WO2015142073A1 - 오디오 신호 처리 방법 및 장치 - Google Patents
오디오 신호 처리 방법 및 장치 Download PDFInfo
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Definitions
- the present invention relates to an audio signal processing method and apparatus, and more particularly, to an audio signal processing method and apparatus capable of synthesizing an object signal and a channel signal and effectively binaural rendering them.
- 3D audio is a series of signal processing, transmission, encoding, and playback methods for providing a realistic sound in three-dimensional space by providing another axis corresponding to the height direction to a sound scene on a horizontal plane (2D) provided by conventional surround audio. Also known as technology.
- a rendering technique is required in which a sound image is formed at a virtual position in which no speaker exists even if a larger number of speakers or a smaller number of speakers are used.
- 3D audio is expected to be an audio solution for ultra-high definition televisions (UHDTVs), as well as sound in vehicles that are evolving into high-quality infotainment spaces, as well as theater sounds, personal 3DTVs, tablets, smartphones, and cloud games. It is expected to be applied in.
- UHDTVs ultra-high definition televisions
- infotainment spaces as well as theater sounds, personal 3DTVs, tablets, smartphones, and cloud games. It is expected to be applied in.
- a channel based signal and an object based signal may exist in the form of a sound source provided to 3D audio.
- a sound source in which a channel-based signal and an object-based signal are mixed, thereby providing a user with a new type of listening experience.
- a performance difference may exist between a channel renderer for processing a channel based signal and an object renderer for processing an object based signal in the audio signal processing apparatus.
- binaural rendering of the audio signal processing apparatus may be implemented based on a channel-based signal.
- the corresponding sound scene may not be reproduced as intended through binaural rendering. Therefore, there is a need to solve various problems that may occur due to the performance difference between the channel renderer and the object renderer.
- the present invention is to solve the above-mentioned problems of the prior art, by implementing an object renderer and a channel renderer corresponding to the spatial resolution that can be provided by the binaural renderer, to produce an output signal that matches the performance of the binaural renderer
- An object of the present invention is to provide an audio signal processing method and apparatus.
- the present invention is to reproduce a multi-channel or multi-object signal in stereo, a filtering process that requires a large amount of computation in the binaural rendering to preserve the three-dimensional effect such as the original signal with a very low computational amount while minimizing sound quality loss It has a purpose to implement.
- the present invention has an object to minimize the diffusion of distortion through a high quality filter when there is distortion in the input signal itself.
- the present invention has an object to implement a finite impulse response (FIR) filter having a very long length to a filter of a smaller length.
- FIR finite impulse response
- the present invention has an object to minimize the distortion of the portion damaged by the missing filter coefficients when performing the filtering using the abbreviated FIR filter.
- the present invention provides an audio signal processing method and an audio signal processing apparatus as follows.
- the present invention includes the steps of receiving an input audio signal including a multi-channel signal; Receiving truncated subband filter coefficients for filtering the input audio signal, wherein the truncated subband filter coefficients are obtained from a Binaural Room Impulse Response (BRIR) filter coefficient for binaural filtering of the input audio signal At least a portion of a subband filter coefficient, wherein the length of the truncated subband filter coefficient is determined based on filter order information obtained at least in part using reverberation time information extracted from the corresponding subband filter coefficient; Obtaining vector information indicating the BRIR filter coefficients corresponding to each channel of the input audio signal; And filtering each subband signal of the multichannel signal using the truncated subband filter coefficients corresponding to the corresponding channel and the subband based on the vector information. It provides an audio signal processing method comprising a.
- BRIR Binaural Room Impulse Response
- the present invention also provides an audio signal processing apparatus for performing binaural rendering on an input audio signal, comprising: a parameterizer for generating a filter of the input audio signal; And a binaural rendering unit configured to receive an input audio signal including a multichannel signal and to filter the input audio signal using the parameter generated by the parameterization unit, wherein the binaural rendering unit includes the parameter.
- BRIR Binaural Room Impulse Response
- a length of the truncated subband filter coefficients is determined based on filter order information obtained by using at least part of reverberation time information extracted from the corresponding subband filter coefficients.
- the BRIR filter coefficients corresponding to each channel of the input audio signal Obtaining vector information, and filtering each subband signal of the multichannel signal using the truncated subband filter coefficients corresponding to the corresponding channel and the subband based on the vector information. Provides ..
- the vector information is converted into the BRIR filter coefficient corresponding to the specific channel. It is characterized by indicating.
- the vector information may have a minimum geometric distance from the position information of the specific channel when a BRIR filter coefficient having position information matching the position information of the specific channel of the input audio signal is not present in the BRIR filter set.
- the BRIR filter coefficients may be indicated by the BRIR filter coefficients corresponding to the specific channel.
- the geometric distance is characterized in that the sum of the absolute value of the absolute deviation and the azimuth deviation of the altitude deviation between the two positions.
- the length of at least one truncated subband filter coefficient is different from the length of the truncated subband filter coefficient of another subband.
- receiving a bitstream of an audio signal comprising at least one of a channel signal and an object signal; Decoding each audio signal included in the bitstream; Receiving virtual layout information corresponding to a Binaural Room Impulse Response (BRIR) filter set for binaural rendering of the audio signal, wherein the virtual layout information includes information of target channels determined based on the BRIR filter set ; Rendering each of the decoded audio signals into a signal of the target channel based on the received virtual layout information; It provides an audio signal processing method comprising a.
- BRIR Binaural Room Impulse Response
- an audio signal processing apparatus comprising: a core decoder for receiving a bitstream of an audio signal including at least one of a channel signal and an object signal, and decoding each audio signal included in the bitstream; And receiving virtual layout information corresponding to a Binaural Room Impulse Response (BRIR) filter set for binaural rendering of the audio signal, wherein the virtual layout information includes information of target channels determined based on the BRIR filter set.
- a renderer for rendering each of the decoded audio signals into a signal of the target channel based on the received virtual layout information; It provides an audio signal processing apparatus comprising a.
- the position set corresponding to the virtual layout information is a subset of the position set corresponding to the BRIR filter set, and the position set of the virtual layout information represents position information of each target channel.
- the BRIR filter set is received from a binaural renderer that performs the binaural rendering.
- the apparatus may further include a mixer configured to generate an output signal for each target channel by mixing each audio signal rendered as the signal of the target channel for each target channel.
- the apparatus further includes a binaural renderer configured to binaurally render the mixed target channel-specific output signal using the BRIR filter coefficients of the BRIR filter set corresponding to the corresponding target channel.
- the binaural renderer converts the BRIR filter coefficients into a plurality of subband filter coefficients, and obtains each of the subband filter coefficients by using at least partially reverberation time information extracted from the corresponding subband filter coefficients. Truncated based on the filtered filter order information, wherein a length of at least one truncated subband filter coefficient is different from a truncated subband filter coefficient of another subband, and each subband of the output signal for each mixed target channel The signal is filtered using the truncated subband filter coefficients corresponding to the corresponding channel and the subband.
- effective binaural rendering may be implemented by performing channel and object rendering based on the data set held by the binaural renderer.
- object rendering may be implemented to provide improved sound quality.
- the amount of computation can be dramatically lowered while minimizing sound quality loss when performing binaural rendering on a multichannel or multiobject signal.
- the present invention provides a method for efficiently performing various types of filtering of a multimedia signal including an audio signal with a low calculation amount.
- FIG. 1 is a block diagram showing an overall audio signal processing system including an audio encoder and an audio decoder according to an embodiment of the present invention.
- FIG. 2 is a block diagram illustrating an arrangement of a multichannel speaker according to an exemplary embodiment of the multichannel audio system.
- FIG. 3 is a diagram schematically showing positions of sound objects constituting a three-dimensional sound scene in a listening space
- FIG. 4 is a block diagram illustrating an audio signal decoder according to an embodiment of the present invention.
- FIG. 5 is a block diagram illustrating an audio decoder according to a further embodiment of the present invention.
- FIG. 6 illustrates an embodiment of the invention performing rendering for an exception object.
- FIG. 7 is a block diagram showing each configuration of a binaural renderer according to an embodiment of the present invention.
- FIG. 8 is a diagram illustrating a filter generation method for binaural rendering according to an embodiment of the present invention.
- FIG 9 illustrates in detail QTDL processing according to an embodiment of the present invention.
- FIG. 10 is a block diagram showing each configuration of the BRIR parameterization unit of the present invention.
- FIG. 11 is a block diagram showing each configuration of a VOFF parameterization unit of the present invention.
- FIG. 12 is a block diagram showing the detailed configuration of the VOFF parameter generating unit of the present invention.
- FIG. 13 is a block diagram showing each configuration of a QTDL parameterization unit of the present invention.
- FIG. 14 illustrates an embodiment of a method for generating FFT filter coefficients for fast convolution in units of blocks.
- FIG. 1 is a block diagram showing an overall audio signal processing system including an audio encoder and an audio decoder according to an embodiment of the present invention.
- the audio encoder 1100 encodes an input sound scene to generate a bitstream.
- the audio decoder 1200 may receive the generated bitstream and decode and render the corresponding bitstream using the audio signal processing method according to an embodiment of the present invention to generate an output sound scene.
- the audio signal processing apparatus may refer to the audio decoder 1200 in a narrow sense, but is not limited thereto and may refer to a detailed configuration included in the audio decoder 1200, and the audio encoder 1100 and the audio decoder may be referred to. It may also refer to an entire audio signal processing system including 1200.
- FIG. 2 is a diagram illustrating a configuration of a multichannel speaker according to an exemplary embodiment of the multichannel audio system.
- a plurality of speaker channels may be used to increase presence, and in particular, a plurality of speakers may be arranged in width, depth, and height directions to provide a sense of presence in three-dimensional space.
- 2 illustrates a speaker layout of 22.2 channels as an embodiment, but the present invention is not limited to a specific channel number or a specific speaker layout.
- a speaker set of 22.2 channels may be composed of three layers, a top layer, a middle layer, and a bottom layer.
- the front side a total of nine speakers may be arranged in the upper layer, three in the front side, three in the middle position, and three in the surround position.
- the middle layer may be arranged in front of five, two in the middle position, three in the surround position can be arranged a total of 10 speakers.
- three speakers may be disposed in front of the lower layer, and two LFE channel speakers may be provided.
- 3 schematically illustrates positions of sound objects constituting a three-dimensional sound scene in a listening space.
- the positions of the respective sound objects 51 constituting the three-dimensional sound scene on the listening space 50 in which the listener 52 listens to the 3D audio may vary in the form of a point source. Can be distributed at a location.
- the sound scene may include a plane wave type sound source, an ambient sound source, etc. in addition to the point source. As such, in order to clearly provide the listeners 52 with the objects and sound sources that are variously distributed in the 3D space, an efficient rendering method is required.
- the audio decoder 1200 of the present invention includes a core decoder 10, a rendering unit 20, a mixer 30, and a post processing unit 40.
- the core decoder 10 decodes the received bitstream and delivers it to the rendering unit 20.
- a signal output from the core decoder 10 and delivered to the rendering unit includes a loudspeaker channel signal 411, an object signal 412, a SAOC channel signal 414, a HOA signal 415, and object metadata. Bitstream 413 and the like.
- the core decoder 10 may use the core codec used when encoding in the encoder. For example, a codec based on MP3, AAC, AC3 or USAC (Unified Speech and Audio Coding) may be used.
- the received bitstream may further include an identifier for identifying whether the signal decoded by the core decoder 10 is a channel signal, an object signal, or a HOA signal.
- the bitstream when the signal to be decoded is the channel signal 411, the bitstream further includes an identifier for identifying which channel (eg, left speaker correspondence, top rear right speaker correspondence, etc.) in each multichannel corresponds to the signal. Can be.
- the signal to be decoded is the object signal 412, which indicates in which position the signal is reproduced, such as object metadata information 425a and 425b obtained by decoding the object metadata bitstream 413. Information can be further obtained.
- the audio decoder may perform flexible rendering to increase the quality of the output audio signal.
- Flexible rendering may mean a process of converting a format of a decoded audio signal based on a loudspeaker arrangement (playback layout) of a real playback environment or a virtual speaker arrangement (virtual layout) of a Binaural Room Impulse Response (BRIR) filter set.
- BRIR Binaural Room Impulse Response
- speakers placed in a living room environment will have different orientation angles and distances compared to standard recommendations. As the height, direction, and distance of the speaker from the speaker differ from the speaker layout according to the specification recommendation, it may be difficult to provide an ideal 3D sound scene when reproducing the original signal at the changed speaker position.
- a flexible rendering that converts an audio signal and corrects a change due to a positional difference between speakers is required.
- the rendering unit 20 renders the signal decoded by the core decoder 10 into the target output signal using the reproduction layout information or the virtual layout information.
- the reproduction layout information indicates the configuration of the target channel and may be expressed as loudspeaker layout information of the reproduction environment.
- the virtual layout information may be obtained based on a Binaural Room Impulse Response (BRIR) filter set used in the binaural renderer 200, wherein a set of positions corresponding to the virtual layout is a BRIR. It may consist of a subset of the position set corresponding to the filter set. In this case, the position set of the virtual layout represents position information of each target channel.
- BRIR Binaural Room Impulse Response
- the rendering unit 20 may include a format converter 22, an object renderer 24, an OAM decoder 25, a SAOC decoder 26, and a HOA decoder 28.
- the rendering unit 20 performs rendering using at least one of the above configurations according to the type of the decoded signal.
- the format converter 22 may also be referred to as a channel renderer, and converts the transmitted channel signal 411 into an output speaker channel signal. That is, the format converter 22 performs conversion between the transmitted channel configuration and the speaker channel arrangement 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 if the transmitted channel arrangement and the channel arrangement to be reproduced are different, the format converter 22 is a channel signal. Perform a downmix or transform on 411. According to an embodiment of the present invention, the audio decoder may generate an optimal downmix matrix using a combination of an input channel signal and an output speaker channel signal, and perform the downmix using the matrix.
- a channel renderer converts the transmitted channel signal 411 into an output speaker channel signal. That is, the format converter 22 performs conversion between the transmitted channel configuration and the speaker channel arrangement 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.
- the channel signal 411 processed by the format converter 22 may include a pre-rendered object signal.
- at least one object signal may be pre-rendered and mixed with the channel signal before encoding the audio signal.
- the mixed object signal 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 for the object based audio signal.
- the object-based audio signal may include individual object waveforms and parametric object waveforms.
- each object signal is provided to the encoder as a monophonic waveform, and the encoder transmits the respective object signals using single channel elements (SCEs).
- SCEs single channel elements
- a parametric object waveform a plurality of object signals are downmixed into at least one channel signal, and characteristics of each object and a relationship between them are represented by a spatial audio object coding (SAOC) parameter.
- SAOC spatial audio object coding
- compressed object metadata corresponding thereto may be transmitted together.
- Object metadata quantizes object attributes in units of time and space to specify the position and gain of each object in three-dimensional space.
- the OAM decoder 25 of the rendering unit 20 receives the compressed object metadata bitstream 413, decodes it, and forwards it to the object renderer 24 and / or the SAOC decoder 26.
- the object renderer 24 uses the object metadata information 425a to render each object signal 412 according to a given playback format.
- each object signal 412 may be rendered to specific output channels based on the object metadata information 425a.
- SAOC decoder 26 recovers the object / channel signal from SAOC channel signal 414 and parametric information.
- the SAOC decoder 26 may generate an output audio signal based on the reproduction layout information and the object metadata information 425b. That is, the SAOC decoder 26 generates a decoded object signal using the SAOC channel signal 414 and performs rendering that maps it to a target output signal.
- 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 415 and the HOA side information and decodes it.
- the HOA decoder 28 generates a sound scene by modeling a channel signal or an object signal with a separate equation. When a location in the space where the speaker is located is selected in the generated sound scene, rendering may be performed with the speaker channel signal.
- HOA Higher Order Ambisonics
- DRC dynamic range control
- the channel-based audio signal and the object-based audio signal processed by the rendering unit 20 are transferred to the mixer 30.
- the mixer 30 generates a mixer output signal by mixing the partial signals rendered in each sub unit of the rendering unit 20. If the partial signals are signals matched to the same position on the reproduction / virtual layout, they are added to each other. If signals matched to non-identical positions, they are mixed into output signals corresponding to separate positions.
- the mixer 30 may determine whether destructive interference occurs between the partial signals added to each other, and may further perform an additional process for preventing this.
- the mixer 30 adjusts delays of the channel-based waveform and the rendered object waveform and adds them in sample units. As such, 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 the multichannel and / or multiobject audio signal transmitted from the mixer 30.
- Such post processing may include dynamic range control (DRC), loudness normalization (LN) and peak limiter (PL).
- DRC dynamic range control
- LN loudness normalization
- PL peak limiter
- the output signal of the speaker renderer 100 may be transmitted to the loudspeaker of the multichannel audio system and output.
- the binaural renderer 200 generates a binaural downmix signal of the multichannel and / or multiobject audio signal.
- the binaural downmix signal is a two-channel audio signal such that each input channel / object signal is 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 on a time domain or a QMF domain.
- BRIR Binaural Room Impulse Response
- DRC dynamic range control
- LN volume normalization
- PL peak limit
- the output signal of the binaural renderer 200 may be transmitted to and output to a two-channel audio output device such as headphones or earphones.
- FIG. 5 is a block diagram illustrating an audio decoder according to a further embodiment of the present invention.
- the same reference numerals are used for the same configuration as the embodiment of FIG. 4, and overlapping descriptions thereof will be omitted.
- the audio decoder 1200-A may further include a rendering setting unit 21 that controls the rendering of the decoded audio signal.
- the rendering setting unit 21 receives the reproduction layout information 401 and / or the BRIR filter set information 402 and generates target format information 421 for rendering the audio signal using the reproduction layout information 401 and / or the BRIR filter set information 402.
- the rendering setting unit 21 may obtain the loudspeaker arrangement of the actual reproduction environment as the reproduction layout information 401, and generate the target format information 421 based on the reproduction layout information 401.
- the target format information 421 may indicate the position (channel) of the loudspeakers of the actual playback environment, or may indicate a superset based on a subset or a combination thereof.
- the rendering setting unit 21 may obtain the BRIR filter set information 402 from the binaural renderer 200 and generate the target format information 421 using the same.
- the target format information 421 indicates target positions (channels) supported by the BRIR filter set of the binaural renderer 200 (ie, capable of binaural rendering), or a subset thereof or a combination thereof. It can represent a superset based.
- the BRIR filter set information 402 may include different target positions from the reproduction layout information 401 indicating the arrangement of the physical loudspeakers, or may include a larger number of target positions. can do.
- the target position of the signal decoded by the core decoder 10 may be provided by the BRIR filter set information 402, but may not be provided by the reproduction layout information 401.
- the rendering setting unit 21 of the present invention generates the target format information 421 using the BRIR filter set information 402 obtained from the binaural renderer 200 when the final output audio signal is a binaural signal. can do.
- the rendering unit 20 performs rendering of the audio signal using the target format information 421 generated as described above, which may occur due to the two-step processing of the rendering and the binaural rendering based on the reproduction layout information 401. Sound degradation can be minimized.
- the rendering setting unit 21 may further obtain information about the type of the final output audio signal.
- the rendering setting unit 21 may generate the target format information 421 based on the reproduction layout information 401 and transmit it to the rendering unit 20.
- the rendering setting unit 21 may generate the target format information 421 based on the BRIR filter set information 402 and transmit it to the rendering unit 20.
- the rendering setting unit 21 may further obtain control information 403 indicating the audio system or the user's selection that the user is using, and use the control information 403 together.
- the target format information 421 may be generated.
- the generated target format information 421 is transferred to the rendering unit 20.
- Each sub unit of the rendering unit 20 may perform flexible rendering using the target format information 421 transmitted from the rendering setting unit 21. That is, the format converter 22 converts the decoded channel signal 411 into an output signal of the target channel based on the target format information 421.
- the object renderer 24 and the SAOC decoder 26 use the object signal 412 and the SAOC channel signal 414 as the output signal of the target channel using the target format information 421 and the object metadata information 425, respectively. Convert to In this case, the mixing matrix for rendering the object signal 412 may be updated based on the target format information 421, and the object renderer 24 may output the object signal 412 using the updated mixing matrix.
- the rendering may be performed by a conversion process of mapping the audio signal to at least one target location (ie, target channel) on the target format.
- the target format information 421 may also be transmitted to the mixer 30, and may be used in a process of mixing partial signals rendered in each sub unit of the rendering unit 20. If the partial signals are signals matched to the same position on the target format, they are added to each other, and if the signals matched to non-identical positions, they may be mixed into output signals corresponding to separate positions.
- the target format may be set according to various methods.
- the rendering setting unit 21 may set a target format having a higher spatial resolution than the obtained reproduction layout information 401 or the BRIR filter set information 402. That is, the rendering setting unit 21 obtains a first target position set, which is a set of original target positions indicated by the reproduction layout information 401 or the BRIR filter set information 402, and combines at least one original target position to add additional target positions. (extra) Create target locations.
- the additional target positions may include a position generated by interpolation between a plurality of original target positions, a position generated by extrapolation, and the like.
- the second target location set may be configured with the set of additional target locations generated as described above.
- the rendering setting unit 21 may generate a target format including the first target position set and the second target position set, and transmit the corresponding target format information 421 to the rendering unit 20.
- the rendering unit 20 may render the audio signal using the high resolution target format information 421 including the additional target position.
- the rendering unit 20 may obtain an output signal mapped to each target position of the target format information 421 through the rendering of the audio signal. If an output signal mapped to an additional target position of the second target position set is obtained, the rendering unit 20 may perform a downmix process of re-rendering the corresponding output signal to the original target position of the first target position set. have.
- the downmix process may be implemented through VBAP (Vector-Based Amplitude Panning) or Amplitude Panning.
- the rendering setting unit 21 may set a target format having a lower spatial resolution than the obtained BRIR filter set information 402. That is, the rendering setting unit 21 may obtain N (N ⁇ M) abbreviated target positions through a subset of M original target positions or a combination thereof, and generate a target format composed of the abbreviated target positions. .
- the rendering setting unit 21 may transmit the corresponding target format information 421 of low resolution to the rendering unit 20, and the rendering unit 20 may perform rendering of the audio signal using the rendering unit 20.
- the computation amount of the rendering unit 20 and the subsequent computation of the binaural renderer 200 may be reduced.
- the rendering setting unit 21 may set different target formats for each sub unit of the rendering unit 20.
- the target format provided to the format converter 22 and the target format provided to the object renderer 24 may be different from each other. If different target formats are provided for each sub unit, the amount of operation can be controlled or the sound quality can be improved for each sub unit.
- the rendering setting unit 21 may set the target format provided to the rendering unit 20 and the target format provided to the mixer 30 differently.
- the target format provided to the rendering unit 20 may have a higher spatial resolution than the target format provided to the mixer 30.
- the mixer 30 can be implemented to accompany the process of downmixing the input signal with high spatial resolution.
- the rendering setting unit 21 may set the target format based on the user's selection, the environment or setting of the device to be used.
- the rendering setting unit 21 may receive such information through the control information 403.
- the control information 403 may vary based on at least one of the calculation performance, the amount of power, and the user's selection that the device can provide.
- the rendering unit 20 is illustrated as performing rendering through different sub-units according to a signal to be rendered, but all or some sub-units may be implemented through an integrated renderer.
- the format converter 22 and the object renderer 24 may be implemented through one integrated renderer.
- At least a part of the output signal of the object renderer 24 may be input to the format converter 22.
- the output signal of the object renderer 24 input to the format converter 22 is used to solve the spatial mismatch that may occur between the two signals due to the difference in the performance of the flexible rendering of the object signal and the flexible rendering of the channel signal.
- the object renderer 24 when the object signal 412 and the channel signal 411 are simultaneously received as inputs, the object renderer 24 does not separately perform flexible rendering based on the target format information 421.
- the output signal can be passed to the format converter 22.
- the output signal of the object renderer 24 transmitted to the format converter 22 may be a signal corresponding to the channel format of the input channel signal 411.
- the format converter 22 may mix the output signal of the object renderer 24 with the channel signal 411 and perform flexible rendering based on the target format information 421 with respect to the mixed signal.
- the object renderer 24 may generate a virtual speaker corresponding to the location of the exception object, and perform rendering by using the real loudspeaker information and the virtual speaker information together.
- FIG. 6 illustrates an embodiment of the present invention for performing rendering on an exception object.
- solid lines denoted by 601 to 609 represent respective target positions supported by the target format, and an area surrounded by the target positions forms an output channel space that can be rendered.
- the dashed lines indicated by 611 to 613 are virtual positions that are not supported by the target format, and may indicate positions of the virtual speakers generated by the object renderer 24.
- the asterisk points denoted by S1 701 to S4 704 indicate the playback position in space in which the specific object S should be rendered at a specific time while moving along the path 700.
- the reproduction position in the space of the object may be obtained based on object metadata information 425.
- the object signal may be rendered based on whether or not the reproduction position of the object matches the target position of the target format.
- the object signal is converted into an output signal of the target channel corresponding to the target position 604. That is, the object signal may be rendered by 1: 1 mapping with the target channel.
- the object signal may be divided into output signals of a plurality of target positions adjacent to the reproduction position. For example, the object signal of S1 701 may be rendered as an output signal of adjacent target positions 601, 602 and 603.
- the object signal When an object signal is mapped to two or three target positions, the object signal may be rendered as an output signal of each target channel by a method such as VBAP (Vector-Based Amplitude Panning). Accordingly, the object signal may be rendered by 1: N mapping with the plurality of target channels.
- VBAP Vector-Based Amplitude Panning
- the object renderer 24 may project the object on the output channel space configured by the target format, and perform rendering from the projected position to the adjacent target position.
- the rendering method from the projected position to the target position may use the above-described rendering method of S1 701 or S2 702. That is, S3 703 and S4 704 may be projected to P3 and P4 on the output channel space, respectively, and the signals of the projected P3 and P4 may be rendered as output signals of adjacent target positions 604, 605 and 607. .
- the object renderer 24 may use the target position and the position of the virtual speaker together to perform rendering of the object. Can be.
- the object renderer 24 renders the object signal as an output signal including at least one virtual speaker signal. For example, when the reproduction position of the object is directly matched with the position of the virtual speaker 611, such as S4 704, the object signal is rendered as an output signal of the virtual speaker 611. However, if there is no virtual speaker that matches the playback position of the object, such as S3 703, the object signal may be rendered as an output signal of the adjacent virtual speaker 611 and the target channels 605 and 607.
- the object renderer 24 re-renders the rendered virtual speaker signal to the output signal of the target channel. That is, the signal of the virtual speaker 611 in which the object signal of S3 703 or S4 704 is rendered may be downmixed to an output signal of an adjacent target channel (eg, 605, 607).
- an adjacent target channel eg, 605, 607
- the target format may include additional target positions 621, 622, 623, and 624 generated by combining the original target positions.
- the binaural renderer 200 is a BRIR parameterization unit 300, high-speed convolution unit 230, late reverberation generation unit 240, QTDL processing unit 250, Mixer & combiner 260 may be included.
- the binaural renderer 200 performs binaural rendering on various types of input signals to generate 3D audio headphone signals (ie, 3D audio two channel signals).
- 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 a HOA signal.
- the binaural renderer 200 when the binaural renderer 200 includes a separate decoder, the input signal may be an encoded bitstream of the aforementioned audio signal.
- Binaural rendering converts the decoded input signal into a binaural downmix signal, so that the surround sound can be experienced while listening to the headphones.
- the binaural renderer 200 may perform binaural rendering using a Binaural Room Impulse Response (BRIR) filter.
- BRIR Binaural Room Impulse Response
- Generalizing binaural rendering using BRIR is M-to-O processing to obtain O output signals for multi-channel input signals with M channels.
- Binaural filtering can be regarded as filtering using filter coefficients corresponding to each input channel and output channel in this process.
- the original filter set H denotes transfer functions from the speaker position of each channel signal to the left and right ear positions.
- One of these transfer functions measured in a general listening room, that is, a room with reverberation, is called a Binaural Room Impulse Response (BRIR).
- the BRIR contains not only the direction information but also the information of the reproduction space.
- the HRTF and an artificial reverberator may be used to replace the BRIR.
- the binaural rendering using the BRIR is described, but the present invention is not limited thereto and may be applied to the binaural rendering using various types of FIR filters including HRIR and HRTF.
- the present invention is applicable not only to binaural rendering of an audio signal but also to various types of filtering operations of an input signal.
- the BRIR may have a length of 96K samples, and multi-channel binaural rendering is performed using M * O different filters, thus requiring a high throughput process.
- the audio signal processing apparatus may refer to the binaural renderer 200 or the binaural rendering unit 220 illustrated in FIG. 7. However, in the present invention, the audio signal processing apparatus may broadly refer to the audio decoder of FIG. 4 or 5 including a binaural renderer.
- an embodiment of a multichannel input signal may be mainly described, but unless otherwise stated, the channel, multichannel, and multichannel input signals respectively include an object, a multiobject, and a multiobject input signal. Can be used as a concept.
- the multichannel input signal may be used as a concept including a HOA decoded and rendered signal.
- the binaural renderer 200 may perform binaural rendering of the input signal on the QMF domain.
- the binaural renderer 200 may receive a multi-channel (N channels) signal of a QMF domain and perform binaural rendering on the multi-channel signal using a BRIR subband filter of the QMF domain.
- binaural rendering may be performed by dividing a channel signal or an object signal of a QMF domain into a plurality of subband signals, convolving each subband signal with a corresponding BRIR subband filter, and then summing them.
- the BRIR parameterization unit 300 converts and edits BRIR filter coefficients and generates various parameters for binaural rendering in the QMF domain.
- the BRIR parameterization unit 300 receives time domain BRIR filter coefficients for a multichannel or multiobject, and converts them into QMF domain BRIR filter coefficients.
- the QMF domain BRIR filter coefficients include a plurality of subband filter coefficients respectively corresponding to the plurality of frequency bands.
- the subband filter coefficients indicate each BRIR filter coefficient of the QMF transformed subband domain.
- Subband filter coefficients may also be referred to herein as BRIR subband filter coefficients.
- the BRIR parameterization unit 300 may edit the 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.
- the BRIR parameterization unit 300 may be included as one component of the binaural renderer 200 or may be provided as a separate device.
- the configuration including the high-speed convolution unit 230, the late reverberation generation unit 240, the QTDL processing unit 250, the mixer & combiner 260 except for the BRIR parameterization unit 300 is The binaural rendering unit 220 may be classified.
- the BRIR parameterization unit 300 may receive, as an input, a BRIR filter coefficient corresponding to at least one position of the virtual reproduction space.
- Each position of the virtual reproduction space may correspond to each speaker position of the multichannel system.
- each BRIR filter coefficient received by the BRIR parameterization unit 300 may be directly matched to each channel or each object of the input signal of the binaural renderer 200.
- each of the received BRIR filter coefficients may have a configuration independent of the input signal of the binaural renderer 200.
- the BRIR filter coefficients received by the BRIR parameterization unit 300 may not directly match the input signal of the binaural renderer 200, and the number of received BRIR filter coefficients may correspond to the channel of the input signal and / or Or it may be smaller or larger than the total number of objects.
- the BRIR parameterization unit 300 may additionally receive the control parameter information and generate the above-described binaural rendering parameter based on the input control parameter information.
- the control parameter information may include a complexity-quality control parameter and the like as described below, and may be used as a threshold for various parameterization processes of the BRIR parameterization unit 300. Based on this input value, the BRIR parameterization unit 300 generates a binaural rendering parameter and transmits it to the binaural rendering unit 220. If the input BRIR filter coefficients or control parameter information are changed, the BRIR parameterization unit 300 may recalculate the binaural rendering parameters and transmit them to the binaural rendering unit.
- the BRIR parameterization unit 300 converts and edits the BRIR filter coefficients corresponding to each channel or each object of the input signal of the binaural renderer 200 to perform the binaural rendering unit 220.
- the corresponding BRIR filter coefficients may be a matching BRIR or fallback BRIR for each channel or each object selected in the BRIR filter set.
- BRIR matching may be determined according to whether or not there is a BRIR filter coefficient targeting the position of each channel or each object in the virtual reproduction space. In this case, location information of each channel (or object) may be obtained from an input parameter signaling a channel layout.
- the corresponding BRIR filter coefficient may be a matching BRIR of the input signal. However, if there is no BRIR filter coefficient that targets the position of a particular channel or object, the BRIR parameterization unit 300 falls back the BRIR filter coefficient that targets the position most similar to that channel or object to the channel or object. It can be provided by BRIR.
- the corresponding BRIR filter coefficient may be selected. For example, a BRIR filter coefficient having the same altitude as the desired position and an azimuth deviation within +/ ⁇ 20 ° may be selected. If there is no corresponding BRIR filter coefficient, a BRIR filter coefficient having a minimum geometric distance from the desired position among the BRIR filter sets may be selected. That is, a BRIR filter coefficient may be selected that minimizes the geometric distance between the location of the BRIR and the desired location.
- the position of the BRIR represents the position of the speaker corresponding to the corresponding BRIR filter coefficients.
- the geometric distance between the two positions may be defined as the sum of the absolute value of the altitude deviation of the two positions and the absolute value of the azimuth deviation.
- the BRIR filter set may be matched to a desired position by interpolating the BRIR filter coefficients.
- the interpolated BRIR filter coefficients may be considered to be part of the BRIR filter set. That is, in this case, the BRIR filter coefficients may be always present at a desired position.
- the BRIR filter coefficients corresponding to each channel or each object of the input signal may be transmitted through separate vector information m conv .
- the vector information m conv indicates a BRIR filter coefficient corresponding to each channel or object of the input signal among the BRIR filter sets. For example, when the BRIR filter coefficients having position information matching the position information of a specific channel of the input signal exist in the BRIR filter set, the vector information m conv indicates that the BRIR filter coefficients correspond to the BRIR filter corresponding to the specific channel. Indicate by count.
- the parameterization unit 300 may determine the BRIR filter coefficients corresponding to each channel or object of the input audio signal in the entire BRIR filter set using the vector information m conv .
- the BRIR parameterization unit 300 may convert and edit all of the received BRIR filter coefficients and transmit the converted BRIR filter coefficients to the binaural rendering unit 220.
- the selection process of the BRIR filter coefficients (or the edited BRIR filter coefficients) corresponding to each channel or each object of the input signal may be performed by the binaural rendering unit 220.
- the binaural rendering parameter generated by the BRIR parameterization unit 300 is transmitted to the rendering unit 220 in a bitstream.
- the binaural rendering unit 220 may decode the received bitstream to obtain binaural rendering parameters.
- the transmitted binaural rendering parameters include various parameters necessary for processing in each subunit of the binaural rendering unit 220, and include transformed and edited BRIR filter coefficients or original BRIR filter coefficients. can do.
- 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 outputs a multi audio signal including a multichannel and / or multiobject signal. Receive.
- an input signal including a multichannel and / or multiobject signal is referred to as a multi audio signal.
- the binaural rendering unit 220 receives the multi-channel signal of the QMF domain, according to an embodiment.
- the input signal of the binaural rendering unit 220 may include a time domain multi-channel signal and a multi-channel signal. Object signals and the like.
- the input signal may be an encoded bitstream of the multi audio signal.
- the present invention will be described based on the case of performing BRIR rendering on the 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 be applied to an audio signal of a single channel or a single object rather than a multi-audio signal.
- the fast convolution unit 230 performs fast convolution between the input signal and the BRIR filter to process direct sound and early reflection on the input signal.
- 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 truncated depending on each subband frequency, and is generated by the BRIR parameterization unit 300. In this case, the length of each truncated subband filter coefficient is determined depending 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 subbands.
- fast convolution may be performed between the QMF domain subband signal and the truncated subband filters of the corresponding QMF domain for each frequency band.
- the truncated subband filter corresponding to each subband signal may be identified through the above-described vector information m conv .
- the late reverberation generator 240 generates a late reverberation signal with respect to the input signal.
- the late reverberation signal represents an output signal after the direct sound and the initial reflection sound generated by the fast convolution unit 230.
- the late reverberation generator 240 may process the input signal based on the reverberation time information determined from each subband filter coefficient transmitted from the BRIR parameterization unit 300.
- the late reverberation generator 240 may generate a mono or stereo downmix signal for the input audio signal and perform late reverberation processing on the generated downmix signal.
- the QMF domain trapped delay line (QTDL) processing unit 250 processes a signal of a high frequency band among the 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 300 and performs tap-delay line filtering in the QMF domain using the received parameter. .
- Parameters corresponding to each subband signal may be identified through the above-described vector information m conv .
- the binaural renderer 200 separates the input audio signal into a low frequency band signal and a high frequency band signal based on a predetermined constant or a predetermined frequency band, and the low frequency band signal is a high speed signal.
- the high frequency band signal may be processed by the QTDL processing unit 250, respectively.
- the fast convolution unit 230, the late reverberation generator 240, and the QTDL processing unit 250 output two QMF domain subband signals, respectively.
- the mixer & combiner 260 performs mixing by combining the output signal of the fast convolution unit 230, the output signal of the late reverberation generator 240, and the output signal of the QTDL processing unit 250. At this time, the combination of the output signal 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 produce a final binaural output audio signal in the time domain.
- FIG. 8 illustrates a filter generation method for binaural rendering according to an embodiment of the present invention.
- an FIR filter transformed into a plurality of subband filters may be used.
- the fast convolution unit of the binaural renderer may perform variable order filtering in the QMF domain by using a truncated subband filter having a different length according to each subband frequency.
- Fk represents a truncated subband filter used for fast convolution for processing direct and early reflections of the QMF subband k.
- Pk also represents a filter used to produce late reverberation of QMF subband k.
- the truncated subband filter Fk is a front filter cut from the original subband filter, and may also be referred to as a front subband filter.
- Pk is also a rear filter after truncation of the original subband filter, and may be referred to as a rear subband filter.
- the QMF domain has a total of K subbands. According to an embodiment, 64 subbands may be used.
- N represents the length (number of taps) of the original subband filter
- N Filter [k] represents the length of the front subband filter of subband k.
- the length N Filter [k] represents the number of taps in the down-sampled QMF domain.
- the filter order for each subband may include parameters extracted from the original BRIR filter, for example, reverberation time (RT) information for each subband filter, and energy decay. Curve) value, energy decay time information, and the like.
- the reverberation time may vary from frequency to frequency, due to the acoustic characteristics of the attenuation in the air for each frequency, the sound absorption of the wall and ceiling material is different. In general, a lower frequency signal has a longer reverberation time. Long reverberation time means that a lot of information remains behind the FIR filter. Therefore, it is preferable to cut the filter for a long time to properly transmit reverberation information.
- the length of each truncated subband filter Fk of the present invention is determined based at least in part on the characteristic information (eg, reverberation time information) extracted from the corresponding subband filter.
- the length of the truncated subband filter Fk may be determined based on additional information obtained by the audio signal processing apparatus, for example, the complexity of the decoder, the complexity level (profile), or the required quality information. .
- the complexity may be determined according to hardware resources of the audio signal processing apparatus or based on 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 an estimated value of the quality of the transmitted audio signal. For example, the higher the bit rate, the higher the quality.
- the length of each truncated subband filter may increase proportionally according to complexity and quality, or may vary at different rates for each band.
- the length of each truncated subband filter may be determined as a multiple of a power unit corresponding to the size unit, for example, a power of two, so as to obtain an additional gain by a fast processing such as an FFT.
- the determined length of the truncated subband filter is longer than the total length of the actual subband filter, the length of the truncated subband filter may be adjusted to the length of the actual subband filter.
- the BRIR parameterization unit of the present invention generates truncated subband filter coefficients corresponding to the lengths of the truncated subband filters determined in this way, and transfers them to the fast convolution unit.
- the fast convolution unit performs frequency domain variable order filtering (VOFF processing) on each subband signal of the multi-audio signal using the truncated subband filter coefficients. That is, for the first subband and the second subband, which are different frequency bands, the fast convolution unit generates the first subband binaural signal by applying the first truncated subband filter coefficients to the first subband signal.
- a second subband binaural signal is generated by applying the second truncated subband filter coefficients to the second subband signal.
- the first truncated subband filter coefficients and the second truncated subband filter coefficients may have different lengths independently from each other, and are obtained from circular filters (prototype filters) having the same time domain. That is, since one time-domain filter is converted into a plurality of QMF subband filters and the lengths of the filters corresponding to each subband are varied, each truncated subband filter is obtained from one circular filter.
- the plurality of QMF-transformed subband filters may be classified into a plurality of groups and used for different processing for each classified group.
- the plurality of subbands are classified into a first subband group Zone 1 of a low frequency and a second subband group Zone 2 of a high frequency based on a preset frequency band QMF band i. Can be.
- VOFF processing may be performed on the input subband signals of the first subband group
- QTDL processing which will be described later, may be performed on the input subband signals of the second subband group.
- the BRIR parameterization unit generates truncated subband filter (front subband filter) coefficients for each subband of the first subband group and transfers the coefficients to the fast convolution unit.
- the fast convolution unit performs VOFF processing on the subband signals of the first subband group by using the received front subband filter coefficients.
- late reverberation processing on the subband signals of the first subband group may be additionally performed by the late reverberation generator.
- the BRIR parameterization unit obtains at least one parameter from each subband filter coefficient of the second subband group and transfers it to the QTDL processing unit.
- the QTDL processing unit performs tap-delay line filtering on each subband signal of the second subband group using the obtained parameter as described below.
- the predetermined frequency (QMF band i) for distinguishing the first subband group and the second subband group may be determined based on a predetermined constant value, and the bit of the transmitted audio input signal may be determined. It may be determined according to the stream 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.
- the plurality of subbands may be classified into three subband groups based on the first frequency band QMF band i and the second frequency band QMF band j as shown in FIG. 8. It may be. That is, the plurality of subbands may include a first subband group Zone 1 which is a low frequency zone smaller than or equal to the first frequency band, and a second subband that is an intermediate frequency zone greater than or equal to the second frequency band and larger than the first frequency band. Band group Zone 2 and a third subband group Zone 3 that is a higher frequency region larger than the second frequency band.
- the first subband group includes a total of 32 subbands having indices of 0 to 31
- the second subband group may include a total of 16 subbands having indices of 32 to 47
- the third subband group may include subbands having indices of the remaining 48 to 63.
- the subband index has a lower value as the subband frequency is lower.
- binaural rendering may be performed only on the subband signals of the first subband group and the second subband group. That is, VOFF processing and late reverberation processing may be performed on the subband signals of the first subband group as described above, and QTDL processing may be performed on the subband signals of the second subband group. In addition, binaural rendering may not be performed on the subband signals of the third subband group.
- the first frequency band (QMF band i) is set to a subband of index Kconv-1
- the second frequency band (QMF band j) is set to a subband of index Kproc-1.
- the values of the information Kproc of the maximum frequency band and the information Kconv of the frequency band performing the convolution may vary depending on the sampling frequency of the original BRIR input, the sampling frequency of the input audio signal, and the like.
- the length of the rear subband filter Pk as well as the front subband filter Fk may be determined based on parameters extracted from the original subband filter. That is, the lengths of the front subband filter and the rear subband filter of each subband are determined based at least in part on the characteristic information extracted from the corresponding subband filter. For example, the length of the front subband filter may be determined based on the first reverberation time information of the corresponding subband filter, and the length of the rear subband filter may be determined based on the second reverberation time information.
- the front subband filter is a filter of the front part cut based on the first reverberation time information in the original subband filter
- the rear subband filter is a section after the front subband filter between the first reverberation time and the second reverberation time.
- the filter may be a later part corresponding to the interval of.
- 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.
- the mixing time for each subband may be estimated to perform high-speed convolution through VOFF processing before the mixing time, and post-reverberation processing may be performed after the mixing time to reflect the common characteristics of each channel.
- the mixing time may cause an error due to bias from a perceptual perspective. Therefore, rather than estimating the correct mixing time and dividing it into a VOFF processing part and a late reverberation processing part on the basis of the boundary, it is excellent in terms of quality to perform fast convolution with the length of the VOFF processing part as long as possible. Accordingly, the length of the VOFF processing 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.
- the model of reducing the filter of the subband to a lower order is possible.
- a typical method is FIR filter modeling using frequency sampling, and it is possible to design a filter that is minimized in terms of least squares.
- the QTDL processing unit 250 uses the one-tap-delay line filter to multi-channel input signals X0, X1,... , Sub-band filtering is performed on X_M-1.
- the multi-channel input signal is received as a subband signal of the QMF domain.
- the one-tap-delay line filter may perform processing for each QMF subband.
- the one-tap-delay line filter performs only one tap convolution on each channel signal.
- the tap used may be determined based on a parameter directly extracted from a BRIR subband filter coefficient corresponding to the corresponding subband signal.
- the parameter includes delay information for the tap to be used in the one-tap-delay line filter and corresponding gain information.
- L_0, L_1,... , L_M-1 represent delays for the BRIR from the M channels to the left ear, respectively, and R_0, R_1,... , R_M-1 represents the delay for the BRIR from the M channel to the right ear, respectively.
- the delay information indicates position information of the maximum peak among the corresponding BRIR subband filter coefficients in order of absolute value, real value, or imaginary value.
- G_L_0, G_L_1,... , G_L_M-1 represent gains corresponding to the delay information of the left channel
- G_R_0, G_R_1,... , G_R_M-1 represents a gain corresponding to each delay information of the right channel.
- Each gain information may be determined based on the total power of the corresponding BRIR subband filter coefficients, the magnitude of the peak corresponding to the corresponding delay information, and the like.
- the corresponding peak value itself in the subband filter coefficients may be used as the gain information
- the weight value of the corresponding peak after energy compensation for the entire subband filter coefficients may be used.
- the gain information is obtained by using both real weight and imaginary weight for the corresponding peak, and thus has a complex value.
- the QTDL processing may be performed only on the input signal of the high frequency band classified based on the predetermined constant or the preset frequency band as described above.
- SBR Spectral Band Replication
- the high frequency band may correspond to the SBR band.
- SBR Spectral Band Replication
- SBR Spectral Band Replication
- the high frequency band is generated by using information of the low frequency band that is encoded and transmitted and additional information of the high frequency band signal transmitted by the encoder.
- high frequency components generated using SBR may cause distortion due to inaccurate harmonics.
- the SBR band is a high frequency band, and as described above, the reverberation time of the frequency band is very short. That is, the BRIR subband filter of the SBR band has less valid information and has a fast attenuation rate. Therefore, the BRIR rendering for the high frequency band that corresponds to the SBR band may be very effective in terms of the amount of computation compared to the quality of sound quality rather than performing the convolution.
- the plurality of channel signals filtered by the one-tap-delay line filter are summed into two channel left and right output signals Y_L and Y_R for each subband.
- the parameters used in each one-tap-delay line filter of the QTDL processing unit 250 may be stored in a memory during initialization of binaural rendering, and QTDL processing may be performed without additional operations for parameter extraction. have.
- the BRIR parameterization unit 300 may include a VOFF parameterization unit 320, a late reverberation parameterization unit 360, and a QTDL parameterization unit 380.
- the BRIR parameterization unit 300 receives the BRIR filter set in the time domain as an input, and each sub unit of the BRIR parameterization unit 300 generates various parameters for binaural rendering using the received BRIR filter set.
- the BRIR parameterization unit 300 may additionally receive a control parameter and generate a parameter based on the input control parameter.
- the VOFF parameterization unit 320 generates truncated subband filter coefficients necessary for frequency domain variable order filtering (VOFF) and corresponding auxiliary parameters. For example, the VOFF parameterization unit 320 calculates frequency band reverberation time information, filter order information, etc. for generating the truncated subband filter coefficients, and performs a fast Fourier transform in block units on the truncated subband filter coefficients. Determine the size of the block to perform. Some parameters generated by the VOFF parameterization unit 320 may be transferred to the late reverberation parameterization unit 360 and the QTDL parameterization unit 380.
- VOFF frequency domain variable order filtering
- the transmitted parameter is not limited to the final output value of the VOFF parameterization unit 320, and may include parameters generated in the middle according to the processing of the VOFF parameterization unit 320, for example, a truncated BRIR filter coefficient in the time domain. have.
- the late reverberation parameterization unit 360 generates a parameter necessary for generating late reverberation.
- the late reverberation parameterization unit 360 may generate downmix subband filter coefficients, IC values, and the like.
- the QTDL parameterization unit 380 generates a parameter for QTDL processing. More specifically, the QTDL parameterization unit 380 receives the subband filter coefficients from the VOFF parameterization unit 320 and generates delay information and gain information in each subband using the subband filter coefficients.
- the QTDL parameterization unit 380 may receive the information (Kproc) of the maximum frequency band to perform binaural rendering and the information (Kconv) of the frequency band to perform convolution as control parameters, and receive Kproc and Kconv. Delay information and gain information can be generated for each frequency band of the subband group serving as a boundary. According to an embodiment, the QTDL parameterization unit 380 may be provided in a configuration included in the VOFF parameterization unit 320.
- Parameters generated by the VOFF parameterization unit 320, the late reverberation parameterization unit 360, and the QTDL parameterization unit 380 are transmitted to a binaural rendering unit (not shown).
- the late reverberation parameterization unit 360 and the QTDL parameterization unit 380 may determine whether to generate parameters according to whether late reverberation processing and QTDL processing are performed in the binaural rendering unit, respectively. If at least one of the late reverberation processing and the QTDL processing is not performed in the binaural rendering unit, the corresponding late reverberation parameterization unit 360 and the QTDL parameterization unit 380 do not generate the parameter or generate the generated parameter. It may not be sent to the binaural rendering unit.
- the VOFF parameterization unit 320 may include a propagation time calculator 322, a QMF converter 324, and a VOFF parameter generator 330.
- the VOFF parameterization unit 320 performs a process of generating truncated subband filter coefficients for VOFF processing using the received time domain BRIR filter coefficients.
- the propagation time calculator 322 calculates propagation time information of the time domain BRIR filter coefficients and cuts the time domain BRIR filter coefficients based on the calculated propagation time information.
- the propagation time information represents the time from the initial sample of the BRIR filter coefficients to the direct sound.
- the propagation time calculator 322 may cut a portion corresponding to the calculated propagation time from the time domain BRIR filter coefficients and remove the same.
- the propagation time may be estimated based on the first point information at which an energy value larger than a threshold value proportional to the maximum peak value of the BRIR filter coefficients appears.
- the propagation time may be different for each channel.
- the propagation time truncation length of all channels must be the same.
- the probability of error occurrence in an individual channel can be reduced.
- the frame energy E (k) for the frame unit index k may be defined first.
- the frame energy E (k) in the k-th frame may be calculated by the following equation.
- N BRIR represents the total number of filters in the BRIR filter set
- N hop represents a preset hop size
- L frm represents a frame size. That is, the frame energy E (k) may be calculated as an average value of the frame energy of each channel for the same time domain.
- the propagation time pt may be calculated by the following equation.
- the propagation time calculation unit 322 shifts by a predetermined hop unit, measures the frame energy, and identifies the first frame in which the frame energy is larger than the preset threshold. At this time, the propagation time may be determined as an intermediate point of the identified first frame.
- the threshold value is illustrated as being set to a value 60 dB lower than the maximum frame energy, but the present invention is not limited thereto, and the threshold value is a value proportional to the maximum frame energy or a predetermined difference from the maximum frame energy. It can be set to a value having.
- the hop size N hop and the frame size L frm may vary based on whether the input BRIR filter coefficients are Head Related Impulse Response (HRIR) filter coefficients.
- the information flag_HRIR indicating whether the input BRIR filter coefficients are HRIR filter coefficients may be received from the outside, or may be estimated using the length of the time domain BRIR filter coefficients.
- the boundary between the early reflection part and the late reverberation part is known as 80ms.
- the propagation time calculator 322 may cut the time domain BRIR filter coefficients based on the calculated propagation time information, and transfer the truncated BRIR filter coefficients to the QMF converter 324.
- the truncated BRIR filter coefficients indicate the filter coefficients remaining after cutting and removing a portion corresponding to the propagation time from the original BRIR filter coefficients.
- the propagation time calculator 322 cuts the time-domain BRIR filter coefficients for each input channel and each output left / right channel, and transmits them to the QMF converter 324.
- the QMF conversion unit 324 performs conversion between the time domain and the QMF domain of the input BRIR filter coefficients. That is, the QMF converter 324 receives the truncated BRIR filter coefficients in the time domain and converts them into a plurality of subband filter coefficients respectively corresponding to the plurality of frequency bands. The converted subband filter coefficients are transferred to the VOFF parameter generator 330, and the VOFF parameter generator 330 generates truncated subband filter coefficients using the received subband filter coefficients. If QMF domain BRIR filter coefficients other than the time domain BRIR filter coefficients are received as inputs to the VOFF parameterization unit 320, the input QMF domain BRIR filter coefficients may bypass the QMF converter 324. . According to another exemplary embodiment, when the input filter coefficients are QMF domain BRIR filter coefficients, the QMF converter 324 may be omitted from the VOFF parameterization unit 320.
- the VOFF parameter generator 330 may include a reverberation time calculator 332, a filter order determiner 334, and a VOFF filter coefficient generator 336.
- the VOFF parameter generator 330 may receive the subband filter coefficients of the QMF domain from the QMF converter 324 of FIG. 11.
- control parameters such as maximum frequency band information Kproc for performing binaural rendering, frequency band information Kconv for performing convolution, and preset maximum FFT size information may be input to the VOFF parameter generator 330. Can be.
- the reverberation time calculator 332 obtains reverberation time information by using the received subband filter coefficients.
- the obtained reverberation time information is transmitted to the filter order determiner 334 and used to determine the filter order of the corresponding subband.
- the reverberation time information may have a bias or a deviation depending on the measurement environment, a uniform value may be used by using a correlation with other channels.
- the reverberation time calculator 332 generates average reverberation time information of each subband, and transmits the average reverberation time information to the filter order determiner 334.
- Average reverberation time information RT k of subband k when reverberation time information of subband filter coefficients for input channel index m, output left / right channel index i, subband index k is RT (k, m, i) Can be calculated through the following equation.
- N BRIR is the total number of filters in the BRIR filter set.
- the reverberation time calculator 332 extracts reverberation time information RT (k, m, i) from each subband filter coefficient corresponding to the multichannel input, and extracts reverberation time information RT for each channel extracted for the same subband. Obtain an average value of (k, m, i) (ie, average reverberation time information RT k ). The obtained average reverberation time information RT k is transmitted to the filter order determiner 334, and the filter order determiner 334 may determine one filter order applied to the corresponding subband.
- the obtained average reverberation time information may include RT20, and other reverberation time information, for example, RT30, RT60, may be obtained according to an embodiment.
- the reverberation time calculating unit 332 determines the filter order as the representative reverberation time information of the corresponding subband as the maximum and / or minimum value of the reverberation time information for each channel extracted for the same subband. May be passed to the unit 334.
- the filter order determiner 334 determines the filter order of the corresponding subband based on the obtained reverberation time information.
- the reverberation time information obtained by the filter order determiner 334 may be average reverberation time information of the corresponding subband, and may be representative of the maximum and / or minimum values of the reverberation time information for each channel, according to an exemplary embodiment. It may also be reverberation time information.
- the filter order is used to determine the length of truncated subband filter coefficients for binaural rendering of the corresponding subband.
- the filter order information N Filter [k] of the corresponding subband may be obtained through the following equation.
- the filter order information may be determined as a power of 2, which is an approximation of an approximated integer value of an integer unit of a log scale of average reverberation time information of a corresponding subband.
- the filter order information may be determined as a power of 2 rounded up, rounded up, or rounded down to average log reverberation time information of the subband. If the original length of the corresponding subband filter coefficients, that is, the length up to the last time slot n end is smaller than the value determined in Equation 7, the filter order information is set to the original length value n end of the subband filter coefficients. Can be replaced. That is, the filter order information may be determined as a smaller value between the reference truncation length determined by Equation 7 and the original length of the subband filter coefficients.
- the filter order determiner 334 may obtain filter order information using a polynomial curve fitting method. To this end, the filter order determiner 334 may obtain at least one coefficient for curve fitting of average reverberation time information. For example, the filter order determiner 334 may curve-fit the average reverberation time information for each subband to a logarithmic linear equation, and obtain the slope value a and the intercept value b of the linear equation.
- Curve-fit filter order information N ' Filter [k] in subband k may be obtained through the following equation using the obtained coefficient.
- the curve-fitted filter order information may be determined as a power of 2, which is an approximation of an integer unit of the polynomial curve-fitted value of the average reverberation time information of the corresponding subband.
- the curve-fitted filter order information may be determined as a power of 2 rounded up, rounded up, or rounded down to the polynomial curve-fitted value of the average reverberation time information of the corresponding subband. .
- the filter order information is the original length value n end of the subband filter coefficient. Can be replaced. That is, the filter order information may be determined as a smaller value between the reference truncation length determined by Equation 8 and the original length of the subband filter coefficients.
- the filter order information using any one of Equations 7 and 8 above. Can be obtained.
- the filter order information may be determined as a value that is not curve-fitted according to Equation (7). That is, the filter order information may be determined based on the average reverberation time information of the corresponding subband without performing curve fitting. This is because HRIR is not affected by room, so the tendency to energy decay is not apparent.
- Filter order information of each subband determined according to the above-described embodiment is transferred to the VOFF filter coefficient generator 336.
- the VOFF filter coefficient generator 336 generates the truncated subband filter coefficients based on the obtained filter order information.
- the truncated subband filter coefficients may include at least one fast Fourier transform (FFT) performed on a predetermined block basis for block-wise fast convolution. It may consist of FFT filter coefficients.
- FFT fast Fourier transform
- the VOFF filter coefficient generator 336 may generate the FFT filter coefficients for block-wise high-speed convolution as described below with reference to FIG. 14.
- the QTDL parameterization unit 380 may include a peak search unit 382 and a gain generator 384.
- the QTDL parameterization unit 380 may receive the subband filter coefficients of the QMF domain from the VOFF parameterization unit 320.
- the QTDL parameterization unit 380 may receive the maximum frequency band information Kproc for binaural rendering and the frequency band information Kconv for convolution as control parameters, and receive Kproc and Kconv. Delay information and gain information can be generated for each frequency band of a subband group (second subband group) serving as a boundary.
- the BRIR subband filter coefficients for the input channel index m, the output left and right channel index i, the subband index k, and the time slot index n of the QMF domain are determined.
- Delay information And gain information Can be obtained as follows.
- n end represents the last time slot of the corresponding subband filter coefficients.
- the delay information may indicate information of a time slot in which the size of the corresponding BRIR subband filter coefficient is maximum, which indicates position information of the maximum peak of the corresponding BRIR subband filter coefficient.
- the gain information may be determined by multiplying the total power value of the corresponding BRIR subband filter coefficients by the sign of the BRIR subband filter coefficients at the maximum peak position.
- the peak search unit 382 obtains the position of the maximum peak in each subband filter coefficient of the second subband group, that is, delay information, based on Equation (7).
- the gain generator 384 obtains gain information for each subband filter coefficient based on Equation (8). Equations 7 and 8 show an example of an equation for obtaining delay information and gain information, but a specific form of the equation for calculating each information may be variously modified.
- fast convolution of a predetermined block unit may be performed for optimal binaural rendering in terms of efficiency and performance.
- High-speed convolution based on FFT reduces the amount of computation as the FFT size increases, but increases the overall processing delay and increases the memory usage. If a high-speed convolution of a BRIR with a length of 1 second with an FFT size that is twice the length is effective, it is efficient in terms of throughput but a delay of 1 second is generated and corresponding buffer and processing memory. You will need An audio signal processing method having a long delay time is not suitable for an application for real time data processing. Since the minimum unit capable of performing decoding in the audio signal processing apparatus is a frame, it is preferable that binaural rendering also performs fast convolution of a block unit in a size corresponding to the frame unit.
- the circular FIR filter is converted into K subband filters, and Fk and Pk are truncated subband filters (front subband filters) and rear subbands of subband k, respectively. Indicates a filter.
- Each subband Band 0 to Band K-1 may represent a subband in the frequency domain, that is, a QMF subband.
- the QMF domain may use 64 subbands in total, but the present invention is not limited thereto.
- N represents the length (number of taps) of the original subband filter
- N Filter [k] represents the length of the front subband filter of subband k.
- the plurality of subbands of the QMF domain includes a first subband group Zone 1 of a low frequency and a second subband group of a high frequency based on a preset frequency band QMF band i. Can be classified as (Zone 2).
- the plurality of subbands may be divided into three subband groups, that is, the first subband group Zone 1 and the second, based on a preset first frequency band QMF band i and a second frequency band QMF band j.
- the subband group Zone 2 and the third subband group Zone 3 may be classified.
- VOFF processing using fast convolution on a block basis may be performed on the input subband signals of the first subband group, and QTDL processing may be performed on the input subband signals of the second subband group.
- the subband signals of the third subband group may not be rendered.
- late reverberation processing may be additionally performed on the input subband signals of the first subband group.
- the VOFF filter coefficient generator 336 of the present invention may generate FFT filter coefficients by performing fast Fourier transform on the truncated subband filter coefficients in predetermined block units in the corresponding subband.
- the length N FFT [k] of the preset block in each subband k is determined based on the preset maximum FFT size 2L. More specifically, the length N FFT [k] of the predetermined block in the subband k may be represented by the following equation.
- 2L is a preset maximum FFT size and N Filter [k] is filter order information of subband k.
- the length NFFT [k] of the preset block is twice the length of the reference filter of the truncated subband filter coefficient ( ) And a smaller value among the preset maximum FFT size 2L.
- the reference filter length represents either a true value or an approximation of a power of 2 of the filter order N Filter [k] (that is, the length of truncated subband filter coefficients) in the corresponding subband k.
- the filter order N Filter [k] is used as the reference filter length in subband k, and if it is not a power of 2 (eg, n end )
- the rounded, rounded, or rounded down power of the filter order N Filter [k] is used as the reference filter length.
- the length N FFT [k] and the reference filter length of a predetermined block Are all powers of two.
- the length of the predetermined block of the corresponding subband N FFT [0] , N FFT [1] is determined by the maximum FFT size (2L), respectively.
- the length N FFT [5] of the predetermined block of the corresponding subband is determined. Is twice the length of the filter Is determined.
- the length N FFT [k] of the block for the fast Fourier transform is It may be determined based on a comparison result between the double value and the preset maximum FFT size (2L).
- the VOFF filter coefficient generator 336 performs fast Fourier transform on the subband filter coefficients truncated in the determined block unit. More specifically, the VOFF filter coefficient generator 336 divides the truncated subband filter coefficients in units of half (N FFT [k] / 2) units of the predetermined block. An area of a dotted line boundary of the VOFF processing part illustrated in FIG. 14 represents subband filter coefficients divided into half units of a preset block. Next, the BRIR parameterization unit generates temporary filter coefficients of a predetermined block unit N FFT [k] by using each divided filter coefficient.
- the first half of the temporary filter coefficients is composed of the divided filter coefficients, and the second half is composed of zero-padded values.
- a temporary filter coefficient of a predetermined block length N FFT [k] is generated using a filter coefficient of half length (N FFT [k] / 2) of the preset block.
- the BRIR parameterization unit performs fast Fourier transform on the generated temporary filter coefficients to generate FFT filter coefficients.
- the FFT filter coefficients generated as described above may be used for fast convolution of a predetermined block unit for the input audio signal.
- the VOFF filter coefficient generator 336 generates FFT filter coefficients by performing fast Fourier transform on the truncated subband filter coefficients in blocks of lengths independently determined for each subband. can do. Accordingly, fast convolution using different numbers of blocks for each subband may be performed. At this time, the number N blk [k] of the blocks in the subband k may satisfy the following equation.
- N blk (k) is a natural number.
- the number N blk [k] of the blocks in the subband k may be determined as a value obtained by dividing a value twice the length of the reference filter in the corresponding subband by the length N FFT [k] of the predetermined block.
- the above-described process of generating FFT filter coefficients in units of blocks may be limitedly performed on the front subband filters Fk of the first subband group.
- the late reverberation processing may be performed by the late reverberation generating unit for the subband signals of the first subband group according to the embodiment.
- late reverberation processing on the input audio signal may be performed based on whether the length of the circular BRIR filter coefficient exceeds a preset value. As described above, whether the length of the circular BRIR filter coefficients exceeds a preset value may be indicated through a flag indicating that (eg, flag_BRIR).
- VOFF processing may be performed on each subband signal of the first subband group.
- the energy compensation may be performed by dividing the filter power up to the cutting point and multiplying the total filter power of the corresponding subband filter coefficients by the filter coefficient before the cutting point based on the filter order information N Filter [k]. .
- the total filter power may be defined as the sum of the powers for the filter coefficients from the initial sample to the last sample n end of the corresponding subband filter coefficients.
- the filter order of each subband filter coefficient may be set differently for each channel.
- the filter order for front channels where the input signal contains more energy may be set higher than the filter order for rear channels containing relatively less energy.
- the resolution reflected after the binaural rendering of the front channel may be increased, and the rendering may be performed on the rear channel with a low calculation amount.
- the division of the front channel and the rear channel is not limited to a channel name assigned to each channel of the multi-channel input signal, and each channel may be classified into a front channel and a rear channel based on a predetermined spatial reference.
- each channel of the multi-channel may be classified into three or more channel groups based on a predetermined spatial criterion, and different filter orders may be used for each channel group.
- different weighted values may be used based on position information of the corresponding channel in the virtual reproduction space.
- the present invention can be applied to a multimedia signal processing apparatus including various types of audio signal processing apparatuses and video signal processing apparatuses.
- the present invention can be applied to a parameterization device for generating a parameter used in the processing of the audio signal processing device and the video signal device.
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Abstract
Description
Claims (10)
- 멀티채널 신호를 포함하는 입력 오디오 신호를 수신하는 단계;상기 입력 오디오 신호의 필터링을 위한 절단된 서브밴드 필터 계수들을 수신하는 단계, 상기 절단된 서브밴드 필터 계수는 상기 입력 오디오 신호의 바이노럴 필터링을 위한 BRIR(Binaural Room Impulse Response) 필터 계수로부터 획득된 서브밴드 필터 계수의 적어도 일 부분이며, 상기 절단된 서브밴드 필터 계수의 길이는 해당 서브밴드 필터 계수에서 추출된 잔향 시간 정보를 적어도 부분적으로 이용하여 획득된 필터 차수 정보에 기초하여 결정됨;상기 입력 오디오 신호의 각 채널에 대응하는 상기 BRIR 필터 계수를 지시하는 벡터 정보를 획득하는 단계; 및상기 벡터 정보에 기초하여, 상기 멀티채널 신호의 각 서브밴드 신호를 해당 채널 및 서브밴드에 대응하는 상기 절단된 서브밴드 필터 계수를 이용하여 필터링 하는 단계;를 포함하는 것을 특징으로 하는 오디오 신호 처리 방법.
- 제1 항에 있어서,상기 벡터 정보는, 상기 입력 오디오 신호의 특정 채널의 위치 정보와 매칭되는 위치 정보를 갖는 BRIR 필터 계수가 BRIR 필터 셋에 존재할 경우, 해당 BRIR 필터 계수를 상기 특정 채널에 대응하는 BRIR 필터 계수로 지시하는 것을 특징으로 하는 오디오 신호 처리 방법.
- 제1 항에 있어서,상기 벡터 정보는, 상기 입력 오디오 신호의 특정 채널의 위치 정보와 매칭되는 위치 정보를 갖는 BRIR 필터 계수가 BRIR 필터 셋에 존재하지 않을 경우, 상기 특정 채널의 위치 정보와 최소의 기하학적 거리를 갖는 BRIR 필터 계수를 상기 특정 채널에 대응하는 BRIR 필터 계수로 지시하는 것을 특징으로 하는 오디오 신호 처리 방법.
- 제3 항에 있어서,상기 기하학적 거리는 두 위치간의 고도 편차의 절대값과 방위각 편차의 절대값을 합산한 값인 것을 특징으로 하는 오디오 신호 처리 방법.
- 제1 항에 있어서,적어도 하나의 상기 절단된 서브밴드 필터 계수의 길이는 다른 서브밴드의 절단된 서브밴드 필터 계수의 길이와 다른 것을 특징으로 하는 오디오 신호 처리 방법.
- 입력 오디오 신호에 대한 바이노럴 렌더링을 수행하기 위한 오디오 신호 처리 장치로서,상기 입력 오디오 신호의 필터를 생성하기 위한 파라메터화부; 및멀티채널 신호를 포함하는 입력 오디오 신호를 수신하고, 상기 파라메터화부에서 생성된 파라메터를 이용하여 상기 입력 오디오 신호를 필터링하는 바이노럴 렌더링 유닛을 포함하되,상기 바이노럴 렌더링 유닛은,상기 파라메터화부로부터 상기 입력 오디오 신호의 필터링을 위한 절단된 서브밴드 필터 계수들을 수신하되, 상기 절단된 서브밴드 필터 계수는 상기 입력 오디오 신호의 바이노럴 필터링을 위한 BRIR(Binaural Room Impulse Response) 필터 계수로부터 획득된 서브밴드 필터 계수의 적어도 일 부분이며, 상기 절단된 서브밴드 필터 계수의 길이는 해당 서브밴드 필터 계수에서 추출된 잔향 시간 정보를 적어도 부분적으로 이용하여 획득된 필터 차수 정보에 기초하여 결정되고,상기 입력 오디오 신호의 각 채널에 대응하는 상기 BRIR 필터 계수를 지시하는 벡터 정보를 획득하고,상기 벡터 정보에 기초하여, 상기 멀티채널 신호의 각 서브밴드 신호를 해당 채널 및 서브밴드에 대응하는 상기 절단된 서브밴드 필터 계수를 이용하여 필터링 하는,오디오 신호 처리 장치.
- 제6 항에 있어서,상기 벡터 정보는, 상기 입력 오디오 신호의 특정 채널의 위치 정보와 매칭되는 위치 정보를 갖는 BRIR 필터 계수가 BRIR 필터 셋에 존재할 경우, 해당 BRIR 필터 계수를 상기 특정 채널에 대응하는 BRIR 필터 계수로 지시하는 것을 특징으로 하는 오디오 신호 처리 장치.
- 제6 항에 있어서,상기 벡터 정보는, 상기 입력 오디오 신호의 특정 채널의 위치 정보와 매칭되는 위치 정보를 갖는 BRIR 필터 계수가 BRIR 필터 셋에 존재하지 않을 경우, 상기 특정 채널의 위치 정보와 최소의 기하학적 거리를 갖는 BRIR 필터 계수를 상기 특정 채널에 대응하는 BRIR 필터 계수로 지시하는 것을 특징으로 하는 오디오 신호 처리 장치.
- 제8 항에 있어서,상기 기하학적 거리는 두 위치간의 고도 편차의 절대값과 방위각 편차의 절대값을 합산한 값인 것을 특징으로 하는 오디오 신호 처리 장치.
- 제6 항에 있어서,적어도 하나의 상기 절단된 서브밴드 필터 계수의 길이는 다른 서브밴드의 절단된 서브밴드 필터 계수의 길이와 다른 것을 특징으로 하는 오디오 신호 처리 장치.
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KR20160124139A (ko) | 2016-10-26 |
US10321254B2 (en) | 2019-06-11 |
CN108600935A (zh) | 2018-09-28 |
EP3122073B1 (en) | 2023-12-20 |
EP3122073A4 (en) | 2017-10-18 |
CN108600935B (zh) | 2020-11-03 |
CN106105269A (zh) | 2016-11-09 |
US20180359587A1 (en) | 2018-12-13 |
US20210195356A1 (en) | 2021-06-24 |
US10771910B2 (en) | 2020-09-08 |
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