AU2024200330B2 - : Pitch Filter for Audio Signals and Method for Filtering an Audio Signal with a Pitch Filter - Google Patents
: Pitch Filter for Audio Signals and Method for Filtering an Audio Signal with a Pitch Filter Download PDFInfo
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Abstract
In some embodiments, a pitch filter for filtering a preliminary audio signal
generated from an audio bitstream is disclosed. The pitch filter has an
5 operating mode selected from one of either: (i) an active mode where the
preliminary audio signal is filtered using filtering information to obtain a filtered
audio signal, and (ii) an inactive mode where the pitch filter is disabled. The
preliminary audio signal is generated in an audio decoder having a coding
mode selected from one of at least two distinct coding modes, and the pitch
10 filter is capable of being selectively operated in either the active mode or the
inactive mode based on control information while the audio decoder is
operating in the coding mode. A method for filtering a preliminary audio signal
with a pitch filter substantially as described above is also disclosed.
Description
This application is a divisional of Australian application No. 2022241555, which is a divisional of Australian Patent No. 2021204569, which is a divisional of Australian Patent No. 2020201647, which is a divisional of Australian Patent No. 2017276206, which is a divisional of Australian Patent No. 2016204303, itself a divisional of Australian Patent No. 2015200065, itself a divisional application of Australian Patent no. 2011273680. The entire contents of all of these patents/applications is incorporated herein by reference. This application claims priority to U.S. Provisional Application No. 61/361,237 filed 2 July 2010 hereby incorporated by reference in entirety for all purposes.
Technical field The present invention generally relates to digital audio coding and more precisely to coding techniques for audio signals containing components of different characters.
Background The following references to and descriptions of prior proposals or products are not intended to be, and are not to be construed as, statements or admissions of common general knowledge in the art.
A widespread class of coding method for audio signals containing speech or singing includes code excitedlinear prediction (CELP) applied in time alternation with different coding methods, including frequency-domain coding methods especially adapted for music or methods of a general nature, to account for variations in character between successive time periods of the audio signal. For example, a simplified Moving Pictures Experts Group (MPEG) Unified Speech and Audio Coding (USAC; see standard ISO/IEC 23003-3) decoder is operable in at least three decoding modes, Ad vanced Audio Coding (AAC; see standard ISO/IEC 13818-7), algebraic CELP
(ACELP) and transform-coded excitation (TCX), as shown in the upper por tion of accompanying figure 2. The various embodiments of CELP are adapted to the properties of the human organs of speech and, possibly, to the human auditory sense. As used in this application, CELP will refer to all possible embodiments and variants, including but not limited to ACELP, wide- and narrow-band CELP, SB-CELP (sub-band CELP), low- and high-rate CELP, RCELP (relaxed CELP), LD CELP (low-delay CELP), CS-CELP (conjugate-structure CELP), CS-ACELP (conjugate-structure ACELP), PSI-CELP (pitch-synchronous innovation CELP) and VSELP (vector sum excited linear prediction). The principles of CELP are discussed by R. Schroeder and S. Atal in Proceedings of the IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP), vol. 10, pp. 937-940, 1985, and some of its applications are de scribed in references 25-29 cited in Chen and Gersho, IEEE Transactions on Speech and Audio Processing, vol. 3, no. 1, 1995. As further detailed in the former paper, a CELP decoder (or, analogously, a CELP speech synthesizer) may include a pitch predictor, which restores the periodic component of an encoded speech signal, and an pulse codebook, from which an innovation se quence is added. The pitch predictor may in turn include a long-delay predic tor for restoring the pitch and a short-delay predictor for restoring formants by spectral envelope shaping. In this context, the pitch is generally understood as the fundamental frequency of the tonal sound component produced by the vocal chords and further coloured by resonating portions of the vocal tract. This frequency together with its harmonics will dominate speech or singing. Generally speaking, CELP methods are best suited for processing solo or one-part singing, for which the pitch frequency is well-defined and relatively easy to determine. To improve the perceived quality of CELP-coded speech, it is common practice to combine it with post filtering (or pitch enhancement by another term). U.S. Patent No. 4 969 192 and section 11 of the paper by Chen and Gersho disclose desirable properties of such post filters, namely their ability to suppress noise components located between the harmonics of the de- tected voice pitch (long-term portion; see section IV). It is believed that an im portant portion of this noise stems from the spectral envelope shaping. The long-term portion of a simple post filter may be designed to have the following transfer function:
HE(Z)=+azZ 2
where T is an estimated pitch period in terms of number of samples and a is a gain of the post filter, as shown in figures 1 and 2. In a manner similar to a comb filter, such a filter attenuates frequencies 1/(2T), 3/(2T), 5/(2T), ...
, which are located midway between harmonics of the pitch frequency, and ad jacent frequencies. The attenuation depends on the value of the gain a. Slightly more sophisticated post filters apply this attenuation only to low fre quencies - hence the commonly used term bass post filter - where the noise is most perceptible. This can be expressed by cascading the transfer function HE described above and a low-pass filter HLP. Thus, the post-processed de coded SE provided by the post filter will be given, in the transform domain, by SE Wz=SWz- aS(Z)PLT (z)HLp(z) ,
where -T T PLT W 2 and S is the decoded signal which is supplied as input to the post filter. Fig ure 3 shows an embodiment of a post filter with these characteristics, which is further discussed in section 6.1.3 of the Technical Specification ETSI TS 126 290, version 6.3.0, release 6. As this figure suggests, the pitch infor mation is encoded as a parameter in the bit stream signal and is retrieved by a pitch tracking module communicatively connected to the long-term predic tion filter carrying out the operations expressed by PLT. The long-term portion described in the previous paragraph may be used alone. Alternatively, it is arranged in series with a noise-shaping filter that preserves components in frequency intervals corresponding to the for mants and attenuates noise in other spectral regions (short-term portion; see section 111), that is, in the 'spectral valleys' of the formant envelope. As an other possible variation, this filter aggregate is further supplemented by a gradual high-pass-type filter to reduce a perceived deterioration due to spec tral tilt of the short-term portion. Audio signals containing a mixture of components of different origins e.g., tonal, non-tonal, vocal, instrumental, non-musical - are not always repro duced by available digital coding technologies in a satisfactory manner. It has more precisely been noted that available technologies are deficient in han dling such non-homogeneous audio material, generally favouring one of the components to the detriment of the other. In particular, music containing sing ing accompanied by one or more instruments or choir parts which has been encoded by methods of the nature described above, will often be decoded with perceptible artefacts spoiling part of the listening experience.
Summary of the invention In order to mitigate at least some of the drawbacks outlined in the pre vious section, the present invention seeks to provide methods and devices adapted for audio encoding and decoding of signals containing a mixture of components of different origins. The invention seeks to provide such methods and devices that are suitable from the point of view of coding efficiency or (perceived) reproduction fidelity or both. The invention seeks to achieve at least one of these objects by provid ing an encoder system, a decoder system, an encoding method, a decoding method and computer program products for carrying out each of the methods, as defined in the independent claims. The dependent claims define embodi ments of the invention. The inventors have realized that some artefacts perceived in decoded audio signals of non-homogeneous origin derive from an inappropriate switch ing between several coding modes of which at least one includes post filtering at the decoder and at least one does not. More precisely, available post filters remove not only interharmonic noise (and, where applicable, noise in spectral valleys) but also signal components representing instrumental or vocal ac companiment and other material of a 'desirable' nature. The fact that the just noticeable difference in spectral valleys may be as large as 10 dB (as noted by Ghitza and Goldstein, IEEE Trans. Acoust., Speech, Signal Processing, vol. ASSP-4, pp. 697-708, 1986) may have been taken as a justification by many designers to filter these frequency bands severely. The quality degrada tion by the interharmonic (and spectral-valley) attenuation itself may however be less important than that of the switching occasions. When the post filter is switched on, the background of a singing voice sounds suddenly muffled, and when the filter is deactivated, the background instantly becomes more sono rous. If the switching takes place frequently, due to the nature of the audio signal or to the configuration of the coding device, there will be a switching ar tefact. As one example, a USAC decoder may be operable either in an ACELP mode combined with post filtering or in a TCX mode without post fil tering. The ACELP mode is used in episodes where a dominant vocal compo nent is present. Thus, the switching into the ACELP mode may be triggered by the onset of singing, such as at the beginning of a new musical phrase, at the beginning of a new verse, or simply after an episode where the accompa niment is deemed to drown the singing voice in the sense that the vocal com ponent is no longer prominent. Experiments have confirmed that an alterna tive solution, or rather circumvention of the problem, by which TCX coding is used throughout (and the ACELP mode is disabled) does not remedy the problem, as reverb-like artefacts appear. Accordingly, in a first and a second aspect, the invention provides an audio encoding method (and an audio encoding system with the correspond ing features) characterized by a decision being made as to whether the de vice which will decode the bit stream, which is output by the encoding method, should apply post filtering including attenuation of interharmonic noise. The outcome of the decision is encoded in the bit stream and is acces sible to the decoding device. By the invention, the decision whether to use the post filter is taken separately from the decision as to the most suitable coding mode. This makes it possible to maintain one post filtering status throughout a period of such length that the switching will not annoy the listener. Thus, the encoding method may prescribe that the post filter will be kept inactive even though it switches into a coding mode where the filter is conventionally active.
It is noted that the decision whether to apply post filtering is normally taken frame-wise. Thus, firstly, post filtering is not applied for less than one frame at a time. Secondly, the decision whether to disable post filtering is only valid for the duration of a current frame and may be either maintained or reas sessed for the subsequent frame. In a coding format enabling a main frame format and a reduced format, which is a fraction of the normal format, e.g., 1/8 of its length, it may not be necessary to take post-filtering decisions for in dividual reduced frames. Instead, a number of reduced frames summing up to a normal frame may be considered, and the parameters relevant for the filter ing decision may be obtained by computing the mean or median of the re duced frames comprised therein. In a third and a fourth aspect of the invention, there is provided an au dio decoding method (and an audio decoding system with corresponding fea tures) with a decoding step followed by a post-filtering step, which includes interharmonic noise attenuation, and being characterized in a step of disa bling the post filter in accordance with post filtering information encoded in the bit stream signal. A decoding method with these characteristics is well suited for coding of mixed-origin audio signals by virtue of its capability to deactivate the post filter in dependence of the post filtering information only, hence independently of factors such as the current coding mode. When applied to coding tech niques wherein post filter activity is conventionally associated with particular coding modes, the post-filtering disabling capability enables a new operative mode, namely the unfiltered application of a conventionally filtered decoding mode. According to another aspect, the present invention provides a method of decoding a bit stream signal as an audio time signal, including the steps of: decoding a bit stream signal as a preliminary audio time signal according to a coding mode selected from a plurality of coding modes, wherein the plurality of coding modes includes at least a first coding mode which includes a post-filtering step and at least a second coding mode which does not include the post-filtering step, wherein the post-filtering step applies a pitch-enhancement filter to the preliminary audio time signal, thereby obtaining an audio time signal, wherein the post-filtering step is selectively omitted responsive to post-filtering information encoded in the bit stream signal, the post-filtering information being indicative of an encoder-side decision of whether or not to omit the post-filtering step, whereby the post-filtering step is selectively omit ted in the first coding mode. In a further aspect, the invention also provides a computer program product for performing one of the above methods. Further still, the invention provides a post filter for attenuating interharmonic noise which is operable in either an active mode or a pass-through mode, as indicated by a post-filtering signal supplied to the post filter. The post filter may include a decision section for autonomously controlling the post filtering activity. According to another aspect, there is provided herein a decoder sys tem for decoding a bit stream signal as an audio time signal, the decoder sys tem including: a decoding section for decoding the bit stream signal as a preliminary audio time signal, wherein the decoding section comprises a code-excited lin ear prediction, CELP, decoding module and a transform-coded excitation, TCX, decoding module; and an interharmonic noise attenuation post filter adapted to receive the preliminary audio time signal, and to supply the audio time signal, wherein the post filter comprises a control section for selectively operating the post filter in one of the following modes: i) a filtering mode, wherein the post filter filters the preliminary audio time signal to obtain a filtered signal and supplies the filtered signal as the au dio time signal; and ii) a pass-through mode, wherein the post filter supplies the preliminary audio time signal as the audio time signal, wherein the decoder system selectively operates in one of the following modes: a) the TCX module is enabled and the post filter is operated in the pass-through mode; b) the CELP module is enabled and, in response to a post-filtering sig nal, the post filter is operated in the filtering mode; and c) the CELP module is enabled and, in response to the post-filtering signal, the post filter is operated in the pass-through mode. In yet another aspect, there is provided herein a method of decoding a bit stream signal as an audio time signal, comprising: decoding the bit stream signal as a preliminary audio time signal in one of a plurality of decoding modes, the plurality of decoding modes comprising code-excited linear prediction, CELP, and transform-coded excitation, TCX, decoding modes; and filtering the preliminary audio time signal with an interharmonic noise attenuation post-filter to obtain the audio time signal, wherein the post-filter comprises a control section for selectively operating the post-filter in one of the following modes: i) a filtering mode, wherein the post filter filters the preliminary audio time signal to obtain a filtered signal and supplies the filtered signal as the au dio time signal; and ii) a pass-through mode, wherein the post-filter supplies the preliminary audio time signal as the audio time signal, wherein decoding the bit stream signal as an audio time signal com prises selectively operating in one of the following modes: a) enabling the TCX decoding mode and operating the post-filter in the pass-through mode; b) enabling the CELP decoding mode and, in response to a post-filter ing signal, operating the post-filter in the filtering mode; and c) enabling the CELP decoding mode and, in response to the post-fil tering signal, operating the post-filter in the pass-through mode. In yet a further aspect, there is provided herein an encoder system for encoding an audio time signal as a bit stream signal, including an encoding section operable in several coding modes, for encoding an audio time signal as a bit stream signal, the encoder system comprising a decision section adapted to decide whether post filtering, which includes attenuation of interharmonic noise, is to be disabled at decoding of the bit stream signal separately from deciding on the coding mode and to encode this decision in the bit stream signal as post filtering information, the decision section being adapted to: detect a co-presence of a signal component with dominant fundamen tal frequency and a signal component located below the fundamental fre quency and, optionally, between its harmonics; and responsive to a positive determination, to take a decision to disable.
According to another aspect, there is provided herein a method of en coding an audio time signal as a bit stream signal, the method including the steps of: encoding an audio time signal as a bit stream signal in one of several coding modes, and deciding whether post filtering, which includes attenuation of interhar monic noise, is to be disabled at decoding of the bit stream signal separately from deciding on the coding mode and encoding this decision in the bit stream signal as post filtering information, wherein deciding whether post fil tering is to be disabled comprises: detecting a co-presence of a signal component with dominant funda mental frequency and a signal component located below the fundamental fre quency and, optionally, between its harmonics; and responsive to a positive determination, deciding to disable post-filter ing.
According to another aspect, there is provided herein a decoder system for decoding a bit stream signal as an audio time signal, the decoder system in cluding: a decoding section for decoding the bit stream signal as a preliminary audio time signal, wherein the decoding section comprises a code-excited lin ear prediction, CELP, decoding module and a transform-coded excitation, TCX, decoding module; and an interharmonic noise attenuation post filter adapted to receive the preliminary audio time signal, and to supply the audio time signal, wherein the post filter comprises a control section for selectively operating the post filter in one of the following modes: i) a filtering mode, wherein the post filter filters the preliminary audio time signal to obtain a filtered signal and supplies the filtered signal as the au dio time signal; and ii) a pass-through mode, wherein the post filter supplies the preliminary audio time signal as the audio time signal, wherein the interharmonic noise attenuation depends on a value of a variable gain, and the control section is operable to enable the pass-through mode by setting the value of the variable gain to zero.
In yet a further aspect, there is provided herein a method of decoding a bit stream signal as an audio time signal, comprising: decoding the bit stream signal as a preliminary audio time signal in one of a plurality of decoding modes, the plurality of decoding modes comprising code-excited linear prediction, CELP, and transform-coded excitation, TCX, decoding modes; and filtering the preliminary audio time signal with an interharmonic noise attenuation post-filter to obtain the audio time signal, wherein the post-filter comprises a control section for selectively operating the post-filter in one of the following modes: i) a filtering mode, wherein the post filter filters the preliminary audio time signal to obtain a filtered signal and supplies the filtered signal as the au dio time signal; and ii) a pass-through mode, wherein the post-filter supplies the preliminary audio time signal as the audio time signal, wherein the interharmonic noise attenuation depends on a value of a variable gain, and the control section is operable to enable the pass-through mode by setting the value of the variable gain to zero.
According to another aspect, there is provided herein a non-transitory com puter readable storage medium containing a program of instructions, which when executed by one or more processors, cause one or more devices to perform a method of decoding a bit stream signal as an audio time signal, the method comprising: decoding the bit stream signal as a preliminary audio time signal in one of a plurality of decoding modes, the plurality of decoding modes comprising code-excited linear prediction, CELP, and transform-coded excitation, TCX, decoding modes; and filtering the preliminary audio time signal with an interharmonic noise attenuation post-filter to obtain the audio time signal, wherein the post-filter comprises a control section for selectively operating the post-filter in one of the following modes: i) a filtering mode, wherein the post filter filters the preliminary audio time signal to obtain a filtered signal and supplies the filtered signal as the au dio time signal; and ii) a pass-through mode, wherein the post-filter supplies the preliminary audio time signal as the audio time signal, wherein the interharmonic noise attenuation depends on a value of a variable gain, and the control section is operable to enable the pass-through mode by setting the value of the variable gain to zero.
According to another aspect, there is provided herein a decoder sys tem for decoding a bit stream signal as an audio time signal, the decoder sys tem including: a decoding section for decoding the bit stream signal as a preliminary audio time signal, wherein the decoding section comprises a code-excited lin ear prediction, CELP, decoding module and a transform-coded excitation, TCX, decoding module; and an interharmonic noise attenuation post filter adapted to receive the preliminary audio time signal, and to supply the audio time signal, wherein the post filter comprises a control section for selectively operating the post filter in one of the following modes: i) a filtering mode, wherein the post filter filters the preliminary audio time signal to obtain a filtered signal and supplies the filtered signal as the au dio time signal; and ii) a pass-through mode, wherein the post filter supplies the preliminary audio time signal as the audio time signal, wherein the interharmonic noise attenuation depends on a value of a variable gain and on pitch information included in the bit stream signal.
According to yet another aspect, there is provided herein a method of decod ing a bit stream signal as an audio time signal, comprising: decoding the bit stream signal as a preliminary audio time signal in one of a plurality of decoding modes, the plurality of decoding modes comprising code-excited linear prediction, CELP, and transform-coded excitation, TCX, decoding modes; and filtering the preliminary audio time signal with an interharmonic noise attenuation post-filter to obtain the audio time signal, wherein the post-filter comprises a control section for selectively operating the post-filter in one of the following modes: i) a filtering mode, wherein the post filter filters the preliminary audio time signal to obtain a filtered signal and supplies the filtered signal as the au dio time signal; and ii) a pass-through mode, wherein the post-filter supplies the preliminary audio time signal as the audio time signal, wherein the interharmonic noise attenuation depends on a value of a variable gain and on pitch information included in the bit stream signal.
According to yet another aspect, there is provided herein a non-transitory computer readable storage medium containing a program of instructions, which when executed by one or more processors, cause one or more devices to perform a method of decoding a bit stream signal as an audio time signal, the method comprising: decoding the bit stream signal as a preliminary audio time signal in one of a plurality of decoding modes, the plurality of decoding modes comprising code-excited linear prediction, CELP, and transform-coded excitation, TCX, decoding modes; and filtering the preliminary audio time signal with an interharmonic noise attenuation post-filter to obtain the audio time signal, wherein the post-filter comprises a control section for selectively operating the post-filter in one of the following modes: i) a filtering mode, wherein the post filter filters the preliminary audio time signal to obtain a filtered signal and supplies the filtered signal as the au dio time signal; and ii) a pass-through mode, wherein the post-filter supplies the preliminary audio time signal as the audio time signal, wherein the interharmonic noise attenuation depends on a value of a variable gain and on pitch information included in the bit stream signal.
As the skilled person will appreciate, an encoder adapted to cooperate with a decoder is equipped with functionally equivalent modules, so as to ena ble faithful reproduction of the encoded signal. Such equivalent modules may be identical or similar modules or modules having identical or similar transfer characteristics. In particular, the modules in the encoder and decoder, re spectively, may be similar or dissimilar processing units executing respective computer programs that perform equivalent sets of mathematical operations. In one embodiment, encoding the present method includes decision making as to whether a post filter which further includes attenuation of spec tral valleys (with respect to the formant envelope, see above). This corre sponds to the short-term portion of the post filter. It is then advantageous to adapt the criterion on which the decision is based to the nature of the post fil ter. One embodiment is directed to a encoder particularly adapted for speech coding. As some of the problems motivating the invention have been observed when a mixture of vocal and other components is coded, the combi nation of speech coding and the independent decision-making regarding post filtering afforded by the invention is particularly advantageous. In particular, such a decoder may include a code-excited linear prediction encoding mod ule. In one embodiment, the encoder bases its decision on a detected sim ultaneous presence of a signal component with dominant fundamental fre quency (pitch) and another signal component located below the fundamental frequency. The detection may also be aimed at finding the co-occurrence of a component with dominant fundamental frequency and another component with energy between the harmonics of this fundamental frequency. This is a situation wherein artefacts of the type under consideration are frequently en countered. Thus, if such simultaneous presence is established, the encoder will decide that post filtering is not suitable, which will be indicated accordingly by post filtering information contained in the bit stream. One embodiment uses as its detection criterion the total signal power content in the audio time signal below a pitch frequency, possibly a pitch fre quency estimated by a long-term prediction in the encoder. If this is greater than a predetermined threshold, it is considered that there are other relevant components than the pitch component (including harmonics), which will cause the post filter to be disabled. In an encoder comprising a CELP module, use can be made of the fact that such a module estimates the pitch frequency of the audio time signal. Then, a further detection criterion is to check for energy content between or below the harmonics of this frequency, as described in more detail above. As a further development of the preceding embodiment including a CELP module, the decision may include a comparison between an estimated power of the audio signal when CELP-coded (i.e., encoded and decoded) and an estimated power of the audio signal when CELP-coded and post-filtered. If the power difference is larger than a threshold, which may indicate that a rele vant, non-noise component of the signal will be lost, and the encoder will de cide to disable the post filter. In an advantageous embodiment, the encoder comprises a CELP mod ule and a TCX module. As is known in the art, TCX coding is advantageous in respect of certain kinds of signals, notably non-vocal signals. It is not common practice to apply post-filtering to a TCX-coded signal. Thus, the encoder may select either TCX coding, CELP coding with post filtering or CELP coding without post filtering, thereby covering a considerable range of signal types. As one further development of the preceding embodiment, the decision between the three coding modes is taken on the basis of a rate-distortion cri terion, that is, applying an optimization procedure known per se in the art. In another further development of the preceding embodiment, the en coder further comprises an Advanced Audio Coding (AAC) coder, which is also known to be particularly suitable for certain types of signals. Preferably, the decision whether to apply AAC (frequency-domain) coding is made sepa rately from the decision as to which of the other (linear-prediction) modes to use. Thus, the encoder can be apprehended as being operable in two super modes, AAC or TCX/CELP, in the latter of which the encoder will select be tween TCX, post-filtered CELP or non-filtered CELP. This embodiment ena bles processing of an even wider range of audio signal types. In one embodiment, the encoder can decide that a post filtering at de coding is to be applied gradually, that is, with gradually increasing gain. Like wise, it may decide that post filtering is to be removed gradually. Such grad ual application and removal makes switching between regimes with and with out post filtering less perceptible. As one example, a singing episode, for which post-filtered CELP coding is found to be suitable, may be preceded by an instrumental episode, wherein TCX coding is optimal; a decoder according to the invention may then apply post filtering gradually at or near the begin ning of the singing episode, so that the benefits of post filtering are preserved even though annoying switching artefacts are avoided. In one embodiment, the decision as to whether post filtering is to be applied is based on an approximate difference signal, which approximates that signal component which is to be removed from a future decoded signal by the post filter. As one option, the approximate difference signal is com puted as the difference between the audio time signal and the audio time sig nal when subjected to (simulated) post filtering. As another option, an encod ing section extracts an intermediate decoded signal, whereby the approxi mate difference signal can be computed as the difference between the audio time signal and the intermediate decoded signal when subjected to post filter ing. The intermediate decoded signal may be stored in a long-term prediction buffer of the encoder. It may further represent the excitation of the signal, im plying that further synthesis filtering (vocal tract, resonances) would need to be applied to obtain the final decoded signal. The point in using an intermedi ate decoded signal is that it captures some of the particularities, notably weaknesses, of the coding method, thereby allowing a more realistic estima tion of the effect of the post filter. As a third option, a decoding section ex tracts an intermediate decoded signal, whereby the approximate difference signal can be computed as the difference between the intermediate decoded signal and the intermediate decoded signal when subjected to post filtering. This procedure probably gives a less reliable estimation than the two first op tions, but can on the other hand be carried out by the decoder in a standalone fashion. The approximate difference signal thus obtained is then assessed with respect to one of the following criteria, which when settled in the affirmative will lead to a decision to disable the post filter: a) whether the power of the approximate difference signal exceeds a predetermined threshold, indicating that a significant part of the signal would be removed by the post filter; b) whether the character of the approximate difference signal is rather tonal than noise-like; c) whether a difference between magnitude frequency spectra of the approximate difference signal and of the audio time signal is unevenly distrib uted with respect to frequency, suggesting that it is not noise but rather a sig nal that would make sense to a human listener; d) whether a magnitude frequency spectrum of the approximate differ ence signal is localized to frequency intervals within a predetermined rele vance envelope, based on what can usually be expected from a signal of the type to be processed; and e) whether a magnitude frequency spectrum of the approximate differ ence signal is localized to frequency intervals within a relevance envelope ob- tained by thresholding a magnitude frequency spectrum of the audio time sig nal by a magnitude of the largest signal component therein downscaled by a predetermined scale factor. When evaluating criterion e), it is advantageous to apply peak tracking in the magnitude spectrum, that is, to distinguish portions having peak-like shapes normally associated with tonal components rather than noise. Components identified by peak tracking, which may take place by some algorithm known per se in the art, may be further sorted by applying a threshold to the peak height, whereby the remaining components are tonal material of a certain magnitude. Such components usually represent relevant signal content rather than noise, which motivates a decision to disable the post filter. In one embodiment of the invention as a decoder, the decision to disa ble the post filter is executed by a switch controllable by the control section and capable of bypassing the post filter in the circuit. In another embodiment, the post filter has variable gain controllable by the control section, or a gain controller therein, wherein the decision to disable is carried out by setting the post filter gain (see previous section) to zero or by setting its absolute value below a predetermined threshold. In one embodiment, decoding according to the present invention in cludes extracting post filtering information from the bit stream signal which is being decoded. More precisely, the post filtering information may be encoded in a data field comprising at least one bit in a format suitable for transmission. Advantageously, the data field is an existing field defined by an applicable standard but not in use, so that the post filtering information does not increase the payload to be transmitted. In other embodiments, an audio decoder for decoding an audio bitstream is disclosed. The decoder includes a first decoding module adapted to operate in a first coding mode and a second decoding module adapted to operate in a second coding mode, the second coding mode being different from the first coding mode. The decoder further includes a pitch filter in either the first coding mode or the second coding mode, the pitch filter adapted to filter a preliminary audio signal generated by the first decoding module or the second decoding module to obtain a filtered signal. The pitch filter is selectively enabled or disabled based on a value of a first parameter encoded in the audio bitstream, the first parameter being distinct from a second parameter encoded in the audio bitstream, the second parameter specifying a current coding mode of the audio decoder. In some embodiments, a pitch filter for filtering a preliminary audio signal generated from an audio bitstream is disclosed. The pitch filter has an operating mode selected from one of either: (i) an active mode where the preliminary audio signal is filtered using filtering information to obtain a filtered audio signal, and (ii) an inactive mode where the pitch filter is disabled. The preliminary audio signal is generated in an audio decoder having a coding mode selected from one of at least two distinct coding modes, and the pitch filter is capable of being selectively operated in either the active mode or the inactive mode based on control information while the audio decoder is operating in the coding mode. The control information may be included in the audio bitstream and may be independent of the coding mode. The filtering information may include pitch information and a gain, wherein the gain or pitch information is included in the audio bitstream. The pitch filter may be adapted to smooth the gain over time during a transition of the pitch filter. The coding mode may be signalled in the audio bitstream as a coding mode parameter. The control information may be a parameter one bit in length, and a first value of the parameter indicates that the pitch filter should be operated in the active mode and a second value of the parameter indicates that the pitch filter should be operated in the inactive mode. The audio bitstream may be segmented into frames of audio content and the control information includes a frame type parameter with one or more first values of the frame type parameter indicating that the pitch filter should be operated in the active mode and a second value of the parameter indicat ing that the pitch filter should be operated in the inactive mode. The frame type parameter may indicate whether a respective frame contains voiced con tent or whether the respective frame contains unvoiced content.
The pitch filter may be a post-filter or a pitch enhancement filter. The post-filter and the pitch enhancement filter may be adapted to attenuate sig nal components between harmonics or attenuate spectral valleys. The post filter and the pitch enhancement filter may be adapted to restore a periodic component of the preliminary audio signal. The first coding mode may include frequency-domain coding or trans form coding and the second coding mode may include linear prediction cod ing. The preliminary audio signal may be an excitation signal, the first cod ing mode may include frequency-domain coding or transform coding, and the second coding mode may include linear prediction. The pitch filter may be implemented with one or more comb filters. The pitch filter may be implemented with a long-term filter and a short term filter. The long-term filter may be a long-term prediction synthesis filter and the short-term filter may be a linear prediction coding synthesis filter, wherein the short-term filter processes the preliminary audio signal after the long-term filter. The pitch filter may have low frequency characteristics. In yet other embodiments, a method for filtering a preliminary audio signal with a pitch filter is disclosed. The pitch filter has an operating mode selected from one of either: an active mode where the preliminary audio signal is filtered using filtering information; or an inactive mode where the preliminary audio signal is not filtered. The method comprises: obtaining the preliminary audio signal, the preliminary audio signal generated from an audio bitstream in a coding mode selected from either a first coding mode or a second coding mode; obtaining control information; and selectively operating the pitch filter in either the active mode or the inactive mode while operating in the coding mode based on the control information. The operating mode of the pitch filter may be determined by the control information, the control information included in the audio bitstream and inde pendent of the coding mode. The pitch filter may have low frequency characteristics.
It is noted that the methods and apparatus disclosed in this section may be applied, after appropriate modifications within the skilled person's abilities including routine experimentation, to coding of signals having several components, possibly corresponding to different channels, such as stereo channels. Throughout the present application, pitch enhancement and post fil tering are used as synonyms. It is further noted that AAC is discussed as a representative example of frequency-domain coding methods. Indeed, apply ing the invention to a decoder or encoder operable in a frequency-domain coding mode other than AAC will only require small modifications, if any, within the skilled person's abilities. Similarly, TCX is mentioned as an exam ple of weighted linear prediction transform coding and of transform coding in general. Features from two or more embodiments described hereinabove can be combined, unless they are clearly complementary, in further embodiments. The fact that two features are recited in different claims does not preclude that they can be combined to advantage. Likewise, further embodiments can also be provided by the omission of certain features that are not necessary or not essential for the desired purpose.
Brief description of the drawings Embodiments of the present invention will now be described with refer ence to the accompanying drawings. However, it is to be understood that the features illustrated in and described with reference to the drawings are not to be construed as limiting on the scope of the invention., In the drawings: figure 1 is a block diagram showing a conventional decoder with post filter; figure 2 is a schematic block diagram of a conventional decoder opera ble in AAC, ACELP and TCX mode and including a post filter permanently connected downstream of the ACELP module; figure 3 is a block diagram illustrating the structure of a post filter; figures 4 and 5 are block diagrams of two decoders according to the in vention; figures 6 and 7 are block diagrams illustrating differences between a conventional decoder (figure 6) and a decoder (figure 7) according to the in vention; figure 8 is a block diagram of an encoder according to the invention; figures 9 and 10 are a block diagrams illustrating differences between a conventional decoder (figure 9) and a decoder (figure 10) according to the invention; and figure 11 is a block diagram of an autonomous post filter which can be selectively activated and deactivated.
Detailed description of embodiments Figure 4 is a schematic drawing of a decoder system 400 according to an embodiment of the invention, having as its input a bit stream signal and as its output an audio signal. As in the conventional decoders shown in figure 1, a post filter 440 is arranged downstream of a decoding module 410 but can be switched into or out of the decoding path by operating a switch 442. The post filter is enabled in the switch position shown in the figure. It would be dis abled if the switch was set in the opposite position, whereby the signal from the decoding module 410 would instead be conducted over the bypass line 444. As an inventive contribution, the switch 442 is controllable by post filter ing information contained in the bit stream signal, so that post filtering may be applied and removed irrespectively of the current status of the decoding mod ule 410. Because a post filter 440 operates at some delay - for example, the post filter shown in figure 3 will introduce a delay amounting to at least the pitch period T - a compensation delay module 443 is arranged on the bypass line 444 to maintain the modules in a synchronized condition at switching. The delay module 443 delays the signal by the same period as the post filter 440 would, but does not otherwise process the signal. To minimize the change-over time, the compensation delay module 443 receives the same signal as the post filter 440 at all times. In an alternative embodiment where the post filter 440 is replaced by a zero-delay post filter (e.g., a causal filter, such as a filter with two taps, independent of future signal values), the com pensation delay module 443 can be omitted.
Figure 5 illustrates a further development according to the teachings of the invention of the triple-mode decoder system 500 of figure 2. An ACELP decoding module 511 is arranged in parallel with a TCX decoding module 512 and an AAC decoding module 513. In series with the ACELP decoding mod ule 511 is arranged a post filter 540 for attenuating noise, particularly noise lo cated between harmonics of a pitch frequency directly or indirectly derivable from the bit stream signal for which the decoder system 500 is adapted. The bit stream signal also encodes post filtering information governing the posi tions of an upper switch 541 operable to switch the post filter 540 out of the processing path and replace it with a compensation delay 543 like in figure 4. A lower switch 542 is used for switching between different decoding modes. With this structure, the position of the upper switch 541 is immaterial when one of the TCX or AAC modules 512, 513 is used; hence, the post filtering in formation does not necessary indicate this position except in the ACELP mode. Whatever decoding mode is currently used, the signal is supplied from the downstream connection point of the lower switch 542 to a spectral band replication (SBR) module 550, which outputs an audio signal. The skilled per son will realize that the drawing is of a conceptual nature, as is clear notably from the switches which are shown schematically as separate physical enti ties with movable contacting means. In a possible realistic implementation of the decoder system, the switches as well as the other modules will be embod ied by computer-readable instructions. Figures 6 and 7 are also block diagrams of two triple-mode decoder systems operable in an ACELP, TCX or frequency-domain decoding mode. With reference to the latter figure, which shows an embodiment of the inven tion, a bit stream signal is supplied to an input point 701, which is in turn per manently connected via respective branches to the three decoding modules 711, 712, 713. The input point 701 also has a connecting branch 702 (not present in the conventional decoding system of figure 6) to a pitch enhance ment module 740, which acts as a post filter of the general type described above. As is common practice in the art, a first transition windowing module 703 is arranged downstream of the ACELP and TCX modules 711, 712, to carry out transitions between the decoding modules. A second transition mod ule 704 is arranged downstream of the frequency-domain decoding module 713 and the first transition windowing module 703, to carry out transition be tween the two super-modes. Further a SBR module 750 is provided immedi ately upstream of the output point 705. Clearly, the bit stream signal is sup plied directly (or after demultiplexing, as appropriate) to all three decoding modules 711, 712, 713 and to the pitch enhancement module 740. Infor mation contained in the bit stream controls what decoding module is to be ac tive. By the invention however, the pitch enhancement module 740 performs an analogous self actuation, which responsive to post filtering information in the bit stream may act as a post filter or simply as a pass-through. This may for instance be realized through the provision of a control section (not shown) in the pitch enhancement module 740, by means of which the post filtering action can be turned on or off. The pitch enhancement module 740 is always in its pass-through mode when the decoder system operates in the fre quency-domain or TCX decoding mode, wherein strictly speaking no post fil tering information is necessary. It is understood that modules not forming part of the inventive contribution and whose presence is obvious to the skilled per son, e.g., a demultiplexer, have been omitted from figure 7 and other similar drawings to increase clarity. As a variation, the decoder system of figure 7 may be equipped with a control module (not shown) for deciding whether post filtering is to be applied using an analysis-by-synthesis approach. Such control module is communica tively connected to the pitch enhancement module 740 and to the ACELP module 711, from which it extracts an intermediate decoded signalsi_DEc(n) representing an intermediate stage in the decoding process, preferably one corresponding to the excitation of the signal. The detection module has the necessary information to simulate the action of the pitch enhancement mod ule 740, as defined by the transfer functions PLT(Z) and HLP(Z) (cf. Background section and figure 3), or equivalently their filter impulse responsespLT(Z) and hLP(n). As follows by the discussion in the Background section, the compo nent to be subtracted at post filtering can be estimated by an approximate dif ference signal sAD(n) which is proportional to [(Si_DEC * PLT) * hLP(n), where * denotes discrete convolution. This is an approximation of the true difference between the original audio signal and the post-filtered decoded signal, namely SORIG(n) - SE(n) = SORIG(n) - (SDEC(n) - a[sDEC * PLT * hLP](n)), where a is the post filter gain. By studying the total energy, low-band energy, tonality, actual magnitude spectrum or past magnitude spectra of this signal, as disclosed in the Summary section and the claims, the control section may find a basis for the decision whether to activate or deactivate the pitch en hancement module 740. Figure 8 shows an encoder system 800 according to an embodiment of the invention. The encoder system 800 is adapted to process digital audio signals, which are generally obtained by capturing a sound wave by a micro phone and transducing the wave into an analog electric signal. The electric signal is then sampled into a digital signal susceptible to be provided, in a suitable format, to the encoder system 800. The system generally consists of an encoding module 810, a decision module 820 and a multiplexer 830. By virtue of switches 814, 815 (symbolically represented), the encoding module 810 is operable in either a CELP, a TCX or an AAC mode, by selectively acti vating modules 811, 812, 813. The decision module 820 applies one or more predefined criteria to decide whether to disable post filtering during decoding of a bit stream signal produced by the encoder system 800 to encode an au dio signal. For this purpose, the decision module 820 may examine the audio signal directly or may receive data from the encoding module 810 via a con nection line 816. A signal indicative of the decision taken by the decision module 820 is provided, together with the encoded audio signal from the en coding module 810, to a multiplexer 830, which concatenates the signals into a bit stream constituting the output of the encoder system 800. Preferably, the decision module 820 bases its decision on an approxi mate difference signal computed from an intermediate decoded signal si_DEC, which can be subtracted from the encoding module 810. The intermediate de coded signal represents an intermediate stage in the decoding process, as discussed in preceding paragraphs, but may be extracted from a correspond ing stage of the encoding process. However, in the encoder system 800 the original audio signal SORIG isavailable so that, advantageously, the approxi mate difference signal is formed as: SORIG(n) - (Si_DEC(n) - a [(si_DEC * PLT) * hLP](n)). The approximation resides in the fact that the intermediate decoded signal is used in lieu of the final decoded signal. This enables an appraisal of the na ture of the component that a post filter would remove at decoding, and by ap plying one of the criteria discussed in the Summary section, the decision module 820 will be able to take a decision whether to disable post filtering. As a variation to this, the decision module 820 may use the original sig nal in place of an intermediate decoded signal, so that the approximate differ ence signal will be [(Si_DEC * PLT) * hLP](n). This is likely to be a less faithful ap proximation but on the other hand makes the presence of a connection line 816 between the decision module 820 and the encoding module 810 optional. In such other variations of this embodiment where the decision module 820 studies the audio signal directly, one or more of the following criteria may be applied: • Does the audio signal contain both a component with dominant funda mental frequency and a component located below the fundamental fre quency? (The fundamental frequency may be supplied as a by-product of the encoding module 810.) • Does the audio signal contain both a component with dominant funda mental frequency and a component located between the harmonics of the fundamental frequency? • Does the audio signal contain significant signal energy below the fun damental frequency? • Is post-filtered decoding (likely to be) preferable to unfiltered decoding with respect to rate-distortion optimality? In all the described variations of the encoder structure shown in fig ure 8 - that is, irrespectively of the basis of the detection criterion - the deci sion section 820 may be enabled to decide on a gradual onset or gradual re moval of post filtering, so as to achieve smooth transitions. The gradual onset and removal may be controlled by adjusting the post filter gain.
Figure 9 shows a conventional decoder operable in a frequency-decod ing mode and a CELP decoding mode depending on the bit stream signal supplied to the decoder. Post filtering is applied whenever the CELP decoding mode is selected. An improvement of this decoder is illustrated in figure 10, which shows an decoder 1000 according to an embodiment of the invention. This decoder is operable not only in a frequency-domain-based decoding mode, wherein the frequency-domain decoding module 1013 is active, and a filtered CELP decoding mode, wherein the CELP decoding module 1011 and the post filter 1040 are active, but also in an unfiltered CELP mode, in which the CELP module 1011 supplies its signal to a compensation delay module 1043 via a bypass line 1044. A switch 1042 controls what decoding mode is currently used responsive to post filtering information contained in the bit stream signal provided to the decoder 1000. In this decoder and that of fig ure 9, the last processing step is effected by an SBR module 1050, from which the final audio signal is output. Figure 11 shows a post filter 1100 suitable to be arranged downstream of a decoder 1199. The filter 1100 includes a post filtering module 1140, which is enabled or disabled by a control module (not shown), notably a bi nary or non-binary gain controller, in response to a post filtering signal re ceived from a decision module 1120 within the post filter 1100. The decision module performs one or more tests on the signal obtained from the decoder to arrive at a decision whether the post filtering module 1140 is to be active or inactive. The decision may be taken along the lines of the functionality of the decision module 820 in figure 8, which uses the original signal and/or an inter mediate decoded signal to predict the action of the post filter. The decision of the decision module 1120 may also be based on similar information as the decision modules uses in those embodiments where an intermediate decoded signal is formed. As one example, the decision module 1120 may estimate a pitch frequency (unless this is readily extractable from the bit stream signal) and compute the energy content in the signal below the pitch frequency and between its harmonics. If this energy content is significant, it probably repre sents a relevant signal component rather than noise, which motivates a deci sion to disable the post filtering module 1140.
A 6-person listening test has been carried out, during which music samples encoded and decoded according to the invention were compared with reference samples containing the same music coded while applying post filtering in the conventional fashion but maintaining all other parameters un changed. The results confirm a perceived quality improvement. Further embodiments of the present invention will become apparent to a person skilled in the art after reading the description above. Even though the present description and drawings disclose embodiments and examples, the invention is not restricted to these specific examples. Numerous modifica tions and variations can be made without departing from the scope of the pre sent invention, which is defined by the accompanying claims. The systems and methods disclosed hereinabove may be implemented as software, firmware, hardware or a combination thereof. Certain compo nents or all components may be implemented as software executed by a digi tal signal processor or microprocessor, or be implemented as hardware or as an application-specific integrated circuit. Such software may be distributed on computer readable media, which may comprise computer storage media (or non-transitory media) and communication media (or transitory media). As is well known to a person skilled in the art, computer storage media includes both volatile and nonvolatile, removable and non-removable media imple mented in any method or technology for storage of information such as com puter readable instructions, data structures, program modules or other data. Computer storage media includes, but is not limited to, RAM, ROM, EEPROM, flash memory or other memory technology, CD-ROM, digital ver satile disks (DVD) or other optical disk storage, magnetic cassettes, magnetic tape, magnetic disk storage or other magnetic storage devices, or any other medium which can be used to store the desired information and which can be accessed by a computer. Further, it is well known to the skilled person that communication media typically embodies computer readable instructions, data structures, program modules or other data in a modulated data signal such as a carrier wave or other transport mechanism and includes any infor mation delivery media.
Throughout the specification and claims the word "comprise" and its derivatives are intended to have an inclusive rather than exclusive meaning unless the contrary is expressly stated or the context requires otherwise. That is, the word "comprise" and its derivatives will be taken to indicate the inclusion of not only the listed components, steps or features that it directly references, but also other components, steps or features not specifically listed, unless the contrary is expressly stated or the context requires otherwise.
In the present specification, terms such as "component", "apparatus", "means" and "device" may refer to singular or plural items and are terms intended to refer to a set of properties, functions or characteristics performed by one or more items having one or more parts. It is envisaged that where a "component", "apparatus", "means" or "device" or similar term is described as being a unitary object, then a functionally equivalent object having multiple components is considered to fall within the scope of the term, and similarly, where a "component", "apparatus", "assembly", "means" or "device" is described as having multiple items, a functionally equivalent but unitary object is also considered to fall within the scope of the term, unless the contrary is expressly stated or the context requires otherwise.
List of embodiments 1. A decoder system (400; 500; 700; 1000) for decoding a bit stream sig nal as an audio time signal, including: a decoding section (410; 511, 512, 513; 711, 712, 713; 1011, 1013) for decoding a bit stream signal as a preliminary audio time signal; and an interharmonic noise attenuation post filter (440; 540; 740; 1040) for filtering the preliminary audio time signal to obtain an audio time signal, characterized by a control section adapted to disable the post filter re sponsive to post-filtering information encoded in the bit stream signal, wherein the preliminary audio time signal is output as the audio time signal.
2. The decoder system of embodiment 1, wherein the post filter is further adapted to attenuate noise located in spectral valleys.
3. The decoder system of embodiment 1, wherein the control section in cludes a switch (442; 541; 1042) for selectively excluding the post filter from the signal processing path of the decoder system, whereby the post filter is disabled.
4. The decoder system of embodiment 1, wherein the post filter has varia ble gain determining the interharmonic attenuation and the control section in cludes a gain controller operable to set the absolute value of the gain below a predetermined threshold, whereby the post filter is disabled.
5. The decoder system of embodiment 1, said decoding section including a speech decoding module.
6. The decoder system of embodiment 1, said decoding section including a code-excited linear prediction, CELP, decoding module (511; 711; 1011).
7. The decoder system of embodiment 5, wherein a pitch frequency esti mated by a long-term prediction section in the encoder is encoded in the bit stream signal.
8. The decoder system of embodiment 7, wherein the post filter is adapted to attenuate spectral components located between harmonics of the pitch frequency.
9. The decoder system of embodiment 1, wherein the bit stream signal contains a representation of a pitch frequency and the post filter is adapted to attenuate spectral components located between harmonics of the pitch fre quency.
10. The decoder system of embodiment 8 or 9, wherein the post filter is adapted to attenuate only such spectral components which are located below a predetermined cut-off frequency.
11. The decoder system of embodiment 6, the decoding section further comprising a transform-coded excitation, TCX, decoding module (512; 712) for decoding a bit stream signal as an au dio time signal, the control section being adapted operate the decoder system in at least the following modes: a) the TCX module is enabled and the post filter is disabled; b) the CELP module and the post filter are enabled; and c) the CELP module is enabled and the post filter is disabled, wherein the preliminary audio time signal and the audio time signal coincide.
12. The decoder system of embodiment 10, the decoding section further comprising an Advanced Audio Coding, AAC, decoding module (513; 713) for decoding a bit stream signal as an au dio time signal, the control section being adapted to operate the decoder also in the following mode: d) the AAC module is enabled and the post filter is disabled.
13. The decoder system of embodiment 1, wherein the bit stream signal is segmented into time frames and the control section is adapted to disable an entire time frame or a sequence of entire time frames.
14. The decoder system of embodiment 13, wherein the control section is further adapted to receive, for each time frame in a Moving Pictures Experts Group, MPEG, bit stream, a data field associated with this time frame and is operable, responsive to the value of the data field, to disable the post filter.
15. The decoder system of embodiment 4, wherein the control section is adapted to decrease and/or increase the gain of the post filter gradually.
16. A decoder system (400; 500; 700; 1000) comprising: a decoding section (410; 511, 512, 513; 711, 712, 713; 1011, 1013) for decoding a bit stream signal as a preliminary audio time signal; and an interharmonic noise attenuation post filter (440; 540; 740; 1040) for filtering the preliminary audio time signal to obtain an audio time signal, characterized in that the decoding section is adapted to generate an intermediate decoded signal representing excitation and to provide this to the control section; and the control section is adapted to compute an approximate difference signal, which approximates the signal component which is to be removed from the decoded signal by the post filter, as a difference between the inter mediate decoded signal and the intermediate decoded signal when subjected to post filtering and to assess at least one of the following criteria: a) whether the power of the approximate difference signal ex ceeds a predetermined threshold; b) whether the character of the approximate difference signal is tonal; c) whether a difference between magnitude frequency spectra of the approximate difference signal and of the audio time signal is une venly distributed with respect to frequency; d) whether a magnitude frequency spectrum of the approximate difference signal is localized to frequency intervals within a predeter mined relevance envelope; and e) whether a magnitude frequency spectrum of the approximate difference signal is localized to frequency intervals within a relevance envelope obtained by thresholding a magnitude frequency spectrum of the audio time signal by a magnitude of the largest signal component therein downscaled by a predetermined scale factor; and, responsive to a positive determination, to disable the post filter, whereby the preliminary audio time signal is output as the audio time signal.
17. An interharmonic noise attenuation post filter (440; 550; 740; 1040; 1140) adapted to receive an input signal, which comprises a preliminary audio signal, and to supply an output audio signal, characterized by a control section for selectively, in accordance with the value of a post-filtering signal, operating the post filter in one of the follow ing modes: i) a filtering mode, wherein it filters the preliminary audio signal to ob tain a filtered signal and supplies this as output audio signal; and ii) a pass-through mode, wherein it supplies the preliminary audio sig nal as output audio signal.
18. The post filter of embodiment 17, wherein the post-filtering signal is in cluded in the input signal.
19. The post filter of embodiment 17, further comprising a decision module (1120) adapted to estimate a pitch frequency of the preliminary audio signal and to assess at least one of the following criteria: a) whether the power of spectral components below the pitch frequency exceed a predetermined threshold; b) whether spectral components below the pitch frequency are tonal; c) whether the power of spectral components between harmon ics of the pitch frequency exceed a predetermined threshold; and d) whether spectral components between harmonics of the pitch frequency are tonal; and, responsive to a positive determination, to take a decision to gen erate a negative post-filtering signal disabling the post filter.
20. A method of decoding a bit stream signal as an audio time signal, in cluding the steps of: decoding a bit stream signal as a preliminary audio time signal; and post-filtering the preliminary audio time signal by attenuating interhar monic noise, thereby obtaining an audio time signal, characterized in that the post-filtering step is selectively omitted re sponsive to post-filtering information encoded in the bit stream signal.
21. The method of embodiment 20, wherein the step of post-filtering further includes attenuating noise located in spectral valleys.
22. The method of embodiment 20, wherein the decoding step includes ap plying a coding method adapted for speech coding.
23. The method of embodiment 20, wherein the decoding step includes ap plying code-excited linear prediction, CELP, decoding.
24. The method of embodiment 22 or 23, wherein the post-filtering step in cludes attenuating spectral components located between harmonics of the pitch frequency, the pitch frequency being extracted from the bit stream signal or estimated in the decoding step.
25. The method of embodiment 20, wherein the post-filtering step includes attenuating only such spectral components which are located below a prede termined cut-off frequency.
26. The method of embodiment 23, wherein the steps of decoding and post-filtering selectively perform one of the following: a) TCX decoding; b) CELP decoding with post filtering; and c) CELP decoding without post filtering.
27. The method of embodiment 26, wherein the steps of decoding and post-filtering selectively perform one of modes a), b), c) and d) Advanced Audio Coding, AAC, decoding.
28. The method of embodiment 20, wherein the bit stream signal is seg mented into time frames and the post-filtering step is omitted for an entire time frame or a sequence of entire time frames.
29. The method of embodiment 28, wherein: the bit stream signal is a Moving Pictures Experts Group, MPEG, bit stream and includes, for each time frame, an associated data field; and the post-filtering step is omitted in a time frame responsive to the value of the associated data field.
30. The method of embodiment 20, wherein said omission of the post-filter ing includes one of the following: full omission of attenuation, partial omission of attenuation, gradually increasing attenuation, and gradually decreasing attenuation.
31. A method of decoding a bit stream signal as an audio time signal, in cluding the steps of: decoding a bit stream signal as a preliminary audio time signal; and post-filtering the preliminary audio time signal by attenuating interhar monic noise, thereby obtaining an audio time signal, characterized in that the step of decoding includes: extracting an intermediate decoded signal representing excitation; computing an approximate difference signal, which approximates the signal component which is to be removed from the decoded signal by the post filter, as a difference between the intermediate decoded signal and the inter mediate decoded signal when subjected to post filtering; assessing at least one of the following criteria: a) whether the power of the approximate difference signal ex ceeds a predetermined threshold; b) whether the character of the approximate difference signal is tonal; c) whether a difference between magnitude frequency spectra of the approximate difference signal and of the audio time signal is une venly distributed with respect to frequency; d) whether a magnitude frequency spectrum of the approximate difference signal is localized to frequency intervals within a predeter mined relevance envelope; e) whether a magnitude frequency spectrum of the approximate difference signal is localized to frequency intervals within a relevance envelope obtained by thresholding a magnitude frequency spectrum of the audio time signal by a magnitude of the largest signal component therein downscaled by a predetermined scale factor; and, responsive to a positive determination, to disable the post filter, whereby the preliminary audio signal is output as the audio time signal.
32. An encoder system (800) for encoding an audio time signal as a bit stream signal, including an encoding section (810) for encoding an audio time signal as a bit stream signal, characterized by a decision section (820) adapted to decide whether post filtering, which includes attenuation of interharmonic noise, is to be disa bled at decoding of the bit stream signal and to encode this decision in the bit stream signal as post filtering information.
33. The encoder system of embodiment 32, the decision section being adapted to decide whether to disable post filtering which further includes at tenuation of noise located in spectral valleys.
34. The encoder system of embodiment 32, the encoding section including a speech coding module.
35. The encoder system of embodiment 32, the encoding section including a code-excited linear prediction, CELP, encoding module.
36. The encoder system of embodiment 32, the decision section being adapted to: detect a co-presence of a signal component with dominant fundamen tal frequency and a signal component located below the fundamental fre quency and, optionally, between its harmonics; and responsive thereto, to take a decision to disable.
37. The encoder system of embodiment 35, the CELP encoding module being adapted to estimate a pitch fre quency in the audio time signal; and the decision section being adapted to detect spectral components lo cated below the estimated pitch frequency and, responsive thereto, to take a decision to disable.
38. The encoder system of embodiment 35, the decision section being adapted to compute a difference between a predicted power of the audio time signal when CELP-coded and a predicted power of the audio time signal when CELP-coded and post-filtered, and, responsive to this difference exceeding a predetermined threshold, to take a decision to disable.
39. The encoder system of embodiment 35, said encoding section further including a transform-coded excitation, TCX, encoding module, wherein the decision section is adapted to select one of the following coding modes: a) TCX coding; b) CELP coding with post filtering; and c) CELP coding without post filtering.
40. The encoder system of embodiment 39, further comprising a coding selector (814) adapted to select one of the following super-modes: i) Advanced Audio Coding, AAC coding, wherein the decision section is disabled; and ii) TCX/CELP coding, wherein the decision section is enabled to select one of coding modes a), b) and c).
41. The encoder system of embodiment 39, the decision section being adapted to decide which mode to use on the basis of a rate-distortion optimi zation.
42. The encoder system of embodiment 32, further adapted to segment the bit stream signal into time frames, the decision section being adapted to decide to disable the post filter in time segments consisting of entire frames.
43. The encoder system of embodiment 32, the decision section being adapted to decide to gradually decrease and/or increase the attenuation of the post filter.
44. The encoder system of embodiment 32, the decision section being adapted to: compute the power of the audio time signal below an estimated pitch frequency; and responsive to this power exceeding a predetermined threshold, to take a decision to disable.
45. The encoder system of embodiment 32, where the decision section is adapted to: derive, from the audio time signal, an approximate difference signal ap proximating the signal component which is to be removed from a future de coded signal by the post filter; assess at least one of the following criteria: a) whether the power of the approximate difference signal ex ceeds a predetermined threshold; b) whether the character of the approximate difference signal is tonal; c) whether a difference between magnitude frequency spectra of the approximate difference signal and of the audio time signal is une venly distributed with respect to frequency; d) whether a magnitude frequency spectrum of the approximate difference signal is localized to frequency intervals within a predeter mined relevance envelope; and e) whether a magnitude frequency spectrum of the approximate difference signal is localized to frequency intervals within a relevance envelope obtained by thresholding a magnitude frequency spectrum of the audio time signal by a magnitude of the largest signal component therein downscaled by a predetermined scale factor; and, responsive to a positive determination, to take a decision to disa ble the post filter.
46. The encoder system of embodiment 45, wherein the decision section is adapted to compute the approximate difference signal as a difference be tween the audio time signal and the audio time signal when subjected to post filtering.
47. The encoder system of embodiment 45, wherein: the encoding section is adapted to extract an intermediate decoded signal representing excitation and to provide this to the decision section; and the decision section is adapted to compute the approximate difference signal as a difference between the audio time signal and the intermediate de coded signal when subjected to post filtering.
48. A method of encoding an audio time signal as a bit stream signal, the method including the step of encoding an audio time signal as a bit stream signal, characterized by the further step of deciding whether post filtering, which includes attenuation of interharmonic noise, is to be disabled at decod ing of the bit stream and encoding this decision in the bit stream signal as post filtering information.
49. The method of embodiment 48, wherein the step of deciding relates to post filtering which further includes attenuation of noise located in spectral valleys.
50. The method of embodiment 48, wherein the step of encoding includes applying a coding method adapted for speech coding.
51. The method of embodiment 48, wherein the step of encoding includes applying code-excited linear prediction, CELP, coding.
52. The method of embodiment 48, further comprising the step of detecting a co-presence of a signal com ponent with dominant fundamental frequency and a signal component located below the fundamental frequency and, optionally, between its harmonics, wherein a decision to disable post filtering is made in the case of a positive detection outcome.
53. The method of embodiment 51, wherein: said step of CELP coding includes estimating a pitch frequency in the audio time signal; and the step of deciding includes detecting spectral components located below the estimated pitch frequency and a decision to disable post filtering is made in the case of a positive detection outcome.
54. The method of embodiment 51, further including the step of computing a difference between a pre dicted power of the audio time signal when CELP-coded and a predicted power of the audio time signal when CELP-coded and post-filtered, wherein a decision to disable post filtering is made if this difference ex ceeds a predetermined threshold.
55. The method of embodiment 51, wherein: the step of encoding includes selectively applying either CELP coding or transform-coded excitation, TCX, coding; and the step of deciding whether post filtering is to be disabled is performed only when CELP coding is applied.
56. The method of embodiment 55, wherein the step of deciding includes selecting, on the basis of a rate-distortion optimization, one of the following operating modes: a) TCX coding; b) CELP coding with post filtering; and c) CELP coding without post filtering.
57. The method of embodiment 55, wherein the step of deciding includes selecting, on the basis of a rate-distortion optimization, one of the following operating modes: a) TCX coding; b) CELP coding with post filtering; c) CELP coding without post filtering; and d) Advanced Audio Coding, AAC coding.
58. The method of embodiment 48, wherein: the step of encoding includes segmenting the audio time signal into time frames and to form a bit stream signal having corresponding time frames; and the step of deciding that post filtering is to be disabled is carried out once in every time frame.
59. The method of embodiment 48, wherein the outcome of the step of de ciding that post filtering is to be disabled is chosen from: no attenuation, full attenuation, partial attenuation, gradually increasing attenuation, and gradually decreasing attenuation.
60. The method of embodiment 48, wherein the step of deciding includes computing the power of the audio time signal below and estimated pitch fre quency and, responsive to this power exceeding a predetermined threshold, to disable the post filter.
61. The method of embodiment 48, wherein: the step of encoding includes deriving, from the audio time signal, an approximate difference signal approximating the signal component which is to be removed from a future decoded signal by the post filter; and the step of deciding includes assessing at least one of the following cri teria: a) whether the power of the approximate difference signal ex ceeds a predetermined threshold; b) whether the character of the approximate difference signal is tonal; c) whether a difference between magnitude frequency spectra of the approximate difference signal and of the audio time signal is une venly distributed with respect to frequency; d) whether a magnitude frequency spectrum of the approximate difference signal is localized to frequency intervals within a predeter mined relevance envelope; and e) whether a magnitude frequency spectrum of the approximate difference signal is localized to frequency intervals within a relevance envelope obtained by thresholding a magnitude frequency spectrum of the audio time signal by a magnitude of the largest signal component therein downscaled by a predetermined scale factor; and, responsive to at least a positive determination, to disable the post filter.
62. The method of embodiment 61, wherein the approximate difference signal is computed as a difference between the audio time signal and the au dio time signal when subjected to post filtering.
63. The method of embodiment 61, wherein: the step of encoding includes extracting an intermediate decoded sig nal representing excitation; and the step of deciding includes computing the approximate difference sig nal as a difference between the audio time signal and the intermediate de coded signal when subjected to post filtering.
64. A computer-program product including a data carrier storing instruc tions for performing the method of any one of embodiment 20 to 31 and 48 to 63.
Claims (18)
1. A decoder system for decoding a bit stream signal as an audio time signal, the decoder system including: a decoding section for decoding the bit stream signal as a preliminary audio time signal, wherein the decoding section comprises a code-excited linear prediction, CELP, decoding module and a transform-coded excitation, TCX, decoding module; and an interharmonic noise attenuation post filter adapted to receive the preliminary audio time signal, and to supply the audio time signal, wherein the post filter comprises a control section for selectively operating the post filter in one of the following modes: i) a filtering mode, wherein the post filter filters the preliminary audio time signal to obtain a filtered signal and supplies the filtered signal as the audio time signal; and ii) a pass-through mode, wherein the post filter supplies the preliminary audio time signal as the audio time signal, wherein the interharmonic noise attenuation depends on a value of a variable gain and on pitch information included in the bit stream signal.
2. The decoder system of claim 1, wherein the decoding section selectively operates in one of the following modes: a) the TCX module is enabled and the post filter is operated in the pass-through mode; b) the CELP module is enabled and, in response to a post-filtering signal, the post filter is operated in the filtering mode; and c) the CELP module is enabled and, in response to the post-filtering signal, the post filter is operated in the pass-through mode.
3. The decoder system of claim 2, the decoding section further comprising an Advanced Audio Coding, AAC, decoding module for decoding a bit stream signal as an audio time signal, the control section being adapted to operate the decoder also in the following mode: d) the AAC module is enabled and the post filter is disabled.
4. The decoder system of claim 1, wherein the post filter is adapted to attenuate only such spectral components which are located below a predetermined cut-off frequency.
5. The decoder system of claim 1, wherein the bit stream signal is a Moving Pictures Experts Group, MPEG, bit stream and is segmented into time frames and the control section is adapted to disable an entire time frame or a sequence of entire time frames; and the control section is further adapted to receive, for each time frame, a data field associated with this time frame and is operable, responsive to the value of the data field, to disable the post filter, whereby the preliminary audio time signal is output as the audio time signal.
6. The decoder system of claim 1, wherein the control section is operable to enable the pass-through mode by setting the value of the variable gain to zero.
7. A method of decoding a bit stream signal as an audio time signal, comprising: decoding the bit stream signal as a preliminary audio time signal in one of a plurality of decoding modes, the plurality of decoding modes comprising code-excited linear prediction, CELP, and transform-coded excitation, TCX, decoding modes; and filtering the preliminary audio time signal with an interharmonic noise attenuation post-filter to obtain the audio time signal, wherein the post-filter comprises a control section for selectively operating the post-filter in one of the following modes: i) a filtering mode, wherein the post filter filters the preliminary audio time signal to obtain a filtered signal and supplies the filtered signal as the audio time signal; and ii) a pass-through mode, wherein the post-filter supplies the preliminary audio time signal as the audio time signal, wherein the interharmonic noise attenuation depends on a value of a variable gain and on pitch information included in the bit stream signal.
8. The method of claim 7, wherein decoding the bit stream signal as an audio time signal comprises selectively operating in one of the following modes: a) enabling the TCX decoding mode and operating the post-filter in the pass through mode; b) enabling the CELP decoding mode and, in response to a post-filtering signal, operating the post-filter in the filtering mode; and c) enabling the CELP decoding mode and, in response to the post-filtering signal, operating the post-filter in the pass-through mode.
9. The method of claim 8, the decoding modes further comprising an Advanced Audio Coding, AAC, decoding mode for decoding a bit stream signal as an audio time signal, the control section being adapted to operate the decoder also in the following mode: d) the AAC decoding mode is enabled and the post filter is disabled.
10. The method of claim 7, wherein the post filter is adapted to attenuate only such spectral components which are located below a predetermined cut-off frequency.
11. The method of claim 7, wherein the bit stream signal is a Moving Pictures Experts Group, MPEG, bit stream and is segmented into time frames and the control section is adapted to disable an entire time frame or a sequence of entire time frames; and the control section is further adapted to receive, for each time frame, a data field associated with this time frame and is operable, responsive to the value of the data field, to disable the post filter, whereby the preliminary audio time signal is output as the audio time signal.
12. The method of claim 7, wherein the control section is operable to enable the pass through mode by setting the value of the variable gain to zero.
13. A non-transitory computer readable storage medium containing a program of instructions, which when executed by one or more processors, cause one or more devices to perform a method of decoding a bit stream signal as an audio time signal, the method comprising: decoding the bit stream signal as a preliminary audio time signal in one of a plurality of decoding modes, the plurality of decoding modes comprising code-excited linear prediction, CELP, and transform-coded excitation, TCX, decoding modes; and filtering the preliminary audio time signal with an interharmonic noise attenuation post-filter to obtain the audio time signal, wherein the post-filter comprises a control section for selectively operating the post-filter in one of the following modes: i) a filtering mode, wherein the post filter filters the preliminary audio time signal to obtain a filtered signal and supplies the filtered signal as the audio time signal; and ii) a pass-through mode, wherein the post-filter supplies the preliminary audio time signal as the audio time signal, wherein the interharmonic noise attenuation depends on a value of a variable gain and on pitch information included in the bit stream signal.
14. The medium of claim 13, wherein decoding the bit stream signal as an audio time signal comprises selectively operating in one of the following modes: a) enabling the TCX decoding mode and operating the post-filter in the pass through mode; b) enabling the CELP decoding mode and, in response to a post-filtering signal, operating the post-filter in the filtering mode; and c) enabling the CELP decoding mode and, in response to the post-filtering signal, operating the post-filter in the pass-through mode.
15. The medium of claim 14, the decoding modes further comprising an Advanced Audio Coding, AAC, decoding mode for decoding a bit stream signal as an audio time signal, the control section being adapted to operate the decoder also in the following mode: d) the AAC decoding mode is enabled and the post filter is disabled.
16. The medium of any claim 13, wherein the post filter is adapted to attenuate only such spectral components which are located below a predetermined cut-off frequency.
17. The medium of claim 13, wherein the bit stream signal is a Moving Pictures Experts Group, MPEG, bit stream and is segmented into time frames and the control section is adapted to disable an entire time frame or a sequence of entire time frames; and the control section is further adapted to receive, for each time frame, a data field associated with this time frame and is operable, responsive to the value of the data field, to disable the post filter, whereby the preliminary audio time signal is output as the audio time signal.
18. The medium of claim 13, wherein the control section is operable to enable the pass-through mode by setting the value of the variable gain to zero.
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