TW201007708A - Apparatus and method for generating a bandwidth extended signal - Google Patents
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Abstract
Description
201007708 六、發明說明: 【發明所屬之技術甸域】 根據本發明的實施例係關於音訊信號處理,且特別地 係關於種用於從輸入信號產生帶寬擴展信號的裝置與方 法種用於基於輸入信號與音訊信號提供帶寬減小信號 的裝置與方法。 音SfUs號的知覺適應性編碼,針對這些信號的有效儲 存與傳輸提供實質資料速率減小已在許多領域中獲得了廣 泛的接受。許多編碼演算法是已知的,例如MPEG 1/2第3 層(“MP3”)或MPEG 4 AAC(高階音訊編碼)。然而,使用此 的編碼’特別是當以最低位元率操作時,可導致主觀音訊 品質的減小’而這通常主要由將被傳送音訊信號帶寬的編 碼器端引起的限制產生。 從WO 98 57436已知的是,在這種情況下使音訊信號在 編碼器端遭受一頻帶限制以及透過一高品質音訊編碼器 (“核心編碼器”)只編碼音訊信號的較低頻帶。然而,較高頻 帶只被非常粗略地特徵化,即透過再現較高頻帶之頻譜包 絡的一組參數。在解碼器端,較高頻帶則被合成。為了達 到這一目的,諸波轉置被提出,其中經解碼音訊信號的較 低頻帶被提供給一濾波器組。較低頻帶的遽波器組通道連 接到較高頻帶的濾波器組通道或者遭“修補,,’且每一經修 補帶通信號遭受包絡調整。屬於一特定分析滤波器組的合 成濾波器組接收較低頻帶中之音訊信號的帶通信號及經較 3 201007708 低頻帶的經包絡調整帶通信號’該等信號遭調和修補到較 高頻帶中。合成濾波器組的輸出信號是相對於其原始帶寬 被擴展的音訊信號,該信號透過以一非常低資料速率操作 的核心編碼器從編碼器端遭傳送到解碼器端。特別地,濾 波器組域中的濾波器組計算與修補可能變成較高的計算工 作。 用於帶寬擴展帶限音訊信號的複雜性降低方法替代使 用低頻信號部分(LF)到高頻範圍(HF)的複製功能,以近似 由頻帶限制產生的資訊丟失。這些方法在下面予以描述: 2002年5月在慕尼克召開的第112屆AES會議(112th AES convention, Munich)通過的 M. Dietz, L. Liljeryd,Κ· Kj0rling及 O. Kunz 的 “Spectral Band Replication, a novel approach in audio coding”; 2002年 5 月在慕尼克召開的第 112 屆 AES會議通過的 S. Meltzer,R. B0hm及F. Henn的 “SBR enhanced audio codecs for digital broadcasting such as “Digital Radio Mondiale”(DRM)”; 2002年5月在慕尼克召開 的第 112屆AES會議通過的T. Ziegler, A. Ehret, Ρ· Ekstrand 及 M. Lutzky 的 “Enhancing mp3 with SBR: Features and Capabilities of the new mp3PRO Algorithm” ;國際標準 ISO/IEC 14496-3:2001/FPDAM 1、“Bandwidth Extension”, ISO/IEC 2002 或 Vasu Iyengar 等人的美國專利 Nr. 5,455,888“Speech bandwidth extension method and apparatus”。 在這些方法中,沒有諧波轉置被執行,但是較低頻帶 201007708 的連續帶通信號被引入到較高頻帶的連續濾波器組通道 中。透過這一點,音说仏波之較南頻帶的粗略近似被實現。 在另一步驟中,透過使用從原始信號獲得之控制資訊的後 處理,信號的這一粗略近似與原始相似。在這裡,例如縮 放因數用於調適頻譜包絡、反向瀘、波、及雜訊基準的增加, 該雜訊基準用於調適音調及丢失諧·波之正弦信號部分的增 補,這也在MPEG-4高效高階音訊編碼(EfficiencyAdvanced Audio Coding)(HE-AAC)標準中予以描述。 除此之外,另外的方法使用用於帶寬擴展的一相位聲 碼器。當施加用於頻譜擴展的該相位聲碼器時,頻率線進 一步彼此遠離移動。若例如透過量化使頻譜中存在間隙, 則透過擴展甚至同樣會增加間隙。在能量調適中,與原始 信號中的各自的線相比較,頻譜中的剩餘線接收太多的能 量。 第13圖顯示使用一相位聲碼器之帶寬擴展13〇〇的一概 要繪示。在這個例子中,兩個補丁 1312、1314被加入到一 信號的低頻帶1302中。也被稱為分頻器(Xover)頻率的該信 號的上截止頻率1320(交越頻率)是鄰近補丁 1312的低端頻 率,而分頻器頻率的兩倍是鄰近補丁 1312的上截止頻率及 下一補丁 1314的下截止頻率。相位聲碼器使信號的低頻帶 1302中的頻率線的頻率加倍,以獲得鄰近補丁 ι312,使信 號的低頻帶1302中的頻率線的頻率增至三倍,以獲得下一 補丁 1314 °因此,鄰近補丁 1312的頻譜密度只是信號的低 頻帶1302之頻譜密度的一半,而下一補丁 1314的頻譜密度 5 201007708 只是信號的低頻帶1302之頻譜密度的三分之一。 透過使頻帶(補丁)中的能量只集中到少數幾條頻率 線,音色中的實質改變導致其不同於原始的。以前較多頻 帶(頻率線)中的能量被加和成較少的剩餘的一些。 相位聲碼器的一些例子及其應用在下面提出: ICASSP’09 的 “Frederik Nagel and Sascha Disch,A Harmonic Bandwidth Extension Method for Audio Codecs” 與 M· Puckette, "Phase-locked Vocoder," IEEE ASSP Conference on Applications of Signal Processing to Audio and Acoustics, Mohonk 1995 ' Robel, A.: Transient detection and preservation in the phase vocoder ; citeseer.ist.psu.edu/679246.html、Laroche L.,Dolson M.: Improved phase vocoder timescale modification of audio, IEEE Trans. Speech and Audio Processing ’ 第 7卷,第 3號, 第323-332頁及美國專利6549884。 一種用於填充間隙的方法在WO 00/45379中顯示。其包 含一種用於使用高頻重建來增強信號源編碼系統的方法與 裝置。該應用旨在透過適應性雜訊基準相加來解決所重建 高頻帶中的不充足雜訊成分的問題。加入雜訊可填充間 隙,但是音訊品質或主觀品質可能不被充分地增加。201007708 VI. DESCRIPTION OF THE INVENTION: [Technical Fields According to the Invention] Embodiments in accordance with the present invention relate to audio signal processing, and in particular to apparatus and methods for generating a bandwidth extension signal from an input signal for input based Signals and audio signals provide means and method for reducing the bandwidth. Perceptual adaptive coding of the SfUs number, which provides substantial data rate reduction for efficient storage and transmission of these signals, has gained wide acceptance in many fields. Many coding algorithms are known, such as MPEG 1/2 Layer 3 ("MP3") or MPEG 4 AAC (High Order Audio Coding). However, the use of this code', especially when operating at the lowest bit rate, can result in a reduction in subjective audio quality' which is typically caused primarily by limitations imposed by the encoder side of the bandwidth of the transmitted audio signal. It is known from WO 98 57436 that in this case the audio signal is subject to a band limitation at the encoder end and only a lower frequency band of the audio signal is encoded by a high quality audio encoder ("core coder"). However, the higher frequency band is only very roughly characterized, i.e., through a set of parameters that reproduce the spectral envelope of the higher frequency band. At the decoder side, the higher frequency bands are synthesized. To achieve this, wave transposition is proposed in which the lower frequency band of the decoded audio signal is provided to a filter bank. The lower frequency band chopper bank channel is connected to the higher band filter bank channel or is "patched," and each repaired bandpass signal suffers from envelope adjustment. Synthesis filter bank reception belonging to a specific analysis filter bank The bandpass signal of the audio signal in the lower frequency band and the envelope-adjusted bandpass signal of the lower band 201007708' are modulated and patched into the higher frequency band. The output signal of the synthesis filter bank is relative to its original An audio signal whose bandwidth is extended, which is transmitted from the encoder side to the decoder side through a core encoder operating at a very low data rate. In particular, filter bank calculation and repair in the filter bank domain may become High computational work. The complexity reduction method for bandwidth extension band-limited audio signals replaces the use of the low-frequency signal portion (LF) to the high-frequency range (HF) copy function to approximate the loss of information due to band limitation. Described below: M. Dietz, L. Lilje, adopted at the 112th AES convention (Munich) in Munich in May 2002 Ryd, K. Kj0rling and O. Kunz, “Spectral Band Replication, a novel approach in audio coding”; S. Meltzer, R. B0hm and F. Henn, adopted at the 112th AES meeting in Munich in May 2002 "SBR enhanced audio codecs for digital broadcasting such as "Digital Radio Mondiale" (DRM)"; T. Ziegler, A. Ehret, Ρ Ekstrand and M adopted at the 112th AES meeting in Munich in May 2002 Lutzky's "Enhancing mp3 with SBR: Features and Capabilities of the new mp3PRO Algorithm"; international standard ISO/IEC 14496-3:2001/FPDAM 1, "Bandwidth Extension", ISO/IEC 2002 or Vasu Iyengar et al. Nr. 5,455,888 "Speech bandwidth extension method and apparatus". In these methods, no harmonic transposition is performed, but the continuous bandpass signal of the lower band 201007708 is introduced into the higher band continuous filter bank channel. Through this, the sound says that the rough approximation of the souther frequency band is realized. In another step, this rough approximation of the signal is similar to the original by using post-processing of the control information obtained from the original signal. Here, for example, the scaling factor is used to adapt the spectral envelope, the inverse chirp, the wave, and the increase of the noise reference, which is used to adjust the complement of the sinusoidal signal portion of the tone and the missing harmonic wave, which is also in MPEG- 4 Efficient Advanced Audio Coding (HE-AAC) standard is described in the standard. In addition to this, another method uses a phase vocoder for bandwidth extension. When the phase vocoder for spectrum spreading is applied, the frequency lines further move away from each other. If, for example, by quantization, there is a gap in the spectrum, the transmission spread will even increase the gap. In energy adaptation, the remaining lines in the spectrum receive too much energy compared to the respective lines in the original signal. Figure 13 shows an overview of the bandwidth extension of a phase vocoder using 13 〇〇. In this example, two patches 1312, 1314 are added to the low frequency band 1302 of a signal. The upper cutoff frequency 1320 (crossover frequency) of the signal, also referred to as the frequency divider (Xover) frequency, is the low end frequency of the adjacent patch 1312, and twice the frequency of the divider is the upper cutoff frequency of the adjacent patch 1312 and The next cutoff frequency for the next patch 1314. The phase vocoder doubles the frequency of the frequency line in the low frequency band 1302 of the signal to obtain a neighboring patch ι 312, which triples the frequency of the frequency line in the low frequency band 1302 of the signal to obtain the next patch 1314 °. The spectral density of the adjacent patch 1312 is only half of the spectral density of the low frequency band 1302 of the signal, while the spectral density 5 201007708 of the next patch 1314 is only one third of the spectral density of the low frequency band 1302 of the signal. By concentrating the energy in the band (patch) to only a few frequency lines, the substantial change in the tone causes it to be different from the original. In the past, more energy in the frequency band (frequency line) was added to the lesser remaining. Some examples of phase vocoders and their applications are presented below: "Frederik Nagel and Sascha Disch, A Harmonic Bandwidth Extension Method for Audio Codecs" by ICASSP'09 with M·Puckette, "Phase-locked Vocoder," IEEE ASSP Conference on Applications of Signal Processing to Audio and Acoustics, Mohonk 1995 ' Robel, A.: Transient detection and preservation in the phase vocoder ; citeseer.ist.psu.edu/679246.html, Laroche L., Dolson M.: Improved phase Vocoder timescale modification of audio, IEEE Trans. Speech and Audio Processing 'Vol. 7, No. 3, pp. 323-332 and U.S. Patent 6,549, 878. A method for filling a gap is shown in WO 00/45379. It includes a method and apparatus for enhancing a signal source coding system using high frequency reconstruction. The application aims to solve the problem of insufficient noise components in the reconstructed high frequency band by adding adaptive noise references. Adding noise can fill the gap, but the audio quality or subjective quality may not be sufficiently increased.
C發明内容;J 本發明的目的是提供音訊信號帶寬擴展的概念,這增 加了帶寬擴展信號的主觀品質。 這一目的透過如申請專利範圍第1項及第^項所述之 201007708 裝置、如申請專利範圍第14項所述之音訊信號及如申請專 利範圍第15項及第16項所述之方法得以解決。 本發明的一實施例提供一種用於從輸入信號產生帶寬 擴展信號的裝置。該輸入信號的第一頻帶由第一解析度資 料表示,第二頻帶由第二解析度資料表示,該第二解析度 低於該第一解析度。該裝置包含一補丁產生器及一組合 器。該補丁產生器受組配以根據第一修補演算法從輸入信 號的第一頻帶產生第一補丁,及受組配以根據第二修補演 算法從輸入信號的第一頻帶產生第二補丁。根據第二修補 演算法產生的第二補丁的頻譜密度較根據第一修補演算法 產生的第一補丁的頻譜密度高。該組合器受組配以組合該 第一補丁、該第二補丁與輸入信號的第一頻帶,以獲得帶 寬擴展信號。用於產生帶寬擴展信號的裝置受組配以根據 第一修補演算法及根據第二修補演算法縮放輸入信號或縮 放第一補丁及第二補丁,藉此帶寬擴展信號滿足一頻譜包 絡準則。 根據本發明的實施例係基於具有低頻譜密度的補丁 (這意味著,例如與輸入信號的低頻帶相比較,該補丁包含 間隙)與具有高頻譜密度的補丁(這意味著,例如與輸入信號 的低頻帶相比較’該補丁只包含較少或不包含間隙)組合, 以擴展輸入信號之頻帶的中心思想。因為兩個補丁係基於 輸入信號產生,輸入信號之低頻帶的高頻率帶寬擴展可能 提供原始音訊信號的良好近似。此外,第一與第二補丁可 在產生之前或之後遭縮放(透過縮放輸入信號),以滿足一頻 7 201007708 譜包絡準則,因為原始音訊信號的頻譜包絡應被認為用於 重建輸入信號的高頻帶。以此方式,帶寬擴展信號的主觀 品質或音訊品質可能被顯著地增加。 在根據本發明的一些實施例中,第一修補演算法是諧 波修補演算法。換言之,第一補丁被產生,藉此只是輸入 信號之第一頻帶中之頻率的整數倍的頻率被包含在第一補 丁中。此外,第二修補演算法可能是一混合修補演算法。 這意味著例如第二補丁可被產生,藉此第二補丁包含是輸 入信號之第一頻帶中之頻率的整數倍的頻率以及不是輸入 信號之第一頻帶中之頻率的整數倍的頻率。因此,第二補 丁的頻譜密度較第一補丁的頻譜密度高。透過組合第一補 丁與第二補丁,第一補丁的丟失頻率線可由第二補丁的頻 率線填充。以此方式,根據第一修補演算法之諧波帶寬擴 展中的間隙可由第二補丁填充,且帶寬擴展信號的音訊品 質可被顯著地提高。 根據本發明的一些實施例與一種基於一輸入信號提供 一帶寬減小信號的裝置有關。該裝置包含一頻譜包絡資料 決定器、一補丁縮放控制資料產生器、及一輸出介面。該 頻譜包絡資料決定器受組配以基於輸入信號的高頻帶決定 頻譜包絡資料。該補丁縮放控制資料產生器受組配以產生 補丁縮放控制資料,該補丁縮放控制資料用於在解碼器端 縮放帶寬減小信號或者用於透過解碼器縮放第一補丁及第 二補丁,藉此由該解碼器產生的帶寬擴展信號滿足一頻譜 包絡準則。該頻譜包絡準則係基於頻譜包絡資料。第一補 201007708 :根據第-修算法從帶寬減讀號的低解產生,及 第-補丁根據第二修補演算法從帶寬減小信號的低頻帶產 2根據第二修補演算法產生的第二補丁的頻譜密度較根 據:一修補演算法產生的第一補丁的頻譜密度高。輸出介 面义組配以組合輸人信號的低頻帶、頻譜包絡資料、及補 y縮放控•料,謂得帶寬減小錢。再者,輸出介面 文組配以提供帶寬減小信制於傳輸或健存。C SUMMARY OF THE INVENTION The purpose of the present invention is to provide a concept of bandwidth extension of audio signals, which increases the subjective quality of the bandwidth extension signal. This object is achieved by the method of claim 10, the method of claim 11, and the method of claim 15 and claim 16, solve. An embodiment of the present invention provides an apparatus for generating a bandwidth extension signal from an input signal. The first frequency band of the input signal is represented by a first resolution data, and the second frequency band is represented by a second resolution data, the second resolution being lower than the first resolution. The device includes a patch generator and a combiner. The patch generator is configured to generate a first patch from a first frequency band of the input signal in accordance with a first patching algorithm and to be configured to generate a second patch from a first frequency band of the input signal in accordance with a second patching algorithm. The second patch generated according to the second patching algorithm has a higher spectral density than the first patch generated according to the first patching algorithm. The combiner is configured to combine the first patch, the second patch, and the first frequency band of the input signal to obtain a bandwidth extension signal. The means for generating the bandwidth extension signal is adapted to scale the input signal or to scale the first patch and the second patch according to the first patching algorithm and according to the second patching algorithm, whereby the bandwidth extension signal satisfies a spectral envelope criterion. Embodiments in accordance with the present invention are based on patches with low spectral density (which means, for example, compared to the low frequency band of the input signal, the patch contains gaps) and patches with high spectral density (which means, for example, with input signals) The low frequency band compares the 'the patch contains only little or no gaps' combination to extend the central idea of the frequency band of the input signal. Since the two patches are generated based on the input signal, the high frequency bandwidth extension of the low frequency band of the input signal may provide a good approximation of the original audio signal. In addition, the first and second patches may be scaled (by scaling the input signal) before or after generation to satisfy the spectral 7 201007708 spectral envelope criterion, since the spectral envelope of the original audio signal should be considered to be used to reconstruct the input signal. frequency band. In this way, the subjective quality or audio quality of the bandwidth extension signal may be significantly increased. In some embodiments in accordance with the invention, the first patching algorithm is a harmonic patching algorithm. In other words, the first patch is generated whereby only the frequency of an integer multiple of the frequency in the first frequency band of the input signal is included in the first patch. In addition, the second patching algorithm may be a hybrid patching algorithm. This means, for example, that a second patch can be generated whereby the second patch contains a frequency that is an integer multiple of the frequency in the first frequency band of the input signal and a frequency that is not an integer multiple of the frequency in the first frequency band of the input signal. Therefore, the spectral density of the second patch is higher than the spectral density of the first patch. By combining the first patch and the second patch, the missing frequency line of the first patch can be filled by the frequency line of the second patch. In this way, the gap in the harmonic bandwidth extension according to the first patching algorithm can be filled by the second patch, and the audio quality of the bandwidth extension signal can be significantly improved. Some embodiments in accordance with the present invention are related to an apparatus for providing a bandwidth reduction signal based on an input signal. The device includes a spectral envelope data determinator, a patch scaling control data generator, and an output interface. The spectral envelope data determinator is configured to determine spectral envelope data based on the high frequency band of the input signal. The patch scaling control data generator is configured to generate patch scaling control data for scaling the bandwidth reduction signal at the decoder side or for scaling the first patch and the second patch by the decoder The bandwidth extension signal produced by the decoder satisfies a spectral envelope criterion. The spectral envelope criterion is based on spectral envelope data. First supplement 201007708: generated according to the low-solution of the bandwidth minus reading number according to the first repair algorithm, and the first patch is generated from the low-band yield of the bandwidth reduction signal according to the second patching algorithm 2 according to the second patching algorithm The spectral density of the patch is higher than that of the first patch generated by a patching algorithm. The output interface group is combined with the low frequency band of the combined input signal, the spectrum envelope data, and the complement y-scale control material, which means that the bandwidth is reduced. Furthermore, the output interface is configured to provide bandwidth reduction signaling for transmission or health.
根據本發明的另外的實施例與包含第-頻帶及第二頻 帶的一音訊信號有關。第—頻帶由第—解析度資料表示, 第二頻帶由第二解析度㈣表示。第二解析度低於第-解 析度。第二解析度資料係基於第二頻帶_譜包絡資料及 第頻帶的補丁縮放控制資料,該補丁縮放控制資料用於 在解碼^端縮放音訊錢或透過解碼魏衫-補丁及第 一補丁,藉此由該解碼器產生的帶寬擴展信號滿足一頻譜 匕、各準則。该頻譜包絡準則係基於頻譜包絡資料。第一補 丁根據第一修補演算法從音訊信號的第—頻帶產生及第 補丁根據第二修補演算法從音訊信號的第一頻帶產生。 根據第二修補演算法產生的第二補丁的料密度較根據第 一修補演算法產生的第-補丁的頻譜密度高。 圖式簡單說明 根據本發明的實施例隨後將參考所附圖式詳細說明, 其中: 第1圖是用於從輸入信號產生帶寬擴展信號之一裝置 的方塊圖; 9 201007708 第2a圖是一所產生第一補丁的概要繪示; 第2b圖是一所產生第一及第二補丁的概要繪示; 第3a圖是用於從輸入信號產生帶寬擴展信號之一裝置 的方塊圖; 第3b圖是一剪輯正弦輸入信號的概要繪示; 第3c圖是一經半波整流正弦輸入信號的概要繪示; 第3d圖是一經全波整流正弦輸入信號的概要繪示;A further embodiment in accordance with the invention is associated with an audio signal comprising a first frequency band and a second frequency band. The first frequency band is represented by the first resolution data, and the second frequency band is represented by the second resolution (four). The second resolution is lower than the first resolution. The second resolution data is based on the second frequency band spectral profile data and the patch scaling control data of the first frequency band, and the patch scaling control data is used to scale the audio money in the decoding end or to decode the Wei shirt-patch and the first patch. The bandwidth extension signal generated by the decoder satisfies a spectrum and criteria. The spectral envelope criterion is based on spectral envelope data. The first patch is generated from the first frequency band of the audio signal according to the first patching algorithm and the first patch is generated from the first frequency band of the audio signal according to the second patching algorithm. The material density of the second patch generated according to the second patching algorithm is higher than the spectral density of the first patch generated according to the first patching algorithm. BRIEF DESCRIPTION OF THE DRAWINGS The embodiments according to the present invention will be described in detail with reference to the accompanying drawings, wherein: FIG. 1 is a block diagram of an apparatus for generating a bandwidth extension signal from an input signal; 9 201007708 Figure 2a is a diagram Generating a summary of the first patch; Figure 2b is a schematic diagram showing the generation of the first and second patches; Figure 3a is a block diagram of one of the means for generating a bandwidth extension signal from the input signal; Is a schematic depiction of a clipped sinusoidal input signal; Figure 3c is a schematic representation of a half-wave rectified sinusoidal input signal; and Figure 3d is a schematic representation of a full-wave rectified sinusoidal input signal;
第4圖是用於從輸入信號產生帶寬擴展信號之一裝置 的方塊圖; 第5a圖是一相位聲碼器之濾波器組實施態樣的概要繪 示; 第5b圖是第5a圖中之濾波器的詳細繪示; 第5c圖是用於處理在第5a圖之一濾波器通道中的振幅 信號與頻率信號的概要繪示; 第6圖是一相位聲碼器之轉換實施態樣的概要繪示;Figure 4 is a block diagram of an apparatus for generating a bandwidth extension signal from an input signal; Figure 5a is a schematic representation of a filter bank implementation of a phase vocoder; Figure 5b is a diagram of Figure 5a. Detailed description of the filter; Figure 5c is a schematic diagram for processing the amplitude signal and the frequency signal in the filter channel of Figure 5a; Figure 6 is a conversion implementation of a phase vocoder Outlined;
第7圖是用於從輸入信號產生帶寬擴展信號之一裝置 的方塊圖, 第8圖是用於從輸入信號產生帶寬擴展信號之一裝置 的方塊圖; 第9圖是用於從輸入信號產生帶寬擴展信號之一裝置 的方塊圖; 第10圖是用於基於輸入信號提供帶寬減小信號之一裝 置的方塊圖; 第11圖是一種用於從輸入信號產生帶寬擴展信號之方 10 201007708 法的流程圖; 第12圖是一種用於基於輸入信號提供帶寬減小信號之 方法的流程圖;以及 第13圖是一已知帶寬擴展演算法的概要繪示。 I:實施方式3 在下文中,相同的參考數字被部分地用於具有相同或 類似功能特性的物件及功能單元,且其關於圖式的描述也 將應用到其他圖式,以減小實施例描述的冗餘。 第1圖顯示根據本發明之一實施例的用於從一輸入信 號102產生一帶寬擴展信號122之一裝置100的方塊圖。輸入 信號102的第一頻帶由第一解析度資料表示,及第二頻帶由 第二解析度資料表示,該第二解析度低於該第一解析度。 該裝置100包含連接到一組合器120的一補丁產生器110。該 補丁產生器120受組配以根據第一修補演算法從輸入信號 102的第一頻帶產生第一補丁 112,及根據第二修補演算法 從輸入信號102的第一頻帶產生第二補丁 114。根據第二修 補演算法產生的第二補丁 114的頻譜密度較根據第一修補 演算法產生的第一補丁 112的頻譜密度高。該組合器120組 合該第一補丁 112、該第二補丁 114及輸入信號102的第一頻 帶,以獲得帶寬擴展信號122。再者,用於產生帶寬擴展信 號122的裝置100根據第一修補演算法及根據第二修補演算 法縮放輸入信號102或縮放第一補丁 112及第二補丁 114,藉 此帶寬擴展信號122滿足一頻譜包絡準則。 頻譜密度的意思是例如一頻帶中的不同頻率或頻率線 11 201007708Figure 7 is a block diagram of an apparatus for generating a bandwidth extension signal from an input signal, and Figure 8 is a block diagram of an apparatus for generating a bandwidth extension signal from an input signal; Figure 9 is for generating from an input signal Block diagram of one of the bandwidth extension signals; Figure 10 is a block diagram of one of the means for providing a bandwidth reduction signal based on the input signal; Figure 11 is a method for generating a bandwidth extension signal from the input signal 10 201007708 Figure 12 is a flow diagram of a method for providing a bandwidth reduction signal based on an input signal; and Figure 13 is a schematic illustration of a known bandwidth extension algorithm. I: Embodiment 3 Hereinafter, the same reference numerals are partially used for articles and functional units having the same or similar functional characteristics, and the description thereof with respect to the drawings will also be applied to other drawings to reduce the description of the embodiments. Redundancy. 1 shows a block diagram of an apparatus 100 for generating a bandwidth extension signal 122 from an input signal 102, in accordance with an embodiment of the present invention. The first frequency band of the input signal 102 is represented by a first resolution data, and the second frequency band is represented by a second resolution data, the second resolution being lower than the first resolution. The device 100 includes a patch generator 110 coupled to a combiner 120. The patch generator 120 is configured to generate a first patch 112 from a first frequency band of the input signal 102 in accordance with a first patching algorithm and a second patch 114 from a first frequency band of the input signal 102 in accordance with a second patching algorithm. The second patch 114 generated according to the second patch algorithm has a higher spectral density than the first patch 112 generated according to the first patch algorithm. The combiner 120 combines the first patch 112, the second patch 114, and the first frequency band of the input signal 102 to obtain a bandwidth extension signal 122. Moreover, the apparatus 100 for generating the bandwidth extension signal 122 scales the input signal 102 or scales the first patch 112 and the second patch 114 according to the first patching algorithm and according to the second patching algorithm, whereby the bandwidth extension signal 122 satisfies a Spectrum envelope criteria. Spectral density means, for example, different frequencies or frequency lines in a frequency band 11 201007708
的密度。例如,從OHz到達i〇kHz的一頻帶(包含具有頻率 4kHz及8kHz之頻率部分)較包含具有頻率、4kHz、 6kHz、8kHz及10kHz之頻率部分的相同頻帶具有一較低頻 譜密度。因為第一補丁 112的頻譜密度較第二補丁 114的頻 譜松'度低,與第一補丁 114相比較,第一補丁 112包含間隙。 因此’第二補丁 114可用來填充這些間隙。因為兩個補丁係 基於輸入信號1G2的第-頻帶’兩個補丁都與相對應於輸入 信號102之原始信號的特性有關。因此,較擴展信號122 可能是原始信號的良好近似’且帶寬擴展信號122的主觀品 質或音訊品質可透過❹職賴概念被顯著地提高。以 此方式,較多的能量可分佈在剩餘的線之間,且例如不自 然聲音可被避免。 例如,第-修補演算法可能是__諸波修補演算法。g 此’補丁產生器110可產生第—補丁 112,該第一補丁 包含是輸入信號102之第一頻帶中的頻率的整數倍的步Density. For example, a frequency band from OHz to i 〇 kHz (including a frequency portion having frequencies of 4 kHz and 8 kHz) has a lower spectral density than the same frequency band including frequency portions having frequencies of 4 kHz, 6 kHz, 8 kHz, and 10 kHz. Because the spectral density of the first patch 112 is lower than the frequency of the second patch 114, the first patch 112 includes a gap compared to the first patch 114. Thus the second patch 114 can be used to fill these gaps. Since both patches are based on the first band of the input signal 1G2, both patches are related to the characteristics of the original signal corresponding to the input signal 102. Therefore, the more extended signal 122 may be a good approximation of the original signal' and the subjective quality or audio quality of the bandwidth extension signal 122 can be significantly improved by the concept of the job. In this way, more energy can be distributed between the remaining lines, and for example, unnatural sound can be avoided. For example, the first patching algorithm may be a __wave patching algorithm. g This patch generator 110 may generate a first patch 112 containing steps that are integer multiples of the frequency in the first frequency band of the input signal 102.
率。譜波帶寬擴展可提供原始信號之音調結構的良好纪 似,但是這一修補演算法將把間隙留在 些間隙可由第二補丁填充。例如,第二修補演 -混合修補演算法,這意味著補了產生器⑽可產生第二承 丁 114,該第二補丁114包含是輪入信號收之第—頻帶中# 頻率的整數倍的頻率(請波頻率)及不是輸人信號iG2之第’ 頻帶中的頻率的整數倍的頻率(非諧波頻率)。非諧波 用於填充第-補丁112巾的_。組合整個第二補丁 ▲ 括讀波頻率)與第-補丁 112也是可能的。在這個例子中: 12 201007708 由於組合第一補丁 112與第二補丁 114的諧波頻率部分產生 的諧波頻率的放大可透過合適地縮放第一補丁 1丨2及/或第 二補丁 114予以考慮。 第一補丁 112及第二補丁 114包含至少部分相同的頻率 範圍。例如,第一補丁 112包含從4kHz到達8kHz的一頻帶, 及第二補丁 114包含從6kHz到達10kHz的一頻帶。在根據本 發明的一些實施例中,第一補丁之頻率的一下截止等於第 二補丁之頻率的一下截止,且第一補丁112之頻率的一上截 止等於第二補丁 114之頻率的一上截止。例如,兩個補丁都 包含從4kHz到達8kHz的一頻帶。 第2a圖及第2b圖顯示根據第一修補演算法212的第一 補丁 112及根據第二修補演算法214的第二補丁114。為了較 佳繪示,第2a圖只顯示第一補丁 112,及第2b圖顯示第一補 丁 112及相對應的第二補丁 114。第2a圖繪示輸入信號102之 第一頻帶202及根據第一修補演算法212產生的兩個第一補 丁 112的例子200。在這個例子中,補丁包含與輸入信號102 的第一頻帶202相同的帶寬。帶寬也可以是不同的。輸入信 號102之第一頻帶202的上截止頻率220由“Xover”頻率(交越 頻率)表示。在第2a圖中所示的例子中,補丁在等於交越頻 率Xover 220的多倍的一頻率處開始。第一補丁 112中的頻 率線是輸入信號102之第一頻帶202中的頻率線的整數倍, 且可例如由一相位聲碼器產生。與輸入信號102的第一頻帶 202相比較,這些第一補丁 112包含關於丟失頻率線的間隙。 第2b圖額外地顯示兩個相對應第二補丁 114的例子 13 201007708 250。這些補丁根據第二修補演算法214產生,且包含諧波 及非諧波頻率。非諧波頻率線可用來填充第一補丁 112中的 間隙。第二補丁 114中的頻率線可例如由非線性失真產生。 以此方式,間隙可能不能被任意地填充,例如用雜訊 填充間隙。間隙可基於輸入信號之第一頻帶中的第一解析 度資料,從而基於原始信號來填充。 輸入彳§號102的第一頻帶可代表,例如在高解析度下編 碼之原始音訊信號的低頻帶。輸入信號1〇2的第二頻帶可代 表’例如原始音訊信號的高頻帶,且可在低解析度下透過 例如頻譜包絡資料、雜訊資料及/或丟失諧波資料的一個或 多個參數來量化。原始音訊信號可以是例如一音訊信號, 該音訊信號在處理或編碼之前由麥克風記錄。 根據第一修補演算法及根據第二修補演算法縮放輸入 信號意味著,例如輸入信號在第一補丁被產生之前根據第 一修補演算法被縮放一次,然後該第一修補基於該所縮放 的輸入信號產生,以及輸入信號在第二補丁被產生之前根 據第二修補演算法被縮放一次,然後該第二補丁基於該所 縮放的輪入信號產生,藉此在組合第一補丁、第二補丁與 輸入信號的第一頻帶之後,帶寬擴展信號滿足一頻譜包絡 準則。可選擇性地,第一補丁與第二補丁在其產生以後遭 縮放’藉此帶寬擴展信號也滿足一頻譜包絡準則。同樣地 與第一補丁及第二補丁的縮放相組合,根據第一修補演算 法及根據第二修補演算法縮放輸入信號是可能的。 組合器120可以是例如一加法器,且帶寬擴展信號122 201007708rate. The spectral bandwidth extension provides a good correlation to the pitch structure of the original signal, but this patching algorithm will leave the gap in the gap and can be filled by the second patch. For example, a second patching-mixing patching algorithm, which means that the complement generator (10) can generate a second socket 114 that contains an integer multiple of the # frequency in the first band of the round-in signal. The frequency (wave frequency) and the frequency (non-harmonic frequency) that is not an integer multiple of the frequency in the 'band' of the input signal iG2. Non-harmonic Used to fill the _ of the first patch 112. It is also possible to combine the entire second patch ▲ including the read wave frequency) with the first patch 112. In this example: 12 201007708 The amplification of the harmonic frequencies generated by combining the harmonic frequency portions of the first patch 112 and the second patch 114 can be considered by appropriately scaling the first patch 1丨2 and/or the second patch 114. . The first patch 112 and the second patch 114 contain at least partially the same frequency range. For example, the first patch 112 includes a frequency band that reaches 8 kHz from 4 kHz, and the second patch 114 includes a frequency band that reaches 10 kHz from 6 kHz. In some embodiments according to the present invention, the next cutoff of the frequency of the first patch is equal to the next cutoff of the frequency of the second patch, and an upper cutoff of the frequency of the first patch 112 is equal to an upper cutoff of the frequency of the second patch 114. . For example, both patches contain a band that reaches 8 kHz from 4 kHz. Figures 2a and 2b show a first patch 112 according to the first patching algorithm 212 and a second patch 114 according to the second patching algorithm 214. For better illustration, Figure 2a shows only the first patch 112, and Figure 2b shows the first patch 112 and the corresponding second patch 114. Figure 2a illustrates an example 200 of a first frequency band 202 of the input signal 102 and two first patches 112 generated in accordance with the first patching algorithm 212. In this example, the patch contains the same bandwidth as the first frequency band 202 of the input signal 102. The bandwidth can also be different. The upper cutoff frequency 220 of the first frequency band 202 of the input signal 102 is represented by the "Xover" frequency (crossover frequency). In the example shown in Figure 2a, the patch begins at a frequency equal to a multiple of the crossover frequency Xover 220. The frequency line in the first patch 112 is an integer multiple of the frequency line in the first frequency band 202 of the input signal 102 and may be generated, for example, by a phase vocoder. These first patches 112 contain a gap with respect to the missing frequency line as compared to the first frequency band 202 of the input signal 102. Figure 2b additionally shows an example of two corresponding second patches 114 13 201007708 250. These patches are generated according to the second patching algorithm 214 and contain harmonic and non-harmonic frequencies. Non-harmonic frequency lines can be used to fill the gaps in the first patch 112. The frequency lines in the second patch 114 can be generated, for example, by nonlinear distortion. In this way, the gap may not be arbitrarily filled, such as filling the gap with noise. The gap may be based on the first resolution data in the first frequency band of the input signal to be populated based on the original signal. The first frequency band of input § § 102 may represent, for example, the low frequency band of the original audio signal encoded at high resolution. The second frequency band of the input signal 1 可 2 may represent, for example, the high frequency band of the original audio signal, and may pass through one or more parameters such as spectral envelope data, noise data, and/or lost harmonic data at low resolution. Quantify. The original audio signal can be, for example, an audio signal that is recorded by the microphone prior to processing or encoding. Scaling the input signal according to the first patching algorithm and according to the second patching algorithm means that, for example, the input signal is scaled once according to the first patching algorithm before the first patch is generated, and then the first patch is based on the scaled input Signal generation, and the input signal is scaled according to a second patching algorithm before the second patch is generated, and then the second patch is generated based on the scaled round-in signal, thereby combining the first patch, the second patch, and After the first frequency band of the input signal, the bandwidth extension signal satisfies a spectral envelope criterion. Alternatively, the first patch and the second patch are scaled after they are generated' whereby the bandwidth extension signal also satisfies a spectral envelope criterion. Similarly, in combination with the scaling of the first patch and the second patch, it is possible to scale the input signal according to the first patching algorithm and according to the second patching algorithm. Combiner 120 can be, for example, an adder, and bandwidth extension signal 122 201007708
可能是第一補丁 112 帶的加權和。 第二補丁 114與輸入信號1〇2之第一頻 /滿足頻〜包絡準則意味著例如帶寬擴展信號的頻譜包 各係'基於輸人號所包含的頻譜包絡資料。頻譜包絡資料 可由一編碼器產生且可代表原始信號的第二頻帶。以此方 式W寬擴展信號的頻譜包絡可能是原始信號之頻 譜包絡 的良好近似。 裝置100也可包含用於解碼輸人信號1G2之第-頻帶的 一核心解碼器。 補T產生H 110與組合器⑶可以是例如專門設計硬體 或一處理器或微控制器的—部分,或可以是受組配以在一 電腦或微控制ϋ上執行的電腦程式。裝置励可以是一解碼 器或音訊解碼器的一部分。 第3a圖顯示根據本發明之一實施例的用於從一輸入信 號102產生一帶寬擴展信號122之一裝置300的方塊圖。在這 個例子中,補丁產生器11〇包含用於產生第一補丁 112的一 相位聲碼器310及用於產生第二補丁 114的一振幅剪輯器 320。相位聲碼器31〇與振幅剪輯器32〇連接到組合器12〇。 相位聲碼器31〇可擴展輸入音訊信號1〇2的第一頻帶,以產 生包含諧波頻率的第一補丁 112。在一非線性處理步驟中, 振幅剪輯器320可剪輯輸入信號1〇2,以產生包含諧波與非 諧波頻率的第二補丁114。作為振幅剪輯器320的替代,同 樣地用在特性曲線之二次區域中的半波整流器、全波整流 器、混合器或二極體可用來基於輸入信號102由一非線性處 15 201007708 理步驟產生非諸波頻率。 第3b圖、第3c圖及第3d圖顯示用以產生非譜波頻率的 經剪輯及/或經整流輸入信號102的例子。第3b圖顯示—經 剪輯正弦輸入信號1〇2的概要繪示350。透過剪輯信號,以 信號斜率380之突變形式的不連續點被產生,且具有較高頻 率的諧波與非諧波部分被產生。 可選擇性地’第3c圖顯示也產生不連續點38〇之一經半 波整流正弦輸入信號102的概要繪示360。 再者,剪輯與整流的組合是可能的。第3d圖顯示產生 參 不同不連續點380之一經剪輯及全波整流正弦輸入信號1〇2 的概要繪示370。 ^ 透過剪輯及/或整流或施加產生不連續點380之非線性 處理的其他方法,具有不同頻率的一寬頻譜可被產生。因 此’根據這種修補演算法產生的補丁可包含一高頻譜密度。 第4圖顯示根據本發明之一實施例的用於從一輸入信 號102產生一帶寬擴展信號122之一裝置400的方塊圖。該裝 置400類似於在第3a圖中所示的裝置,但是額外地包含一頻 ® 譜線選擇器410。相位聲碼器310與振幅剪輯器320連接到頻 譜線選擇器410,而該頻譜線選擇器410連接到組合器120。 頻譜線選擇器410可選擇第二補丁 114中的多個頻率線,以 獲得可與第一補丁互補的一經修改第二補丁414。若第一補 丁 112中的相對應頻率線丟失,則第二補丁 114中的頻率線 可被選擇。換言之,頻譜線選擇器410選擇第二補丁 114中 的頻率線,用於填充第一補丁 112中的間隙,且可忽視已包 16 201007708 含在第一補丁 112中的第二補丁 114中的頻率。以此方式, 經修改第二補丁414在已包含在第一補丁 112中的頻率處可 包含間隙。 在這個例子中,組合器120包含第一補丁 ^2、經修改 第二補丁 414及輸入信號102的第一頻帶。 頻譜線選擇益410可以是例如補丁產生器no的一部分 (如在第4圖中所示)或一獨立單元。 在下文中,參考第5圖及第6圖,根據本發明的相位聲 碼器310之可能的實施態樣被緣示。第5a圖顯示〆相位聲碼 器的濾波器組實施態樣’其中一音訊信號被鎮給一輸入端 500且在一輸出端510被獲得。特別地,在第5a圖中所繪示 的概要滤波器組中的每一通道包括—帶通滤波器501及一 下游振盪器502。來自每一通道的所有振盪器的輸出信號透 過一組合器來組合,該組合器被實施為例如一加法器且在 503處予以指示’以獲得輸出信號。每一濾波器5〇1被實施, 藉此其一方面提供一振幅信號,另—方面提供一頻率信 號。該振幅信號與該頻率信號是說明濾波器5〇1中的振幅隨 著時間的逝去之發展的時間信號,而頻率信號代表透過濾 波器501濾波的信號的頻率之發展。 遽波器501的概要設置在第5b圖中予以繪示。第5a圖中 的每一濾波器501可如在第5b圖中那樣設置,然而,其中只 是被提供給兩個輸入混合器551及加法器552的頻率^從通 道到通道是不同的。混合器551的混合器兩個輸出信號都被 低通濾波器553濾波’其中低通信號是不同的,因為它們由 17 201007708 相位偏離90。的本地振盪器頻率(LO頻率)產生。較高低通濾 波器553提供一正交信號554,而較低濾波器553提供一同相 信號555。這兩個信號(即Q及〗)被提供給從直角表示產生幅 度相位表示的一座標轉換器556。第5a圖中的幅度信號或振 幅信號隨著時間的逝去分別在輸出端557被輸出。相位信號 被提供給一相位解開器(unwrapper)558。在元件558的輸出 端’不再存在通常在〇與360°之間的相位值,而存在線性增 加的相位值。這一“所解開的”相位值被提供給一相位/頻率 轉換器559,該相位/頻率轉換器559可被實施為例如一簡單 參 相位差5十算器’該相位差δ十算|§從在一目前時間點的相位 減去一先前時間點的相位,以獲得目前時間點的頻率值, 或用於獲得相位變異量之近似的任何其他裝置。這一頻率 值被加到濾波器通道1中的恆定頻率值&,以在輸出端56〇獲 得一時變頻率。在輸出端56〇處的頻率值具有一直流分量 及一交流分量=濾波器通道中之信號的目前頻率脫離平均 - 頻率的頻率偏移。May be the weighted sum of the first patch 112. The first frequency/satisfaction-envelope-envelope criterion of the second patch 114 and the input signal 1〇2 means that, for example, the spectrum packet of the bandwidth extension signal is based on the spectral envelope data contained in the input number. The spectral envelope data may be generated by an encoder and may represent a second frequency band of the original signal. The spectral envelope of the spread signal in this way may be a good approximation of the spectral envelope of the original signal. Apparatus 100 may also include a core decoder for decoding the first frequency band of input signal 1G2. The T-generation H 110 and the combiner (3) may be, for example, a portion specially designed for hardware or a processor or a microcontroller, or may be a computer program that is configured to execute on a computer or micro-controller. The device excitation can be part of a decoder or audio decoder. Figure 3a shows a block diagram of an apparatus 300 for generating a bandwidth extension signal 122 from an input signal 102, in accordance with an embodiment of the present invention. In this example, the patch generator 11A includes a phase vocoder 310 for generating the first patch 112 and an amplitude clipper 320 for generating the second patch 114. The phase vocoder 31 and the amplitude clipper 32 are connected to the combiner 12A. The phase vocoder 31 〇 expands the first frequency band of the input audio signal 1 〇 2 to produce a first patch 112 containing harmonic frequencies. In a non-linear processing step, the amplitude clipper 320 can clip the input signal 1〇2 to produce a second patch 114 containing harmonic and non-harmonic frequencies. As an alternative to the amplitude clipper 320, a half-wave rectifier, full-wave rectifier, mixer or diode similarly used in the quadratic region of the characteristic curve can be used to generate a nonlinearity based on the input signal 102 by a non-linearity 15 201007708 step. Non-wave frequencies. Figures 3b, 3c, and 3d show examples of clipped and/or rectified input signals 102 used to generate non-spectral frequencies. Figure 3b shows a summary 350 of the clipped sinusoidal input signal 1〇2. By clipping the signal, discontinuities in the form of abrupt changes in signal slope 380 are generated, and harmonic and non-harmonic portions having higher frequencies are generated. Optionally, Figure 3c shows a schematic depiction 360 of a half-wave rectified sinusoidal input signal 102 that also produces a discontinuous point 38〇. Furthermore, a combination of clipping and rectification is possible. Figure 3d shows a summary depiction 370 of one of the different discontinuities 380 and the full-wave rectified sinusoidal input signal 1〇2. ^ By clipping and/or rectifying or applying other methods of non-linear processing that produce discontinuous points 380, a wide spectrum of different frequencies can be generated. Therefore, patches generated according to this patching algorithm can include a high spectral density. Figure 4 shows a block diagram of an apparatus 400 for generating a bandwidth extension signal 122 from an input signal 102, in accordance with an embodiment of the present invention. This device 400 is similar to the device shown in Figure 3a, but additionally includes a frequency ® line selector 410. Phase vocoder 310 and amplitude clipper 320 are coupled to spectral line selector 410, which is coupled to combiner 120. The spectral line selector 410 can select a plurality of frequency lines in the second patch 114 to obtain a modified second patch 414 that is complementary to the first patch. If the corresponding frequency line in the first patch 112 is lost, the frequency line in the second patch 114 can be selected. In other words, the spectral line selector 410 selects the frequency line in the second patch 114 for filling the gap in the first patch 112, and can ignore the frequency of the second patch 114 included in the first patch 112 that has been wrapped 16 201007708 . In this manner, the modified second patch 414 can include a gap at the frequencies already included in the first patch 112. In this example, combiner 120 includes a first patch ^2, a modified second patch 414, and a first frequency band of input signal 102. The spectral line selection benefit 410 can be, for example, part of the patch generator no (as shown in Figure 4) or a separate unit. In the following, with reference to Figures 5 and 6, possible implementations of phase vocoder 310 in accordance with the present invention are illustrated. Figure 5a shows a filter bank implementation of a phase vocoder. One of the audio signals is clocked to an input 500 and obtained at an output 510. In particular, each channel in the summary filter bank illustrated in Figure 5a includes a bandpass filter 501 and a downstream oscillator 502. The output signals from all of the oscillators of each channel are combined by a combiner that is implemented, for example, as an adder and indicated at 503 to obtain an output signal. Each filter 5〇1 is implemented whereby it provides an amplitude signal on the one hand and a frequency signal on the other. The amplitude signal and the frequency signal are time signals indicating the evolution of the amplitude in the filter 5〇1 with time, and the frequency signal represents the development of the frequency of the signal filtered by the filter 501. The outline of the chopper 501 is shown in Figure 5b. Each of the filters 501 in Fig. 5a can be set as in Fig. 5b, however, only the frequency supplied from the two input mixers 551 and the adder 552 is different from the channel to the channel. The two output signals of the mixer of the mixer 551 are both filtered by the low pass filter 553 'where the low pass signals are different because they are phase shifted by 90 from 17 201007708. The local oscillator frequency (LO frequency) is generated. The higher low pass filter 553 provides a quadrature signal 554 and the lower filter 553 provides an in-phase signal 555. These two signals (i.e., Q and 〖) are provided to a scale converter 556 that produces a magnitude phase representation from a right angle representation. The amplitude or amplitude signals in Figure 5a are output at output 557, respectively, over time. The phase signal is provided to a phase unwrapper 558. At the output end of element 558 there is no longer a phase value typically between 〇 and 360°, and there is a linearly increasing phase value. This "unwrapped" phase value is provided to a phase/frequency converter 559 which can be implemented, for example, as a simple parametric phase difference 5 voltameter 'the phase difference δ ten calculation| § subtracting the phase of a previous time point from the phase at a current time point to obtain the frequency value at the current time point, or any other means for obtaining an approximation of the phase variation. This frequency value is applied to the constant frequency value & in filter channel 1 to obtain a time varying frequency at output 56. The frequency value at the output 56〇 has a DC component and an AC component = the current frequency of the signal in the filter channel deviates from the average-frequency frequency offset.
因此,如在第5a圖及第5b圖中所繪示的,相位聲竭n Q 實現頻譜資訊與時間資訊的分離。頻譜資訊被包含在特定 通=或頻率0,其提供每-通道之頻率的直流部分,而時 1貝民刀别被i含在頻率偏移或隨著時間的逝之幅度演化 中〇 第5C圖顯示根據本發明的—處理,該處理被執行用於 產生’特別地使用相位聲碼器⑽來產生第一補丁且較詳 細地被插入在第5碉中所繪示電路的虛線位置。 18 201007708 對於時間縮放而言,例如每一通道中的振幅信號A(t) 或每一通道中之信號f(t)的頻率可被降頻或内插。為了達到 轉置目的,因為其對本發明有用,一内插(及信號A⑴與f(t) 的時間延伸或擴展)被執行,以獲得擴展信號A,⑴及f⑴, 其中該内插受擴展因數598控制。例如,該擴展因數可被選 擇’藉此相位聲碼器產生諸波頻率。透過相位變化的内插, 即透過加法器552加入恆定頻率之前的值,第5a圖中的每個 ^ 振盈器5〇2的頻率沒有被改變。然而,總音訊信號的時間改 變被減慢,即透過因數2。結果是具有原始補丁的一時間擴 展音調’即具有其諧波的原始基本波。 ’ 透過執行在第5c圖中所繪示的信號處理,音訊信號可 被縮小回到其原始期間,例如透過以一因數2的降頻,而所 有頻率同時被加倍。這導致一因數2補丁轉置,然而,其中 ' 一音訊信號被獲得,該音訊信號與原始音訊信號具有相同 的長度(即相同的樣本數目)。 Φ 作為在第5a®中所繪示濾波器組實施態樣的替代,— 相位聲碼器的轉換實施態樣也可如在第6圖中所描述那樣 使用。在這裡,音訊信號698被饋給一FFT處理器,或較一 般地,一短時傅利葉轉換(STFT)處理器6〇〇中作為一時間樣 本序列。FFT處理器_被實施,以執行音訊信號的時_ 窗化’用以而後透過後續的FFT計算幅度頻譜以及相位頻續 兩者,其中這-計算針對連續頻言普予以執行,該連續頻^ 與強重疊的音訊信號區塊有關。 曰 在一極端情況下,針對每-新音訊信號樣本,—新頻 19 201007708 譜可被計算’其中-新頻譜也可只針對每-個第二十個新 樣本來6十算。在兩個頻譜之間的樣本的距離‘a,較佳地由控 制器602給出。控制器6〇2進一步被實施,以饋給一贿處 理器該IFFT處理器6〇4被實施以在一重疊相加操作中 操作。特別地’ IFFT處理器6〇4被實施,藉此其基於一幅度 頻譜與一相位頻譜透過每一頻譜執行一個IFFT來執行一短 時傅立葉反轉換’用以而後執行一重疊相加操作以獲得 產生的時間信號。重疊相加操作受組配以消除由分析視窗 引起的區塊效應 0 時間信號的時間擴展透過在被11?1^處理器6〇4處理時 兩個頻譜之間的距離‘b,大於用於產生FFT頻譜的頻譜之間 的距離a來實現。基本思想是透過較分析FFT只被進一步 隔開的反FFT來擴展音訊信號。因此,所合成音訊信號中的 頻譜改變較原始音訊信號中的較緩慢地發生。Therefore, as depicted in Figures 5a and 5b, phase exhaustion n Q separates the spectral information from the temporal information. The spectrum information is included in the specific pass = or frequency 0, which provides the DC portion of the frequency per channel, while the 1 民 刀 is not included in the frequency offset or the evolution of the time lapse of the 5C The figure shows a process according to the invention, which is performed for generating a 'specially used phase vocoder (10) to generate a first patch and inserted in more detail in the dashed position of the circuit depicted in the fifth. 18 201007708 For time scaling, for example, the amplitude signal A(t) in each channel or the frequency of the signal f(t) in each channel can be down or interpolated. In order to achieve the transposition purpose, as it is useful for the present invention, an interpolation (and time extension or extension of signals A(1) and f(t)) is performed to obtain spread signals A, (1) and f(1), where the interpolation is spread factor 598 control. For example, the spreading factor can be selected 'by this phase vocoder to generate the wave frequencies. Through the interpolation of the phase change, that is, the value before the constant frequency is added through the adder 552, the frequency of each of the vibrators 5〇2 in Fig. 5a is not changed. However, the time change of the total audio signal is slowed down by a factor of two. The result is a one-time extended pitch with the original patch', the original fundamental wave with its harmonics. By performing the signal processing illustrated in Figure 5c, the audio signal can be reduced back to its original period, e.g., by a factor of 2 down-conversion, and all frequencies are simultaneously doubled. This results in a factor 2 patch transpose, however, where 'an audio signal is obtained, the audio signal has the same length (i.e., the same number of samples) as the original audio signal. Φ As an alternative to the implementation of the filter bank shown in 5a®, the phase vocoder conversion implementation can also be used as described in Figure 6. Here, the audio signal 698 is fed to an FFT processor, or more generally, a short time Fourier transform (STFT) processor 6 作为 as a time sample sequence. The FFT processor_ is implemented to perform time-windowing of the audio signal for subsequent calculation of the amplitude spectrum and phase repetition through subsequent FFTs, wherein this calculation is performed for successive frequencies, the continuous frequency ^ Associated with strongly overlapping audio signal blocks.曰 In an extreme case, for each new audio signal sample, the new frequency 19 201007708 spectrum can be calculated 'where - the new spectrum can also be calculated for each of the twentieth new samples. The distance 'a of the samples between the two spectra is preferably given by controller 602. The controller 〇2 is further implemented to feed a bribe processor. The IFFT processor 〇4 is implemented to operate in an overlap-and-add operation. In particular, the 'IFFT processor 〇4 is implemented, whereby it performs a short-time Fourier inverse conversion by performing an IFFT on each spectrum based on an amplitude spectrum and a phase spectrum, and then performs an overlap-and-add operation to obtain The generated time signal. The overlap-and-add operation is assembled to eliminate the block effect caused by the analysis window. The time spread of the time signal is greater than the distance 'b between the two spectra when processed by the 11?1^ processor 6〇4. This is achieved by generating the distance a between the spectra of the FFT spectrum. The basic idea is to extend the audio signal by an inverse FFT that is only further separated by the analysis FFT. Therefore, the spectral change in the synthesized audio signal occurs more slowly than in the original audio signal.
然而,若在方塊606中沒有相位重新縮放,這將導致頻 率人工因素。例如當其中45。的連續相位值被實施的一單頻 窗被考慮時,這意味著這-濾波||組巾的信號的相位以1/8 G 週期之一速率增加,及按每—時間區間45。,其中這裡的時 間區間是連續FFT之間的時間區間。若現在該反FFT彼此被 較遠地隔開,這意味著45。的相位增加橫跨一較長時間區間 發生。這意味著這一信號部分的頻率偶然地被修改。為了 肩除這種人工因素,相位被完全相同的因數重新縮放,其 中音訊信號透過該因數在時間上被擴展。每一FFT頻譜值的 相位因此以因數b/a被增加,藉此偶然頻率修改被消除。 20 201007708 儘管在第5c圖中所繪示的實施例中,對於在第5a圖之 慮波器組實施態樣中的一個信號振盪器而言,擴展透過振 幅/頻率控制信號的内插被實現,但是第6圖中的擴展透過 大於兩個FFT頻譜之間之距離的兩個11?17丁頻譜之間的距離 (即‘b’大於‘a’)來實現,然而,其中為了預防人工因素,相 位重新縮放根據比例‘b/a’被執行。例如距離‘b,可被選擇, 藉此相位聲碼器產生諧波頻率。 第7圖顯示根據本發明之一實施例的用於從一輸入信 號102產生帶寬擴展信號122之一裝置7〇〇的方塊圖。該裝置 700類似於在第1圖中所示的裝置,但是包含一功率控制器 710、第一功率調整裝置720及第二功率調整裝置730。功率 控制器710連接到第一功率調整裝置72〇及第二功率調整裝 置730。第一功率調整裝置72〇與第二功率調整裝置73〇連接 到補丁產生器110。功率控制器71〇可基於輸入信號所包含 的頻譜包絡資料及基於輸入信號所包含的補丁縮放控制資 料控制根據第一與第二修補演算法之輸入信號的縮放。可 選擇性地,不是輸入信號中所包含的補丁縮放控制資料, 而疋至少一個所儲存的補丁縮放控制參數可被使用。補丁 縮放控制參數可被補了縮放控制參數記憶體儲存,該補丁 縮放控制參數記憶體可以是功率控制器71〇的一部分或一 獨立單元。第一功率調整裝置72〇可根據第一修補演算法縮 放輸入信號102’而第二功率調整裝置73()可根據第二修補 演算法縮放輸入k號102。換言之,輸入信號1〇2可被預處 理,藉此第-與第二補丁可被產生,藉此帶寬擴展信號滿 21 201007708 足頻譜包絡準則。為此,頻譜包絡資料可定義帶寬擴展信 號122的頻譜包絡,且補丁縮放控制資料或補丁縮放控制參 數可設定第一補丁 112與第二補丁 114之間的比例,或可設 定第一補丁 112及/或第二補丁 114的絕對值。如在第7圖中所 示,第一功率調整裝置720與第二功率調整裝置730可以是 功率控制器710的一部分或者獨立單元。亦如在第7圖中所 示,功率控制器710可以是補丁產生器no的一部分或者_ 獨立單元。功率調整裴置720、730可以是例如受功率控制 器710控制的放大器或濾波器。 ❿ 可選擇性地’縮放在補丁產生以後被完成。合適地, 第8圖顯示根據本發明之一實施例的用於從一輸入信號1〇2 產生一帶寬擴展信號122之一裝置800的方塊圖。該裝置8〇〇 類似於在第7圖中所示的裝置,但是功率調整裝置720、73〇 遭配置在補丁產生器11〇與組合器12〇之間。在這個例子 中’補丁產生器110連接到第一功率調整裝置72〇及連接到 第二功率調整裝置73〇。第一功率調整裝置72〇與第二功率 調整裝置730連接到組合器12〇。以此方式,第一補丁 112可 ® 根據第一修補演算法透過第一功率調整裝置72〇來縮放,而 第二補丁 114可根據第二修補演算法透過第二功率調整裝 置730來縮放。該等功率調整裝置基於如前所述的頻譜包絡 資料及補丁縮放控制資料或補丁縮放控制參數再次受功率 控制器710控制。 可選擇性地,同樣地只縮放或功率調整兩個補丁中的 一個,接著在組合所組合補丁與輸入信號1〇2之第—頻帶之 22 201007708 前透過組合器12 0組合補丁及縮放所組合補丁是町能的。換 5之,第一個補丁可被縮放,以在兩個補丁之間實現一預 定義比例(例如基於補丁縮放控制資料),然後所組合補丁被 縮放(例如基於頻譜包絡資料),以滿足頻譜包絡準則。 補丁縮放控制資料可包含例如用於功率分佈縮放的一 簡單因數或多個參數。補丁縮放控制資料可指系例如在全 部第二頻帶或全部高頻帶上的第一補丁與第二補丁之間的 _ 功率比或在全部第二頻帶或全部高頻帶上的第〆補丁及/ - 或第二補丁之功率的絕對值且可由至少一個參數表示。可 選擇性地,針對一起構成第二頻帶或高頻帶之多個子頻帶 中的每一子頻帶,補丁縮放資料包含一因數,例如類似於 頻譜帶寬複製應用中的每一子頻帶的頻譜包絡資料。可選 . 擇性地,補丁縮放資料也可指示濾波器的轉換函數。例如, 用於縮放第一補丁的濾波器之轉換函數中的參數及/或用 於縮放第一補丁的渡波器之轉換函數中的參數^包含在輸 ^ 入k號中。以此方式,參數可代表一頻率函數。另一備選 可以是代表第一補丁與第二補丁之微分函數的補丁縮放控 制參數。根據這個實施例,輸入信號的縮放或第一補丁與 第二補丁的縮放可基於包含至少一個參數的補丁縮放控制 資料。 第9圖顯示根據本發明之一實施例的用於從一輸入信 號102產生帶寬擴展信號122之一裝置900的方塊圖。該裝置 900類似於在第8圖中所示的裝置,但是額外地包含一雜訊 加法器910、一丟失諧波加法器920、一雜訊功率調整裝置 23 201007708 940及一丢失諧波功率調整裝置950。雜訊加法器910連接到 雜訊功率調整裝置940,雜訊功率調整裝置940連接到組合 器120 °丢失諧波加法器920連接到丟失諧波功率調整裝置 950 ’丢失諧波功率調整裝置95〇連接到組合器12〇。再者, 功率控制器710連接到雜訊功率調整裝置94〇及丟失諧波功 率調整裝置950。雜訊加法器910可基於輸入信號1〇2所包含 的雜訊資料產生一雜訊補丁912。 雜訊補丁912可透過雜訊功率調整裝置940來縮放。功 率控制器710可基於輸入信號102所包含的頻譜包絡資料及 參 /或雜訊縮放資料控制雜訊功率調整裝置94〇。以此方式, 原始信號的雜訊可被近似,以提高帶寬擴展信號的音訊品 質。 丟失諸波加法器920可基於輸入信號所包含的丟失諧 波資料產生一丟失諧波補丁922。該丟失諧波補丁922可包 含諸波頻率’該等諧波頻率可只在原始信號的高頻帶中發 生’因此若只是關於輸入信號1〇2之第一頻帶之原始信號之 低頻帶的資訊是可得的’不能被再現。丟失諧波資料可提 〇 供有關這些丟失諧波的資訊。丟失諧波補丁922可透過丟失 諧波功率調整裝置950來縮放。功率控制器710可基於頻譜 包絡資料或基於輸入信號1〇2所包含的丟失諧波縮放資料 控制丟失諧波功率調整裝置950。 組合器120可組合第一補丁112、第二補丁114、輸入信 號102的第一頻帶、雜訊補丁912及丟失諧波補丁922,以獲 得帶寬擴展信號122。功率控制器710與功率調整裝置組合 24 201007708 可基於頻譜包絡資料縮放第一補丁 112、第二補丁 114、雜 訊補丁912及丟失諧波補丁922,藉此頻譜包絡準則被滿足。 第10圖顯示根據本發明之一實施例的用於基於一輸入 信號1002提供一帶寬減小信號1032之一裝置1000的方塊 圖。該裝置1000包含一頻譜包絡資料決定器1010、一補丁 縮放控制資料產生器1020及一輸出介面1030。頻譜包絡資 料決定器1010與補丁縮放控制資料產生器1020連接到輸出 介面1030。頻譜包絡資料決定器1010可基於輸入信號10〇2 9 \ 的一高頻帶決定頻譜包絡資料1012。補丁縮放控制資料產 生器1020可產生補丁縮放控制資料1〇22,該補丁縮放控制 資料1022用於在解碼器端縮放帶寬減小信號1032或用於透 過解碼器縮放第一補丁及第二補丁,藉此由該解碼器產生 的帶寬擴展信號滿足一頻譜包絡準則。該頻譜包絡準則係 基於頻譜包絡資料。第一補丁根據第一修補演算法從帶寬 減小信號1032的第一頻帶產生,而第二補丁根據第二修補 | 演算法從帶寬減小信號1032的第一頻帶產生。根據第二修 補演算法產生的第二補丁的頻譜密度較根據第一修補演算 法產生的第一補丁的頻譜密度高。輸出介面1030組合輸入 信號1002的一低頻帶、頻譜包絡資料1012及補丁縮放控制 資料1022,以獲得帶寬減小信號1032。再者,輸出介面1030 提供帶寬減小信號1032用於傳輸或儲存。 裝置1000也可能包含用於編碼輸入信號之低頻帶的一 核心編碼器。該核心編碼器可以是例如一差動編碼器、一 滴編碼器或一知覺音訊編碼器。 25 201007708 編μ的Γ ‘切料轉#付之 的一部分。補丁縮放控 = 資料可指示例如在全部高頻帶上衫數。補丁縮放控制 間的-功率比或在全部高頻帶上的第—補τ與第-補丁之 丁之功率的絕對值且可由至少 τ&/或第二補 地,補丁_料包含針對-起構成:高=:= 帶中的每-子頻帶所決定的一因 ^之多個子頻However, if there is no phase rescaling in block 606, this will result in a frequency artifact. For example, when 45. When a single frequency window in which the continuous phase value is implemented is considered, this means that the phase of the signal of the -filtered || tissue is increased at a rate of 1/8 G period, and is 45 per time interval. Where the time interval is the time interval between consecutive FFTs. If the inverse FFTs are now separated from each other far away, this means 45. The phase increase occurs across a longer time interval. This means that the frequency of this signal portion is accidentally modified. To overcome this artifact, the phase is rescaled by exactly the same factor, by which the audio signal is spread over time. The phase of each FFT spectral value is therefore increased by a factor b/a, whereby accidental frequency modifications are eliminated. 20 201007708 Although in the embodiment illustrated in Figure 5c, the interpolation of the extended transmission amplitude/frequency control signal is implemented for a signal oscillator in the embodiment of the wave filter set of Figure 5a. , but the extension in Figure 6 is achieved by the distance between the two 11?17-but spectra of the distance between the two FFT spectra (ie, 'b' is greater than 'a'), however, in order to prevent artificial factors The phase rescaling is performed according to the ratio 'b/a'. For example, the distance 'b can be selected, whereby the phase vocoder produces a harmonic frequency. Figure 7 shows a block diagram of a device 7 for generating a bandwidth extension signal 122 from an input signal 102, in accordance with an embodiment of the present invention. The device 700 is similar to the device shown in Figure 1, but includes a power controller 710, a first power adjustment device 720, and a second power adjustment device 730. The power controller 710 is coupled to the first power adjustment device 72 and the second power adjustment device 730. The first power adjustment device 72A and the second power adjustment device 73A are connected to the patch generator 110. The power controller 71 can control the scaling of the input signals according to the first and second patching algorithms based on the spectral envelope data contained in the input signal and based on the patch scaling control information contained in the input signal. Alternatively, instead of the patch scaling control data contained in the input signal, at least one of the stored patch scaling control parameters can be used. The patch scaling control parameter may be supplemented by a scaling control parameter memory store, which may be part of the power controller 71 or a separate unit. The first power adjustment means 72 can scale the input signal 102' according to the first patching algorithm and the second power adjustment means 73() can scale the input k number 102 according to the second patching algorithm. In other words, the input signal 1 〇 2 can be pre-processed whereby the first and second patches can be generated whereby the bandwidth extension signal is full 21 201007708 full spectrum envelope criterion. To this end, the spectrum envelope data may define a spectrum envelope of the bandwidth extension signal 122, and the patch scaling control data or the patch scaling control parameter may set a ratio between the first patch 112 and the second patch 114, or may set the first patch 112 and / or the absolute value of the second patch 114. As shown in Figure 7, the first power adjustment device 720 and the second power adjustment device 730 can be part of a power controller 710 or a separate unit. As also shown in Figure 7, power controller 710 can be part of patch generator no or a separate unit. Power conditioning devices 720, 730 can be, for example, amplifiers or filters that are controlled by power controller 710. ❿ Selectively 'scaling' is done after the patch is generated. Suitably, Figure 8 shows a block diagram of an apparatus 800 for generating a bandwidth extension signal 122 from an input signal 1 〇 2 in accordance with an embodiment of the present invention. The device 8 is similar to the device shown in Fig. 7, but the power adjusting devices 720, 73 are disposed between the patch generator 11 and the combiner 12A. In this example, the patch generator 110 is connected to the first power adjustment device 72 and to the second power adjustment device 73. The first power adjustment device 72A and the second power adjustment device 730 are connected to the combiner 12A. In this manner, the first patch 112 can be scaled by the first power adjustment device 72 according to the first patching algorithm, and the second patch 114 can be scaled by the second power adjustment device 730 according to the second patching algorithm. The power adjustment means are again controlled by the power controller 710 based on the spectral envelope data and the patch scaling control data or the patch scaling control parameters as previously described. Alternatively, only one of the two patches can be scaled or power adjusted, and then combined with the combination of the patch and the zoom by the combiner 120 before combining the combined patch with the first band of the input signal 1〇2, 201007708 The patch is the town. In the fifth case, the first patch can be scaled to achieve a predefined ratio between the two patches (eg, based on patch scaling control data), and then the combined patches are scaled (eg, based on spectral envelope data) to meet the spectrum. Envelope criteria. The patch scaling control data may include, for example, a simple factor or a plurality of parameters for power distribution scaling. The patch scaling control data may refer to, for example, a _ power ratio between a first patch and a second patch on all second frequency bands or all high frequency bands or a third patch on all second frequency bands or all high frequency bands and/or Or the absolute value of the power of the second patch and may be represented by at least one parameter. Alternatively, for each of the plurality of sub-bands that together comprise the second or high frequency band, the patch scaling data includes a factor, such as spectral envelope data similar to each sub-band in the spectral bandwidth replication application. Optional. Optionally, the patch scaling data also indicates the filter's conversion function. For example, the parameter in the conversion function of the filter for scaling the first patch and/or the parameter ^ in the conversion function of the waver for scaling the first patch are included in the input k number. In this way, the parameters can represent a frequency function. Another alternative may be a patch scaling control parameter that represents a differential function of the first patch and the second patch. According to this embodiment, the scaling of the input signal or the scaling of the first patch and the second patch may be based on patch scaling control data including at least one parameter. Figure 9 shows a block diagram of an apparatus 900 for generating a bandwidth extension signal 122 from an input signal 102, in accordance with an embodiment of the present invention. The device 900 is similar to the device shown in FIG. 8, but additionally includes a noise adder 910, a lost harmonic adder 920, a noise power adjustment device 23 201007708 940, and a lost harmonic power adjustment. Device 950. The noise adder 910 is connected to the noise power adjustment device 940, and the noise power adjustment device 940 is connected to the combiner 120°. The loss harmonic adder 920 is connected to the lost harmonic power adjustment device 950 'Lost harmonic power adjustment device 95〇 Connected to the combiner 12〇. Furthermore, the power controller 710 is connected to the noise power adjustment device 94 and the lost harmonic power adjustment device 950. The noise adder 910 can generate a noise patch 912 based on the noise information contained in the input signal 1〇2. The noise patch 912 can be scaled by the noise power adjustment device 940. The power controller 710 can control the noise power adjustment device 94 based on the spectral envelope data and the reference or noise scaling data contained in the input signal 102. In this way, the noise of the original signal can be approximated to improve the audio quality of the bandwidth extension signal. The lost wave adder 920 can generate a missing harmonic patch 922 based on the missing harmonic data contained in the input signal. The lost harmonic patch 922 can include wave frequencies 'the harmonic frequencies can occur only in the high frequency band of the original signal'. Therefore, if only the low frequency band of the original signal of the first frequency band of the input signal 1〇2 is The available 'can't be reproduced. Loss of harmonic data provides information about these missing harmonics. The lost harmonic patch 922 can be scaled by the lost harmonic power adjustment device 950. The power controller 710 can control the lost harmonic power adjustment device 950 based on the spectral envelope data or based on the lost harmonic scaling data contained in the input signal 1〇2. The combiner 120 can combine the first patch 112, the second patch 114, the first frequency band of the input signal 102, the noise patch 912, and the lost harmonic patch 922 to obtain the bandwidth extension signal 122. The power controller 710 is combined with the power adjustment device. 24 201007708 The first patch 112, the second patch 114, the noise patch 912, and the missing harmonic patch 922 can be scaled based on the spectral envelope data, whereby the spectral envelope criteria are satisfied. Figure 10 shows a block diagram of an apparatus 1000 for providing a bandwidth reduction signal 1032 based on an input signal 1002, in accordance with an embodiment of the present invention. The apparatus 1000 includes a spectral envelope data determiner 1010, a patch scaling control data generator 1020, and an output interface 1030. The spectral envelope data determiner 1010 is coupled to the patch scale control data generator 1020 to the output interface 1030. The spectral envelope data determinator 1010 can determine the spectral envelope data 1012 based on a high frequency band of the input signal 10 〇 2 9 \ . The patch scaling control data generator 1020 can generate patch scaling control data 1102 for scaling the bandwidth reduction signal 1032 at the decoder side or for scaling the first patch and the second patch through the decoder. Thereby the bandwidth extension signal produced by the decoder satisfies a spectral envelope criterion. The spectral envelope criteria are based on spectral envelope data. The first patch is generated from the first frequency band of the bandwidth reduction signal 1032 according to the first patching algorithm, and the second patch is generated from the first frequency band of the bandwidth reduction signal 1032 according to the second patching algorithm. The second patch generated according to the second patch algorithm has a higher spectral density than the first patch generated according to the first patching algorithm. The output interface 1030 combines a low frequency band of the input signal 1002, the spectral envelope data 1012, and the patch scaling control data 1022 to obtain a bandwidth reduction signal 1032. Furthermore, the output interface 1030 provides a bandwidth reduction signal 1032 for transmission or storage. Apparatus 1000 may also include a core encoder for encoding the low frequency band of the input signal. The core encoder can be, for example, a differential encoder, a drop encoder or a perceptual audio encoder. 25 201007708 Edited by μ ‘cutting the turn# part of the payment. Patch Scale Control = Data can indicate, for example, the number of shirts on all high bands. The patch-to-control control-to-power ratio or the absolute value of the power of the first-complement τ and the first-patch on all high-bands and may be composed of at least τ&/or second complement, patch_inclusive :High =:= Multiple sub-frequency of one factor determined by each sub-band in the band
例如類似於頻譜帶寬 複製應用中的母—子頻帶的頻譜包絡資料。可選擇性地, 補丁縮放資料也可指示濾的轉換函數。例如,用於縮 放第-補T的濾波H之轉換錢巾的參數及/或用於縮放 第二補丁的濾、波器之轉換函數中的參數可被蚊用於產 生補丁縮放控制資料。以此方式,參數可基於—頻率函數 產生ϋ選可以S產生代表第—補丁與第二補丁之微 分函數的補丁縮放控制參數。 ❹ 補丁縮放控制資料1022可透過分析輸入信號1〇〇2及基 於對輸入信號1002的分析選擇儲存在—補丁縮放控制參數 記憶體中的補丁縮放控制參數來產生,以獲得補丁縮放控 制資料1022。 可選擇性地’補丁縮放控制 > 料1 〇22的產生可透過一 分析合成方法來實現。為此’補丁縮放控制資料產生器1〇2〇 可額外地包含一補丁產生器(如針對解碼器之所述)及一比 較器。該補丁產生器可根據第一修補演算法從輸入信號 1002的低頻帶產生第一補丁,及根據第二修補演算法從輸 26 201007708 入馆號1002的低頻帶產生第二補丁。根據第二修補演算法 產生的第二補丁的頻譜密度較根據第一修補演算法產生的 第—補丁的頻譜密度高。該比較器可比較該第一補丁、該 第二補丁及該輸入信號的高頻帶,以獲得補丁縮放控制資 料1022。換言之,鈿述概念也被應用到裝置丨〇〇〇。以此方 式,裝置1000可透過比較補丁或所組合補丁與輸入信號來 擷取補丁縮放控制資料1022,其中該輸入信號可能是例如 ❻ 一原始音訊仏號。此外,裝置1000也可包含如前所述的一 ; 頻β普線選擇器、一功率控制器、一雜訊加法器及/或一丟失 諧波加法器。以此方式,雜訊資料、雜訊補丁縮放控制資 料、丟失諧波資料及/或丟失諧波補丁縮放控制資料也可透 過一種分析合成方法來擷取。 根據本發明的一些實施例與包含第一頻帶及第二頻帶 的一音訊信號有關。第一頻帶由第一解析度資料表示,且 第二頻帶由第二解析度資料表示,其中第二解析度低於第 Q 一解析度。第二解析度資料係基於第二頻帶的頻譜包絡資 料及第二頻帶的補丁縮放控制資料,該補丁縮放控制資料 用於在解碼器端縮放音訊信號或用於透過解碼器縮放第一 補丁及第二補丁,藉此透過該解碼器產生的帶寬擴展信號 滿足一頻譜包絡準則。該頻譜包絡準則係基於頻譜包絡資 料。第一補丁根據第一修補演算法從音訊信號的第一頻帶 產生’及第二補丁根據第二修補演算法從音訊信號的第一 頻帶產生。根據第二修補演算法產生的第二補丁的頻譜密 度較根據第一修補演算法產生的第一補丁的頻譜密度高。 27 201007708 音訊信號可以疋例如原始音訊信號為基礎的帶寬 減小信號。音訊信號的第〜頻帶可代表在高解析度下編碼 之原始音訊信號的一低頻帶。音訊信號的第二頻帶可代表 原始音訊信號的一咼頻帶,且可透過至少兩個參數來量 化:由頻譜包絡資料表示的〜頻譜包絡參數及由補丁縮放 控制資料表㈣-補丁縮故控制參數。基於這種音訊信 號’根據上述概念的解碼器可產生一帶寬擴展信號,該帶 寬擴展信號提供原始音訊信說的良好近似且與已知概念相 比較具有提高的音訊品質。 ® 第11圖顯不根據本發明之一實施例的用於從一輸入信 號產生-帶寬擴展信號之—方法丨的流程圖。該輸人冑 號的第-頻帶由第-解析度資料表示,及第二頻帶由第二 解析度資料表示,該第二解析度低於該第—解析度。财 法1100包含以下步驟:產生⑽第_補丁、產生112〇第二 補丁、縮放1130輸入彳§號或縮放113〇第一補丁及第二補丁 以及組合114〇第一補丁、第二補丁與輸入信號的第一頻 ▼ ’以獲得帶寬擴展信號。第一補丁根據第一修補演算法 ® 從輸入h號的第一頻帶產生111〇,及第二補丁根據第二修 補廣算法從輸入信號的第1帶產生112〇。根據第二修補 及算法產生山⑽第工補了的賴密度較根據第 一修補演 算法產生1U0的第一補丁的頻譜密度高。輸入信號可根據 第修補演算法及根據第二修補演算法來縮放113〇,或第 一補丁及第二補丁可被縮放113〇 ’藉此帶寬擴展信號滿足 一頻谱包絡準則。 28 201007708 再者,方法1100可透過根據上述概念的步驟來擴展。 方法1100可例如被實現為用於在一電腦或微控制器上執行 的電腦程式。 第12圖顯示根據本發明之一實施例的用於基於一輪人 信號提供一帶寬減小信號之一方法1200的流程圖。該方法 1200包含以下步驟:基於輸入信號的一高頻帶決定121〇頻 譜包絡資料、產生1220補丁縮放控制資料、組合1230輸入 信號的低頻帶、頻譜包絡資料與補丁縮放控制資料,以獲 得帶寬減小信號以及提供1240該帶寬減小信號用於傳輸或 儲存。補丁縮放控制資料被產生1220,其用於在解碼器端 縮放帶寬減小信號或用於透過解碼器縮放第一補丁及第二 補丁’藉此透過該解碼器產生的帶寬擴展信號滿足一頻譜 包絡準則。該頻譜包絡準則係基於該頻譜包絡資料。第一 補丁根據第一修補演算法從帶寬減小信號的一低頻帶產 生’及第二補丁根據第二修補演算法從帶寬減小信號的該 低頻帶產生。根據第二修補演算法產生的第二補丁的頻譜 密度較根據第一修補演算法產生的第一補丁的頻譜密度 南。 再者’方法1200可透過根據上述概念的步驟來擴展。 方法1200可例如被實現為用於在一電腦或微控制器上執行 的電腦程式。 根據本發明的一些實施例與一種裝置有關,該裝置用 於使用用於帶寬擴展的一相位聲碼器與用於一較密集頻譜 的非線性失真或雜訊填充相組合來產生一帶寬擴展信號。 29 201007708 當施加用於帶寬擴展的相位聲碼器時,頻率線進一步遠 離。若例如透過量化使_巾存在_,财過擴展同= 間隙甚至被增加。在能量調適中’頻譜中的剩餘線接收太 多的能量。這透過用雜訊或另外的諧波填充間隙來避免, 該等雜訊或另外的諸波可能由信號的非線性失真獲得。這 樣’較多的能量可能分佈在剩餘線之間。透過將頻帶中的 能量只集中到少數幾條頻率線,產生不自然或金屬似的聲 音。以前較多頻帶中的能量被加和到剩餘的一些。For example, it is similar to the spectral envelope data of the mother-subband in the spectrum bandwidth replication application. Alternatively, the patch scaling data may also indicate a filtered conversion function. For example, the parameters of the conversion wiper used to scale the filter H of the first complement T and/or the parameters in the filter function of the filter used to scale the second patch may be used by the mosquito to generate patch scale control data. In this manner, the parameter can generate a patch scaling control parameter that can generate a differential function representative of the first patch and the second patch based on the frequency function.补丁 Patch scaling control data 1022 can be generated by analyzing input signal 1〇〇2 and selecting patch scaling control parameters stored in the patch scaling control parameter memory based on analysis of input signal 1002 to obtain patch scaling control data 1022. Alternatively, the 'patch scaling control' > material 1 〇22 generation can be achieved by an analytical synthesis method. To this end, the patch scaling control data generator 1 〇 2 〇 may additionally include a patch generator (as described for the decoder) and a comparator. The patch generator may generate a first patch from the low frequency band of the input signal 1002 according to the first patching algorithm, and generate a second patch from the low frequency band of the access number 1002 according to the second patching algorithm. The second patch generated according to the second patching algorithm has a higher spectral density than the first patch generated according to the first patching algorithm. The comparator compares the first patch, the second patch, and the high frequency band of the input signal to obtain patch scaling control information 1022. In other words, the concept of narration is also applied to the device. In this manner, device 1000 can retrieve patch scaling control data 1022 by comparing patches or combined patches and input signals, which may be, for example, an original audio nickname. In addition, apparatus 1000 can also include a frequency beta line selector, a power controller, a noise adder, and/or a loss harmonic adder as previously described. In this way, noise data, noise patch scaling control data, lost harmonic data, and/or missing harmonic patch scaling control data can also be retrieved through an analytical synthesis method. Some embodiments in accordance with the present invention are associated with an audio signal comprising a first frequency band and a second frequency band. The first frequency band is represented by a first resolution data, and the second frequency band is represented by a second resolution data, wherein the second resolution is lower than the Qth resolution. The second resolution data is based on the spectral envelope data of the second frequency band and the patch scaling control data of the second frequency band, the patch scaling control data is used for scaling the audio signal at the decoder end or for scaling the first patch and the first through the decoder The second patch, whereby the bandwidth extension signal generated by the decoder satisfies a spectral envelope criterion. This spectral envelope criterion is based on spectral envelope data. The first patch is generated from the first frequency band of the audio signal according to the first patching algorithm and the second patch is generated from the first frequency band of the audio signal according to the second patching algorithm. The spectral density of the second patch generated according to the second patching algorithm is higher than the spectral density of the first patch generated according to the first patching algorithm. 27 201007708 Audio signals can reduce the signal based on, for example, the original audio signal. The first band of the audio signal may represent a low frequency band of the original audio signal encoded at high resolution. The second frequency band of the audio signal may represent a frequency band of the original audio signal and may be quantized by at least two parameters: a spectral envelope parameter represented by the spectral envelope data and a patch scaling control data table (4) - a patching control parameter . A decoder based on such an audio signal' based on the above concept can generate a bandwidth extension signal that provides a good approximation of the original audio message and has improved audio quality compared to known concepts. ® Figure 11 shows a flow chart of a method for generating a -bandwidth spread signal from an input signal in accordance with an embodiment of the present invention. The first frequency band of the input signal is represented by the first resolution data, and the second frequency band is represented by the second resolution data, and the second resolution is lower than the first resolution. The financial method 1100 includes the steps of: generating (10) a _ patch, generating 112 〇 a second patch, scaling a 1130 input 彳§ number or zooming 113 〇 a first patch and a second patch, and combining 114 〇 a first patch, a second patch, and an input The first frequency of the signal ▼ 'to obtain a bandwidth extension signal. The first patch generates 111〇 from the first frequency band of the input h number according to the first patching algorithm, and the second patch generates 112〇 from the first band of the input signal according to the second patching algorithm. According to the second patching and algorithm generation, the density of the patch (10) is higher than that of the first patch that generates 1U0 according to the first patching algorithm. The input signal may be scaled according to a first patching algorithm and according to a second patching algorithm, or the first patch and the second patch may be scaled 113' to thereby satisfy a spectral envelope criterion. 28 201007708 Again, method 1100 can be extended by the steps in accordance with the concepts above. Method 1100 can be implemented, for example, as a computer program for execution on a computer or microcontroller. Figure 12 shows a flow diagram of a method 1200 for providing a bandwidth reduction signal based on a round of human signals, in accordance with an embodiment of the present invention. The method 1200 includes the steps of: determining a 121 〇 spectral envelope data based on a high frequency band of the input signal, generating 1220 patch scaling control data, combining a low frequency band of the 1230 input signal, spectral envelope data, and patch scaling control data to obtain a bandwidth reduction The signal and the 1240 bandwidth reduction signal are provided for transmission or storage. The patch scaling control data is generated 1220 for scaling the bandwidth reduction signal at the decoder side or for scaling the first patch and the second patch by the decoder 'by which the bandwidth extension signal generated by the decoder satisfies a spectral envelope Guidelines. The spectral envelope criterion is based on the spectral envelope data. The first patch is generated from a low frequency band of the bandwidth reduction signal according to the first patching algorithm' and the second patch is generated from the low frequency band of the bandwidth reduction signal according to the second patching algorithm. The spectral density of the second patch generated according to the second patching algorithm is greater than the spectral density of the first patch generated according to the first patching algorithm. Further, the method 1200 can be extended by the steps according to the above concept. Method 1200 can be implemented, for example, as a computer program for execution on a computer or microcontroller. Some embodiments in accordance with the present invention are related to an apparatus for generating a bandwidth extension signal using a phase vocoder for bandwidth extension in combination with nonlinear distortion or noise filling for a denser spectrum . 29 201007708 When a phase vocoder for bandwidth expansion is applied, the frequency line is further away. If, for example, the _ towel exists in the _, the financial expansion is equal to the gap = even increased. The remaining lines in the spectrum in the energy adaptation receive too much energy. This is avoided by filling the gap with noise or another harmonic, which may be obtained by nonlinear distortion of the signal. Such more energy may be distributed between the remaining lines. Unnatural or metallic sounds are produced by concentrating the energy in the frequency band to only a few frequency lines. In the past, the energy in more bands was added to the rest.
若頻譜中不存在間隙,但是至少雜訊是存在的,則一 部分能量保持在雜訊基準中。透過應用非線性失真,頻譜 可再次被增加密度,一方面透過失真所產生的雜訊,另二 方面透過*合適選擇將*真的信號部分所引人的另外的諸 波部分。 所以帶寬擴展信號可以是例如一經濾波失真信號與在 相位聲碼器幫助下產生之一信號的加權和。換言之,帶寬 擴展信號可以是第-補丁、第二補τ與輪人信號之第:頻 帶的加權和。 根據本發明的一些實施例與適用於所有音1應用 概念有關’其中在該等音訊應用中不是全部帶寬都是可用 的。例如,對於使用數位無線電服務之音 言,網際網路串流化或其他音訊通訊應用 可被應用。 訊内容的廣播而 、所描述的概念 儘管本發明已依據若干實施例予 入本發明之範圍内的改變、變更、及等效。 以描述,但是存在落 也應注意的是, 30 201007708 存在實施本發明之方法與組成的許多備選方法。因此所意 欲的是以下所附申請專利範圍被理解為包括落入本發明之 真實精神與範圍内的所有這些改變、變更、及等效。If there is no gap in the spectrum, but at least the noise is present, then some of the energy remains in the noise reference. By applying nonlinear distortion, the spectrum can be increased in density again, on the one hand, through the noise generated by the distortion, and on the other hand, through the appropriate selection of the other parts of the wave that are introduced by the *true signal portion. The bandwidth extension signal can therefore be, for example, a weighted sum of a filtered distortion signal and one of the signals produced with the help of the phase vocoder. In other words, the bandwidth extension signal can be the weighted sum of the first patch, the second complement τ, and the wheel signal: the frequency band. Some embodiments in accordance with the present invention are related to the application of all audio 1 application concepts where not all of the bandwidth is available in the audio applications. For example, for the use of digital radio services, Internet streaming or other audio communication applications can be applied. </ RTI> <RTIgt; </ RTI> <RTIgt; </ RTI> <RTIgt; </ RTI> <RTIgt; </ RTI> <RTIgt; </ RTI> <RTIgt; By way of description, but it should be noted that 30 201007708 there are many alternative ways of implementing the method and composition of the present invention. It is therefore intended that the appended claims be interpreted as
特別地,其指出的是,本發明方案依據情況也可在軟 體中實施。實施可能是在一數位儲存媒體上,特別是具有 電氣可讀控制信號的一軟式磁碟或一CD,其中該等電氣可 漬控制信號可與一可規劃電腦系統協同工作,藉此相對應 的方法被執行。一般而言,本發明因此也存在於一電腦程 式產品中,該電腦程式產品具有儲存在機器可讀載體上的 弋竭备3亥電腦程式產品在一電腦上被執行時,該程式 焉用於執行本發明方法。換句話說,本發明因此也可被實 見為具有㈣碼的電腦程式,當該電腦程式產品在一電腦 上被執行時,該程式碼用於執行本方法。 【陶式簡單說明】 第1囫是用於從輪入信號產生帶寬擴展信號之-裝置 的方塊圖; 第2a圖是-所產生第—補丁的概要繪示; 第2b圖是-所產生第一及第二補丁的概要緣示; 第3a圖是用於從輸人㈣產生帶寬擴展信號之 的方塊圖; 第3b圖是輯正弦輸人信號的概要繪示; =圖是-經半波整流正弦輪人信號的概要繪示; ^圖是—經域錢正弦輪人信_概要緣示; 4圖是用於從輸入信號產生帶寬擴展信號之-裝置 31 201007708 的方塊圖, 第5 a圖是一相位聲碼器之濾波器組實施態樣的概要繪 示; 第5b圖是第5a圖中之濾波器的詳細繪示; 第5c圖是用於處理在第5a圖之一濾波器通道中的振幅 信號與頻率信號的概要繪示; 第6圖是一相位聲碼器之轉換實施態樣的概要繪示;In particular, it is pointed out that the solution of the invention can also be implemented in software depending on the situation. The implementation may be on a digital storage medium, in particular a floppy disk or a CD having an electrically readable control signal, wherein the electrically smear control signals can cooperate with a programmable computer system, whereby corresponding The method is executed. In general, the present invention is also embodied in a computer program product having a deliberately stored computer program product stored on a machine readable carrier for execution on a computer. The method of the invention is carried out. In other words, the present invention can thus also be embodied as a computer program having a (four) code for performing the method when the computer program product is executed on a computer. [Simple Description] The first block is a block diagram of the device for generating a bandwidth extension signal from the wheeled signal; the second block is a summary of the generated first patch; the second block is - the generated A summary of the first and second patches; Figure 3a is a block diagram for generating a bandwidth extension signal from the input (4); Figure 3b is a schematic representation of the sinusoidal input signal; = Figure is - half-wave A schematic diagram of the rectified sinusoidal human signal; ^ is a graph of the sinusoidal sinusoidal _ summary; 4 is a block diagram of the device 31 201007708 for generating a bandwidth extension signal from the input signal, 5a The figure is a schematic diagram of a filter bank implementation of a phase vocoder; Figure 5b is a detailed depiction of the filter in Figure 5a; Figure 5c is a filter used to process a filter in Figure 5a A schematic diagram of the amplitude signal and the frequency signal in the channel; FIG. 6 is a schematic diagram showing a conversion implementation of a phase vocoder;
第7圖是用於從輸入信號產生帶寬擴展信號之一裝置 的方塊圖, 第8圖是用於從輸入信號產生帶寬擴展信號之一裝置 的方塊圖; 第9圖是用於從輸入信號產生帶寬擴展信號之一裝置 的方塊圖, 第10圖是用於基於輸入信號提供帶寬減小信號之一裝 置的方塊圖;Figure 7 is a block diagram of an apparatus for generating a bandwidth extension signal from an input signal, and Figure 8 is a block diagram of an apparatus for generating a bandwidth extension signal from an input signal; Figure 9 is for generating from an input signal a block diagram of one of the bandwidth extension signals, and FIG. 10 is a block diagram of an apparatus for providing a bandwidth reduction signal based on the input signal;
第11圖是一種用於從輸入信號產生帶寬擴展信號之方 法的流程圖, 第12圖是一種用於基於輸入信號提供帶寬減小信號之 方法的流程圖;以及 第13圖是一已知帶寬擴展演算法的概要繪示。 【主要元件符號說明】 102.. .輸入信號 1312、1314··.補丁 110.. .補丁產生器 120…組合器 112、114、414、200、250、 122...帶寬擴展信號 32 201007708Figure 11 is a flow chart of a method for generating a bandwidth extension signal from an input signal, and Figure 12 is a flow chart of a method for providing a bandwidth reduction signal based on an input signal; and Figure 13 is a known bandwidth A summary of the extended algorithm. [Description of main component symbols] 102.. Input signal 1312, 1314··. Patch 110.. Patch generator 120... Combiner 112, 114, 414, 200, 250, 122... Bandwidth extension signal 32 201007708
202.. .頻帶 212、214…修補演算法 220、1320.·.截止頻率/X〇ver 100、300、400、700、800、 900、1000...裝置 310.. .相位聲碼器 320.. .振幅剪輯器 350…剪輯正弦輸入信號 360…半波整流正弦輸入信號 370…經剪輯及全波整流正弦 輸入信號 380.. .不連續點 410.. .頻譜線選擇器 500.. .輸入端 501.. .帶通濾波器 502.. .下游振盪器 503、552…加法器 510、557、560...輪出端 551.. .輸入混合器 553.. .低通濾波器 554…正交信號 555…同相信號 556.. .座標轉換器 558.. .相位解開器 559…相位/頻率轉換器 598.. .擴展因數 600…短時傅利葉轉換(SXFT) 處理器 602.. .控制器 604.. .1.FT 處理器 606…方塊 698…音訊信號 710…功率控制器 720、730…功率調整裝置 910…雜訊加法器 912…雜訊補丁 920·.·丟失諧波加法器 922.. .丟失諧波補丁 940···雜訊功率調整裝置 950…丟失諧波功率調整裝置 1002.. .帶寬減小信號 1010.. .頻譜包絡資料決定器 1012.. .頻譜包絡資料 1020.. .補丁縮放控制資料產 生器 1022.. .補丁縮放控制資料 1030.. .輸出介面 1032.. .帶寬減小信號 33 201007708 1100、1200...方法 1302···信號的低頻帶 1110~1140、1210~1240…流程 步驟 1300...帶寬擴展202.. . Band 212, 214... patching algorithm 220, 1320.. cutoff frequency / X〇ver 100, 300, 400, 700, 800, 900, 1000... device 310.. phase vocoder 320 .. Amplitude Clipper 350... Clip Sine Input Signal 360... Half Wave Rectified Sinusoidal Input Signal 370... Clipped and Full Wave Rectified Sinusoidal Input Signal 380.. Discontinuity Point 410.. Spectrum Line Selector 500.. . Input 501.. Bandpass Filter 502.. Downstream Oscillator 503, 552... Adder 510, 557, 560... Round Out End 551.. Input Mixer 553.. Low Pass Filter 554 ... quadrature signal 555... in-phase signal 556.. coordinate converter 558.. phase unwinder 559... phase/frequency converter 598.. expansion factor 600... short time Fourier transform (SXFT) processor 602. . controller 604..1. FT processor 606...block 698...audio signal 710...power controller 720,730...power adjustment device 910...noise adder 912...noise patch 920···loss of harmonics Adder 922.. Lost Harmonic Patch 940··· Noise Power Adjustment Device 950... Lost Harmonic Power Adjustment Device 1002.. Bandwidth Reduction Signal 1010.. Spectrum Envelope Data Determinator 1012.. Spectrum Envelope Data 1020.. Patch Zoom Control Data Generator 1022.. Patch Zoom Control Data 1030.. Output Interface 1032.. Bandwidth Reduction Signal 33 201007708 1100, 1200 ...method 1302···the low frequency band of the signal 1110~1140, 1210~1240...flow step 1300...bandwidth expansion
3434
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