fixed codebook 128, and first codebook search controlled by distortion minimizing section 126 performs a first codebook search.
- a driving sound source signal for a channel is generated.
- the LPC analysis unit 121 performs linear prediction analysis on the first channel signal to obtain an LPC coefficient that is spectrum envelope information.
- the LPC quantization unit 122 quantizes the LPC coefficient, outputs the obtained quantized LPC coefficient to the LPC synthesis filter 123 and the LPC inverse filter 111, and outputs an LPC quantum index ⁇ 2 indicating the quantized LPC coefficient. To do.
- adaptive codebook 127 outputs the driving sound source to multiplier 129 in accordance with the instruction from distortion minimizing section 126.
- fixed codebook 128 outputs a driving sound source to multiplier 130 in accordance with an instruction from distortion minimizing section 126.
- Multiplier 129 and multiplier 130 multiply the outputs from adaptive codebook 127 and fixed codebook 128 by the adaptive codebook gain and fixed codebook gain in accordance with instructions from distortion minimizing section 126, and output the result to adder 131.
- the adder 131 outputs the driving signal of the monaural signal predicted by the sound source prediction unit 112 from each codebook. Add the driving sound source signal.
- the LPC synthesis filter 123 uses the quantized LPC coefficient output from the LPC quantization unit 122 as a filter coefficient, is driven as an LPC synthesis filter by the driving sound source signal output from the adder 131, and adds the synthesized signal. Output to device 124.
- the adder 124 also calculates the coding distortion by subtracting the composite signal from the first channel signal power, and outputs it to the perceptual weighting unit 125.
- the auditory weighting unit 125 performs auditory weighting on the encoded distortion using the perceptual weighting filter using the LPC coefficient output from the LPC analysis unit 121 as a filter coefficient, and outputs the result to the distortion minimizing unit 126.
- Distortion minimizing section 126 obtains each index of adaptive codebook 127 and fixed codebook 128 for each subframe such that the code distortion that is output through perceptual weighting section 125 is minimized, These indexes are output as the sign key parameter P3. Note that the driving sound source signal of the first channel signal when the codebook distortion is minimized is expressed as EXC "(n) in the above equation (6)!
- the driving sound source (the output of the adder 131) when the code distortion is minimized is fed back to the adaptive codebook 127 for each subframe.
- stereo code frame section 104 includes adaptive codebook 147 and fixed codebook 148 for the second channel, and generates a driving excitation signal for the second channel by codebook search.
- the adder 151 adds a driving excitation signal that outputs each codebook power to the driving excitation signal of the monaural signal predicted by the excitation prediction unit 115.
- these drive sound source signals are multiplied by appropriate gains by multipliers 116, 149, and 150.
- the LPC synthesis filter 143 uses the LPC coefficient that is LPC-analyzed by the LPC analysis unit 141 and quantized by the LPC quantization unit 142, based on the second channel drive sound source signal output from the adder 151. And outputs the combined signal to the adder 144.
- the adder 144 calculates the coding distortion by subtracting the synthesized signal from the second channel signal and outputs it to the perceptual weighting unit 145.
- Distortion minimizing section 146 obtains each index of adaptive codebook 147 and fixed codebook 148 for each subframe so that the coding distortion output through perceptual weighting section 145 is minimized. Is output as the sign parameter P5.
- the mark The driving sound source signal of the first channel signal when the distortion of the book is minimized is expressed in the above equation (6) as EXC "(n)! /.
- the generated code key parameters P1 to P5 are sent to the decoding device as the code key parameters of the stereo signal, and are used when decoding the second channel signal.
- stereo coding section 104 of the enhancement layer performs CELP coding using the monaural signal prior to the second channel with respect to the first channel.
- the second channel is efficiently encoded using the result of the CELP code key of the first channel.
- the CELP code signal of the first channel is used.
- the first channel drive sound source is predicted from the monaural signal drive sound source to improve the prediction efficiency and the code rate is reduced.
- the channel is encoded as usual by LPC analysis.
- the prediction accuracy of the driving sound sources of the first channel and the second channel is improved, and as a result, the coding rate can be reduced while preventing the sound quality deterioration of the decoded signal with respect to the stereo audio signal. Further, according to the present embodiment, the circuit scale can be reduced.
- the force described with reference to an example in which the monaural signal is obtained as an average of the first channel and the second channel is not limited to this, and other methods may be used.
- stereo code encoding section 104 performs CELP code encoding on the first channel using a driving signal of a monaural signal first, and the second channel is the first channel.
- the code key is efficiently processed. Therefore, the code accuracy of the first channel that performs the first code influence also on the code accuracy of the second channel. Therefore, if more bits are allocated to the CELP code key of the first channel than the CELP code key of the second channel, the code key performance of the code key device can be improved.
- the “first channel” and “second channel” used in Embodiment 1 are specifically the R channel or the L channel in the stereo signal.
- the first channel and the second channel force are not particularly limited as to which of the R channel and the L channel, and the case where both of them may be applied has been described.
- the first channel is limited to a specific channel by the following method, that is, if one of the R channel and the L channel is selected as the first channel, the code performance of the scalable coding apparatus is further improved. be able to.
- FIG. 6 is a block diagram showing the main configuration of the enhancement layer of the scalable coding apparatus according to Embodiment 2 of the present invention. Note that the same components as those of the scalable coding apparatus shown in Embodiment 1 are denoted by the same reference numerals, and the description thereof is omitted.
- the first channel signal is LPC analyzed by the LPC analysis unit 201-1, and quantized by the LPC quantization unit 202-1, and then quantized by the LPC inverse filter 203-1! / Then, the driving sound source signal of the first channel signal is calculated using the quantized LPC coefficient and output to the channel signal determination unit 204.
- the LPC analysis unit 201-2, the LPC quantization unit 202-2, and the LPC inverse filter 203-2 perform the same processing as the first channel signal on the second channel signal.
- the channel signal determination unit 204 calculates the cross-correlation function between the input driving sound source signal of the first channel signal and the second channel signal and the driving sound source signal of the monaural signal by the following equations (7) and (8 ).
- the channel signal determination unit 204 calculates m that maximizes the calculated ⁇ (m) and ⁇ (m).
- a channel selection flag indicating the selected channel is output to the channel signal selection unit 205.
- the channel selection flag is output to the decoding apparatus for each frame as a code key parameter together with the LPC quantization index and codebook index.
- Channel signal selection section 205 receives an input stereo signal (R channel signal, L channel signal) based on the channel selection flag output from channel signal determination section 204, and is input to stereo coding section 104. Are classified as the first channel signal and the second channel signal.
- the channel having the higher correlation with the monaural signal is selected and used as the first channel of stereo coding unit 104.
- the encoding performance of the encoding device can be improved.
- the stereo code unit 104 performs the CELP code signal using the driving signal of the monaural signal before the first channel
- the second channel uses the CELP code signal of the first channel. Efficiently sign using the result. Therefore, the code accuracy of the first channel that performs the first code influences the accuracy of the second channel. That is, it is easily understood that if the channel having the higher correlation with the monaural signal is set as the first channel as in the present embodiment, the code accuracy of the first channel is improved.
- the channel selection flag can be sent together so that a plurality of frames other than each frame select the same channel signal. Alternatively, first, after calculating the cross-correlation function of several frames, it may be determined which channel signal is the first channel and the channel selection flag is sent first.
- Embodiment 3 of the present invention discloses a method for changing the bit distribution in the scalable code generator according to the present invention.
- the scalable coding apparatus performs the coding of the first channel signal and the coding of the second channel signal, so that the coding code is distributed to both the first channel and the second channel. If the number of bits can be increased, both the code distortion of the first channel and the code distortion of the second channel can be reduced.
- the influence on the second channel code distortion when the number of bits for the first channel is increased is not limited to the negative aspect.
- the second channel drive sound source signal is predicted from the monaural signal drive sound source signal and the first channel signal drive sound source signal in the scalable coding apparatus according to the present invention (see FIG. 4).
- the sign distortion of the second channel signal depends on the coding distortion of the first channel signal. Therefore, if the mutual dependency between the first channel code distortion and the second channel coding distortion is taken into account, the number of bits allocated to the first channel increases, and the first channel code distortion As the signal decreases, the sign distortion of the second channel signal also decreases. That is, in the scalable coding apparatus according to the present invention, the influence of the increase in the number of bits for the first channel on the coding distortion of the second channel includes a positive aspect.
- the overall code efficiency of the scalable encoding device is improved by adaptively allocating the number of bits to the first channel and the second channel.
- the number of bits is adaptively applied to the first channel and the second channel so that the first channel code distortion and the second channel code distortion are equal. To distribute.
- Scalable coding apparatus 300 has a basic configuration similar to that of scalable coding apparatus 100 (see FIG. 1) shown in the first embodiment.
- the block diagram showing the configuration of the dredging device 300 is omitted.
- the stereo code key unit 304 of the scalable code key device 300 is different from the stereo code key unit 104 shown in Embodiment 1 in part in configuration and operation, and thus is given a different code.
- Scalable code device 30 Bit allocation at 0 is performed within the stereo code section 304.
- FIG. 7 is a block diagram showing a main configuration inside stereo coding unit 304 according to the present embodiment.
- Stereo code key section 304 has the same basic configuration as stereo code key section 104 (see FIG. 2) shown in the first embodiment, and the same reference numerals are given to the same components. The description is omitted.
- the stereo code key unit 304 according to the present embodiment is different from the stereo code key unit 104 shown in the first embodiment in that it further includes a code book selection unit 318.
- CELP code key unit 314 and CELP code key unit 317 have the same basic configuration as CELP code key unit 114 and CELP code key unit 117 shown in the first embodiment. There are differences in some configurations and operations. These differences will be described below.
- CELP code key unit 314 outputs the LPC quantum key index for the first channel and the codebook index for the first channel to the codebook selection unit 318 instead of outputting them as coding parameters. This differs from the CELP code key unit 114 shown in the first embodiment.
- the CELP code key unit 314 further outputs the minimum code key distortion of the first channel signal to the code book selection unit 318, and the code book selection index 318 for the first channel is fed back. This is different from the CELP code key unit 114 shown in the first embodiment.
- the minimum code distortion of the first channel is obtained by a closed loop distortion minimization process performed to minimize the encoding distortion of the first channel in the CELP code key section 314. This is the minimum encoding distortion of one channel signal.
- CELP code key unit 317 outputs the second channel LPC quantum key index and the second channel code book index to code book selection unit 318 instead of outputting them as coding parameters. This differs from the CELP code key unit 117 shown in the first embodiment.
- the CELP code key unit 317 further outputs the minimum code key distortion of the second channel signal to the code book selection unit 318, and the code book selection index for the second channel is fed back from the code book selection unit 318.
- the minimum code distortion of the second channel is obtained from the closed loop distortion minimization process performed to minimize the encoding distortion of the second channel in the CELP encoder 317. The minimum value of the sign distortion of the second channel signal.
- Codebook selection section 318 receives from LLP quantization index for the first channel, codebook index for the first channel, and minimum coding distortion of the first channel signal from CELP code section 314.
- the CELP code input unit 317 receives the LPC quantization index for the second channel, the codebook index for the second channel, and the minimum coding distortion of the second channel signal.
- the codebook selection unit 318 performs codebook selection processing using these inputs, feeds back the codebook selection index for the first channel to the CELP code input unit 314, and the second channel to the CELP encoding unit 317. Feed back the codebook selection index.
- the codebook selection processing in the codebook selection unit 318 means that the minimum coding distortion of the first channel signal and the minimum coding distortion of the second channel signal are equalized. This is a process of changing the number of bits allocated to the heel part 317 and indicating the change information of the number of bits using the codebook selection index for the first channel and the codebook selection index for the second channel.
- Codebook selection section 318 includes first channel LPC quantization index P2, first channel codebook index P3, second channel LPC quantum index P4, second channel codebook index P5, and Bit allocation selection information P6 is output as a sign key parameter.
- FIG. 8 is a block diagram illustrating in more detail the internal configuration of stereo coding unit 304 according to the present embodiment.
- This figure mainly shows the internal configuration of CELP code key section 314 in more detail, and the internal configuration of CELP code key section 317 is the same as the internal configuration of CELP code key section 314. The explanation is omitted. In this figure, the description of the same parts as those shown in FIG. 5 of the first embodiment will be omitted, and only the different parts will be described.
- Fixed codebook 328 includes first fixed codebook 328-1 to n-th fixed codebook 328-n, and any one of first fixed codebook 328-1 to n-th fixed codebook 328-n This is different from fixed codebook 128 described in Embodiment 1 in that the driving sound source is output and the output destination of the driving sound source is switching unit 321 instead of multiplier 130.
- the first fixed codebook 328-1 to the nth fixed codebook 328-n are n fixed codebooks having different bit rates, so that the fixed codebook 328 uses the switching unit 321 to output a driving sound source. By changing the number of sign bits for the first channel.
- the number of bits required by the fixed codebook than the number of bits required by the adaptive codebook In this case, changing the number of allocated bits in the fixed codebook 328 is more effective in improving the coding distortion than changing the number of allocated bits in the adaptive codebook 127. Therefore, in this embodiment, the number of bits allocated to both channels is changed by changing the fixed codebook index of fixed codebook 328 instead of the codebook index of adaptive codebook 127.
- the LPC quantization unit 322 does not output the LPC quantum index for the first channel as the code parameter, but outputs it to the codebook selection unit 318, as described in Embodiment 1. This is different from the LPC quantization unit 122.
- Distortion minimizing section 326 outputs the first channel codebook index to codebook selecting section 318 instead of outputting it as a code key parameter, and further outputs the first channel signal to codebook selecting section 318. It differs from the distortion minimizing section 126 shown in Embodiment 1 in that it outputs the minimum coding distortion.
- the minimum code distortion of the first channel signal means that the codebook selection unit 318 switches the distortion minimizing unit 326 from the first fixed codebook 328-1 to the nth fixed codebook 328-n based on the instruction. However, this is the minimum value of the first channel signal encoding distortion that is finally obtained by performing the closed-loop distortion minimization process to minimize the first channel code distortion.
- the codebook selection unit 318 receives the LPC quantum index for the first channel and the codebook index for the first channel from the LPC quantization unit 322, and receives the first channel signal from the distortion minimization unit 326. The minimum code distortion is input. Similarly, the codebook selection unit 318 receives the LPC quantization index for the second channel, the codebook index for the second channel, and the minimum code distortion of the second channel signal from the CELP code key unit 317. The The codebook selection unit 318 performs codebook selection processing using these inputs, feeds back the codebook selection index for the first channel to the switching unit 321, and feeds the codebook for the second channel to the CELP encoding unit 317. Feedback selection index.
- the codebook selection index for the first channel is an index indicating each of the first fixed codebook 328-1 to the nth fixed codebook 328-n used by the fixed codebook 328 for the first channel code. It is.
- the codebook selection unit 318 includes the LPC quantization index P2 for the first channel, the codebook index P3 for the first channel, the LPC quantization index P4 for the second channel, and the second channel.
- the codebook index P5 for use and the bit allocation selection information P6 are each output as the code parameter.
- Switching section 321 switches the path between fixed codebook 328 and multiplier 130 based on the codebook selection index input from codebook selection section 318. For example, when the codebook indicated by the codebook selection index input from the codebook selection unit 318 is the second fixed codebook 328-2, the switching unit 321 selects the driving sound source of the second fixed codebook 328-2. Output to the multiplier 130.
- FIG. 9 is a flowchart showing the procedure of bit allocation processing in codebook selection section 318.
- the processing shown in this figure is performed in units of frames, and bit allocation is performed so that the coding distortion of the first channel signal and the coding distortion of the second channel signal are equal.
- codebook selection section 318 allocates the minimum number of bits for both channels and initializes the bit allocation processing. That is, the codebook selection unit 318 instructs the fixed codebook 328 to use the fixed codebook having the minimum bit rate, for example, the second fixed codebook 32-2, via the codebook selection index for the first channel. To do.
- the processing of the codebook selection unit 318 for the second channel is the same as the processing for the first channel.
- minimum coding distortion of the first channel signal and minimum coding distortion of the second channel signal are input to codebook selection section 318. That is, when using, for example, the second fixed codebook 32-2 as the fixed codebook 328, the distortion minimizing section 326 obtains the minimum value of the coding distortion of the first channel signal in such a case, and sends it to the codebook selection section 318. Output.
- the fixed codebook used by fixed codebook 328 is the one specified by codebook selection section 318 in the step prior to ST3020.
- the processing in the second channel is the same as the processing in the first channel.
- codebook selecting section 318 compares the minimum coding distortion of the first channel signal with the minimum coding distortion of the second channel signal. If the minimum code distortion of the first channel signal is larger than the minimum code distortion of the second channel signal, codebook selection section 318 increases the number of bits for the first channel in ST3040. That is, the codebook selection unit 318 instructs the fixed codebook 328 to use the fixed codebook having a higher bit rate, for example, the fourth fixed codebook 328-4, via the codebook selection index for the first channel. . on the other hand, When the minimum coding distortion of the first channel signal is smaller than the minimum coding distortion of the second channel signal, the codebook selection unit 318 increases the number of bits for the second channel in ST3050! ] The method for increasing the number of bits for the second channel is the same as the method for increasing the number of bits for the first channel.
- ST3060 it is determined whether or not the total number of bits already allocated to both channels has reached the upper limit value. When the sum of the number of bits allocated to both channels reaches the upper limit value, it returns to ST3020, and until the sum of the number of bits allocated to both channels reaches the upper limit value, the codebook selection unit 318 operates from ST3020 onwards. Repeat the process of ST3060.
- codebook selection section 318 first allocates the minimum bit rate for both channels, and maintains equality between the coding distortion of the first channel signal and the coding distortion of the second channel signal. However, the number of bits allocated to both channels is gradually increased, and finally a predetermined upper limit number of bits is allocated to both channels. In other words, the total number of bits allocated to both channels gradually increases from the minimum value according to the progress of processing, and finally reaches a predetermined upper limit value.
- FIG. 10 is a flowchart showing another procedure of bit allocation processing in codebook selection section 318.
- the processing shown in this figure is also performed on a frame-by-frame basis, similar to the processing shown in FIG. 9. Make an allocation.
- the processing shown in FIG. 9 shows that the sum of the number of bits allocated to both channels gradually increases from the minimum value according to the progress of processing and finally reaches a predetermined upper limit value.
- the initial power bit number for both channels is distributed equally to both channels until the code distortion of the first channel signal and the code distortion of the second channel signal are equal. Adjust the percentage of numbers.
- the detailed operation of each component of the scalable coding apparatus 300 in each step of the processing procedure will not be described (see the description of FIG. 10).
- codebook selection section 318 distributes a predetermined upper limit number of bits evenly to both channels, and initializes bit allocation processing.
- codebook selection section 318 receives the minimum coding distortion of the first channel signal and the minimum coding distortion of the second channel signal.
- the codebook selection unit 318 performs the minimum code of the first channel signal. Compare the coding distortion with the minimum coding distortion of the second channel signal. When the minimum code distortion of the first channel signal is larger than the minimum code distortion of the second channel signal, the codebook selection unit 318 increases the number of bits for the first channel and increases the number of bits for the second channel in ST3140. Decrease the number of bits.
- the increase in the number of bits for the first channel is the same as the decrease in the number of bits for the second channel.
- the codebook selection unit 318 reduces the number of bits for the first channel and reduces the second channel in ST3150. Increase the number of bits for.
- the decrease in the number of bits for the first channel is the same as the increase in the number of bits for the second channel.
- codebook selecting section 318 determines whether or not the difference between the minimum coding distortion of the first channel signal and the minimum coding distortion of the second channel signal is a predetermined value or less.
- codebook selecting section 318 determines that the difference between the minimum coding distortion of the first channel signal and the minimum coding distortion of the second channel signal is equal to or less than a predetermined value, Judgment distortion is equal to the minimum coding distortion of the second channel signal. If the difference between these two minimum code distortions is not less than or equal to the predetermined value, the process returns to ST3120, and the codebook selection unit 318 determines whether the difference between the two minimum code distortions is equal to or less than the predetermined value. Repeat the process.
- the procedure shown in this figure is different from the initialization of the bit allocation process shown in Fig. 9 in that the predetermined upper limit number of bits is evenly distributed to both channels in initialization.
- the predetermined upper limit number of bits is set so that the encoding distortion of the first channel signal and the encoding distortion of the second channel signal are equal to those in the procedure shown in FIG. To channel.
- the predetermined upper limit number of bits is set to both channels so that the code distortion of the first channel signal and the code distortion of the second channel signal are equal. Therefore, it is possible to reduce the code distortion of the encoder apparatus and improve the encoder performance of the encoder apparatus.
- bit allocation is performed so that the encoding distortion of the first channel signal and the encoding distortion of the second channel signal are equalized has been described as an example.
- the sum of the sign distortion of the first channel signal and the sign distortion of the second channel signal is minimized.
- bit allocation may be performed.
- the method of allocating bits so that the sum of the sign distortion of the first channel signal and the sign distortion of the second channel signal is minimized is that the coding distortion of one of the channel signals increases due to the increase in the number of bits. This method is optimally applied when the degree of improvement in the sign distortion of the other channel signal is significantly greater than the degree of improvement in the other channel signal.
- bit allocation processing is initialized by allocating more bits to the first channel than to the second channel. Also good. Furthermore, the value of the cross-correlation function between the monaural signal and the first channel signal and the value of the cross-correlation function between the monaural signal and the second channel signal are obtained.
- the bit allocation processing may be initialized by adaptively increasing the number of bits to be allocated. This improved initialization process can reduce the number of loop processes required to equalize the minimum code distortion of the first channel signal and the minimum code distortion of the second channel signal. And bit allocation processing can be shortened.
- code codes other than the fixed codebook index are used as a target for changing the bit distribution. It may be a parameter. For example, code key information such as LPC parameters, adaptive codebook lag, and sound source gain parameters may be adaptively changed.
- bit allocation may be performed based on information other than code distortion.
- bit allocation may be performed based on the prediction gain of the sound source prediction unit.
- the value of the cross-correlation function between the monaural signal and the first channel signal and the phase between the monaural signal and the second channel signal You may perform bit allocation using the value of a cross correlation function, etc.
- the value of the cross-correlation function between the monaural signal and the first channel signal and the value of the cross-correlation function between the monaural signal and the second channel signal are obtained, and more bits are assigned to the channel with the smaller value of the cross-correlation function. Allocate numbers.
- the number of bits allocated to the first channel may be adaptively increased in consideration of the fact that the code distortion of the second channel signal depends on the code distortion of the first channel signal.
- the scalable encoding device and scalable encoding method according to the present invention are not limited to the above embodiments, and can be implemented with various modifications. For example, each embodiment can be implemented in combination as appropriate.
- the fixed codebook may be called a fixed excitation codebook, a noise codebook, a stochastic codebook, or a random codebook.
- the adaptive codebook may also be referred to as an adaptive excitation codebook.
- the LSP is sometimes called LSF (Line Spectral Frequency), and the LSP may be read as LSF.
- LSF Line Spectral Frequency
- ISP Interference Spectrum Pairs
- the present invention is realized as an ISP code ⁇ Z decoding device. Can be used.
- the scalable coding apparatus can be installed in a communication terminal apparatus and a base station apparatus in a mobile communication system, and thereby has a function and effect similar to the above.
- An apparatus, a base station apparatus, and a mobile communication system can be provided.
- the power described with reference to an example in which the present invention is configured by nodeware can be realized by software.
- a scalable code encoding method according to the present invention is described by describing an algorithm of the scalable code encoding method according to the present invention in a programming language, storing the program in a memory, and causing the information processing means to execute the program. Functions similar to those of the apparatus can be realized.
- each functional block used in the description of each of the above embodiments is typically realized as an LSI that is an integrated circuit. These may be individually integrated into one chip, or part or One chip may be included to include everything.
- 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 a field programmable gate array (FPGA) that can be programmed after LSI manufacturing, or a reconfigurable processor that can reconfigure the connection or setting of circuit cells inside the LSI.
- FPGA field programmable gate array
- the scalable code frame apparatus and the scalable code frame method according to the present invention can be applied to applications such as a communication terminal apparatus and a base station apparatus in a mobile communication system.
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Abstract
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CN2006800191271A CN101185123B (zh) | 2005-05-31 | 2006-05-29 | 可扩展编码装置及可扩展编码方法 |
US11/915,617 US8271275B2 (en) | 2005-05-31 | 2006-05-29 | Scalable encoding device, and scalable encoding method |
DE602006015461T DE602006015461D1 (de) | 2005-05-31 | 2006-05-29 | Einrichtung und verfahren zur skalierbaren codierung |
JP2007518977A JP4948401B2 (ja) | 2005-05-31 | 2006-05-29 | スケーラブル符号化装置およびスケーラブル符号化方法 |
EP06746967A EP1887567B1 (en) | 2005-05-31 | 2006-05-29 | Scalable encoding device, and scalable encoding method |
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EP (1) | EP1887567B1 (ja) |
JP (1) | JP4948401B2 (ja) |
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GB2453117A (en) * | 2007-09-25 | 2009-04-01 | Motorola Inc | Down-mixing a stereo speech signal to a mono signal for encoding with a mono encoder such as a celp encoder |
WO2009116280A1 (ja) * | 2008-03-19 | 2009-09-24 | パナソニック株式会社 | ステレオ信号符号化装置、ステレオ信号復号装置およびこれらの方法 |
JP5413839B2 (ja) * | 2007-10-31 | 2014-02-12 | パナソニック株式会社 | 符号化装置および復号装置 |
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US8489403B1 (en) * | 2010-08-25 | 2013-07-16 | Foundation For Research and Technology—Institute of Computer Science ‘FORTH-ICS’ | Apparatuses, methods and systems for sparse sinusoidal audio processing and transmission |
US9183842B2 (en) * | 2011-11-08 | 2015-11-10 | Vixs Systems Inc. | Transcoder with dynamic audio channel changing |
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US20090271184A1 (en) | 2009-10-29 |
EP1887567B1 (en) | 2010-07-14 |
CN101185123A (zh) | 2008-05-21 |
JP4948401B2 (ja) | 2012-06-06 |
EP1887567A4 (en) | 2009-07-01 |
DE602006015461D1 (de) | 2010-08-26 |
JPWO2006129615A1 (ja) | 2009-01-08 |
US8271275B2 (en) | 2012-09-18 |
CN101185123B (zh) | 2011-07-13 |
EP1887567A1 (en) | 2008-02-13 |
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