KR20090026504A - Method and apparatus for assessing audio signal spectrum - Google Patents
Method and apparatus for assessing audio signal spectrum Download PDFInfo
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- KR20090026504A KR20090026504A KR1020070091531A KR20070091531A KR20090026504A KR 20090026504 A KR20090026504 A KR 20090026504A KR 1020070091531 A KR1020070091531 A KR 1020070091531A KR 20070091531 A KR20070091531 A KR 20070091531A KR 20090026504 A KR20090026504 A KR 20090026504A
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- 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
- G10L19/02—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 spectral analysis, e.g. transform vocoders or subband vocoders
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- G10L25/48—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 specially adapted for particular use
- G10L25/51—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 specially adapted for particular use for comparison or discrimination
- G10L25/60—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 specially adapted for particular use for comparison or discrimination for measuring the quality of voice signals
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Abstract
The present invention is to evaluate the voice signal processing performance of any system utilizing the digital signal processing technology of the voice signal, to set the overall signal processing environment of any system, and to apply to any system under the set system environment The speech signal is coded and compressed according to the codec, the compressed speech signal is adapted to the set system environment, decoded according to the speech codec, reconstructed into a final synthesized speech signal, and then the original spectrum and the final The final synthesized spectrum of the reconstructed speech signal is compared using a predetermined spectrum evaluation method, and the distortion degree according to the difference is determined to evaluate the speech processing performance of an arbitrary system.
Description
The present invention relates to a digital signal processing method and apparatus for speech signals, and more particularly, to a method and apparatus for evaluating a spectrum of a speech signal recovered after speech signal processing.
Digital signal processing technology for converting analog voice signals into digital voice signals, coding and compressing them, decoding the compressed voice signals, combining them, and restoring them into analog voice signals is used in various fields. It is becoming. For example, it is used in technical fields such as voice communication, digital broadcasting, and voice recognition, and in addition, there are various technical fields in which digital signal processing technology of voice signals, such as storage and playback of video and music contents, is utilized. Accordingly, digital signal processing schemes for voice signals have been variously developed, and digital signal processing for voice signals of various schemes may be performed by a voice codec corresponding to each scheme.
Each voice codec has performance characteristics according to a codec-specific digital signal processing scheme, and thus, the distortion degree of the voice signal restored by the same voice codec is maintained at a constant level with respect to the original voice signal. However, since the type of the system to which the voice codec is applied or provided may be different, even the voice signal processed by the same voice codec, the reconstructed voice signal may have a different distortion degree than the original voice signal. In other words, the distortion of the reconstructed speech signal may be evaluated differently according to the characteristics of the system to which the speech codec is applied. The voice codec is a technology related to direct processing of a voice signal, that is, compression and reconstruction according to coding and decoding of the voice signal. However, due to the characteristics of a system to which the voice codec is applied, a compressed voice signal may be transmitted through a wired or wireless communication network. In order to improve the transmission characteristic of the audio codec, a process other than the speech signal processing unique to the speech codec may be added. Therefore, even if the same voice codec is used, the sound quality of the voice signal that is finally restored and provided to the user may vary depending on which system the voice codec is used. In this case, the system includes a system associated with each of various technical fields in which digital signal processing of a voice signal may be utilized. For example, a mobile communication system, a voice recognition system, a remote voice recognition system, a digital broadcasting system, and the like may be included.
Voice signals, on the other hand, include voiced sounds, unvoiced sounds and noise. Among them, voiced sound occupies the largest part of voice signal entropy and includes the most characteristic of voice signal. And the characteristic of the voiced sound is most effectively represented by the characteristic of the spectrum of the speech signal. Accordingly, in the digital signal processing of the speech signal, the processing performance of the voiced sound included in the speech signal has the most influence on the distortion degree, and in particular, the coding and decoding processing of the spectral features of the speech signal determine the sound quality of the restored speech signal.
According to this characteristic, the speech signal processing performance evaluation of the system to which the speech codec is applied is not the performance evaluation of the speech codec itself, but the speech distortion processing evaluation must be performed on the entire system to which the speech codec is applied, so that the valid speech signal processing performance can be evaluated. In addition, when applying a specific voice codec to any system, it is desirable to be able to evaluate the speech signal processing performance according to a certain system and a specific voice codec combination in advance before actually implementing any system completely. In particular, in the case of a system for transmitting and receiving a voice signal through a communication network, if the voice signal processing performance evaluation of the system is performed in advance, waste of resources may be prevented. Furthermore, if the speech signal processing performance according to a combination of any system and various speech codecs can be evaluated in advance, it is possible to identify a speech codec or a speech codec to be preferably applied to any system. In particular, if the spectral characteristics of the speech signal can be used to evaluate the speech signal processing performance of an arbitrary system, it will be possible to evaluate the speech signal processing performance more accurately using less signal processing.
Accordingly, the present invention can provide a speech spectrum evaluation method and apparatus capable of evaluating speech processing performance of a system to which a speech codec is applied.
In addition, the present invention can provide a speech spectrum evaluation method and apparatus capable of testing a speech codec exhibiting good speech signal processing performance when applied to any system utilizing digital signal processing of speech signals.
In addition, the present invention can provide a speech spectrum evaluation method and apparatus capable of evaluating speech processing performance by combining various speech codecs in any system without implementing any system utilizing digital signal processing of speech signals. In addition, it is possible to provide a method and apparatus for evaluating the speech processing capability of a system using an actual implemented speech signal according to the degree of spectral distortion.
In order to solve the above problems, the present invention is a voice signal spectrum evaluation method of a spectrum evaluation device for evaluating the voice signal processing performance of any system utilizing digital signal processing of a voice signal, the system of the any system Setting a signal processing environment according to a feature, setting a voice codec selected to be applied to the arbitrary system; and when a voice signal is input, according to the voice signal processing method of the voice codec under the set signal processing environment, the voice Analyzing the signal to detect the original spectrum, and processing the speech signal into a final speech signal according to the speech signal processing method of the speech codec under the set signal processing environment, and converting the final spectrum from the final speech signal. Detecting and the original spectrum And compared according to the rating system set spectrum the final spectrum pre-detecting the spectral distortion includes the step of evaluating the performance of audio signal processing.
The predetermined spectrum evaluation method may include a spectral flatness detection method, a segmental signal-to-noise ratio measure (Seg-SNR) method, a linear predictive coding parameter measure (LLR) method, and a log likelihood ratio (LLR). ) One of the measuring methods and the capstrum distance measuring method.
The process of detecting the original spectrum by analyzing the voice signal may include receiving a voice signal by activating a microphone which is the same as a microphone type set in the arbitrary system, and according to a noise removing method set according to the signal processing environment. Removing the noise included in the voice signal, converting the noise-removed voice signal into a voice signal in a frequency domain according to a voice signal processing method of the voice codec under the set signal processing environment, and analyzing the original signal Extracting the spectral features.
The detecting of the final spectrum from the final speech signal may include encoding or decoding the converted speech signal according to a speech signal processing scheme of the speech codec under the signal processing environment, and adapting the final speech signal to the final speech signal. Converting to a speech signal and detecting the final spectrum from the final speech signal.
The process of evaluating the speech signal processing performance may include detecting the distortion degree of the speech signal by comparing the original spectrum and the final spectrum according to the preset spectrum evaluation method, and comparing the predetermined reference value with the detected distortion degree. And notifying, according to the result, whether the spectral performance has been passed for the arbitrary system.
The final speech signal is a speech signal in a format finally converted by performing a final process of speech signal processing in the arbitrary system.
The present invention can evaluate the speech processing performance of a system to which the speech codec is applied according to the distortion degree of the speech spectrum, and exhibits good speech signal processing performance when applied to any system utilizing digital signal processing of speech signals. The voice codec can be tested, and the voice processing performance can be evaluated by combining various voice codecs in any system without implementing any system. As a result, it is possible to save resources related to the construction of an arbitrary system, and to perform an evaluation and inspection of whether a voice signal processing function suitable for providing a voice service is provided before being released into a real product.
Hereinafter, exemplary embodiments of the present invention will be described in detail with reference to the accompanying drawings. Note that the same components in the drawings are represented by the same reference numerals and symbols as much as possible even though they are shown in different drawings. In addition, in describing the present invention, when it is determined that a detailed description of a related known function or configuration may unnecessarily obscure the subject matter of the present invention, the detailed description thereof will be omitted.
The present invention is to evaluate the voice signal processing performance of any system utilizing the digital signal processing technology of the voice signal, to set the overall signal processing environment of any system, and to apply to any system under the set system environment The speech signal is processed according to the speech codec, and the processed speech signal is adapted to the set system environment, thereby changing the speech signal to be finally used in the set system. Then, the original spectrum of the original speech signal and the final spectrum of the finally changed speech signal are compared using a predetermined spectrum evaluation method, and the degree of distortion according to the difference is determined to evaluate the speech processing performance of an arbitrary system. The arbitrary system may be any system that utilizes digital signal processing technology of a voice signal, such as a mobile communication system, a voice recognition system, a remote voice recognition system, a digital broadcast system, and the like. In addition, the digital signal processing technology of the voice signal may be various voice signal codecs. If any system is a far-field speech recognition system, a signal processing environment of the speech recognition system, for example, a type of a microphone collecting a voice signal, a method of removing noise from a signal collected from the microphone, and performing a recognition process Set the signal transmission environment to the server. The A voice codec to be included in the far speech recognition system is set. With the basic environment set up as described above, the present invention processes the voice signal according to the set microphone type and noise reduction method, and codes and compresses the voice signal according to the A voice codec. The compressed speech signal is then adapted to the set signal transmission environment. For example, if the compressed voice signal is larger than the set signal transmission standard, the transmission environment may be set to recompress and transmit the compressed voice signal in another manner, and to decompress the received voice signal. If such a transmission environment is set, the compressed voice signal is compressed / restored in a manner set for signal transmission. Then, the compressed speech signal is decoded and synthesized according to the A speech codec and converted into a final speech signal. In this case, the final voice signal is a voice signal used directly in a voice service process such as voice recognition or a call. In addition, the original spectrum of the original speech signal and the final spectrum of the final speech signal are compared by using a predetermined spectrum evaluation method to determine the degree of distortion, and the speech processing performance of the far speech recognition system is evaluated. The spectral evaluation method is a method of measuring distortion, a spectral flatness detection method, a segmental signal-to-noise ratio measure (SEG-SNR) method, a linear predictive coding parameter measure (linear predictive coding parameter measure) method The LLR (Log Likelihood Ratio) measuring method, the capstrum distance measuring method, and the spectral distortion measurement method newly defined according to the present invention may be set to any one.
According to an exemplary embodiment of the present invention, according to a voice codec set on a set system environment, the entire voice signal may be digitally processed, followed by extracting and evaluating a spectral component. It can also be evaluated. Accordingly, an embodiment of extracting and evaluating a spectral component after digital signal processing of the voice signal itself will be described with reference to FIGS. 1 and 2.
1 is a diagram showing the configuration of a first audio signal spectrum evaluation apparatus according to an embodiment of the present invention. Referring to FIG. 1, a first audio signal spectrum evaluation apparatus includes a voice
The
The voice
The
The
In the spectrum evaluation method set in the
In the spectral distortion measure for calculating the spectrum of the original speech signal and the compressed speech signal, the spectrum S (ω) of the linear predictive polynomial of the P th order is expressed by Equation 1.
Where S (ω) is the spectrum,
Is the nth linear predictive coefficient. For n> P the corresponding linear prediction coefficient is zero. Where N represents the number of spectral samples and is usually chosen to be much larger than P. The spectral distortion (SD) according to this is shown in Equation 2.
Where M is the total number of frames used,
Wow Denote the original spectral characteristics of the original speech signal and the final spectral characteristics of the final speech signal, respectively, for the m th frame. The spectral feature may use a linear prediction based feature. And May be a compressed spectral characteristic of the compressed speech signal or a composite spectral characteristic of the synthesized speech signal according to system characteristics.A measure of spectral distortion can be seen as a measure of the distance between the original speech signal and the smoothed spectra of the synthesized speech signal. To find smoothed spectra, cepstral or filter-bank analysis techniques can be used. And other weights for spectral distortion, such as using a different standard instead of the Euclidean standard used in Equation 2, or a fixed frequency domain weighting filter or variable weights calculated from the original and compressed spectra. Other methods such as the use of can be used. For example, a spectral flatness detection method, a segmental signal-to-noise ratio measure (Seg-SNR) method, a linear predictive coding parameter measure method, and a log likelihood ratio measurement method , Spectral distortion can be measured using a capstrum distance measurement method.
The first audio signal spectrum evaluation apparatus may have the following assumptions according to an embodiment of the present invention. The first audio signal spectrum evaluation device is configured on the assumption that it analyzes and compresses the whole, but not the part of the speech signal, and decodes the compressed speech signal. That is, the compressed speech signal coded and compressed by the
2 illustrates an operation of the first spectrum evaluation device configured as described above. FIG. 2 is a flowchart illustrating an operation of a first spectrum evaluation apparatus according to a system characteristic when the final speech signal is a decoded synthesized speech signal. Referring to FIG. 2, in
On the other hand, according to another embodiment of the present invention, the spectrum evaluation apparatus may be configured as shown in FIG. 2 illustrates a configuration of a second spectrum evaluation apparatus according to another embodiment of the present invention, and the second spectrum evaluation apparatus evaluates the speech spectrum by coding / decoding only the spectral components of the speech signal. Referring to FIG. 2, the apparatus for evaluating a second voice signal spectrum includes a voice
The
The voice
The
The
The
The
The second spectrum evaluation device does not use the compression spectrum parameters for the selected speech codec, is not affected by the static performance of the compression spectrum, and the dynamic behavior of the compression (inter-frame effects). Reflects well. In contrast, the first spectral distortion measurement method reflects static distortion.
In addition, the second spectrum evaluation apparatus may selectively set only the codec process associated with the spectrum without actually implementing or setting a complete voice codec. Therefore, it may not be affected by the distortion caused by other parts of the selected voice codec.
In addition, the evaluation of the new spectral coding method is very simple, and spectral performance evaluation is possible without making any assumptions about the rest of the speech codec algorithm. In another aspect, this method also means that the appropriate choice of short-term spectrum quantization scheme for a particular speech coding method is independent of other parts of the codec. And the spectral evaluation method by the second spectrum evaluation device is also very closely related to the result of subjective sound quality judgment, which takes into account the short-term spectral envelope of the linear prediction model. This can be a good model for the human auditory system.
An operation process of the second spectrum evaluation device configured as described above is illustrated in FIG. 4. Referring to FIG. 4, in
Since the first spectrum evaluation apparatus compresses and restores the speech signal itself according to the set speech codec, the distortion degree of the final spectrum may be affected by other components constituting the speech signal in addition to the spectrum depending on the type of the speech codec. . On the contrary, since the second spectrum evaluation apparatus extracts only the spectral components from the speech signal and compresses and restores the extracted spectral components according to the set speech codec, the second spectrum evaluation apparatus can determine the degree of distortion of the spectral components itself compared to the first spectrum evaluation apparatus. . Therefore, the speech processing performance of the system can be evaluated by selecting an appropriate device from the first spectrum evaluation device and the second spectrum evaluation device according to the type and characteristics of the system utilizing the digital signal processing of the voice signal.
The example which applies this invention to a network robot system is demonstrated. In the network robot system, the network robot processes the voice signal input through the microphone from the robot terminal, removes the noise and the echo sound through the voice preprocessing unit, and then codes (compresses) the spectral feature extracted from the voice signal itself or the voice signal. Transfer to the server at The remote server decodes the transmitted voice signal or voice feature signal, restores the synthesized voice signal, synthesizes the voice signal, and performs a voice service. For example, when the voice service is a voice call, voice communication with the receiver may be used as an input signal of the voice recognizer.
The speech performance evaluation method according to the present invention is applied to such a network robot system as follows. First, if the network robot system is simulated with the first spectrum evaluation device, the
In the above description of the present invention, specific embodiments have been described, but various modifications can be made without departing from the scope of the present invention. In the above example, the first spectrum evaluation device and the second spectrum evaluation device are independently configured. However, in the embodiment of the present invention, the first spectrum evaluation device may be included in one device. In addition, although the final
1 is a view showing the configuration of a first spectrum evaluation apparatus according to an embodiment of the present invention;
2 is a view showing an operation process of the first spectrum evaluation apparatus according to an embodiment of the present invention;
3 is a view showing the configuration of a second spectrum evaluation apparatus according to another embodiment of the present invention;
4 is a view showing the operation of the second spectrum evaluation apparatus according to another embodiment of the present invention.
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Cited By (3)
Publication number | Priority date | Publication date | Assignee | Title |
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CN108364661A (en) * | 2017-12-15 | 2018-08-03 | 海尔优家智能科技(北京)有限公司 | Visualize speech performance appraisal procedure, device, computer equipment and storage medium |
KR20220074415A (en) | 2020-11-27 | 2022-06-03 | 주식회사 에스알유니버스 | Method and apparatus for automated evaluation of synthetic speech based on artificial intelligence |
WO2024080590A1 (en) * | 2022-10-14 | 2024-04-18 | 삼성전자주식회사 | Electronic device and method for detecting signal error |
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Cited By (4)
Publication number | Priority date | Publication date | Assignee | Title |
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CN108364661A (en) * | 2017-12-15 | 2018-08-03 | 海尔优家智能科技(北京)有限公司 | Visualize speech performance appraisal procedure, device, computer equipment and storage medium |
CN108364661B (en) * | 2017-12-15 | 2020-11-24 | 海尔优家智能科技(北京)有限公司 | Visual voice performance evaluation method and device, computer equipment and storage medium |
KR20220074415A (en) | 2020-11-27 | 2022-06-03 | 주식회사 에스알유니버스 | Method and apparatus for automated evaluation of synthetic speech based on artificial intelligence |
WO2024080590A1 (en) * | 2022-10-14 | 2024-04-18 | 삼성전자주식회사 | Electronic device and method for detecting signal error |
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