WO2009110751A2 - 오디오 신호 처리 방법 및 장치 - Google Patents
오디오 신호 처리 방법 및 장치 Download PDFInfo
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- WO2009110751A2 WO2009110751A2 PCT/KR2009/001081 KR2009001081W WO2009110751A2 WO 2009110751 A2 WO2009110751 A2 WO 2009110751A2 KR 2009001081 W KR2009001081 W KR 2009001081W WO 2009110751 A2 WO2009110751 A2 WO 2009110751A2
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- G—PHYSICS
- G11—INFORMATION STORAGE
- G11B—INFORMATION STORAGE BASED ON RELATIVE MOVEMENT BETWEEN RECORD CARRIER AND TRANSDUCER
- G11B20/00—Signal processing not specific to the method of recording or reproducing; Circuits therefor
- G11B20/00007—Time or data compression or expansion
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; 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
- G10L19/20—Vocoders using multiple modes using sound class specific coding, hybrid encoders or object based coding
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- G—PHYSICS
- G11—INFORMATION STORAGE
- G11B—INFORMATION STORAGE BASED ON RELATIVE MOVEMENT BETWEEN RECORD CARRIER AND TRANSDUCER
- G11B20/00—Signal processing not specific to the method of recording or reproducing; Circuits therefor
- G11B20/00007—Time or data compression or expansion
- G11B2020/00014—Time or data compression or expansion the compressed signal being an audio signal
Definitions
- the present invention relates to an audio signal processing method and apparatus capable of efficiently encoding and decoding all kinds of various types of audio signals.
- perceptual audio coder optimized for music is a method of reducing the amount of information in the encoding process by using a masking principle, which is a human listening psychoacoustic theory on the frequency axis.
- a linear prediction based coder optimized for speech is a method of reducing the amount of information by modeling speech utterance on the time axis.
- An object of the present invention is to provide an audio signal processing method and apparatus capable of compressing and restoring various kinds of audio signals with higher efficiency.
- An embodiment of an audio signal processing method of the present invention may include identifying whether a coding type of an audio signal is a music signal coding type by using first type information, and when the coding type of the audio signal is not a music signal coding type, Identifying whether a coding type of the audio signal is a voice signal coding type or a mixed signal coding type using second type information, and when the coding type of the audio signal is a mixed signal coding type, spectral from the audio signal Extracting data and linear prediction coefficients, generating reverse signals by inverse frequency transforming the spectral data, and performing linear prediction coding on the linear prediction coefficients and the residual signals to restore an audio signal And using the extension base signal and the band extension information which are partial regions of the reconstructed audio signal. And a step of restoring the high frequency region signal.
- an embodiment of an audio processing apparatus of the present invention includes a demultiplexer for extracting first type information, second type information, and band extension information from a bitstream, and a coding type of an audio signal using the first type information. If the coding type of the audio signal is not a music signal coding type, identifying whether the coding type of the audio signal is a voice signal coding type or a mixed signal coding type using second type information. Next, a decoder determiner for determining a decoding method, an information extractor for extracting spectral data and linear prediction coefficients from the audio signal, and the spectral data when the coding type of the audio signal is a mixed signal coding type.
- a frequency converter for generating a residual signal for linear prediction by frequency conversion, the linear prediction coefficient and the register And a linear prediction unit for linearly predicting and coding the dual signal to restore the audio signal, and a bandwidth extension decoding unit for reconstructing the high frequency region signal by using the extension base signal and the band extension information which are partial regions of the reconstructed audio signal.
- the audio signal may include a plurality of subframes, and the second type information may exist in units of the subframes.
- the bandwidth of the high frequency region signal may be generated not equal to the bandwidth of the extension base signal, wherein the band extension information is a filter range applied to the reconstructed audio signal, a start frequency of the extension base signal, and the It may include information of any one or more of the end frequency.
- the audio signal is a frequency domain signal. If the coding type of the audio signal is a voice signal coding type, the audio signal is a time domain signal, and the coding type of the audio signal. If the mixed signal coding type, the audio signal may be an MDCT domain signal.
- the linear prediction coefficient modes may be extracted, and the linear prediction coefficients having a variable bit size corresponding to the extracted modes may be extracted.
- an audio coding scheme suitable for each audio signal characteristic provides a more efficient compression and reconstruction of an audio signal.
- FIG. 1 is a block diagram illustrating an audio encoding apparatus according to an embodiment of the present invention.
- FIG. 2 is a block diagram illustrating an audio encoding apparatus according to another embodiment of the present invention.
- FIG. 3 is a block diagram showing a detailed configuration of the bandwidth preprocessor 150 according to an embodiment of the present invention.
- FIG. 4 is a flowchart illustrating a method of encoding an audio signal using audio type information according to an embodiment of the present invention.
- FIG 5 shows an example of an audio bitstream structure encoded by the present invention.
- FIG. 6 is a block diagram illustrating an audio decoding apparatus according to an embodiment of the present invention.
- FIG. 7 is a block diagram illustrating an audio decoding apparatus according to another embodiment of the present invention.
- FIG. 8 is a block diagram showing a detailed configuration of the bandwidth expansion unit 250 according to an embodiment of the present invention.
- FIG. 9 illustrates the configuration of a product in which an audio decoding apparatus according to an embodiment of the present invention is implemented.
- FIG. 10 illustrates an example of a relationship between products in which an audio decoding apparatus according to an embodiment of the present invention is implemented.
- FIG. 11 is a flowchart illustrating an audio decoding method according to an embodiment of the present invention.
- 'Coding' may be interpreted as encoding or decoding in some cases, and information is a term including all values, parameters, coefficients, elements, and the like. .
- the term 'audio signal' in the present invention refers to all signals that can be visually identified during reproduction as a concept that is distinguished from a video signal.
- the audio signal may be, for example, a speech signal or a similar signal centered on human pronunciation (hereinafter, referred to as a 'speech signal'), a machine sound and a music centered sound. (music) signal or a similar signal (hereinafter referred to as a 'music signal'), and a 'mixed signal' in which the voice signal and the music signal are mixed.
- An object of the present invention is to provide a method and apparatus for encoding and decoding audio signals classified into three types according to characteristics of each signal.
- the classification of the audio signal is only a criterion classified for explanation of the present invention, and even if the audio signal is classified by another method, it is obvious that the technical idea of the present invention is equally applicable. .
- FIG. 1 is a block diagram illustrating an audio encoding apparatus according to an embodiment of the present invention.
- FIG. 1 illustrates a process of classifying an input audio signal according to a predetermined criterion and selecting and encoding an audio encoding method suitable for each classified audio signal.
- a signal classification unit (Sound Activity Detector) 100 for analyzing a characteristic of an input audio signal and classifying it into any one type of a voice signal, a music signal, or a mixed signal of voice and music, and the signal
- the linear prediction modeling unit 110 for encoding a speech signal among the signal types determined by the classification unit 100, the psychoacoustic modeling unit 120 for encoding a music signal, and a mixed signal for encoding a mixed signal of speech and music
- the modeling unit 130 is included.
- the switching unit 101 may select a coding scheme suitable for this.
- the switching unit 101 will be described later in detail with reference to audio signal coding type information generated by the signal classification unit 100 (for example, first type information and second type information, and FIGS. 2 and 3). Is operated as a control signal.
- the mixed signal modeling unit 130 may include a linear prediction unit 131, a residual signal extraction unit 132, and a frequency converter 133.
- a linear prediction unit 131 for example, a linear prediction unit 131
- a residual signal extraction unit 132 for example, first type information and second type information, and FIGS. 2 and 3
- a frequency converter 133 for example, a frequency converter 133.
- the signal classification unit 100 generates a control signal for classifying an input audio signal type and selecting an audio encoding scheme suitable for the input audio signal. For example, the signal classifying unit 100 classifies whether an input audio signal is a music signal, a voice signal, or a mixed signal in which both voice and music signals are mixed. That is, the reason for classifying the type of the audio signal input as described above is to select an optimal coding method among audio coding methods to be described later for each audio signal type. As a result, the signal classification unit 100 may correspond to a process of analyzing an input audio signal and selecting an optimal audio coding scheme.
- the signal classification unit 100 analyzes an input audio signal to generate audio coding type information, and the generated audio coding type information is used as a criterion for selecting a coding scheme, as well as the final audio signal. It is included in the form of a bitstream and transmitted to the decoding apparatus or the receiving apparatus. A decoding method and apparatus using the audio coding type information will be described later in detail with reference to FIGS. 6 to 8 and 11.
- the audio coding type information generated by the signal classification unit 100 may include, for example, first type information and second type information. This will be described later in FIGS. 4 and 5.
- the signal classification unit 100 determines the audio signal type according to the characteristics of the input audio signal. For example, if the input audio signal is a signal that is better modeled with a specific coefficient and a residual signal, it is determined as a voice signal, whereas the signal is a signal that cannot be well modeled with a specific coefficient and a residual signal. In this case, it is determined as a music signal. In addition, when it is difficult to determine any one of the voice signal and the music signal, it may be determined as a mixed signal. Specifically, for example, when the signal is modeled as a specific coefficient and a residual signal, when the energy level ratio of the residual signal to the energy level of the signal is smaller than a predetermined reference value, the signal may be modeled.
- the signal may be determined as a signal well modeled by linear prediction that predicts the current signal from the past signal, and thus, may be determined as a music signal.
- the input signal may be encoded using a speech encoder optimized for the speech signal.
- the linear prediction modeling is performed using a coding scheme suitable for the speech signal.
- the unit 110 is used.
- the linear prediction modeling unit 110 has various methods, for example, an Algebraic Code Excited Linear Prediction (ACELP) coding method or an Adaptive Multi-Rate (AMR) coding and an Adaptive Multi-Rate Wideband (AMR-WB) coding. The method can be applied.
- ACELP Algebraic Code Excited Linear Prediction
- AMR Adaptive Multi-Rate
- AMR-WB Adaptive Multi-Rate Wideband
- the linear prediction modeling unit 110 may linearly predict and encode an input audio signal in units of frames, and extract and quantize prediction coefficients for each frame.
- a method of extracting prediction coefficients using a 'Levinson-Durbin algorithm' is widely used.
- linear prediction is performed for each frame.
- the modeling method can be applied.
- the input audio signal when the input audio signal is classified as a music signal by the signal classification unit 100, the input signal may be encoded using a music encoder optimized for the music signal.
- the psychoacoustic modeling unit 120 is used as a suitable coding scheme.
- the psychoacoustic modeling unit 120 is configured based on a perceptual audio coder.
- the signal classification unit 100 classifies an input audio signal into a mixed signal in which voice and music are mixed
- the input signal may be encoded using an encoder optimized for the mixed signal.
- the mixed signal modeling unit 130 is used as an encoding method suitable for the mixed signal.
- the mixed signal modeling unit 130 may code the mixed prediction method by modifying the aforementioned linear prediction modeling method and the psychoacoustic modeling method. That is, the mixed signal modeling unit 130 performs linear predictive coding on an input signal and then obtains a residual signal that is a difference between the linearly predicted result signal and the original signal, and the residual signal is frequency transform coded. The way you do it.
- FIG. 1 illustrates an example in which the mixed signal modeling unit 130 includes a linear predictor 131, a residual signal extractor 132, and a frequency converter 133.
- the linear prediction unit 131 performs linear prediction analysis on the input signal to extract linear prediction coefficients representing the characteristics of the signal, and uses the linear prediction coefficients extracted by the residual signal extracting unit 132.
- a residual signal from which duplicate components are removed from the input signal is extracted.
- the residual signal may have a shape such as white noise since redundancy is removed.
- the linear predictor 131 may linearly encode an input audio signal in units of frames, and extract and quantize prediction coefficients for each frame. That is, for example, when the input audio signal is composed of a plurality of frames or a plurality of super frames having a plurality of frames as one unit, linear prediction is performed for each frame.
- the modeling method can be applied.
- the residual signal extractor 132 receives a residual signal coded through the linear predictor 131 and an original audio signal that has passed through the signal classifier 100, and is a residual signal that is a difference signal between the two signals. Extract the signal.
- the frequency converter 133 calculates a masking threshold value or a signal-to-mask ratio (SMR) of the residual signal by frequency domain converting the input residual signal using a method such as MDCT. Code dual signals.
- the frequency converter 133 may code a signal of residual audio tendency using TCX.
- the linear prediction modeling unit 110 and the linear prediction unit 131 linearly analyze the input audio signal to extract a linear prediction coefficient (LPC) reflecting audio characteristics.
- LPC linear prediction coefficient
- a method using a variable bit may be considered.
- the signal classification unit 100 generates and generates coding type information of an audio signal into two types of information, and includes the same in a bitstream to transmit to the decoding apparatus.
- audio coding type information according to the present invention will be described in detail with reference to FIGS. 4 and 5.
- FIG. 4 is a flowchart illustrating a method of encoding an audio signal using coding type information of the audio signal according to an embodiment of the present invention.
- the present invention proposes a method of expressing a type of an audio signal by dividing it into first type information and second type information. That is, for example, when the input audio signal is determined to be a music signal (S100), the signal classification unit 100 may switch the switching unit (eg, to select an appropriate coding scheme (for example, the psychoacoustic modeling method of FIG. 2). By controlling 101, encoding is performed according to the selected encoding scheme (S110). Subsequently, the control information is configured as first type information and included in the encoded audio bitstream for transmission. In relation to this, the first type information eventually serves as coding identification information indicating that the coding type of the audio signal is a music signal coding time, which is utilized in decoding the audio signal in the decoding method and apparatus.
- S100 music signal
- the signal classification unit 100 may switch the switching unit (eg, to select an appropriate coding scheme (for example, the psychoacoustic modeling method of FIG. 2).
- the control information is configured as first type information and included in the
- the signal classifying unit 100 may select the coding scheme (for example, the linear prediction modeling method of FIG. 1) suitable for the switching unit 101 (S120). By controlling, the encoding is performed according to the selected encoding scheme (S130). In addition, if it is determined that the input audio signal is a mixed signal (S120), the signal classifying unit 100 sets the switching unit 101 to select an appropriate coding scheme (for example, the mixed signal modeling method of FIG. 1). By controlling, encoding is performed according to the selected encoding scheme (S140).
- the coding scheme for example, the linear prediction modeling method of FIG. 1
- control information indicating either the speech signal coding type or the mixed signal coding type is configured as second type information, and is included in the encoded audio bitstream together with the first type information for transmission.
- the second type information eventually serves as a coding identification information indicating that a coding type of an audio signal is either a voice signal coding type or a mixed signal coding type, which is the first type described above in the decoding method and apparatus. It is used to decode the audio signal along with the information.
- the first type information and the second type information may be transmitted either only the first type information or both of the first type information and the second type information according to characteristics of the input audio signal.
- the input audio signal coding type is a music signal coding type
- only the first type information may be included in the bitstream and transmitted, and the second type information may not be included in the bitstream (Fig. 5 (a)). That is, since the second type information is included in the bitstream only when the input audio signal coding type is the voice signal coding type or the mixed signal coding type, the second type information prevents unnecessary bits to represent the coding type of the audio signal. It works.
- the first type information is a music signal coding type.
- the first type information is used as a voice signal coding type or a mixed signal coding type. It is obvious that it can be used as information to indicate. That is, according to the coding environment to which the present invention is applied, by using the audio coding type that is frequently generated as the first type information, the number of bits of the overall bitstream is reduced.
- FIG 5 shows an example of an audio bitstream structure encoded by the present invention.
- FIG. 5A illustrates a case where an input audio signal corresponds to a music signal, and includes only the first type information 301 in the bitstream and does not include the second type information.
- the bitstream includes audio data coded with a coding type corresponding to the first type information 301 (for example, the AAC bitstream 302).
- FIG. 5B illustrates a case where the input audio signal corresponds to a voice signal, and includes both the first type information 311 and the second type information 312 in the bitstream.
- the bitstream includes audio data coded with a coding type corresponding to the second type information 312 (for example, the AMR bitstream 313).
- FIG. 5C illustrates a case where an input audio signal corresponds to a mixed signal, and includes both first type information 321 and second type information 322 in the bitstream.
- the bitstream includes audio data coded with a coding type corresponding to the second type information 322 (for example, the AAC bitstream 313 to which TXC is applied).
- 5 (a) to 5 (c) show only information included in an audio bitstream encoded by the present invention as an example, and it will be apparent that various applications are possible within the scope of the present invention.
- the present invention adds information for identifying AMR and AAC as examples of coding schemes, but various coding schemes are applicable, and coding identification information for identifying them may also be used in various ways.
- 5 (a) to (c) of the present invention are applicable to one super frame, unit frame, or subframe. That is, it is possible to provide audio signal coding type information for each preset frame unit.
- a frequency bandwidth expansion process and a channel number change process are performed. It may be done.
- the bandwidth preprocessor may generate high frequency components using low frequency components, and as an example of the bandwidth preprocessor 150, the modified and improved SBR (Spectral Band Replication) and HBE (High Band Extension) are available.
- SBR Spectral Band Replication
- HBE High Band Extension
- the channel number changing process reduces the bit allocation by encoding channel information of the audio signal into additional information.
- An example of the channel number changing process may include a downmix channel generator (FIGS. 2 and 140).
- the downmix channel generator 140 may be, for example, applied to a parametric stereo (PS) scheme.
- PS is a technique of coding a stereo signal and downmixes the stereo signal to a mono signal.
- the downmix channel generator 140 downmixes the input multi-channel audio signal to generate spatial information related to the downmix signal and the restoration of the downmixed signal.
- the signal when a 48 kHz stereo signal is transmitted using the SBR (Spectral Band Replication) and the PS (Parametric Stereo), the signal is passed through the SBR / PS, and a 24 kHz, mono signal signal is left, which may be encoded through an encoder.
- the reason why the input signal of the encoder is 24 kHz is because the high frequency component is coded through the SBR and downsampled to half of the existing frequency while passing through the SBR, and the reason for the mono signal is that stereo audio is extracted as a parameter through the mono This is because it changes to the sum of signal and additional audio.
- FIG. 2 illustrates an encoding apparatus including the above-described downmix channel generator 140 and the bandwidth preprocessor 150 as an encoding preprocessing process.
- the operations of the linear prediction modeling unit 110, the psychoacoustic modeling unit 120, the mixed signal modeling unit 130, and the switching unit 101 are the same.
- the signal classification unit 100 has the same content of generating the second type information under the first type information, but additionally, controls to control the operations of the downmix channel generator 140 and the bandwidth preprocessor 150. Will generate a signal.
- the downmix channel generator 140 and the bandwidth preprocessor 150 are analyzed as an encoding preprocessing process by analyzing an input audio signal to determine an audio signal type and analyzing the number of channels and a frequency bandwidth in the audio signal. ) Generates control signals 100b and 100c for controlling the operation and the operation range.
- FIG. 3 is a block diagram showing a detailed configuration of the bandwidth preprocessor 150 according to an embodiment of the present invention.
- the bandwidth preprocessor 150 for band extension includes a high frequency region remover 151, an extension information generator 152, and a spatial information inserter 153.
- the high frequency region remover 151 receives a downmix signal and spatial information from the downmix channel generator 140.
- the high frequency region removing unit 151 generates reconstruction information including the start frequency and the end frequency of the low frequency downmix signal and the extended basic signal (to be described later) from which the high frequency signal corresponding to the high frequency region of the downmix signal is removed. do.
- the reconstruction information may be determined based on characteristics of the input signal.
- the start frequency of the high frequency signal is half of the total bandwidth of the input signal.
- the reconstruction information may determine the start frequency as a frequency corresponding to less than or equal to half of the entire bandwidth according to the characteristics of the input signal.
- the reconstruction information is the start frequency at the end of the bandwidth. It may represent a frequency located in.
- the reconstruction information may be determined using at least one of a signal size, a length of a segment used in coding, and a type of a source, but is not limited thereto.
- the extension information generator 152 uses the downmix signal and the spatial information generated by the downmix channel generator 14 to generate extension information for determining an extension base signal to be used for decoding.
- the extended base signal is a frequency signal of the downmix signal used to restore the high frequency signal of the downmix signal removed by the high frequency region remover 151 during decoding, and may be a signal of a low frequency signal or a low frequency signal.
- the downmix signal may be bandpass filtered to separate the low frequency signal into a low frequency band region and a middle frequency band region. In this case, only the low frequency band region may be used.
- Extension information can be generated.
- a boundary frequency for dividing the low frequency band region and the day frequency frequency band region may be set to a fixed value.
- the signal classification unit 100 analyzes the ratio of voice and music to the mixed signal. The information may be variably determined for each frame.
- the extension information may correspond to information about a downmix signal that is not removed by the high frequency region removing unit 151, but is not limited thereto.
- the extension information may be information about some signals of the downmix signal.
- the extension information when the extension information is information on some signals of the downmix signal, the extension information may include a start frequency and an end frequency of the extension base signal, and further include a range of a filter applied to the frequency signal of the downmix signal. It may include.
- the spatial information insertion unit 153 inserts the restoration information generated by the high frequency region removing unit 121 and the extension information generated by the extension information generation unit 122 into the spatial information generated by the downmix channel generator 140. The generated new spatial information.
- FIG. 6 is a diagram illustrating a decoding apparatus according to an embodiment of the present invention.
- the decoding apparatus may restore a signal from an input bitstream by performing an inverse process of an encoding process performed in the encoding apparatus described with reference to FIG. 1.
- the decoding apparatus may include a demultiplexer 210, a decoder determiner 220, a decoder 230, and a synthesizer 240.
- the decoder 230 may include a plurality of decoders 231, 232, and 233 which perform decoding by different methods, which are operated under the control of the decoder determiner 220.
- the decoder 230 may include a linear prediction decoder 231, a psychoacoustic decoder 232, and a mixed signal decoder 233.
- the mixed signal decoder 233 may include an information extractor 234, a frequency converter 235, and a linear predictor 236.
- the demultiplexer 210 extracts a plurality of encoded signals and additional information for decoding the signals from an input bitstream. For example, the first type information and the second type information (included only when necessary) included in the aforementioned bitstream are extracted and transmitted to the decoder determiner 220.
- the decoder determiner 220 determines one of decoding methods in the decoders 231, 232, and 233 from the first type information and the second type information (which are included only when necessary). However, the decoder determiner 220 may determine the decoding method using the additional information extracted from the bitstream. However, when there is no additional information in the bitstream, the decoder determining unit 220 may determine the decoding method by an independent determination method. have. The determination method may utilize the features of the signal classification unit (FIGS. 1 and 100) described above.
- the linear prediction decoder 231 in the decoder 230 is capable of decoding an audio signal of a voice signal type.
- the psychoacoustic decoder 232 decodes an audio signal of a music signal type.
- the mixed signal decoder 233 decodes an audio signal of a mixed type of voice and music.
- the mixed signal decoder 233 extracts spectral data and linear predictive coefficients from an audio signal, and inverse-frequency transforms the spectral data to generate a residual signal for linear prediction.
- a linear predictor 236 for linearly predicting and coding the linear predictive coefficient and the residual signal to generate an output signal.
- the decoded signals are synthesized by the combiner 240 and restored to an audio signal before being encoded.
- the post-processing process is performed to increase bandwidth and change the number of channels for the decoded audio signal using any one of the linear prediction decoder 231, the psychoacoustic decoder 232, and the mixed signal decoder 233. It means the process.
- the post-processing process may include a bandwidth extension decoding unit 250 and a plurality of channel generators 260 corresponding to the downmix channel generator 140 and the bandwidth preprocessor 150 of FIG. 2.
- the extension information generated in the aforementioned bandwidth preprocessor 150 is extracted from the bitstream by the demultiplexer 210 and used.
- Spectral data of another band (for example, a high frequency band) is generated from some or all of the spectral data from the extension information included in the audio signal bitstream.
- a block may be generated by grouping into units having similar characteristics in extending the frequency band. This is like creating an envelope region by grouping type slots (or samples) with a common envelope (or envelope characteristic).
- the bandwidth extension decoding unit 250 includes an extension base region determiner 251, a high frequency region reconstructor 252, and a bandwidth expander 253.
- the extended base area determiner 251 determines an extended base area among the received downmix signals based on the received extended information, and generates an extended base signal as a result.
- the downmix signal may be a signal represented in a frequency domain, and the extension base signal refers to a part of a frequency domain of the downmix signal in a frequency domain.
- the extension information may be used to determine the extension base signal, and may be a range of a filter for filtering a start frequency and an end frequency of the extension base signal, or a portion of the downmix signal.
- the high frequency region recovery unit 252 receives a downmix signal and extension information, and receives the extension base signal. Thereafter, the high frequency region signal of the downmix signal removed by the encoding end may be restored using the extension base signal and the extension information. In this case, reconstruction information received from the encoding apparatus may be further used.
- the high frequency region signal may not be included in the downmix signal but may be a high frequency region signal included in the original signal.
- the high frequency region signal may not be an integer multiple of the downmix signal, and the bandwidth of the high frequency region signal may not be the same as the bandwidth of the extension base signal.
- the apparatus and method for extending bandwidth according to an embodiment of the present invention does not use all of the downmix signals from which a high frequency region has been removed from an encoding terminal as the extension base signal, and a signal corresponding to some frequency regions of the downmix signals. By using this, it is possible to use a bandwidth extension technique even when the high frequency region to be restored is not an integer multiple of the downmix signal.
- the high frequency region recovery unit 252 may further include a time extension downmix signal generator (not shown) and a frequency signal extension unit (not shown).
- the time extension downmix signal generator may extend the downmix signal to the time domain by applying the extension information to the extension base signal.
- the frequency signal extension unit may expand the signal in the frequency domain of the downmix signal by reducing the number of samples of the time extension downmix signal.
- the bandwidth extension unit 253 combines the downmix signal and the high frequency region signal when the high frequency region restoration unit 252 includes only the restored high frequency region signal and does not include the low frequency region signal. Generate this extended extended downmix signal.
- the high frequency region signal may not be an integer multiple of the downmix signal. Accordingly, the bandwidth extension technique according to an embodiment of the present invention may be used for upsampling into a signal that is not a multiple relationship.
- the extended downmix signal finally generated by the bandwidth extension 253 is input to the multiple channel generator 260 and converted into multiple channels.
- the demultiplexer 210 extracts first type information and second type information (if necessary) from the input bitstream. In addition, the demultiplexer 210 extracts information (eg, band extension information, reconstruction information, etc.) for post-processing.
- the decoder determiner 220 first determines a coding type of the received audio signal by using first type information among the extracted information (S1000). If the coding type of the received audio signal is a music signal coding type, the psychoacoustic decoder 232 in the decoder 230 may be utilized, and determined by each frame or subframe, determined by the first type information. The coding scheme to be applied is determined, and then decoding is performed by applying a coding scheme suitable for this (S1100).
- the decoder determiner 220 first uses the second type information. It is determined whether the coding type of the received audio signal is a voice signal coding type or a mixed signal coding type (S1200).
- the linear prediction decoder 231 in the decoder 230 may be used, and each frame or subframe may be utilized by using coding identification information extracted from the bitstream. A coding scheme applied to each star is determined, and then decoding is performed by applying a suitable coding scheme (S1300).
- the mixed signal decoder 233 in the decoder 230 may be utilized, and each frame or subframe may be determined by the second type information.
- the coding scheme to be applied is determined, and then decoding is performed by applying a coding scheme suitable for this (S1400).
- the bandwidth extension decoding unit 250 is a post-processing step of decoding an audio signal using any one of the linear prediction decoder 231, the psychoacoustic decoder 232, and the mixed signal decoder 233.
- a frequency band extension process may be performed at step S1500.
- the bandwidth extension decoding unit 250 decodes the band extension information extracted from the audio signal bitstream to obtain spectral data of another band (eg, a high frequency band) from some or all of the spectral data. Will be created.
- a process of generating a plurality of channels in the multi-channel generator 260 may be performed with respect to the audio signal in which the bandwidth is generated after the band extension process (S1600).
- FIG. 9 is a diagram illustrating a configuration of a product on which a decoding apparatus according to an embodiment of the present invention is implemented.
- FIG. 10 is a diagram illustrating a relationship between products in which a decoding apparatus according to an embodiment of the present invention is implemented.
- the wired / wireless communication unit 910 receives a bitstream through a wired / wireless communication scheme.
- the wired / wireless communication unit 910 may include at least one of a wired communication unit 910A, an infrared communication unit 910B, a Bluetooth unit 910C, and a wireless LAN communication unit 910D.
- the user authentication unit 920 performs user authentication by inputting user information, and includes one or more of a fingerprint recognition unit 920A, an iris recognition unit 920B, a face recognition unit 920C, and a voice recognition unit 920D.
- the fingerprint, iris information, facial contour information, and voice information may be input, converted into user information, and the user authentication may be performed by determining whether the user information matches the existing registered user data. .
- the input unit 930 is an input device for a user to input various types of commands, and may include one or more of a keypad unit 930A, a touch pad unit 930B, and a remote controller unit 930C. It is not limited.
- the signal decoding unit 950 analyzes signal characteristics using the received bitstream and frame type information, and decodes the signal using a decoding unit corresponding to the corresponding signal characteristics to generate an output signal.
- the controller 950 receives input signals from the input apparatuses and controls all processes of the signal decoding unit 940 and the output unit 960.
- the output unit 960 is a component in which an output signal generated by the signal decoding unit 940 is output, and may include a speaker unit 960A and a display unit 960B. When the output signal is an audio signal, the output signal is output to the speaker, and when the output signal is a video signal, the output signal is output through the display.
- FIG. 10 illustrates a relationship between a terminal and a server corresponding to the product illustrated in FIG. 9.
- the first terminal 1001 and the second terminal 1002 are each terminals. It can be seen that they can communicate the data to the bitstream in both directions through the wired or wireless communication unit.
- the server 1003 and the first terminal 1001 may also perform wired or wireless communication with each other.
- the audio signal processing method according to the present invention can be stored in a computer-readable recording medium which is produced as a program for execution in a computer, and multimedia data having a data structure according to the present invention can also be stored in a computer-readable recording medium.
- the computer readable recording medium includes all kinds of storage devices in which data that can be read by a computer system is stored. Examples of computer-readable recording media include ROM, RAM, CD-ROM, magnetic tape, floppy disk, optical data storage, and the like, and may also be implemented in the form of a carrier wave (for example, transmission over the Internet). Include.
- the bitstream generated by the encoding method may be stored in a computer-readable recording medium or transmitted using a wired / wireless communication network.
Abstract
Description
Claims (15)
- 오디오 복호화기를 포함하는 오디오 신호 처리 장치내에서,제1 타입정보를 이용하여 오디오 신호의 코딩타입이 음악신호 코딩타입인지 를 식별하는 단계;상기 오디오 신호의 코딩타입이 음악신호 코딩타입이 아닌 경우, 제 2 타입정보를 이용하여 상기 오디오 신호의 코딩타입이 음성신호 코딩타입인지, 혼합신호 코딩타입인지를 식별하는 단계;상기 오디오 신호의 코딩타입이 혼합신호 코딩타입인 경우, 상기 오디오 신호로부터 스펙트럴 데이터와 선형예측 계수를 추출하는 단계;상기 스펙트럴 데이터를 역 주파수 변환하여 선형 예측에 대한 레지듀얼 신호를 생성하는 단계;상기 선형예측 계수 및 상기 레지듀얼 신호를 선형 예측 코딩하여, 오디오 신호를 복원하는 단계; 및상기 복원된 오디오 신호의 일부 영역인 확장 기초 신호 및 대역 확장 정보를 이용하여 고주파 영역 신호를 복원하는 단계를 포함하는 것을 특징으로 하는 오디오 신호 처리 방법.
- 제 1항에 있어서,상기 오디오 신호는 복수의 서브 프레임으로 구성되며, 상기 제 2 타입 정보는 상기 서브 프레임 단위로 존재하는 것을 특징으로 하는 오디오 신호 처리 방법.
- 제 1항에 있어서,상기 고주파 영역 신호의 대역폭은 상기 확장 기초 신호의 대역폭과 동일하지 아니한 것을 특징으로 하는 오디오 신호 처리 방법.
- 제 1항에 있어서,상기 대역 확장 정보는 상기 복원된 오디오 신호에 적용되는 필터 범위, 상기 확장 기초 신호의 시작 주파수 및 종료 주파수 중 어느 하나 이상의 정보를 포함하는 것을 특징으로 하는 오디오 신호 처리 방법.
- 제 1항에 있어서,상기 오디오 신호의 코딩타입이 음악신호 코딩타입이면 상기 오디오 신호는 주파수 도메인 신호이고, 상기 오디오 신호의 코딩타입이 음성신호 코딩타입이면 상기 오디오 신호는 타임 도메인 신호이며, 상기 오디오 신호의 코딩타입이 혼합신호 코딩타입이면 상기오디오 신호는 MDCT 도메인 신호인 것을 특징으로 하는 오디오 신호 처리 방법.
- 제 1항에 있어서,상기 선형 예측 계수를 추출하는 단계는,선형 예측 계수 모드를 추출하고, 상기 추출된 모드에 해당하는 가변비트수 크기의 선형 예측 계수를 추출하는 것을 특징으로 하는 오디오 신호 처리 방법.
- 비트스트림으로부터 제1 타입정보, 제2 타입정보, 대역 확장 정보를 추출하는 디멀티플렉서;상기 제1 타입정보를 이용하여 오디오 신호의 코딩타입이 음악신호 코딩타입인지를 식별하고, 상기 오디오 신호의 코딩타입이 음악신호 코딩타입이 아닌 경우, 제 2 타입정보를 이용하여 상기 오디오 신호의 코딩타입이 음성신호 코딩타입인지 또는 혼합신호 코딩타입인지를 식별한 후, 복호화 방식을 결정하는 복호화기 결정부;상기 오디오 신호의 코딩타입이 혼합신호 코딩타입인 경우, 상기 오디오 신호로부터 스펙트럴 데이터와 선형예측 계수를 추출하는 정보추출부;상기 스펙트럴 데이터를 역 주파수 변환하여 선형 예측에 대한 레지듀얼 신호를 생성하는 주파수 변환부;상기 선형예측 계수 및 상기 레지듀얼 신호를 선형 예측 코딩하여, 오디오 신호를 복원하는 선형 예측부; 및상기 복원된 오디오 신호의 일부 영역인 확장 기초 신호 및 대역 확장 정보를 이용하여 고주파 영역 신호를 복원하는 대역폭 확장 디코딩부를 포함하는 것을 특징으로 하는 오디오 신호 처리 장치.
- 제 7항에 있어서,상기 오디오 신호는 복수의 서브 프레임으로 구성되며, 상기 제 2 타입 정보는 상기 서브 프레임 단위로 존재하는 것을 특징으로 하는 오디오 신호 처리 장치.
- 제 7항에 있어서,상기 고주파 영역 신호의 대역폭은 상기 확장 기초 신호의 대역폭과 동일하지 아니한 것을 특징으로 하는 오디오 신호 처리 장치.
- 제 7항에 있어서,상기 대역 확장 정보는 상기 복원된 오디오 신호에 적용되는 필터 범위, 상기 확장 기초 신호의 시작 주파수 및 상기 종료 주파수 중 어느 하나 이상의 정보를 포함하는 것을 특징으로 하는 오디오 신호 처리 장치.
- 제 7항에 있어서,상기 오디오 신호의 코딩타입이 음악신호 코딩타입이면 상기 오디오 신호는 주파수 도메인 신호이고, 상기 오디오 신호의 코딩타입이 음성신호 코딩타입이면 상기 오디오 신호는 타임 도메인 신호이며, 상기 오디오 신호의 코딩타입이 혼합신호 코딩타입이면 상기오디오 신호는 MDCT 도메인 신호인 것을 특징으로 하는 오디오 신호 처리 장치.
- 제 1항에 있어서,상기 선형 예측 계수를 추출하는 단계는,선형 예측 계수 모드를 추출하고, 상기 추출된 모드에 해당하는 가변비트수 크기의 선형 예측 계수를 추출하는 것을 특징으로 하는 오디오 신호 처리 장치.
- 오디오 신호를 처리하는 오디오 부호화기를 포함하는 오디오 신호 처리 장치 내에서,오디오 신호의 고주파 대역 신호를 제거하고, 상기 고주파 대역 신호를 복원하기 위한 대역 확장 정보를 생성하는 단계;상기 오디오 신호의 코딩타입을 결정하는 단계;상기 오디오 신호가 음악신호이면, 음악신호 코딩타입으로 코딩됨을 나타내는 제1 타입정보를 생성하는 단계;상기 오디오 신호가 음악신호가 아니면, 음성신호 코딩타입과 혼합신호 코딩 타입 중 어느 하나로 코딩됨을 나타내는 제2 타입정보를 생성하는 단계;상기 오디오 신호의 코딩타입이 혼합신호 코딩타입인 경우, 상기 오디오 신호를 선형 예측 코딩하여 선형예측 계수를 생성하는 단계;상기 선형 예측 코딩에 대한 레지듀얼 신호를 생성하는 단계;상기 레지듀얼 신호를 주파수 변환하여 스펙트럴 계수를 생성하는 단계; 및상기 제 1 타입정보, 상기 제 2 타입정보, 상기 선형예측 계수 및 레지듀얼 신호를 포함하는 오디오 비트스트림을 생성하는 단계를 포함하는 오디오 신호 처리 방법.
- 오디오 신호의 고주파 대역 신호를 제거하고, 상기 고주파 대역 신호를 복원하기 위한 대역 확장 정보를 생성하는 대역폭 전처리부;입력 오디오 신호의 코딩타입을 결정하되, 상기 오디오 신호가 음악신호이면, 음악신호 코딩타입으로 코딩됨을 나타내는 제1 타입정보를 생성하고, 상기 오디오 신호가 음악신호가 아니면, 음성신호 코딩타입과 혼합신호 코딩 타입 중 어느 하나로 코딩됨을 나타내는 제2 타입정보를 생성하는 신호분류부;상기 오디오 신호의 코딩타입이 혼합신호 코딩타입인 경우, 상기 오디오 신호를 선형 예측 코딩하여 선형예측 계수를 생성하는 선형예측 모델링부;상기 선형 예측에 대한 레지듀얼 신호를 생성하는 레지듀얼 신호추출부; 및상기 레지듀얼 신호를 주파수 변환하여 스펙트럴 계수를 생성하는 주파수 변환부를 포함하는 오디오 신호 처리 장치.
- 제 11항에 있어서,상기 오디오 신호는 복수의 서브 프레임으로 구성되며, 상기 제2 타입정보는 상기 서브 프레임별로 생성되는 것을 특징으로 하는 오디오 신호 처리 장치.
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ES09717694.5T ES2464722T3 (es) | 2008-03-04 | 2009-03-04 | Método y aparato para procesar una señal de audio |
CA2717584A CA2717584C (en) | 2008-03-04 | 2009-03-04 | Method and apparatus for processing an audio signal |
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AU2009220341A AU2009220341B2 (en) | 2008-03-04 | 2009-03-04 | Method and apparatus for processing an audio signal |
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CA2717584A1 (en) | 2009-09-11 |
WO2009110751A3 (ko) | 2009-10-29 |
KR20100134623A (ko) | 2010-12-23 |
CN102007534A (zh) | 2011-04-06 |
EP2259254A2 (en) | 2010-12-08 |
ES2464722T3 (es) | 2014-06-03 |
RU2010140365A (ru) | 2012-04-10 |
EP2259254B1 (en) | 2014-04-30 |
RU2452042C1 (ru) | 2012-05-27 |
JP2011514558A (ja) | 2011-05-06 |
CN102007534B (zh) | 2012-11-21 |
EP2259254A4 (en) | 2013-02-20 |
AU2009220341B2 (en) | 2011-09-22 |
US20100070272A1 (en) | 2010-03-18 |
US8135585B2 (en) | 2012-03-13 |
JP5108960B2 (ja) | 2012-12-26 |
CA2717584C (en) | 2015-05-12 |
AU2009220341A1 (en) | 2009-09-11 |
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