RU2789688C1 - Improved harmonic transformation based on a block of sub-bands - Google Patents

Abstract

FIELD: computer technology.

SUBSTANCE: invention relates to the field of computer technology for processing audio data. The effect is achieved by creating a signal of the analyzed subrange associated with the frequency band of the input audio signal; extracting a frame L of input discrete values from a series of complex-valued analyzed discrete values; applying a block input step to a set of complex-valued analyzed samples before extracting the next frame of L input samples; determining a frame of processed discrete values from a frame of input discrete values by determining for each processed discrete value of the frame: the phase of the processed discrete value by shifting the phase of the corresponding input discrete value; and the amplitude of the processed sample value based on the amplitude of the corresponding input sample value and the amplitude of the predetermined input sample value; and determining a synthesis subband signal by superimposing and adding the samples of the set of frames of the processed samples.

EFFECT: reducing the computational complexity of the harmonic transformation to improve the quality of the audio signal.

19 cl, 7 dwg

Description

TECHNICAL APPLICATION

This document relates to source audio coding systems that use a harmonic transformation technique for high frequency reconstruction (HFR), as well as to digital effects processors such as exciters, in which the generation of harmonic distortion adds brightness to the processed signal, and to time expanders, in which the signal duration increases while maintaining the spectral composition.

BACKGROUND OF THE INVENTION

отображается в синусоиду с частотой

, где

– целое число, определяющее порядок преобразования. В отличие от этого, HFR на основе модуляции с одной боковой полосой (SSB) отображает синусоиду с частотой

в синусоиду с частотой

, где

– фиксированный сдвиг частоты. Для данного сигнала с низкочастотной полосой пропускания в результате SSB-преобразования, как правило, появляется артефакт диссонирующего «звона». По причине указанных артефактов HFR на основе гармонического преобразования, в общем, более предпочтительно, чем HFR на основе SSB.WO 98/57436 created the concept of transform as a way of recreating a high frequency band from a lower frequency band of an audio signal. By using this concept for audio coding, significant bit rate savings can be achieved. In an HFR-based audio coding system, a low-bandwidth signal is passed to the core waveform encoder, and high frequencies are regenerated using a transform and at an extremely low bit rate of side information describing the target spectrum shape at the decoder side. For low bit rates, when the bandwidth of the underlying coded signal is narrow, recreating the high frequency band with pleasant characteristics becomes increasingly important. The harmonic transformation defined in WO 98/57436 performs well for complex musical material in a low crossover frequency situation. Document WO 98/57436 is incorporated herein by reference. The principle of harmonic transformation is that a sinusoid with a frequency

mapped to a sinusoid with a frequency

, Where

is an integer specifying the conversion order. In contrast, single-sideband (SSB) modulation-based HFR displays a sinusoid with a frequency

into a sinusoid with a frequency

, Where

– fixed frequency shift. For a given low-bandwidth signal, the SSB conversion typically results in a dissonant “ringing” artifact. Because of these artifacts, harmonic transform-based HFR is generally preferred over SSB-based HFR.

In order to achieve improved audio quality, high quality harmonic transform-based HFR methods typically use complex, high frequency resolution, high oversampling, complex modulated filter banks to achieve the desired audio quality. High frequency resolution is typically used to avoid unwanted intermodulation distortion that occurs when processing non-linear or processing different subband signals that can be viewed as the sum of a series of sinusoids. For sufficiently narrow subranges, i.e. with sufficiently high frequency resolution, high-quality harmonic transform-based HFR methods tend to have no more than one sinusoid in each subband. As a result, intermodulation distortion caused by non-linear processing can be avoided. On the other hand, a high degree of time oversampling can be useful in order to avoid aliasing distortion that can be caused by filter banks and non-linear processing. In addition, a certain degree of frequency oversampling may be necessary for short-term non-periodic signals to avoid pre-echo caused by non-linear processing of subband signals.

In addition, harmonic transform-based HFR methods generally use processing based on two groups of filter banks. The first part of HFR based on harmonic transformation typically uses a high frequency and time resolution analysis/synthesis filter bank and/or frequency oversampling to generate a high frequency signal component from a low frequency signal component. The second part of HFR based on harmonic transformation typically uses a filter bank with a relatively low frequency resolution, such as a QMF filter bank, which is used to apply additional spectral information or HFR information to the high frequency component, i.e. to perform the so-called. HFR processing, in order to generate a high-frequency component with the desired spectrum shape. The second part of the filter banks is also used to combine the low frequency signal component with the modified high frequency signal component to create a decoded audio signal.

As a result of using a sequence of two groups of filter banks and using high frequency and time resolution analysis/synthesis filter banks and/or frequency oversampling, the computational complexity of a harmonic transform-based HFR can be relatively high. Accordingly, there is a need to provide harmonic transform-based HFR methods with reduced computational complexity, while at the same time providing good audio signal quality for various types of audio signals (eg, short non-periodic and stationary audio signals).

BRIEF DESCRIPTION OF THE INVENTION

According to one of the features, the so-called. subband block-based harmonic transformation can be used to suppress combinational products caused by non-linear processing of subband signals, i.e. the combinational components in the subbands can be suppressed or reduced by performing block-based non-linear processing of the harmonic transducer subband signals. As a result, a harmonic transform can be applied that uses an analysis/synthesis filter bank with a relatively low frequency resolution and/or a relatively low degree of oversampling. For example, a QMF filter bank may be used.

Block-based non-linear processing of a sub-band block-based harmonic transformation system includes processing a time block of complex sub-band samples. The processing of a block of complex subband samples may include a well-known phase modification of the complex subband samples and a superposition of several modified samples to form a subband output sample. Such block-based processing has the net effect of suppressing or reducing combinational products that would otherwise occur with subband inputs comprising multiple sinusoids.

In view of the fact that analyzer/synthesizing filter banks with relatively low frequency resolution can be used for subband block-based harmonic transform, and because a lower degree of oversampling may be required, harmonic transform based on block-based subband processing, may have less high computational complexity compared to high quality harmonic converters, i.e. with harmonic converters having high frequency resolution and using discrete value processing. At the same time, it has been experimentally shown that for many types of audio signals, the quality of the audio signal that can be achieved using a block-based harmonic transformation is almost the same as using a sample-based harmonic transformation. However, it has been observed that the audio quality obtained for short duration non-periodic audio signals is generally reduced compared to the audio quality that can be achieved with high quality harmonic transducers, i.e. harmonic converters using high frequency resolution. It was found that the reduced quality for short-term non-periodic signals may be due to the blurring of time caused by block processing.

преобразования. Как правило, каждый порядок

преобразования гармонического преобразования на блочной основе требует отличающейся структуры блоков анализирующих и синтезирующих фильтров.In addition to the quality issues raised above, the complexity of the subband block-based harmonic transform still remains higher than the complexity of the simplest SSB-based HFR methods. This is true because synthesizing the required bandwidth usually requires multiple signals with different orders.

transformations. Generally, each order

block-based harmonic transformation transformations require a different structure of analysis and synthesis filter banks.

нескольких порядков может использоваться один блок анализирующих/синтезирующих фильтров, например, блок QMF-фильтров. Кроме того, для гармонического преобразования (т.е. для первой части HFR на основе гармонического преобразования) и для HFR-обработки (т.е. для второй части HFR на основе гармонического преобразования) может применяться одна и та же пара блоков анализирующих/синтезирующих фильтров, и, таким образом, полная HFR на основе гармонического преобразования может возлагаться на один единственный блок анализирующих/синтезирующих фильтров. Иными словами, на стороне ввода для генерирования ряда сигналов анализируемых поддиапазонов, которые затем поддаются обработке посредством гармонического преобразования и HFR-обработке, может использоваться лишь один единственный блок анализирующих фильтров. В конечном итоге, для генерирования декодированного сигнала на стороне вывода может использоваться лишь один единственный блок синтезирующих фильтров.In view of the above analysis, there is a particular need to improve the quality of the subband block-based harmonic transform for short-term non-periodic and voice signals while maintaining the quality of stationary signals. As will be described below, quality improvement can be achieved by fixed, or signal adaptive, modification of the non-linear block processing. In addition, there is a need to further reduce the complexity of the harmonic transform based on the block of subbands. As will be described below, a reduction in computational complexity can be achieved by efficiently implementing a multi-order subband block-based transform within a single pair of analysis and synthesis filter banks. As a result, for the harmonic transformation

multiple orders, a single analysis/synthesizing filter bank, such as a QMF filter bank, may be used. In addition, for harmonic transformation (i.e., for the first part of HFR based on harmonic transformation) and for HFR processing (i.e., for the second part of HFR based on harmonic transformation), the same pair of analyzers/synthesizers can be used. filters, and thus a complete HFR based on harmonic transformation can be assigned to one single analysis/synthesis filter bank. In other words, on the input side, only one single analysis filter bank can be used to generate a number of analyzed subband signals, which are then amenable to harmonic transformation and HFR processing. Ultimately, only one single synthesis filter bank can be used to generate the decoded signal at the output side.

In one aspect, a system is described that is configured to generate a time-stretched and/or frequency-converted signal from an input signal. The system may include an analysis filter bank configured to generate an analysis subband signal from an input signal. The analyzed subband may be associated with one of the frequency bands of the input signal. The analyzed subband signal may include a number of complex-valued analyzed samples, each of which has a phase and amplitude. The analysis filter bank may be a quadrature mirror filter bank, a windowed discrete Fourier transform, or a wavelet transform. In particular, the analysis filter bank may be a 64-point quadrature mirror filter bank. Thus, the analysis filter bank may have a low frequency resolution.

анализа по времени, и/или блок анализирующих фильтров может иметь разнос

анализируемых частот так, чтобы полоса частот, связанная с сигналом анализируемого поддиапазона имела номинальную ширину

, и/или блок анализирующих фильтров мог содержать количество N анализируемых поддиапазонов, N>1, где n – индекс анализируемого поддиапазона,

. Следует отметить, что из-за наложения смежных частотных полос фактическая спектральная ширина анализируемого поддиапазона может быть больше

. Однако разнос частот между смежными анализируемыми поддиапазонами, как правило, имеет вид

разноса анализируемых частот.The analysis filter block can apply a step to the input signal

analysis over time, and/or the analysis filter bank may have a spacing

analyzed frequencies so that the bandwidth associated with the analyzed subband signal has a nominal width

, and/or the analysis filter block could contain the number N of analyzed subranges, N >1, where n is the index of the analyzed subrange,

. It should be noted that due to the overlap of adjacent frequency bands, the actual spectral width of the analyzed subband may be greater than

. However, the frequency spacing between adjacent analyzed subbands, as a rule, has the form

separation of the analyzed frequencies.

преобразования поддиапазона и коэффициента S растягивания поддиапазона. По меньшей мере, один из коэффициентов, Q или S, может быть больше единицы. Модуль обработки поддиапазонов может включать экстрактор блоков, сконфигурированный для извлечения кадра, состоящего из L входных дискретных значений, из ряда комплекснозначных анализируемых дискретных значений. Длина L кадра может быть больше единицы, однако, в некоторых вариантах осуществления изобретения длина L кадра может быть равна единице. В альтернативном варианте или в дополнение, экстрактор блоков может быть сконфигурирован для применения величины скачка блока из р дискретных значений к ряду анализируемых дискретных значений перед извлечением следующего кадра, состоящего из L входных дискретных значений. В результате многократного применения величины скачка блока к ряду анализируемых дискретных значений может генерироваться набор кадров, состоящих из входных дискретных значений.The system may include a subband processing module configured to determine a synthesis subband signal from an analyzed subband signal using a coefficient

a subband transform; and a subband stretch factor S. At least one of the coefficients, Q or S , may be greater than one. The subband processing module may include a block extractor configured to extract a frame of L input samples from a set of complex valued samples to be analyzed. The frame length L may be greater than one, however, in some embodiments of the invention, the frame length L may be equal to one. Alternatively, or in addition, the block extractor may be configured to apply a block jump value of p samples to the set of analyzed samples before extracting the next frame of L input samples. As a result of multiple application of the block jump value to a number of analyzed discrete values, a set of frames consisting of input discrete values can be generated.

It should be noted that the frame length L and/or the block hop amount p can be arbitrary numbers and are not necessarily integers. In this or other cases, the block extractor may be configured to interpolate two or more parsed samples to obtain a frame input sample consisting of L input samples. For example, if the frame length and/or block hop amount are fractional numbers, the input sample of a frame consisting of the input samples can be obtained by interpolating two or more contiguous analyzed samples.

Alternatively, or in addition, the block extractor may be configured to downsample the number of parsed samples to obtain a frame input sample consisting of L input samples. In particular, the block extractor may be configured to downsample the number of analyzed samples with a subband transform factor Q. Thus, the block extractor can contribute to harmonic transformation and/or time stretching by performing a downsampling operation.

. В частности, величина сдвига фазы может задаваться предварительно определяемым входным дискретным значением, умноженным на

, плюс параметр

коррекции фазы. Параметр

коррекции фазы может определяться экспериментально для ряда входных сигналов, имеющих конкретные акустические свойства.The system, and in particular the subband processing module, may include a non-linear frame processing module configured to determine a frame of processed samples based on a frame of input samples. The determination may be repeated for a set of frames consisting of input samples, and thus a set of frames consisting of processed samples is generated. The determination can be performed by determining for each processed discrete value of the frame the phase of the processed discrete value - by shifting the phase of the corresponding input discrete value. In particular, the non-linear frame processor may be configured to determine the phase of the processed sample by shifting the phase of the corresponding input sample by a phase shift amount that is based on a predetermined input sample from a frame of input samples, a transform factor Q , and a factor S subrange stretching. The amount of phase shift can be based on a predefined input sample value multiplied by

. In particular, the amount of phase shift may be given by a predetermined input sample value multiplied by

, plus the parameter

phase correction. Parameter

phase correction can be determined experimentally for a number of input signals having specific acoustic properties.

In one of the preferred embodiments of the invention, the predetermined input sample is the same for each processed frame sample. In particular, the predetermined input sample may be the center sample of a frame of input samples.

, умноженная на амплитуду предварительно определяемого входного дискретного значения, возведенную в степень

. Как правило, параметром геометрического взвешивания амплитуды является

. Кроме того, параметр геометрического взвешивания амплитуды

может являться функцией коэффициента Q преобразования поддиапазона и коэффициента S растягивания поддиапазона. В частности, параметром геометрического взвешивания амплитуды может являться

, что в результате приводит к снижению вычислительной сложности.Alternatively, or in addition, the determination may be performed by determining, for each frame sample to be processed, the amplitude of the processed sample based on the amplitude of the corresponding input sample and the amplitude of the predetermined input sample. In particular, the non-linear frame processor may be configured to determine the amplitude of the processed sample as the average of the amplitude of the corresponding input sample and the amplitude of the predetermined input sample. The amplitude of the sample to be processed may be defined as the geometric mean of the amplitude of the corresponding input sample and the amplitude of the predetermined input sample. More specifically, the geometric mean may be defined as the amplitude of the corresponding input sample value raised to the power

multiplied by the amplitude of the predetermined input discrete value raised to the power

. Typically, the amplitude geometric weighting parameter is

. In addition, the amplitude geometric weighting parameter

may be a function of the subband transform factor Q and the subband stretch factor S. In particular, the amplitude geometric weighting parameter can be

, resulting in a reduction in computational complexity.

It should be noted that the predetermined input sample used to determine the amplitude of the processed sample may be different from the predetermined input sample used to determine the phase of the processed sample. However, in the preferred embodiment of the invention, both predetermined input discrete values are the same.

In general, a non-linear frame processing module may be used in the system to control the degree of harmonic transformation and/or time stretching. It can be shown that by determining the amplitude of the processed sample from the amplitude of the corresponding input sample and the amplitude of the predetermined input sample, the performance of the system with respect to short-term non-periodic and/or voice signals can be improved.

The system, and in particular the non-linear subband processing module, may include an overlay and add module configured to determine a synthesis subband signal by overlaying and adding samples from a set of frames of processed samples. The overlay and add block may apply the jump amount to successive frames of processed samples. This hop amount may be equal to the block hop amount p multiplied by the subband stretch factor S. Thus, the overlay and add module can be used in the system to control the amount of time stretching and/or harmonic transformation.

скачка, могут предусматривать набор дискретных значений, имеющих, в значительной мере, постоянную величину К.The system, and in particular the subband processing module, may include a windowing module located upstream of the overlay and add module. The windowing module may be configured to apply a windowing function to a frame consisting of processed samples. Thus, the window function can be applied to a set of frames consisting of processed discrete values before the overlay and add operation. The window function may have a length that corresponds to the frame length L. The window function may be a Gaussian window, a cosine window, a raised cosine window, a Hamming window, a Hann window, a rectangular window, a Bartlett window, and/or a Blackman window. Typically, a window function includes a set of window function samples, and superimposed and added window function samples from the set of window functions shifted by the amount

jump, may provide for a set of discrete values, having, to a large extent, a constant value K .

синтеза по времени, и/или блок синтезирующих фильтров имеет разнос

синтезируемых частот, и/или блок синтезирующих фильтров содержит количество М синтезируемых поддиапазонов,

, где m – индекс синтезируемого поддиапазона,

.The system may include a synthesis filter bank configured to generate a time-stretched and/or frequency-transformed signal from the synthesis subband signal. The synthesized subband may be associated with a frequency band of the time-stretched and/or frequency-converted signal. The synthesis filter bank may be the corresponding filter bank, or the inverse of the filter bank, or the transform, of the analysis filter bank. In particular, the synthesis filter bank may be a 64-point inverse quadrature mirror filter bank. In one embodiment of the invention, the synthesis filter bank applies a step to the synthesis subband signal

time synthesis, and/or the synthesis filter bank has a spacing

synthesized frequencies, and/or the block of synthesizing filters contains the number M of synthesized subbands,

, where m is the index of the synthesized subband,

.

It should be noted that typically the analysis filter bank is configured to generate a number of analysis subband signals; the subband processing module is configured to determine a plurality of synthesis subband signals from the plurality of analyzed subband signals; and the synthesis filter bank is configured to generate a time-stretched and/or frequency-transformed signal from the plurality of synthesis subband signals.

физического растягивания во времени и/или преобразованным по частоте с коэффициентом

физического преобразования по частоте. В этом случае коэффициент растягивания поддиапазона может иметь вид

, коэффициент преобразования поддиапазона может иметь вид

; и/или индекс n анализируемого поддиапазона, связанный с сигналом анализируемого поддиапазона, и индекс m синтезируемого поддиапазона, связанный с сигналом синтезируемого поддиапазона, могут быть связаны соотношением

. Если выражение

не является целым числом, n может выбираться как ближайшее, т.е. ближайшее меньшее или большее, целое число к члену

.In one embodiment of the invention, the system may be configured to generate a signal that is spread out in time by a factor

physical stretching in time and / or converted in frequency with a coefficient

physical frequency conversion. In this case, the subband stretch factor can be of the form

, the subband transformation coefficient can be of the form

; and/or the analysis subband index n associated with the analysis subband signal and the synthesis subband index m associated with the synthesis subband signal may be related by

. If the expression

is not an integer, n can be chosen as the closest, i.e. nearest lesser or greater integer to the term

.

The system may include a control data receiving module configured to receive control data indicative of instantaneous acoustic properties of the input signal. Said instantaneous acoustic properties can, for example, be reflected by classifying the input signal into different acoustic property classes. These classes may include a class of transient properties for a short-term non-periodic signal, and/or a class of stationary properties for a stationary signal. The system may include a signal classifier or may receive control data from a signal classifier. The signal classifier may be configured to analyze the instantaneous acoustic properties of the input signal and/or configured to provide control data indicative of the instantaneous acoustic properties.

The subband processing module may be configured to determine the synthesis subband signal given the control data. In particular, the block extractor may be configured to set the frame length L in accordance with the control data. In one of the embodiments of the invention, a small frame length L is set if the control data reflects a short-term non-periodic signal; and/or a large frame length L is set if the control data reflects a stationary signal. In other words, the frame length L may be reduced for short non-periodic signal parts compared to the frame length L used for stationary signal parts. Thus, the instantaneous acoustic properties of the input signal can be taken into account in the subband processing module. As a result, the efficiency of the system in relation to short non-periodic and/or voice signals can be increased.

As described above, the analysis filter bank is typically configured to generate a number of analysis subband signals. In particular, the analysis filter bank may be configured to generate a second analysis subband signal from the input signal. Said second analysis subband signal is typically associated with a different frequency band of the input signal than the analysis subband signal. The second analysis subband signal may include a number of complex-valued second analysis samples.

The subband processing module may include a second block extractor configured to extract a set of second input samples by applying a block jump value p to a set of second samples to be analyzed, i. in one of the preferred embodiments of the invention, the second block extractor uses a length L =1 frame. Typically, every second input sample corresponds to a frame consisting of input samples. This correspondence may be related to timing and/or sampling features. In particular, the second input sample and the corresponding frame consisting of the input samples may refer to the same time points of the input signal.

The subband processing module may include a second non-linear frame processing module configured to determine a frame of second processing samples from a frame of input samples and a corresponding second input sample. Determination of a frame consisting of the second processing samples may be performed by determining, for each second processing sample of the frame, the phase of the second processing sample by shifting the phase of the corresponding input sample by a phase shift amount that is based on the corresponding second input sample, the conversion factor Q and a subband stretch factor S. In particular, the phase shift may be performed as described herein, where the second processed sample is taken instead of a predetermined input sample. In addition, determination of a frame consisting of the second processing samples may be performed by determining, for each second processing sample of the frame, the amplitude of the second processing sample based on the amplitude of the corresponding input sample and the amplitude of the corresponding second input sample. In particular, the amplitude may be determined as described herein, where the second processed sample is taken in place of the predetermined input sample.

Thus, a second non-linear frame processing module can be used to extract a frame, or set of frames, consisting of processed samples from frames taken from two different analyzed subband signals. In other words, a single synthesis subband signal may be derived from two or more different analyzed subband signals. As described herein, this can be useful when a single pair of analysis and synthesis filterbanks are used for a number of harmonic transposition orders and/or time stretching degrees.

является целым числом n, то сигнал синтезируемого поддиапазона может определяться на основе кадра, состоящего из обрабатываемых дискретных значений, т.е. сигнал синтезируемого поддиапазона может определяться из единичного сигнала анализируемого поддиапазона, соответствующего целочисленному индексу n. В альтернативном варианте или в дополнение, можно оговорить, что если член

не является целым числом, где n является ближайшим целым числом, то сигнал синтезируемого поддиапазона может определяться на основе кадра, состоящего из вторых обрабатываемых дискретных значений, т.е. сигнал синтезируемого поддиапазона может определяться из двух сигналов анализируемых поддиапазонов, соответствующих ближайшему целочисленному значению индекса n и смежному целочисленному значению индекса. В частности, сигнал второго анализируемого поддиапазона может соответствовать индексу анализируемого поддиапазона n+1 или n–1.In order to determine one or two analysis subbands that should contribute to the synthesis subband with index m , the ratio between the frequency resolution of the analysis and synthesis filterbanks can be taken into account. In particular, it can be stipulated that if a member

is an integer n , then the synthesis subband signal may be determined based on the frame consisting of processed samples, i.e. a synthesis subband signal may be determined from a unit analysis subband signal corresponding to an integer index n . Alternatively, or in addition, it may be stipulated that if a member

is not an integer, where n is the nearest integer, then the synthesis subband signal may be determined based on the frame consisting of the second processed samples, i. the synthesis subband signal may be determined from the two analyzed subband signals corresponding to the nearest integer index value n and the adjacent integer index value. In particular, the second analysis subband signal may correspond to an analysis subband index n +1 or n -1.

According to the following feature, a system configured to generate a time-stretched and/or frequency-converted signal from an input signal is described. This system is particularly adapted to generate a time-stretched and/or frequency-transformed signal under the action of a control signal and thus to take into account the instantaneous acoustic properties of the input signal. This may be particularly well suited to improve the transient response of the system.

The system may include a control data receiving module configured to receive control data indicative of instantaneous acoustic properties of the input signal. In addition, the system may include an analysis filter bank configured to generate an analysis subband signal from an input signal; where the analyzed subband signal includes a number of complex-valued analyzed discrete values, each of which has a phase and amplitude. In addition, the system may include a subband processing module configured to determine the synthesis subband signal from the analyzed subband signal using the subband transform coefficient Q , the subband stretch coefficient S , and control data. Typically, at least one of the coefficients, Q or S , is greater than one.

The subband processing module may include a block extractor configured to extract a frame containing L input samples from a set of complex-valued analyzed samples. The frame length L may be greater than one. In addition, the block extractor may be configured to set the frame length L in accordance with the control data. The block extractor may also be configured to apply a block jump amount of p samples to the set of analyzed samples before extracting the next frame of L input samples; thus, a set of frames consisting of input discrete values is generated.

As described above, the subband processing module may include a non-linear frame processing module configured to determine a frame of processed samples from a frame of input samples. This can be done by determining, for each processed sample of the frame, the phase of the processed sample by shifting the phase of the corresponding input sample; and by determining, for each frame sample to be processed, the amplitude of the processed sample based on the amplitude of the corresponding input sample.

In addition, as described above, the system may include an overlay and add module configured to determine a synthesis subband signal by overlaying and adding samples from a set of frames consisting of processed samples; and a synthesis filter bank configured to generate a time-stretched and/or frequency-transformed signal from the synthesis subband signal.

In another aspect, a system configured to generate a time-stretched and/or frequency-converted signal from an input signal is described. The system may be particularly well adapted to perform a number of time stretching and/or frequency transform operations in a single pair of analysis and synthesis filter banks. The system may include an analysis filter bank configured to generate first and second analysis subband signals from an input signal, where each of the first and second subband signals includes a number of complex-valued analysis samples, referred to as the first and second analysis samples, respectively, and each the analyzed discrete value has a phase and an amplitude. Typically, the first and second signals of the analyzed subbands correspond to different frequency bands of the input signal.

The system may also include a subband processing module configured to determine a synthesis subband signal from the first and second analyzed subband signals using the subband transform factor Q and the subband stretch factor S. Typically, at least one of the coefficients, Q or S , is greater than one. The subband processing module may include a first block extractor configured to extract a frame consisting of L first input samples from a set of first analyzed samples; frame length L is greater than one. The first block extractor may be configured to apply a block hop amount of p samples to a set of first parsed samples before extracting the next frame of the L first input samples; and thus a set of frames consisting of the L first input samples is generated. Further, the subband processing module may include a second block extractor configured to extract a set of second input samples by applying a block jump value p to a set of second analyzed samples; where every second input sample corresponds to a frame consisting of the first input samples. The first and second block extractors may have any of the features described herein.

The subband processing module may include a non-linear frame processing module configured to determine a frame of processed samples from a frame of first input samples and a corresponding second input sample. This can be done by determining for each processed sample of the frame the phase of the processed sample - by shifting the phase of the corresponding first input sample; and/or determining, for each frame sample to be processed, the amplitude of the processed sample based on the amplitude of the corresponding first input sample and the amplitude of the corresponding second input sample. In particular, the non-linear frame processing module may be configured to determine the phase of the processed sample by shifting the phase of the corresponding first input sample by a phase shift amount that is based on the corresponding second input sample, the transform factor Q , and the subband stretch factor S.

In addition, the subband processing module may include an overlay and add module configured to determine a synthesis subband signal by overlaying and adding samples from a set of frames consisting of processed samples, where the overlay and addition module may apply a hop amount to successive frames consisting of processed discrete values. The hop amount may be equal to the block hop amount p times the subband stretch factor S. Finally, the system may include a synthesis filter bank configured to generate a time-stretched and/or frequency-transformed signal from the synthesis subband signal.

It should be noted that the various components of the systems described herein may include any or all of the features described herein in relation to those components. This applies in particular to the analysis and synthesis filter banks, the subband processor, the non-linear processor, the block extractors, the overlay and add module, and/or the windowing module described in various parts of this document.

The systems described in this document may include a number of sublayers processing modules. Each module for processing the sublayapsons can be configured to determine the intermediate signal of the synthesized sublayer with the use of various coefficients of Q transformation and/or various coefficients of the sub -projection S. Systems can also include, in a descending direction relative to a number of sub disperse processing modules and in the ascending direction relative to the synthesizing filter unit, a merging module configured for the merger of the corresponding intermediate signals of the synthesized subhyanias into the synthesized sublayer signal. Thus, systems can be used to perform a number of stretching operations in time and/or harmonic transformation, despite the use of only the only pair of blocks of analyzing and synthesizing filters.

The systems may include, upstream of the analysis filter bank, a core decoder configured to decode the bitstream into an input signal. The systems may also include an HFR processing module downstream of the fusion module (if the fusion module is present) and upstream of the synthesis filter bank. The HFR processing unit may be configured to apply spectral band information extracted from the bitstream to the synthesis subband signal.

According to another aspect, an external device is described for decoding a received signal including at least a low frequency component of an audio signal. The external device may include a system according to any of the features and characteristics described herein for generating a high frequency audio signal component from a low frequency audio signal component.

According to the following feature, a method for generating a time-stretched and/or frequency-converted signal from an input signal is described. This method is particularly well adapted to improve the transient response of time stretching and/or frequency conversion operations. The method may include the step of generating an analysis subband signal from an input signal, where the analysis subband signal includes a number of complex valued analysis samples, each having phase and amplitude.

The method may generally include the step of determining a synthesis subband signal from an analyzed subband signal using a transform factor Q and a subband stretch factor S. Typically, at least one of the coefficients, Q or S , is greater than one. In particular, the method may include the step of extracting a frame consisting of L input samples from a set of complex-valued analyzed samples, where the frame length L is typically greater than one. In addition, a block jump value of p samples may be applied to the set of analyzed samples before extracting the next frame containing L input samples; thus, a set of frames consisting of input discrete values is generated. Further, the method may include the step of determining a frame of processed samples from a frame of input samples. This can be done by determining, for each successive frame sample, the phase of the processed sample by shifting the phase of the corresponding input sample.

Alternatively, or in addition, for each frame sample to be processed, the amplitude of the processed sample may be determined based on the amplitude of the corresponding input sample and the amplitude of the predetermined input sample.

The method may also include the step of determining the synthesis subband signal by overlaying and adding the samples from the set of frames, consisting of their processed samples.

Ultimately, a time-stretched and/or frequency-converted signal can be generated from the synthesis subband signal.

In another aspect, a method for generating a time-stretched and/or frequency-converted signal from an input signal is described. This method is particularly well adapted to improve the efficiency of time stretch and/or frequency transform operations in combination with short-term non-periodic input signals. The method may include the step of receiving control data indicative of instantaneous acoustic properties of the input signal. The method may also include the step of generating an analysis subband signal from an input signal, where the analysis subband signal includes a number of complex valued analysis samples, each having phase and amplitude.

In the next step, a synthesis subband signal can be determined from the analyzed subband signal using the subband transform factor Q , the subband stretch factor S , and control data. Typically, at least one of the coefficients, Q or S , is greater than one. In particular, the method may include the step of extracting a frame consisting of L input samples from a set of complex-valued analyzed samples, where the frame length L is typically greater than one, and where the frame length L is set in accordance with the control data. In addition, the method may include the step of applying a block hop value of p samples to a number of parsed subbands before extracting the next frame of L input samples, thereby generating a set of frames of input samples. Then, from a frame consisting of input discrete values, a frame consisting of processed discrete values can be determined by determining, for each processed discrete value of the frame, the phase of the processed discrete value by shifting the phase of the corresponding input discrete value, and the amplitude of the processed discrete value - based on the amplitude of the corresponding input discrete value.

The synthesis subband signal may be determined by superimposing and adding samples from a set of frames consisting of processed samples, and a time-stretched and/or frequency-converted signal may be generated from the synthesis subband signal.

According to the following feature, a method for generating a time-stretched and/or frequency-converted signal from an input signal is described. The technique may be particularly well adapted to perform a series of time stretching and/or frequency transform operations using a single pair of analysis/synthesis filter banks. At the same time, the method is well adapted to the processing of short-term non-periodic input signals. The method may include the step of creating first and second analysis subband signals from an input signal, where each of the analysis subband signals, the first and second, includes a number of complex-valued analysis samples, referred to as the first and second analysis samples, respectively, where each analysis sample has phase and amplitude.

In addition, the method may include the step of determining a synthesis subband signal from the first and second analyzed subband signals using a subband transform coefficient Q and a subband stretch coefficient S , where at least one of the coefficients, Q or S , is typically greater than one. In particular, the method may include the step of extracting a frame consisting of L first input samples from a set of first analyzed samples, where the frame length L is typically greater than one. A block jump value of p samples may be applied to the set of first parsed samples before extracting the next frame of the L first input samples, to thereby generate a set of frames of the first input samples. The method may also include the step of extracting a set of second input samples by applying a block jump value p to a set of second analyzed samples, where each second source sample corresponds to a frame consisting of the first input samples.

The method continues by determining a frame consisting of the processed samples, from a frame consisting of the first input samples, and from the corresponding second input sample. This can be done by determining for each processed sample of the frame the phase of the processed sample - by shifting the phase of the corresponding first input sample, and the amplitude of the processed discrete value - based on the amplitude of the corresponding first input discrete value and the amplitude of the corresponding second input discrete value.

Then, by superimposing and adding the samples from the set of frames consisting of the processed samples, the synthesis subband signal can be determined. Ultimately, a time-stretched and/or frequency-converted signal can be generated from the synthesis subband signal.

According to another aspect, a program implemented in software is described. A program implemented in software may be adapted to execute on a processor and to perform the method steps and/or implement the features and characteristics described herein when implemented on a computing device.

According to another aspect, a storage medium has been described. The storage medium may include a program implemented in software adapted to execute on a processor and to perform the method steps and/or implement the features and characteristics described herein when implemented on a computing device.

According to another aspect, a computer program product is described. The computer program product may include executable instructions for performing the steps of the method and/or for implementing the features and characteristics described herein when executed on a computer.

It should be noted that the methods and systems, including their preferred embodiments, as described in this patent application, can be used alone or in combination with other methods and systems disclosed in this document. In addition, all features of the methods and systems described in this patent application can be arbitrarily combined. In particular, the characteristic features of the claims can be freely combined with each other.

BRIEF DESCRIPTION OF GRAPHICS

The present invention will be described below by way of illustrative examples, without limiting the scope or spirit of the invention, with reference to the accompanying drawings, where:

fig. 1 illustrates the principle of an example of harmonic transformation based on a block of subbands;

fig. 2 illustrates the principle of operation of an example of non-linear processing of a block of subbands with a single input subband;

fig. 3 illustrates the principle of operation of an example of non-linear processing of a block of subbands with two input subbands;

fig. 4 illustrates an example scenario for applying a subband block transform using multiple transform orders in an HFR enhanced audio codec;

fig. 5 illustrates an exemplary subband block multiorder transform operation scenario applying a separate analysis filter bank for each transform order;

fig. 6 illustrates an example scenario for efficient operation of a multi-order subband block transform using a single 64-band QMF filter bank; And

fig. 7 illustrates the step response for stretching an example audio signal over time by a factor of two based on a block of subbands.

DESCRIPTION OF THE PREFERRED EMBODIMENTS

The embodiments of the invention described below are merely illustrative of the principles of the present invention for enhanced harmonic transform based on a block of subbands. It should be understood that modifications and changes to the arrangements and details described herein will be understood by those skilled in the art. Therefore, the intention is to be limited only by the scope of the following claims and not by the specific details provided herein for the purpose of describing and explaining embodiments of the invention.

Fig. 1 illustrates the principle of an example of transform, time stretch, or combination of transform and time stretch based on a block of subbands. The time-domain input signal is applied to the analysis filter bank 101, which creates a plurality, or series, of complex-valued subband signals. This set of subband signals is provided to subband processor 102, which is influenced by control data 104. Each output subband of subband processor 102 can be obtained by processing one or two input subbands, or even by superimposing the result for several the specified processed subranges. A plurality, or series, of complex-valued output subbands is provided to synthesis filter bank 103, which in turn outputs a modified time-domain signal. The control data 104 improves the quality of the modified signal in the time domain for some types of signals. Control data 104 may be associated with a signal in the time domain. In particular, the control data 104 may be related to the type of time domain signal, or dependent on the type of time domain signal supplied to the analysis filter bank 101 . For example, control data 104 may indicate whether a time domain signal, or an instantaneous time domain signal chunk, is a stationary signal, or a time domain signal is a short-term, non-periodic signal.

Fig. 2 illustrates the operation of an example of non-linear subband block processing 102 with a single input subband. Given the physical time stretch and/or transform target values, and the physical parameters of the analysis and synthesis filter blocks 101 and 103, the subband stretch time and transform parameters, as well as the source subband index, which may also be referred to as the analysis subband index, can be output for each a target subband index, which may also be referred to as a synthesized subband index. The goal of processing a block of subbands is to implement an appropriate transform, time stretch, or combination of transform and time stretch on a complex valued source subband signal to generate a target subband signal.

In the non-linear subband block processing 102, the block extractor 201 samples a finite-size sampled frame from the complex-valued input signal. The frame may be determined by the position of the input pointer and the subband transform factor. The specified frame is subjected to non-linear processing in a non-linear processing module 202, and then processed by a finite length window in module 203. The window 203 may be a Gaussian window, a cosine window, a Hamming window, a Hann window, a rectangular window, a Bartlett window, a Blackman window, etc. d. The resulting samples are added to the previous output samples in overlay and add module 204, where the position of the output frame may be determined by the position of the output pointer. The input pointer has a fixed increment, also called the block jump value, and the output pointer has an increment equal to that of the source pointer, by a factor corresponding to the subband stretch factor, i.e. the block jump value is multiplied by the subband stretch factor. Repeating this chain of operations will generate an output signal with a duration equal to the duration of the input subband signal, with a factor corresponding to the subband stretch factor (up to the length of the synthesis window), and with complex frequencies transformed with the subband transform factor.

The control data 104 may affect the non-linear block-based processing 102 of any of the processed blocks 201, 202, 203, 204. In particular, the control data 104 may control the length of the blocks extracted by the block extractor 201. In one embodiment of the invention, the block length is reduced if the control data 104 indicates that the time domain signal is a short-term non-periodic signal, while the block length is increased or maintained at a higher level if the control data 104 indicates that that the time domain signal is a stationary signal. Alternatively, or in addition, the control data 104 may affect the non-linear processing unit 202, such as the parameters used in the non-linear processing unit 202, and/or the windowing unit 203, such as the window used by the method processing unit 203. window.

Fig. 3 illustrates the operation of an example of non-linear subband block processing 102 with two input subbands. Given the target values of the physical time stretch and transform, and the physical parameters of the analysis and synthesis filter blocks 102 and 103, the time stretch and subband transform parameters are output, as well as two source subband indices for each target subband index. The goal of subband block processing is to implement an appropriate transform, time stretch, or combination of transform and time stretch on the combination of two complex-valued source subband signals to generate a target subband signal. The block extractor 301-1 samples a finite size frame consisting of samples from the first complex valued source subband, and the block extractor 301-2 samples the finite size frame consisting of samples from the second complex valued source subband. In one embodiment of the invention, one of the block extractors 301-1 and 301-2 may generate a single subrange sample, i. one of the block extractors 301-1 and 301-2 may apply a block length equal to one discrete value. The frames may be defined by a common input pointer position and a subband transform factor. The two frames extracted by block extractors 301-1 and 301-2, respectively, are non-linearly processed in module 302. Non-linear processing module 302 typically generates a single output frame from two input frames. The output frame is then processed by a finite length window method in module 203. The above process is repeated for the set of frames that are generated from the set of frames extracted from the two subband signals using the block hop amount. The set of output frames is superimposed and added in module 204 overlay and addition. Repeating this chain of operations will generate an output signal with a duration that is equal to the duration of the longest of the two input subband signals with the subband stretch factor (up to the length of the synthesis window). In case the two input subband signals carry the same frequencies, the output signal will have the complex frequencies converted with the subband transform factor.

As described in the context of FIG. 2, the control data 104 may be used to modify the operation of various non-linear processing units 102, such as the operation of the block extractors 301-1 and 301-2. In addition, it should be noted that the operations described above are generally performed for all analysis subband signals produced by the analysis filter bank 101 and for all synthesis subband signals that are input to the synthesis filter bank 103 .

In the following description with reference to FIG. 1 to 3, and by adding appropriate mathematical terminology, a description will be given of the principles of time stretching and transformation based on a block of subbands.

The two main configuration parameters for the harmonic converter and/or time stretcher in general are:

– требуемый коэффициент физического растягивания во времени; и•

– required coefficient of physical stretching in time; And

– требуемый коэффициент физического преобразования.•

is the required physical conversion factor.

и

– для приводимых ниже четырех параметров блока фильтров, каждый из которых измеряется в физических единицах измерения. В приведенных выше отношениях:The filter banks 101 and 103 may be of any complex exponent modulated type such as QMF or windowed DFT or wavelet transform, for example. The analysis filter bank 101 and the modulation synthesis filter bank 103 may be uniformly or non-uniformly arranged and may be determined from a wide range of prototype filters and/or windows. Although all of these second order choices have an impact on subsequent design details such as phase corrections and subband display control, the main system design parameters for subband processing can generally be derived from knowledge of two relationships −

And

– for the four filter bank parameters below, each of which is measured in physical units. In the above relationship:

–шаг по времени дискретного значения поддиапазона, или шаг по времени блока 101 анализирующих фильтров (например, измеряемый в секундах [c]);•

– subband sample time step, or analysis filter bank time step 101 (eg, measured in seconds [s]);

– разнос частот поддиапазонов блока 101 анализирующих фильтров (например, измеряемый в Герцах [1/c]);•

- a variety of frequencies of sublayers of the block 101 of analyzing filters (for example, measured in Hertza [1/c]);

–шаг по времени дискретного значения поддиапазона, или шаг по времени блока 103 синтезирующих фильтров (например, измеряемый в секундах [c]); и•

– subband sample time step, or synthesis filter bank time step 103 (eg, measured in seconds [s]); And

– разнос частот поддиапазонов блока 103 синтезирующих фильтров (например, измеряемый в Герцах [1/c]).•

– the frequency spacing of the subbands of the synthesis filter bank 103 (eg, measured in Hertz [1/s]).

To configure the subband processing module 102, the following parameters must be calculated:

;• S – subrange stretching factor, ie. a stretch factor that is applied in subband processing module 102 to achieve an overall physical time stretch of the signal in the time domain by a factor

;

; и• Q is the subband conversion factor, ie. a transform factor that is applied in subband processing module 102 to achieve an overall physical signal transform in the time domain with a factor

; And

• correspondence between source and target subband indices, where n denotes the index of the analyzed subband included in the subband processing module 102, and m denotes the corresponding synthesized subband at the output of the subband processing module 102.

дискретных значений анализируемых поддиапазонов на входе в модуль 102 обработки поддиапазонов. Эти

дискретных значений будут растягиваться в

дискретных значений модулем 102 обработки поддиапазонов, который применяет коэффициент S растягивания поддиапазона. На выходе блока 102 синтезирующих фильтров эти

дискретных значений приводят к выходному сигналу, имеющему физическую длительность

. Поскольку последняя указанная длительность должна совпадать с заданной величиной

, т.е. поскольку длительность выходного сигнала во временной области должна быть растянута во времени по сравнению с входным сигналом во временной области c коэффициентом физического растягивания во времени

, получается следующее правило проектирования:In determining the subband stretch factor S, it was observed that the input signal to the synthesis filter bank 101 with physical duration D corresponds to the number

discrete values of the analyzed subbands at the input to the subband processing module 102 . These

discrete values will stretch in

samples by subband processing module 102, which applies a subband stretch factor S. At the output of the synthesizing filter block 102, these

discrete values lead to an output signal having a physical duration

. Since the last specified duration must match the specified value

, i.e. since the duration of the output signal in the time domain must be stretched in time compared to the input signal in the time domain with a physical time stretch factor

, the following design rule is obtained:

. (1)

. (1)

, наблюдается, что входная синусоида в блок 101 анализирующих фильтров с физической частотой

будет приводить к комплексному сигналу анализируемого поддиапазона с зависящей от дискретного времени частотой

, и главный вклад вносится в анализируемый поддиапазон с индексом

. Выходная синусоида на выходе блока 103 синтезирующих фильтров с требуемой преобразованной физической частотой

будет генерироваться путем снабжения синтезируемого поддиапазона индексом

c комплексным сигналом поддиапазона с дискретной частотой

. В данном контексте следует проявлять осторожность во избежание синтеза ступенчатых выходных частот, отличающихся от

. Как правило, этого можно избежать путем создания соответствующих вторых порядков выбора так, как это обсуждалось выше, например, путем выбора соответствующих блоков анализирующих/синтезирующих фильтров. Дискретная частота

на выходе модуля 102 обработки поддиапазонов должна соответствовать зависящей от дискретного времени частоте

на входе модуля 102 обработки поддиапазонов, умноженной на коэффициент Q преобразования поддиапазона. Т.е. путем приравнивания

и

можно определить соотношение между коэффициентом

физического преобразования и коэффициентом Q преобразования поддиапазона:In order to determine the subband transform coefficient Q that is applied in the subband processing module 102 to achieve the physical transform

, it is observed that the input sinusoid into the analysis filter bank 101 with a physical frequency

will result in a complex signal of the analyzed subband with a discrete time-dependent frequency

, and the main contribution is made to the analyzed subrange with the index

. The output sinusoid at the output of the synthesis filter block 103 with the required converted physical frequency

will be generated by supplying the synthesized subband with an index

with a complex subband signal with a discrete frequency

. In this context, care should be taken to avoid synthesizing stepped output frequencies other than

. Typically, this can be avoided by creating appropriate second order selections as discussed above, for example by selecting appropriate analysis/synthesis filter banks. Discrete frequency

at the output of the subband processing module 102 must correspond to a discrete time-dependent frequency

at the input of subband processing module 102 multiplied by the subband transform factor Q. Those. by equalization

And

it is possible to determine the ratio between the coefficient

physical conversion and subband conversion coefficient Q :

. (2)

. (2)

Similarly, the corresponding source or analysis subband index n of the subband processing module 102 for a given target or synthesized subband index m should obey the relationship:

. (3)

. (3)

, т.е. разнос частот блока 103 синтезирующих фильтров соответствует разносу частот блока 101 анализирующих фильтров, умноженному на коэффициент физического преобразования, и может применяться взаимно однозначное отображение индекса анализируемого поддиапазона в индекс синтезируемого поддиапазона, n=m. В других вариантах осуществления изобретения отображение индекса поддиапазона может зависеть от деталей параметров блока фильтров. В частности, если отношение разносов частот блока 103 синтезирующих фильтров и блока 101 анализирующих фильтров отличается от коэффициента

физического преобразования, для данного целевого поддиапазона могут быть заданы один или два исходных поддиапазона. В случае двух исходных поддиапазонов может оказаться более предпочтительным использование двух смежных исходных поддиапазонов, соответственно, с индексами n и n+1. То есть первый и второй исходные поддиапазоны задаются как

, или

.In one of the options for implementing the invention, it is accepted that

, i.e. the frequency spacing of the synthesis filter bank 103 corresponds to the frequency spacing of the analysis filter bank 101 multiplied by the physical transform coefficient, and a one-to-one mapping of the analysis subband index to the synthesis subband index, n = m , may be applied. In other embodiments of the invention, the subband index display may depend on the details of the filter bank parameters. In particular, if the frequency spacing ratio of the synthesis filter bank 103 and the analysis filter bank 101 differs from the coefficient

physical transformation, one or two source subbands may be specified for a given target subband. In the case of two source subbands, it may be more preferable to use two adjacent source subbands, respectively, with indices n and n +1. That is, the first and second initial sub -dispactions are set as

, or

.

– входной сигнал в экстрактор 201, блоков и пусть р – входной шаг блока. Т.е.

– это комплекснозначный сигнал поддиапазона для анализируемого поддиапазона с индексом n. Блок, извлекаемый экстрактором 201 блоков, можно без потери общности рассматривать как определяемый L=2R+1 дискретных значений:The subband processing of FIG. 2 with a single source subband will be described below as a function of the subband processing parameters S and Q . Let

is the input signal to the extractor 201, blocks and let p be the input step of the block. Those.

is the complex-valued subband signal for the analyzed subband with index n . The block extracted by the block extractor 201 can, without loss of generality, be considered as defined by L =2 R +1 discrete values:

, (4)

, (4)

блок извлекается из последовательных дискретных значений, а при

понижающая дискретизация выполняется так, чтобы входные адреса растягивались с коэффициентом Q. Если Q – целое число, эта операция, как правило, является простой для выполнения, в то время как для нецелочисленных значений Q может потребоваться интерполяция. Данное утверждение также справедливо и для нецелочисленных значений приращения р, т.е. входного шага блока. В одном из вариантов осуществления изобретения к комплекснозначному сигналу поддиапазона могут применяться короткие интерполирующие фильтры, например, фильтры, содержащие два отвода фильтра. Например, если требуется дискретное значение с дробным временным индексом k+0,5, к достаточному качеству может приводить интерполяция с двумя отводами в форме

.where the integer l is the block count index, L is the block length, and R is an integer, R >0. Note that when

the block is extracted from successive discrete values, and when

downsampling is done so that the input addresses are stretched by a factor Q . If Q is an integer, this operation is usually easy to perform, while interpolation may be required for non-integer values of Q. This statement is also true for non-integer values of the increment p , i.e. block input step. In one embodiment of the invention, short interpolation filters, such as filters comprising two filter taps, may be applied to the complex-valued subband signal. For example, if a discrete value with a fractional time index k +0.5 is required, a two-tap interpolation of the form

.

An interesting special case of formula (4) is the case R = 0, where the extracted block consists of a single discrete value, i.e. block length L =1.

, где

– амплитуда комплексного числа, и

– фаза комплексного числа, блок 202 нелинейной обработки, генерирующий выходной кадр

из входного кадра

, преимущественно определяется с коэффициентом модификации фазы

черезIn the polar performance of a complex number

, Where

- amplitude of a complex number, and

– complex number phase, non-linear processing unit 202 generating output frame

from the input frame

, mainly determined with the coefficient of phase modification

through

,

, (5)

,

, (5)

– параметр геометрического взвешивания амплитуды. Случай

соответствует чистой модификации фазы извлеченного блока. Параметр

коррекции фазы зависит от деталей блока фильтров и индексов исходного и целевого поддиапазонов. В одном из вариантов осуществления изобретения параметр

коррекции фазы может определяться экспериментально путем развертки набора входных синусоид. Кроме того, параметр

коррекции фазы может быть выведен путем изучения разности фаз смежных комплексных синусоид целевых поддиапазонов или путем оптимизации эффективности для входного сигнала, относящегося к типу импульса Дирака. Коэффициент Т модификации фазы должен быть целым числом так, чтобы коэффициенты Т–1 и 1 в линейной комбинации фаз в первой строке формулы (5) были целыми числами. При таком допущении, т.е. при допущении того, что коэффициент Т модификации фазы является целым числом, результат нелинейной модификации хорошо определяется даже тогда, кода фазы являются неоднозначными за счет добавления произвольных целочисленных множителей

.Where

is the amplitude geometric weighting parameter. Happening

corresponds to the net phase modification of the extracted block. Parameter

phase correction depends on the details of the filter bank and the indices of the source and target subbands. In one of the embodiments of the invention, the parameter

phase correction can be determined experimentally by sweeping a set of input sinusoids. In addition, the parameter

The phase correction can be derived by examining the phase difference of adjacent target subband complex sinusoids, or by optimizing the efficiency for an input signal related to the Dirac pulse type. The phase modification coefficient T must be an integer so that the coefficients T –1 and 1 in the linear combination of phases in the first line of formula (5) are integers. With this assumption, i.e. assuming that the phase modification factor T is an integer, the result of the non-linear modification is well defined even when the phase codes are ambiguous by adding arbitrary integer factors

.

Thus, formula (5) determines that the phase of the output frame sample is determined by shifting the phase of the corresponding input frame sample by a constant shift amount. This shift constant may depend on the modification factor T , which in turn depends on the subband stretch factor and/or the subband transform factor.

коррекции фазы, который может определяться экспериментально.In addition, the constant amount of the shift may depend on the phase of the particular sample of the input frame from the input frame. This particular input frame sample is held fixed to determine the phase of all output frame samples of that block. In the case of formula (5), the phase of the central discrete value of the input frame is used as the phase of the special discrete value of the input frame. In addition, the constant value of the shift may depend on the parameter

phase correction, which can be determined experimentally.

The second line of formula (5) specifies that the amplitude of the output frame sample may depend on the amplitude of the corresponding input frame sample. In addition, the amplitude of an output frame sample may depend on the amplitude of a specific sample of the input frame. This particular input frame sample can be used to determine the amplitude for all frame output samples. In the case of formula (5), the central discrete value of the input frame is used as a special discrete value of the input frame. In one embodiment of the invention, the output frame sample amplitude may correspond to the geometric mean of the amplitudes of the corresponding input frame sample and the particular input frame sample.

In the windowing module 203, a window of length L is applied to the output frame, resulting in a windowed output frame:

. (6)

. (6)

Finally, frames are assumed to continue with zeros, and the overlay and addition operation is given by the equation:

, (7)

, (7)

в S раз больше длительности входного сигнала

, т.е. сигнал синтезируемого поддиапазона является растянутым с коэффициентом S растягивания поддиапазона по сравнению с сигналом анализируемого поддиапазона. Следует отметить, что данное наблюдение, как правило, предполагает, что длина L окна пренебрежимо мала по сравнению с длительностью сигнала.where it should be noted that the overlay and add module 204 applies the block step Sp , i. e. a time step that is S times greater than the input block step p . Due to the indicated difference in time steps in formulas (4) and (7), the duration of the output signal

S times the duration of the input signal

, i.e. the synthesis subband signal is stretched by a subband stretch factor S compared to the analyzed subband signal. It should be noted that this observation generally assumes that the length L of the window is negligible compared to the duration of the signal.

In the case where a complex sine wave is used as input to the subband processing 102, i. e. the signal of the analyzed subband corresponds to a complex sinusoid

, (8)

, (8)

by applying formulas (4) - (7) it can be determined that the output signal of the subband processing 102, i.e. the corresponding signal of the synthesized subband has the form:

. (9)

. (9)

будет трансформироваться в комплексную синусоиду с зависящей от дискретного времени частотой

при условии, что сдвиги окна с шагом Sp суммируются в одну и ту же постоянную величину K для всех k:Thus, a complex sinusoid with a discrete time-dependent frequency

will be transformed into a complex sinusoid with a discrete time-dependent frequency

provided that window shifts with step Sp sum to the same constant value K for all k :

(10)

(10)

и

. Если входной шаг блока р=1 и R=0, все вышеприведенное, т.е., в особенности, формула (5), сокращается до точечного правила модификации фазы, или правила модификации фазы на основе дискретного значения:To illustrate, consider a special case of a pure transformation, where

And

. If the block input step p = 1 and R = 0, all of the above, i.e., especially formula (5), is reduced to a point phase modification rule, or a phase modification rule based on a discrete value:

. (11)

. (eleven)

становится очевидным, когда в пределах сигнала

анализируемого поддиапазона рассматривается сумма синусоид. Трудность, связанная с точечным правилом (11) для суммы синусоид с частотами

заключается в том, что в выходном сигнале обработки 102, т.е. в сигнале

синтезируемого поддиапазона, будут присутствовать не только желательные частоты

, но и частоты комбинационных составляющих в форме

. Использование блока

и удовлетворяющей окну формулы (10), как правило, приводит к подавлению этих комбинационных составляющих. С другой стороны, длинный блок будет приводить к большей степени нежелательного размывания времени для кратковременных непериодических сигналов. Кроме того, для сигналов в форме последовательности импульсов, например, человеческого голоса в случае гласных звуков или однотонного инструмента, с относительно низким основным тоном, комбинационные составляющие могут оказаться желательными, как это описано в документе WO 2002/052545. Данный документ ссылкой включен в настоящее описание.Advantage of using block size

becomes apparent when within the signal

of the analyzed subrange, the sum of the sinusoids is considered. The difficulty associated with the point rule (11) for the sum of sinusoids with frequencies

lies in the fact that in the output signal of the processing 102, i.e. in signal

of the synthesized subband, not only the desired frequencies will be present

, but also the frequencies of the combinational components in the form

. Block usage

and satisfies the window of formula (10), as a rule, leads to the suppression of these combination components. On the other hand, a long block will result in a greater degree of unwanted time blur for short-term non-periodic signals. In addition, for pulse train signals, such as the human voice in the case of vowels, or a monophonic instrument with a relatively low fundamental, combinational components may be desirable, as described in WO 2002/052545. This document is incorporated herein by reference.

. Наблюдалось (см., например, фиг. 7), что выбор параметра геометрического взвешивания амплитуды

улучшает переходную характеристику обработки 102 на основе блока поддиапазонов по сравнению с использованием чистой модификации фазы с

, в то же время, c сохранением достаточной мощности подавления интермодуляционного искажения для стационарных сигналов. Наиболее привлекательно значение параметра взвешивания амплитуды

, для которого формула (5) нелинейной обработки сокращается до следующих этапов вычислений:To solve the problem of relatively low processing efficiency 102 based on a block of subbands for short-term non-periodic signals, it is proposed to use in formula (5) a non-zero value of the geometric amplitude weighting parameter

. It has been observed (see, for example, Fig. 7) that the choice of the amplitude geometric weighting parameter

improves the transient response of subband block processing 102 compared to using a pure phase modification with

, while maintaining sufficient IM distortion suppression power for stationary signals. The most attractive value of the amplitude weighting parameter

, for which the formula (5) of nonlinear processing is reduced to the following calculation steps:

. (12)

. (12)

в формуле (5). Иными словами, определение амплитуды дискретных значений выходного кадра на основе формулы (5) геометрических средних с использованием параметра взвешивания амплитуды

может реализовываться без каких-либо дополнительных затрат на вычислительную сложность. В то же время, эффективность гармонического преобразователя для кратковременных непериодических сигналов увеличивается при сохранении эффективности для стационарных сигналов.These computational steps represent an amount of computational complexity that is equivalent compared to the work of pure phase modulation in the case

in formula (5). In other words, determining the amplitude of discrete values of the output frame based on the formula (5) of geometric averages using the amplitude weighting parameter

can be implemented without any additional cost in computational complexity. At the same time, the efficiency of the harmonic converter for short-term non-periodic signals increases while maintaining the efficiency for stationary signals.

As has been described in the context of FIG. 1, 2, and 3, subband processing 102 can be further enhanced by applying control data 104. In one embodiment of the invention, two configurations of subband processing 102 that share the same value can be used to implement signal-adaptive subband processing. K in formula (11) and can involve different block lengths. A conceptual starting point in designing a subband processing module with signal-adapted configuration switching would be to have two configurations run in parallel, with a selector switch at their outputs, where the position of the selector switch depends on the control data 104. Sharing the value of K will ensure that switching will occur without abrupt transitions in the case of a single output complex sinusoid. For generalized signals, hard switching at the subband signal level will be automatically windowed by the surrounding filter bank design 101, 103 so that no switching artifacts are introduced into the final output signals. It can be shown that when the block sizes differ significantly and the update rate of the control data is not very high, the superposition and addition process of formula (7) can produce an output signal similar to that of the above-described conceptual switching system, at the computational cost of a system having the configuration with the longest block. Thus, there is no loss in computational complexity associated with the work of adaptation to the signal. As discussed above, the shorter block length configuration is more suitable for short-term non-periodic signals and low pitch periodic signals, while the longer block length configuration is more suitable for stationary signals. Thus, a signal classifier can be used to classify audio fragments into a short-term non-periodic signal class and a non-transient signal class and pass this classification information as control data 104 to the subband processing module 102 with signal-adaptive configuration switching. Subband processing module 102 may use control data 104 to specify certain processing parameters, such as block length in block extractors.

– входной сигнал поддиапазона в первый экстрактор 301-1 блоков, и пусть

– входной сигнал поддиапазона во второй экстрактор 301-2 блоков. Блок, извлекаемый экстрактором 301-1 блоков определяется формулой (4), и блок, извлекаемый экстрактором 301-2 состоит из единственного дискретного значения:Below, the description of subband processing will be extended to the case of FIG. 3 with two subband inputs. Only the changes made to the single input case will be described. In other words, reference is made to the information provided above. Let

is the subband input to the first block extractor 301-1, and let

– subband input signal to the second block extractor 301-2. The block extracted by the block extractor 301-1 is determined by formula (4), and the block extracted by the extractor 301-2 consists of a single discrete value:

. (13)

. (13)

, который может определяться формулойThose. in the described embodiment, the first block extractor 301-1 uses a block length L while the second block extractor 301-2 uses a block length 1. In this case, the non-linear processing 302 generates an output frame

, which can be determined by the formula

, (14)

, (14)

and the rest of the processing in modules 203 and 204 is similar to the processing described in the context of the single input case. In other words, it is proposed to replace the specific frame sample in formula (5) with a single subband sample derived from the corresponding second signal of the analyzed subband.

блока 103 синтезирующих фильтров к разносу частот

блока 101 анализирующих фильтров отличается от требуемого коэффициента

физического преобразования, может оказаться полезным определение дискретных значений синтезируемого поддиапазона с индексом m из двух анализируемых поддиапазонов, соответственно, с индексами n и n+1. При заданном индексе m соответствующий индекс n может даваться целочисленным значением, полученным путем усечения индекса анализируемого поддиапазона, заданного формулой (3). Один из сигналов анализируемых поддиапазонов, например, сигнал анализируемого поддиапазона, соответствующий индексу n, подается в первый экстрактор 301-1 блоков, и второй сигнал анализируемого поддиапазона, например, сигнал, соответствующий индексу n+1, подается во второй экстрактор 301-2 блоков. На основе этих двух сигналов анализируемых поддиапазонов в соответствии с описанной выше обработкой определяется сигнал синтезируемого поддиапазона, соответствующий индексу m. Направление смежных сигналов анализируемых поддиапазонов в два экстрактора 301-1 и 302-1 блоков может основываться на остатке, получаемом при усечении значения индекса по формуле (3), т.е. на разности точного значения индекса, данного по формуле (3), и усеченного целочисленного значения n, полученного из формулы (3). Если остаток больше 0,5, то сигнал анализируемого поддиапазона, соответствующий индексу n, может направляться во второй экстрактор 301-2 блоков, в противном случае этот сигнал анализируемого поддиапазона может направляться в первый 301-1 экстрактор блоков.In one embodiment of the invention, where the frequency separation ratio

block 103 synthesis filters to frequency spacing

block 101 analyzing filters differs from the required coefficient

physical transformation, it may be useful to determine the discrete values of the synthesized subrange with index m from the two analyzed subranges, respectively, with indices n and n +1. Given an index m, the corresponding index n may be given an integer value obtained by truncating the analysis subband index given by formula (3). One of the analysis subband signals, e.g., the analysis subband signal corresponding to index n , is supplied to the first block extractor 301-1, and the second analysis subband signal, e.g., the signal corresponding to index n +1, is supplied to the second block extractor 301-2. Based on these two analyzed subband signals, the synthesis subband signal corresponding to index m is determined in accordance with the above-described processing. The direction of adjacent analysis subband signals to the two block extractors 301-1 and 302-1 may be based on the remainder obtained by truncating the index value according to formula (3), i.e. on the difference between the exact value of the index given by formula (3) and the truncated integer value n obtained from formula (3). If the remainder is greater than 0.5, then the analysis subband signal corresponding to index n may be sent to the second block extractor 301-2, otherwise, this analysis subband signal may be sent to the first block extractor 301-1.

. Этот базовый сигнал с низкой полосой пропускания также может именоваться низкочастотной составляющей звукового сигнала. Сигнал с низкой частотой дискретизации

может подвергаться повторной дискретизации до выходной частоты дискретизации

посредством 32-полосного блока 402 комплексных модулированных QMF-фильтров, а затем – 64-полосного блока 405 синтезирующих QMF-фильтров (обратных QMF-фильтров). Оба блока 402 и 405 фильтров имеют одинаковые физические параметры

и

, и блок 404 HFR-обработки, как правило, пропускает немодифицированные более низкие поддиапазоны, соответствующие базовому сигналу с низкочастотной полосой пропускания. Высокочастотное информационное наполнение выходного сигнала получается путем снабжения более высоких поддиапазонов 64-полосного блока 405 синтезирующих QMF-фильтров выходными полосами из модуля 403 многократного преобразователя, подвергнутыми формированию и модификации спектра, выполняемыми модулем 404 HFR-обработки. Многократный преобразователь 403 в качестве входного сигнала принимает декодированный базовый сигнал и выводит множество сигналов поддиапазонов, которые представляют 64-полосный анализ суперпозиции, или комбинации, нескольких преобразованных составляющих сигнала. Иными словами, сигнал на выходе многократного преобразователя 403 должен соответствовать преобразованным сигналам синтезируемых поддиапазонов, которые могут подаваться в блок 103 синтезирующих фильтров, который в случае фиг. 4 представлен блоком 405 обратных QMF-фильтров. Fig. 4 illustrates an example scenario for applying a subband block based transform using multiple transform orders in an HFR enhanced audio codec. The transmitted bitstream is received by the core decoder 401, which produces a low-bandwidth decoded core signal with a sampling rate of

. This low bandwidth base signal may also be referred to as the low frequency component of the audio signal. Low sample rate signal

can be resampled up to the output sample rate

through a 32-band complex modulated QMF filter bank 402, and then a 64-band QMF synthesis filter bank 405 (inverse QMF filters). Both filter banks 402 and 405 have the same physical parameters.

And

, and the HFR processing block 404 typically passes unmodified lower subbands corresponding to the base signal with a low frequency bandwidth. The high-frequency output signal content is obtained by supplying the higher subbands of the 64-band QMF synthesis filter block 405 with output bands from the multiple transducer module 403 subjected to spectrum shaping and modification by the HFR processing module 404. The multiple transducer 403 takes the decoded base signal as input and outputs a plurality of subband signals that represent a 64-band analysis of a superposition, or combination, of several transformed signal components. In other words, the signal at the output of the multiple converter 403 must correspond to the converted synthesis subband signals that can be applied to the synthesis filter bank 103, which in the case of FIG. 4 is represented by an inverse QMF filter block 405 .

, и 

). Для кратковременных непериодических составляющих базового сигнала HFR-обработка иногда может компенсировать плохую переходную характеристику многократного преобразователя 403, но последовательно высокое качество, как правило, может достигаться только в том случае, если является удовлетворительной переходная характеристика самого многократного преобразователя. Как описывается в настоящем документе, управляющий сигнал 104 преобразователя может влиять на работу многократного преобразователя 403 и, таким образом, обеспечивать удовлетворительную переходную характеристику многократного преобразователя 403. В альтернативном варианте или в дополнение, приведенная выше схема геометрического взвешивания (см., например, формулу (5) и/или формулу (4)) может вносить вклад в улучшение переходной характеристики гармонического преобразователя 403.Possible implementations of multiple converter 403 are described in the context of FIG. 5 and 6. The purpose of multiple transform 403 is that, if HFR processing 404 is bypassed, each component corresponds to an integer physical transform without time-stretching of the base signal (

, And

). For short-term non-periodic components of the base signal, HFR processing can sometimes compensate for the poor transient response of the multiple transducer 403, but consistently high quality can generally only be achieved if the transient response of the multiple transducer itself is satisfactory. As described herein, transducer control signal 104 can influence the operation of multiple transducer 403 and thus provide satisfactory transient response to multiple transducer 403. Alternatively or in addition, the above geometric weighting scheme (see, for example, formula ( 5) and/or formula (4)) may contribute to improving the transient response of the harmonic converter 403.

, генерируются и доставляются три порядка преобразования

. Модуль 504 слияния выбирает и комбинирует соответствующие поддиапазоны из каждой ветви коэффициентов преобразования в единое множество QMF-поддиапазонов, подаваемых в модуль HFR-обработки.Fig. 5 illustrates an exemplary scenario for a multi-order subband block transform module 403 applying a separate filter block 502-2, 502-3, 502-4 for each transform order. In the illustrated example, in the region of a 64-band QMF filter bank operating at the output sample rate

, three orders of transformation are generated and delivered

. Merging module 504 selects and combines respective subbands from each transform coefficient branch into a single set of QMF subbands supplied to the HFR processing module.

. Цель непосредственно заключается в том, чтобы последовательность обработки из 64-полосного QMF-анализа 502-2, модуля 503-2 обработки поддипазонов и 64-полосного QMF-синтеза 405 приводила к физическому преобразованию с

и

(т.е. без растягивания). При идентификации указанных трех блоков как модулей 101, 102 и 103 по фиг. 1, обнаруживается, что

и

, таким образом, формулы (1) - (3) приводят к следующим техническим условиям для модуля 503-2 обработки поддиапазонов. Модуль 503-2 обработки поддиапазонов должен выполнить растягивание поддиапазона до

, преобразование поддиапазона до

(т.е. отсутствует) и обеспечить соответствие между исходными поддиапазонами с индексом n и целевыми поддиапазонами с индексом m, имеющее вид n=m (см. формулу (3)).Consider the first case

. The intention is directly that the processing sequence from the 64-band QMF analysis 502-2, the sub-band processing module 503-2, and the 64-band QMF synthesis 405 results in a physical transformation with

And

(i.e. without stretching). By identifying these three blocks as modules 101, 102 and 103 of FIG. 1, it is found that

And

, thus formulas (1) to (3) lead to the following specifications for the subband processing module 503-2. The subband processing module 503-2 must perform subband stretching to

, converting the subrange to

(i.e., absent) and provide a correspondence between the source subranges with index n and the target subranges with index m, having the form n = m (see formula (3)).

пример системы включает преобразователь 501-3 частоты дискретизации, который понижает входную частоту дискретизации с коэффициентом 3/2 – от

до

. Цель непосредственно заключается в том, чтобы последовательность обработки, состоящей из 64-полосного QMF-анализа 502-3, модуля 503-3 обработки поддипазонов и 64-полосного QMF-синтеза 405, приводила к физическому преобразованию с

и

(т.е. без растягивания). При идентификации приведенных выше трех блоков, соответственно, как модулей 101, 102 и 103 по фиг. 1, обнаруживается, что из-за повторной дискретизации

и

, таким образом, формулы (1) - (3) приводят к следующим техническим условиям для модуля 503-3 обработки поддиапазонов. Модуль 503-3 обработки поддиапазонов должен выполнить растягивание поддиапазона до

, преобразование поддиапазона до

(т.е. не выполнять) и обеспечить соответствие между исходными поддиапазонами с индексом n и целевыми поддиапазонами с индексом m, имеющее вид n=m (см. формулу (3)).For the occasion

The example system includes a 501-3 sample rate converter that downsamples the input sample rate by a factor of 3/2 - from

before

. The goal is directly that the processing sequence consisting of the 64-band QMF analysis 502-3, the sub-band processing module 503-3, and the 64-band QMF synthesis 405 results in a physical transformation with

And

(i.e. without stretching). By identifying the above three blocks, respectively, as modules 101, 102, and 103 of FIG. 1, it is found that due to resampling

And

, thus formulas (1) to (3) lead to the following specifications for the subband processing module 503-3. The subband processing module 503-3 must perform subband stretching to

, converting the subrange to

(i.e., do not perform) and provide a match between the source subranges with index n and the target subranges with index m, having the form n = m (see formula (3)).

пример системы включает преобразователь 501-4 частоты дискретизации, который понижает входную частоту дискретизации в два раза – от

до

. Цель непосредственно заключается в том, чтобы последовательность обработки, состоящей из 64-полосного QMF-анализа 502-4, модуля 503-4 обработки поддипазонов и 64-полосного QMF-синтеза 405, приводила к физическому преобразованию с

и

(т.е. без растягивания). При идентификации приведенных выше трех блоков, соответственно, как модулей 101, 102 и 103 по фиг. 1, обнаруживается, что из-за повторной дискретизации

и

, формулы (1) - (3) приводят к следующим техническим условиям для модуля 503-4 обработки поддиапазонов. Модуль 503-4 обработки поддиапазонов должен выполнить растягивание поддиапазона до S=4, преобразование поддиапазона до

(т.е. не выполнять) и обеспечить соответствие между исходными поддиапазонами с индексом n и целевыми поддиапазонами с индексом m, имеющее вид условия n=m.For the occasion

an example system includes a 501-4 sample rate converter that downsamples the input sample rate by a factor of two, from

before

. The goal is directly that the processing sequence consisting of the 64-band QMF analysis 502-4, the sub-band processing module 503-4, and the 64-band QMF synthesis 405 results in a physical transformation with

And

(i.e. without stretching). By identifying the above three blocks, respectively, as modules 101, 102, and 103 of FIG. 1, it is found that due to resampling

And

, formulas (1) to (3) lead to the following specifications for the subband processing module 503-4. The subband processing module 503-4 must perform subband stretching to S =4, subband transformation to

(i.e., do not execute) and match between source subranges at index n and target subranges at index m, having the condition n = m .

, то переходная характеристика многократного преобразователя будет улучшена по сравнению со случаем

.As a conclusion for the example of the scenario in FIG. 5, modules 504-2 through 503-4 all perform pure subband signal stretches and employ the single non-linear subband input block processing described in the context of FIG. 2. Control signal 104, if present, can simultaneously affect the operation of all three subband processing modules. In particular, control signal 104 can be used to simultaneously switch between long block processing and short block processing depending on the type (short-term non-periodic or non-transient) of the input signal sample. Alternatively, or in addition, if the three modules 504-2 through 504-4 use a non-zero amplitude geometric weighting parameter

, then the transient response of the multiple converter will be improved compared to the case

.

и

. Иными словами, используется только один блок 502-2 анализирующих фильтров и один блок 405 синтезирующих фильтров, что, таким образом, снижает общую вычислительную сложность многократного преобразователя.Fig. 6 illustrates an example scenario for efficient operation of a multi-order subband block transform employing a single 64-band QMF analysis filter bank. In fact, the use of three separate QMF analysis blocks and two sample rate converters of FIG. 5 results in a rather high computational complexity as well as some inconvenience in implementing frame-based processing due to the sampling rate conversion in module 501-3, i. e. due to fractional sample rate conversion. Therefore, it is proposed to replace two transformation branches, including modules 501-3→502-3→503-3 and 501-4→502-4→503-4, with modules 603-3 and 603-4, respectively, while branch 502-2→ 503-2 compared to FIG. 5 remains unchanged. All three transform orders are performed in the filter bank area with reference to FIG. 1, where

And

. In other words, only one analysis filter bank 502-2 and one synthesis filter bank 405 are used, thus reducing the overall computational complexity of the multiple transducer.

технические условия для модуля 603-3 обработки поддиапазонов, имеющие вид формул (1) - (3), заключаются в том, что модуль 603-3 обработки поддиапазонов должен выполнять растягивание поддиапазона с S=2 и преобразование поддиапазона с Q=3/2, так, чтобы соответствие между исходными поддиапазонами с индексом n и целевыми поддиапазонами с индексом m имело вид

. Для случая

технические условия для модуля 603-4 обработки поддиапазонов, имеющие вид формул (1) - (3), заключаются в том, что модуль 603-4 обработки поддиапазонов должен выполнять растягивание поддиапазона с S=2 и преобразование поддиапазона с Q=2, так, чтобы соответствие между исходными диапазонами с индексом n целевыми поддиапазонами с индексом m имело вид

.When

the specification for the subband processing module 603-3, which is in the form of formulas (1) to (3), is that the subband processing module 603-3 must perform subband stretching with S =2 and subband transformation with Q =3/2, so that the correspondence between source subranges with index n and target subranges with index m has the form

. For the occasion

the specification for the subband processing module 603-4, which is in the form of formulas (1) to (3), is that the subband processing module 603-4 must perform subband stretching with S =2 and subband transformation with Q =2, so, so that the correspondence between the source ranges with index n and the target subranges with index m has the form

.

, то переходная характеристика многократного преобразователя может быть улучшена по сравнению со случаем

.As can be seen, formula (3) does not necessarily provide an integer index n for the target subrange with index m . Therefore, for convenience, when determining the target subrange, two adjacent source subranges can be considered as described above (using formula (14)). In particular, this may be convenient for target subranges with index m for which formula (3) provides for a non-integer value of n . On the other hand, target subbands with index m for which formula (3) provides for an integer value of index n can be determined from a single source subband with index n (using formula (5)). In other words, it is proposed that a sufficiently high quality harmonic transformation can be achieved by using subband processing modules 603-3 and 603-4, which both use non-linear subband block processing with two input subbands, as described in the context of FIG. 3. In addition, control signal 104, if present, can simultaneously affect the operation of all three subband processing modules. Alternatively, or in addition, if all three modules 503-2, 603-3, 603-4 use a non-zero amplitude geometric weighting parameter

, then the transient response of the multiple transducer can be improved compared to the case

.

. Нижняя панель изображает выходной сигнал растягивания во времени тогда, когда для нелинейной обработки блоков по формуле (5) используется параметр геометрического взвешивания амплитуды

. Как видно, переходная характеристика в последнем случае значительно улучшена. В частности, видно, что обработка поддиапазонов с использованием параметра взвешивания

приводит к артефактам 701, которые существенно уменьшаются (см. ссылочную позицию 702) при обработке поддиапазонов с использованием параметра взвешивания

.Fig. 7 illustrates an example step response for time stretching by a factor of two based on a block of subbands. The top panel shows the input signal, which is the attack of the castanets sampled at 16 kHz. The system based on the design of FIG. 1 is designed with a 64-band QMF analysis filter bank 101 and a 64-band QMF synthesis filter bank 103. The subband processing module 102 is configured to implement subband stretching by a factor S =2, with no subband transformation ( Q =1), and with a direct one-to-one mapping from source subbands to target subbands. Block analysis step p =1, and block size radius R =7, i.e. block length L =15 subband samples, which corresponds to 15∙64=960 samples in the signal domain (time domain). The w window is a power cosine window, i.e. cosine raised to the second power. The middle panel of Fig. 7 depicts the time stretch output when pure phase modification is applied by module 102, i.e. for nonlinear processing of blocks according to formula (5), the weighting parameter is used

. The bottom panel depicts the time stretch output when the geometric amplitude weighting parameter is used for the non-linear processing of blocks according to formula (5).

. As can be seen, the transient response in the latter case is significantly improved. In particular, it can be seen that the processing of subranges using the weighting parameter

results in artifacts 701 that are substantially reduced (see numeral 702) when subbands are processed using the weighting parameter

.

This document describes a method and system for HFR based on harmonic transformation and/or time stretch. The method and system can be implemented with significantly reduced computational complexity compared to conventional harmonic transform-based HFR, yet provide high quality harmonic transform for both stationary and short-term non-periodic signals. The described harmonic transform-based HFR uses non-linear block-based subband processing. To adapt the non-linear subband processing to the type of signal, such as a short-term non-periodic or non-transient signal, the use of signal-dependent control data is proposed. In addition, in order to improve the transient response of the harmonic transformation using block-based non-linear processing of subbands, the use of a geometric weighting parameter is proposed. Finally, for harmonic transform-based HFR, a low complexity method and system is described that uses a single pair of analysis/synthesis filter banks for harmonic transform and HFR processing. The described methods and systems can be used in various decoding devices, such as multimedia receivers, external video/audio devices, mobile devices, audio players, video players, and so on.

The methods and systems for transformation and/or high frequency reconstruction and/or time stretching described herein may be implemented as software, firmware and/or hardware. Some components may be implemented, for example, as software running on a digital signal processor or microprocessor. Other components may be implemented, for example, as hardware or as application-specific integrated circuits. The signals encountered in the described methods and systems may be stored on media such as random access memory or optical storage media. They may be transmitted over networks such as radio networks, satellite networks, wireless networks, or wired networks such as the Internet. Typical devices using the methods and systems described herein are portable electronic devices or other consumer equipment that is used to store and/or reproduce audio signals. The methods and system can also be used in computer systems, such as Internet web servers, that store and make available for download audio signals, such as music signals.

Claims (40)

1. An audio processing device, comprising a subband processing module configured to determine a synthesis subband signal from an analyzed subband signal; wherein the signal of the analyzed subrange includes a number of complex-valued analyzed discrete values at different points in time, each of which has a phase and amplitude; wherein the signal of the analyzed subrange is associated with the frequency band of the input audio signal; wherein the subband processing module comprises a block extractor configured to re-extract a frame of L input samples from a set of complex-valued analyzed samples; wherein the frame length L is greater than one; And applying a block input step to a set of complex-valued analyzed samples, before extracting the next frame of the L input samples; thus generating a frame set of L input samples; a non-linear frame processing module configured to determine a frame of processed samples from a frame of input samples by determining for each frame sample to be processed: the phase of the processed discrete value due to the phase shift of the corresponding input discrete value; And the amplitude of the processed discrete value based on the amplitude of the corresponding input discrete value and the amplitude of the predetermined input discrete value; and an overlay and add module configured to determine a synthesis subband signal by overlaying and adding the samples from the set of frames of the processed samples; in this case, the input step of the block is equal to one discrete value, and wherein the signal of the synthesized subband is associated with the frequency band of the signal, stretched in time and / or converted in frequency relative to the input audio signal, while one or more of the block extractor, the non-linear frame processing module and the overlay and addition module are implemented at least partially, one or more hardware devices. 2. The audio processing device according to claim. 1, characterized in that the block extractor is configured to downsampling a number of complex-valued analyzed discrete values with a coefficient

subrange transformations. 3. An audio processing device according to claim 1, characterized in that the block extractor is configured to interpolate two or more complex-valued analyzed samples to extract an input sample. 4. The sound processing device according to claim 1, characterized in that the non-linear frame processing module is configured to determine the amplitude of the processed discrete value as the average value of the amplitude, the corresponding input discrete value and the amplitude of the predetermined input discrete value. 5. The audio processing device according to claim 4, characterized in that the non-linear frame processing module is configured to determine the amplitude of the processed discrete value as the geometric average of the amplitude value, the corresponding input discrete value and the amplitude of the predetermined input discrete value. 6. The sound processing device according to claim 5, characterized in that the geometric average value is defined as the amplitude of the corresponding input discrete value erected to the degree

multiplied by the amplitude of a preliminary input discrete value, erected to a degree

, while the amplitude geometric weighting parameter is

. 7. The sound processing device according to claim 6, characterized in that the geometric weighing parameter of the amplitude

is a function of the coefficient

subrange and ratio conversions

subrange stretching. 8. The sound processing device according to claim 7, characterized in that the amplitude geometric weighting parameter is expressed as

. 9. The sound processing device according to claim 1, characterized in that the non-linear frame processing module (202) is configured to determine the phase of the processed sample value by shifting the phase of the corresponding input sample value by a phase offset value that is based on a predetermined input sample value from frame of input discrete values, coefficient

conversion and coefficient

subrange stretching. 10. An audio processing device according to claim 9, characterized in that the phase offset value is based on a predetermined input sample value multiplied by

. 11. The audio processing device according to claim 10, characterized in that the phase offset value is given by a predetermined input discrete value multiplied by

, plus the parameter

phase correction. 12. Sound processing device according to claim 11, characterized in that the parameter

phase correction is determined experimentally for a number of input signals with certain acoustic properties. 13. An audio processing device according to claim 1, characterized in that the predetermined input sample value is the same for each processed frame sample value. 14. An audio processing device according to claim 1, characterized in that the predetermined input sample is the center sample of the frame of input samples. 15. The sound processing device according to claim 1, characterized in that the overlay and addition module applies the block step to the next frames of the processed discrete values, and the block step is equal to the input block step multiplied by the coefficient

subrange stretching. 16. An audio processing device according to claim 1, characterized in that the subband processing module further comprises a windowing module, which is located before the overlay and addition module, and is configured to apply a window function to the frame of processed samples. 17. Sound processing device according to claim 1, characterized in that the subband processing module is configured to determine a plurality of synthesis subband signals from the plurality of analyzed subband signals; a number of analyzed subband signals associated with a number of frequency bands of the input audio signal; And a number of signals of the synthesized subband is associated with a number of frequency bands of the signal, stretched in time and/or converted in frequency relative to the input audio signal. 18. A method performed by an audio processing device for generating a synthesis sub-band signal associated with a signal bandwidth that is time-stretched and/or frequency-converted relative to an input audio signal, the method comprising: creating an analyzed sub-band signal associated with the bandwidth of the input audio signal; wherein the signal of the analyzed subrange includes a number of complex-valued analyzed discrete values at different points in time, each of which has a phase and amplitude; extracting a frame L of input discrete values from a series of complex-valued analyzed discrete values; wherein the frame length L is greater than one; applying a block input step to a set of complex-valued analyzed samples before extracting the next frame of L input samples; thus generating a frame set of input samples; defining a frame of processed discrete values from a frame of input discrete values by defining for each processed discrete value of the frame: the phase of the processed discrete value due to the phase shift of the corresponding input discrete value; And the amplitude of the processed discrete value based on the amplitude of the corresponding input discrete value and the amplitude of the predetermined input discrete value; And determining the signal of the synthesized subband by superimposing and adding the discrete values of the set of frames of the processed discrete values; in this case, the input step of the block is equal to one discrete value, and wherein one or more of generating an analysis subband signal, extracting a frame, applying an input block step, determining a frame of processed samples, and determining a synthesized subband signal is implemented at least in part by one or more hardware devices. 19. A non-volatile storage medium containing a program implemented in software adapted for execution on a processor and for performing the steps of the method according to claim 18 when implemented on an audio processing device.

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