KR102230308B1 - Method and apparatus for processing multimedia signals - Google Patents

Abstract

The present invention relates to a signal processing method and apparatus for effectively rendering a multimedia signal, and more particularly, to a multimedia signal processing method and apparatus for implementing a multimedia signal rendering with a low computational amount while minimizing quality loss.
To this end, the present invention is a multimedia signal processing apparatus, comprising a rendering unit for receiving a multimedia signal and filtering a plurality of subband signals of the multimedia signal, wherein the rendering unit corresponds to each subband signal of the multimedia signal Obtaining a set of subband filter coefficient(s) to perform, and filtering a corresponding subband signal using the obtained set of subband filter coefficient(s), the subband filter coefficient(s) of each subband The set provides a multimedia signal processing apparatus having a variable length in a frequency domain and a multimedia signal processing method using the same.

Description

Multimedia signal processing method and apparatus {METHOD AND APPARATUS FOR PROCESSING MULTIMEDIA SIGNALS}

The present invention relates to a signal processing method and apparatus for effectively rendering a multimedia signal, and more particularly, to a multimedia signal processing method and apparatus for implementing a multimedia signal rendering with a low computational amount while minimizing quality loss.

Binaural rendering for listening to a multichannel signal in stereo has a problem that requires a large amount of computation as the length of the target filter increases. In particular, when using a BRIR (Binaural Room Impulse Response) filter that reflects the characteristics of a recording room, its length can range from 48,000 to 96,000 samples. Here, as in the 22.2 channel format, if the number of input channels increases, the amount of computation is enormous.

, 해당 채널의 좌, 우 BRIR 필터를 각각

,

, 출력 신호를

,

이라고 하면, 바이노럴 필터링(binaural filtering)은 다음과 같은 식으로 표현할 수 있다.input signal of the i-th channel

, Set the left and right BRIR filters of the corresponding channel, respectively.

,

, The output signal

,

If, binaural filtering can be expressed in the following equation.

이며, *는 콘볼루션(convolution)을 의미한다. 위의 시간-도메인 콘볼루션은 일반적으로 고속 퓨리에 변환(Fast Fourier Transform, FFT)에 기반한 고속 콘볼루션(fast convolution)을 이용하여 수행된다. 고속 콘볼루션을 이용하여 바이노럴 렌더링을 수행하는 경우, 입력 채널수에 해당하는 횟수의 FFT와 출력 채널수에 해당하는 횟수의 역 고속 퓨리에 변환(Inverse FFT)을 수행해야 한다. 게다가 멀티채널 오디오 코덱과 같이 실시간 재생 환경에서의 경우 딜레이를 고려해야 하기 때문에 블록 단위(block-wise)의 고속 콘볼루션을 수행해야 하며, 이는 전체 길이에 대하여 단순히 고속 콘볼루션을 수행했을 때보다 더 많은 연산량을 소모할 수 있다.From here

And * means convolution. The above time-domain convolution is generally performed using fast convolution based on Fast Fourier Transform (FFT). When binaural rendering is performed using fast convolution, the FFT of the number of input channels and the inverse fast Fourier transform of the number of output channels must be performed. In addition, in the case of a real-time playback environment such as a multi-channel audio codec, since delay must be considered, block-wise high-speed convolution must be performed, which is more than simply performing high-speed convolution for the entire length. It can consume the amount of computation.

However, most coding schemes are performed in the frequency domain, and in some coding schemes (eg, HE-AAC, USAC, etc.), the last step of the decoding process is performed in the QMF domain. Therefore, as shown in Equation 1 above, when binaural filtering is performed in the time domain, it is very inefficient because an operation for QMF synthesis as many as the number of channels is additionally required. Therefore, there is an advantage when binaural rendering is performed directly in the QMF domain.

In the present invention, when reproducing a multi-channel or multi-object signal in stereo, a filtering process that requires a large amount of computation in binaural rendering to preserve a three-dimensional effect like the original signal is implemented with a very low computational amount while minimizing sound quality loss. Has a purpose for it.

In addition, an object of the present invention is to minimize the spread of distortion through a high-quality filter when there is distortion in the input signal itself.

In addition, the present invention has an object of implementing a Finite Impulse Response (FIR) filter having a very long length as a filter having a smaller length.

In addition, the present invention has an object of minimizing distortion of a portion damaged by missing filter coefficients when performing filtering using the abbreviated FIR filter.

In order to solve the above problems, the present invention provides an audio signal processing method and an audio signal processing apparatus as follows.

First, the present invention includes the steps of receiving a multi-audio signal including a multi-channel or multi-object signal; Receiving truncated subband filter coefficients for filtering the multi-audio signal, wherein the truncated subband filter coefficient is obtained from Binaural Room Impulse Response (BRIR) filter coefficients for binaural filtering of the multi-audio signal. It is at least a part of the subband filter coefficient, and the length of the truncated subband filter coefficient is determined based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficient, and at least one The length of the truncated subband filter coefficients is different from the length of the truncated subband filter coefficients of other subbands; And filtering the subband signal using the truncated subband filter coefficients corresponding to each subband signal of the multi-audio signal. It provides an audio signal processing method comprising a.

In addition, as an audio signal processing apparatus for performing binaural rendering on a multi-audio signal including a multi-channel or multi-object signal, the multi-audio signal includes a plurality of sub-band signals, and each sub-band signal A high-speed convolution unit for rendering the direct sound and the initial reflection sound part of the signal; And a late reverberation generator for performing rendering of a late reverberation part for each of the subband signals, wherein the high-speed convolution unit receives truncated subband filter coefficients for filtering the multi-audio signal, wherein the truncation The subband filter coefficient obtained is at least a part of the subband filter coefficient obtained from the Binaural Room Impulse Response (BRIR) filter coefficient for binaural filtering of the multi-audio signal, and the length of the truncated subband filter coefficient corresponds to It is determined based on filter order information obtained by at least partially using the characteristic information extracted from the subband filter coefficients, and the length of at least one of the truncated subband filter coefficients is of the truncated subband filter coefficients of the other subbands. Different from the length, it provides an audio signal processing apparatus, characterized in that filtering the subband signal using the truncated subband filter coefficient corresponding to each subband signal of the multi-audio signal.

In this case, the characteristic information includes first reverberation time information of a corresponding subband filter coefficient, and the filter order information has one value for each subband.

In addition, the length of the truncated subband filter is characterized in that it has a multiple value of a power of 2.

According to an embodiment of the present invention, the plurality of subband filter coefficients and the plurality of subband signals each include a first subband group having a low frequency and a second subband group having a high frequency based on a preset frequency band, , Wherein the filtering is performed on the truncated subband filter coefficients and subband signals of the first subband group.

In addition, according to an embodiment of the present invention, the filtering is performed using a front subband filter coefficient truncated based at least in part on first reverberation time information of a corresponding subband filter coefficient. And a reverberation processing step of the subband signal corresponding to a period after the front subband filter coefficient.

In this case, the reverberation processing step includes receiving a downmix subband filter coefficient for each subband, and the downmix subband filter coefficient combines rear subband filter coefficients for each channel or each object of a corresponding subband. And the rear subband filter coefficient is obtained from a section after the front subband filter coefficient among corresponding subband filter coefficients; Generating a downmix subband signal for each subband, wherein the downmix subband signal is generated by downmixing subband signals for each channel or each object of a corresponding subband; And generating two-channel left and right subband reverberation signals by using the downmix subband signal and the downmix subband filter coefficient corresponding thereto.

According to an embodiment of the present invention, the downmix subband signal is a mono subband signal, the downmix subband filter coefficient reflects an energy reduction characteristic of a reverberant for a corresponding subband signal, and the filtered mono subband Generating a decorrelation signal for the band signal; And generating two-channel left and right signals by adding weights between the filtered mono subband signal and the decorrelation signal.

According to another embodiment of the present invention, receiving a multi-audio signal including a multi-channel or multi-object signal, each of the multi-audio signals includes a plurality of subband signals, and the plurality of subband signals are preset. Including a signal of a first subband group having a low frequency and a signal of a second subband group having a high frequency based on the frequency band; Receiving at least one parameter corresponding to each subband signal of the second subband group, wherein the at least one parameter is a Binaural Room Impulse Response (BRIR) corresponding to each subband signal of the second subband group Extracted from subband filter coefficients; Performing tap-delay line filtering on the subband signals of the second subband group using the received parameters; It provides an audio signal processing method comprising a.

In addition, as an audio signal processing apparatus for performing binaural rendering on a multi-audio signal including a multi-channel or multi-object signal, the multi-audio signal each includes a plurality of subband signals, and the plurality of subbands The signals include a signal of a first subband group having a low frequency and a signal of a second subband group having a high frequency based on a preset frequency band, and performing rendering of each subband signal of the first subband group. A high-speed convolution unit for; And a tap-delay line processing unit for performing rendering of each subband signal of the second subband group, wherein the tap-delay line processing unit comprises: Receiving at least one parameter, wherein the at least one parameter is extracted from a Binaural Room Impulse Response (BRIR) subband filter coefficient corresponding to each subband signal of the second subband group, and using the received parameter It provides an audio signal processing apparatus, characterized in that tap-delay line filtering is performed on the subband signals of the second subband group.

In this case, the parameter is characterized in that it includes one delay information for a corresponding BRIR subband filter coefficient and one gain information corresponding to the delay information.

According to an embodiment of the present invention, the tap-delay line filtering is characterized in that the one-tap-delay line filtering using the parameter.

In addition, the delay information is characterized in that it indicates position information on a maximum peak among the BRIR subband filter coefficients.

In addition, the delay information is characterized by having an integer value in units of samples in the QMF domain.

In addition, the gain information is characterized by having a complex value.

According to an embodiment of the present invention, the step of summing the filtered multi-audio signal into two channel left and right subband signals for each subband; Combining the summed left and right subband signals with left and right subband signals generated from multi-audio signals of the first subband group; And QMF synthesizing the combined left and right subband signals, respectively. It characterized in that it further comprises.

According to another embodiment of the present invention, there is provided a method comprising: receiving a multimedia signal having a plurality of subbands; Receiving at least one prototype rendering filter coefficient for filtering each subband signal of the multimedia signal; Converting the circular rendering filter coefficients into a plurality of subband filter coefficients; Truncating each of the subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, wherein at least one of the truncated subband filter coefficients has a different length Different from the length of the truncated subband filter coefficients of the band; And filtering the multimedia signal using the truncated subband filter coefficients corresponding to each of the subband signals. It provides a multimedia signal processing method comprising a.

In addition, as a multimedia signal processing apparatus having a plurality of subbands, receiving at least one prototype (proto-type) rendering filter coefficients for filtering each subband signal of the multimedia signal, and receiving the plurality of circular rendering filter coefficients Transforming into subband filter coefficients, and truncating each subband filter coefficient based on filter order information obtained by at least partially using characteristic information extracted from a corresponding subband filter coefficient, wherein at least one of the truncated sub A parameterization unit, wherein the length of the band filter coefficient is different from the length of the truncated subband filter coefficient of the other subband; And a rendering unit that receives the multimedia signal and filters the multimedia signal by using the truncated subband filter coefficients corresponding to each of the subband signals.

In this case, the multimedia signal includes a multi-channel or multi-object signal, and the circular rendering filter coefficient is a BRIR filter coefficient in a time domain.

In addition, the characteristic information includes energy decay time information of a corresponding subband filter coefficient, and the filter order information has one value for each subband.

According to another embodiment of the present invention, receiving a multi-audio signal including a multi-channel or multi-object signal, wherein each of the multi-audio signals includes a plurality of subband signals, and the plurality of subband signals are Including a signal of a first subband group having a low frequency and a signal of a second subband group having a high frequency based on the set frequency band; Receiving truncated subband filter coefficients for filtering the multi-audio signal of the first subband group, wherein the truncated subband filter coefficient is a Binaural Room Impulse (BRIR) for binaural filtering of the multi-audio signal. Response) It is at least a part of the subband filter coefficient of the first subband group obtained from the filter coefficient, and the length of the truncated subband filter coefficient is obtained by at least partially using the characteristic information extracted from the corresponding subband filter coefficient. Determined based on the obtained filter order information; And filtering the subband signals of the first subband group using the truncated subband filter coefficients. Receiving at least one parameter corresponding to each subband signal of the second subband group, the at least one parameter being extracted from subband filter coefficients corresponding to each subband signal of the second subband group ; And performing tap-delay line filtering on the subband signals of the second subband group using the received parameters. It provides an audio signal processing method comprising a.

In addition, as an audio signal processing apparatus for performing binaural rendering on a multi-audio signal including a multi-channel or multi-object signal, the multi-audio signal each includes a plurality of subband signals, and the plurality of subbands The signals include a signal of a first subband group having a low frequency and a signal of a second subband group having a high frequency based on a preset frequency band, and performing rendering of each subband signal of the first subband group. A high-speed convolution unit for; And a tap-delay line processing unit for performing rendering of each subband signal of the second subband group, wherein the high-speed convolution unit comprises truncation for filtering the multi-audio signal of the first subband group. Received subband filter coefficients, wherein the truncated subband filter coefficients are subband filter coefficients of a first subband group obtained from Binaural Room Impulse Response (BRIR) filter coefficients for binaural filtering of the multi-audio signal And the length of the truncated subband filter coefficient is determined based on filter order information obtained at least partially using characteristic information extracted from the corresponding subband filter coefficient, and the truncated subband filter coefficient The subband signals of the first subband group are filtered using, and the tap-delay line processing unit receives at least one parameter corresponding to each subband signal of the second subband group, wherein the at least one The parameter of is extracted from the Binaural Room Impulse Response (BRIR) subband filter coefficient corresponding to each subband signal of the second subband group, and is applied to the subband signal of the second subband group using the received parameter. An apparatus for processing an audio signal, characterized in that tap-delay line filtering is performed.

At this time, the left and right subband signals of the two channels generated by filtering the subband signals of the first subband group and the left of the two channels generated by tap-delay line filtering the subband signals of the second subband group. Combining the right subband signals; And QMF synthesizing the combined left and right subband signals, respectively. It characterized in that it further comprises.

According to an embodiment of the present invention, it is possible to significantly reduce the amount of computation while minimizing sound quality loss when performing binaural rendering for a multi-channel or multi-object signal.

According to an embodiment of the present invention, it is possible to perform high-quality binaural rendering of a multi-channel or multi-object audio signal, which was not possible in real-time processing in a conventional low-power device.

1 is a block diagram showing an audio signal decoder according to an embodiment of the present invention.
2 is a block diagram showing each configuration of a binaural renderer according to an embodiment of the present invention.
3 to 7 are diagrams showing various embodiments of an audio signal processing apparatus according to the present invention.
8 to 10 are diagrams illustrating a method of generating an FIR filter for binaural rendering according to an embodiment of the present invention.
11 to 14 are views showing various embodiments of the P-part rendering unit of the present invention.
15 and 16 are diagrams showing various embodiments of the QTDL processing of the present invention.

The terms used in the present specification have been selected as currently widely used general terms as possible while considering functions in the present invention, but this may vary according to the intention, custom, or the emergence of new technologies of technicians in the field. In addition, in certain cases, there are terms arbitrarily selected by the applicant, and in this case, the meaning will be described in the description of the corresponding invention. Therefore, it should be noted that the terms used in the present specification should be interpreted based on the actual meaning of the term and the entire contents of the present specification, not a simple name of the term.

1 is a block diagram showing an audio signal decoder according to an embodiment of the present invention. The audio signal decoder of the present invention includes a core decoder 10, a rendering unit 20, a mixer 30, and a post processing unit 40.

First, the core decoder 10 decodes a loudspeaker channel signal, a discrete object signal, an object downmix signal, and a pre-rendered signal. According to an embodiment, in the core decoder 10, a codec based on Unified Speech and Audio Coding (USAC) may be used. The core decoder 10 decodes the received bitstream and delivers it to the rendering unit 20.

The rendering unit 20 renders the signal decoded by the core decoder 10 using reproduction layout information. The rendering unit 20 may include a format converter 22, an object renderer 24, an OAM decoder 25, an SAOC decoder 26, and an HOA decoder 28. The rendering unit 20 performs rendering using any one of the above configurations according to the type of the decoded signal.

The format converter 22 converts the transmitted channel signal into an output speaker channel signal. That is, the format converter 22 converts between the transmitted channel configuration and the speaker channel configuration to be reproduced. If the number of output speaker channels (e.g., 5.1 channels) is less than the number of transmitted channels (e.g., 22.2 channels) or the transmitted channel configuration and the channel configuration to be reproduced are different, the format converter 22 Downmix the signal. The audio signal decoder of the present invention may generate an optimal downmix matrix using a combination of an input channel signal and an output speaker channel signal, and perform downmixing using the matrix. According to an embodiment of the present invention, a channel signal processed by the format converter 22 may include a pre-rendered object signal. According to an embodiment, before encoding an audio signal, at least one object signal may be pre-rendered and mixed into a channel signal. The object signal mixed in this way may be converted into an output speaker channel signal by the format converter 22 together with the channel signal.

The object renderer 24 and the SAOC decoder 26 perform rendering on an object-based audio signal. Object-based audio signals may include individual object waveforms and parametric object waveforms. In the case of individual object waveforms, each object signal is provided to an encoder in a monophonic waveform, and the encoder transmits each object signal using single channel elements (SCEs). In the case of a parametric object waveform, a plurality of object signals are downmixed to at least one channel signal, and characteristics of each object and a relationship between them are expressed by a spatial audio object coding (SAOC) parameter. The object signals are downmixed and encoded with the core codec, and parametric information generated at this time is transmitted to the decoder together.

Meanwhile, when an individual object waveform or a parametric object waveform is transmitted to the audio signal decoder, compressed object metadata corresponding thereto may be transmitted together. Object metadata quantizes object properties in units of time and space to designate a position and a gain value of each object in a three-dimensional space. The OAM decoder 25 of the rendering unit 20 receives the compressed object metadata, decodes it, and transmits it to the object renderer 24 and/or the SAOC decoder 26.

The object renderer 24 renders each object signal according to a given reproduction format using object metadata. In this case, each object signal may be rendered to specific output channels based on object metadata. The SAOC decoder 26 restores an object/channel signal from the decoded SAOC transport channels and parametric information. The SAOC decoder 26 may generate an output audio signal based on reproduction layout information and object metadata. In this way, the object renderer 24 and the SAOC decoder 26 may render the object signal as a channel signal.

The HOA decoder 28 receives a Higher Order Ambisonics (HOA) signal and HOA side information, and decodes it. The HOA decoder 28 generates a sound scene by modeling a channel signal or an object signal using a separate equation. When a location in the space where the speaker is located in the generated sound scene is selected, rendering can be performed with a speaker channel signal.

Meanwhile, although not shown in FIG. 1, when an audio signal is transmitted to each component of the rendering unit 20, dynamic range control (DRC) may be performed as a preprocessing process. DRC limits the dynamic range of the reproduced audio signal to a certain level. Sounds that are less than a preset threshold are adjusted to be louder, and sounds that are louder than a preset threshold are adjusted to be smaller.

The channel-based audio signal and object-based audio signal processed by the rendering unit 20 are transmitted to the mixer 30. The mixer 30 adjusts the delay between the channel-based waveform and the rendered object waveform, and sums them in units of samples. The audio signal summed by the mixer 30 is delivered to the post processing unit 40.

The post processing unit 40 includes a speaker renderer 100 and a binaural renderer 200. The speaker renderer 100 performs post processing for outputting a multi-channel and/or multi-object audio signal transmitted from the mixer 30. Such post processing may include dynamic range control (DRC), loudness normalization (LN), and peak limiter (PL).

The binaural renderer 200 generates a binaural downmix signal of a multi-channel and/or multi-object audio signal. The binaural downmix signal is a two-channel audio signal that allows each input channel/object signal to be represented by a virtual sound source located in three dimensions. The binaural renderer 200 may receive an audio signal supplied to the speaker renderer 100 as an input signal. Binaural rendering is performed based on a Binaural Room Impulse Response (BRIR) filter, and may be performed in a time domain or a QMF domain. According to an embodiment, as a post-processing process of binaural rendering, the above-described dynamic range control (DRC), volume normalization (LN), peak limiting (PL), and the like may be additionally performed.

2 is a block diagram showing each configuration of a binaural renderer according to an embodiment of the present invention. As shown, the binaural renderer 200 according to an embodiment of the present invention includes a BRIR parameterization unit 210, a high-speed convolution unit 230, a late reverberation generation unit 240, a QTDL processing unit 250, It may include a mixer & combiner 260.

The binaural renderer 200 generates a 3D audio headphone signal (ie, a 3D audio 2-channel signal) by performing binaural rendering on various types of input signals. In this case, the input signal may be an audio signal including at least one of a channel signal (ie, a speaker channel signal), an object signal, and an HOA signal. According to another embodiment of the present invention, when the binaural renderer 200 includes a separate decoder, the input signal may be an encoded bitstream of the audio signal described above. The binaural rendering converts the decoded input signal into a binaural downmix signal so that you can experience a surround sound when listening with headphones.

, 서브밴드 도메인에서의 시간 인덱스를 l이라고 하면, QMF 도메인에서의 바이노럴 렌더링은 다음과 같은 식으로 표현할 수 있다.According to an embodiment of the present invention, the binaural renderer 200 may perform binaural rendering of an input signal on a QMF domain. For example, the binaural renderer 200 may receive a multi-channel (N channels) signal in the QMF domain and perform binaural rendering on the multi-channel signal using a BRIR subband filter in the QMF domain. The k-th subband signal of the i-th channel that has passed through the QMF analysis filter bank

, Assuming that the time index in the subband domain is l, the binaural rendering in the QMF domain can be expressed as follows.

이며,

은 시간 도메인 BRIR 필터를 QMF 도메인의 서브밴드 필터로 변환한 것이다.here,

Is,

Is transformed from the time domain BRIR filter into a subband filter in the QMF domain.

That is, the binaural rendering may be performed by dividing a channel signal or an object signal in the QMF domain into a plurality of subband signals, convolving each subband signal with a corresponding BRIR subband filter, and then adding them.

The BRIR parameterization unit 210 transforms and edits BRIR filter coefficients for binaural rendering in the QMF domain, and generates various parameters. First, the BRIR parameterization unit 210 receives time domain BRIR filter coefficients for multi-channels or multi-objects, and converts them into QMF domain BRIR filter coefficients. In this case, the QMF domain BRIR filter coefficient includes a plurality of subband filter coefficients respectively corresponding to a plurality of frequency bands. In the present invention, the subband filter coefficient indicates each BRIR filter coefficient of the QMF-transformed subband domain. In this specification, the subband filter coefficient may also be referred to as a BRIR subband filter coefficient. The BRIR parameterization unit 210 may edit a plurality of BRIR subband filter coefficients of the QMF domain, respectively, and transmit the edited subband filter coefficients to the high-speed convolution unit 230 or the like. According to an embodiment of the present invention, the BRIR parameterization unit 210 may be included as a component of the binaural renderer 200 or may be provided as a separate device. According to an embodiment, a configuration including a high-speed convolution unit 230, a late reverberation generation unit 240, a QTDL processing unit 250, and a mixer & combiner 260 excluding the BRIR parameterization unit 210 It may be classified as a binaural rendering unit 220.

According to an embodiment, the BRIR parameterization unit 210 may receive a BRIR filter coefficient corresponding to at least one location in the virtual reproduction space as an input. Each position of the virtual reproduction space may correspond to a position of each speaker of the multi-channel system. According to an embodiment, each BRIR filter coefficient received by the BRIR parameterization unit 210 may be directly matched to each channel or each object of an input signal of the binaural renderer 200. On the other hand, according to another embodiment of the present invention, each of the received BRIR filter coefficients may have a configuration independent of the input signal of the binaural renderer 200. That is, at least some of the BRIR filter coefficients received by the BRIR parameterization unit 210 may not be directly matched to the input signal of the binaural renderer 200, and the number of received BRIR filter coefficients is the channel and/or of the input signal. Alternatively, it may be smaller or larger than the total number of objects.

According to an embodiment of the present invention, the BRIR parameterization unit 210 transforms and edits the BRIR filter coefficients corresponding to each channel or object of an input signal of the binaural renderer 200, and the binaural rendering unit 220 Can be passed on. The corresponding BRIR filter coefficient may be a matching BRIR or a fallback BRIR for each channel or object. BRIR matching may be determined according to whether there is a BRIR filter coefficient targeting the position of each channel or each object in the virtual reproduction space. If there is a BRIR filter coefficient targeting at least one of each channel of an input signal or a location of each object, the corresponding BRIR filter coefficient may be a matching BRIR of the input signal. However, if there is no BRIR filter coefficient targeting the location of a specific channel or object, the binaural rendering unit 220 applies the BRIR filter coefficient targeting the location most similar to the corresponding channel or object to the corresponding channel or object. It can be provided as a fallback for BRIR.

Meanwhile, according to another embodiment of the present invention, the BRIR parameterization unit 210 may transform and edit all of the received BRIR filter coefficients and transfer them to the binaural rendering unit 220. In this case, the selection of BRIR filter coefficients (or edited BRIR filter coefficients) corresponding to each channel or object of the input signal may be performed by the binaural rendering unit 220.

The binaural rendering unit 220 includes a high-speed convolution unit 230, a late reverberation generation unit 240 and a QTDL processing unit 250, and includes a multi-channel and/or multi-object signal. Receive. In this specification, an input signal including a multi-channel and/or multi-object signal will be referred to as a multi-audio signal. In FIG. 2, the binaural rendering unit 220 is shown to receive a multi-channel signal in the QMF domain according to an embodiment, but the input signal of the binaural rendering unit 220 includes a time domain multi-channel signal and a multi-channel signal. Object signals and the like may be included. In addition, when the binaural rendering unit 220 additionally includes a separate decoder, the input signal may be an encoded bitstream of the multi-audio signal. In addition, in the present specification, the present invention will be described based on a case of performing BRIR rendering on a multi-audio signal, but the present invention is not limited thereto. That is, the features provided by the present invention may be applied to other types of rendering filters other than BRIR, and may also be applied to an audio signal of a single channel or a single object rather than a multi-audio signal.

The high-speed convolution unit 230 processes direct sound and early reflection on the input signal by performing high-speed convolution between the input signal and the BRIR filter. To this end, the high-speed convolution unit 230 may perform high-speed convolution using a truncated BRIR. The truncated BRIR includes a plurality of subband filter coefficients that are cut dependently on each subband frequency, and is generated by the BRIR parameterization unit 210. In this case, the length of each truncated subband filter coefficient is determined dependently on the frequency of the corresponding subband. The fast convolution unit 230 may perform variable order filtering in the frequency domain by using truncated subband filter coefficients having different lengths according to the subbands. That is, for each frequency band, high-speed convolution may be performed between the QMF domain subband audio signal and the truncated subband filters of the QMF domain corresponding thereto. In the present specification, a direct sound & early reflection (D&E) part may be referred to as an F (front)-part.

The late reverberation generator 240 generates a late reverberation signal for the input signal. The late reverberation signal represents the direct sound generated by the high-speed convolution unit 230 and the output signal after the initial reflection sound. The late reverberation generator 240 may process an input signal based on reverberation time information determined from each subband filter coefficient transmitted from the BRIR parameterization unit 210. According to an embodiment of the present invention, the late reverberation generator 240 may generate a mono or stereo downmix signal for an input audio signal and perform a late reverberation process on the generated downmix signal. In this specification, the Late Reverberation (LR) part may be referred to as a P (parametric)-part.

The QMF domain tapped delay line (QTDL) processing unit 250 processes a signal of a high frequency band among input audio signals. The QTDL processing unit 250 receives at least one parameter corresponding to each subband signal of a high frequency band from the BRIR parameterization unit 210, and performs tap-delay line filtering in the QMF domain using the received parameter. . According to an embodiment of the present invention, the binaural renderer 200 separates the input audio signal into a low frequency band signal and a high frequency band signal based on a preset constant or a preset frequency band, and the low frequency band signal is high-speed. In the convolution unit 230 and the late reverberation generation unit 240, the high frequency band signal may be processed by the QTDL processing unit 250, respectively.

The high-speed convolution unit 230, the late reverberation generation unit 240, and the QTDL processing unit 250 respectively output two-channel QMF domain subband signals. The mixer & combiner 260 combines the output signal of the high-speed convolution unit 230, the output signal of the late reverberation generation unit 240, and the output signal of the QTDL processing unit 250 to perform mixing. In this case, the combination of the output signals is performed separately for the left and right output signals of the two channels. The binaural renderer 200 QMF synthesizes the combined output signal to generate a final output audio signal in the time domain.

Hereinafter, various embodiments of the high-speed convolution unit 230, the late reverberation generation unit 240, the QTDL processing unit 250, and combinations thereof of FIG. 2 will be described in detail with reference to the respective drawings.

3 to 7 illustrate various embodiments of an audio signal processing apparatus according to the present invention. In the present invention, the audio signal processing apparatus may refer to the binaural renderer 200 or the binaural rendering unit 220 shown in FIG. 2 in a narrow sense. However, in the present invention, the audio signal processing apparatus may refer to the audio signal decoder of FIG. 1 including a binaural renderer in a broad sense. Each binaural renderer illustrated in FIGS. 3 to 7 may represent only a partial configuration of the binaural renderer 200 illustrated in FIG. 2 for convenience of description. In addition, in the present specification, an embodiment of a multi-channel input signal may be mainly described, but unless otherwise noted, the channel, multi-channel, and multi-channel input signals each include an object, a multi-object, and a multi-object input signal. Can be used as a concept. In addition, the multi-channel input signal may be used as a concept including a HOA decoded and rendered signal.

3 shows a binaural renderer 200A according to an embodiment of the present invention. When the binaural rendering using BRIR is generalized, it is M-to-O processing to obtain O output signals for a multichannel input signal having M channels. Binaural filtering can be viewed as filtering using filter coefficients corresponding to each input channel and output channel in this process. In FIG. 3, the original filter set H refers to transfer functions from the speaker position of each channel signal to the positions of the left and right ears. Among these transfer functions, measurements measured in a general listening room, that is, a space with reverberation, is called Binaural Room Impulse Response (BRIR). On the other hand, what was measured in the anechoic chamber so that there is no effect of the regeneration space is called Head Related Impulse Response (HRIR), and the transfer function for this is called Head Related Transfer Function (HRTF). Therefore, unlike HRTF, BRIR contains not only direction information but also information of a reproduction space. According to an embodiment, the BRIR may be replaced by using an HRTF and an artificial reverberator. In the present specification, binaural rendering using BRIR is described, but the present invention is not limited thereto, and the same can be applied to binaural rendering using various types of FIR filters. Meanwhile, the BRIR may have a sample length of 96K as described above, and multi-channel binaural rendering is performed using M*O different filters, thus requiring a high computational processing process.

According to an embodiment of the present invention, the BRIR parameterization unit 210 may generate modified filter coefficients from the original filter set H to optimize the amount of computation. The BRIR parameterization unit 210 separates the original filter coefficient into an F (front)-part coefficient and a P (parametric)-part coefficient. Here, the F-part represents the direct sound and early reflection (D&E) parts, and the P-part represents the late reverberation (LR) part. For example, an original filter coefficient having a 96K sample length may be divided into an F-part cut only up to the previous 4K sample and a P-part corresponding to the remaining 92K samples.

The binaural rendering unit 220 receives F-part coefficients and P-part coefficients from the BRIR parameterization unit 210, respectively, and renders a multi-channel input signal by using them. According to an embodiment of the present invention, the high-speed convolution unit 230 shown in FIG. 2 renders a multi-audio signal using the F-part coefficient received from the BRIR parameterization unit 210, and a late reverberation generator 240 ) May render a multi-audio signal using the P-part coefficient received from the BRIR parameterization unit 210. That is, the high-speed convolution unit 230 and the late reverberation generation unit 240 may correspond to the F-part rendering unit and the P-part rendering unit of the present invention, respectively. According to an embodiment, F-part rendering (binaural rendering using F-part coefficients) is implemented with a conventional Finite Impulse Response (FIR) filter, and P-part rendering (binaural rendering using P-part coefficients) Rendering) can be implemented in a parametric way. Meanwhile, the complexity-quality control input provided by the user or the control system may be used to determine information generated by the F-part and/or the P-part.

4 is a binaural renderer 200B according to another embodiment of the present invention, and illustrates a more detailed method of implementing F-part rendering. For convenience of explanation, the P-part rendering unit is omitted in FIG. 4. In addition, although FIG. 4 shows a filter implemented in the QMF domain, the present invention is not limited thereto and can be applied to all subband processing in other domains.

Referring to FIG. 4, F-part rendering may be performed by the high-speed convolution unit 230 on the QMF domain. For rendering in the QMF domain, the QMF analysis unit 222 includes time domain input signals x0, x1, ... x_M-1 to QMF domain signals X0, X1, ... Convert to X_M-1. At this time, the input signals x0, x1, ... x_M-1 may be a multi-channel audio signal, for example, a channel signal corresponding to a 22.2 channel speaker. A total of 64 subbands may be used for the QMF domain, but the present invention is not limited thereto. Meanwhile, according to an embodiment of the present invention, the QMF analysis unit 222 may be omitted from the binaural renderer 200B. In the case of HE-AAC or USAC using SBR (Spectral Band Replication), processing is performed in the QMF domain, so the binaural renderer 200B directly performs QMF domain signals X0, X1,… without QMF analysis. X_M-1 can be received as an input. Accordingly, when a QMF domain signal is received as a direct input as described above, the QMF used by the binaural renderer according to the present invention is characterized in that it is the same as the QMF used by the previous processing unit (for example, SBR). The QMF synthesizing unit 244 QMF synthesizes the two-channel left and right signals Y_L and Y_R on which binaural rendering has been performed, and generates two-channel output audio signals yL and yR in the time domain.

5 to 7 illustrate embodiments of binaural renderers 200C, 200D, and 200E respectively performing F-part rendering and P-part rendering. In the embodiments of FIGS. 5 to 7, the F-part rendering is performed by the high-speed convolution unit 230 on the QMF domain, and the P-part rendering is performed by the late reverberation generation unit 240 on the QMF domain or time domain. do. In the embodiments of FIGS. 5 to 7, detailed descriptions of parts overlapping with the embodiments of the previous drawings will be omitted.

Referring to FIG. 5, the binaural renderer 200C may perform both F-part rendering and P-part rendering in the QMF domain. That is, the QMF analysis unit 222 of the binaural renderer 200C is the time domain input signal x0, x1, ... x_M-1 to QMF domain signals X0, X1, ... It is converted into X_M-1 and transmitted to the high-speed convolution unit 230 and the late reverberation generation unit 240, respectively. The high-speed convolution unit 230 and the late reverberation generation unit 240 are QMF domain signals X0, X1, ... By rendering X_M-1, output signals Y_L, Y_R, Y_Lp, and Y_Rp of two channels are generated, respectively. In this case, the high-speed convolution unit 230 and the late reverberation generation unit 240 may perform rendering using the F-part filter coefficients and the P-part filter coefficients respectively received from the BRIR parameterization unit 210. The output signals Y_L and Y_R of the F-part rendering and the output signals Y_Lp and Y_Rp of the P-part rendering are combined for each of the left and right channels in the mixer & combiner 260 and transmitted to the QMF synthesis unit 224. The QMF synthesizing unit 224 QMF synthesizes the input two-channel left and right signals to generate two-channel output audio signals yL and yR in the time domain.

Referring to FIG. 6, the binaural renderer 200D may perform F-part rendering in the QMF domain and P-part rendering in the time domain, respectively. The QMF analysis unit 222 of the binaural renderer 200D converts the time domain input signal to QMF and transmits it to the high-speed convolution unit 230. The high-speed convolution unit 230 generates two-channel output signals Y_L and Y_R by F-part rendering the QMF domain signal. The QMF synthesis unit 224 converts the output signal of F-part rendering into a time domain output signal and transmits it to the mixer & combiner 260. Meanwhile, the late reverberation generation unit 240 directly receives the time domain input signal and performs P-part rendering. The output signals yLp and yRp of the P-part rendering are transmitted to the mixer & combiner 260. The mixer & combiner 260 generates two-channel output audio signals yL and yR in the time domain by combining the F-part rendering output signal and the P-part rendering output signal in the time domain, respectively.

In the embodiments of FIGS. 5 and 6, F-part rendering and P-part rendering are performed in parallel, respectively, whereas according to the embodiment of FIG. 7, the binaural renderer 200E includes F-part rendering and P-part rendering can be performed sequentially, respectively. That is, the high-speed convolution unit 230 F-parts rendering the QMF-converted input signal, and the F-part-rendered two-channel signals Y_L and Y_R are converted into time domain signals by the QMF synthesis unit 224 and then reverberate later. It may be transmitted to the generation unit 240. The late reverberation generator 240 generates two-channel output audio signals yL and yR in the time domain by performing P-part rendering on the input two-channel signal.

5 to 7 illustrate an embodiment of performing F-part rendering and P-part rendering, respectively, and binaural rendering may be performed by combining or modifying the embodiments of each drawing. For example, in each embodiment, the binaural renderer may individually perform P-part rendering for each of the input multi-audio signals, but downmixes the input signal into a 2-channel left/right signal or a mono signal and then down. It is also possible to perform P-part rendering on the mixed signal.

<Variable Order Filtering in Frequency-domain, VOFF>

8 to 10 illustrate a method of generating an FIR filter for binaural rendering according to an embodiment of the present invention. According to an embodiment of the present invention, for binaural rendering in the QMF domain, an FIR filter converted into a plurality of subband filters in the QMF domain may be used. In this case, subband filters cut dependently on each subband frequency may be used for F-part rendering. That is, the high-speed convolution unit of the binaural renderer may perform variable-order filtering in the QMF domain by using truncated subband filters having different lengths according to subbands. The filter generation embodiments of FIGS. 8 to 10 described below may be performed by the BRIR parameterization unit 210 of FIG. 2.

8 shows an embodiment of a length according to each QMF band of a QMF domain filter used for binaural rendering. In the embodiment of FIG. 8, the FIR filter is converted into I QMF subband filters, and Fi denotes a truncated subband filter of QMF subband i. A total of 64 subbands may be used for the QMF domain, but the present invention is not limited thereto. In addition, N represents the length (number of taps) of the original subband filter, and the lengths of the cut subband filter are expressed as N1, N2, and N3, respectively. At this time, the lengths N, N1, N2, and N3 represent the number of taps in the down-sampled QMF domain.

According to an embodiment of the present invention, a truncated subband filter having different lengths (N1, N2, N3) according to each subband may be used for F-part rendering. In this case, the cut subband filter is a filter of a front part cut from the original subband filter, and may also be referred to as a front subband filter. In addition, a rear portion of the original subband filter after truncation may be referred to as a rear subband filter, and may be used for P-part rendering.

In the case of rendering using a BRIR filter, the filter order (i.e., filter length) for each subband is the parameters extracted from the original BRIR filter, such as reverberation time (RT) information for each subband filter, and EDC (Energy Decay Curve) value, energy decay time information, and the like. The reverberation time may vary depending on the frequency due to the acoustic characteristics of different levels of attenuation in the air for each frequency and the degree of sound absorption according to the material of the wall and ceiling. In general, a signal having a lower frequency has a longer reverberation time. If the reverberation time is long, it means that a lot of information remains behind the FIR filter. Therefore, it is preferable to use the filter by cutting it long to properly transmit the reverberation information. Accordingly, the length of each truncated subband filter of the present invention is determined based at least in part on characteristic information (eg, reverberation time information) extracted from the corresponding subband filter.

The length of the cut subband filter may be determined according to various embodiments. First, according to an embodiment, each subband is classified into a plurality of groups, and a length of each truncated subband filter may be determined according to the classified group. According to the example of FIG. 8, each subband can be classified into three zones (Zone 1, Zone 2, and Zone 3), and the truncated subband filters of Zone 1 corresponding to a low frequency are Zone 1 corresponding to a high frequency. It may have a longer filter order (ie, filter length) than the truncated subband filters of Zone 2 and Zone 3. In addition, as it goes to the high frequency region, the filter order of the truncated subband filter in the region may gradually decrease.

According to another embodiment of the present invention, the length of each truncated subband filter may be independently and variably determined for each subband according to characteristic information of the original subband filter. The length of each truncated subband filter is determined based on the truncated length determined in the corresponding subband, and is not affected by the length of the truncated subband filters of neighboring or other subbands. For example, the length of part or all of the truncated subband filter in Zone 2 may be longer than the length of at least one truncated subband filter in Zone 1.

According to another embodiment of the present invention, frequency domain variable order filtering may be performed only on some of subbands classified into a plurality of groups. That is, truncated subband filters having different lengths may be generated only for subbands belonging to some of the classified at least two groups. According to an embodiment, the group in which the truncated subband filter is generated may be a subband group (for example, Zone 1) classified as a low frequency band based on a preset constant or a preset frequency band.

The length of the truncated filter may be determined based on additional information obtained by the audio signal processing apparatus, such as a decoder's complexity, a complexity level (profile), or required quality information. The complexity may be determined according to a hardware resource of the audio signal processing apparatus or may be determined according to a value directly input by the user. The quality may be determined according to a user's request, or may be determined by referring to a value transmitted through the bitstream or other information included in the bitstream. In addition, the quality may be determined according to a value obtained by estimating the quality of the transmitted audio signal. For example, the higher the bit rate, the higher the quality may be regarded. In this case, the length of each truncated subband filter may increase proportionally according to complexity and quality, or may change at different rates for each band. In addition, the length of each truncated subband filter may be determined by a corresponding size unit, such as a multiple of a power of 2, in order to obtain an additional gain by high-speed processing such as an FFT to be described later. On the other hand, when the determined length of the cut subband filter is longer than the total length of the actual subband filter, the length of the cut subband filter may be adjusted to the length of the actual subband filter.

The BRIR parameterization unit generates truncated sub-band filter coefficients (F-part coefficients) corresponding to each truncated sub-band filter determined according to the above-described embodiment, and transfers them to the high-speed convolution unit. The fast convolution unit performs frequency domain variable-order filtering on each subband signal of the multi-audio signal by using the truncated subband filter coefficients.

9 shows another embodiment of the length of each QMF band of a QMF domain filter used for binaural rendering. In the embodiment of FIG. 9, overlapping descriptions of parts that are the same as or corresponding to the embodiment of FIG. 8 will be omitted.

In the embodiment of FIG. 9, Fi_L and Fi_R represent a truncated subband filter (front subband filter) used for F-part rendering of QMF subband i, respectively, and Pi is used for P-part rendering of QMF subband i. It represents the rear subband filter to be used. N denotes the length (number of taps) of the original subband filter, and NiF and NiP denote the lengths of the front subband filter and the rear subband filter of subband i, respectively. As mentioned above, NiF and NiP represent the number of taps in the down-sampled QMF domain.

According to the embodiment of FIG. 9, lengths of not only the front subband filter but also the rear subband filter may be determined based on parameters extracted from the original subband filter. That is, the lengths of the front and rear subband filters of each subband are determined based at least in part on characteristic information extracted from the corresponding subband filter. For example, the length of the front subband filter may be determined based on first reverberation time information of the subband filter, and the length of the rear subband filter may be determined based on second reverberation time information. That is, the front subband filter is a filter of the front part cut based on the first reverberation time information in the original subband filter, and the rear subband filter is a section after the front subband filter and is between the first reverberation time and the second reverberation time. It can be a filter at the end corresponding to the section of. According to an embodiment, the first reverberation time information may be RT20 and the second reverberation time information may be RT60, but the present invention is not limited thereto.

Within the second reverberation time, there is a part that changes from the early reflection sound part to the late reverberation part. That is, there is a point at which the transition from a section having a deterministic characteristic to a section having a stochastic characteristic exists, and this point is called a mixing time in terms of the BRIR of the entire band. In the case of a section before the mixing time, information providing direction is mainly present for each location, which is unique for each channel. On the other hand, in the case of the late reverberation part, since each channel has a common characteristic, it may be efficient to process a plurality of channels at once. Therefore, by estimating the mixing time for each subband, high-speed convolution is performed through F-part rendering before the mixing time, and processing that reflects common characteristics of each channel is performed through P-part rendering after the mixing time. have.

However, estimating the mixing time may cause errors due to bias from a perceptual point of view. Therefore, rather than estimating an accurate mixing time and dividing it into an F-part and a P-part based on a corresponding boundary, it is superior in terms of quality to perform high-speed convolution with the length of the F-part as long as possible. Accordingly, the length of the F-part, that is, the length of the front subband filter may be longer or shorter than the length corresponding to the mixing time according to the complexity-quality control.

In addition to this, in addition to the method of cutting as described above to reduce the length of each subband filter, when the frequency response of a specific subband is monotonic, modeling that reduces the filter of the corresponding subband to a lower order is possible. As a representative method, there is an FIR filter modeling using frequency sampling, and a filter that is minimized in terms of least squares can be designed.

According to an embodiment of the present invention, the length of the front subband filter and/or the rear subband filter for each subband may have the same value for each channel of the corresponding subband. Measurement errors may exist in the BRIR, and error elements such as bias also exist in estimating the reverberation time. Accordingly, in order to reduce this effect, the length of the filter may be determined based on a correlation between channels or between subbands. According to an embodiment, the BRIR parameterization unit extracts first characteristic information (for example, first reverberation time information) from subband filters corresponding to each channel of the same subband, and combines the extracted first characteristic information. Accordingly, one filter order information (or first cutoff point information) for a corresponding subband may be obtained. The front subband filters for each channel of the corresponding subband may be determined to have the same length based on the obtained filter order information (or first cutting point information). Similarly, the BRIR parameterization unit extracts second characteristic information (e.g., second reverberation time information) from subband filters corresponding to each channel of the same subband, and combines the extracted second characteristic information to obtain the corresponding subband. It is possible to obtain second cutting point information to be commonly applied to the rear subband filter corresponding to each channel of. Here, the front subband filter is a filter of the front part cut based on the first cutoff point information in the original subband filter, and the rear subband filter is a section after the front subband filter, between the first cutoff point and the second cutoff point. It can be a filter at the end corresponding to the section of

Meanwhile, according to another embodiment of the present invention, only F-part processing may be performed on subbands of a specific subband group. At this time, if processing is performed using only the filter up to the first cut point for the corresponding subband, the user can perceive it by the energy difference of the processed filter compared to when the processing is performed using the entire subband filter. A level of distortion can occur. In order to prevent such distortion, energy compensation may be performed for a region not used for processing in a corresponding subband filter, that is, a region after the first cutting point. The energy compensation can be performed by dividing the filter power to the first cutting point of the subband filter by the F-part coefficient (front subband filter coefficient), and multiplying the energy of the desired region, such as the total power of the subband filter. Do. Thus, the energy of the F-part coefficient can be adjusted to be equal to the energy of the entire subband filter. In addition, although the P-part coefficient is transmitted from the BRIR parameterization unit, the binaural rendering unit may not perform P-part processing based on the complexity-quality control. In this case, the binaural rendering unit may perform the energy compensation for the F-part coefficient by using the P-part coefficient.

In the F-part processing according to the above-described methods, filter coefficients of truncated subband filters having different lengths for each subband are obtained from one time domain filter (ie, proto-type filter). That is, since one time domain filter is converted into a plurality of QMF subband filters and the lengths of filters corresponding to each subband are varied, each truncated subband filter is obtained from one circular filter.

The BRIR parameterization unit generates front subband filter coefficients (F-part coefficients) corresponding to each of the front subband filters determined according to the above-described embodiment, and transfers the coefficients to the high-speed convolution unit. The fast convolution unit performs frequency domain variable order filtering on each subband signal of the multi-audio signal using the received front subband filter coefficients. In addition, the BRIR parameterization unit may generate rear subband filter coefficients (P-part coefficients) corresponding to each rear subband filter determined according to the above-described embodiment, and transmit the generated rear subband filter coefficients (P-part coefficients) to the late reverberation generation unit. The late reverberation generator may perform reverberation processing on each subband signal by using the received rear subband filter coefficients. According to an embodiment of the present invention, the BRIR parameterization unit may generate a downmix subband filter coefficient (downmix P-part coefficient) by combining rear subband filter coefficients for each channel, and transfer the result to the late reverberation generation unit. As described later, the late reverberation generator may generate 2-channel left and right subband reverberation signals using the received downmix subband filter coefficients.

10 shows another embodiment of a method of generating an FIR filter used for binaural rendering. In the embodiment of FIG. 10, duplicate descriptions of parts that are the same as or correspond to the embodiments of FIGS. 8 and 9 will be omitted.

Referring to FIG. 10, a plurality of QMF-transformed subband filters are classified into a plurality of groups, and different processing may be applied to each classified group. For example, a plurality of subbands are classified into a low frequency first subband group (Zone 1) and a high frequency second subband group (Zone 2) based on a preset frequency band (QMF band i). Can be. In this case, F-part rendering may be performed on the input subband signals of the first subband group, and QTDL processing to be described later may be performed on the input subband signals of the second subband group.

Accordingly, the BRIR parameterization unit generates front subband filter coefficients for each subband of the first subband group, and transfers the coefficients to the high-speed convolution unit. The fast convolution unit performs F-part rendering on the subband signals of the first subband group using the received front subband filter coefficients. According to an embodiment, the P-part rendering of the subband signals of the first subband group may be additionally performed by the late reverberation generator. In addition, the BRIR parameterization unit acquires at least one parameter from each subband filter coefficient of the second subband group and transmits the obtained parameter to the QTDL processing unit. The QTDL processing unit performs tap-delay line filtering on each subband signal of the second subband group as described later using the acquired parameters. According to an embodiment of the present invention, a preset frequency (QMF band i) for dividing the first subband group and the second subband group may be determined based on a predetermined constant value, or a bit of the transmitted audio input signal. It may also be determined according to thermal characteristics. For example, in the case of an audio signal using SBR, the second subband group may be set to correspond to the SBR band.

According to another embodiment of the present invention, the plurality of subbands may be classified into three subband groups based on a preset first frequency band (QMF band i) and a second frequency band (QMF band j). That is, the plurality of subbands is a first subband group (Zone 1) that is a low frequency region less than or equal to the first frequency band, and a second subband group that is an intermediate frequency region that is greater than or equal to the second frequency band. It may be classified into a band group (Zone 2) and a third subband group (Zone 3) that is a higher frequency region than the second frequency band. In this case, F-part rendering and QTDL processing may be performed on the subband signals of the first subband group and the subband signals of the second subband group, respectively, as described above, and the subbands of the third subband group Rendering may not be performed on signals.

<Late reverberation rendering>

Next, various embodiments of P-part rendering according to the present invention will be described with reference to FIGS. 11 to 14. That is, various embodiments of the late reverberation generator 240 of FIG. 2 that performs P-part rendering in the QMF domain will be described with reference to FIGS. 11 to 14. In the embodiments of FIGS. 11 to 14, it is assumed that the multi-channel input signal is received as a subband signal in the QMF domain. Accordingly, processing of each of the components of FIGS. 11 to 14, that is, the decorrelator 241, the subband filtering unit 242, the IC matching unit 243, the downmix unit 244, and the energy attenuation matching unit 246 May be performed for each QMF subband. In the embodiments of FIGS. 11 to 14, detailed descriptions of parts overlapping with the embodiments of the previous drawings will be omitted.

In the above-described embodiments of FIGS. 8 to 10, Pi (P1, P2, P3, ...) corresponding to the P-part corresponds to the rear part of each subband filter removed by the frequency variable cutting, and is typically used for late reverberation. It contains information about. According to the complexity-quality control, the length of the P-part may be defined as the entire filter after the cut point of each subband filter, or may be defined as a smaller length by referring to the second reverberation time information of the corresponding subband filter. have.

P-part rendering may be performed independently for each channel, or may be performed for downmixed channels. In addition, the P-part rendering may be applied by different processing for each subband group or each subband, or the same processing for all subbands. At this time, the processing applicable to the P-part includes energy reduction compensation for the input signal, tap-delay line filtering, processing using an Infinite Impulse Response (IIR) filter, processing using an artificial reverberator, and frequency (FIIC). -Independent interaural coherence (FDIC) compensation, frequency-dependent interaural coherence (FDIC) compensation, etc. may be included.

On the other hand, for parametric processing of the P-part, it is important to preserve two characteristics, namely, energy decay relief (EDR) and frequency-dependent interaural coherence (FDIC). First, when the P-part is observed from the energy point of view, it can be seen that the EDR for each channel is the same or similar. Since each channel has a common EDR, it is reasonable from an energy point of view to downmix all channels to one or two channels and then perform P-part rendering on the downmixed channels. At this time, the P-part rendering operation that must perform M convolutions for M channels is reduced to M-to-O downmix and one (or two) convolution, thereby reducing the gain of a considerable amount of computation. Can provide.

Next, it is necessary to compensate the FDIC in P-part rendering. There are several methods of estimating the FDIC, but the following equation can be used.

는 임펄스 응답

의 STFT(Short Time Fourier Transform) 계수, n은 시간 인덱스, i는 주파수 인덱스, k는 프레임 인덱스, m은 출력 채널 인덱스(L, R)를 나타낸다. 또한, 분자의 함수

는 입력 x의 실수 값을 출력하고,

는 x의 복소 켤레(complex conjugate) 값을 나타낸다. 상기 수식에서 분자 부분은 실수 값 대신 절대값을 취하는 함수로 치환될 수도 있다.here,

Is the impulse response

STFT (Short Time Fourier Transform) coefficient, n is a time index, i is a frequency index, k is a frame index, and m is an output channel index (L, R). Also, the function of the molecule

Outputs the real value of the input x,

Represents the complex conjugate of x. In the above equation, the numerator part may be substituted with a function taking an absolute value instead of a real value.

Meanwhile, in the present invention, since binaural rendering is performed in the QMF domain, FDIC may be defined by the following equation.

는 BRIR의 서브밴드 필터를 나타낸다.Where i is the subband index, k is the time index in the subband,

Denotes a subband filter of BRIR.

The FDIC of the late reverberation part is a parameter that is mainly affected by the position of the two microphones when BRIR is recorded, and is not affected by the position of the speaker, that is, direction and distance. Assuming that the listener's head is a sphere, BRIR's theoretical FDIC (IC _ideal ) can satisfy the following equation.

Here, r is the distance between the two ears of the listener, that is, the distance between the two microphones, and k is the frequency index.

When analyzing the FDIC using the BRIR of multiple channels, it can be seen that the initial reflection sound mainly included in the F-part is very different for each channel. That is, the FDIC of the F-part changes very differently for each channel. On the other hand, in the case of the high frequency band, the FDIC changes very significantly, but this is because a large measurement error occurs due to the characteristic of the high frequency band signal where energy rapidly attenuates, and when the average for each channel is taken, the FDIC converges to almost zero. . On the other hand, even in the case of the P-part, a difference in FDIC for each channel occurs due to a measurement error, but it can be seen that the average convergence to the sync function shown in Equation 5 is obtained. According to an embodiment of the present invention, it is possible to implement a late reverberation generator for P-part rendering based on the above-described characteristics.

11 shows a late reverberation generation unit 240A according to an embodiment of the present invention. According to the embodiment of FIG. 11, the late reverberation generation unit 240A may include a subband filtering unit 242 and downmix units 244a and 244b.

The subband filtering unit 242 uses the P-part coefficients to provide multi-channel input signals X0, X1, ... , X_M-1 is filtered for each subband. The P-part coefficient is received from the BRIR parameterization unit (not shown) as described above, and may include coefficients of rear subband filters having different lengths for each subband. The subband filtering unit 242 performs high-speed convolution between the QMF domain subband signal and the corresponding rear subband filter of the QMF domain for each frequency. In this case, the length of the rear subband filter may be determined based on RT60 as described above, but may be set to a value larger or smaller than RT60 according to complexity-quality control.

The multi-channel input signals are respectively left channel signals X_L0, X_L1, ... by the subband filtering unit 242. , X_L_M-1 and right channel signals X_R0, X_R1,… , Rendered as X_R_M-1. The downmixing units 244a and 244b downmix the rendered plurality of left channel signals and the plurality of right channel signals for each of the left and right channels, respectively, to generate two channel left and right output signals Y_Lp and Y_Rp.

12 shows a late reverberation generation unit 240B according to another embodiment of the present invention. According to the embodiment of FIG. 12, the late reverberation generation unit 240B includes a decoder 241, an IC matching unit 243, downmix units 244a and 244b, and energy attenuation matching units 246a and 246b. can do. In addition, for processing by the late reverberation generation unit 240B, the BRIR parameterization unit (not shown) may include an IC estimation unit 213 and a downmix subband filter generation unit 216.

According to the embodiment of FIG. 12, the late reverberation generation unit 240B may reduce the amount of computation by using the same energy decay characteristics for each channel of the late reverberation part. That is, the late reverberation generator 240B performs decoding and interaural coherence (IC) adjustment for each multi-channel signal, and downmixes the adjusted input signal and decoding signal for each channel into left and right channel signals. Afterwards, by compensating the energy attenuation of the downmixed signal, the left and right output signals of 2 channels are generated. More specifically, the decorrelator 241 includes each of the multi-channel input signals X0, X1, ... , Decorrelation signals D0, D1,… for X_M-1 , Create D_M-1. The decorrelator 241 is a kind of preprocessor for adjusting coherence between both ears, and a phase randomizer may be used. You can change it.

라고 했을 때, IC 매칭이 수행된 좌, 우 채널 신호 X_L, X_R은 다음 수식과 같이 표현될 수 있다.Meanwhile, the IC estimating unit 213 of the BRIR parameterization unit (not shown) estimates the IC value and transmits it to the binaural rendering unit (not shown). The binaural rendering unit may store the received IC value in the memory 255 and transfer the received IC value to the IC matching unit 243. The IC matching unit 243 may directly receive an IC value from the BRIR parameterization unit or may obtain an IC value previously stored in the memory 255. The input signals and decorrelation signals for each channel are left channel signals X_L0, X_L1, ... from the IC matching unit 243. , X_L_M-1 and right channel signals X_R0, X_R1,… , Rendered as X_R_M-1. The IC matching unit 243 performs weight summation between the decorrelation signal and the original input signal for each channel by referring to the IC value, and adjusts the coherence between the two channel signals through this. At this time, since the input signal for each channel is a signal in the subband domain, the aforementioned FDIC can be matched. The original channel signal is X, the decorrelation channel signal is D, and the IC of the corresponding subband is

In the case of, the left and right channel signals X_L and X_R on which IC matching has been performed may be expressed as the following equation.

(East order of the abdomen)

, k번째 채널에서의 N번째 샘플까지의 절단된 BRIR을

, N번째 샘플 이후의 절단된 부분의 에너지를 보상한 다운믹스 서브밴드 필터 계수를

라고 한다면,

는 다음과 같은 수식을 이용하여 구할 수 있다.The downmixing units 244a and 244b downmix the plurality of left channel signals and the plurality of right channel signals rendered through IC matching for each of the left and right channels to generate two-channel left and right rendering signals. Next, the energy attenuation matching units 246a and 246b generate 2 channel left and right output signals Y_Lp and Y_Rp by reflecting energy attenuation of the 2 channel left and right rendering signals, respectively. The energy attenuation matching units 246a and 246b perform energy attenuation matching using the downmix subband filter coefficients obtained from the downmix subband filter generator 216. The downmix subband filter coefficients are generated by combining the rear subband filter coefficients for each channel of the subband. For example, the downmix subband filter coefficient may include a subband filter coefficient obtained by taking a root of an average of square amplitude responses of rear subband filter coefficients for each channel for a corresponding subband. Therefore, the downmix subband filter coefficient reflects the energy reduction characteristic of the late reverberation part for the corresponding subband signal. The downmix subband filter coefficient may include a downmix subband filter coefficient downmixed to mono or stereo according to an embodiment, and, like FDIC, directly received from the BRIR parameterizer or from a value previously stored in the memory 225 Can be obtained. BRIR in which the F-part in the kth channel among M channels is cut

, the truncated BRIR up to the Nth sample in the kth channel

, The downmix subband filter coefficient compensated for the energy of the truncated portion after the Nth sample

If you say,

Can be obtained using the following equation.

here,

13 shows a late reverberation generation unit 240C according to another embodiment of the present invention. Each configuration of the late reverberation generator 240C of FIG. 13 may be the same as that of the late reverberation generator 240B described in the embodiment of FIG. 12, and a data processing order between the components may be partially different.

According to the embodiment of FIG. 13, the late reverberation generation unit 240C may further reduce the amount of computation by using the same FDIC for each channel of the late reverberation part. That is, the late reverberation generation unit 240C downmixes each multi-channel signal into left and right channel signals, adjusts the IC of the downmixed left and right channel signals, and then adjusts the energy for the adjusted left and right channel signals. By compensating for attenuation, two-channel left and right output signals can be generated.

More specifically, the decorrelator 241 includes each of the multi-channel input signals X0, X1, ... , Decorrelation signals D0, D1,… for X_M-1 , Create D_M-1. Next, the downmixing units 244a and 244b downmix the multi-channel input signal and the decorrelation signal, respectively, to generate two-channel downmix signals X_DMX and D_DMX. The IC matching unit 243 weights the two-channel downmix signal by referring to the IC value, and adjusts the coherence between the two channel signals through this. The energy attenuation compensation units 246a and 246b perform energy compensation for each of the left and right channel signals X_L and X_R on which IC matching is performed by the IC matching unit 243, and thus output signals X_Lp and Y_Rp of the two channels. Create In this case, the energy compensation information used for energy compensation may include a downmix subband filter coefficient for each subband.

14 shows a late reverberation generation unit 240D according to another embodiment of the present invention. Each configuration of the late reverberation generating unit 240D of FIG. 14 may be the same as that of the late reverberation generating units 240B and 240C described in the embodiments of FIGS. 12 and 13, but has a more simplified feature.

First, the down-mixing unit 244 includes multi-channel input signals X0, X1, ... , X_M-1 is downmixed for each subband to generate a mono downmix signal (ie, a mono subband signal) X_DMX. The energy attenuation matching unit 246 reflects energy attenuation for the generated mono downmix signal. In this case, a downmix subband filter coefficient for each subband may be used to reflect energy attenuation. Next, the decorrelator 241 generates a decorrelation signal D_DMX of the mono downmix signal in which the energy attenuation is reflected. The IC matching unit 243 adds weights of the mono downmix signal reflecting the energy attenuation and the decorrelation signal with reference to the FDIC value, thereby generating the left and right output signals Y_Lp and Y_Rp of two channels. According to the embodiment of FIG. 14, since energy attenuation matching is performed only once for the mono downmix signal X_DMX, the amount of computation can be further saved.

<QTDL processing of high frequency band>

Next, various embodiments of the QTDL processing of the present invention will be described with reference to FIGS. 15 and 16. That is, various embodiments of the QTDL processing unit 250 of FIG. 2 that performs QTDL processing in the QMF domain will be described with reference to FIGS. 15 and 16. 15 and 16, it is assumed that the multi-channel input signal is received as a subband signal in the QMF domain. Accordingly, in the embodiments of FIGS. 15 and 16, the tap-delay line filter and the one-tap-delay line filter may perform processing for each QMF subband. In addition, QTDL processing may be performed only on an input signal of a high frequency band classified based on a preset constant or a preset frequency band as described above. If SBR (Spectral Band Replication) is applied to the input audio signal, the high frequency band may correspond to the SBR band. In the embodiments of FIGS. 15 and 16, detailed descriptions of parts that overlap with the embodiments of the previous drawings will be omitted.

SBR (Spectral Band Replication), which is used for efficient encoding of high frequency bands, is a tool to secure the bandwidth as much as the original signal by re-extending the narrowed band width due to discarding the high frequency band signal during low bit rate encoding. . In this case, the high frequency band is generated by using the information of the low frequency band transmitted after being encoded and the additional information of the high frequency band signal transmitted from the encoder. However, high frequency components generated using SBR may cause distortion due to incorrect harmonic generation. In addition, the SBR band is a high frequency band, and as described above, the reverberation time of the corresponding frequency band is very short. That is, the BRIR subband filter of the SBR band has little effective information and has a fast attenuation rate. Therefore, for BRIR rendering for a high frequency band similar to the SBR band, rendering using a few effective taps rather than performing convolution may be very effective in terms of the amount of operation compared to the quality of sound quality.

15 shows a QTDL processing unit 250A according to an embodiment of the present invention. According to the embodiment of FIG. 15, the QTDL processing unit 250A uses a tap-delay line filter to provide multi-channel input signals X0, X1, ... , Filtering is performed for each subband for X_M-1. The tap-delay line filter performs convolution of only a few preset taps for each channel signal. In this case, the number of taps used may be determined based on a parameter directly extracted from the BRIR subband filter coefficients corresponding to the corresponding subband signal. The parameter includes delay information for each tap to be used in the tap-delay line filter and gain information corresponding thereto.

The number of taps used in the tap-delay line filter may be determined by complexity-quality control. The QTDL processing unit 250A receives a set of parameters (gain information, delay information) corresponding to the number of taps for each channel and subband from the BRIR parameterization unit based on the predetermined number of taps. In this case, the received parameter set is extracted from the BRIR subband filter coefficients corresponding to the corresponding subband signal, and may be determined according to various embodiments. For example, from among a plurality of peaks of the corresponding BRIR subband filter coefficient, a set of parameters for each of the peaks extracted by the predetermined number of taps in the order of absolute value, real value, or imaginary value may be received. have. At this time, the delay information of each parameter indicates the position information of the corresponding peak, and has an integer value in units of samples in the QMF domain. Also, the gain information is determined based on the size of a peak corresponding to the corresponding delay information. In this case, a corresponding peak value in the subband filter coefficients itself may be used as the gain information, but a weight value of the corresponding peak after energy compensation for all subband filter coefficients is performed may be used. The gain information is obtained by using a real weight and an imaginary weight for a corresponding peak together, and thus has a complex value.

The plurality of channel signals filtered by the tap-delay line filter are summed into two channel left and right output signals Y_L and Y_R for each subband. Meanwhile, parameters used in each tap-delay line filter of the QTDL processing unit 250A may be stored in a memory during an initialization process of binaural rendering, and QTDL processing may be performed without an additional operation for parameter extraction.

16 shows a QTDL processing unit 250B according to another embodiment of the present invention. According to the embodiment of FIG. 16, the QTDL processing unit 250B uses a one-tap-delay line filter to provide multi-channel input signals X0, X1, ... , Filtering is performed for each subband for X_M-1. The one-tap-delay line filter can be understood as performing convolution in only one tap for each channel signal. In this case, the tap used may be determined based on a parameter directly extracted from the BRIR subband filter coefficient corresponding to the corresponding subband signal. As described above, the parameter includes delay information extracted from the BRIR subband filter coefficient and gain information corresponding thereto.

In Fig. 16, L_0, L_1, ... L_M-1 represents the delay for the BRIR from the M channels to the left ear, respectively, R_0, R_1, ... , R_M-1 denotes a delay for BRIR from M channels to the right ear, respectively. In this case, the delay information indicates position information on the maximum peak in the order of absolute value size, real value size, or imaginary value size among corresponding BRIR subband filter coefficients. In addition, in FIG. 16, G_L_0, G_L_1, ... , G_L_M-1 denotes a gain corresponding to each delay information of the left channel, and G_R_0, G_R_1, ... , G_R_M-1 denotes a gain corresponding to each delay information of the right channel, respectively. As described above, each gain information is determined based on the size of a peak corresponding to the corresponding delay information. In this case, a corresponding peak value in the subband filter coefficients itself may be used as the gain information, but a weight value of the corresponding peak after energy compensation for all subband filter coefficients is performed may be used. The gain information is obtained by using a real weight and an imaginary weight for a corresponding peak together, and thus has a complex value.

15, a plurality of channel signals filtered by a one-tap-delay line filter are summed into two channel left and right output signals Y_L and Y_R for each subband. In addition, parameters used in each one-tap-delay line filter of the QTDL processing unit 250B may be stored in memory during the initialization process of binaural rendering, and QTDL processing may be performed without an additional operation for parameter extraction. have.

In the above, the present invention has been described through specific embodiments, but those skilled in the art can modify and change without departing from the spirit and scope of the present invention. That is, although the present invention has been described with respect to an embodiment of binaural rendering for a multi-audio signal, the present invention can be equally applied and extended to various multimedia signals including video signals as well as audio signals. Therefore, what can be easily inferred by a person belonging to the technical field to which the present invention belongs from the detailed description and examples of the present invention is construed as belonging to the scope of the present invention.

200: binaural renderer 210: BRIR parameterization unit
220: binaural rendering unit 230: high-speed convolution unit
240: late reverberation generation unit 250: QTDL processing unit

Claims (12)

As a multimedia signal processing device,
And a rendering unit for receiving a multimedia signal including at least one of a multi-channel signal or a multi-object signal and filtering a plurality of subband signals of the multimedia signal,
The rendering unit,
Obtaining a set of subband filter coefficient(s) corresponding to each subband signal of the multimedia signal,
Filtering a corresponding subband signal using the obtained set of subband filter coefficient(s),
A multimedia signal processing apparatus in which a set of subband filter coefficient(s) of each subband has a variable length in a frequency domain based on a filter order determined for each subband. The method of claim 1,
The set of each subband filter coefficient(s) is a set truncated from the set of prototype subband filter coefficients,
The truncation length of each set is determined based on the filter order obtained based at least in part on characteristic information extracted from circular subband filter coefficients of the corresponding subband. The method of claim 2,
The characteristic information includes energy decay time information of circular subband filter coefficients of a corresponding subband. The method of claim 3,
The multimedia signal includes an audio signal,
The set of circular subband filter coefficients is a set of BRIR subband filter coefficients obtained from a set of Binaural Room Impulse Response (BRIR) filter coefficients corresponding to the audio signal. The method of claim 4,
The characteristic information includes reverberation time information of circular subband filter coefficients of a corresponding subband. The method of claim 2,
The filter order is a multimedia signal processing apparatus having a single value for each subband. As a multimedia signal processing method,
Receiving a multimedia signal including a plurality of subband signals, the multimedia signal including at least one of a multi-channel signal and a multi-object signal;
Obtaining a set of subband filter coefficient(s) corresponding to each subband signal of the multimedia signal; And
Filtering a corresponding subband signal using the obtained set of subband filter coefficient(s); Including,
A multimedia signal processing method in which a set of subband filter coefficient(s) of each subband has a variable length in a frequency domain based on a filter order determined for each subband. The method of claim 7,
The set of each subband filter coefficient(s) is a set truncated from the set of prototype subband filter coefficients,
The truncation length of each set is determined based on the filter order obtained based at least in part on characteristic information extracted from circular subband filter coefficients of the corresponding subband. The method of claim 8,
The characteristic information includes energy decay time information of circular subband filter coefficients of a corresponding subband. The method of claim 9,
The multimedia signal includes an audio signal,
The set of circular subband filter coefficients is a set of BRIR subband filter coefficients obtained from a set of Binaural Room Impulse Response (BRIR) filter coefficients corresponding to the audio signal. The method of claim 10,
The characteristic information includes reverberation time information of circular subband filter coefficients of a corresponding subband. The method of claim 8,
The method of processing a multimedia signal in which the filter order has a single value for each subband.

Images