Spectral Translation/Folding in the Subband Domain
The present invention relates to a new method and apparatus for improvement of High Frequency Reconstruction (HFR) techniques using frequency translation or folding or a combination thereof. The proposed invention is applicable to audio source coding systems, and offers significantly reduced computational complexity. This is accomplished by means of frequency translation or folding in the subband domain, preferably integrated with spectral envelope adjustment in the same domain. The concept of dissonance guard-band filtering is further presented. The proposed invention offers a low-complexity, intermediate quality HFR method useful in speech and natural audio coding applications.
Latest Dolby Labs Patents:
- Coding and decoding of interleaved image data
- Nested entropy encoding
- Method for encoding/decoding an intra-picture prediction mode using two intra-prediction mode candidate, and apparatus using such a method
- Systems and methods for covariance smoothing
- Methods, apparatus and systems for generation, transportation and processing of immediate playout frames (IPFs)
This application is a continuation of U.S. patent application Ser. No. 15/370,054 filed Dec. 6, 2016, which is a continuation of U.S. patent application Ser. No. 14/964,836 filed Dec. 10, 2015, now U.S. Pat. No. 9,548,059, issued on Jan. 17, 2017, which is a continuation of U.S. patent application Ser. No. 13/969,708 filed Aug. 19, 2013, now U.S. Pat. No. 9,245,534, issued on Jan. 26, 2016, which is a continuation of U.S. patent application Ser. No. 13/460,797 filed Apr. 30, 2012, now U.S. Pat. No. 8,543,232, issued on Sep. 24, 2013, which is a continuation of U.S. patent application Ser. No. 12/703,553 filed Feb. 10, 2012, now U.S. Pat. No. 8,412,365, issued on Apr. 2, 2013, which is a continuation of U.S. patent application Ser. No. 12/253,135 filed Oct. 16, 2008, now U.S. Pat. No. 7,680,552, issued on Mar. 16, 2010, which is a continuation of U.S. patent application Ser. No. 10/296,562 filed Jan. 6, 2004, now U. S. Pat. No. 7,483,753, issued on Jan. 27, 2009, which is a national-stage entry of International patent application no. PCT/SE01/01171 filed May 23, 2001, which claims the benefit of International application no.0001926-5 filed on May 23, 2000, all of which are hereby incorporated by reference.
TECHNICAL FIELDThe present invention relates to a new method and apparatus for improvement of High Frequency Reconstruction (HFR) techniques, applicable to audio source coding systems. Significantly reduced computational complexity is achieved using the new method. This is accomplished by means of frequency translation or folding in the subband domain, preferably integrated with the spectral envelope adjustment process. The invention also improves the perceptual audio quality through the concept of dissonance guard-band filtering. The proposed invention offers a low-complexity, intermediate quality HFR method and relates to the PCT patent Spectral Band Replication (SBR) [WO 98/57436].
BACKGROUND OF THE INVENTIONSchemes where the original audio information above a certain frequency is replaced by gaussian noise or manipulated lowband information are collectively referred to as High Frequency Reconstruction (HFR) methods. Prior-art HFR methods are, apart from noise insertion or non-linearities such as rectification, generally utilizing so-called copy-up techniques for generation of the highband signal. These techniques mainly employ broadband linear frequency shifts, i.e. translations, or frequency inverted linear shifts, i.e. foldings. The prior-art HFR methods have primarily been intended for the improvement of speech codec performance. Recent developments in highband regeneration using perceptually accurate methods, have however made HFR methods successfully applicable also to natural audio codecs, coding music or other complex programme material, PCT patent [WO 98/57436]. Under certain conditions, simple copy-up techniques have shown to be adequate when coding complex programme material as well. These techniques have shown to produce reasonable results for intermediate quality applications and in particular for codec implementations where there are severe constraints for the computational complexity of the overall system.
The human voice and most musical instruments generate quasistationary tonal signals that emerge from oscillating systems. According to Fourier theory, any periodic signal may be expressed as a sum of sinusoids with frequencies ƒ, 2ƒ, 3ƒ, 4ƒ, 5ƒ etc. where ƒ is the fundamental frequency. The frequencies form a harmonic series. Tonal affinity refers to the relations between the perceived tones or harmonics. In natural sound reproduction such tonal affinity is controlled and given by the different type of voice or instrument used. The general idea with HFR techniques is to replace the original high frequency information with information created from the available lowband and subsequently apply spectral envelope adjustment to this information. Prior-art HFR methods create highband signals where tonal affinity often is uncontrolled and impaired. The methods generate non-harmonic frequency components which cause perceptual artifacts when applied to complex programme material. Such artifacts are referred to in the coding literature as “rough” sounding and are perceived by the listener as distortion.
Sensory dissonance (roughness), as opposed to consonance (pleasantness), appears when nearby tones or partials interfere. Dissonance theory has been explained by different researchers, amongst others Plomp and Levelt [“Tonal Consonance and Critical Bandwidth” R. Plomp, W. J. M. Levelt JASA , Vol 38, 1965], and states that two partials are considered dissonant if the frequency difference is within approximately 5 to 50% of the bandwidth of the critical band in which the partials are situated. The scale used for mapping frequency to critical bands is called the Bark scale. One bark is equivalent to a frequency distance of one critical band. For reference, the function
can be used to convert from frequency (f) to the bark scale (z). Plomp states that the human auditory system can not discriminate two partials if they differ in frequency by approximately less than five percent of the critical band in which they are situated, or equivalently, are separated less than 0.05 Bark in frequency. On the other hand, if the distance between the partials are more than approximately 0.,5 Bark, they will be perceived as separate tones.
Dissonance theory partly explains why prior-art methods give unsatisfactory performance. A set of consonant partials translated upwards in frequency may become dissonant. Moreover, in the crossover regions between instances of translated bands and the lowband the partials can interfere, since they may not be within the limits of acceptable deviation according to the dissonance-rules.
SUMMARY OF THE INVENTIONThe present invention provides a new method and device for improvements of translation or folding techniques in source coding systems. The objective includes substantial reduction of computational complexity and reduction of perceptual artifacts. The invention shows a new implementation of a subsampled digital filter bank as a frequency translating or folding device, also offering improved crossover accuracy between the lowband and the translated or folded bands. Further, the invention teaches that crossover regions, to avoid sensory dissonance, benefits from being filtered. The filtered regions are called dissonance guard-bands, and the invention offers the possibility to reduce dissonant partials in an uncomplicated and accurate manner using the subsampled filterbank.
The new filterbank based translation or folding process may advantageously be integrated with the spectral envelope adjustment process. The filterbank used for envelope adjustment is then used for the frequency translation or folding process as well, in that way eliminating the need to use a separate filterbank or process for spectral envelope adjustment. The proposed invention offers a unique and flexible filterbank design at a low computational cost, thus creating a very effective translation/folding/envelope-adjusting system.
In addition, the proposed invention is advantageously combined with the Adaptive Noise-Floor Addition method described in PCT patent [SE00/00159]. This combination will improve the perceptual quality under difficult programme material conditions.
The proposed subband domain based translation of folding technique comprise the following steps:
- filtering of a lowband signal through the analysis part of a digital filterbank to obtain a set of subband signals;
- repatching of a number of the subband signals from consecutive lowband channels to consecutive highband channels in the synthesis part of a digital filterbank;
- adjustment of the patched subband signals, in accordance to a desired spectral envelope; and
- filtering of the adjusted subband signals through the synthesis part of a digital filterbank, to obtain an envelope adjusted and frequency translated or folded signal in a very effective way.
Attractive applications of the proposed invention relates to the improvement of various types of intermediate quality codec applications, such as MPEG 2 Layer III, MPEG 2/4 AAC, Dolby AC-3, NTT TwinVQ, AT&T/Lucent PAC etc. where such codecs are used at low bitrates. The invention is also very useful in various speech codecs such as G. 729 MPEG-4 CELP and HVXC etc to improve perceived quality. The above codecs are widely used in multimedia, in the telephone industry, on the Internet as well as in professional multimedia applications.
The present invention is described by way of illustrative examples, not limiting the scope or spirit of the invention, with reference to the accompanying drawings, in which:
New filter bank based translating or folding techniques will now be described. The signal under consideration is decomposed into a series of subband signals by the analysis part of the filterbank. The subband signals are then repatched, through reconnection of analysis—and synthesis subband channels, to achieve spectral translation or folding or a combination thereof.
In the illustrative, but not limiting, descriptions below it is assumed that an L-channel filter bank splits the input signal x(n) into L subband signals. The input signal, with sampling frequency ƒs, is bandlimited to frequency ƒc. The analysis filters of a maximally decimated filter bank (
The reconstruction range start channel, denoted M, is determined by
The number of source area channels is denoted S (1≦S≦M). Performing spectral reconstruction through translation on {circumflex over (x)}(n) according to the present invention, in combination with envelope adjustment, is accomplished by repatching the subband signals as
vM+k(n)=eM+k(n)vM−S−P+k(n), (3)
where kε[0, S−1], (−1)S+P=1, i.e. S+P is an even number, P is an integer offset (0≦P≦M−S) and eM+k(n) is the envelope correction. Performing spectral reconstruction through folding on {circumflex over (x)}(n) according to the present invention, is further accomplished by repatching the subband signals as
vM+k(n)=eM+k(n)v*M−P−S−k(n), (4)
where kε[0, S−1], (−1)S+P=−1, i.e. S+P is an odd integer number, P is an integer offset (1−S≦P≦M−2S+1) and eM+k(n) is the envelope correction. The operator [*] denotes complex conjugation. Usually, the repatching process is repeated until the intended amount of high frequency bandwidth is attained.
It should be noted that, through the use of the subband domain based translation and folding, improved crossover accuracy between the lowband and instances of translated or folded bands is achieved, since all the signals are filtered through filterbank channels that have matched frequency responses.
If the frequency ƒc of x(n) is too high, or equivalently ƒs is too low, to allow an effective spectral reconstruction, i.e. M+S>L, the number of subband channels may be increased after the analysis filtering. Filtering the subband signals with a QL-channel synthesis filter bank, where only the L lowband channels are used and the upsampling factor Q is chosen so that QL is an integer value, will result in an output signal with sampling frequency Qƒs. Hence, the extended filter bank will act as if it is an L-channel filter bank followed by an upsampler. Since, in this case, the L(Q−1) highband filters are unused (fed with zeros), the audio bandwidth will not change—the filter bank will merely reconstruct an upsampled version of {circumflex over (x)}(n). If, however, the L subband signals are repatched to the highband channels, according to Eq.(3) or (4), the bandwidth of {circumflex over (x)}(n) will be increased. Using this scheme, the upsampling process is integrated in the synthesis filtering. It should be noted that any size of the synthesis filter bank may be used, resulting in different sampling rates of the output signal.
Referring to
Using the same analysis filterbank and an input signal with the same frequency contents,
Sensory dissonance may develop in the translation or folding process due to adjacent band interference, i.e. interference between partials in the vicinity of the crossover region between instances of translated bands and the lowband. This type of dissonance is more common in harmonic rich, multiple pitched programme material. In order to reduce dissonance, guard-bands are inserted and may preferably consist of small frequency bands with zero energy, i.e. the crossover region between the lowband signal and the replicated spectral band is filtered using a bandstop or notch filter. Less perceptual degradation will be perceived if dissonance reduction using guard-bands is performed. The bandwidth of the guard-bands should preferably be around 0.5 Bark. If less, dissonance may result and if wider, comb-filter-like sound characteristics may result.
In filterbank based translation or folding, guard-bands could be inserted and may preferably consist of one or several subband channels set to zero. The use of guardbands changes Eq.(3) to
vM+D+k(n)=eM+D+k(n)vM−S−P+k(n) (5)
vM+D+k(n)=eM+D+k(n)v*M−P−S−k(n). (6)
D is a small integer and represents the number of filterbank channels used as guardband. Now P+S+D should be an even integer in Eq.(5) and an odd integer in Eq.(6). P takes the same values as before.
In order to make the spectral envelope continuous, the dissonance guard-bands may be partially reconstructed using a random white noise signal, i.e. the subbands are fed with white noise instead of being zero. The preferred method uses Adaptive Noise-floor Addition (ANA) as described in the PCT patent application [SE00/00159]. This method estimates the noise-floor of the highband of the original signal and adds synthetic noise in a well-defined way to the recreated highband in the decoder.
Practical ImplementationsThe present invention may be implemented in various kinds of systems for storage or transmission of audio signals using arbitrary codecs.
The above-described embodiments are merely illustrative for the principles of the present invention for improvement of High Frequency Reconstruction (HFR) techniques using filterbank-based frequency translation or folding. It is understood that modifications and variations of the arrangements and the details described herein will be apparent to others skilled in the art. It is the intent, therefore, to be limited only by the scope of the impending patent claims and not by the specific details presented by way of description and explanation of the embodiments herein.
Claims
1. A method for decoding coded signals, the coded signals comprising a coded lowband audio signal and coded envelope data, comprising:
- separating the coded lowband audio signal from the coded signals;
- audio decoding the coded lowband audio signal to obtain a decoded audio signal;
- decoding the coded envelope data to obtain decoded envelope data;
- obtaining an envelope adjusted and frequency-translated signal, comprising: filtering the decoded audio signal using an analysis filterbank to obtain complex-valued subband signals within a source range, wherein each complex-valued subband signal is represented by a real-valued component and an imaginary-valued component;
- patching the real-valued component and the imaginary-valued component of a complex-valued subband signal with index i within the source range to a complex-valued subband signal with index j within a reconstruction range, wherein the source range comprises frequencies lower than frequencies in the reconstruction range;
- patching the real-valued component and the imaginary-valued component of a complex-valued subband signal with index i+1 within the source range to a complex-valued subband signal with index j+1 within a reconstruction range;
- applying an envelope adjustment to the patched complex-valued subband signals within the reconstruction range in response to the coded envelope data; and
- filtering the patched and envelope adjusted complex-valued subband signals within the reconstruction range using a synthesis filterbank to obtain the envelope adjusted and frequency translated signal.
2. A method according to claim 1, wherein the analysis filterbank and the synthesis filterbank are obtained by cosine or sine modulation of a lowpass prototype filter.
3. A method according to claim 1, wherein the analysis filterbank and the synthesis filterbank are obtained by complex-exponential-modulation of a lowpass prototype filter.
4. A method according to claim 2, wherein the lowpass prototype filter is designed so that a transition band of channels of the analysis filterbank and the synthesis filterbank overlaps a passband of neighbouring channels only.
5. A method according to claim 1, in which the synthesis filterbank comprises a dissonance guard band, the dissonance guard band being positioned between synthesis filterbank channels in the source range and synthesis filterbank channels in the reconstruction range.
6. A method according to claim 5, in which one or several of the channels in the dissonance guard band are fed with zeros or gaussian noise; whereby dissonance related artifacts are attenuated.
7. A method according to claim 5, in which a bandwidth of the dissonance guard band is approximately one half Bark.
8. A method according to claim 1, in which the step of patching implements a first iteration step, and in which the method further comprises another step of patching implementing a second iteration step, wherein in the second iteration step, subband signals within the source range for the second iteration step comprise the subband signals within the reconstruction range for the first iteration step.
9. A decoder for decoding coded signals, the coded signals comprising a coded lowband audio signal and coded envelope data, comprising:
- a separator for separating the coded lowband audio signal from the coded signals;
- an audio decoder for audio decoding the coded lowband audio signal to obtain a decoded audio signal;
- an envelope data decoder for decoding the coded envelope data to obtain decoded envelope data;
- an apparatus for obtaining an envelope adjusted and frequency-translated signal, comprising: an analysis filterbank for filtering the decoded audio signal using an analysis filterbank to obtain complex-valued subband signals within a source range, wherein each complex-valued subband signal is represented by a real-valued component and an imaginary-valued component,
- a high frequency reconstruction/envelope adjustment unit for:
- patching the real-valued component and the imaginary-valued component of a complex-valued subband signal with index i within the source range to a complex-valued subband signal with index j within a reconstruction range, wherein the source range comprises frequencies lower than frequencies in the reconstruction range;
- patching the real-valued component and the imaginary-valued component of a complex-valued subband signal with index i+1 within the source range to a complex-valued subband signal with index j+1 within a reconstruction range; and
- applying an envelope adjustment to the patched complex-valued subband signals within the reconstruction range in response to the decoded envelope data; and
- a synthesis filterbank for filtering the patched and envelope adjusted complex-valued subband signals within the reconstruction range using a synthesis filterbank to obtain the envelope adjusted and frequency translated signal.
10. A decoder according to claim 9, in which the coded signals further comprise envelope data,
- in which the separator is further arranged to separate the envelope data from the coded signals,
- wherein the decoder further comprises an envelope decoder for decoding the envelope data to obtain spectral envelope information,
- wherein the spectral envelope information is fed to the apparatus for obtaining an envelope adjusted and frequency-translated signal and is used to apply the spectral envelope adjustment.
11. A non-transitory computer readable storage medium comprising a sequence of instructions which, when executed by a processing device, cause the processing device to perform a method for decoding coded signals, the coded signals comprising a coded lowband audio signal and coded envelope data, comprising:
- separating the coded lowband audio signal from the coded signals;
- audio decoding the coded lowband audio signal to obtain a decoded audio signal;
- decoding the coded envelope data to obtain decoded envelope data;
- obtaining an envelope adjusted and frequency-translated signal, comprising: filtering the decoded audio signal using an analysis filterbank to obtain complex-valued subband signals within a source range, wherein each complex-valued subband signal is represented by a real-valued component and an imaginary-valued component;
- patching the real-valued component and the imaginary-valued component of a complex-valued subband signal with index i within the source range to a complex-valued subband signal with index j within a reconstruction range, wherein the source range comprises frequencies lower than frequencies in the reconstruction range;
- patching the real-valued component and the imaginary-valued component of a complex-valued subband signal with index i+1 within the source range to a complex-valued subband signal with index j+1 within a reconstruction range;
- applying an envelope adjustment to the patched complex-valued subband signals within the reconstruction range in response to the coded envelope data; and
- filtering the patched and envelope adjusted complex-valued subband signals within the reconstruction range using a synthesis filterbank to obtain the envelope adjusted and frequency translated signal.
Type: Application
Filed: Mar 1, 2017
Publication Date: Jun 22, 2017
Applicant: Dolby International AB (Amsterdam Zuidoost)
Inventors: Lars G. Liljeryd (Stocksund), Per Ekstrand (Saltsjobaden), Fredrik Henn (Huddinge), Kristofer Kjoerling (Solna)
Application Number: 15/446,485