WO2012149843A1 - 音频信号编解码方法和设备 - Google Patents
音频信号编解码方法和设备 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
- G10L19/008—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
- G10L19/02—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
- G10L19/032—Quantisation or dequantisation of spectral components
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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
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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
- G10L19/002—Dynamic bit allocation
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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
- G10L19/02—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
- G10L19/028—Noise substitution, i.e. substituting non-tonal spectral components by noisy source
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- G—PHYSICS
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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
- G10L19/0204—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 using subband decomposition
Definitions
- Embodiments of the present invention relate to the field of audio codec technology, and more particularly, to an audio signal encoding and decoding method and apparatus.
- the input time domain signal is first transformed into the frequency domain, and the subband normalization factor, ie, the envelope information of the spectrum, is extracted in the frequency domain.
- the quantized subband normalization factor is then used to normalize the spectrum to obtain normalized spectral information. Then determine the bit allocation for each subband and quantize the normalized spectrum so that the audio signal is encoded as a quantity Entropy information and normalized spectrum information, output bit rate stream.
- the decoding end is the inverse of the encoding end.
- the encoding end cannot encode all frequency bands, and the decoding end needs to use bandwidth extension technology to recover the frequency band that is not encoded at the encoding end.
- Simultaneously encoded subbands also have more zero-frequency points due to the limitations of the quantizer, requiring noise-filled modules to improve performance.
- the decoded subband normalization factor is applied to the decoded normalized spectral coefficients to obtain the reconstructed spectral coefficients, and then inversely transformed to obtain the output time domain audio signal.
- the high-frequency harmonics are divided into some scattered bits for encoding, but the distribution on the time axis is not continuous, so that the high-frequency harmonics reconstructed during decoding are intermittent, and excessive noise is introduced. , Reconstructed audio quality is poor. Summary of the invention
- Embodiments of the present invention provide an audio signal encoding and decoding method and device, which can improve audio quality.
- an audio signal encoding method including: dividing a frequency band of an audio signal into a plurality of sub-bands, and quantizing a sub-band normalization factor of each sub-band; according to the quantized sub-band normalization factor, or according to The quantized subband normalization factor and code rate information, determine the signal bandwidth of the bit allocation; assign bits to the subbands within the determined signal bandwidth; encode the spectral coefficients of the audio signal according to the bits allocated by each subband .
- an audio signal decoding method including: obtaining a quantized subband normalization factor; according to the quantized subband normalization factor, or according to the quantized subband normalization factor and code Rate information, determining the signal bandwidth of the bit allocation; allocating bits to the subbands within the determined signal bandwidth; decoding the normalized spectrum according to the bits allocated by each subband; performing noise filling on the decoded normalized spectrum And bandwidth extension to obtain a normalized full-band spectrum; according to the normalized full-band spectrum and subband A factor that obtains the spectral coefficients of the audio signal.
- an audio signal encoding apparatus including: a quantization unit, configured to divide a frequency band of an audio signal into a plurality of sub-bands, and quantize a sub-band normalization factor of each sub-band; and a first determining unit, configured to: Determining a signal bandwidth of the bit allocation according to the subband normalization factor quantized by the quantization unit, or according to the quantized subband normalization factor and the code rate information; the first allocation unit, configured to determine the signal of the first determining unit Subband allocation bits within the bandwidth; a coding unit for encoding the spectral coefficients of the audio signal according to the bits allocated by the allocation unit for each subband.
- an audio signal decoding apparatus including: an obtaining unit, configured to obtain a quantized subband normalization factor; and a second determining unit, configured to perform normalization according to the quantized subband obtained by the obtaining unit a factor, or a signal bandwidth of the bit allocation according to the quantized subband normalization factor and code rate information; a second allocation unit, configured to allocate a bit to a subband within a signal bandwidth determined by the second determining unit; a unit, configured to decode a normalized spectrum according to a bit allocated by the second allocation unit for each sub-band; and an expansion unit configured to perform noise filling and bandwidth expansion on the decoded normalized spectrum to obtain a normalized Full-band spectrum; recovery unit for obtaining spectral coefficients of the audio signal based on the normalized full-band spectrum and sub-band normalization factor.
- the quantized subband normalization factor or the code rate information is used to determine the signal bandwidth of the bit allocation, so that the number of bits can be concentrated to effectively encode and decode the determined signal bandwidth, and the audio is improved. quality.
- FIG. 1 is a flowchart of an audio signal encoding method according to an embodiment of the present invention, without any creative labor.
- 2 is a flow chart of an audio signal decoding method according to an embodiment of the present invention.
- Figure 3 is a block diagram of an audio signal encoding apparatus in accordance with one embodiment of the present invention.
- FIG. 4 is a block diagram of an audio signal encoding apparatus according to another embodiment of the present invention.
- FIG. 5 is a block diagram of an audio signal decoding apparatus in accordance with one embodiment of the present invention.
- FIG. 6 is a block diagram of an audio signal decoding apparatus according to another embodiment of the present invention. Mode for carrying out the invention
- 1 is a flow chart of an audio signal encoding method according to an embodiment of the present invention. 101. Divide a frequency band of the audio signal into a plurality of sub-bands, and quantize the sub-band normalization factor of each sub-band.
- the MDCT transform is taken as an example for description below.
- the input audio signal is subjected to MDCT transform to obtain frequency domain coefficients.
- the MDCT transform here may include several processes of windowing, time domain aliasing, and discrete DCT transform.
- adding a sine window to the input time domain signal x ( n ) n 0,..., 2L-l L is the frame length of the signal
- n L,..., 2L-l
- the frequency domain envelope is then extracted from the MDCT coefficients and quantized.
- the entire frequency band is divided into subbands of different frequency domain resolutions, the normalization factor of each subband is extracted, and the subband normalization factor is quantized.
- a frequency band corresponding to a bandwidth of 16 kHz such as a frame length of 20 ms (640 samples)
- a frequency band corresponding to a bandwidth of 16 kHz such as a frame length of 20 ms (640 samples)
- a frame length of 20 ms 640 samples
- a child can be defined as:
- S P is the starting point of the subband
- E P is the ending point of the subband
- P is the total number of subbands.
- the normalization factor After the normalization factor is obtained, it can be quantified in the log domain to obtain the quantized subband normalization factor wnorm.
- bit-allocated signal bandwidth sfm_ 1 imi t may be defined as a partial bandwidth of the audio signal, such as a partial bandwidth at low frequencies (Tsfm_l imi t or an intermediate partial bandwidth).
- a ratio factor fact can be determined based on the code rate information, the ratio factor fact being greater than 0 and less than or equal to one.
- the smaller the code rate the smaller the ratio factor.
- different fact rates can be obtained according to Table 1 below to take the corresponding fact value.
- f ac t qx (0.5+ bitrate-value/128000), where bi trate_value is the value of the code rate such as 24000, and q is the correction factor.
- the partial bandwidth is then determined based on the ratio factor fact and the quantized subband normalization factor wnorm.
- the spectral energy in each sub-band can be obtained according to the quantized sub-band normalization factor, and the spectral energy in each sub-band is accumulated from the low frequency to the high frequency until the accumulated spectral energy is greater than the total spectral energy of all sub-bands.
- the product of the ratio factor fact taking the bandwidth below the current subband as part of the bandwidth.
- spectral energy can be obtained according to the following normalization factor according to the following equation:
- the subbands are added one by one from the low frequency to the high frequency, and the spectrum energy energy_limit is accumulated, and it is judged whether energy.1 imi t > factxenergy_sum 0 is satisfied. If it is not satisfied, the spectrum energy of the subband is continuously accumulated. If satisfied, the current subband is the last subband of the defined partial bandwidth, output The current subband number sfm_limit is used to represent the defined partial bandwidth, ie 0 sfm_limit;
- the rate factor information is used to determine the ratio factor fact.
- fact can be determined by a subband normalization factor. For example, first obtain the harmonic level or noise level of the audio signal based on the subband normalization factor. noise-leveh In general, the larger the harmonic level of the audio signal, the smaller the noise level. The noise level is taken as an example for explanation.
- the noise level noise_level can be obtained as follows.
- wnorm is the decoded subband normalization factor and sfm is the number of subbands for the entire frequency band.
- the fact When the noise-level is large, the fact is also large; when the noise-level is small, the fact is also small. If the harmonic level is used as a parameter, the fact is smaller when the harmonic level is larger; the fact is larger when the harmonic level is smaller.
- the low frequency partial bandwidth of 0 _ sfm_limit has been described above as an example, the embodiment of the present invention is not limited thereto.
- the above partial bandwidth may also be other forms as needed, for example, may be a partial bandwidth between a certain non-zero low frequency point and sfm_limit.
- the following iterative method can be used: a) finding the subband corresponding to the largest rim value, allocating a certain bit; b) then making the wmorm value of the subband Correspondingly less; c) Repeat steps a ⁇ b until the bit allocation is complete. 104. Encode the spectral coefficients of the audio signal according to the bits allocated by each subband.
- trellis vector quantization scheme that can be used for coding coefficients, or other quantization
- the quantized subband normalization factor or the code rate information is used to determine the signal bandwidth of the bit allocation, so that the number of bits can be concentrated to effectively encode and decode the determined signal bandwidth, and the audio is improved. quality.
- the bit allocation is performed within the signal bandwidth (Tsfn limit.
- Tsfn limit By limiting the bandwidth sfm_limit for bit allocation, the bit number pair is more concentrated at a lower bit rate
- the effective coding of the frequency band also makes the bandwidth expansion of the uncoded frequency band more efficient. This is mainly because if the bit allocation bandwidth is not limited, the high frequency harmonics will be divided into some scattered bits for encoding, but at the time. The distribution on the axis is not continuous, so that the reconstruction of the high-frequency harmonics is intermittent. If the scattered bits are more concentrated to the low frequency by limiting the bit allocation bandwidth, the low-frequency signal is better encoded, and the high-frequency harmonics pass the low-frequency signal. Bandwidth expansion, which makes the high frequency harmonic signals more continuous.
- the subband normalization factor of the subband in the bandwidth may be first determined.
- the adjustment makes it possible to allocate more bits in the high frequency band in the bandwidth.
- the adjusted intensity can be adaptive to the bit rate. The main consideration is that if the lower band energy in this bandwidth is larger and the bits are larger, the quantization bit is saturated, and this adjustment can be used to increase the middle and high frequency quantization bits in the band. More harmonics are also beneficial for higher frequency bandwidth extensions.
- the subband normalization factor of the intermediate subband of the partial bandwidth is used as the subband normalization factor of each subband after the intermediate subband, and the sfm_limit/2 subbands can be normalized.
- the factor is a subband normalization factor for each subband within the range of the frequency band sfm_l imi t/2 s sfm_ l imi t . If sfm_l imi t/2 is not an integer, it can be rounded up or down. At this time, the adjusted subband normalization factor can be used when performing bit allocation.
- audio signal frame classification can be further considered when applying the encoding and decoding method of the embodiment of the present invention.
- the embodiments of the present invention can adopt different codec strategies for different classifications, thereby improving the coding and decoding quality of different signals.
- the audio signal can be divided into various types such as No i se (noise), Harmonic (harmonic), Trans ient (transient).
- the noise-like signal is divided into noisy se mode, in which the spectrum is relatively flat; the signal with steep phase is divided into Trans i ent mode, and the spectrum is also flat; the harmonic signal is divided into Harmonic mode, then the spectrum The change is large and contains more information.
- Embodiments of the present invention may determine that the frame of the audio signal belongs to a harmonic type or a non-harmonic type prior to 101 of Fig. 1, and if the frame of the audio signal belongs to a harmonic type, the method of Fig. 1 is continued.
- the signal bandwidth of the bit allocation can be defined in accordance with the embodiment of Figure 1, i.e., the signal bandwidth of the bit allocation of the frame is limited to the partial bandwidth of the frame.
- the signal bandwidth of the bit allocation may be limited to a partial bandwidth according to the embodiment of FIG. 1, or the signal bandwidth of the bit allocation may not be limited, for example, the bit allocation bandwidth of such a frame is determined as the frame. The full bandwidth.
- Audio signal frames can be classified by peak-to-average ratio.
- the peak-to-average ratio of each of the sub-bands of all or a portion of the sub-bands of the frame (eg, a partial sub-band of high frequencies) is obtained.
- the peak-to-average ratio refers to the ratio of the peak energy or amplitude of the sub-band to the average energy or amplitude of the sub-band.
- the embodiments of the present invention are not limited to the example of classifying according to the peak-to-average ratio parameter, and may be classified according to other parameters.
- the bandwidth sfm_l imi t it is more efficient to concentrate the selected frequency band by concentrating the number of bits at a low code rate, and also making bandwidth expansion of the uncoded frequency band more effective, mainly because If the bit allocation bandwidth is not limited, the high frequency harmonics will be divided into some scattered bits for encoding, but the distribution on the time axis is not continuous, so that the reconstruction of the high frequency harmonics is intermittent, if the bandwidth is allocated by limiting bits. These scattered bits are more concentrated into the low frequency, so that the low frequency signal is better encoded, and the high frequency harmonics are expanded by the low frequency signal, so that the high frequency harmonic signal is more continuous.
- FIG. 2 is a flow chart of an audio signal decoding method according to an embodiment of the present invention.
- the quantized subband normalization factor can be obtained by decoding the bit stream.
- 202 Determine a signal bandwidth of the bit allocation according to the quantized subband normalization factor or according to the quantized subband normalization factor and the code rate information. 202 is similar to 102 in Fig. 1, and therefore the description will not be repeated.
- the quantized subband normalization factor or the code rate information is used to determine the signal bandwidth of the bit allocation, so that the number of bits can be concentrated to effectively encode and decode the determined signal bandwidth, and the audio is improved. quality.
- the embodiment of the present invention has no limitation on the order of execution of noise filling and bandwidth expansion in 205. You can perform noise filling before performing bandwidth expansion, or you can perform bandwidth expansion before performing noise filling. In addition, the embodiment of the present invention may perform bandwidth expansion on a part of the frequency band and perform noise filling on the other part of the frequency band first. These variations are all within the scope of embodiments of the invention.
- bandwidth expansion can be performed to obtain a normalized full-band spectrum. For example, based on the current frame and before it
- the bit allocation of the frame, the first frequency band is determined as the frequency band to be copied.
- N is a positive integer. It is generally desirable to select a plurality of consecutive sub-bands with bit allocation as the range of the first frequency band. Then, based on the spectral coefficients of the first frequency band, the spectral coefficients of the high frequency band are obtained.
- a correlation between a bit allocated by a current frame and a bit allocated by a previous N frame may be acquired, and the first frequency band is determined according to the acquired correlation.
- the bit allocated by the current frame be R-current
- the bit allocated in the previous frame be R_previous
- the obtained top_band may be taken as the first band upper limit and top.band/2 as the first band lower limit. If the difference between the lower limit of the first band of the previous frame and the lower limit of the first band of the current frame is less than 1 kHz, the lower limit of the first band of the previous frame may be taken as the lower limit of the first band of the current frame. This is mainly to ensure continuity of the first frequency band to be extended, thereby ensuring continuity of the extended high frequency spectrum. Then cache the current R-current in the middle, as the R_previous of the next middle. If top-band/2 is not an integer, it can be rounded up or down.
- noise filling first An example of performing noise filling first is described above.
- the embodiment of the present invention is not limited thereto, and the bandwidth extension may be performed first, and the background noise is filled in the extended full frequency band.
- the noise filling method can be similar to the above example.
- last_sfm _ high_sfm above-described range can be estimated by the decoder noise_level value, further adjustment band filled in the last-sfm- high_sfm range of background noise.
- noise-level refer to equation (8) above.
- the noise-level is obtained by decoding the subband normalization factor to distinguish the intensity level of the padding noise, so that no coded bits are transmitted.
- the background noise in the high frequency band can be adjusted using the obtained noise level as follows.
- y(k) ( (1 - noise _ level ) * y norm (k) + noise _ level * noise _ CB(k) ) * wnor m (9)
- U k ) is the normalized coefficient after decoding
- nQise - CBG is a noise code book.
- the present invention can also adjust the spectral coefficients of the first frequency band first, and then use the adjusted spectral coefficients for bandwidth extension to further improve the performance of the high frequency band.
- a normalized length can be obtained according to the spectral flatness information and the high-band signal type, the spectral coefficients of the first frequency band are normalized using the obtained normalized length, and the normalized first frequency band is processed The spectral coefficient is used as the spectral coefficient of the high frequency band.
- the above spectral flatness information may include: a mean peak ratio of each subband in the first frequency band, a correlation of a time domain signal corresponding to the first frequency band, or a zero crossing rate of a time domain signal corresponding to the first frequency band.
- the average peak ratio is exemplified below, but the embodiment of the present invention is not limited thereto, and other spectral flatness information may be similarly adjusted.
- the peak-to-average ratio is the ratio of the peak energy or amplitude of a subband to the average energy or amplitude of that subband.
- the peak-to-average ratio of each sub-band in the first frequency band is obtained according to the spectral coefficient of the first frequency band, and whether the sub-band is a harmonic sub-band is determined according to the peak-to-average ratio value and the maximum peak value in the sub-band, and is recorded.
- the number of harmonic subbands n_band is then adaptively determined according to the signal type of n_band and the high frequency band itself. ength. norm. harm: n band
- Length _ norm— harm a * ⁇ 1 + -
- Adaptive signal type such as harmonic signal
- the spectral coefficients of the first frequency band can then be normalized using the obtained normalized length, and the spectral coefficients of the normalized first frequency band can be used as the spectral coefficients of the high frequency band.
- the decoder can further consider the audio signal frame classification.
- the embodiment of the present invention can adopt different coding and decoding strategies for different classifications, thereby improving the coding and decoding quality of different signals.
- the method of classifying the audio signal frame can be referred to the coding end, and therefore will not be described again.
- Classification information indicating the frame type can be extracted from the code stream.
- the signal bandwidth of the bit allocation can be defined in accordance with the embodiment of Figure 2, i.e., the signal bandwidth of the bit allocation of the frame is limited to a portion of the bandwidth of the frame.
- the signal bandwidth of the bit allocation may be limited to a partial bandwidth according to the embodiment of FIG. 2, or the signal bandwidth of the bit allocation may not be limited according to the prior art, for example, bit allocation of such a frame.
- the bandwidth is determined to be the full bandwidth of the frame.
- the embodiment of the present invention can improve the quality of the harmonic signal without reducing the quality of the non-harmonic signal.
- FIG. 3 is a block diagram of an audio signal encoding apparatus in accordance with one embodiment of the present invention.
- the audio signal encoding apparatus 30 of Fig. 3 includes a quantizing unit 31, a first determining unit 32, a first assigning unit 33, and an encoding unit 34.
- the quantizing unit 31 divides the frequency band of the audio signal into a plurality of sub-bands, and quantizes the sub-band normalization factor of each sub-band.
- the first determining unit 32 determines the signal bandwidth of the bit allocation based on the subband normalization factor quantized by the quantization unit 31 or based on the quantized subband normalization factor and code rate information.
- the first allocation unit 33 allocates bits to subbands within the signal bandwidth determined by the first determining unit 32.
- the encoding unit 34 encodes the spectral coefficients of the audio signal based on the bits allocated by the first allocation unit 33 for each subband.
- the coded bandwidth of the bit allocation is determined according to the quantized subband normalization factor or code rate information, so that the number of bits can be concentrated.
- the determined signal bandwidth is effectively coded to improve audio quality.
- FIG. 4 is a block diagram of an audio signal encoding apparatus according to another embodiment of the present invention.
- the audio signal encoding apparatus 40 of Fig. 4 the same or similar portions as those of Fig. 3 are denoted by the same reference numerals.
- the first determining unit 32 may define the bit-allocated signal bandwidth as a partial bandwidth of the audio signal.
- the first determining unit 32 may include a first ratio factor determining module 321.
- the first ratio factor determination module 321 can determine a ratio factor fact based on the code rate information, the ratio factor fact being greater than Q and less than or equal to one.
- the first determining unit 32 may include a second ratio factor determining module 322 instead of the first ratio factor determining module 321.
- the second ratio factor determination module 322 obtains the harmonic level or noise level of the audio signal based on the subband normalization factor, and determines the ratio factor fact based on the harmonic level or noise level.
- the first determining unit 32 further includes a first bandwidth determining module 323. After the ratio factor fac t is obtained, the first bandwidth determining module 323 can determine the partial bandwidth based on the ratio factor fact and the quantized subband normalization factor.
- the first bandwidth determining module 323 when determining the partial bandwidth, acquires spectral energy in each subband according to the quantized subband normalization factor, and accumulates from the low frequency to the high frequency. The spectral energy in each subband is until the accumulated spectral energy is greater than the product of the total spectral energy of all subbands and the ratio factor fact, and the bandwidth below the current subband is taken as the partial bandwidth.
- the audio signal encoding device 40 may further include a classifying unit 35 for classifying frames of the audio signal.
- the classification unit 35 may determine that the frame of the audio signal belongs to a harmonic type or a non-harmonic type, and if the frame of the audio signal belongs to a harmonic type, the quantization unit 31 is triggered.
- the type of frame may be determined based on the mean peak ratio.
- the classification unit 35 acquires all or part of the frame.
- the peak-to-average ratio of each sub-band in the sub-band, when the number of sub-bands whose peak-to-average ratio is greater than the first threshold is greater than or equal to the second threshold, determining that the frame belongs to a harmonic type, and the sub-average ratio is greater than the first threshold
- the first determining unit 32 can limit the signal bandwidth of the bit allocation to the partial bandwidth of the frame for the frame belonging to the harmonic type.
- the first allocation unit 33 may include a subband normalization factor adjustment module 331 and a bit allocation module 332.
- Subband normalization factor adjustment module 331 adjusts the subband normalization factor of the subbands within the determined signal bandwidth, and bit allocation module 332 performs bit allocation based on the adjusted subband normalization factor.
- the first allocation unit 33 may use the subband normalization factor of the intermediate subband of the partial bandwidth determined by the first determining unit 32 as the subband normalization factor of each subband after the intermediate subband.
- the quantized subband normalization factor or the code rate information is used to determine the signal bandwidth of the bit allocation, so that the number of bits can be concentrated to effectively encode and decode the determined signal bandwidth, and the audio is improved. quality.
- FIG. 5 is a block diagram of an audio signal decoding apparatus in accordance with one embodiment of the present invention.
- the audio signal decoding apparatus 50 of Fig. 5 includes an acquisition unit 51, a second determination unit 52, a second assignment unit 53, a decoding unit 54, an extension unit 55, and a restoration unit 56.
- the obtaining unit 51 obtains the quantized subband normalization factor.
- the second determining unit 52 determines the signal bandwidth of the bit allocation according to the quantized subband normalization factor acquired by the obtaining unit 51, or according to the quantized subband normalization factor and the code rate information.
- the second allocation unit 53 allocates bits to subbands within the signal bandwidth determined by the second determining unit 52.
- the decoding unit 54 decodes the normalized spectrum based on the bits allocated by the second allocation unit 53 for each subband.
- the spreading unit 55 performs noise filling and bandwidth expansion on the normalized spectrum decoded by the decoding unit 54, to obtain a normalized full-band spectrum.
- Recovery order The element 56 obtains the spectral coefficients of the audio signal based on the normalized full-band spectrum and the sub-band normalization factor obtained by the spreading unit 55.
- the quantized subband normalization factor or the code rate information is used to determine the signal bandwidth of the bit allocation, so that the number of bits can be concentrated to effectively encode and decode the determined signal bandwidth, and the audio is improved. quality.
- FIG. 6 is a block diagram of an audio signal decoding apparatus according to another embodiment of the present invention.
- the audio signal decoding device 60 of Fig. 6 the same or similar portions as those of Fig. 5 are denoted by the same reference numerals.
- the second determining unit 52 of the audio signal decoding device 60 can define the signal bandwidth of the bit allocation as a partial bandwidth of the audio signal.
- the second determining unit 52 may include a third ratio factor determining unit 521 for determining a ratio factor fact based on the code rate information, the ratio factor fac t being greater than 0 and less than or equal to 1.
- the second determining unit 52 may include a fourth ratio factor determining unit 522 for acquiring a harmonic level or a noise level of the audio signal according to the subband normalization factor, and determining a ratio factor fact according to the harmonic level or the noise level. .
- the second determining unit 52 further includes a second bandwidth determining module 523.
- the second bandwidth determining module 523 can determine the partial bandwidth based on the ratio factor fact and the quantized subband normalization factor.
- the second bandwidth determining module 523 when determining the partial bandwidth, acquires spectral energy in each subband according to the quantized subband normalization factor, and accumulates from the low frequency to the high frequency. The spectral energy in each subband is until the accumulated spectral energy is greater than the product of the total spectral energy of all subbands and the ratio factor fact, and the bandwidth below the current subband is taken as the partial bandwidth.
- the extension unit 55 may include a first frequency band determination. Module 551 and spectral coefficient acquisition module 552.
- the first frequency band determining module 551 determines the first frequency band according to the bit allocation of the current frame and its previous N frames, where N is a positive integer
- the spectral coefficient obtaining module 552 obtains the spectral coefficients of the high frequency frequency band according to the spectral coefficients of the first frequency band.
- the first frequency band determining module 551 may acquire a correlation between a bit allocated by the current frame and a bit allocated by the previous N frame, and determine the first frequency band according to the acquired correlation.
- the audio signal decoding apparatus 60 may further include an adjustment unit 57 for obtaining a noise level based on the subband normalization factor and adjusting the background noise in the high frequency band using the obtained noise level.
- the spectral coefficient acquisition module 552 can obtain a normalized length according to the spectral flatness information and the high-band signal type, and use the obtained normalized length to the spectrum of the first frequency band.
- the coefficients are normalized, and the spectral coefficients of the normalized first frequency band are used as the spectral coefficients of the high frequency band.
- the spectrum flatness information may include: a mean peak ratio of each subband in the first frequency band, a correlation of a time domain signal corresponding to the first frequency band, or a zero crossing rate of a time domain signal corresponding to the first frequency band.
- the quantized subband normalization factor or the code rate information is used to determine the signal bandwidth of the bit allocation, so that the number of bits can be concentrated to effectively encode and decode the determined signal bandwidth, and the audio is improved. quality.
- a codec system may include the above-described audio signal encoding device or audio signal decoding device.
- the disclosed systems, devices, and methods may be implemented in other ways.
- the device embodiments described above are merely illustrative.
- the division of the unit is only a logical function division.
- there may be another division manner for example, multiple units or components may be combined or Can be integrated into another system, or some features can be ignored or not executed.
- the coupling or direct coupling or communication connection shown or discussed may be an indirect coupling or communication connection through some interface, device or unit, and may be electrical, mechanical or otherwise.
- the units described as separate components may or may not be physically separate, and the components displayed as units may or may not be physical units, i.e., may be located in one place, or may be distributed over multiple network units. Some or all of the units may be selected according to actual needs to achieve the purpose of the solution of the embodiment.
- each functional unit in each embodiment of the present invention may be integrated into one processing unit, or each unit may exist physically separately, or two or more units may be integrated into one unit.
- the functions may be stored in a computer readable storage medium if implemented in the form of a software functional unit and sold or used as a standalone product. Based on such understanding, the technical solution of the present invention, which is essential or contributes to the prior art, or a part of the technical solution, may be embodied in the form of a software product, which is stored in a storage medium, including Several instructions to make one
- the computer device (which may be a personal computer, server, or network device, etc.) performs all or part of the steps of the methods described in various embodiments of the present invention.
- the foregoing storage medium includes: a USB flash drive, a removable hard disk, a read-only memory (ROM), a random access memory (RAM, Random Acces s Memory), a magnetic disk or an optical disk, and the like, which can store program codes. medium.
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- Audiology, Speech & Language Pathology (AREA)
- Human Computer Interaction (AREA)
- Acoustics & Sound (AREA)
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EP12731282.5A EP2613315B1 (en) | 2011-07-13 | 2012-03-22 | Method and device for coding an audio signal |
EP16160249.5A EP3174049B1 (en) | 2011-07-13 | 2012-03-22 | Audio signal coding method and device |
JP2014519382A JP5986199B2 (ja) | 2011-07-13 | 2012-03-22 | 音声信号の符号化と復号化の方法および装置 |
ES12731282.5T ES2612516T3 (es) | 2011-07-13 | 2012-03-22 | Método y dispositivo de codificación y decodificación de señal de audio |
KR1020167035436A KR101765740B1 (ko) | 2011-07-13 | 2012-03-22 | 오디오 신호 코딩 및 디코딩 방법 및 장치 |
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US13/532,237 US9105263B2 (en) | 2011-07-13 | 2012-06-25 | Audio signal coding and decoding method and device |
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US15/981,645 US10546592B2 (en) | 2011-07-13 | 2018-05-16 | Audio signal coding and decoding method and device |
US16/731,897 US11127409B2 (en) | 2011-07-13 | 2019-12-31 | Audio signal coding and decoding method and device |
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US10971162B2 (en) | 2012-12-06 | 2021-04-06 | Huawei Technologies Co., Ltd. | Method and device for decoding signal |
US11610592B2 (en) | 2012-12-06 | 2023-03-21 | Huawei Technologies Co., Ltd. | Method and device for decoding signal |
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US9984697B2 (en) | 2018-05-29 |
US20200135219A1 (en) | 2020-04-30 |
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US20130018660A1 (en) | 2013-01-17 |
EP3174049B1 (en) | 2019-01-09 |
US10546592B2 (en) | 2020-01-28 |
US9105263B2 (en) | 2015-08-11 |
KR20160149326A (ko) | 2016-12-27 |
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JP6702593B2 (ja) | 2020-06-03 |
JP2018106208A (ja) | 2018-07-05 |
ES2718400T3 (es) | 2019-07-01 |
JP5986199B2 (ja) | 2016-09-06 |
EP2613315B1 (en) | 2016-11-02 |
KR20140005358A (ko) | 2014-01-14 |
JP2016218465A (ja) | 2016-12-22 |
JP6321734B2 (ja) | 2018-05-09 |
CN102208188B (zh) | 2013-04-17 |
EP3174049A1 (en) | 2017-05-31 |
US20150302860A1 (en) | 2015-10-22 |
US20180261234A1 (en) | 2018-09-13 |
EP2613315A1 (en) | 2013-07-10 |
KR101765740B1 (ko) | 2017-08-07 |
JP2014523549A (ja) | 2014-09-11 |
EP2613315A4 (en) | 2013-07-10 |
KR20160028511A (ko) | 2016-03-11 |
ES2612516T3 (es) | 2017-05-17 |
PT2613315T (pt) | 2016-12-22 |
KR101602408B1 (ko) | 2016-03-10 |
CN102208188A (zh) | 2011-10-05 |
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